the sample size.

This is already done in avcodec_decode_audio3()

Also return AVERROR_PATCHWELCOME.

They only operate on stereo content, so the extra param is not necessary and
also allows for simplifying the code.

Now they only do stereo interleaving.

This should also fix decoding of mono 24-bit.

It is identical for 16-bit and 24-bit, so there is no need to have duplicate
code.

The bits are not wasted, they are additional low bits that are added to the
16-bit decompressed samples to increase the output sample depth.

Also rearranges some functions for easier cleanup on failure.

reduces memory usage when the stream has fewer than MAX_CHANNELS

check frame header channel count against header/container channel count.

It is already done when using it to set sample_fmt.

Otherwise the delay expressed in has_b_frames increases with every
avcodec_close/avcodec_open.
Fixes fate-ea-dct with more than 1 thread.

Fixes Ticket592

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

Add whitespace.

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

Fixes Ticket561

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

Cleanup the options and add explicit fragment_size avoption.

* 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.