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Commit Graph

8 Commits

Author SHA1 Message Date
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