Created attachment 102011 [details]
more /etc/pulse/*

Created attachment 102012 [details]
full gdb backtrace

Created attachment 102013 [details]
asound.conf

as you are customize pulseaudio to use dmix as default sink



http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076475.html

you may need to use Alexander 's program whether dmix plugin should return zero for snd_pcm_rewindable and should alsa lib should return error when disable period wake-up when using dmix

strange that stop threshold is zero when pulseaudio using dmix


aplay -v -D plug:dmix Foo.wav

(In reply to comment #4)

As the configuration is now, and while pulseaudio is running `aplay -v -D plug:dmix Foo.wav` plays.
> as you are customize pulseaudio to use dmix as default sink

I added the following block and changed device="default" to device="dmixer",
pcm.dmixer {
    type dmix
        ipc_key 1024
        ipc_key_add_uid false
        ipc_perm 0660
        slave {
            pcm "hw:0,0"
            period_time 0
                period_size 1024
                buffer_size 8192
                rate 44100
                channels 2
        }
}

Now I get:
ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.0.27.2/work/alsa-lib-1.0.27.2/src/pcm/pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
aplay: main:722: audio open error: Device or resource busy

> http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076475.html
> 
> you may need to use Alexander 's program whether dmix plugin should return
> zero for snd_pcm_rewindable and should alsa lib should return error when
> disable period wake-up when using dmix

I am not sure how to obtain Alexander's program or how to check for these values.

Created attachment 102024 [details]
pulseaudio with dmixer as default

http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076480.html

does your sound card suspend and resume normally when not using pulseaudio ?

aplay -D hw:0,0 stereo.wav

aplay -D plughw:0,0 stereo.wav

aplay -D plug:dmix stereo.wav

refer to your log

dmix usually set stop threshold to boundary 


(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   start_threshold  : 1
(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   stop_threshold   : 0
(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   silence_threshold: 0
(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   silence_size : 0

(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   boundary     : 4611686018427387904
(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   appl_ptr     : 0
(   0.189|   0.000) D: [pulseaudio] alsa-util.c:   hw_ptr       : 8138786
(   0.189|   0.000) D: [alsa-sink-ALC887 Analog] alsa-sink.c: Thread starting up
(   0.197|   0.007) I: [alsa-sink-ALC887 Analog] core-util.c: Failed to acquire real-time scheduling: No such file or directory
(   0.198|   0.000) I: [alsa-sink-ALC887 Analog] alsa-sink.c: Starting playback.


hwptr already ahead of appl_ptr before pulseaudio call snd_pcm_start






his program pcm_rewindable.c want to check when did hw ptr update  by writing square wave and rewind in periods

(In reply to comment #7)
> does your sound card suspend and resume normally when not using pulseaudio ?
> 
> aplay -D hw:0,0 stereo.wav
> 
> aplay -D plughw:0,0 stereo.wav
> 
> aplay -D plug:dmix stereo.wav

These commands work fine.

> his program pcm_rewindable.c want to check when did hw ptr update  by
> writing square wave and rewind in periods

I ran it with the above devices and I'm attaching the output.

Created attachment 102059 [details]
pcm_rewindable.c output

 
you can change dmix format to use S16_LE instead of 32 bits 

http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/alsa.conf;hb=HEAD

defaults.pcm.dmix.format 


http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=614a66bb2a94ee64e37935fed7a0adc1e822dc2e

http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=5256e150eb34cf599e79839feaff7398ed67a499

there are patches which remove support of rewind from rate plugin

(   0.187|   0.000) I: [pulseaudio] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
(   0.187|   0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) failed: Invalid argument
(   0.187|   0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) failed: Invalid argument
(   0.188|   0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) failed: Invalid argument
(   0.188|   0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) failed: Invalid argument
(   0.188|   0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 341 ms
(   0.188|   0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 96000 samples), period size second (to 96000 samples).
(   0.188|   0.000) I: [pulseaudio] alsa-util.c: Device default doesn't support 44100 Hz, changed to 48000 Hz.
(   0.188|   0.000) I: [pulseaudio] alsa-util.c: Device default doesn't support sample format s16le, changed to s32le.
(   0.188|   0.000) I: [pulseaudio] alsa-util.c: ALSA period wakeups disabled

you can change dmix default rate and format to reduce the coversion

http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=70b11d614d6f7c13ea8c3fe81e7b34430fba389d;hp=8f16428f9cb66026bb62fe67ae10a32240a16db4

if the logic of dmix_rewind is so complex, it it unliksly dmix_rewindable is so simple

http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=c88672d86fe713e8f049df895fc3b64c472fbf5d;hp=058dde8b7da6f7b725e4ef8b9b237f2a5c6ff01e#patch3

340|   0.021) D: [alsa-sink-ALC887 Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
(   0.361|   0.021) D: [alsa-sink-ALC887 Analog] alsa-sink.c: Cutting sleep time for the initial iterations by half.
(  49.612|  49.250) D: [alsa-source-USB Audio] alsa-util.c: Got POLLERR from ALSA
(  49.612|   0.000) W: [alsa-source-USB Audio] alsa-util.c: Got POLLNVAL from ALSA
(  49.612|   0.000) D: [alsa-source-USB Audio] alsa-util.c: PCM state is DISCONNECTED
(  49.612|   0.000) W: [alsa-source-USB Audio] alsa-util.c: Could not recover from POLLERR|POLLNVAL|POLLHUP with snd_pcm_prepare(): No such device
(  49.612|   0.000) I: [alsa-sink-ALC887 Analog] alsa-sink.c: Scheduling delay of 39174.92 ms > 9.21 ms, you might want to investigate this to improve latency...
(  49.612|   0.000) D: [alsa-sink-ALC887 Analog] alsa-util.c: Got POLLERR from ALSA
(  49.612|   0.000) D: [alsa-sink-ALC887 Analog] alsa-util.c: Got POLLOUT from ALSA
(  49.612|   0.000) D: [alsa-sink-ALC887 Analog] alsa-util.c: PCM state is SUSPENDED

seem USB audio disconnect

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