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Rework on audio buffer management

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Use function to get samples to be sent to fill audio buffers to a
level. This replaces the function that only shows how much data
is in the buffer.

This way the function itself can control how much data will be
sent.
This commit is contained in:
Andreas Eversberg
2017-03-04 06:35:38 +01:00
parent 9d0e6b82b7
commit 4201717f36
11 changed files with 56 additions and 44 deletions
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22
src/common/sound_alsa.c
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@@ -323,8 +323,10 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE)
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
@@ -363,8 +365,10 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
return 0;
PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
/* recover read */
if (rc == -EPIPE)
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
@@ -391,14 +395,15 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
}
/*
* get playback buffer fill
* get playback buffer space
*
* return number of frames */
int sound_get_inbuffer(void *inst)
* return number of samples to be sent */
int sound_get_tosend(void *inst, int latspl)
{
sound_t *sound = (sound_t *)inst;
int rc;
snd_pcm_sframes_t delay;
int tosend;
rc = snd_pcm_delay(sound->phandle, &delay);
if (rc < 0) {
@@ -406,12 +411,15 @@ int sound_get_inbuffer(void *inst)
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency and enable real time scheduling\n");
else
PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE)
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
return delay;
tosend = latspl - delay;
return tosend;
}
int sound_is_stereo_capture(void *inst)