<html> <head> <meta content="text/html; charset=utf-8" http-equiv="Content-Type"> </head> <body bgcolor="#FFFFFF" text="#000000"> <div class="moz-cite-prefix"> Thanks all, Finally I`m using this way to get the available frames: static int32_t audio_pcm_avail(ni_mediahal_audio_dev_type type) { snd_pcm_sframes_t av1 = snd_pcm_avail(pcm); snd_pcm_uframes_t av2 = 0; snd_htimestamp_t ht; int err = snd_pcm_htimestamp(pcm, &av2, &ht); struct timespec t; clock_gettime(CLOCK_MONOTONIC, &t); int64_t curTime = (int64_t) (t.tv_sec) * 1000000000LL + t.tv_nsec; int64_t alsaTime = (int64_t) (ht.tv_sec) * 1000000000LL + ht.tv_nsec; //LOG_AUDIOIO("av1=%d, av2=%u, curTime=%lld, alsaTime=%lld err=%d", av1, av2, curTime, alsaTime, err); double timeDiff = (double)(curTime - alsaTime) * 0.000000001; int32_t diffFrames = (int32_t)(timeDiff * audio_dev_params[type].rate); //LOG_AUDIOIO("timeDiff=%g, frameDiff=%d, rate=%d", timeDiff, diffFrames, audio_dev_params[type].rate); if (diffFrames >= (int)audio_dev_params[type].period_size) { diffFrames = 0; } return av1 + diffFrames; } get the diff time of current time and alsa timestamp, than caculate the diff frames with sample rate. 在 2015年06月10日 19:42, Raymond Yau 写道: </div> <blockquote cite="mid:CAN8cciZCYOSxLjzNiHryQu3dzFtOmbdX5t_7bbTd1MkZHCFfHQ@mail.gmail.com" type="cite"> >> > >> > >> >> > >> >> > >> > >> > >> > 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. >> > >> >> > >> If your sound card only increment hw_ptr only at interrupt occur, you >> > >> need to increase default_rewind_safeguard from 256 bytes to your >> > >> selected period size >> > > >> > > >> > > No. PulseAudio, in timer-scheduling mode, does not use periods at all. You need to change the driver so that it reports SNDRV_PCM_INFO_BATCH, so that PulseAudio does not try to use this mode. >> > > >> > > >> > >> >> > >> This mean that your sound card won't work with timer scheduling or >> > >> dynamic latency, you can only archieve low latency by decrease period size >> > >> Why do pulseaudio enable timer scheduling when most sound card use IRQ ? >> > > >> > > >> > > Because most broken sound cards driver authors forget to report SNDRV_PCM_INFO_BATCH? >> > >> > Why pulseaudio rely on the flag if your program can find out the granulatity ? >> >> AFAIK, there isn't a way to figure out granularity. Having this would be nice as we could be more intelligent about our tsched behaviour. > > > There is not only no way to query granularity, in some cases it is simply unknown. As for my approach (of measuring it directly), I currently think (but do not insist) that it is not suitable for inclusion into PulseAudio, because it is based on using a silent "test sound", busy-looping and repeatedly querying the position until it plays out. This would be unreliable if there is an unrelated CPU usage spike, and I think that busy-looping in general is not welcome. <a moz-do-not-send="true" href="https://bugs.freedesktop.org/show_bug.cgi?id=86262#c19">https://bugs.freedesktop.org/show_bug.cgi?id=86262#c19</a> Seem hwptr of snd-usb-audio are not that bad around 240 to 288 frames (less than period size) but not as good as snd-hda-intel 32 frames or oxygen 8 frames How accurate do pulseaudio need to use timer base scheduling ? <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>