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audio: fix initial audio PTS 

Commit 5e25a3d2 broke handling of the initial frame (the one decoded
with initial_audio_decode()). It didn't update the pts_offset field,
leading to a shift in timestamps by one audio frame.

Fix by calling the actual decode function in a single place. This
requires slightly more changes than what would be necessary to fix the
bug, but it also somewhat simplifies the data flow.
This commit is contained in:
wm4 2015-01-14 22:14:46 +01:00
parent 1a522f2976
commit 4cabd08e8a
1 changed files with 26 additions and 25 deletions
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43
audio/decode/dec_audio.c
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@ -153,23 +153,33 @@ void audio_uninit(struct dec_audio *d_audio)
MP_VERBOSE(d_audio, "Uninit audio filters...\n"); MP_VERBOSE(d_audio, "Uninit audio filters...\n");
af_destroy(d_audio->afilter); af_destroy(d_audio->afilter);
uninit_decoder(d_audio); uninit_decoder(d_audio);
talloc_free(d_audio->waiting);
talloc_free(d_audio); talloc_free(d_audio);
} }
static int decode_new_frame(struct dec_audio *da)
{
while (!da->waiting) {
int ret = da->ad_driver->decode_packet(da, &da->waiting);
if (ret < 0)
return ret;
if (da->waiting) {
da->pts_offset += da->waiting->samples;
da->decode_format = *da->waiting;
mp_audio_set_null_data(&da->decode_format);
}
}
return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
}
/* Decode packets until we know the audio format. Then reinit the buffer. /* Decode packets until we know the audio format. Then reinit the buffer.
* Returns AD_OK on success, negative AD_* code otherwise. * Returns AD_OK on success, negative AD_* code otherwise.
* Also returns AD_OK if already initialized (and does nothing). * Also returns AD_OK if already initialized (and does nothing).
*/ */
int initial_audio_decode(struct dec_audio *da) int initial_audio_decode(struct dec_audio *da)
{ {
while (!da->waiting) { return decode_new_frame(da);
int ret = da->ad_driver->decode_packet(da, &da->waiting);
if (ret < 0)
return ret;
}
talloc_steal(da, da->waiting);
da->decode_format = *da->waiting;
return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
} }
static bool copy_output(struct af_stream *afs, struct mp_audio_buffer *outbuf, static bool copy_output(struct af_stream *afs, struct mp_audio_buffer *outbuf,
@ -208,31 +218,22 @@ int audio_decode(struct dec_audio *da, struct mp_audio_buffer *outbuf,
if (copy_output(afs, outbuf, minsamples, false)) if (copy_output(afs, outbuf, minsamples, false))
break; break;
struct mp_audio *mpa = da->waiting; res = decode_new_frame(da);
da->waiting = NULL;
if (!mpa) {
res = da->ad_driver->decode_packet(da, &mpa);
if (res < 0) { if (res < 0) {
// drain filters first (especially for true EOF case) // drain filters first (especially for true EOF case)
copy_output(afs, outbuf, minsamples, true); copy_output(afs, outbuf, minsamples, true);
break; break;
} }
if (!mpa)
continue;
da->pts_offset += mpa->samples;
da->decode_format = *mpa;
mp_audio_set_null_data(&da->decode_format);
}
// On format change, make sure to drain the filter chain. // On format change, make sure to drain the filter chain.
if (!mp_audio_config_equals(&afs->input, mpa)) { if (!mp_audio_config_equals(&afs->input, da->waiting)) {
da->waiting = talloc_steal(da, mpa);
copy_output(afs, outbuf, minsamples, true); copy_output(afs, outbuf, minsamples, true);
res = AD_NEW_FMT; res = AD_NEW_FMT;
break; break;
} }
struct mp_audio *mpa = da->waiting;
da->waiting = NULL;
if (af_filter_frame(afs, mpa) < 0) if (af_filter_frame(afs, mpa) < 0)
return AD_ERR; return AD_ERR;
} }