Add "prog" parameter value, and deprecate numeric values.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>

This broke compilation on darwin, revert until a better solution is found.

This reverts commit a812b599.

swr_convert is not properly buffering packed input audio when the
output is not large enough, and when resampling is not actually needed
(same samplerate and no SWR_FLAG_RESAMPLE).

buf_set() is only handling the first channel and leaving the others as-is.

Sample program to reproduce the problem is here https://gist.github.com/2431768Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This adds a hand-optimized assembly version for get_cabac much like the
existing one, but it works if the table offsets are RIP-relative.
Compared to the non-RIP-relative version this adds 2 lea instructions
and it needs one extra register.
There is a surprisingly large performance improvement over the c version (more
so than the generated assembly seems to suggest) just in get_cabac, I measured
roughly 40% faster for get_cabac on a K8. However, overall the difference is
not that big, I measured roughly 5% on a test clip on a K8 and a Core2.
Hopefully it still compiles on x86 32bit...
v2: incorporated feedback from Loren Merritt to avoid rip-relative movs
for every table, and got rid of unnecessary @GOTPCREL.
v3: apply similar fixes to the the decode_significance functions, and use
same macro arguments for non-pic case.
v4: prettify inline asm arguments, add a non-fast-cmov version (as I expect
the c code to be faster otherwise since both cmov and sbb suck hard on a
Prescott, even can't construct the mask with a 64bit shift as that's just as
terrible - it's quite difficult to find usable instructions on that chip...).
This is tested to work but not on a P4, in theory it _should_ be fast there.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  avcodec: add a cook parser to get subpacket duration
  FATE: allow lavf tests to alter input parameters
  FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
  FATE: replace the acodec-g726 test with 4 new encode/decode tests
  FATE: replace current g722 encoding tests with an encode/decode test
  FATE: add a pattern rule for generating asynth wav files
  FATE: optionally write a WAVE header in audiogen
  avutil: add audio fifo buffer

Conflicts:
	doc/APIchanges
	libavcodec/version.h
	libavutil/avutil.h
	tests/Makefile
	tests/codec-regression.sh
	tests/fate/voice.mak
	tests/lavf-regression.sh
	tests/ref/acodec/g722
	tests/ref/acodec/g726
	tests/ref/acodec/pcm_s24daud
	tests/ref/lavf/dv_fmt
	tests/ref/lavf/gxf
	tests/ref/lavf/mxf
	tests/ref/lavf/mxf_d10
	tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

This also matches the rest of the demuxer which will return
partial packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes jittery video playback of rm files with cook audio.

Fixes out of array write.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This fixes out of array writes.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes a segfault with Ogg output, libtheora not compiled in
and no codec specified.

Change some lavf tests to avoid resampling and channel mixing.

This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.

Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.

Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.

The functions operate on the sample level rather than the byte level and work
with all audio sample formats.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This will allow decoding a single undamaged slice even if all others are lost
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This allows us to clear outside of the main thread for example.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The native libavfilter hflip filter does the same thing.

Signed-off-by: Stefano Sabatini <stefasab@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Previously too little data could lead to a false detection.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  dv: Initialize encoder tables during encoder init.
  dv: Replace some magic numbers by the appropriate #define.
  FATE: pass the decoded output format and audio source file to enc_dec_pcm
  FATE: specify the input format when decoding in enc_dec_pcm()
  x86inc: support AVX abstraction for 2-operand instructions
  configure: detect PGI compiler and set suitable flags
  avconv: check for an incompatible changing channel layout
  avio: make AVIOContext.av_class pointer to const
  nutdec: add malloc check and fix const to non-const conversion warnings

Conflicts:
	ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Generating warnings when casting const away leads to tight constraints
on the code if one wants to avoid warnings. This is especially true for
generic code that is supposed to work with both const and non const.
These tight constrains cause people to waste time trying to find a
way to write code so it doesnt generate any warning, while people
should rather spend their time thinking on how to write fast,
clean, maintainable and bug free code.
Removing this class of warnings fixes this issue.
Approved-by: Nicolas George <nicolas.george@normalesup.org>
Approved-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This fixes out of array reads

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

See ticket #1228.

Yasm was fixed in its r2161 and yasm 0.8.0 (Apr 2010) contained this fix.
Nasm was fixed in 2.06 (Jun 2009):
https://groups.google.com/group/alt.lang.asm/browse_thread/thread/fcc85bbc3745d893

I tested with yasm  0.7.99 and yasm 1.2.0.7, where this works fine.

I also tested with nasm. The nasm shipping with Xcode is too old to understand
ffmpeg's assembly, before and after the patch. Nasm 2.10 fails to compile
fft_mmx.asm on trunk with

  libavcodec/x86/fft_mmx.asm:88: panic: section ".text" has already been specified with alignment 32, conflicts with new alignment of 16

but builds fine if I change the two alignment "16"s in x86inc.asm to "32". With this patch,
nasm 2.10 fails with

  libavcodec/x86/fft_mmx.asm:39: panic: section ".rodata" has already been specified with alignment 32, conflicts with new alignment of 16

instead, but again builds fine with s/16/32/.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>