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audio: strictly align audio on spdif frames

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We still have a sample-based buffer between filters and audio outputs.
In order to avoid cutting frames into half (which can upset receivers),
we strictly need to align the boundaries on which we cut the audio.
This commit is contained in:
wm4 2015-11-04 16:47:11 +01:00
parent 3ad03f6673
commit db1fc3628f
1 changed files with 7 additions and 3 deletions
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10
player/audio.c
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@ -485,7 +485,7 @@ static bool get_sync_samples(struct MPContext *mpctx, int *skip)
} }
int align = af_format_sample_alignment(out_format.format); int align = af_format_sample_alignment(out_format.format);
*skip = (-ptsdiff * play_samplerate) / align * align; *skip = (int)(-ptsdiff * play_samplerate) / align * align;
return true; return true;
} }
@ -538,6 +538,7 @@ void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
struct mp_audio out_format = {0}; struct mp_audio out_format = {0};
ao_get_format(mpctx->ao, &out_format); ao_get_format(mpctx->ao, &out_format);
double play_samplerate = out_format.rate / mpctx->audio_speed; double play_samplerate = out_format.rate / mpctx->audio_speed;
int align = af_format_sample_alignment(out_format.format);
// If audio is infinitely fast, somehow try keeping approximate A/V sync. // If audio is infinitely fast, somehow try keeping approximate A/V sync.
if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) && if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) &&
@ -563,7 +564,6 @@ void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
mpctx->audio_status == STATUS_PLAYING) mpctx->audio_status == STATUS_PLAYING)
{ {
int samples = ceil(opts->sync_audio_drop_size * play_samplerate); int samples = ceil(opts->sync_audio_drop_size * play_samplerate);
int align = af_format_sample_alignment(out_format.format);
samples = (samples + align / 2) / align * align; samples = (samples + align / 2) / align * align;
skip_duplicate = mpctx->last_av_difference >= 0 ? -samples : samples; skip_duplicate = mpctx->last_av_difference >= 0 ? -samples : samples;
@ -573,6 +573,8 @@ void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
mpctx->audio_drop_throttle += 1 - drop_limit - samples / play_samplerate; mpctx->audio_drop_throttle += 1 - drop_limit - samples / play_samplerate;
} }
playsize = playsize / align * align;
int status = AD_OK; int status = AD_OK;
bool working = false; bool working = false;
if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) { if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
@ -664,7 +666,7 @@ void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
double samples = (endpts - written_audio_pts(mpctx) - opts->audio_delay) double samples = (endpts - written_audio_pts(mpctx) - opts->audio_delay)
* play_samplerate; * play_samplerate;
if (playsize > samples) { if (playsize > samples) {
playsize = MPMAX(samples, 0); playsize = MPMAX((int)samples / align * align, 0);
audio_eof = true; audio_eof = true;
partial_fill = true; partial_fill = true;
} }
@ -684,6 +686,8 @@ void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
struct mp_audio data; struct mp_audio data;
mp_audio_buffer_peek(mpctx->ao_buffer, &data); mp_audio_buffer_peek(mpctx->ao_buffer, &data);
if (audio_eof || data.samples >= align)
data.samples = data.samples / align * align;
data.samples = MPMIN(data.samples, mpctx->paused ? 0 : playsize); data.samples = MPMIN(data.samples, mpctx->paused ? 0 : playsize);
int played = write_to_ao(mpctx, &data, playflags); int played = write_to_ao(mpctx, &data, playflags);
assert(played >= 0 && played <= data.samples); assert(played >= 0 && played <= data.samples);