This affects Annex B streams (such as demuxed from .ts and others). It
also handles the format change in reinit-large_420_8-to-small_420_8.h264
correctly.
Instead of passing through the extradata, create it on the fly it from
the currently active SPS and PPS. Since reconstructing the PPS and SPS
NALs would be very complicated and verbose, we use the NALs as they
originally appeared in the bitstream.
The code for writing the extradata is somewhat derived from
libavformat/avc.c, but it's small and different enough that sharing it
is not really worth it.
We assume an upper bound of 4096 bytes for each raw SPS/PPS. It's hard
to determine an exact maximum size, but this value was was considered
high enough and safe.
Needed for the following VideotoolBox commit.
The current one, while correct, does not yield the best possible
results. The specificiations suggest another formula, which results
in quality gains in the decoded output from fate tests. This
justifies changing said formula.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is only (mis-)used to set the dsp fucntions clear_block(s). But
these functions always work on 16bits-wide elements, which make
the parameter useless and actually harmful, as it causes all content
on more than 8-bits to not use accelerated functions.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes deadlock with threads
Found-by: Paul B Mahol
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The variable is not a constant and can lead to race conditions
Fixes: repro.webm (not reproducable with FFmpeg alone)
Found-by: Dale Curtis <dalecurtis@google.com>
Tested-by: Dale Curtis <dalecurtis@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Commits 43bc5cf9 and c5371f77 add code for skipping initial zeros in mp3
packets. This code forgot to report to the user that data was skipped at
all.
Since audio codecs allow partial packet decoding, the user application
has to rely on the return value. It will remove the data reported as
consumed by the decoder, and feed it to the decoder again. This resulted
in the mp3 frame after the zero region to be decoded over and over
again, until the zero region was finally skipped by the application.
Fix this by including the amount of skipped bytes to the number of
consumed bytes returned by the decode call.
Fixes trac ticket #4890.
CID 1256 is specified as using the same table for luma and chroma,
which is the same as CID 1235 luma table. This is consistent with
the format supposedly being RGB, although most sequences seem to
actually be YCbCr-encoded.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
VideoToolbox also implements a software decoder for h264, and will fallback to
using it if the file cannot be decoded on the GPU. In these cases though,
we want the hwaccel to fail so that we can use the libavcodec software decoder
instead of the Apple one.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
* commit '4885bde3187a2bb0cae85b67796e07db233bf77f':
motion_est_template: Fix undefined left shift of negative number
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '948f3c19a8bd069768ca411212aaf8c1ed96b10d':
lavc: Make AVPacket.duration int64, and deprecate convergence_duration
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This bit is 1 in some samples, and seems to coincide with interlaced
mbs and CID1260. 2008 specs do not know about it, and maintain qscale
is 11 bits. This looks oversized, but may help larger bitdepths.
Currently, it leads to an obviously incorrect qscale value, meaning
its syntax is shifted by 1. However, reading 11 bits also leads to
obviously incorrect decoding: qscale seems to be 10 bits.
However, as most profiles still have 11bits qscale, the feature is
restricted to the CID1260 profile (this flag is dependent on
a higher-level flag located in the header).
The encoder writes 12 bits of syntax, last and first bits always 0,
which is now somewhat inconsistent with the decoder, but ends up with
the same effect (progressive + reserved bit).
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Note that convergence_duration had another meaning, one which was in
practice never used. The only real use for it was a 64 bit replacement
for the duration field. It's better just to make duration 64 bits, and
to get rid of it.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit '83847cc8fa97e0fc637a0962bafb837acdb6eacc':
qsvenc: do not try to close the encoder if the session is NULL
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
File libopenh264enc.c has been modified so that the encoder uses av_log()
to log messages (error, warning, info, etc.) instead of logging them
directly to stderr. At the time the encoder is created, the current
libav log level is mapped to an equivalent libopenh264 log level. This
log level, and a message logging function that invokes av_log() to
actually log messages, are then set on the encoder.
This contains further changes and simplifications by Michael Niedermayer
and Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
pix_fmt was declared presumably to shorten the argument passed to the function.
However, it is currently not being used for such a purpose.
This patch simply removes it instead.
This fixes -Wunused-but-set-variable reported at e.g:
http://fate.ffmpeg.org/log.cgi?time=20150919194249&log=compile&slot=x86_64-darwin-gcc-4.9.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It appears vdpau drivers can return constrained baseline as unsupported,
even if libvdpau knows about the symbol, and the main profile is
supported.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
