* commit '8ac0f6767bf63d3e6b308ee6648ff02598b81e03':
dcadec: allow the decoder to change the channel layout mid-stream
cook: use av_dlog() for debug logging instead of av_log() with AV_LOG_ERROR
cook: move samples_per_frame from COOKSubpacket to where it is used
cook: use av_get_channel_layout_nb_channels() instead of cook_count_channels()
cook: reverse a condition so that the code makes more sense
cook: remove unneeded COOKContext variable, sample_rate
cook: remove unneeded COOKContext variable, bit_rate
cook: use AVCodecContext.channels instead of keeping a private copy
bmvaudio: set channel layout at init() rather than validating it
atrac1: do not keep a copy of channel count in the private context
dsicinaudio: set channels and channel layout
g722dec: set channel layout at initialization instead of validating it
amrwbdec: set channels, channel_layout, and sample_rate
amrnbdec: set channels, channel_layout, and sample_rate
dca_parser: allow the parser to change the sample rate
lavc: check channel count after decoder init
lavc: move SANE_NB_CHANNELS to internal.h and use it in the PCM decoders
Conflicts:
libavcodec/dcadec.c
libavcodec/pcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids using bits_per_coded_sample for this information.
bits_per_coded_sample should be 4 for this codec normally,
since two samples are encoded into one 8 bit codeword.
In principle, this might be info that needs to be passed from
a demuxer, and in that case, a private AVOption isn't the best
choice, but no such samples are available at the moment, so
that use case is purely theoretical at the moment.
Signed-off-by: Martin Storsjö <martin@martin.st>
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.
