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audio/out: prevent underruns with spdif under certain conditions
The player tries to avoid splitting frames with spdif (sample alignment stuff). This can in certain corner cases with certain drivers lead to the situation that ao_get_space() returns a number higher than 0 and lower than the audio frame size. The playloop will round this down to 0 bytes and do nothing, leading to a missed wakeup. This can lead to underruns or playback completely getting stuck. It can be reproduced by playing AC3 passthrough with no video and: --ao=null --ao-null-buffer=0.256 --ao-null-outburst=6100 This commit attempts to fix it by allowing the playloop to write some additional data (to get a complete frame), that will be buffered within the AO ringbuffer even if the audio device doesn't want it.
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@ -182,6 +182,7 @@ static int unlocked_get_space(struct ao *ao)
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struct ao_push_state *p = ao->api_priv;
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int space = mp_audio_buffer_get_write_available(p->buffer);
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if (ao->driver->get_space) {
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int align = af_format_sample_alignment(ao->format);
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// The following code attempts to keep the total buffered audio to
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// ao->buffer in order to improve latency.
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int device_space = ao->driver->get_space(ao);
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@ -191,6 +192,7 @@ static int unlocked_get_space(struct ao *ao)
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// byte based and doesn't do proper chunked processing.
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int min_buffer = ao->buffer + 64;
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int missing = min_buffer - device_buffered - soft_buffered;
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missing = (missing + align - 1) / align * align;
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// But always keep the device's buffer filled as much as we can.
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int device_missing = device_space - soft_buffered;
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missing = MPMAX(missing, device_missing);
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