vorbisdec: read the previous window flag for long windows

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.
avcodec: add a Vorbis parser to get packet duration

This also allows for removing some of the Vorbis-related hacks.
thp: set audio packet durations
mpegaudio_parser: be less picky about the start position

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
mpegaudio_parser: do not ignore information from the first parsed frame

Update some demuxing and seeking fate tests.
tta demuxer: set packet duration
vqf: set packet duration

Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
psx-str: fix audio pts

Each packet has 18 sectors with 224/channels samples in each sector.
ea: fix audio pts

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.
segafilm: use the sample rate as the time base for audio streams
voc/avs: Do not change the sample rate mid-stream.

Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.
tiertexseq: set audio stream start time to 0

Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.
tiertexseq: set correct block_align for audio
wmaenc: limit block_align to MAX_CODED_SUPERFRAME_SIZE

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
wmaenc: limit allowed sample rate to 48kHz

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
wmaenc: require a large enough output buffer to prevent overwrites

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
wmaenc: check final frame size against output packet size

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.
wmaenc: return s->block_align instead of recalculating it
wmaenc: fix m/s stereo encoding for the first frame

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
fate: Add sunrast regression test

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