Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
Also break some long lines, remove codec function placeholder comments
and add spaces in sample/pixel format lists.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
Earlier, bits per sample was defined as 8, since
bits_per_coded_sample was used to indicate whether to ignore
the lower bits of the codeword, having values 6, 7 or 8.
g722 encodes 2 samples into one byte codeword, therefore the
bits per sample is 4. By changing this, the generated timestamps
for streams encoded with g722 become correct.
This makes timestamp generation for g722 data correct (both when
encoding and when demuxing from raw g722 files).
Signed-off-by: Martin Storsjö <martin@martin.st>
