<html> <head> <meta content="text/html; charset=utf-8" http-equiv="Content-Type"> </head> <body bgcolor="#FFFFFF" text="#000000"> <div class="moz-cite-prefix">Hi below is what the terminate shows when running pcm_avail.c uid=0 gid=1007@nutshell:/ # alsactl_test min_period_size: 8 frames, dir: 0 Playback hwparams: FIFO size is 8 Hardware PCM card 0 'rsnd-dai.0-dirana3.0' 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 : 1073741824 appl_ptr : 0 hw_ptr : 0 Playing silence Available: 0, loop iteration: 0 Available: 1024, loop iteration: 1469 Available: 2048, loop iteration: 5609 Available: 3072, loop iteration: 9667 All I got is just the 4 lines. BR, Lixin 在 2015年06月09日 11:38, Raymond Yau 写道: </div> <blockquote cite="mid:CAN8cciZ7fsgFxzrQTHs_6cOMERsU5kzTXkFHF-WCgjrRV8jDqg@mail.gmail.com" type="cite"> > > I found some audio noise problem when I trying to set the sink latency to a lower value. > > here is the alsa dump: > > D/NMAudio ( 1959): Its setup is: > D/NMAudio ( 1959): stream : PLAYBACK > D/NMAudio ( 1959): access : MMAP_INTERLEAVED > D/NMAudio ( 1959): format : S16_LE > D/NMAudio ( 1959): subformat : STD > D/NMAudio ( 1959): channels : 2 > D/NMAudio ( 1959): rate : 22050 > D/NMAudio ( 1959): exact rate : 22050 (22050/1) > D/NMAudio ( 1959): msbits : 16 > D/NMAudio ( 1959): buffer_size : 8192 > D/NMAudio ( 1959): period_size : 2048 > D/NMAudio ( 1959): period_time : 92879 > D/NMAudio ( 1959): tstamp_mode : ENABLE > D/NMAudio ( 1959): period_step : 1 > D/NMAudio ( 1959): avail_min : 7751 > D/NMAudio ( 1959): period_event : 0 > D/NMAudio ( 1959): start_threshold : -1 > D/NMAudio ( 1959): stop_threshold : 1073741824 > D/NMAudio ( 1959): silence_threshold: 0 > D/NMAudio ( 1959): s > > here is the log when problem happened: <a moz-do-not-send="true" href="https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/linux/dmaengine.h">https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/linux/dmaengine.h</a> * @DMA_RESIDUE_GRANULARITY_SEGMENT: Residue is updated after each successfully * completed segment of the transfer (For cyclic transfers this is after each * period). This is typically implemented by having the hardware generate an * interrupt after each transferred segment and then the drivers updates the * outstanding residue by the size of the segment. Another possibility is if * the hardware supports scatter-gather and the segment descriptor has a field * which gets set after the segment has been completed. The driver then counts * the number of segments without the flag set to compute the residue. * @DMA_RESIDUE_GRANULARITY_BURST: Residue is updated after each transferred * burst. This is typically only supported if the hardware has a progress * register of some sort (E.g. a register with the current read/write address * or a register with the amount of bursts/beats/bytes that have been * transferred or still need to be transferred). */ How accurate can you sound card hw_ptr increase ? period size or DMA brust size ? <a moz-do-not-send="true" href="http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081501.html">http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081501.html</a> you can try Alexander's pcm_avail.c <fieldset class="mimeAttachmentHeader"></fieldset> <pre wrap="">_______________________________________________ pulseaudio-discuss mailing list <a class="moz-txt-link-abbreviated" href="mailto:pulseaudio-discuss@lists.freedesktop.org">pulseaudio-discuss@lists.freedesktop.org</a> <a class="moz-txt-link-freetext" href="http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss">http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss</a> </pre> </blockquote> </body> </html>