Commit Graph

15 Commits

Author SHA1 Message Date
wm4 7b8862760d audio: add missing declaration 2015-01-14 22:15:00 +01:00
wm4 97becbc31b audio: add some utility functions for refcounted frames
Used in the following commits.
2015-01-13 20:14:25 +01:00
wm4 7d6e58471f audio: make mp_audio_config_to_str return a stack-allocated string
Simpler overall.
2014-11-25 11:11:31 +01:00
wm4 46d6fb9dc1 audio: add mp_audio_make_writeable() 2014-11-10 22:02:05 +01:00
wm4 0b26d8c666 audio: add function to convert AVFrame to mp_audio references
This is somewhat duplicated from ad_lavc.c and af_lavfi.c, but will
eventually be used by both.
2014-11-10 22:02:04 +01:00
wm4 5d46e44160 audio: add mp_audio_pool
A helper to allocate refcounted audio frames from a pool. This will
replace the static buffer many audio filters use (af->data), because
such static buffers are incompatible with refcounting.
2014-11-10 18:15:22 +01:00
wm4 9388f69f67 audio: use AVBufferRef to allocate audio frames
A first step towards refcounted audio frames.

Amazingly, the API just does what we want, and the code becomes
simpler. We will need to NIH allocation from a pool, though.
2014-11-10 10:43:15 +01:00
Marcoen Hirschberg 31a10f7c38 af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriate
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
2014-05-28 21:38:00 +02:00
wm4 addfcf9ce3 audio: better rejection of invalid formats
This includes the case when lavc decodes audio with more than 8
channels, which our audio chain currently does not support.

the changes in ad_lavc.c are just simplifications. The code tried to
avoid overriding global parameters if it found something invalid, but
that is not needed anymore.
2013-11-27 00:16:05 +01:00
wm4 cc5083cfe0 mp_audio: use av_malloc (cargo cult for libav*)
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.)
2013-11-12 23:35:33 +01:00
wm4 d2e7467eb2 audio/filter: prepare filter chain for non-interleaved audio
Based on earlier work by Stefano Pigozzi.

There are 2 changes:

1. Instead of mp_audio.audio, 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.
2013-11-12 23:16:31 +01:00
wm4 d9582ad0a4 audio/filters: add af_force
Its main purpose is for testing in case channel layout stuff breaks, in
particular in connection with old audio filters.
2013-05-12 21:24:56 +02:00
wm4 3b1956608d audio: print channel map additionally to channel count on terminal 2013-05-12 21:24:56 +02:00
wm4 7971bb87cb af: use mp_chmap for mp_audio, include channel map in format negotiation
Now af_lavrresample pretends to reorder the channels, although it
doesn't yet, and nothing sets non-standard layouts either.
2013-05-12 21:24:54 +02:00
wm4 f7a427676c audio: add some setters for mp_audio, and require filters to use them
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.
2013-05-12 21:24:54 +02:00