* commit '5d8bea3b':
  xtea: add av_xtea_alloc()
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

* commit 'b469832d':
  rc4: extend av_rc4_init() doxy
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

* commit 'ae365453':
  rc4: add av_rc4_alloc()
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

* commit '7a7df34c':
  blowfish: add av_blowfish_alloc()

Conflicts:
	doc/APIchanges
	libavutil/version.h
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

* commit 'cd4d9df2':
  asfdec: free AVDictionaries properly when closing the demuxer
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

* commit '90e648c5':
  fate/mp3: use the f32le format as output
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Commit 0a551cbe introduced "activation_bytes" option, and not specifying
this option (while calling ffmpeg / ffprobe) causes the program to quit
early. Before this commit, ffprobe was capable of processing metadata in
.aax files.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

This commit adds a short description to the aac_coder option of the
AAC encoder in order to be consistent with the other options.
Generally, right now, the 'FAAC' method works fine with speech and
low broadband spectrum audio. 'Fast' is just as the name suggests.
'ANMR' still needs work and 'Twoloop', the default, works well with
every type of audio.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>

Requested-by: Ganesh Ajjanagadde <gajjanag@mit.edu> (original patch reviewer)
This reverts commit 293e25be.

Reviewed-by: Nicolas George <george@nsup.org>
Merged-by: Michael Niedermayer <michael@niedermayer.cc>

This commit removes a redundant argument from the functions in aaccoder.
The argument lambda was redundant as it was just a copy of s->lambda,
to which all functions have access to anyway. This cleans up the function
pointers a bit which is helpful as there are a lot of other search_for_*
functions under development and with them populated it gets messy.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

libx264 added support for NV21 input recently.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

All DSD codes have 1 bit per sample.
https://en.wikipedia.org/wiki/Direct_Stream_DigitalSigned-off-by: Ihar A. Tumashyk <itumashyk@gmail.com>
Reviewed-by: Ganesh Ajjanagadde <gajjanag@mit.edu>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

add http_accept,
add http_handshake and move handshake logic there,
handle connection closing.
Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>

Signed-off-by: rogerdpack <rogerpack2005@gmail.com>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

* commit 'a0797950':
  fate/mp3: specify the number of output samples instead of filesize
Merged-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

Those tests test the float decoder, so it is preferable to dump the
decoder output directly instead of converting it to s16.

This is not dependent on the output format.

Can be used to fix Ticket4577
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Needed for old compilers like GCC 4.2

Tested by trac user brad. Fixes ticket #4745
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>

Between 1.5 and 2.5 times faster
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>

faults in fate otherwise breaks the terminal.
To reproduce, add a abort() into wav_read_header()
run make fate-acodec-adpcm-ima_wav

This reverts commit 92e62f49.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>