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.

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

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.

audio_resample can not reduce the number of channels

This avoids warnings from the overflow checker and simplifies the code.
Signed-off-by: Mans Rullgard <mans@mansr.com>

During slice threading only MpegEncContext is passed to
mpeg_decode_slice, remove a wrong cast and change the function
definition to take MpegEncContext pointer.

This reverts commit da22ba7d since it
broke slice threading. Slice threading just duplicates MpegEncContext
so every value used during mpeg_decode_slice has to be in it.
A second patch will fix the illusion that Mpeg1Context is available
in mpeg_decode_slice.

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

This fixes assembling using "nasm".