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

Prevents warnings because the dst and src overlap (are the same) in the
memcpy() inside the function.

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

Prevents using the invalid mode as an index in a static array, which
would generate invalid reads.

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

This prevents us from reading before the start of the buffer, and thus
prevents crashes resulting from this behaviour. Fixes bug 237.

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

x86-64 is guaranteed to have at least SSE2, therefore the MMX/MMX2
functions will never be used in practice.

On 64bit platforms with 32bit int, this means we won't have to sign-
extend the integer anymore.

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

Decodes 16-bit WMA Lossless encoded files. 24-bit is not supported yet.

Bitstream parser written by Andreas Öman with contributions from
Baptiste Coudurier and Ulion.

Includes a number of bug-fixes from Benjamin Larsson, Michael Niedermayer and
Konstantin Shishkov, shine and polish by Diego Biurrun.
Signed-off-by: Diego Biurrun <diego@biurrun.de>

Fixes CVE-2011-3937

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

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>

It's not supposed to be set by demuxers.

Set avg_frame_rate and r_frame_rate instead.

The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.

Demuxers are not supposed to set it.

Demuxers are not supposed to set it.
Set stream timebase and framerates instead (this is a cfr container with
no timestamps).

Demuxers are not supposed to set it.
Set r_frame_rate and avg_frame_rate instead.