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:
>> >
>> > >>
>> > >>
>> > >> >
>> > >> > 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.
https://bugs.freedesktop.org/show_bug.cgi?id=86262#c19
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 ?
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