It is already checked by avcodec_open2().

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.

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.