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

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

Fixes Ticket592
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes Ticket561
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master: (22 commits)
  g722dec: check output buffer size before decoding
  g722dec: cosmetics: reindent/linewrap
  g722dec: remove the use of lowres for half-rate decoding.
  tta: check for extradata allocation failure in tta demuxer
  tta: check for allocation failure of decode_buffer
  tta: use correct frame_length calculation.
  tta: add support for decoding 24-bit sample format
  cosmetics: indentation
  tta: remove pointless braces
  tta: check output buffer size after adjusting frame length for last frame
  tta: fix reading of format in TTA header.
  tta: remove useless commented-out lines
  tta: check remaining bitstream size while reading unary value
  lavf: deprecate AVStream.stream_copy
  avconc: split choose_codec() to choose_decoder/choose_encoder.
  lavf: simplify by using FFMAX/FFMIN.
  mpegenc: add preload private option.
  cosmetics: simplify latm_decode_init
  latm: avoid unnecessary reinit of the aac decoder
  aacdec: initialize sbr context only in new channel elements
  ...

Conflicts:
	avconv.c
	libavcodec/resample.c
	libavcodec/tta.c
	libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This should be changed to LIBAVDEVICE_IDENT once it exists.
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: Stefano Sabatini <stefasab@gmail.com>

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

Without this patch each displayed second is incremented 1 frame early,
second 0 is only 24 frames long where as every other second is 25 frames
long.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

It is broken because an AVCodecContext can be opened/closed multiple
times, and sample_rate is getting divided by 2 each time that happens.

This removes the only use of lowres for audio.

using a floating-point calculation is not necessary.

Note that this will not work in most cases with avconv and avplay due to the
AVCODEC_MAX_AUDIO_FRAME_SIZE limit, but it will decode correctly if given a
large enough output buffer.

TTA does not support float at all, and format 2 is encrypted TTA.

It's only used in avconv, so it properly belongs to OutputStream struct
there.

Prevents -c copy from working for input streams and allows to move
stream_copy variable from AVStream to OutputStream.

Deprecate AVFormatContext.preload.

The situation was not clear when support was added but it is now:
CELT and Opus are really two different codecs.
The current code supports CELT via libcelt, not Opus.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

audio_resample can not reduce the number of channels

* cus/stable:
  ffplay: add frame drop statistics
  ffplay: consider estimated time of filter in early frame drop
  ffplay: reimplement early frame drop
Merged-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  adpcm: use sign_extend()
  mpeg12: fix mpeg_decode_slice context parameter type
  Revert "mpeg12: move full_pel from MpegEncContext to Mpeg1Context"

Conflicts:
	libavcodec/mpeg12.c
	libavcodec/mpeg12.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Marton Balint <cus@passwd.hu>