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* af: add replaygain_data field to af_stream and af_instanceAlessandro Ghedini2014-04-041-0/+1
| | | | Closes #664
* command: allow changing filters before video chain initializationwm42014-03-301-2/+2
| | | | | | | Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
* audio: remove sample rate limit checkswm42014-03-301-7/+1
| | | | | | | | | | | This played the file at a wrong sample rate if the rate was out of certain bounds. A comment says this was for the sake of libaf/af_resample.c. This resampler has been long removed. Our current resampler (libav/swresample) checks supported sample rates on reconfiguration, and will error out if a sample rate is not supported. And I think that is the correct behavior.
* af: add metadata field to af_stream and af_instanceAlessandro Ghedini2014-03-131-0/+1
| | | | | | This allows to propagate metadata information to audio filters. Closes #632
* audio: don't downmix when doing digital passthroughwm42014-03-101-1/+2
| | | | | | This obviously doesn't work. It wasn't much of a problem in the past because most passthrough formats use 2 channels, which is also the default for downmix.
* audio: make --channels option always force the output layoutwm42014-03-101-6/+1
| | | | | | Use the --channels value directly on the AO, instead of doing it only in the --channels=stereo (default) case and if the decoder output is not stereo.
* audio: don't write audio when pausedwm42014-03-091-0/+5
| | | | | | | | | | This is probably "safer". Without it, we will play 1 sample, because the logic was written in a way to decode 1 sample if audio is paused. 1 sample usually will initialize the audio PTS, but not play any real audio. Also see previous commit. In ancient times, this actually used 1 byte (instead of 1 sample), so clearly no sample was written, unless the audio was 8-bit mono.
* audio: remove handling of partially written datawm42014-03-091-7/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove the ao_buffer_playable_samples field. This contained the number of samples that fill_audio_out_buffers() wanted to write to the AO (i.e. this data was supposed to be played at some point), but ao_play() rejected it due to partial fill. This could happen with many AOs, notably those which align all written data to an internal period size (often called "outburst" in the AO code), and the accepted number of samples is rounded down to period boundaries. The left-over samples at the end were still kept in mpctx->ao_buffer, and had to be played later. The reason ao_buffer_playable_samples had to exist was to make sure that at EOF, the correct number of left-over samples was played (and not possibly other data in the buffer that had to be sliced off due to endpts in fill_audio_out_buffers()). (You'd think you could just slice the entire buffer, but I suspect this wasn't done because the end time could actually change due to A/V sync changes. Maybe that was the reason it's so complicated.) Some commits ago, ao.c gained internal buffering, and ao_play() will never return partial writes - as long as you don't try to write more samples than ao_get_space() reports. This is always the case. The only exception is filling the audio buffers while paused. In this case, we decode and play only 1 sample in order to initialize decoding (e.g. on seeking). Actually playing this 1 sample is in fact a bug, but even of the AO doesn't have period size alignment, you won't notice it. In summary, this means we can safely remove the code.
* audio/out: make ao struct opaquewm42014-03-091-42/+56
| | | | | | We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
* encode: don't access ao->ptswm42014-03-071-0/+4
| | | | | | | | | | This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
* client API: add events for video and audio reconfigwm42014-02-171-0/+3
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* player: fix an assert when reinitializing audio in some caseswm42014-02-091-0/+3
| | | | | | | | This sometimes happened when changing playback speed (= reinitializing audio) after seeking of playback start. The assertion in audio.c:441 was triggered, because buffer_playable_samples wasn't reset correctly when the audio buffer was cleared or shortened. The assertion is correct and should hold up any time.
* Fix audio delay inversionMartin Herkt2014-01-061-2/+2
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* player: add infrastructure to select multiple tracks at oncewm42013-12-241-1/+1
| | | | | Of course this does not allow decoding multiple tracks at once; it just adds some minor infrastructure, which could be used to achieve this.
* player: redo demuxer stream selectionwm42013-12-241-3/+3
| | | | | | | Use struct track to decide what stream to select. Add a "selected" field and use that in some places instead of checking mpctx->current_track.
* audio: mp_msg conversionswm42013-12-211-0/+2
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* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-2/+2
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* Move options/config related files from mpvcore/ to options/wm42013-12-171-1/+1
| | | | | | | | | Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
* Rename mp_core.h to core.hwm42013-12-171-1/+1
| | | | Get rid of the mp_ prefix.
* Move mpvcore/player/ to player/wm42013-12-171-0/+471