Using this to change the encoder based on mono/stereo rather than looking
at the number of input channels.

This not only avoids sudden changes in audio quality, but also increases
quality when we don't have enough bits to code a wide stereo image.

opus_decode() and opus_multistream_decode() now apply soft clipping
before converting to 16-bit int. This should produce better a higher
quality result than hard clipping like we were doing before. The _float()
API isn't affected, but the clipping function is exported so users can
manually apply the soft clipping.

This is done using an adaptive beta and an estimate of the speech
and music detection confidence

Avoids biasing the decision when it's all silence/noise.

Variable duration option renamed to OPUS_SET_EXPERT_FRAME_DURATION,
with new API.
Also moves up the analysis to avoid having to do int->float conversion
on large buffers.

1) In cases where the SILK desired bandwidth went down, then quickly up, we
count get stuck in a mode with the LP variation going the wrong way.
2) Bandwidth detection can no longer force SILK to go below wideband to
avoid switches that require redundancy.

Now supports OPUS_AUTO and OPUS_BITRATE_MAX

Also fixes a bug with stereo streams where the initial memory was only
using the left channel.

The search is based on minimizing the bitrate increase due to transients
by considering we can reduce the "transient boost" by reducing the frame
size, at the cost of increasing the normal overhead.

Also, the encode+decode mode now produces an output of the same size
as the original.

This improves quality of CELT->SILK switches by not having the SILK encoder
attempt to encode a discontinuity. Overall, it's a small PQ improvement,
confirmed by some listening.

This was causing periodic glitches in the right channel in hybrid mode.

Fixes an MSVC double truncation warning.

This fixes a truncation warning in MSVC:

warning C4305: 'initializing' : truncation from 'double' to
'const opus_val16'.

opus_val16 is a float in the floating point build and an
int16 in the fixed-point build. The table isn't used in
the fixed-point build, but the compiler would presumedly
warn in that case as well.

Also fixes an issue with 40- and 60-ms frames using the repacketizer.

Can now work up to full-band and uses lsb_depth to fix the noise issue.

Calling PLC/FEC with a different size was not updating it

PLC and FEC now return exactly the number of samples specified for the
buffer rather than (usually) returning the size of the last packet.
Doc and tests are updated accordingly.

This avoids linker errors with VS2010 of the form
    failure during conversion to COFF: invalid file or corrupt
    [c:\users\giles\opus\source\opus_demo.vcxproj]

The problem does not occur if the project files are updated
to build under VS2012.

This was reported by Juri Aedla and is limited to reading memory up
to about 60 kB beyond the compressed buffer. This can only be triggered
by a compressed packet more than about 16 MB long, so it's not a problem
for RTP. In theory, it *could* crash an Ogg decoder if the memory just after
the incoming packet is out-of-range.

The README doesn't say anything about this particular define.
Refer people to config.h and the other build files for examples.