It allows to check if a given codec can be written into a container.

Metadata currently is written only at the start of the file in normal
cases, when transcoding from a rtmp source metadata could be
written later and the offset recorded can exceed 32bit.

Signed-off-by: Anton Khirnov <anton@khirnov.net>

ID3v1 are legacy tags with several limitations; furthermore
avconv/ffmpeg writes the tags in UTF-8 which probably has near-0
software support.

Add a -write_id3v1 option to be able to turn it on; disabled by default.

Signed-off-by: Anton Khirnov <anton@khirnov.net>

MpegTSContext's AVClass member can be NULL.

strsep() is not POSIX compliant, and thus not supported on some platform.
Fix compilation on Solaris.

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

New streams are always added to the last output file.

Move find_codec_or_die(), choose_codec(), new_output_stream() and
read_avserver_streams() below transcode() where they belong.

It's only used in opt_output_file(), so move it right above
opt_output_file().

parse_forced_key_frames() is only used in new_video_stream(), so move it
right above new_video_stream().

Its only reason for its existence was ffmpeg's inability to properly assign
AVOptions to streams. Now this is not a problem anymore, so 'ab' should
go.

It's often mistakenly used as 'same quality', emphasize that it's not
true in the manual.

The new option doesn't depend on its placement wrt -new* options (which
don't exist anymore) and works in a similar way as per-stream AVOptions.

-[vas]codec remain as aliases to -codec:[vas]

Nothing except x264 uses them and native x264 presets should be used
with private options.

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

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

* qatar/master:
  Fix NASM include directive
  dsputil_mmx: Honor HAVE_AMD3DNOW
  lavf,lavd: remove all usage of AVFormatParameters from demuxers.
  jack: add 'channels' private option.
  VC-1: fix reading of custom PAR.
  Remove redundant and dubious video codec detection by its extradata
  mpeg12: remove repeat-field code disabled since May 2002
  patch checklist: suggest fate instead of regression tests
  Turn on resampling on sudden size change instead of bailing out during recode.
  avtools: reinitialise filter chain when input video stream changes dimensions

Conflicts:
	Makefile
	avconv.c
	doc/developer.texi
	ffplay.c
	libavcodec/x86/dsputil_mmx.c
	libavdevice/libdc1394.c

Merged-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: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

AVFormatParameters are converted into corresponding private options in
av_open_input_file/stream() compat wrappers, so accessing them from
demuxers is redundant.

Get rid of AVFormatParameters usage.

Custom PAR num/denum are in 1-256 range.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

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

Since SDL has no audio buffer fullness info, one can get a much precise audio
clock based on the last time of the audio callback and the elapsed time since.

To achieve this I introduced the audio_current_pts and audio_current_pts_drift
variables (similar to video_current_pts and video_current_pts_drift) and
calculate them in the end of the audio callback, when VideoState->audio_clock
is already updated. The reference time I use is from the start of the audio
callback, because this way the amount of time used for audio decoding is not
interfereing with calculation.

I also replaced the audio_write_get_buf_size function with a calculated
variable because when the audio frame decoding is in progress audio_buf_size
and audio_buf_index are not stable, so using them from other threads are not a
good idea.

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>

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

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

* qatar/master:
  aacenc: Fix number of coefficients used in a LFE channel.
  aacenc: Fix a segfault with grouped psymodel.

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

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

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