rtpdec_hevc: do not print an error message if the received packet has a valid header but lacks additional bytes as payload
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Consider a file created with something like:

    cat file1.mp3 file2.mp3 > result.mp3

Then if file2.mp3 has gapless information, result.mp3 would stop playing
something in the middle. This happens because the gapless info directs
the decoder to discard all samples after a certain position. To make
matters worse, the gapless info of file2.mp3 will be used when playing
the file1.mp3 part, because the gapless info is located at the end of
the file.

While handling concatenated gapless files correctly would be insane and
a lot of effort (especially without scanning the whole file on opening),
it's easy to prevent at least early EOF. Playback will happen to work,
even if it's slightly broken.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The fourth substream is being discarded, since its not raw audio data,
but an encoded Atmos stream which needs a specialized decoder.

Fixes decoding of the true hd stream from Transformers\ -\ Age\ of\ Extinction\ 2014\ 1080P-003.mkv
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Found-by: Kierank
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Found-by: Kierank
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* commit '103391ca':
  dca: Remove some commented-out cruft
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The extra bits have already been masked, so this was not doing anything.

Noticed-by: Justin Ruggles
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The issue lies in actually dead code ("for now it's not used").

Noticed-by: Justin Ruggles
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The code already had skipping of initial padding, but discarding
trailing frame padding was missing.

This is somewhat questionable, because it will make the decoder discard
any data after the declared file size in the LAME header. But note that
skipping full frames at the end of the stream is required. Encoders
actually create such files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Also add mmxext versions of vsad8 and vsad_intra8, and sse2 versions of
vsad16 and vsad_intra16.
Since vsad8 and vsad16 are not bitexact, they are accordingly marked as
approximate.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>

Found-by: Nicholas Robbins <nickrobbins-at-yahoo.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

As suggested, this just adds _DEFAULT_SOURCE while preserving
_SVID_SOURCE for whatever old systems still need it.
Signed-off-by: Daniel Bomar <dbdaniel42@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

See https://code.google.com/p/webp/issues/detail?id=206
for a description of the problem/fix.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This patch makes the decoder follow the recommendation of the spec.
There is some disagreement (see "[FFmpeg-devel] [PATCH]: libavcodec/webp")
about what would be best to be written in the spec, so in case the spec
is changed again, this potentially would need to be amended or reverted

in the latest git commits of libilbc developers removed WebRtc_xxx typedefs

This commit uses int types instead,
it's safe to apply also for previous versions since
WebRtc_Word16 was always a typedef of int16_t and
WebRtc_UWord16 a typedef of uint16_t
Reviewed-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

drop "#define CLIP av_clip_uint8"
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* postprocwork:
  postproc/postprocess: use av_strtok()
  postprocess: make some variables in pp_get_mode_by_name_and_quality() const
  postproc: simplify forwarding return codes
  libpostproc/postprocess: avoid some if()
  fate: add fate-filter-pp1

This is merged instead of just fast forward pushed due to a bug in the git hook
which does not allow commits to change filter-video.mak except merge commits.
filter-video.mak contains a few tabs, which are needed due to Makefile syntax
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This tests a few more filters with forced quantizers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

No point in having the sad8 functions separate now that the loop is no
longer unrolled.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>

* commit '2bb2c2bd':
  rtpenc_chain: Pass the initial time_base hint on to the chained muxer

See: [FFmpeg-devel] [PATCH] avformat/rtpenc_chain: Set timebase
Merged-by: Michael Niedermayer <michaelni@gmx.at>