x86: vc1: fix and enable optimised loop filter

The problem is that the ssse3 psign instruction does the wrong
thing here.  Commit ea60dfe incorrectly removed a macro emulating
this instruction for pre-ssse3 code.  However, the emulation is
incorrect, and the code relies on the behaviour of the macro.
Specifically, the psign sets destination elements to zero where
the corresponding source element is zero, whereas the emulation
only negates destination elements where the source is negative.

Furthermore, the PSIGNW_MMX macro in x86util.asm is totally bogus,
which is why the original VC-1 code had an additional right shift
when using it.  Since the psign instruction cannot be used here,
skip all the macro hell and use the working instruction sequence
directly.

None of this was noticed due a stray return statement in
ff_vc1dsp_init_mmx() which meant that only the mmx version of the
loop filter was ever used (before being removed in ea60dfe).

Signed-off-by: Mans Rullgard <mans@mansr.com>
os_support: Include all the necessary headers for the win32 open function

io.h is required for open and _wopen, and fcntl.h is required for
the O_CREAT flag. On mingw, fcntl.h is included by os_support.h (and
the mingw fcntl.h includes io.h), but include it explicitly here
since this implementation requires it.

Also move the #undef open up. open must not be defined to ff_win32_open
while including the headers that declare the open function. On mingw,
this happened in os_support.h before open was redirected.

Signed-off-by: Martin Storsjö <martin@martin.st>
os_support: Don't compare a negative number against socket descriptors

The fds are unsigned integers in the windows definition of struct
sockfds. Due to this, the comparison if (fds[i].fd > n) was always
false.

Signed-off-by: Martin Storsjö <martin@martin.st>
network: Check for struct pollfd

We need to include winsock2.h here, to make sure we have the
real pollfd struct definition, if one exists, before defining the
fallback poll function.

Signed-off-by: Martin Storsjö <martin@martin.st>
os_support: Rename the poll fallback function to ff_poll

The fallback function is a non-static function, we shouldn't be
defining non-static functions outside of the proper ff/av prefix
namespaces.

This is especially important for a function like poll, which
other parties (other libraries, or executables linking these
libraries) also might provide similar but incompatible fallbacks for.

Signed-off-by: Martin Storsjö <martin@martin.st>
os_support: Add fallback definitions for stat flags

Mingw headers provide similar defines already (unconditional #defines,
without any #undef or #ifdef around it), while MSVC doesn't have
them.

Signed-off-by: Martin Storsjö <martin@martin.st>
cmdutils: Pass the actual chosen encoder to filter_codec_opts

This allows passing the right options to encoders when there's more
than one encoder for a certain codec id.

Signed-off-by: Martin Storsjö <martin@martin.st>
attributes: Add a definition of av_always_inline for MSVC

Signed-off-by: Martin Storsjö <martin@martin.st>
libavcodec: Add more AAC profiles

TODO: Are the mpeg2 ones official or specific to the fraunhofer
encoder?
Add support for AAC encoding via the external library fdk-aac
libfdk-aac: Support setting VBR bitrate modes

This requires a modified version of the fdk-aac library that
accepts setting these.

The general consensus seems to be to use the qscale flag for
enabling vbr mode. There doesn't seem to be any consistent way to
indicate the actual desired quality though. Both libfaac and
libmp3lame calculate avctx->global_quality / FF_QP2LAMBDA and set
that as the libraries VBR quality parameters, with wildly different
results. On libmp3lame, the VBR quality parameter is between 0 (best)
and 10 (worst), while the scale goes in the opposite direction for
libfaac, where higher quality values gives you better quality.

Therefore, for now, I just pass the actual value of
avctx->global_quality through. You can set it to values between 1
and 5:
1 - about 32 kbps/channel
2 - about 40 kbps/channel
3 - about 48-56 kbps/channel
4 - about 64 kbps/channel
5 - about 80-96 kbps/channel
avconv: Set audio filter time base to the sample rate

If the output frame size is smaller than the input sample rate,
and the input stream time base corresponds exactly to the input
frame size (getting input packet timestamps like 0, 1, 2, 3, 4 etc),
the output timestamps from the filter will be like
0, 1, 2, 3, 4, 4, 5 ..., leadning to non-monotone timestamps later.

A concrete example is input mp3 data having frame sizes of 1152
samples, transcoded to aac with 1024 sample frames.

By setting the audio filter time base to the sample rate, we will
get sensible timestamps for all output packets, regardless of
the ratio between the input and output frame sizes.

The actual rescaling call for rescaling timestamps before sending
them to the filter might be messed up though, comments welcome.