Attachment 96017 Details for Bug 76328 – Output of pulseaudio -vvvvv while recording in audacity.

Output of pulseaudio -vvvvv while recording in audacity.

pulseaudio_log (text/plain), 735.21 KB, created by Dhaval on 2014-03-18 18:06:56 UTC

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Creator: Dhaval

Created: 2014-03-18 18:06:56 UTC

Size: 735.21 KB

patch

obsolete

>I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
>I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
>D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
>D: [pulseaudio] core-util.c: RealtimeKit worked.
>I: [pulseaudio] core-util.c: Successfully gained nice level -11.
>I: [pulseaudio] main.c: This is PulseAudio 4.0
>D: [pulseaudio] main.c: Compilation host: x86_64-redhat-linux-gnu
>D: [pulseaudio] main.c: Compilation CFLAGS: -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches  -m64 -mtune=generic -W -Wextra -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -fno-common -fdiagnostics-show-option
>D: [pulseaudio] main.c: Running on host: Linux x86_64 3.13.5-200.fc20.x86_64 #1 SMP Mon Feb 24 16:51:35 UTC 2014
>D: [pulseaudio] main.c: Found 2 CPUs.
>I: [pulseaudio] main.c: Page size is 4096 bytes
>D: [pulseaudio] main.c: Compiled with Valgrind support: no
>D: [pulseaudio] main.c: Running in valgrind mode: no
>D: [pulseaudio] main.c: Running in VM: no
>D: [pulseaudio] main.c: Optimized build: yes
>D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled.
>I: [pulseaudio] main.c: Machine ID is fbb0c4896a9c0f680475c0a07edd2c5a.
>I: [pulseaudio] main.c: Session ID is 1.
>I: [pulseaudio] main.c: Using runtime directory /run/user/1000/pulse.
>I: [pulseaudio] main.c: Using state directory /home/dhaval/.pulse.
>I: [pulseaudio] main.c: Using modules directory /usr/lib64/pulse-4.0/modules.
>I: [pulseaudio] main.c: Running in system mode: no
>I: [pulseaudio] main.c: Fresh high-resolution timers available! Bon appetit!
>D: [pulseaudio] memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472
>I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 MMXEXT 3DNOW 3DNOWEXT 
>I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions.
>I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers.
>I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions.
>I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers.
>I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions.
>I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions.
>D: [pulseaudio] database-tdb.c: Opened TDB database '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-device-volumes.tdb'
>I: [pulseaudio] module-device-restore.c: Successfully opened database file '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-device-volumes'.
>I: [pulseaudio] module.c: Loaded "module-device-restore" (index: #0; argument: "").
>D: [pulseaudio] database-tdb.c: Opened TDB database '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-stream-volumes.tdb'
>I: [pulseaudio] module-stream-restore.c: Successfully opened database file '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-stream-volumes'.
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 added for object /org/pulseaudio/stream_restore1
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry0
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry1
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry2
>I: [pulseaudio] module.c: Loaded "module-stream-restore" (index: #1; argument: "").
>D: [pulseaudio] database-tdb.c: Opened TDB database '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-card-database.tdb'
>I: [pulseaudio] module-card-restore.c: Successfully opened database file '/home/dhaval/.pulse/fbb0c4896a9c0f680475c0a07edd2c5a-card-database'.
>I: [pulseaudio] module.c: Loaded "module-card-restore" (index: #2; argument: "").
>I: [pulseaudio] module.c: Loaded "module-augment-properties" (index: #3; argument: "").
>I: [pulseaudio] module.c: Loaded "module-switch-on-port-available" (index: #4; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-udev-detect.so': success
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:14.2/sound/card0 is busy: no
>D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="pci-0000_00_14.2" card_name="alsa_card.pci-0000_00_14.2" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"'
>D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus session bus 6a5fd7c25bf84c63c152014053271c4a as :1.9887
>D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio0'
>I: [pulseaudio] (alsa-lib)utils.c: could not open configuration file /usr/share/alsa/ucm/HDA ATI SB/HDA ATI SB.conf
>I: [pulseaudio] (alsa-lib)parser.c: error: could not parse configuration for card HDA ATI SB
>I: [pulseaudio] (alsa-lib)main.c: error: failed to import HDA ATI SB use case configuration -2
>I: [pulseaudio] alsa-ucm.c: UCM not available for card HDA ATI SB
>D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
>D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
>D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-front-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-rear-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-internal-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Internal Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-internal-mic', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-dock-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-dock-mic', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Select' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost (+20dB)' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-mic', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-linein'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-aux'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-video'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-tvtuner'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-fm'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-mic-line'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-headphone-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Auto-Mute Mode' succeeded (volume=0, switch=0, enumeration=1).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-headphone-mic', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-headset-mic'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headset Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headset Mic Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headset Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headset Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headset' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-headset-mic', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1ae87e0, direction=2
>D: [pulseaudio] alsa-mixer.c: Path analog-input-front-mic (Front Microphone), direction=2, priority=85, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=60.75
>D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front Mic, direction=2, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Front Mic, alsa_name='Front Mic Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Front Mic Phantom, alsa_name='Front Mic Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Path analog-input-rear-mic (Rear Microphone), direction=2, priority=82, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=60.75
>D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic, direction=2, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front Mic, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Rear Mic, alsa_name='Rear Mic Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Rear Mic Phantom, alsa_name='Rear Mic Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Path analog-input-linein (Line In), direction=2, priority=81, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=30
>D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Line, direction=2, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front Mic, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line, alsa_name='Line Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Phantom, alsa_name='Line Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog 4-channel Input (analog-4-channel-input)
>D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
>I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
>D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Looking at profile input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
>D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
>I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2)
>I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:0: No such file or directory
>D: [pulseaudio] alsa-mixer.c: Caching failure to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
>D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0
>D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
>I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
>D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-mono - will not be able to open output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-stereo - will not be able to open output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-4-channel-input - will not be able to open output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:iec958-stereo - will not be able to open output:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-lineout'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Front Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Front Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out CLFE Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out CLFE Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Surround Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Surround Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Side Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Out Side Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-speaker'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Speaker Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Center Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-output-speaker', none of required-any elements preset.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-desktop-speaker'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' failed.
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Phantom Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headset' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=3, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=2, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=2, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=2, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=2, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones-2'
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=0, switch=1, enumeration=0).
>D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' failed.
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b26ca0, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-114, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
>D: [pulseaudio] alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open surround40:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 11888 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround40:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround40:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b25ca0, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
>D: [pulseaudio] alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open surround41:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 7925 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround41:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround41:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b40cf0, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
>D: [pulseaudio] alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open surround50:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 7925 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround50:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround50:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b417a0, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
>D: [pulseaudio] alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open surround51:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 7925 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround51:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround51:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b77290, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
>D: [pulseaudio] alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open surround71:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 5944 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround71:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround71:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-lineout'.
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1aea3c0, direction=1
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo)
>D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
>I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1p' failed (-2)
>I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:0: No such file or directory
>D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-mono - will not be able to open output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-stereo - will not be able to open output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-4-channel-input - will not be able to open output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open output:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
>D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
>I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0
>I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory
>D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-mono - will not be able to open output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-stereo - will not be able to open output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-4-channel-input - will not be able to open output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-40
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
>D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
>I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0
>I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory
>D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-mono - will not be able to open output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-stereo - will not be able to open output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-4-channel-input - will not be able to open output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-51
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/DTS) (iec958-dts-surround-51)
>D: [pulseaudio] alsa-util.c: Trying dca:0 with SND_PCM_NO_AUTO_FORMAT ...
>I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM dca:0
>I: [pulseaudio] alsa-util.c: Error opening PCM device dca:0: No such file or directory
>D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-mono - will not be able to open output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-stereo - will not be able to open output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-4-channel-input - will not be able to open output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:iec958-stereo - will not be able to open output:iec958-dts-surround-51
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
>D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open hdmi:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo supported.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-0'
>D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=3 Jack' succeeded (not found)
>D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying):
>D: [pulseaudio] alsa-mixer.c: Path Set 0x1b6d330, direction=1
>D: [pulseaudio] alsa-mixer.c: Path hdmi-output-0 (HDMI / DisplayPort), direction=1, priority=59, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf
>D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=3, alsa_name='HDMI/DP,pcm=3 Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-mono - will not be able to open input:analog-mono
>D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-stereo
>D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
>D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo+input:analog-stereo supported.
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-4-channel-input - will not be able to open input:analog-4-channel-input
>D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo
>D: [pulseaudio] alsa-mixer.c: Profile set 0x1a9e140, auto_profiles=yes, probed=yes, n_mappings=7, n_profiles=15, n_decibel_fixes=0
>D: [pulseaudio] alsa-mixer.c: Mapping analog-stereo (Analog Stereo), priority=60, channel_map=front-left,front-right, supported=yes, direction=0
>D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-40 (Analog Surround 4.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-41 (Analog Surround 4.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,lfe, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-50 (Analog Surround 5.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-51 (Analog Surround 5.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-71 (Analog Surround 7.1), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo (Digital Stereo (HDMI)), priority=54, channel_map=front-left,front-right, supported=yes, direction=1
>D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo (Analog Stereo Input), priority=60, supported=yes n_input_mappings=1, n_output_mappings=0
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo (Analog Stereo Output), priority=6000, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-stereo
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo (Analog Stereo Duplex), priority=6060, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-stereo
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 (Analog Surround 4.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-40
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo (Analog Surround 4.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-40
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 (Analog Surround 4.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-41
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo (Analog Surround 4.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-41
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 (Analog Surround 5.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-50
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo (Analog Surround 5.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-50
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 (Analog Surround 5.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-51
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo (Analog Surround 5.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-51
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 (Analog Surround 7.1 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-71
>D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo (Analog Surround 7.1 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output analog-surround-71
>D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo (Digital Stereo (HDMI) Output), priority=5400, supported=yes n_input_mappings=0, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo
>D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo+input:analog-stereo (Digital Stereo (HDMI) Output + Analog Stereo Input), priority=5460, supported=yes n_input_mappings=1, n_output_mappings=1
>D: [pulseaudio] alsa-mixer.c: Input analog-stereo
>D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo
>I: [pulseaudio] module-card-restore.c: Restoring port latency offsets for card alsa_card.pci-0000_00_14.2.
>I: [pulseaudio] card.c: Created 0 "alsa_card.pci-0000_00_14.2"
>D: [pulseaudio] module-alsa-card.c: Found 8 jacks.
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] module-alsa-card.c: Jack 'Line Out Front Jack' is now plugged in
>D: [pulseaudio] device-port.c: Setting port analog-output-lineout to status yes
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: finding port analog-output-lineout
>D: [pulseaudio] module-alsa-card.c: Jack 'Line Out CLFE Jack' is now unplugged
>D: [pulseaudio] module-alsa-card.c: Jack 'Line Out Surround Jack' is now unplugged
>D: [pulseaudio] module-alsa-card.c: Jack 'Line Out Side Jack' is now unplugged
>D: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged
>D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: finding port analog-output-headphones
>D: [pulseaudio] module-alsa-card.c: Jack 'Front Mic Jack' is now unplugged
>D: [pulseaudio] device-port.c: Setting port analog-input-front-mic to status no
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-front-mic
>D: [pulseaudio] module-alsa-card.c: Jack 'Rear Mic Jack' is now plugged in
>D: [pulseaudio] device-port.c: Setting port analog-input-rear-mic to status yes
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-rear-mic
>D: [pulseaudio] module-alsa-card.c: Jack 'Line Jack' is now unplugged
>D: [pulseaudio] device-port.c: Setting port analog-input-linein to status no
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-linein
>D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0'
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>I: [pulseaudio] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples).
>I: [pulseaudio] alsa-util.c: ALSA period wakeups disabled
>I: [pulseaudio] alsa-sink.c: Successfully opened device front:0.
>I: [pulseaudio] alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo).
>I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode.
>I: [pulseaudio] alsa-sink.c: Successfully enabled timer-based scheduling mode.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Added 2 ports
>D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: sink:alsa_output.pci-0000_00_14.2.analog-stereo:null
>I: [pulseaudio] sink.c: Created sink 0 "alsa_output.pci-0000_00_14.2.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
>I: [pulseaudio] sink.c:     alsa.resolution_bits = "16"
>I: [pulseaudio] sink.c:     device.api = "alsa"
>I: [pulseaudio] sink.c:     device.class = "sound"
>I: [pulseaudio] sink.c:     alsa.class = "generic"
>I: [pulseaudio] sink.c:     alsa.subclass = "generic-mix"
>I: [pulseaudio] sink.c:     alsa.name = "VT1708S Analog"
>I: [pulseaudio] sink.c:     alsa.id = "VT1708S Analog"
>I: [pulseaudio] sink.c:     alsa.subdevice = "0"
>I: [pulseaudio] sink.c:     alsa.subdevice_name = "subdevice #0"
>I: [pulseaudio] sink.c:     alsa.device = "0"
>I: [pulseaudio] sink.c:     alsa.card = "0"
>I: [pulseaudio] sink.c:     alsa.card_name = "HDA ATI SB"
>I: [pulseaudio] sink.c:     alsa.long_card_name = "HDA ATI SB at 0xf9ff4000 irq 16"
>I: [pulseaudio] sink.c:     alsa.driver_name = "snd_hda_intel"
>I: [pulseaudio] sink.c:     device.bus_path = "pci-0000:00:14.2"
>I: [pulseaudio] sink.c:     sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
>I: [pulseaudio] sink.c:     device.bus = "pci"
>I: [pulseaudio] sink.c:     device.vendor.id = "1002"
>I: [pulseaudio] sink.c:     device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
>I: [pulseaudio] sink.c:     device.product.id = "4383"
>I: [pulseaudio] sink.c:     device.product.name = "M4A785TD Motherboard"
>I: [pulseaudio] sink.c:     device.form_factor = "internal"
>I: [pulseaudio] sink.c:     device.string = "front:0"
>I: [pulseaudio] sink.c:     device.buffering.buffer_size = "352768"
>I: [pulseaudio] sink.c:     device.buffering.fragment_size = "176384"
>I: [pulseaudio] sink.c:     device.access_mode = "mmap+timer"
>I: [pulseaudio] sink.c:     device.profile.name = "analog-stereo"
>I: [pulseaudio] sink.c:     device.profile.description = "Analog Stereo"
>I: [pulseaudio] sink.c:     device.description = "Built-in Audio Analog Stereo"
>I: [pulseaudio] sink.c:     alsa.mixer_name = "VIA VT1708S"
>I: [pulseaudio] sink.c:     alsa.components = "HDA:11060397,1043836c,00100000"
>I: [pulseaudio] sink.c:     module-udev-detect.discovered = "1"
>I: [pulseaudio] sink.c:     device.icon_name = "audio-card-pci"
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source.c: Created source 0 "alsa_output.pci-0000_00_14.2.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
>I: [pulseaudio] source.c:     device.description = "Monitor of Built-in Audio Analog Stereo"
>I: [pulseaudio] source.c:     device.class = "monitor"
>I: [pulseaudio] source.c:     alsa.card = "0"
>I: [pulseaudio] source.c:     alsa.card_name = "HDA ATI SB"
>I: [pulseaudio] source.c:     alsa.long_card_name = "HDA ATI SB at 0xf9ff4000 irq 16"
>I: [pulseaudio] source.c:     alsa.driver_name = "snd_hda_intel"
>I: [pulseaudio] source.c:     device.bus_path = "pci-0000:00:14.2"
>I: [pulseaudio] source.c:     sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
>I: [pulseaudio] source.c:     device.bus = "pci"
>I: [pulseaudio] source.c:     device.vendor.id = "1002"
>I: [pulseaudio] source.c:     device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
>I: [pulseaudio] source.c:     device.product.id = "4383"
>I: [pulseaudio] source.c:     device.product.name = "M4A785TD Motherboard"
>I: [pulseaudio] source.c:     device.form_factor = "internal"
>I: [pulseaudio] source.c:     device.string = "0"
>I: [pulseaudio] source.c:     module-udev-detect.discovered = "1"
>I: [pulseaudio] source.c:     device.icon_name = "audio-card-pci"
>I: [pulseaudio] alsa-sink.c: Using 2.0 fragments of size 176384 bytes (999.91ms), buffer size is 352768 bytes (1999.82ms)
>I: [pulseaudio] alsa-sink.c: Time scheduling watermark is 20.00ms
>D: [pulseaudio] alsa-sink.c: hwbuf_unused=0
>D: [pulseaudio] alsa-sink.c: setting avail_min=87310
>D: [pulseaudio] alsa-mixer.c: Activating path analog-output-lineout
>D: [pulseaudio] alsa-mixer.c: Path analog-output-lineout (Line Out), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=42, min_dB=-177, max_dB=0
>D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Line Out, alsa_name='Line Out Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Phantom, alsa_name='Line Out Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front, alsa_name='Line Out Front Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Front Phantom, alsa_name='Line Out Front Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE, alsa_name='Line Out CLFE Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out CLFE Phantom, alsa_name='Line Out CLFE Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround, alsa_name='Line Out Surround Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Surround Phantom, alsa_name='Line Out Surround Phantom Jack', detection unavailable
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side, alsa_name='Line Out Side Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Line Out Side Phantom, alsa_name='Line Out Side Phantom Jack', detection unavailable
>I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume.
>I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -177.00 dB to 0.00 dB.
>I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB
>I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
>I: [pulseaudio] alsa-sink.c: Using hardware mute control.
>D: [pulseaudio] alsa-util.c: snd_pcm_dump():
>D: [pulseaudio] alsa-util.c: Soft volume PCM
>D: [pulseaudio] alsa-util.c: Control: PCM Playback Volume
>D: [pulseaudio] alsa-util.c: min_dB: -51
>D: [pulseaudio] alsa-util.c: max_dB: 0
>D: [pulseaudio] alsa-util.c: resolution: 256
>D: [pulseaudio] alsa-util.c: Its setup is:
>D: [pulseaudio] alsa-util.c:   stream       : PLAYBACK
>D: [pulseaudio] alsa-util.c:   access       : MMAP_INTERLEAVED
>D: [pulseaudio] alsa-util.c:   format       : S16_LE
>D: [pulseaudio] alsa-util.c:   subformat    : STD
>D: [pulseaudio] alsa-util.c:   channels     : 2
>D: [pulseaudio] alsa-util.c:   rate         : 44100
>D: [pulseaudio] alsa-util.c:   exact rate   : 44100 (44100/1)
>D: [pulseaudio] alsa-util.c:   msbits       : 16
>D: [pulseaudio] alsa-util.c:   buffer_size  : 88192
>D: [pulseaudio] alsa-util.c:   period_size  : 44096
>D: [pulseaudio] alsa-util.c:   period_time  : 999909
>D: [pulseaudio] alsa-util.c:   tstamp_mode  : ENABLE
>D: [pulseaudio] alsa-util.c:   period_step  : 1
>D: [pulseaudio] alsa-util.c:   avail_min    : 87310
>D: [pulseaudio] alsa-util.c:   period_event : 0
>D: [pulseaudio] alsa-util.c:   start_threshold  : -1
>D: [pulseaudio] alsa-util.c:   stop_threshold   : 6205960286516543488
>D: [pulseaudio] alsa-util.c:   silence_threshold: 0
>D: [pulseaudio] alsa-util.c:   silence_size : 0
>D: [pulseaudio] alsa-util.c:   boundary     : 6205960286516543488
>D: [pulseaudio] alsa-util.c: Slave: Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
>D: [pulseaudio] alsa-util.c: Its setup is:
>D: [pulseaudio] alsa-util.c:   stream       : PLAYBACK
>D: [pulseaudio] alsa-util.c:   access       : MMAP_INTERLEAVED
>D: [pulseaudio] alsa-util.c:   format       : S16_LE
>D: [pulseaudio] alsa-util.c:   subformat    : STD
>D: [pulseaudio] alsa-util.c:   channels     : 2
>D: [pulseaudio] alsa-util.c:   rate         : 44100
>D: [pulseaudio] alsa-util.c:   exact rate   : 44100 (44100/1)
>D: [pulseaudio] alsa-util.c:   msbits       : 16
>D: [pulseaudio] alsa-util.c:   buffer_size  : 88192
>D: [pulseaudio] alsa-util.c:   period_size  : 44096
>D: [pulseaudio] alsa-util.c:   period_time  : 999909
>D: [pulseaudio] alsa-util.c:   tstamp_mode  : ENABLE
>D: [pulseaudio] alsa-util.c:   period_step  : 1
>D: [pulseaudio] alsa-util.c:   avail_min    : 87310
>D: [pulseaudio] alsa-util.c:   period_event : 0
>D: [pulseaudio] alsa-util.c:   start_threshold  : -1
>D: [pulseaudio] alsa-util.c:   stop_threshold   : 6205960286516543488
>D: [pulseaudio] alsa-util.c:   silence_threshold: 0
>D: [pulseaudio] alsa-util.c:   silence_size : 0
>D: [pulseaudio] alsa-util.c:   boundary     : 6205960286516543488
>D: [pulseaudio] alsa-util.c:   appl_ptr     : 0
>D: [pulseaudio] alsa-util.c:   hw_ptr       : 0
>D: [pulseaudio] alsa-sink.c: Read hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Thread starting up
>I: [alsa-sink-VT1708S Analog] core-util.c: Clamping rlimit-rttime to 200000 for RealtimeKit
>D: [alsa-sink-VT1708S Analog] core-util.c: RealtimeKit worked.
>I: [alsa-sink-VT1708S Analog] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] module-device-restore.c: Could not set format on sink alsa_output.pci-0000_00_14.2.analog-stereo
>D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
>D: [pulseaudio] alsa-util.c: Managed to open front:0
>I: [pulseaudio] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [pulseaudio] alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples).
>I: [pulseaudio] alsa-util.c: ALSA period wakeups disabled
>I: [pulseaudio] alsa-source.c: Successfully opened device front:0.
>I: [pulseaudio] alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo).
>I: [pulseaudio] alsa-source.c: Successfully enabled mmap() mode.
>I: [pulseaudio] alsa-source.c: Successfully enabled timer-based scheduling mode.
>I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0
>I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory
>I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0'
>D: [pulseaudio] alsa-mixer.c: Added 3 ports
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-switch-on-port-available.c: Switching initial port for source 'alsa_input.pci-0000_00_14.2.analog-stereo' to '(null)'
>I: [pulseaudio] module-device-restore.c: Restoring volume for source alsa_input.pci-0000_00_14.2.analog-stereo: front-left: 65540 / 100%,   front-right: 65540 / 100%
>I: [pulseaudio] source.c: Created source 1 "alsa_input.pci-0000_00_14.2.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
>I: [pulseaudio] source.c:     alsa.resolution_bits = "16"
>I: [pulseaudio] source.c:     device.api = "alsa"
>I: [pulseaudio] source.c:     device.class = "sound"
>I: [pulseaudio] source.c:     alsa.class = "generic"
>I: [pulseaudio] source.c:     alsa.subclass = "generic-mix"
>I: [pulseaudio] source.c:     alsa.name = "VT1708S Analog"
>I: [pulseaudio] source.c:     alsa.id = "VT1708S Analog"
>I: [pulseaudio] source.c:     alsa.subdevice = "0"
>I: [pulseaudio] source.c:     alsa.subdevice_name = "subdevice #0"
>I: [pulseaudio] source.c:     alsa.device = "0"
>I: [pulseaudio] source.c:     alsa.card = "0"
>I: [pulseaudio] source.c:     alsa.card_name = "HDA ATI SB"
>I: [pulseaudio] source.c:     alsa.long_card_name = "HDA ATI SB at 0xf9ff4000 irq 16"
>I: [pulseaudio] source.c:     alsa.driver_name = "snd_hda_intel"
>I: [pulseaudio] source.c:     device.bus_path = "pci-0000:00:14.2"
>I: [pulseaudio] source.c:     sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
>I: [pulseaudio] source.c:     device.bus = "pci"
>I: [pulseaudio] source.c:     device.vendor.id = "1002"
>I: [pulseaudio] source.c:     device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
>I: [pulseaudio] source.c:     device.product.id = "4383"
>I: [pulseaudio] source.c:     device.product.name = "M4A785TD Motherboard"
>I: [pulseaudio] source.c:     device.form_factor = "internal"
>I: [pulseaudio] source.c:     device.string = "front:0"
>I: [pulseaudio] source.c:     device.buffering.buffer_size = "352768"
>I: [pulseaudio] source.c:     device.buffering.fragment_size = "176384"
>I: [pulseaudio] source.c:     device.access_mode = "mmap+timer"
>I: [pulseaudio] source.c:     device.profile.name = "analog-stereo"
>I: [pulseaudio] source.c:     device.profile.description = "Analog Stereo"
>I: [pulseaudio] source.c:     device.description = "Built-in Audio Analog Stereo"
>I: [pulseaudio] source.c:     alsa.mixer_name = "VIA VT1708S"
>I: [pulseaudio] source.c:     alsa.components = "HDA:11060397,1043836c,00100000"
>I: [pulseaudio] source.c:     module-udev-detect.discovered = "1"
>I: [pulseaudio] source.c:     device.icon_name = "audio-card-pci"
>I: [pulseaudio] alsa-source.c: Using 2.0 fragments of size 176384 bytes (999.91ms), buffer size is 352768 bytes (1999.82ms)
>I: [pulseaudio] alsa-source.c: Time scheduling watermark is 20.00ms
>D: [pulseaudio] alsa-source.c: hwbuf_unused=0
>D: [pulseaudio] alsa-source.c: setting avail_min=87310
>D: [pulseaudio] alsa-mixer.c: Activating path analog-input-rear-mic
>D: [pulseaudio] alsa-mixer.c: Path analog-input-rear-mic (Rear Microphone), direction=2, priority=82, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-16.5, max_dB=60.75
>D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Rear Mic, direction=2, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
>D: [pulseaudio] alsa-mixer.c: Element Front Mic, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
>D: [pulseaudio] alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
>D: [pulseaudio] alsa-mixer.c: Jack Rear Mic, alsa_name='Rear Mic Jack', detection possible
>D: [pulseaudio] alsa-mixer.c: Jack Rear Mic Phantom, alsa_name='Rear Mic Phantom Jack', detection unavailable
>I: [pulseaudio] alsa-source.c: Successfully enabled deferred volume.
>I: [pulseaudio] alsa-source.c: Hardware volume ranges from -16.50 dB to 60.75 dB.
>I: [pulseaudio] alsa-source.c: Fixing base volume to -60.75 dB
>I: [pulseaudio] alsa-source.c: Using hardware volume control. Hardware dB scale supported.
>I: [pulseaudio] alsa-source.c: Using hardware mute control.
>D: [pulseaudio] alsa-util.c: snd_pcm_dump():
>D: [pulseaudio] alsa-util.c: Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
>D: [pulseaudio] alsa-util.c: Its setup is:
>D: [pulseaudio] alsa-util.c:   stream       : CAPTURE
>D: [pulseaudio] alsa-util.c:   access       : MMAP_INTERLEAVED
>D: [pulseaudio] alsa-util.c:   format       : S16_LE
>D: [pulseaudio] alsa-util.c:   subformat    : STD
>D: [pulseaudio] alsa-util.c:   channels     : 2
>D: [pulseaudio] alsa-util.c:   rate         : 44100
>D: [pulseaudio] alsa-util.c:   exact rate   : 44100 (44100/1)
>D: [pulseaudio] alsa-util.c:   msbits       : 16
>D: [pulseaudio] alsa-util.c:   buffer_size  : 88192
>D: [pulseaudio] alsa-util.c:   period_size  : 44096
>D: [pulseaudio] alsa-util.c:   period_time  : 999909
>D: [pulseaudio] alsa-util.c:   tstamp_mode  : ENABLE
>D: [pulseaudio] alsa-util.c:   period_step  : 1
>D: [pulseaudio] alsa-util.c:   avail_min    : 87310
>D: [pulseaudio] alsa-util.c:   period_event : 0
>D: [pulseaudio] alsa-util.c:   start_threshold  : -1
>D: [pulseaudio] alsa-util.c:   stop_threshold   : 6205960286516543488
>D: [pulseaudio] alsa-util.c:   silence_threshold: 0
>D: [pulseaudio] alsa-util.c:   silence_size : 0
>D: [pulseaudio] alsa-util.c:   boundary     : 6205960286516543488
>D: [pulseaudio] alsa-util.c:   appl_ptr     : 0
>D: [pulseaudio] alsa-util.c:   hw_ptr       : 0
>D: [pulseaudio] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [pulseaudio] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Thread starting up
>D: [pulseaudio] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] core-util.c: RealtimeKit worked.
>I: [alsa-source-VT1708S Analog] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5.
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-alsa-card.c: No ELD device found for port hdmi-output-0.
>I: [pulseaudio] module.c: Loaded "module-alsa-card" (index: #6; argument: "device_id="0" name="pci-0000_00_14.2" card_name="alsa_card.pci-0000_00_14.2" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"").
>I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:14.2/sound/card0 (alsa_card.pci-0000_00_14.2) module loaded.
>I: [pulseaudio] module-udev-detect.c: Found 1 cards.
>I: [pulseaudio] module.c: Loaded "module-udev-detect" (index: #5; argument: "tsched=1").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-jackdbus-detect.so': failure
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-bluetooth-policy.so': success
>I: [pulseaudio] module.c: Loaded "module-bluetooth-policy" (index: #7; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-bluetooth-discover.so': success
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-bluez5-discover.so': success
>D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus system bus da43340d374557bda234f26653271c06 as :1.922
>I: [pulseaudio] module.c: Loaded "module-bluez5-discover" (index: #9; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-bluez4-discover.so': failure
>I: [pulseaudio] module.c: Loaded "module-bluetooth-discover" (index: #8; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-esound-protocol-unix.so': success
>I: [pulseaudio] module.c: Loaded "module-esound-protocol-unix" (index: #10; argument: "").
>I: [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #11; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-gconf.so': failure
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_14.2.analog-stereo'.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_14.2.analog-stereo'.
>I: [pulseaudio] module.c: Loaded "module-default-device-restore" (index: #12; argument: "").
>I: [pulseaudio] module.c: Loaded "module-rescue-streams" (index: #13; argument: "").
>I: [pulseaudio] module.c: Loaded "module-always-sink" (index: #14; argument: "").
>I: [pulseaudio] module.c: Loaded "module-intended-roles" (index: #15; argument: "").
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>I: [pulseaudio] module.c: Loaded "module-suspend-on-idle" (index: #16; argument: "").
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-console-kit.so': success
>D: [pulseaudio] module.c: Checking for existence of '/usr/lib64/pulse-4.0/modules/module-systemd-login.so': success
>I: [pulseaudio] client.c: Created 0 "Login Session 1"
>D: [pulseaudio] module-systemd-login.c: Added new session 1
>I: [pulseaudio] module.c: Loaded "module-systemd-login" (index: #17; argument: "").
>I: [pulseaudio] module.c: Loaded "module-position-event-sounds" (index: #18; argument: "").
>D: [pulseaudio] module-role-cork.c: Using role 'phone' as trigger role.
>D: [pulseaudio] module-role-cork.c: Using roles 'music' and 'video' as cork roles.
>I: [pulseaudio] module.c: Loaded "module-role-cork" (index: #19; argument: "").
>I: [pulseaudio] module.c: Loaded "module-filter-heuristics" (index: #20; argument: "").
>I: [pulseaudio] module.c: Loaded "module-filter-apply" (index: #21; argument: "").
>D: [pulseaudio] main.c: Got org.PulseAudio1!
>D: [pulseaudio] main.c: Got org.pulseaudio.Server!
>I: [pulseaudio] main.c: Daemon startup complete.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>E: [pulseaudio] bluez5-util.c: GetManagedObjects() failed: org.freedesktop.systemd1.LoadFailed: Unit dbus-org.bluez.service failed to load: No such file or directory.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>I: [pulseaudio] client.c: Created 1 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-augment-properties.c: Found /usr/share/applications/audacity.desktop.
>I: [pulseaudio] client.c: Freed 1 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 2 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 2 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 3 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 3 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 4 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 4 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 5 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "8000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 0 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 8000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=131072.00 ms, minreq=128.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] protocol-native.c: Requested latency=128.00 ms, Received latency=128.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 131200.00 ms = 130816.00 ms + 2*128.00 ms + 128.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 128.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=328192
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352228 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88057
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88057
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352228 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1189
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352228 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 23860 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 5965
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 5965
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 23860 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1088
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 0 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 5 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 6 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "9600"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 1 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 9600Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=109226.67 ms, minreq=106.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=106.67 ms, Received latency=106.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 109333.33 ms = 109013.33 ms + 2*106.67 ms + 106.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 106.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=333956
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352328 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88082
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88082
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352328 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1252
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352328 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 18132 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 4533
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 4533
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 18132 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1179
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 1 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 6 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 7 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "11025"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 2 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 11025Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=95108.93 ms, minreq=92.88 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=92.88 ms, Received latency=92.88 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 95201.81 ms = 94923.17 ms + 2*92.88 ms + 92.88 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 92.88ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=336388
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352040 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88010
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88010
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352040 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1378
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352040 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 15980 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 3995
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 3995
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 15980 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1348
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 2 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 7 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 8 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "12000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 3 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 12000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=87381.33 ms, minreq=85.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=85.33 ms, Received latency=85.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 87466.67 ms = 87210.67 ms + 2*85.33 ms + 85.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 85.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=336388
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352284 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88071
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88071
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352284 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1184
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352284 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 15544 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 3886
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 3886
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 15544 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1159
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 3 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 8 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 9 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "15000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 4 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 15000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=69905.07 ms, minreq=68.27 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=68.27 ms, Received latency=68.27 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 69973.33 ms = 69768.53 ms + 2*68.27 ms + 68.27 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 68.27ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=340728
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352324 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88081
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88081
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352324 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1262
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352324 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 11328 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 2832
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 2832
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 11328 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1325
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 4 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 9 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 10 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "16000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 5 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 16000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=65536.00 ms, minreq=64.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=64.00 ms, Received latency=64.00 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 65600.00 ms = 65408.00 ms + 2*64.00 ms + 64.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 64.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=340480
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352300 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88075
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88075
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352300 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1195
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352300 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 11576 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 2894
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 2894
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 11576 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1109
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 5 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 10 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 11 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "22050"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 6 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 22050Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=47554.47 ms, minreq=46.44 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=46.44 ms, Received latency=46.44 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 47600.91 ms = 47461.59 ms + 2*46.44 ms + 46.44 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 46.44ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=344580
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352196 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88049
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88049
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352196 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1436
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352196 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 7624 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1906
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1906
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 7624 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1403
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 6 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 11 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 12 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "24000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 7 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 24000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=43690.67 ms, minreq=42.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=42.67 ms, Received latency=42.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 43733.33 ms = 43605.33 ms + 2*42.67 ms + 42.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 42.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=344580
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352344 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88086
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88086
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352344 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1266
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352344 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 7640 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1910
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1910
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 7640 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1212
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 7 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 12 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 13 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "32000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 8 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 32000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=32768.00 ms, minreq=32.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=32.00 ms, Received latency=32.00 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 32800.00 ms = 32704.00 ms + 2*32.00 ms + 32.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 32.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=346624
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 351940 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 87985
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 87985
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 351940 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1249
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 351940 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 5692 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1423
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1423
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 5692 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1247
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 8 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 13 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 14 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "44100"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 9 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=23777.23 ms, minreq=23.22 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 23800.45 ms = 23730.79 ms + 2*23.22 ms + 23.22 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 23.22ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=348676
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87611
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352348 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88087
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88087
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352348 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1321
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352348 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87610
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 3340 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 835
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 835
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 3340 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1325
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 9 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 14 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 15 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "48000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87610
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 10 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 48000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=21845.33 ms, minreq=21.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=21.33 ms, Received latency=21.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21866.67 ms = 21802.67 ms + 2*21.33 ms + 21.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 21.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=348676
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87651
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352152 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88038
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88038
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352152 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1177
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352152 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87650
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 3340 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 835
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 835
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 3340 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1151
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 10 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 15 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 16 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "88200"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87650
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 11 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 88200Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=11888.62 ms, minreq=11.61 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=11.61 ms, Received latency=11.61 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 11900.23 ms = 11865.40 ms + 2*11.61 ms + 11.61 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 11.61ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=350724
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87938
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352304 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88076
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88076
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352304 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1276
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352304 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 1404 bytes.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 351
>I: [pulseaudio] sink-input.c: Freeing input 11 "ALSA Playback"
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 351
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 1404 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1225
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] client.c: Freed 16 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 17 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "96000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 12 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 96000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=10922.67 ms, minreq=10.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=10.67 ms, Received latency=10.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10933.33 ms = 10901.33 ms + 2*10.67 ms + 10.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 10.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=350724
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352352 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88088
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88088
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352352 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1280
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352352 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 1616 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 404
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 404
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 1616 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1278
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 12 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 17 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 18 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "176400"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 13 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 176400Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=5944.31 ms, minreq=5.80 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=5.80 ms, Received latency=5.80 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5950.11 ms = 5932.70 ms + 2*5.80 ms + 5.80 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 5.80ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=351748
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88065
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352376 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352376 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1405
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352376 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88017
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 732 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 183
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 183
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 732 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1388
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 13 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 18 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 19 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "192000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 14 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=5461.33 ms, minreq=5.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=5.33 ms, Received latency=5.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5466.67 ms = 5450.67 ms + 2*5.33 ms + 5.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 5.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=351748
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88066
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352364 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88091
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88091
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352364 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1268
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352364 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 728 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 182
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 182
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 728 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1257
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 14 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 19 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 20 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 20 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 21 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 21 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 22 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 0 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 8000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 256.00 ms = 128.00 ms + 128.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5263
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5263
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 0 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 22 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 23 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 1 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 9600Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 213.33 ms = 106.67 ms + 106.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 1 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 23 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 24 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 2 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 11025Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 185.76 ms = 92.88 ms + 92.88 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 2 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 24 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 25 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 3 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 12000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 170.67 ms = 85.33 ms + 85.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 3 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 25 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 26 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 4 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 15000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 136.53 ms = 68.27 ms + 68.27 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 4 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 26 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 27 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 5 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 16000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 128.00 ms = 64.00 ms + 64.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 5 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 27 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 28 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 6 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 22050Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 92.88 ms = 46.44 ms + 46.44 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 6 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 28 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 29 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 7 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 24000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 85.33 ms = 42.67 ms + 42.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 7 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 29 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 30 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 8 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 32000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 64.00 ms = 32.00 ms + 32.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 8 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 30 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 31 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 9 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 46.44 ms = 23.22 ms + 23.22 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 9 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 31 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 32 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 10 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 48000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 42.67 ms = 21.33 ms + 21.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 10 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 32 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 33 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 11 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 88200Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 23.22 ms = 11.61 ms + 11.61 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 11 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 33 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 34 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 12 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 96000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21.33 ms = 10.67 ms + 10.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 12 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 34 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 35 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 13 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 176400Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 11.61 ms = 5.80 ms + 5.80 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88017
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 13 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 35 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 36 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 14 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10.67 ms = 5.33 ms + 5.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 14 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 36 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 37 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 37 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 38 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 38 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 39 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "8000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 15 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 8000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=131072.00 ms, minreq=128.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=128.00 ms, Received latency=128.00 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 131200.00 ms = 130816.00 ms + 2*128.00 ms + 128.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 128.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=328192
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88001
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 334876 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 83719
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 83719
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 334876 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1255
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 334876 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 24124 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 6031
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 6031
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 24124 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1269
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 15 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 39 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 40 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "9600"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88001
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 16 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 9600Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=109226.67 ms, minreq=106.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=106.67 ms, Received latency=106.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 109333.33 ms = 109013.33 ms + 2*106.67 ms + 106.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 106.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=333956
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352376 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352376 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1399
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352376 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 18280 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 4570
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 4570
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 18280 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1367
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 16 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 40 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 41 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "11025"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 17 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 11025Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=95108.93 ms, minreq=92.88 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=92.88 ms, Received latency=92.88 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 95201.81 ms = 94923.17 ms + 2*92.88 ms + 92.88 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 92.88ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=336388
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 351736 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 87934
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 87934
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 351736 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1401
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 351736 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 15928 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 3982
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 3982
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 15928 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1403
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 17 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 41 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 42 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "12000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 18 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 12000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=87381.33 ms, minreq=85.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=85.33 ms, Received latency=85.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 87466.67 ms = 87210.67 ms + 2*85.33 ms + 85.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 85.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=336388
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352348 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88087
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88087
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352348 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1286
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352348 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 15900 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 3975
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 3975
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 15900 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1295
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 18 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 42 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 43 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "15000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 19 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 15000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=69905.07 ms, minreq=68.27 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=68.27 ms, Received latency=68.27 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 69973.33 ms = 69768.53 ms + 2*68.27 ms + 68.27 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 68.27ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=340728
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352360 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88090
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88090
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352360 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1404
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352360 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 11616 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 2904
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 2904
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 11616 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1400
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 19 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 43 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 44 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "16000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 20 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 16000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=65536.00 ms, minreq=64.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=64.00 ms, Received latency=64.00 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 65600.00 ms = 65408.00 ms + 2*64.00 ms + 64.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 64.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=340480
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352284 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88071
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88071
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352284 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1181
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352284 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 11356 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 2839
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 2839
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 11356 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1199
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 20 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 44 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 45 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "22050"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 21 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 22050Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=47554.47 ms, minreq=46.44 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=46.44 ms, Received latency=46.44 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 47600.91 ms = 47461.59 ms + 2*46.44 ms + 46.44 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 46.44ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=344580
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352184 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88046
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88046
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352184 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1265
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352184 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 7528 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1882
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1882
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 7528 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1303
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 21 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 45 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 46 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "24000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 22 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 24000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=43690.67 ms, minreq=42.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=42.67 ms, Received latency=42.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 43733.33 ms = 43605.33 ms + 2*42.67 ms + 42.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 42.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=344580
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352276 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88069
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88069
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352276 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1124
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352276 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 7480 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1870
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1870
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 7480 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1204
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 22 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 46 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 47 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "32000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 23 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 32000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=32768.00 ms, minreq=32.00 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=32.00 ms, Received latency=32.00 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 32800.00 ms = 32704.00 ms + 2*32.00 ms + 32.00 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 32.00ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=346624
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87311
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352172 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88043
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88043
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352172 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1262
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352172 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 5716 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 1429
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 1429
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 5716 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1261
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 23 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 47 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 48 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "44100"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 24 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=23777.23 ms, minreq=23.22 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 23800.45 ms = 23730.79 ms + 2*23.22 ms + 23.22 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 23.22ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=348676
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87611
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352376 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88094
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352376 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1391
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352376 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87610
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 3616 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 904
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 904
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 3616 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1375
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 24 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 48 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 49 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "48000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87610
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 25 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 48000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=21845.33 ms, minreq=21.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=21.33 ms, Received latency=21.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21866.67 ms = 21802.67 ms + 2*21.33 ms + 21.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 21.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=348676
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87651
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352228 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88057
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88057
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352228 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1299
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352228 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87650
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 3576 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 894
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 894
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 3576 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1294
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 25 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 49 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 50 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "88200"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87650
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 26 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 88200Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=11888.62 ms, minreq=11.61 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=11.61 ms, Received latency=11.61 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 11900.23 ms = 11865.40 ms + 2*11.61 ms + 11.61 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 11.61ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=350724
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87938
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352392 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88098
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88098
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352392 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1415
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352392 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 1604 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 401
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 401
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 1604 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1370
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 26 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 50 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 51 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "96000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 27 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 96000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=10922.67 ms, minreq=10.67 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=10.67 ms, Received latency=10.67 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10933.33 ms = 10901.33 ms + 2*10.67 ms + 10.67 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 10.67ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=350724
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352372 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88093
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88093
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352372 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1271
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352372 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 1748 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 437
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 437
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 1748 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1271
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 27 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 51 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 52 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "176400"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87937
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 20.00ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 28 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 176400Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=5944.31 ms, minreq=5.80 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=5.80 ms, Received latency=5.80 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5950.11 ms = 5932.70 ms + 2*5.80 ms + 5.80 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 5.80ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=351748
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88065
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352408 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88102
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88102
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352408 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1378
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352408 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88017
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 544 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 136
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 136
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 544 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1400
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 28 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 52 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 53 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "192000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] sink.c: Suspending sink alsa_output.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 48000
>D: [pulseaudio] source.c: Suspending source alsa_output.pci-0000_00_14.2.analog-stereo.monitor due to changing the sample rate.
>I: [pulseaudio] source.c: Changed sampling rate successfully
>I: [pulseaudio] sink.c: Changed sampling rate successfully
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Trying resume...
>I: [alsa-sink-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-sink-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-sink-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=87310
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Time scheduling watermark is 18.38ms
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Resumed successfully...
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Starting playback.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 29 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=5461.33 ms, minreq=5.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=5.33 ms, Received latency=5.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5466.67 ms = 5450.67 ms + 2*5.33 ms + 5.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 5.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=351748
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88066
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 352344 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 88086
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 88086
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 352344 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1250
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 352344 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 568 bytes.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 142
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 142
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 568 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 29 "ALSA Playback"
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1220
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] client.c: Freed 53 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 54 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 54 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 55 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 55 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 56 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 15 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 8000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 256.00 ms = 128.00 ms + 128.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5953
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5953
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>I: [alsa-source-VT1708S Analog] alsa-source.c: Increasing wakeup watermark to 8.00 ms
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87808
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 15 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 56 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 57 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87808
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 16 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 9600Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 213.33 ms = 106.67 ms + 106.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 16 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 57 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 58 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 17 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 11025Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 185.76 ms = 92.88 ms + 92.88 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 17 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 58 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 59 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 18 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 12000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 170.67 ms = 85.33 ms + 85.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 18 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 59 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 60 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 19 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 15000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 136.53 ms = 68.27 ms + 68.27 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 19 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 60 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 61 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 20 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 16000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 128.00 ms = 64.00 ms + 64.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 20 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 61 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 62 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 21 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 22050Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 92.88 ms = 46.44 ms + 46.44 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 21 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 62 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 63 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 22 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 24000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 85.33 ms = 42.67 ms + 42.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 22 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 63 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 64 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 23 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 32000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 64.00 ms = 32.00 ms + 32.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 23 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 64 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 65 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 24 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 46.44 ms = 23.22 ms + 23.22 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 24 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 65 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 66 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 25 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 48000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 42.67 ms = 21.33 ms + 21.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 25 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 66 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 67 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 26 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 88200Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 23.22 ms = 11.61 ms + 11.61 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 26 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 67 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 68 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 27 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 96000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21.33 ms = 10.67 ms + 10.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 27 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 68 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 69 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 28 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 176400Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 11.61 ms = 5.80 ms + 5.80 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88017
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 28 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 69 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 70 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 29 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10.67 ms = 5.33 ms + 5.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 29 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 70 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 71 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 71 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 72 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 72 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 73 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Created 74 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 30 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10.67 ms = 5.33 ms + 5.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>I: [alsa-source-VT1708S Analog] alsa-source.c: Increasing minimal latency to 1.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "192000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 30 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=16.00 ms, minreq=5.33 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=5.33 ms, Received latency=5.33 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=12288, base=4, prebuf=4096, minreq=4096 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=12288, base=4, prebuf=4096, minreq=4096 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21.33 ms = 5.33 ms + 2*5.33 ms + 5.33 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 5.33ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=351748
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88066
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 332496 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 83124
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 83124
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 332496 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1255
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 332496 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>I: [pulseaudio] source-output.c: Freeing output 30 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 73 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 628 bytes.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 157
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 157
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 628 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1246
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>I: [pulseaudio] sink-input.c: Freeing input 30 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 74 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 75 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 31 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5.80 ms = 2.90 ms + 2.90 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352212
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=70
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352212
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=70
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>I: [alsa-source-VT1708S Analog] alsa-source.c: Increasing minimal latency to 2.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352212
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=70
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 31 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 75 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 76 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Playback, because it lacks role.
>D: [pulseaudio] sink-input.c: Negotiated format: pcm, format.sample_format = "\"float32le\""  format.rate = "192000"  format.channels = "1"  format.channel_map = "\"mono\""
>I: [pulseaudio] sink-input.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00 
>D: [pulseaudio] resampler.c:     +------
>D: [pulseaudio] resampler.c: O00 | 1.000
>D: [pulseaudio] resampler.c: O01 | 1.000
>I: [pulseaudio] remap_sse.c: Using SSE2 mono to stereo remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] sink-input.c: Created input 31 "ALSA Playback" on alsa_output.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] sink-input.c:     media.name = "ALSA Playback"
>I: [pulseaudio] sink-input.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] sink-input.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] sink-input.c:     native-protocol.version = "29"
>I: [pulseaudio] sink-input.c:     application.process.id = "18069"
>I: [pulseaudio] sink-input.c:     application.process.user = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.host = "dhaval"
>I: [pulseaudio] sink-input.c:     application.process.binary = "audacity"
>I: [pulseaudio] sink-input.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] sink-input.c:     window.x11.display = ":0"
>I: [pulseaudio] sink-input.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] sink-input.c:     application.process.session_id = "1"
>I: [pulseaudio] sink-input.c:     application.icon_name = "audacity"
>I: [pulseaudio] sink-input.c:     module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Requested tlength=11.60 ms, minreq=2.90 ms
>D: [pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
>D: [pulseaudio] protocol-native.c: Requested latency=2.90 ms, Received latency=2.90 ms
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=8912, base=4, prebuf=2228, minreq=2228 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=8912, base=4, prebuf=2228, minreq=2228 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 14.51 ms = 5.80 ms + 2*2.90 ms + 2.90 ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Latency set to 2.90ms
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=352212
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88123
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requesting rewind due to latency change.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 351572 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 87893
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 87893
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 351572 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1288
>D: [alsa-sink-VT1708S Analog] sink-input.c: Have to rewind 351572 bytes on render memblockq.
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink-input.c: Requesting rewind due to corking
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 0 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Mhmm, actually there is nothing to rewind.
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: hwbuf_unused=0
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: setting avail_min=88000
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Got hardware volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-sink-VT1708S Analog] sink.c: Volume not changing
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Requested to rewind 352768 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Limited to 276 bytes.
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: before: 69
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: after: 69
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Rewound 276 bytes.
>D: [alsa-sink-VT1708S Analog] sink.c: Processing rewind...
>D: [alsa-sink-VT1708S Analog] sink.c: latency = 1271
>D: [alsa-sink-VT1708S Analog] source.c: Processing rewind...
>D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] sink-input.c: Freeing input 31 "ALSA Playback"
>I: [pulseaudio] client.c: Freed 76 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 77 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88001
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 32 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5.80 ms = 2.90 ms + 2.90 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352260
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=64
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352260
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=64
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88017
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 32 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 77 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 78 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 33 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 5.80 ms = 2.90 ms + 2.90 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352260
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=64
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 2.90ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=352260
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=64
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88017
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 33 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 78 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 79 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 34 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 8000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 256.00 ms = 128.00 ms + 128.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5263
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 128.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=328192
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=5263
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 34 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 79 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 80 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 35 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 9600Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 213.33 ms = 106.67 ms + 106.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 106.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=333956
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3822
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 35 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 80 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 81 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 36 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 11025Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 185.76 ms = 92.88 ms + 92.88 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 92.88ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 36 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 81 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 82 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 37 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 12000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 170.67 ms = 85.33 ms + 85.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 85.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=336388
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=3214
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 37 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 82 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 83 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 38 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 15000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 136.53 ms = 68.27 ms + 68.27 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 68.27ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340728
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 38 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 83 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 84 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 39 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 16000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 128.00 ms = 64.00 ms + 64.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 64.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=340480
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=2191
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 39 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 84 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 85 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 40 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 22050Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 92.88 ms = 46.44 ms + 46.44 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 46.44ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 40 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 85 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 86 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 41 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 24000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 85.33 ms = 42.67 ms + 42.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 42.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=344580
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=1166
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 41 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 86 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 87 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 42 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 32000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 64.00 ms = 32.00 ms + 32.00 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 32.00ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=346624
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=655
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 42 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 87 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 88 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 43 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 46.44 ms = 23.22 ms + 23.22 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 23.22ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=442
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 43 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 88 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 89 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87610
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 44 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 48000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 42.67 ms = 21.33 ms + 21.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 21.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348676
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=482
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 44 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 89 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 90 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 45 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 88200Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 23.22 ms = 11.61 ms + 11.61 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 45 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 90 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 91 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 46 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 96000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 21.33 ms = 10.67 ms + 10.67 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 10.67ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=256
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 46 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 91 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 92 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87937
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 47 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 176400Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 11.61 ms = 5.80 ms + 5.80 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.80ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=128
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88017
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 47 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 92 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 93 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 48000
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 21845 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 18.38ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 48000 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'speex-float-1'
>I: [pulseaudio] resampler.c: Using float32le as working format.
>I: [pulseaudio] resampler.c: Choosing speex quality setting 1.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 48 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 192000Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 10.67 ms = 5.33 ms + 5.33 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 5.33ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=351748
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=129
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88000
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 48 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 93 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 94 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 94 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 95 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>I: [pulseaudio] client.c: Freed 95 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] client.c: Created 96 "Native client (UNIX socket client)"
>D: [pulseaudio] protocol-native.c: Protocol version: remote 29, local 29
>I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
>D: [pulseaudio] protocol-native.c: SHM possible: yes
>D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
>D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for audacity
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>D: [pulseaudio] source.c: Suspending source alsa_input.pci-0000_00_14.2.analog-stereo due to changing the sample rate.
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0020, suspending
>I: [alsa-source-VT1708S Analog] alsa-source.c: Device suspended...
>I: [pulseaudio] alsa-source.c: Updating rate for device front:0, new rate is 44100
>I: [pulseaudio] source.c: Changed sampling rate successfully
>D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0000, resuming
>I: [alsa-source-VT1708S Analog] alsa-source.c: Trying resume...
>I: [alsa-source-VT1708S Analog] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
>D: [alsa-source-VT1708S Analog] alsa-util.c: Maximum hw buffer size is 23777 ms
>D: [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 samples), period size second (to 44096 samples).
>I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=88001
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310
>I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is 20.00ms
>I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully...
>I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 49 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 49.98 ms = 24.99 ms + 24.99 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 24.99ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348364
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=443
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 24.99ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348364
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=443
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87533
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 49 "ALSA Capture"
>D: [pulseaudio] module-intended-roles.c: Not setting device for stream ALSA Capture, because it lacks role.
>I: [pulseaudio] source-output.c: Trying to change sample rate
>I: [pulseaudio] source-output.c: Rate changed to 44100 Hz
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming.
>I: [pulseaudio] resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
>D: [pulseaudio] resampler.c: Channel matrix:
>D: [pulseaudio] resampler.c:        I00   I01 
>D: [pulseaudio] resampler.c:     +------------
>D: [pulseaudio] resampler.c: O00 | 0.500 0.500
>I: [pulseaudio] remap.c: Using generic matrix remapping
>I: [pulseaudio] resampler.c: Using resampler 'copy'
>I: [pulseaudio] resampler.c: Using s16le as working format.
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
>I: [pulseaudio] source-output.c: Created output 50 "ALSA Capture" on alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch 44100Hz and channel map mono
>I: [pulseaudio] source-output.c:     media.name = "ALSA Capture"
>I: [pulseaudio] source-output.c:     application.name = "ALSA plug-in [audacity]"
>I: [pulseaudio] source-output.c:     native-protocol.peer = "UNIX socket client"
>I: [pulseaudio] source-output.c:     native-protocol.version = "29"
>I: [pulseaudio] source-output.c:     application.process.id = "18069"
>I: [pulseaudio] source-output.c:     application.process.user = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.host = "dhaval"
>I: [pulseaudio] source-output.c:     application.process.binary = "audacity"
>I: [pulseaudio] source-output.c:     application.language = "en_US.UTF-8"
>I: [pulseaudio] source-output.c:     window.x11.display = ":0"
>I: [pulseaudio] source-output.c:     application.process.machine_id = "fbb0c4896a9c0f680475c0a07edd2c5a"
>I: [pulseaudio] source-output.c:     application.process.session_id = "1"
>I: [pulseaudio] source-output.c:     application.icon_name = "audacity"
>I: [pulseaudio] source-output.c:     module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"
>D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
>D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
>I: [pulseaudio] protocol-native.c: Final latency 49.98 ms = 24.99 ms + 24.99 ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 24.99ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348364
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=443
>D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 24.99ms
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=348364
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=443
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>I: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo idle for too long, suspending ...
>D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0004, suspending
>I: [alsa-sink-VT1708S Analog] alsa-sink.c: Device suspended...
>D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
>D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_14.2.
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0
>D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87533
>D: [alsa-source-VT1708S Analog] alsa-source.c: Requested volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Got hardware volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
>D: [alsa-source-VT1708S Analog] alsa-source.c: Calculated software volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB (accurate-enough=yes)
>D: [alsa-source-VT1708S Analog] source.c: Volume not changing
>D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
>D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
>I: [pulseaudio] source-output.c: Freeing output 50 "ALSA Capture"
>I: [pulseaudio] client.c: Freed 96 "ALSA plug-in [audacity]"
>I: [pulseaudio] protocol-native.c: Connection died.
>I: [pulseaudio] main.c: Got signal SIGINT.
>I: [pulseaudio] main.c: Exiting.
>I: [pulseaudio] main.c: Daemon shutdown initiated.
>I: [pulseaudio] module.c: Unloading "module-filter-apply" (index: #21).
>I: [pulseaudio] module.c: Unloaded "module-filter-apply" (index: #21).
>I: [pulseaudio] module.c: Unloading "module-filter-heuristics" (index: #20).
>I: [pulseaudio] module.c: Unloaded "module-filter-heuristics" (index: #20).
>I: [pulseaudio] module.c: Unloading "module-role-cork" (index: #19).
>I: [pulseaudio] module.c: Unloaded "module-role-cork" (index: #19).
>I: [pulseaudio] module.c: Unloading "module-position-event-sounds" (index: #18).
>I: [pulseaudio] module.c: Unloaded "module-position-event-sounds" (index: #18).
>I: [pulseaudio] module.c: Unloading "module-systemd-login" (index: #17).
>D: [pulseaudio] module-systemd-login.c: Removing session 1
>I: [pulseaudio] client.c: Freed 0 "Login Session 1"
>I: [pulseaudio] module.c: Unloaded "module-systemd-login" (index: #17).
>I: [pulseaudio] module.c: Unloading "module-suspend-on-idle" (index: #16).
>I: [pulseaudio] module.c: Unloaded "module-suspend-on-idle" (index: #16).
>I: [pulseaudio] module.c: Unloading "module-intended-roles" (index: #15).
>I: [pulseaudio] module.c: Unloaded "module-intended-roles" (index: #15).
>I: [pulseaudio] module.c: Unloading "module-always-sink" (index: #14).
>I: [pulseaudio] module.c: Unloaded "module-always-sink" (index: #14).
>I: [pulseaudio] module.c: Unloading "module-rescue-streams" (index: #13).
>I: [pulseaudio] module.c: Unloaded "module-rescue-streams" (index: #13).
>I: [pulseaudio] module.c: Unloading "module-default-device-restore" (index: #12).
>I: [pulseaudio] module.c: Unloaded "module-default-device-restore" (index: #12).
>I: [pulseaudio] module.c: Unloading "module-native-protocol-unix" (index: #11).
>I: [pulseaudio] module.c: Unloaded "module-native-protocol-unix" (index: #11).
>I: [pulseaudio] module.c: Unloading "module-esound-protocol-unix" (index: #10).
>I: [pulseaudio] module.c: Unloaded "module-esound-protocol-unix" (index: #10).
>I: [pulseaudio] module.c: Unloading "module-bluez5-discover" (index: #9).
>I: [pulseaudio] module.c: Unloaded "module-bluez5-discover" (index: #9).
>I: [pulseaudio] module.c: Unloading "module-bluetooth-discover" (index: #8).
>I: [pulseaudio] module.c: Unloaded "module-bluetooth-discover" (index: #8).
>I: [pulseaudio] module.c: Unloading "module-bluetooth-policy" (index: #7).
>I: [pulseaudio] module.c: Unloaded "module-bluetooth-policy" (index: #7).
>I: [pulseaudio] module.c: Unloading "module-alsa-card" (index: #6).
>D: [alsa-sink-VT1708S Analog] alsa-sink.c: Thread shutting down
>I: [pulseaudio] sink.c: Freeing sink 0 "alsa_output.pci-0000_00_14.2.analog-stereo"
>I: [pulseaudio] source.c: Freeing source 0 "alsa_output.pci-0000_00_14.2.analog-stereo.monitor"
>D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
>D: [alsa-source-VT1708S Analog] alsa-source.c: Thread shutting down
>I: [pulseaudio] source.c: Freeing source 1 "alsa_input.pci-0000_00_14.2.analog-stereo"
>I: [pulseaudio] card.c: Freed 0 "alsa_card.pci-0000_00_14.2"
>D: [pulseaudio] ratelimit.c: 127 events suppressed
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry)
>I: [pulseaudio] module.c: Unloaded "module-alsa-card" (index: #6).
>I: [pulseaudio] module.c: Unloading "module-udev-detect" (index: #5).
>I: [pulseaudio] module.c: Unloaded "module-udev-detect" (index: #5).
>I: [pulseaudio] module.c: Unloading "module-switch-on-port-available" (index: #4).
>I: [pulseaudio] module.c: Unloaded "module-switch-on-port-available" (index: #4).
>I: [pulseaudio] module.c: Unloading "module-augment-properties" (index: #3).
>I: [pulseaudio] module.c: Unloaded "module-augment-properties" (index: #3).
>I: [pulseaudio] module.c: Unloading "module-card-restore" (index: #2).
>I: [pulseaudio] module.c: Unloaded "module-card-restore" (index: #2).
>I: [pulseaudio] module.c: Unloading "module-stream-restore" (index: #1).
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 removed from object /org/pulseaudio/stream_restore1
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry0
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry1
>D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry2
>I: [pulseaudio] module.c: Unloaded "module-stream-restore" (index: #1).
>I: [pulseaudio] module.c: Unloading "module-device-restore" (index: #0).
>I: [pulseaudio] module.c: Unloaded "module-device-restore" (index: #0).
>I: [pulseaudio] main.c: Daemon terminated.

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Attachments on bug 76328: 96017

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