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This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
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Remove the awkward planarization. It had to be done because the AC3
encoder requires planar formats, but now we support them natively.
Try to simplify buffer management with mp_audio_buffer.
Improve checking for buffer overflows and out of bound writes. In
theory, these shouldn't happen due to AC3 fixed frame sizes, but being
paranoid is better.
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The format negotiation is the same, except don't confusingly copy the
input format into af->data, just to overwrite it later. af->data should
alwass contain the output format, and the existing code was just a very
misguided use of the af_test_output() helper function.
Just set af->data to the output format immediately, and modify the input
format properly.
Also, if format negotiation fails (and needs another iteration), don't
initialize the libavcodec encoder.
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Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
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Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
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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.
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Also, remove all af_fmt2str usages.
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Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
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In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
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This is done in af_lavrresample now, and as part of format negotiation.
Also remove the remaining reorder_channel calls. They were redundant
and did nothing.
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Now af_lavrresample pretends to reorder the channels, although it
doesn't yet, and nothing sets non-standard layouts either.
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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.
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avcodec_encode_audio() was deprecated, and was finally removed from
Libav and FFmpeg git.
This appears to work. I get heavy A/V desync with -ao alsa and -ao pcm,
but this was already so before this change.
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This fixes operation with current ffmpeg releases.
Note that this planarization is slow and should be reverted once proper
planar audio support is there in mpv.
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Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.
The two commits are separate, because git is bad at tracking renames
and content changes at the same time.
Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
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Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
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