Attached patch (together with demuxing patch) allows WMP/msacm G723.1 codec decode files encoded by FFmpeg.

Tested with both 6400 and 5333 mode
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

This should have no effect as it was not used in this case
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

OpenJPEG doesn't have a max lowres limit, so don't enforce an arbitrary one.
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
  rtmp: Move the CONFIG_ condition into the if conditions
  aac: Mention abbreviation as well in long_name
  build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
  doc: Add Git configuration section
  configure: Add a dependency on https for rtmpts
  rtp: Only choose static payload types if the sample rate and channels are right

Conflicts:
	doc/git-howto.texi
	libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This replaces the use of the no_packet and no_frame arrays.

Attached patch fixes remuxing of G723.1 in wav, so the output is playable by WMP.
(It's still not enough for encoding - probably some extradata should be added to the output file
to make it playable by WMP/win codec)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Our implementation of RTMPE is heavily based on librtmp.
Signed-off-by: Martin Storsjö <martin@martin.st>

This reverts commit 885fc058.
This commit caused timestamps in case of generic seeking to become
inconsistent.

This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.
Signed-off-by: Martin Storsjö <martin@martin.st>

Found-by: durandal_1707
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes read of uninitialized memory
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This is not implemented and doesnt work.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

Most people know the codec as "AAC" and not "Advanced Audio Coding".

  Make it possible to use the run bisect sub command. As with all
other ffbisect commands, revisions that do not contain the needed
tools are skipped.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>

In file libavcodec/x86/dsputil_mmx.c, function ff_put_pixels_clamped_mmx(), there are two assembly code blocks. In the first block (in the unrolled loop), the instructions "movq 8%3, %%mm1 \n\t" etc have problem.
For above instruction, it is clear what the programmer wants: a load from p + 8. But this assembly code doesn’t guarantee that. It only works if the compiler puts p in a register to produce an instruction like this: “movq 8(%edi), %mm1”. During compiler optimization, it is possible that the compiler will be able to constant propagate into p. Suppose p = &x[10000]. Then operand 3 can become 10000(%edi), where %edi holds &x. And the instruction becomes “movq 810000(%edx)”. That is, it will stride by 810000 instead of 8.
This will cause the segmentation fault.
This error was fixed in the second block of the assembly code, but not in the unrolled loop.

How to reproduce:
This error is exposed when we build the ffmpeg using Intel C++ Compiler, IPO+PGO optimization. The ffmpeg was crashed when decoding a mjpeg video.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The ffrtmpcrypt protocol depends on external libraries, which are
also required to compile the header file.

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

The rtmpts protocol uses https implicitly, via the ffrtmphttp
protocol, but the ffrtmphttp protocol is also useable for plain
rtmpt without https, so the dependency needs to be added here instead.
Signed-off-by: Martin Storsjö <martin@martin.st>

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

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

Also while here sort chunk ids.
Signed-off-by: Paul B Mahol <onemda@gmail.com>

The rtmpts protocol uses https implicitly, via the ffrtmphttp
protocol, but the ffrtmphttp protocol is also useable for plain
rtmpt without https, so the dependency needs to be added here instead.
Signed-off-by: Martin Storsjö <martin@martin.st>

If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.

G722 is a special case where the normal rules don't apply.
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Fix a segfault.

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

Signed-off-by: Paul B Mahol <onemda@gmail.com>

* qatar/master:
  v410dec: Implement explode mode support
  zerocodec: fix direct rendering.
  wav: init st to NULL to avoid a false-positive warning.
  wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
  h264: refactor NAL decode loop
  RTMPTE protocol support
  RTMPE protocol support
  rtmp: Add ff_rtmp_calc_digest_pos()
  rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
  swscale: add missing HAVE_INLINE_ASM check.
  lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
  vc1: Add a test for interlaced field pictures
  swscale: Mark all init functions as av_cold
  swscale: x86: Drop pointless _mmx suffix from filenames
  lavf: use conditional notation for default codec in muxer declarations.
  swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
  dsputil: ppc: cosmetics: pretty-print
  dsputil: x86: add SHUFFLE_MASK_W macro
  configure: respect CC_O setting in check_cc

Conflicts:
	Changelog
	configure
	libavcodec/v410dec.c
	libavcodec/zerocodec.c
	libavformat/asfenc.c
	libavformat/version.h
	libswscale/utils.c
	libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Try and decode broken files, but still fail if explode
mode is enabled.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Set picture type before calling get_buffer.
This allows the DR application to make better decisions.
It also fixes a resource leak in case of missing reference frames
since it would call get_buffer but never release_buffer.
Also use FFSWAP to ensure that the AVFrame is properly initialized
in the next get_buffer (in particular that data[0] is NULL).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.

Fixes warning:
    libavformat/wav.c: In function ‘wav_read_header’:
    libavformat/wav.c:499:44: warning: ‘st’ may be used uninitialized in this function [-Wmaybe-uninitialized]
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

The audio and video code paths were too different,
most of the decoding has been rewritten.