path: root/audio/audio.c
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* audio: fix audio data memory leakwm42013-11-141-1/+1
| | | | | Practically all audio decoding and filtering code leaked sample data memory after uninitialization due to a simple logic bug (or typo).
* mp_audio: use av_malloc (cargo cult for libav*)wm42013-11-121-3/+27
| | | | | | | | | | | | | | | libav* is generally freaking horrible, and might do bad things if the data pointer passed to it are not aligned. One way to be sure that the alignment is correct is allocating all pointers using av_malloc(). It's possible that this is not needed at all, though. For now it might be better to keep this, since the mp_audio code is intended to replace another buffer in dec_audio.c, which is currently av_malloc() allocated. The original reason why this uses av_malloc() is apparently because libavcodec used to directly encode into mplayer buffers, which is not the case anymore, and thus (probably) doesn't make sense anymore. (The commit subject uses the word "cargo cult", after all.)
* audio/filter: prepare filter chain for non-interleaved audiowm42013-11-121-11/+137
| | | | | | | | | | | | | | | | | | Based on earlier work by Stefano Pigozzi. There are 2 changes: 1. Instead of, mp_audio.planes[0] must be used. 2. mp_audio.len used to contain the size of the audio in bytes. Now mp_audio.samples must be used. (Where 1 sample is the smallest unit of audio that covers all channels.) Also, some filters need changes to reject non-interleaved formats properly. Nothing uses the non-interleaved features yet, but this is needed so that things don't just break when doing so.
* audio: replace af_fmt2str_short -> af_fmt_to_strwm42013-11-071-1/+1
| | | | Also, remove all af_fmt2str usages.
* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-061-1/+1
| | | | Followup commit. Fixes all the files references.
* audio: add channel map selection functionwm42013-05-121-2/+1
| | | | | | | | | | | | | | | | | | | | The point is selecting a minimal fallback. The AOs will call this through the AO API, so it will be possible to add options affecting the general channel layout selection. It provides the following mechanism to AOs: - forcing the correct channel order - downmixing to stereo if no layout is available - allow 5.1 <-> 5.1(side) fallback - handling "unknown" channel layouts This is quite weak and lots of code/complexity for little gain. All AOs already made sure the channel order was correct, and the fallback is of little value, and could perhaps be done in the frontend instead, like stereo downmixing with --channels=2 is handled. But I'm not really sure how this stuff should _really_ work, and the new code will hopefully provides enough flexibility to make radical changes to channel layout negotiation easier.
* audio/filters: add af_forcewm42013-05-121-0/+6
| | | | | Its main purpose is for testing in case channel layout stuff breaks, in particular in connection with old audio filters.
* audio: print channel map additionally to channel count on terminalwm42013-05-121-0/+15
* af: use mp_chmap for mp_audio, include channel map in format negotiationwm42013-05-121-2/+20
| | | | | Now af_lavrresample pretends to reorder the channels, although it doesn't yet, and nothing sets non-standard layouts either.
* audio: add some setters for mp_audio, and require filters to use themwm42013-05-121-0/+38
mp_audio has some redundant fields. Setters like mp_audio_set_format() initialize these properly. Also move the mp_audio struct to a the file audio.c. We can remove a mysterious line of code from af.c: in.format |= af_bits2fmt(in.bps * 8); I'm not sure if this was ever actually needed, or if it was some kind of "make it work" quick-fix that works against the way things were supposed to work. All filters etc. now set the format correctly, so if there ever was a need for this code, it's definitely gone.