mirror of https://github.com/mpv-player/mpv
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:
parent
1a522f2976
commit
4cabd08e8a
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@ -153,23 +153,33 @@ void audio_uninit(struct dec_audio *d_audio)
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MP_VERBOSE(d_audio, "Uninit audio filters...\n");
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af_destroy(d_audio->afilter);
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uninit_decoder(d_audio);
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talloc_free(d_audio->waiting);
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talloc_free(d_audio);
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}
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static int decode_new_frame(struct dec_audio *da)
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{
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while (!da->waiting) {
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int ret = da->ad_driver->decode_packet(da, &da->waiting);
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if (ret < 0)
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return ret;
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if (da->waiting) {
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da->pts_offset += da->waiting->samples;
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da->decode_format = *da->waiting;
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mp_audio_set_null_data(&da->decode_format);
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}
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}
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return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
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}
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/* Decode packets until we know the audio format. Then reinit the buffer.
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* Returns AD_OK on success, negative AD_* code otherwise.
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* Also returns AD_OK if already initialized (and does nothing).
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*/
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int initial_audio_decode(struct dec_audio *da)
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{
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while (!da->waiting) {
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int ret = da->ad_driver->decode_packet(da, &da->waiting);
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if (ret < 0)
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return ret;
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}
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talloc_steal(da, da->waiting);
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da->decode_format = *da->waiting;
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return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
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return decode_new_frame(da);
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}
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static bool copy_output(struct af_stream *afs, struct mp_audio_buffer *outbuf,
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@ -208,31 +218,22 @@ int audio_decode(struct dec_audio *da, struct mp_audio_buffer *outbuf,
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if (copy_output(afs, outbuf, minsamples, false))
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break;
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struct mp_audio *mpa = da->waiting;
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da->waiting = NULL;
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if (!mpa) {
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res = da->ad_driver->decode_packet(da, &mpa);
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res = decode_new_frame(da);
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if (res < 0) {
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// drain filters first (especially for true EOF case)
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copy_output(afs, outbuf, minsamples, true);
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break;
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}
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if (!mpa)
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continue;
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da->pts_offset += mpa->samples;
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da->decode_format = *mpa;
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mp_audio_set_null_data(&da->decode_format);
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}
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// On format change, make sure to drain the filter chain.
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if (!mp_audio_config_equals(&afs->input, mpa)) {
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da->waiting = talloc_steal(da, mpa);
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if (!mp_audio_config_equals(&afs->input, da->waiting)) {
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copy_output(afs, outbuf, minsamples, true);
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res = AD_NEW_FMT;
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break;
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}
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struct mp_audio *mpa = da->waiting;
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da->waiting = NULL;
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if (af_filter_frame(afs, mpa) < 0)
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return AD_ERR;
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}
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