[pulseaudio-discuss] Need help to reduce delay with USB, mic using pulseaudio

ABDUL MOIZ abdulmoiz786 at gmail.com
Thu Aug 25 02:59:43 PDT 2011


Hi,


On Wed, Aug 24, 2011 at 11:11 PM, Pierre-Louis Bossart <
pierre-louis.bossart at linux.intel.com> wrote:

> > 3) In the latency reported by module-loopback.c, the second number
> > (i.e.
> > the internal jitter buffer) dominates, both in USB => same USB and USB
> > => onboard audio cases: 63.91+733.32+11.65 ms
> >
> > So, for me, it looks like the USB master clock (from which the USB
> > audio
> > clock is derived) is too unstable WRT the system timer as a reference,
> > but this doesn't apply to the "internal audio vs system timer"
> > situation.
>
> Nah. It's more a configuration issue. I tried with pactl load-module
> module-loopback latency_msec=20 and I get a really low latency. Can you
> play
> with this latency_msec parameter and report the log?
> Thanks,
> -Pierre
>

           We have tried with the below command

          $ pactl load-module module-loopback latency_msec=20

         But still delay problem is there. (we have also tried with
latency_msec=10,30,40,100)
         you can find the attached log file

   Regards
   Abdul Moiz


>
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> pulseaudio-discuss at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
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I: client.c: Created 7 "Native client (UNIX socket client)"
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
I: module-stream-restore.c: Restoring mute state for sink input sink-input-by-media-role:abstract.
I: alsa-sink.c: Trying resume...
I: alsa-util.c: cannot disable ALSA period wakeups
I: alsa-util.c: ALSA period wakeups were not disabled
I: alsa-sink.c: Resumed successfully...
I: alsa-sink.c: Starting playback.
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: sink-input.c: Created input 1 "Loopback from AK4571                    Analog Mono" on alsa_output.platform-soc-audio.1.analog-stereo with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink-input.c:     media.name = "Loopback from AK4571                    Analog Mono"
I: sink-input.c:     media.role = "abstract"
I: sink-input.c:     media.icon_name = "audio-card-usb"
I: sink-input.c:     module-stream-restore.id = "sink-input-by-media-role:abstract"
I: alsa-source.c: Trying resume...
I: alsa-util.c: cannot disable ALSA period wakeups
I: alsa-util.c: ALSA period wakeups were not disabled
I: alsa-source.c: Resumed successfully...
I: alsa-source.c: Starting capture.
I: resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
I: remap.c: Using mono to stereo remapping
I: resampler.c: Using resampler 'copy'
I: resampler.c: Using s16le as working format.
I: source-output.c: Created output 2 "Loopback to PandaHDMI Analog Stereo" on alsa_input.usb-AKM_AK4571-00-default.analog-mono with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source-output.c:     media.name = "Loopback to PandaHDMI Analog Stereo"
I: source-output.c:     media.role = "abstract"
I: source-output.c:     media.icon_name = "audio-card"
I: source-output.c:     module-stream-restore.id = "source-output-by-media-role:abstract"
I: module-loopback.c: Max request changed
I: module-loopback.c: Max request changed
I: module-loopback.c: Skipping 0 bytes
I: module.c: Loaded "module-loopback" (index: #22; argument: "latency_msec=20").
I: client.c: Freed 7 "pactl"
I: protocol-native.c: Connection died.
I: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.platform-soc-audio.1.analog-stereo.
I: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.platform-soc-audio.1.analog-stereo.
I: module-device-restore.c: Synced.
I: module-loopback.c: New rate of 47237 Hz not within 2? of 44051 Hz, forcing smaller adjustment
I: module-loopback.c: New rate of 47246 Hz not within 2? of 44139 Hz, forcing smaller adjustment



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