This allows user to select quantisation matrix from different profile,
stamp frames with custom vendor string and change target bitrate.

Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.

Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.

fixes writing of uninitialized packet data

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

Fixes pre-processing with latest versions of nasm.

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

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.

CC:libav-stable@libav.org

Currently we have an assert() that prevents the frame from being too large,
but it is more user-friendly to give an error message instead of aborting on
assert(). This condition is quite unlikely due to the minimum bit rate check
in encode_init(), but it is still worth having.

The maximum theoretical frame size is around 17000 bytes. Although in
practice it will generally be much smaller, we require a larger buffer
just to be safe.

CC: libav-stable@libav.org

ff_wma_init() allows up to 50kHz, but this generates an exponent band
size table that requires 65 bands. The code assumes 25 bands in many
places, and using sample rates higher than 48kHz will lead to buffer
overwrites.

CC:libav-stable@libav.org

This is near the theoretical limit for wma frame size and is the most that
our decoder can handle. Allowing higher bit rates will just end up padding
each frame with empty bytes.

Fixes invalid writes for avconv when using very high bit rates.

CC:libav-stable@libav.org

Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.

Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.

The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.

Each packet has 18 sectors with 224/channels samples in each sector.

Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.

Update some demuxing and seeking fate tests.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

This also allows for removing some of the Vorbis-related hacks.

When reading sequentially, we are using the actual flag from the previous
frame, but when seeking we do not know what the previous window flag was, so
we need to read it from the bitstream.

Also, don't bother with saving/restoring data, av_init_packet doesn't
touch it.

This allows it to be used with get_bits without the thread of overreads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org