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>

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 885fc05865.
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>

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

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

Also while here sort chunk ids.

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

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>

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: Martin Storsjö <martin@martin.st>

This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.

Signed-off-by: Martin Storsjö <martin@martin.st>

This function is used for calculating digest position for RTMP handshake
packets.

Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Martin Storsjö <martin@martin.st>

This removes the use of macro nesting in these code constructs, which
makes it easier to parse in pre-processors.

If there is a 'SMV0' tag, but no 'fmt' tag, it is possible that
'st' got used as NULL.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

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

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

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

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

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

'mono' came before 'mono16' in the list of colorspace alternatives,
causing files of the latter type to be parsed as the former.

Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>

This makes usage of the macro look more natural when
used with array entries.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

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

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

Reveiwed-by: compn <tempn@twmi.rr.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This happens with for example mplayer.
Fixing it in ffmpeg allows new ffmpeg to be compiled with older mplayer
which would not be possible if the fix was just in mplayer.

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

The new function is called avformat_match_stream_specifier()
and does not include logging the error message.

Opus in Matroska is still a draft.

This patch also introduces CODEC_ID_OPUS.

It already skips any extra bytes at the end, and apparently there are some
samples in the wild with larger 'kuki' chunks.

If skip_samples is set and timestamps are synthesized using durations,
make them start at -skip_samples (rescaled) instead of 0,
so that the timestamp of the first undiscarded sample is 0.

They are only supported if pthread is available.

It is used to store a value read by avio_rb64().

according to the CAF specification:
"... the value for mChunkSize can be greater than the actual valid content
of the packet table chunk"

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

This fixes seeking in filecopy.avi of Ticket504

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

Increase feedback provided in avformat_find_stream_info().

In particular, fix trac issue #218.

Improve feedback.