| Summary: | "Transistor" game makes every client playing at double speed | ||
|---|---|---|---|
| Product: | PulseAudio | Reporter: | rkfg <rkfg> |
| Component: | core | Assignee: | pulseaudio-bugs |
| Status: | RESOLVED MOVED | QA Contact: | pulseaudio-bugs |
| Severity: | normal | ||
| Priority: | medium | CC: | lennart |
| Version: | unspecified | ||
| Hardware: | x86-64 (AMD64) | ||
| OS: | Linux (All) | ||
| Whiteboard: | |||
| i915 platform: | i915 features: | ||
| Attachments: |
pulseaudio log
PA startup log ALSA time test ALSA test log Fixed PA log pcm_avail log ALSA info Pulse log pactl list log |
||
The issue seems to go away after I set the default sample spec to s32le (default is s16le). I also noticed that with s16le both Chrome and Transistor use "resample method: (null)" (according to pacmd list-sink-inputs), and with s32le it becomes "resample method: copy" for both. And no, Chrome isn't the culprit as it happens even when it's not running, I've tried. The speedup happens right after starting the game, I've run a song in Chrome and launched Transistor. First time it was ok in both Chrome and the game but on the second run the song immediately sped up to double speed. I have no idea how the sample spec (not the sample rate!) affects the speed but it's what it is. SBLive! seems not to support 44100 Hz as it switches to 48k by default and even if I set 44100 explicitly in the daemon.conf. Though here paman and pacmd disagree with each other, pacmd says "sample spec: s16le 2ch 48000Hz" and paman says "Default Sample Type: s16le 2ch 44100Hz". Mumble notification sounds doesn't crackle as they do on ALC889 @ 44100Hz so I believe pacmd is correct here. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/sound/pci/ca0106/ca0106_main.c static struct snd_pcm_hardware snd_ca0106_playback_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), .rate_min = 48000, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, .buffer_bytes_max = ((65536 - 64) * 8), .period_bytes_min = 64, .period_bytes_max = (65536 - 64), .periods_min = 2, .periods_max = 8, .fifo_size = 0, }; seem support both s16_le and s32_le but require SNDRV_PCM_INFO_SYNC_START post the pulseaudio verbose log and aplay -D hw:CARD=CA0106 --dump-hw-params any.wav > aplay -D hw:CARD=CA0106 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
HW Params of device "hw:CARD=CA0106":
--------------------
ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT: S16_LE S32_LE
SUBFORMAT: STD
SAMPLE_BITS: [16 32]
FRAME_BITS: [32 64]
CHANNELS: 2
RATE: [48000 192000]
PERIOD_TIME: (41 170667)
PERIOD_SIZE: [8 8192]
PERIOD_BYTES: [64 65472]
PERIODS: [2 8]
BUFFER_TIME: (83 341334)
BUFFER_SIZE: [16 16384]
BUFFER_BYTES: [64 65536]
TICK_TIME: ALL
--------------------
aplay: set_params:1239: Channels count non available
The pulseaudio log is attached to the first message. This is the debug level, AFAIK, it's the most verbose one.
how about aplay -D surround40:CARD=CA0106 --dump-hw-params any.wav aplay -D surround51:CARD=CA0106 --dump-hw-params any.wav pulseaudio does not support SND_PCM_ACCESS_MMAP_COMPLEX and fallback to use SND_PCM_ACCESS_RW_INTERLEAVED this force pulseaudio to use irq polling instead of timer scheduling for those surround devices http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa-util.c?id=045c1d602dcba57868845ba3270510593c39480f + if (!_use_mmap) + _use_tsched = FALSE; + your log did not contain the probing of your ca0106 and those errors alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77) and pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-November/022670.html https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1315212 static snd_pcm_uframes_t
snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
{
struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ca0106_pcm *epcm = runtime->private_data;
unsigned int ptr, prev_ptr;
int channel = epcm->channel_id;
int timeout = 10;
if (!epcm->running)
return 0;
prev_ptr = -1;
do {
ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
ptr = (ptr >> 3) * runtime->period_size;
ptr += bytes_to_frames(runtime,
snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel));
if (ptr >= runtime->buffer_size)
ptr -= runtime->buffer_size;
if (prev_ptr == ptr)
return ptr;
prev_ptr = ptr;
} while (--timeout);
dev_warn(emu->card->dev, "ca0106: unstable DMA pointer!\n");
return 0;
}
pointer callback return value from hardware register,
do the sound card report realtime accurate update poistion ?
http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/tests/alsa-time-test.c
try different fill rate
do the sound card report hw_ptr better than period size ?
Created attachment 111535 [details] PA startup log Here's the startup log. > aplay -D surround40:CARD=CA0106 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "surround40:CARD=CA0106": -------------------- ACCESS: MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [64 128] CHANNELS: 4 RATE: [48000 192000] PERIOD_TIME: (41 170667) PERIOD_SIZE: [8 8192] PERIOD_BYTES: [64 131072] PERIODS: [2 8] BUFFER_TIME: (83 341334) BUFFER_SIZE: [16 16384] BUFFER_BYTES: [128 262144] TICK_TIME: ALL -------------------- aplay: set_params:1239: Channels count non available > aplay -D surround51:CARD=CA0106 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "surround51:CARD=CA0106": -------------------- ACCESS: MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [96 192] CHANNELS: 6 RATE: [48000 192000] PERIOD_TIME: (41 170667) PERIOD_SIZE: [8 8192] PERIOD_BYTES: [96 196608] PERIODS: [2 8] BUFFER_TIME: (83 341334) BUFFER_SIZE: [16 16384] BUFFER_BYTES: [192 393216] TICK_TIME: ALL -------------------- aplay: set_params:1239: Channels count non available Created attachment 111536 [details]
ALSA time test
The output of alsa-time-test.c
It ends with:
a.out: alsa-time-test.c:244: main: Assertion `(unsigned) avail <= buffer_size' failed.
zsh: abort ./a.out > timetest.log
I think it's ok.
(In reply to rkfg from comment #7) > Created attachment 111536 [details] > ALSA time test > > The output of alsa-time-test.c > > It ends with: > a.out: alsa-time-test.c:244: main: Assertion `(unsigned) avail <= > buffer_size' failed. > zsh: abort ./a.out > timetest.log > > I think it's ok. how did you run the program ? the program is hardcoded to use 44100Hz , it should run continously without any error if fillrate is equal to period size if sound card can report hw_ptr with better grannularity, the program can still run continously when fillrate is smaller than period size (e.g. dma brust size ) (In reply to rkfg from comment #6) > Created attachment 111535 [details] > PA startup log > > Here's the startup log. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)control.c: Invalid CTL iec958:0 Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: Unable to attach to mixer iec958:0: No such file or directory Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Device or resource busy git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=src/conf/cards/CA0106.conf;hb=HEAD seem related to lock in iec958 playback when pulseaudio try to open iec958 for playback and capture need to use asym plugin if your ca0106 support both iec958 playback and capture }
type hooks
slave.pcm {
type hw
card $CARD
}
hooks.0 {
type ctl_elems
hook_args [
{
name "IEC958 Front Playback Volume"
index 0
lock true
preserve true
value [ 207 207 ] # Puts 0x30303030 in the Volume register. 0xff - 0x30 = 0xcf = 207
}
{
name "IEC958 Playback Switch"
lock true
preserve true
value 1
}
{
interface PCM
name "IEC958 Playback Default"
index 1
lock true
preserve true
optional true
value [ $AES0 $AES1 $AES2 $AES3 ]
}
{
# for compatibility with older drivers
name "IEC958 Playback Default"
index 1
lock true
preserve true
optional true
value [ $AES0 $AES1 $AES2 $AES3 ]
}
]
}
}
alternative is to force pulseaudio not to open iec958 for capture by removing all paths-input of iec958 from http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-sets/default.conf [Mapping iec958-stereo] device-strings = iec958:%f channel-map = left,right paths-input = iec958-stereo-input paths-output = iec958-stereo-output priority = 5 APLAY **** List of PLAYBACK Hardware Devices **** card 0: CA0106 [CA0106], device 0: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 1: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 2: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 3: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: CA0106 [CA0106], device 0: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 1: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 2: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 3: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 (In reply to Raymond from comment #8) > > how did you run the program ? > > the program is hardcoded to use 44100Hz , it should run continously without > any error if fillrate is equal to period size > > if sound card can report hw_ptr with better grannularity, the program can > still run continously when fillrate is smaller than period size (e.g. dma > brust size ) I did: gcc alsa-time-test.c -lasound ./a.out That's all. What exactly should I do now and what information do you need? http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html pcm.name { type asym # Asym PCM playback STR # Playback slave name # or playback { # Playback slave definition pcm STR # Slave PCM name # or pcm { } # Slave PCM definition } capture STR # Capture slave name # or capture { # Capture slave definition pcm STR # Slave PCM name # or pcm { } # Slave PCM definition } } pcm.NAME { type hooks # PCM with hooks slave STR # Slave name # or slave { # Slave definition pcm STR # Slave PCM name # or pcm { } # Slave PCM definition } hooks { ID STR # Hook name (see pcm_hook) # or ID { } # Hook definition (see pcm_hook) } } you need to put the current ca0106 iec958 slave pcm and hook into playback slave and create capture slave of asym plugin (In reply to Raymond from comment #14) > you need to put the current ca0106 iec958 slave pcm and hook into playback > slave and create capture slave of asym plugin Sorry, I don't understand. I'm not that familiar with low-level ALSA configs, I only did some basic operations with ~/.asoundrc What file exactly should I change and what should I write into it? /usr/share/alsa/cards/CA0106.conf Good, the file is there and it has those lines you mentioned in #10. Now how should I modify them? Could you provide a diff to apply? I don't really get what "to put the current ca0106 iec958 slave pcm and hook into playback slave and create capture slave of asym plugin" means. If the alternative method is effectively equal to ALSA hacks, I can go that way, it seems to be much easier.
>alternative is to force pulseaudio not to open iec958 for capture
>by removing all paths-input of iec958 from
Should I comment out the entire [Mapping iec958-stereo] section or just "paths-input = iec958-stereo-input"?
CA0106.pcm.iec958.0 {
@args [ CARD AES0 AES1 AES2 AES3 ]
@args.CARD {
type string
}
@args.AES0 {
type integer
}
@args.AES1 {
type integer
}
@args.AES2 {
type integer
}
@args.AES3 {
type integer
}
+ type asym
+ playback.pcm {
type hooks
slave.pcm {
type hw
card $CARD
}
hooks.0 {
type ctl_elems
hook_args [
{
name "IEC958 Front Playback Volume"
index 0
lock true
preserve true
value [ 207 207 ] # Puts 0x30303030 in the Volume register. 0xff - 0x30 = 0xcf = 207
}
{
name "IEC958 Playback Switch"
lock true
preserve true
value 1
}
{
interface PCM
name "IEC958 Playback Default"
index 1
lock true
preserve true
optional true
value [ $AES0 $AES1 $AES2 $AES3 ]
}
{
# for compatibility with older drivers
name "IEC958 Playback Default"
index 1
lock true
preserve true
optional true
value [ $AES0 $AES1 $AES2 $AES3 ]
}
]
}
+ }
+ capture.pcm {
+ type hw
+ card $CARD
+ }
}
for alsa-time-test.c
dev = argc > 1 ? argv[1] : "front:0";
cap = argc > 2 ? atoi(argv[2]) : 0;
fillrate = argc > 3 ? atoi(argv[3]) : 1;
there are three parameter which allow you to change
device
capture or playback
fillrate
most sound cards cannot use default fill rate 1
for hda-intel, the hw_ptr increase by 16/32 frames for each dma transfer, fill rate can be lower than the period size
Created attachment 111585 [details]
ALSA test log
I've applied those changes for the ALSA config. I ran the test as ./a.out front:0 0 4 > alsa-test.log and here's the output. I stopped it with Ctrl-C. When I set the fillrate less than 4, it stops with the assertion failure like before. Setting the fillrate to 1 with playback test mode (./a.out front:0 1 1 > alsa1-test.log) doesn't fail.
That aside, I don't understand how IEC capture is related to the playback issue though. The game doesn't use microphone.
those locks in hook plugin prevent you running aplay -D iec958:CARD=CA0106 stereo48000.wav and arecord -D iec958:CARD=CA0106 test.wav at same time those message "Cannot lock ctl elem" should disaapear when you can playback and capture iec958 at same time ? post pulseaudio verbose log http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081501.html you can try Alexander's pcm_avail.c to find out whether your ca0106 can report DMA_RESIDUE_GRANULARITY_BURST ? do the error alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77) appear when you fix the error "cannot lock ctl elem" as pulseaudio start the probing of those surround40, 51 and71 ? http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011913.html : module-alsa-sink.c: Using 4 fragments of size 8816 bytes, buffer time is 99.95ms D: module-alsa-sink.c: hwbuf_unused_frames=0 D: module-alsa-sink.c: setting avail_min=1 I: module-alsa-sink.c: Volume ranges from 0 to 31. I: module-alsa-sink.c: Volume ranges from -46.50 dB to 0.00 dB. I: alsa-util.c: ALSA device lacks separate volumes control for channel 'rear-left' I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Multi PCM D: alsa-util.c: Channel bindings: D: alsa-util.c: 0: slave 0, channel 0 D: alsa-util.c: 1: slave 0, channel 1 D: alsa-util.c: 2: slave 1, channel 0 D: alsa-util.c: 3: slave 1, channel 1 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : RW_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 4 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 4408 D: alsa-util.c: period_size : 1102 D: alsa-util.c: period_time : 24988 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1102 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : -1 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 1155530752 D: alsa-util.c: Slave #0: Hooks PCM D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 4408 D: alsa-util.c: period_size : 1102 D: alsa-util.c: period_time : 24988 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1102 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_thr D: module-alsa-sink.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 D: module-alsa-sink.c: Read hardware volume: 0: 87% 1: 87% 2: 87% 3: 87% I: module-alsa-sink.c: Starting playback. I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed the playback stream has been started and pcm state is running but pulseaudio still call snd_pcm_start() seem default sample 16 bits has no effect on format , pulseaudio prefer 32 bits Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] card.c: Created 2 "alsa_card.pci-0000_06_00.0" Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: cannot disable ALSA period wakeups Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: ALSA period wakeups were not disabled Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-sink.c: Successfully opened device front:0. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-sink.c: Successfully enabled timer-based scheduling mode. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] module-device-restore.c: Restoring port for sink sink:alsa_output.pci-0000_06_00.0.analog-stereo. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_06_00.0.analog-stereo: front-left: 65536 / 100%, front-right: 65536 / 100% Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] sink.c: Default and alternate sample rates are the same. Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] sink.c: Created sink 0 "alsa_output.pci-0000_06_00.0.analog-stereo" with sample spec s32le 2ch 48000Hz and channel map front-left,front-right Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] sink.c: alsa.resolution_bits = "32" Created attachment 111615 [details]
Fixed PA log
It now doesn't report "Cannot lock ctl elem".
Created attachment 111616 [details]
pcm_avail log
Here's the output of pcm_avail.c program.
(In reply to Raymond from comment #25) > seem default sample 16 bits has no effect on format , pulseaudio prefer 32 > bits No, it does. I have these options uncommented in my daemon.conf: default-sample-format = s32le default-sample-rate = 48000 I have them enabled for some time and PA startup logs were provided with them on. If that's wrong tell me and I'll provide logs without them. (In reply to rkfg from comment #27) > Created attachment 111616 [details] > pcm_avail log > > Here's the output of pcm_avail.c program. avail increment at 8 frames which is less than min period size 16 and period size 1024 this mean some tsched feature are supported hwbuf_unused can be set to non zero (e.g. multiple of 8 frames instead of period size) min_period_size: 16 frames, dir: 0 Playback hwparams: FIFO size is 0 Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 Playing silence Available: 64, loop iteration: 0 Available: 72, loop iteration: 8 Available: 80, loop iteration: 16 Available: 88, loop iteration: 24 Available: 96, loop iteration: 31 (In reply to rkfg from comment #26) > Created attachment 111615 [details] > Fixed PA log > > It now doesn't report "Cannot lock ctl elem". it is strange that pulseaudio still don't probe surround profiles pactl list should list those surround profiles similar to https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1299857 device.vendor.name = "Creative Labs" device.product.name = "CA0106 Soundblaster" device.string = "0" device.description = "CA0106 Soundblaster" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" profiles: output:analog-stereo: Analog Stereo Output (priority 6000) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (priority 6060) output:analog-stereo+input:iec958-stereo: Analog Stereo Output + Digital Stereo (IEC958) Input (priority 6055) output:analog-surround-40: Analog Surround 4.0 Output (priority 700) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (priority 760) output:analog-surround-40+input:iec958-stereo: Analog Surround 4.0 Output + Digital Stereo (IEC958) Input (priority 755) can you post the output of alsa-info.sh ? did it support 7.1 since there are only 4 jacks ? speaker-test -c 4 -t wav -D surround40:CARD=CA0106 speaker-test -c 6 -t wav -D surround51:CARD=CA0106 speaker-test -c 8 -t wav -D surround71:CARD=CA0106 Created attachment 111649 [details]
ALSA info
Here's the output of alsa-info.sh. The speaker test works, all three commands without errors. However, I only have 2 speakers so I hear "front left" and "front right".
analog source seem supported by pulseaudio http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/mixer/paths/analog-input.conf.common?id=67e3925795ad939f2c9acb3aa93122a18672fade pactl list do pulseaudio show four input ports : phone, mic, line in and aux SImple mixer control 'Analog Source',0 Capabilities: cenum Items: 'Phone' 'Mic' 'Line in' 'Aux' Item0: 'Mic' Simple mixer control 'CAPTURE feedback',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Simple mixer control 'Digital Source',0 Capabilities: cenum Items: 'IEC958 out' 'i2s mixer out' 'IEC958 in' 'i2s in' 'AC97 in' 'SRC out' Item0: 'i2s in' Simple mixer control 'Shared Mic/Line in',0 Capabilities: cenum Items: 'Line in' 'Mic in' Item0: 'Mic in' CA0106 use specific name, you may need to add them to http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths/analog-output.conf Simple mixer control 'Analog Center/LFE',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 207 [81%] [0.00dB] Front Right: Playback 207 [81%] [0.00dB] Simple mixer control 'Analog Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 207 [81%] [0.00dB] Front Right: Playback 207 [81%] [0.00dB] Simple mixer control 'Analog Rear',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 207 [81%] [0.00dB] Front Right: Playback 207 [81%] [0.00dB] Simple mixer control 'Analog Side',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 207 [81%] [0.00dB] Front Right: Playback 207 [81%] [0.00dB] Jan 01 18:52:27 homecomp pulseaudio[14807]: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo
seem cannot opem iec958 for input
can you specify device 0
+ capture.pcm {
+ type hw
+ card $CARD
+ device 0
+
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-sets/default.conf [Mapping iec958-stereo] device-strings = iec958:%f channel-map = left,right paths-input = iec958-stereo-input paths-output = iec958-stereo-output priority = 5 as pulseaudio already define iec958-stereo-input when there is iec958-stereo-output.conf why there is no iec958-stereo-input.conf http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths Created attachment 111662 [details] Pulse log (In reply to Raymond from comment #35) > can you specify device 0 > > + capture.pcm { > + type hw > + card $CARD > + device 0 > + Added this, here's the startup log. Created attachment 111663 [details]
pactl list log
Log of pactl list.
arecord -v -d 1 -D iec958:CARD=CA0106 -f dat -t wav --dump-hw-params test.wav device.description = "CA0106 Soundblaster (SB0410 SBLive! 24-bit) Analog Stereo" alsa.mixer_name = "CA0106" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Ports: analog-input-mic: Microphone (priority: 8700) analog-input-aux;input-microphone+input-microphone: Analog Input / Microphone / Microphone (priority: 8000) analog-input-aux;input-microphone+input-linein: Analog Input / Microphone / Line In (priority: 8000) analog-input-aux;input-linein+input-microphone: Analog Input / Line In / Microphone (priority: 8000) analog-input-aux;input-linein+input-linein: Analog Input / Line In / Line In (priority: 8000) analog-input-aux;input+input-microphone: Analog Input / Input / Microphone (priority: 8000) analog-input-aux;input+input-linein: Analog Input / Input / Line In (priority: 8000) Active Port: analog-input-mic Formats: pcm seem phone is still missing SImple mixer control 'Analog Source',0 Capabilities: cenum Items: 'Phone' 'Mic' 'Line in' 'Aux' Item0: 'Mic' Simple mixer control 'Line in',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 0 [0%] [-99999.99dB] Front Right: Capture 0 [0%] [-99999.99dB] Simple mixer control 'Mic',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 255 [100%] [24.00dB] Front Right: Capture 255 [100%] [24.00dB] Simple mixer control 'Phone',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 0 [0%] [-99999.99dB] Front Right: Capture 0 [0%] [-99999.99dB] Simple mixer control 'Aux',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 0 [0%] [-99999.99dB] Front Right: Capture 0 [0%] [-99999.99dB] Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "iec958:CARD=CA0106": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [32 64] CHANNELS: 2 RATE: [48000 192000] PERIOD_TIME: (41 170334) PERIOD_SIZE: [8 8176] PERIOD_BYTES: [64 32704] PERIODS: 2 BUFFER_TIME: (83 340667) BUFFER_SIZE: [16 16352] BUFFER_BYTES: [64 65408] TICK_TIME: ALL -------------------- Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 8160 period_size : 4080 period_time : 85000 tstamp_mode : NONE period_step : 1 avail_min : 4080 period_event : 0 start_threshold : 1 stop_threshold : 8160 silence_threshold: 0 silence_size : 0 boundary : 9187343239835811840 appl_ptr : 0 hw_ptr : 0 overrun!!! (at least 0.017 ms long) Status: state : XRUN trigger_time: 847573.10484226 tstamp : 847573.10498485 delay : 0 avail : 8160 avail_max : 8160 ^CAborted by signal Interrupt... does analog capture also overrun ? arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/ca0106/ca0106_mixer.c if your sb0410 only have four jacks and support 7.1 it need snd_ca0106_capture_line_in_side_out which change blue line in jack as side line out jack if (emu->details->i2c_adc == 1) { ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls); if (emu->details->gpio_type == 1) err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); else /* gpio_type == 2 */ err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu)); if (err < 0) return err; } It does:
> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
overrun!!! (at least 0.084 ms long)
^CAborted by signal Interrupt...
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/sound/pci/ca0106/ca0106_main.c GENERAL INFO: * Model: SB0410 * P17 Chip: CA0106-DAT * AC97 Codec: None * ADC: WM8775EDS (4 Channel) * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support) /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */ { .serial = 0x10061102, .name = "Live! 7.1 24bit [SB0410]", .gpio_type = 1, .i2c_adc = 1 } , BTW, my mic is connected to the blue jack, not the usual pink one. This is the only configuration I've found to be working, others result in noise or silence. The mic is a bit more quiet (not _almost silent_ as if it's connected to a Line-In jack, just not comfortable enough) than it was on the ALC889 internal card so I had to increase the volume via software means. There's no boost setting in the alsamixer but PulseAudio can handle that. (In reply to rkfg from comment #43) > It does: > > arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav > Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, > Stereo > overrun!!! (at least 0.084 ms long) > ^CAborted by signal Interrupt... https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_main.c?id=34fdeb2d07102e07ecafe79dec170bd6733f2e56 you have to report to the author if overrun occur do it work better if you specify buffer size power of two (eg. 1024, 2048 or 4096 arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size= 2048 test.wav It happens more likely with small buffers and sometimes doesn't happen with larger buffer. However, this behavior is not consistent: %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=2048 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.025 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=4096 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.023 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=8192 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.023 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=16384 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.028 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=32768 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.045 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=65536 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.039 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=131072 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.046 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=262144 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=262144 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.039 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=524288 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=524288 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.046 ms long) ^CAborted by signal Interrupt... %[homecomp]:[/tmp/test]> arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 --buffer-size=524288 test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo overrun!!! (at least 0.044 ms long) ^CAborted by signal Interrupt... Ok, what about the original issue? I don't quite get what we are doing here. It looks like we're debugging something completely unrelated, for example, I don't use the digital output like iec958 and this game doesn't capture sound so this is probably unrelated, too. I'm fine with improving ALSA but all this looks like other bugs and should probably be filed separately... Raymond tends to be side-tracked from time to time and ask questions which are related to your original issue. Also apologies for not having read the entire thread through. PulseAudio closes the audio stream five seconds after no applications are active, so that's probably when the issue goes back to normal too. The resampler changed from "null" to "copy" means that we have a resampler engine active for the stream, but all it does is to convert from s16le (as being input by Chrome) to s32le. You're probably hitting a driver issue, but I wonder how we can prove it. Also the fact that it seems to hit one application only is a bit strange, maybe it's related to the requested latency by that specific application or something. did your sb0410 have these two chips ? ADC: WM8775EDS (4 Channel) DAC: CS4382 if you look at cs4382 datasheet , it has three modes which can double or quad the speed for wm8775 which does not have any boost for mic, you have to ask the author who add support of mic line in switch since the datasheet does not mention any usage of input pins as mic https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers you have to file upstream bug for overrun when recording using arecord http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=1af088e39b75a0a0897c7036487b143e983cd423;hp=57b5076c30b3453ee843912c0aeb3df8dbee3f68 it is strange the pulseaudio no longer probe surround40 , 51, and 71 your log have no traces of surround21 too http://git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=test/chmap.c;hb=HEAD chmap -D surround40 query chmap -D surround40 get (In reply to Raymond from comment #49) > did your sb0410 have these two chips ? > > ADC: WM8775EDS (4 Channel) > DAC: CS4382 > My card looks exactly like this: http://www.ixbt.com/multimedia/creative-live!24bit/card-big.jpg Probably, it has all these chips. http://www.ixbt.com/multimedia/creative-live!24bit.shtml seem need specific jack to support 7.1 control.29 {
iface MIXER
name 'Phone Capture Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 255'
dbmin -9999999
dbmax 2400
dbvalue.0 -9999999
dbvalue.1 -9999999
}
}
control.30 {
iface MIXER
name 'Mic Capture Volume'
value.0 255
value.1 255
comment {
access 'read write'
type INTEGER
count 2
range '0 - 255'
dbmin -9999999
dbmax 2400
dbvalue.0 2400
dbvalue.1 2400
}
}
control.31 {
iface MIXER
name 'Line in Capture Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 255'
dbmin -9999999
dbmax 2400
dbvalue.0 -9999999
dbvalue.1 -9999999
}
}
control.32 {
iface MIXER
name 'Aux Capture Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 255'
dbmin -9999999
dbmax 2400
dbvalue.0 -9999999
dbvalue.1 -9999999
}
}
refer to table 11 wm8775 datasheet
0dB is at 0xcf and 24 dB at 0xff
Four Stereo ADC Inputs with Analogue Gain Adjust from +24dB to –21dB in 0.5dB Steps Digital Gain Adjust from -21.5dB to -103dB. table 10 of wm8775 mention that 0.5 dB per steps and min 0 is mute As max 255 is +24 dB , 0xcf is 0 dB , step 1 is -103 dB Source #1 State: SUSPENDED Name: alsa_input.pci-0000_06_00.0.analog-stereo Description: CA0106 Soundblaster (SB0410 SBLive! 24-bit) Analog Stereo Driver: module-alsa-card.c Sample Specification: s32le 2ch 48000Hz Channel Map: front-left,front-right Owner Module: 8 Mute: no Volume: front-left: 99957 / 153% / 11.00 dB, front-right: 99957 / 153% / 11.00 dB balance 0.00 Base Volume: 26090 / 40% / -24.00 dB Monitor of Sink: n/a Latency: 0 usec, configured 0 usec Flags: HARDWARE HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY seem chmap -Dsurround41:CARD=CA0106 query fail but chmap -Dsurround41:CARD=CA0106 get return channel map in ca0106.conf there is no definition of ca0106.pcm.surround41 just <confdir:pcm/surround21.conf> <confdir:pcm/surround41.conf> <confdir:pcm/surround50.conf> Do surround21 , surround41 appear in aplay -L static unsigned int i2c_adc_init[][2] = {
{ 0x17, 0x00 }, /* Reset */
{ 0x07, 0x00 }, /* Timeout */
{ 0x0b, 0x22 }, /* Interface control */
{ 0x0c, 0x22 }, /* Master mode control */
{ 0x0d, 0x08 }, /* Powerdown control */
{ 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */
{ 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */
{ 0x10, 0x7b }, /* ALC Control 1 */
{ 0x11, 0x00 }, /* ALC Control 2 */
{ 0x12, 0x32 }, /* ALC Control 3 */
{ 0x13, 0x00 }, /* Noise gate control */
{ 0x14, 0xa6 }, /* Limiter control */
{ 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */
seem default is linein and 0dB
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_mixer.c?id=fff36e472b4315df77513f4339c5c199c6aad28b static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); the scale only define step 1 is -103.50 dB and step size 0.5dB for unknown reason which pulseaudio did not probe surround40 pulseaudio still using timer base scheduling in your last log but it was strange tradtional mode was used in your first log 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.name = "CA0106" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.id = "ca0106" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.subdevice = "0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.subdevice_name = "subdevice #0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.device = "0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.card = "0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.card_name = "CA0106" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.long_card_name = "Live! 7.1 24bit [SB0410] at 0xcf00 irq 19" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.driver_name = "snd_ca0106" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.bus_path = "pci-0000:06:00.0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1c.3/0000:05:00.0/0000:06:00.0/sound/card0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.bus = "pci" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.vendor.id = "1102" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.vendor.name = "Creative Labs" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.product.id = "0007" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.product.name = "CA0106 Soundblaster (SB0410 SBLive! 24-bit)" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.string = "front:0" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.buffering.buffer_size = "65536" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.buffering.fragment_size = "32768" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.access_mode = "mmap+timer" Jan 02 20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: device.profile.name = "analog-stereo" Dec 25 19:33:27 homecomp pulseaudio[1911]: module-stream-restore.id = "sink-input-by-application-name:paplay" Dec 25 19:33:27 homecomp pulseaudio[1911]: Requested tlength=2000.00 ms, minreq=20.00 ms Dec 25 19:33:27 homecomp pulseaudio[1911]: Traditional mode enabled, modifying sink usec only for compat with minreq. Dec 25 19:33:27 homecomp pulseaudio[1911]: Requested latency=1960.00 ms, Received latency=341.33 ms Dec 25 19:33:27 homecomp pulseaudio[1911]: memblockq requested: maxlength=4194304, tlength=192000, base=2, prebuf=190082, minreq=1920 maxrewind=0 Dec 25 19:33:27 homecomp pulseaudio[1911]: memblockq sanitized: maxlength=4194304, tlength=192000, base=2, prebuf=190082, minreq=1920 maxrewind=0 Dec 25 19:33:27 homecomp pulseaudio[1911]: Final latency 2341.33 ms = 1960.00 ms + 2*20.00 ms + 341.33 ms Dec 25 19:33:27 homecomp pulseaudio[1911]: Latency set to 26.00ms Dec 25 19:33:27 homecomp pulseaudio[1911]: hwbuf_unused=60544 Dec 25 19:33:27 homecomp pulseaudio[1911]: setting avail_min=15617 the driver need to provide accurate hw_ptr position in order to set hwbuf_unused to non zero the actual buffer used = buffer size (65536) - hwbuf_unused (60544) is 5012 bytes (In reply to Raymond from comment #57) > Do surround21 , surround41 appear in > > aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default:CARD=CA0106 CA0106, CA0106 Default Audio Device sysdefault:CARD=CA0106 CA0106, CA0106 Default Audio Device front:CARD=CA0106,DEV=0 CA0106, CA0106 Front speakers rear:CARD=CA0106,DEV=0 CA0106, CA0106 Rear speakers center_lfe:CARD=CA0106,DEV=0 CA0106, CA0106 Center and Subwoofer speakers side:CARD=CA0106,DEV=0 CA0106, CA0106 Side speakers surround21:CARD=CA0106,DEV=0 CA0106, CA0106 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=CA0106,DEV=0 CA0106, CA0106 4.0 Surround output to Front and Rear speakers surround41:CARD=CA0106,DEV=0 CA0106, CA0106 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=CA0106,DEV=0 CA0106, CA0106 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=CA0106,DEV=0 CA0106, CA0106 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=CA0106,DEV=0 CA0106, CA0106 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=CA0106,DEV=0 CA0106, CA0106 IEC958 (S/PDIF) Digital Audio Output dmix:CARD=CA0106,DEV=0 CA0106, CA0106 Direct sample mixing device dmix:CARD=CA0106,DEV=1 CA0106, CA0106 Direct sample mixing device dmix:CARD=CA0106,DEV=2 CA0106, CA0106 Direct sample mixing device dmix:CARD=CA0106,DEV=3 CA0106, CA0106 Direct sample mixing device dsnoop:CARD=CA0106,DEV=0 CA0106, CA0106 Direct sample snooping device dsnoop:CARD=CA0106,DEV=1 CA0106, CA0106 Direct sample snooping device dsnoop:CARD=CA0106,DEV=2 CA0106, CA0106 Direct sample snooping device dsnoop:CARD=CA0106,DEV=3 CA0106, CA0106 Direct sample snooping device hw:CARD=CA0106,DEV=0 CA0106, CA0106 Direct hardware device without any conversions hw:CARD=CA0106,DEV=1 CA0106, CA0106 Direct hardware device without any conversions hw:CARD=CA0106,DEV=2 CA0106, CA0106 Direct hardware device without any conversions hw:CARD=CA0106,DEV=3 CA0106, CA0106 Direct hardware device without any conversions plughw:CARD=CA0106,DEV=0 CA0106, CA0106 Hardware device with all software conversions plughw:CARD=CA0106,DEV=1 CA0106, CA0106 Hardware device with all software conversions plughw:CARD=CA0106,DEV=2 CA0106, CA0106 Hardware device with all software conversions plughw:CARD=CA0106,DEV=3 CA0106, CA0106 Hardware device with all software conversions default:CARD=PCH HDA Intel PCH, ALC889 Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, ALC889 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Digital IEC958 (S/PDIF) Digital Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog Direct sample mixing device dmix:CARD=PCH,DEV=1 HDA Intel PCH, ALC889 Digital Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=1 HDA Intel PCH, ALC889 Digital Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=1 HDA Intel PCH, ALC889 Digital Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC889 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=1 HDA Intel PCH, ALC889 Digital Hardware device with all software conversions hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 3 HDMI Audio Output dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=7 HDA NVidia, HDMI 1 Direct sample mixing device dmix:CARD=NVidia,DEV=8 HDA NVidia, HDMI 2 Direct sample mixing device dmix:CARD=NVidia,DEV=9 HDA NVidia, HDMI 3 Direct sample mixing device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=7 HDA NVidia, HDMI 1 Direct sample snooping device dsnoop:CARD=NVidia,DEV=8 HDA NVidia, HDMI 2 Direct sample snooping device dsnoop:CARD=NVidia,DEV=9 HDA NVidia, HDMI 3 Direct sample snooping device hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 1 Direct hardware device without any conversions hw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 2 Direct hardware device without any conversions hw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 3 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 1 Hardware device with all software conversions plughw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 2 Hardware device with all software conversions plughw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 3 Hardware device with all software conversions what is the smallest buffer size which aplay without underrun ? 1024,512,256,128,64,... aplay -D front:CARD=CA0106 -v --buffer-size=1024 stereo48000.wav 19:29:42 homecomp pulseaudio[1911]: Protocol version: remote 26, local 29 Dec 25 19:29:42 homecomp pulseaudio[1911]: Got credentials: uid=1000 gid=1000 success=1 Dec 25 19:29:42 homecomp pulseaudio[1911]: SHM possible: yes Dec 25 19:29:42 homecomp pulseaudio[1911]: Negotiated SHM: yes Dec 25 19:29:42 homecomp pulseaudio[1911]: Looking for .desktop file for Transistor.bin.x86_64 Dec 25 19:29:42 homecomp pulseaudio[1911]: Not setting device for stream ALSA Playback, because it lacks role. Dec 25 19:29:42 homecomp pulseaudio[1911]: Negotiated format: pcm, format.sample_format = "\"s16le\"" format.rate = "48000" format.channels = "2" format.channel_map = "\"front-left,front-right\"" Dec 25 19:29:42 homecomp pulseaudio[1911]: Sink alsa_output.pci-0000_06_00.0.analog-stereo becomes busy, resuming. Dec 25 19:29:42 homecomp pulseaudio[1911]: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 Dec 25 19:29:42 homecomp pulseaudio[1911]: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 Dec 25 19:29:42 homecomp pulseaudio[1911]: Created input 15 "ALSA Playback" on alsa_output.pci-0000_06_00.0.analog-stereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right Dec 25 19:29:42 homecomp pulseaudio[1911]: media.name = "ALSA Playback" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.name = "ALSA plug-in [Transistor.bin.x86_64]" Dec 25 19:29:42 homecomp pulseaudio[1911]: native-protocol.peer = "UNIX socket client" Dec 25 19:29:42 homecomp pulseaudio[1911]: native-protocol.version = "26" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.id = "9536" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.user = "rkfg" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.host = "homecomp" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.binary = "Transistor.bin.x86_64" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.language = "en_GB.UTF-8" Dec 25 19:29:42 homecomp pulseaudio[1911]: window.x11.display = ":0.0" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.machine_id = "3be8f38f291dd2003af500d051c0a849" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.session_id = "10" Dec 25 19:29:42 homecomp pulseaudio[1911]: module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [Transistor.bin.x86_64]" Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested tlength=85.33 ms, minreq=21.33 ms Dec 25 19:29:42 homecomp pulseaudio[1911]: Early requests mode enabled, configuring sink latency to minreq. Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested latency=21.33 ms, Received latency=26.00 ms Dec 25 19:29:42 homecomp pulseaudio[1911]: memblockq requested: maxlength=4194304, tlength=16384, base=4, prebuf=11396, minreq=4992 maxrewind=0 Dec 25 19:29:42 homecomp pulseaudio[1911]: memblockq sanitized: maxlength=4194304, tlength=16384, base=4, prebuf=11396, minreq=4992 maxrewind=0 Dec 25 19:29:42 homecomp pulseaudio[1911]: Final latency 111.33 ms = 33.33 ms + 2*26.00 ms + 26.00 ms Dec 25 19:29:42 homecomp pulseaudio[1911]: Latency set to 26.00ms Dec 25 19:29:42 homecomp pulseaudio[1911]: hwbuf_unused=60544 Dec 25 19:29:42 homecomp pulseaudio[1911]: setting avail_min=15617 Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested volume: front-left: 52559 / 80% / -5.75 dB, front-right: 52559 / 80% / -5.75 dB Dec 25 19:29:42 homecomp pulseaudio[1911]: Got hardware volume: front-left: 53066 / 81% / -5.50 dB, front-right: 53066 / 81% / -5.50 dB Dec 25 19:29:42 homecomp pulseaudio[1911]: Calculated software volume: front-left: 64910 / 99% / -0.25 dB, front-right: 64910 / 99% / -0.25 dB (accurate-enough=yes) Dec 25 19:29:42 homecomp pulseaudio[1911]: Volume not changing Dec 25 19:29:42 homecomp pulseaudio[1911]: Requesting rewind due to end of underrun. Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested to rewind 65536 bytes. Dec 25 19:29:42 homecomp pulseaudio[1911]: Limited to 3904 bytes. Dec 25 19:29:42 homecomp pulseaudio[1911]: before: 976 Dec 25 19:29:42 homecomp pulseaudio[1911]: after: 976 Dec 25 19:29:42 homecomp pulseaudio[1911]: Rewound 3904 bytes. Dec 25 19:29:42 homecomp pulseaudio[1911]: Processing rewind... Dec 25 19:29:42 homecomp pulseaudio[1911]: latency = 65877 Dec 25 19:29:42 homecomp pulseaudio[1911]: Have to rewind 3904 bytes on render memblockq. Dec 25 19:29:42 homecomp pulseaudio[1911]: Have to rewind 3904 bytes on render memblockq. Dec 25 19:29:42 homecomp pulseaudio[1911]: Have to rewind 3904 bytes on render memblockq. Dec 25 19:29:42 homecomp pulseaudio[1911]: Processing rewind... Dec 25 19:29:42 homecomp pulseaudio[1911]: Wakeup from ALSA! Dec 25 19:29:42 homecomp pulseaudio[1911]: Wakeup from ALSA! Dec 25 19:29:42 homecomp pulseaudio[1911]: Wakeup from ALSA! Dec 25 19:30:19 homecomp pulseaudio[1911]: Requesting rewind due to corking Dec 25 19:30:19 homecomp pulseaudio[1911]: Requested to rewind 65536 bytes. Dec 25 19:30:19 homecomp pulseaudio[1911]: Limited to 4544 bytes. Dec 25 19:30:19 homecomp pulseaudio[1911]: before: 1136 Dec 25 19:30:19 homecomp pulseaudio[1911]: after: 1136 Dec 25 19:30:19 homecomp pulseaudio[1911]: Rewound 4544 bytes. Dec 25 19:30:19 homecomp pulseaudio[1911]: Processing rewind... Dec 25 19:30:19 homecomp pulseaudio[1911]: latency = 1234 Dec 25 19:30:19 homecomp pulseaudio[1911]: Have to rewind 4544 bytes on render memblockq. Dec 25 19:30:19 homecomp pulseaudio[1911]: Have to rewind 4544 bytes on render memblockq. Dec 25 19:30:19 homecomp pulseaudio[1911]: Have to rewind 4544 bytes on render memblockq. Dec 25 19:30:19 homecomp pulseaudio[1911]: Have to rewind 4544 bytes on implementor. Dec 25 19:30:19 homecomp pulseaudio[1911]: Processing rewind... Dec 25 19:30:19 homecomp pulseaudio[1911]: Found underrun 4864 bytes ago (128 bytes ahead in playback buffer) Dec 25 19:30:19 homecomp pulseaudio[1911]: Found underrun 4864 bytes ago (64 bytes ahead in playback buffer) Dec 25 19:30:19 homecomp pulseaudio[1911]: Latency set to 26.00ms Dec 25 19:30:19 homecomp pulseaudio[1911]: hwbuf_unused=60544 Dec 25 19:30:19 homecomp pulseaudio[1911]: setting avail_min=15617 This command doesn't work, it says "aplay: set_params:1239: Channels count non available". The -D default:CARD=CA0106 variant works and it does not overrun. I only had overruns on recording (at least, explicit). That said, I've made a huge improvement in my system, I compiled a pf-kernel and set CONFIG_PREEMPT=y, CONFIG_HZ_1000=y and CONFIG_HZ=1000. The default Debian kernel is on 250 Hz and is not preemptible. Now that my kernel supports preemption I (crossing fingers) don't have ALSA over- and underruns anymore. They were quite sudden and audible as clicks/gaps. I saw this in the log: Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: Scheduling delay of 11.42 ms > 11.31 ms, you might want to investigate this to improve latency... Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: Underrun! Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: Increasing minimal latency to 1.00 ms Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: Latency set to 21.33ms Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: hwbuf_unused=57352 Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: setting avail_min=7651 Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: Latency set to 21.33ms Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: hwbuf_unused=57352 Jan 4 20:25:15 homecomp pulseaudio[26848]: [alsa-sink-ca0106] alsa-sink.c: setting avail_min=7651 With the preempt kernel this doesn't happen at all (I grepped the log) and sound is just as good as with pure ALSA. However, arecord still drops overrun for CA0106. It doesn't do it for PCH (ALC889) though. if you cannot find stereo 48000Hz wav file aplay -D plughw:CARD=CA0106 -v --buffer-size=1024 any.wav try to find minimum buffer size 1024,512, 256,128,64,... without underrun 1024 frames = 21.33 ms x 48000Hz 4096 = 85.33 ms x 48000Hz if request tlength 85.33ms is buffer time and 21.33ms is period time not sure why pulseaudio received latency is 26ms alsa_output.pci-0000_06_00.0.analog-stereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right Dec 25 19:29:42 homecomp pulseaudio[1911]: media.name = "ALSA Playback" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.name = "ALSA plug-in [Transistor.bin.x86_64]" Dec 25 19:29:42 homecomp pulseaudio[1911]: native-protocol.peer = "UNIX socket client" Dec 25 19:29:42 homecomp pulseaudio[1911]: native-protocol.version = "26" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.id = "9536" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.user = "rkfg" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.host = "homecomp" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.binary = "Transistor.bin.x86_64" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.language = "en_GB.UTF-8" Dec 25 19:29:42 homecomp pulseaudio[1911]: window.x11.display = ":0.0" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.machine_id = "3be8f38f291dd2003af500d051c0a849" Dec 25 19:29:42 homecomp pulseaudio[1911]: application.process.session_id = "10" Dec 25 19:29:42 homecomp pulseaudio[1911]: module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [Transistor.bin.x86_64]" Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested tlength=85.33 ms, minreq=21.33 ms Dec 25 19:29:42 homecomp pulseaudio[1911]: Early requests mode enabled, configuring sink latency to minreq. Dec 25 19:29:42 homecomp pulseaudio[1911]: Requested latency=21.33 ms, Received latency=26.00 ms # aplay -D plughw:CARD=CA0106 -v --buffer-size=128 /usr/share/sounds/alsa/Side_Left.wav
Playing WAVE '/usr/share/sounds/alsa/Side_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
Plug PCM: Route conversion PCM (sformat=S16_LE)
Transformation table:
0 <- 0
1 <- 0
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 1
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 128
period_size : 32
period_time : 666
tstamp_mode : NONE
period_step : 1
avail_min : 32
period_event : 0
start_threshold : 128
stop_threshold : 128
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Slave: Hardware PCM card 0 'CA0106' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 128
period_size : 32
period_time : 666
tstamp_mode : NONE
period_step : 1
avail_min : 32
period_event : 0
start_threshold : 128
stop_threshold : 128
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
appl_ptr : 0
hw_ptr : 0
128 seems to be fine. It's not consistent, sometimes 64 works ok, sometimes it overruns (says so in the console and hitches). 128 has never overrun, neither visually (saying it in the console), nor audible (clicks/pops).
Tried with a stereo 48k file (converted /usr/share/sounds/KDE-Im-Cant-Connect.ogg to wav, tested with ffprobe that it's truly 48k/2chan) and 64 bytes buffer size: # aplay -D plughw:CARD=CA0106 -v --buffer-size=64 /tmp/kde.wav Playing WAVE '/tmp/kde.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Plug PCM: Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 64 period_size : 16 period_time : 333 tstamp_mode : NONE period_step : 1 avail_min : 16 period_event : 0 start_threshold : 64 stop_threshold : 64 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 underrun!!! (at least 0.010 ms long) Status: state : XRUN trigger_time: 260136.459825170 tstamp : 260136.459834181 delay : 0 avail : 64 avail_max : 64 underrun!!! (at least 0.020 ms long) Status: state : XRUN trigger_time: 260136.460064991 tstamp : 260136.460084242 delay : 0 avail : 64 avail_max : 64 No underruns with 128b buffer. how about CD audio (44100Hz stereo) using plughw? Interesting. I've converted that file to 44100 Hz and now it reveals a strange behavior. When I play it with buffer size of 64 it sounds like the pitch is lower than it should be (even if it doesn't overrun though it does from time to time). When the size is 128 it sounds fine. # aplay -D plughw:CARD=CA0106 -v --buffer-size=64 /tmp/kde44100.wav Playing WAVE '/tmp/kde44100.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 58 period_size : 14 period_time : 333 tstamp_mode : NONE period_step : 1 avail_min : 14 period_event : 0 start_threshold : 58 stop_threshold : 58 silence_threshold: 0 silence_size : 0 boundary : 4179340454199820288 Slave: Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 64 period_size : 16 period_time : 333 tstamp_mode : NONE period_step : 1 avail_min : 16 period_event : 0 start_threshold : 64 stop_threshold : 64 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 underrun!!! (at least 0.116 ms long) Status: state : XRUN trigger_time: 301642.243139271 tstamp : 301642.243254519 delay : 2 avail : 56 avail_max : 56 # aplay -D plughw:CARD=CA0106 -v --buffer-size=128 /tmp/kde44100.wav Playing WAVE '/tmp/kde44100.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 117 period_size : 29 period_time : 666 tstamp_mode : NONE period_step : 1 avail_min : 29 period_event : 0 start_threshold : 117 stop_threshold : 117 silence_threshold: 0 silence_size : 0 boundary : 4215369251218784256 Slave: Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 128 period_size : 32 period_time : 666 tstamp_mode : NONE period_step : 1 avail_min : 32 period_event : 0 start_threshold : 128 stop_threshold : 128 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 This pitch (or maybe speed?) weirdness occurs only on playing stereo files with rate of 44100. I've tried playing music and it indeed sounds a bit transposed. Bufsize of 128 frames also seems to produce underruns after some time, my testing clip was too small to reveal it. So it required bufsize 512 to be played completely without underruns: # aplay -D plughw:CARD=CA0106 -v --buffer-size=512 /tmp/ksp44100.wav Playing WAVE '/tmp/ksp44100.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 529 period_size : 132 period_time : 3000 tstamp_mode : NONE period_step : 1 avail_min : 132 period_event : 0 start_threshold : 529 stop_threshold : 529 silence_threshold: 0 silence_size : 0 boundary : 4764808405757984768 Slave: Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 576 period_size : 144 period_time : 3000 tstamp_mode : NONE period_step : 1 avail_min : 144 period_event : 0 start_threshold : 576 stop_threshold : 576 silence_threshold: 0 silence_size : 0 boundary : 5188146770730811392 appl_ptr : 0 hw_ptr : 0 The same song converted to 48000 Hz plays without underruns with buffer size of 256 frames: Playing WAVE '/tmp/ksp48000.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Plug PCM: Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 256 period_size : 64 period_time : 1333 tstamp_mode : NONE period_step : 1 avail_min : 64 period_event : 0 start_threshold : 256 stop_threshold : 256 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 -- GitLab Migration Automatic Message -- This bug has been migrated to freedesktop.org's GitLab instance and has been closed from further activity. You can subscribe and participate further through the new bug through this link to our GitLab instance: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/308. |
Use of freedesktop.org services, including Bugzilla, is subject to our Code of Conduct. How we collect and use information is described in our Privacy Policy.
Created attachment 111331 [details] pulseaudio log The game uses MonoGame as the engine, it outputs audio via ALSA. Often, but not every time, when I start the game the sound is played at double speed. What's worse all other apps also start playing audio too fast, including paplay, Chrome etc. Even those which are started after I exit the game. This can be fixed by restarting PA or after killing all clients connections with paman (if there are none after I exit the game, sound seems to be working correctly). I believe no client should affect other clients like this, no matter how it's broken. A friend of mine completed Transistor without such issues using PA. I also found that something like this was long ago fixed in Wine but this game is native. At least more native than pure Win32 apps+Wine and it doesn't use Wine. It uses Mono though and FMOD as the sound engine. My soundcard is CA0106 Soundblaster (SB0410 SBLive! 24-bit), CPU is Intel Core i7-2600, 32Gb RAM, NVIDIA GeForce 770 GTX driver v340.65, Debian Jessie GNU/Linux amd64, Awesome WM 3.4.15, kernel v3.16. I also have another sound chip, ALC889 Analog. It's disabled in pavucontrol. You can also see some ALSA underruns, I couldn't fix it no matter what priorities and realtime priorities I set. They happen on both snd-hda-intel and snd-ca0106.