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dec_audio: fix segment boudnary switching
Some bugs in this code are exposed by e.g. playing lossless audio files with --ad-lavc-threads=16. (libavcodec doesn't really support threaded audio decoding, except for lossless files.) In these cases, a major amount of audio can be buffered, which makes incorrect handling of this buffering obvious. For one, draining the decoder can take a while, so if there's a new segment, we shouldn't read audio. The segment end check was completely wrong, and used the start value.
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@@ -200,7 +200,9 @@ void audio_work(struct dec_audio *da)
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if (da->current_frame)
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return;
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if (!da->packet && demux_read_packet_async(da->header, &da->packet) == 0) {
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if (!da->packet && !da->new_segment &&
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demux_read_packet_async(da->header, &da->packet) == 0)
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{
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da->current_state = DATA_WAIT;
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return;
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}
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@@ -211,6 +213,7 @@ void audio_work(struct dec_audio *da)
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da->packet = NULL;
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}
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bool had_input_packet = !!da->packet;
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bool had_packet = da->packet || da->new_segment;
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int ret = da->ad_driver->decode_packet(da, da->packet, &da->current_frame);
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@@ -233,12 +236,12 @@ void audio_work(struct dec_audio *da)
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fix_audio_pts(da);
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bool segment_end = true;
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bool segment_end = !da->current_frame && !had_input_packet;
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if (da->current_frame) {
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mp_audio_clip_timestamps(da->current_frame, da->start, da->end);
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if (da->current_frame->pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
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segment_end = da->current_frame->pts >= da->start;
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segment_end = da->current_frame->pts >= da->end;
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if (da->current_frame->samples == 0) {
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talloc_free(da->current_frame);
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da->current_frame = NULL;
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