* qatar/master:
smacker: Sanity check huffman tables found in the headers.
smacker: remove dead store
qdm2: Check data block size for bytes to bits overflow.
mxfdec: Fix files with essence containers larger than 2 GiB.
mxfdec: Employ correct printf conversion specifiers for POSIX int types.
vc1: always read the bfraction element for interlaced fields
fate: add XWD image regression test
lavf: prevent infinite loops while flushing in avformat_find_stream_info
matroskadec: Pad AAC extradata.
ismindex: Fix build on mingw
Conflicts:
libavformat/mxfdec.c
libavformat/utils.c
tests/lavf-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes not yet fixed parts of CVE-2011-3946.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously, it would not be read if refdist_flag was not set, however
according to the spec and the reference decoder, it should always be read.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master: (22 commits)
wma: Clip WMA1 and WMA2 frame length to 11 bits.
movenc: Don't require frame_size to be set for modes other than mov
doc: Update APIchanges with info on muxer flushing
movenc: Reindent a block
tools: Remove some unnecessary #undefs.
rv20: prevent calling ff_h263_decode_mba() with unset height/width
tools: K&R reformatting cosmetics
Ignore generated aviocat and ismindex tools.
build: Automatically include architecture-specific library Makefile snippets.
indeo5: prevent null pointer dereference on broken files
pktdumper: Use usleep instead of sleep
cosmetics: Remove some unnecessary block braces.
Drop unnecessary prefix from *sink* variable and struct names.
Add a tool for creating smooth streaming manifests
movdec: Calculate an average bit rate for fragmented streams, too
movenc: Write the sample rate instead of time scale in the stsd atom
movenc: Add a separate ismv/isma (smooth streaming) muxer
movenc: Allow the caller to decide on fragmentation
libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
movenc: Add support for writing fragmented mov files
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
ffmpeg.c
ffplay.c
libavfilter/Makefile
libavformat/Makefile
libavformat/avformat.h
libavformat/movenc.c
libavformat/movenc.h
libavformat/version.h
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If no data was seen for a stream decoder are returning 0 when fed with
empty packets for flushing. We can stop flushing when the decoder does
not return delayed delayed frames anymore. Changes try_decode_frame()
return value to got_picture or negative error.
CC: libav-stable@libav.org
The MDCT buffers in the decoder are only sized for up to 11 bits. The
reverse engineered documentation for WMA1/2 headers say that that for
all samplerates above 32kHz 11 bits are used. 12 and 13 bit support
were added for WMAPro. I was unable to make any Microsoft tools generate
a test file at a samplerate above 48kHz.
Discovered by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The field frame_size isn't written to the output anywhere except
than in mov.
This facilitates stream copy from formats that don't set frame_size.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add some space around operators and wrap a comment
that extends past the 80 char "limit"/guideline.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes a double release of the current frame on deinit.
Fixes CVE-2011-3934
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The index of the motion vector has to be checked before being
multiplied by 2 for the array index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes part2 of CVE-2011-3929
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes part1 of CVE-2011-3929
Possibly fixes part of CVE-2011-3936
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
MinGW doesn't have sleep, only _sleep (which is deprecated),
Sleep (which is defined in winbase.h and not in the standard
C headers) and usleep.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can also optionally split the file into individual fragments,
which allows it to be served from any web server without any
server side support.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes calculation of trackDuration if the MOVIentry array
is cleared. This is required by the fragmentation support in the
next patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit makes the check specific to the case that needs it.
Regression was introduced by
commit 62adc60b97
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Fri Dec 16 06:13:04 2011 +0100
avidec: Check that the header chunks fit in the available filesize.
Fixes Ticket771
Bug found by: Diana Elena Muscalu
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Although it has been deprecated for a long time, its intended
replacement (request_channel_layout) is not actually used anywhere, so
request_channels is currently the only way to access that functionality.
Allows our users to still build against a libpostproc with the old
API/ABI. Distributions can use this option to defer the soname bump.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously this was just checked in case of slice threads,
but frame threads do not support this either currently.
Making them support this is of course the long term goal
Fixes bug155
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reduces the delay when opening the video with quicktime.
Idea-by: Maksym Veremeyenko <verem@m1stereo.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The previous implementation assumed that a new picture would always
supersede the previous picture. Similarly, presentation segments
were assumed to pertain to the most-recently-read picture.
However, each presentation segment may refer to 0 or more pictures
by their ID. Picture IDs may repeat, and a repeated picture ID
indicates that the old picture for that ID is no longer needed
and may be discarded.
The new implementation allocates a buffer with one slot for each
possible picture ID (the picture ID is a 16-bit field) and
properly decodes presentation segments so that all relevant
pictures are output upon encountering a display segment.
Given that most PGS streams are unlikely to use more than a small
fraction of the available picture IDs, it would probably be better
to use a more memory-efficient data structure. I'm lazy though, so
I leave this to a more motivated individual.
I've tested the code with MKV files in VLC (a recent revision from
their git repo) and with HandBrake (a version that I hacked up to
use ffmpeg's PGS subtitle decoder).
Review-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes memory corruption when seeking in broken streams.
a random mpeg4 in nut file was used to debug.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This matches the spec as well as the reference decoder, and fixes a bug
with interlaced frame decoding.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master: (25 commits)
riff: fix invalid av_freep() calls on EOF in ff_read_riff_info
pam: Fix a typo that broke writing and reading PAM files.
mxfdec: fix memleak on av_realloc failures
mxfdec: Do not parse slices or DeltaEntryArrays.
mxfdec: hybrid demuxing/seeking solution
mxfdec: Add Avid's essence element key.
mfxdec: Separate mxf_essence_container_uls for audio and video.
mxfdec: Compute packet offsets properly.
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack.
mxfdec: use av_dlog() for 'no corresponding source package found'
mxfdec: Make mxf->partitions sorted by offset.
mxfdec: parse ThisPartition
mxfdec: Speed up metadata and index parsing.
mxfdec: Make sure DataDefinition is consistent between material track and source track.
mxfdec: add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
mxfdec: Add hack that adjusts the n_delta calculation when system items are present.
mxfdec: Parse IndexTableSegments and convert them into AVIndexEntry arrays.
mxfdec: Move FooterPartition to MXFContext and make sure it is never zero.
mxfdec: check return value of avio_seek
mxfdec: skip to end of structural sets
...
Conflicts:
configure
libavcodec/pnm.c
libavformat/mxfdec.c
libavformat/riff.c
libavformat/rtsp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This was a regression that came in when I switched to using the
h.264 annex b filter all the time. As the filter modifies extradata,
its use violates the statelessness assumption that exists in the
'ffmpeg' command line tool, and maybe elsewhere. It assumes that
a docoder can be reinitalised and pointed to an existing stream and
get the same results.
For now, the only way to meet this requirement is to backup the
extradata.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Makes it possible to select the name/path of the tool for compiling
the non-inline assembly code.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
It is a really bad idea to assign a video codec id
when we have set codec_type to audio and vice versa.
Prevents detection of mp2 in mxf as mpeg2video.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers are not needlessly
parsed. This speeds up parsing large files a lot.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes 0001GL.MXF.V1.mxf_opatom.mxf and 0001GL00.MXF.A1.mxf_opatom.mxf
getting two streams each due to both using the same SourcePackageID.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes reading of partition packs. The code stops reading after the
operational pattern and should skip the array of essence container
labels that follow.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Now that 0.8 is out we can reapply this commit. It breaks shared
avserver builds due to avserver using internal libavformat symbols,
which are now hidden, so this commit also disables avserver with
--enable-shared.
the written length was off by 2 causing aac decoders to fail with the data.
lucky the encoder was marked as experimental and not used much
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some applications use the j2c extension for jpeg2000 codestream files.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.
This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.
Signed-off-by: Martin Storsjö <martin@martin.st>
This check isn't relevant in the way the code currently works.
Also change a case of if (x == 0) into if (!x).
Signed-off-by: Martin Storsjö <martin@martin.st>
Original commit:
commit 2473a45c85
Author: Janne Grunau <janne-libav@jannau.net>
Date: Wed Jan 18 10:53:41 2012 +0100
threads: change the default for threads back to 1
Using threaded decoding by default breaks backward compatibility if
AVHWAccel is used or if an appliction sets threadunsafe callbacks.
Avconv and avplay still use -threads auto if not specified.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec: Use our own SSRC in the SDES field when sending RRs
Finalize changelog for 0.8 Release
Prepare for 0.8 Release
threads: change the default for threads back to 1
threads: update slice_count and slice_offset from user context
aviocat: Remove useless includes
doc/APIChanges: fill in missing dates and hashes
Revert "avserver: fix build after the next bump."
mpegaudiodec: switch error detection check to AV_EF_BUFFER
lavf: rename fer option and document resulting (f_)err_detect options
lavc: rename err_filter option to err_detect and document it
mpegvideo: fix invalid memory access for small video dimensions
movenc: Reorder entries in the MOVIentry struct, for tigheter packing
rtsp: Remove extern declarations for variables that don't exist
aviocat: Flush the output before closing
Conflicts:
Changelog
RELEASE
libavcodec/mpegaudiodec.c
libavcodec/pthread.c
libavformat/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using threaded decoding by default breaks backward compatibility if
AVHWAccel is used or if an appliction sets threadunsafe callbacks.
Avconv and avplay still use -threads auto if not specified.
This temporarily (until 0.8 is released) reverts commit
8e1340abc3. That commit breaks shared
builds because of symbol hiding. Reverting it will enable shared builds
for 0.8
When either video dimension is only one macroblock, subtractions
based on v_edge_pos and the macroblock size may be negative. In
that situation, an unsigned comparison isn't sufficent to test for
MV overruns, because a limit of (unsigned)-1 will let any other
value pass.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
That way all mix levels as exported by the parser
will have the same meaning.
Previously the 3bit center mix level for eac3 was
used to index in a 4 entry table leading to out of array reads.
this change removes the table and offsets the ac3 variable by 4
so it matches the meanings for eac3 except the reserved case.
The reserved case is then explicitly handled.
Idea-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Sometimes the scan finds nothing that qualifies for addition to
the array and pos is zero after the loops. The code forces pos to
1 and the array is then processed as if it had one valid element in it,
producing some amusing but not very useful results.
I don't see the rationale for this. If pos is zero coming out of the
loops, the only appropriate thing to do is set t->angle to zero. The
attached patch does that. It's worked properly in several tests so far.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
As the test is run during fate and the benchmark is useless for fate
this very slightly speeds up fate. Its also consistent with the other
tests.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Add a tool that uses avio to read and write, doing a plain copy of data
ARM: fix build with FFT enabled and MDCT disabled
lavf: force single-threaded decoding in avformat_find_stream_info
avidec: migrate last of lavf from FF_ER_* to AV_EF_*
avserver: fix build after the next bump.
Conflicts:
libavformat/Makefile
libavformat/avidec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Originally, sizeof(struct MOVIentry) was 48, after the reordering,
it is 40 in my build configuration.
When writing really long mov/mp4 files, this can make a difference
- this saves a bit over 2 MB of memory per hour of video (down to
10.3 MB per hour from 12.3 MB per hour initially) for a video with
75 packets per second - 25 fps + 50 audio packets (which is the
case for AMR audio).
Signed-off-by: Martin Storsjö <martin@martin.st>
The H.264 decoder needs SPS and PPS for initialization during
multi-threaded decoding. When probed single-threaded SPS and PPS are
copied to extradata and are available for proper initialization of
the decoder before the first frame is decoded.
* qatar/master:
mpeg12: check for available bits to avoid an infinite loop
fate: add some shorthands to run groups of tests
fate: Give some tests more sensible names.
cosmetics: Rename ffsink to avsink.
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/dpcm.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/pcm.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/wma.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This one was missed in the previous fraps fix, the
allocation is exactly the same in both cases.
Fixes fraps-v5 under valgrind.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The ABI differences are negligible, but its easier for all if
all distros have libpostproc HEAD under the same soname and
debian bumped soname without consulting upstream, so as silly as
it is following this is probably the least pain for all.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is needed in case the get_buffer() callback doesnt set
width/height.
Ideally all decoders would make calls through some wraper
to the callbacks and that wraper would call ff_init_buffer_info()
But until thats done, the default reget buffer must call this
itself as it needs the values for the changed size check later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tta: cast output data pointer to the correct type
avconv: fix -frames for video encoders with delay.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
There is no guarantee that the casted double which is assigned to the
variable will be contained in an int (also if it is almost sure for most
non-alien architectures).
Use long long int to contain such values instead of an int, which is
required to contain at least 64 bits, so it is guaranteed to contain also
int64_t values, which are used by some fields.
In particular, should fix trac ticket #921.
This fixes parallel FATE (make fate -j4) failing under valgrind with:
Syscall param ioctl(TCSET{S,SW,SF}) points to uninitialised byte(s)
at 0x5D98B23: tcsetattr (tcsetattr.c:88)
by 0x43D66C: term_init (ffmpeg.c:734)
by 0x43CD8D: main (ffmpeg.c:5071)
Address 0x7fefffdd0 is on thread 1's stack
Uninitialised value was created by a stack allocation
at 0x43D5B0: term_init (ffmpeg.c:716)
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The same as av_fast_malloc but uses av_mallocz and keeps extra
always-0 padding.
This does not mean the memory will be 0-initialized after each call,
but actually only after each growth of the buffer.
However this makes sure that
a) all data anywhere in the buffer is always initialized
b) the padding is always 0
c) the user does not have to bother with adding the padding themselves
Fixes another valgrind warning about use of uninitialized data,
this time with fate-vsynth1-jpegls.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This combination is quite odd and almost certainly a bug if
it happens.
Reviewed-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The tool is useful for testing the internal arithmetic evaluation engine
(indeed I plan to use it in FATE), and provides a handy calculator when
you can't rely on bc ;-).
They allow to implement the if/then/else logic, which cannot be
implemented otherwise.
For example the expression:
A*B + not(A)*C
always evaluates to NaN if B is NaN, even in the case where A is 0.
* qatar/master:
rv34: add NEON rv34_idct_add
rv34: 1-pass inter MB reconstruction
add SMJPEG muxer
avformat: split out common SMJPEG code
pictordec: Use bytestream2 functions
avconv: use avcodec_encode_audio2()
pcmenc: use AVCodec.encode2()
avcodec: bump minor version and add APIChanges for the new audio encoding API
avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
avcodec: add a public function, avcodec_fill_audio_frame().
rv34: Intra 16x16 handling
rv34: Inter/intra MB code split
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/pictordec.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/rv34dsp.asm
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Overall almost 4% faster, idct_add down from 350 to 85 cycles, idct_dc_add
down from 83 to 30 cycles.
squash: rv34 idct rearrange partial register loads
This reworks the frame skipping code such that the reference
buffers are still updated according to the header.
However it also ensures that the current frame will not end
up in any reference buffer.
Also fixes a hang with frame-multithreading, probably because
get_buffer was already called and would have reset the progress,
however the frame could remain in framep due to the missing update
(or it could be assigned to next_framep and a skip_frame skip would
then write it into framep - there might be even more failure modes).
Sample might become available at samples/nsv/vp8.nsv
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Enhances seeking by demuxing until the requested timestamp is reached within
the segment selected by the seek code using the playlist info.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is required by the spec and fixes video-1frag.ogg.48.ogg. (FPE)
Based on the debuging work of Oana Stratulat and ubitux.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
mov: cosmetics - move a line to a better position and add a comment
Oana Andreea Stratulat submitted a similar patch to trac, but forgot
to notify the ML about it.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.
This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.
* qatar/master:
fate: split ADPCM and DPCM test references into separate files.
mov, mxfdec: Employ more meaningful return values.
lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer()
wavpack: fix clipping for 32-bit lossy mode
vb: Use bytestream2 functions
Conflicts:
libavcodec/utils.c
libavcodec/vb.c
libavformat/mxfdec.c
tests/fate/dpcm.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Extract processing of intra 16x16 blocks from intra macroblock
processing.
Also implement a function performing inverse transform and block
reconstruction for DC-only blocks in 1 pass instead of 2.
Split inter/intra macroblock handling code. This will allow further
optimizations such as performing inverse transform and block reconstruction
in a single pass as well as specialize code.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Do not fail audio decoding with avcodec_decode_audio3 if user has set a
custom get_buffer. Strictly speaking, this was never allowed by the API,
but it seems that some software packages did so anyways. In order to
unbreak applications (cf. http://bugs.debian.org/655890), this change
clarifies the API and overrides the custom get_buffer() with the defaults.
This change is inspired by a similar
commit (c3846e3eba) in FFmpeg.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Current code would just return uninitialized data with no way
to detect this condition.
Instead, fill the whole GUID with 0 in that case.
Fixes valgrind uninitialized data errors in fate-seek-lavf_asf.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reference decoder clips data before shifting it to final range and also
forces 32-bit lossy mode to be actually 24-bit lossy mode in order to be
able to perform proper clipping.
This is not a real error and memsetting always even when the
size did not change is overkill, but it still should be
an acceptable trade-off.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
sgidec: Use bytestream2 functions to prevent buffer overreads.
cosmetics: Move static and inline attributes to more standard places.
configure: provide libavfilter/version.h header to get_version()
swscale: change yuv2yuvX code to use cpuflag().
libx264: Don't leave max_b_frames as -1 if the user didn't set it
FATE: convert output to rgba for the targa tests which currently output pal8
fate: add missing reference files for targa tests in 9c2f9b0e2
FATE: enable the 2 remaining targa conformance suite tests
targa: add support for rgb555 palette
FATE: fix targa tests on big-endian systems
Conflicts:
libavcodec/sgidec.c
libavcodec/targa.c
libswscale/x86/output.asm
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At the very least this should fix warnings about unused static
functions if one or more of these is not defined.
However even compilation might be broken if the compiler does
not optimize the function away completely.
This actually happens in case of the AVX function, since the
function pointer is used in an assignment that is not under
an #if and thus probably only optimized away after the function
was already marked as used.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We do this for all other codec_tag checks in mpegvideo*/h26*
doing it here too makes the code more consistent.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While we correctly "register" the side data when we split it,
the application (in this case FFmpeg) might not update the
AVPacket pool it uses to finally free the packet, thus
causing a leak.
This also makes the av_dup_packet unnecessary which could
cause an even worse leak in this situation.
Also change the code to not modify the user-provide AVPacket at all.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* cus/stable:
ffplay: silence buffer size must be a multiple of frame size
ffplay: use swr_set_compensation for audio synchronization
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: add tests for targa
ARM: fix Thumb-mode simple_idct_arm
ARM: 4-byte align start of all asm functions
rgb2rgb: rgb12to15()
swscale-test: fix stack overread.
swscale: fix invalid conversions and memory problems.
cabac: split cabac.h into declarations and function definitions
cabac: Mark ff_h264_mps_state array as static, it is only used within cabac.c.
cabac: Remove ff_h264_lps_state array.
Conflicts:
libswscale/rgb2rgb.h
libswscale/swscale_unscaled.c
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some of these encoders may produce invalid bitstreams, which should not
be done without the user knowing.
Some of these decoders may be unfinished and may contain security issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
max_b_frames is initialized to -1 for libx264, to allow
distinguishing between an explicit user set 0 and a default not
touched 0 (see bb73cda2).
If max_b_frames is left as -1, this affects dts generation (where
expressions like max_b_frames != 0 are used), so make sure it is
left at the default 0 after the libx264 init function returns.
This avoids unnecessarily producing dts != pts when using
profile=baseline.
Signed-off-by: Martin Storsjö <martin@martin.st>
The alignment directive must obviously precede the label.
This was never noticed in ARM mode since the location is
already aligned there.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Due to apprent bugs in the GNU assembler and/or linker, relocations
can be incorrectly processed if the alignment of a Thumb instruction
is changed in the output file compared to the input object.
This fixes crashes in h264 decoding with Thumb enabled. No effect in
ARM mode since everything is 4-byte aligned there.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes a segmentation fault when doing a transcoding and a stream
copy of the same input stream at the same time, e.g.:
ffmpeg -i in.mkv -c:v mpeg2video transcode.m2v -c copy copy.ts
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With the added benefit that allowing -segment_list_size 0 makes it
possible to keep all segment entries in the list file.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes problems where rgbToRgbWrapper() is called even though it doesn't
support this particular conversion (e.g. converting from RGB444 to
anything). Thirdly, fixes issues where rgbToRgbWrapper() is called for
non-native endiannness conversions (e.g. RGB555BE on a LE system).
Fourthly, fixes crashes when converting from e.g. monowhite to
monowhite, which calls planarCopyWrapper() and overwrites/reads because
n_bytes != n_pixels.
* qatar/master: (21 commits)
utils: Check for extradata size overflows.
ARM: rv34: fix asm syntax in dc transform functions
avio: Fix the value of the deprecated URL_FLAG_NONBLOCK
rv34: fix and optimise frame dependency checking
rv34: NEON optimised dc only inverse transform
avprobe: use avio_size() instead of deprecated AVFormatContext.file_size.
ffmenc: remove references to deprecated AVFormatContext.timestamp.
lavf: undeprecate read_seek().
avserver: remove code using deprecated CODEC_CAP_PARSE_ONLY.
lavc: replace some remaining FF_I_TYPE with AV_PICTURE_TYPE_I
lavc: ifdef out parse_only AVOption
nellymoserdec: SAMPLE_FMT -> AV_SAMPLE_FMT
mpegvideo_enc: ifdef out/replace references to deprecated codec flags.
riff: remove references to sonic codec ids
indeo4: add some missing static and const qualifiers
rv34: DC-only inverse transform
avconv: use AVFrame.width/height/format instead of corresponding AVCodecContext fields
lavfi: move version macros to a new installed header version.h
vsrc_buffer: release the buffer on uninit.
rgb2rgb: rgb12tobgr12()
...
Conflicts:
avconv.c
doc/APIchanges
ffprobe.c
libavfilter/Makefile
libavfilter/avfilter.h
libswscale/rgb2rgb.c
libswscale/rgb2rgb.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes standalone compilation of some decoders with --disable-optimizations.
cabac.h defines some inline functions that use symbols from cabac.c. Without
optimizations these inline functions are not eliminated and linking fails with
references to non-existing symbols.
Splitting the inline functions off into their own header and only #including
it in the places where the inline functions are used allows #including cabac.h
from anywhere without ill effects.
This isn't used in practice anywhere within libav at the moment,
but change it for consistency until it is removed.
URL_RDONLY/WRONLY were fixed in commit 5b81e29593 (after the
values that actually were used were changed at the major bump,
in commit cbea3ac8), but this flag was unintentionally left unfixed.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using the double variant causes several pointless conversions between
double and int.
Worse, one of the conversions is in an inner loop together with a
function using MMX, resulting in undefined behaviour.
Based on debugging by Ray Simard.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Tested-by: Ray Simard <rhs.ffmpeg@sylvan-glade.com>
The sporadic threading errors during fate-rv30 were caused by calling
ff_thread_await_progress with mb row -1 as argument. That returns
immediately since progress is initialized to -1. Not yet computed
motion vectors from the reference could be used for the first
macroblocks.
This fixes dithering for rgb555le, it appears gcc had moved the
setup of the variables after the asm or something like that.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_destruct_packet() always frees the packet data even when the demuxer
is going to re-use it, thus causing crashes when decoding audio
frames (as implemented in a pending patch).
av_free_packet() is used instead, as it allows each demuxer to set the
right packet data releasing mechanism through the pkt->destruct callback.
When decoding coefficients, detect whether the block is DC-only, and take
advantage of this knowledge to perform DC-only inverse transform.
This is achieved by:
- first, changing the 108x4 element modulo_three_table into a 108 element
table (kind of base4), and accessing each value using mask and shifts.
- then, checking low bits for 0 (as they represent the presence of higher
frequency coefficients)
Also provide x86 SIMD code for the DC-only inverse transform.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master:
fft: init functions with INIT_XMM/YMM.
pcmenc: set frame_size to 0.
gsm demuxer: use generic seeking instead of a gsm-specific function.
gsm demuxer: return packets with only 1 gsm block at a time.
avcodec: add GSM parser
doc: Replace ffmpeg references in avserver config file by avconv.
doc: Fix names of av_log color environment variables.
Fix a bunch of platform name and other typos.
Add some missing changelog entries and release 0.8_beta2
No longer build libpostproc by default
wtv: fix memleaks during normal operation
threads: add CODEC_CAP_AUTO_THREADS for libvpx and xavs
Conflicts:
Changelog
RELEASE
cmdutils.c
configure
doc/ffserver.conf
doc/platform.texi
ffplay.c
libavcodec/Makefile
libavcodec/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also change synchronize_audio to only calculate the wanted number of samples
instead of doing the actual synchronization, and make swr_convert handle the
sample addition or deletion.
This new method replaces the old buggy way which simply deleted or
multiplied samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
This is required to handle clobbering of XMM registers on Win64
correctly. Fixes FFT and all tests depending on FFT on Win64.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes some cases where the clipping was entirely missing.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Thanks (for the comments and review) -to: Reimar, beastd, Ronald
This reverts commit 0efd48dfd1.
Reason for the revert is that the code seems based on some
misunderstanding on how the code works.
Conflicts:
libavdevice/v4l2.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The WAVE demuxer returns packets with many blocks per frame, which needs to be
parsed into single blocks. This has a side-effect of fixing the timestamps.
* qatar/master: (22 commits)
rv34: frame-level multi-threading
mpegvideo: claim ownership of referenced pictures
aacsbr: prevent out of bounds memcpy().
ipmovie: fix pts for CODEC_ID_INTERPLAY_DPCM
sierravmd: fix audio pts
bethsoftvideo: Use bytestream2 functions to prevent buffer overreads.
bmpenc: support for PIX_FMT_RGB444
swscale: fix crash in fast_bilinear code when compiled with -mred-zone.
swscale: specify register type.
rv34: use get_bits_left()
avconv: reinitialize the filtergraph on resolution change.
vsrc_buffer: error on changing frame parameters.
avconv: fix -copyinkf.
fate: Update file checksums after the mov muxer change in a78dbada55
movenc: Don't store a nonzero creation time if nothing was set by the caller
bmpdec: support for rgb444 with bitfields compression
rgb2rgb: allow conversion for <15 bpp
doc: fix stray reference to FFmpeg
v4l2: use C99 struct initializer
v4l2: poll the file descriptor
...
Conflicts:
avconv.c
libavcodec/aacsbr.c
libavcodec/bethsoftvideo.c
libavcodec/kmvc.c
libavdevice/v4l2.c
libavfilter/vsrc_buffer.c
libswscale/swscale_unscaled.c
libswscale/x86/input.asm
tests/ref/acodec/alac
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
tests/ref/vsynth1/dnxhd_1080i
tests/ref/vsynth1/mpeg4
tests/ref/vsynth1/qtrle
tests/ref/vsynth1/svq1
tests/ref/vsynth2/dnxhd_1080i
tests/ref/vsynth2/mpeg4
tests/ref/vsynth2/qtrle
tests/ref/vsynth2/svq1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* richardpl/sws:
rgb2rgb: remove unused bgr8torgb8()
rgb2rgb: rgb12tobgr12()
rgb2rgb: allow conversion for <15 bpp
bmpenc: support for PIX_FMT_RGB444
bmpdec: support for rgb444 with bitfields compression
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Statistics for bourne.rmvb -an -f null
1 thread: 37.12s user 0.03s system 99% cpu 37.174 total
2 threads: 47.63s user 0.24s system 185% cpu 25.807 total
4 threads: 41.21s user 0.30s system 327% cpu 12.674 total
Under certain conditions pictures could be released before they were
returned with frame-threading. Broken mv computation in the upcoming
rv34 frame-threading patch was caused by this.
To prevent contexts from running out of available pictures the loop
releasing "unused" pictures has to be run for B frames too.
Fixes Libav Bug 195.
This doesn't make the code handle sample rate or upsample/downsample
change properly but this is still a good sanity check.
Based on change by Michael Niedermayer.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This reverts commit 77d88b872d.
The revert fixes actual overflows and a segfault as the variables
are signed and can be negative.
Conflicts:
libswscale/swscale.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If the creation time is stored in the file as a zero, the
mov demuxer skips exporting the creation time. Currently,
files muxed without a creation time get demuxed with a
Jan 1st 1970 creation timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
This test does not work on all platforms and until it does
it just hides new failures, which is really bad.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: Add tests for more AAC features.
aacps: Add missing newline in error message.
fate: Add tests for vc1/wmapro in ism.
aacdec: Add a fate test for 5.1 channel SBR.
aacdec: Turn off PS for multichannel files that use PCE based configs.
cabac: remove put_cabac_u/ueg from cabac-test.
swscale: RGB4444 and BGR444 input
FATE: add test for xWMA demuxer.
FATE: add test for SMJPEG demuxer and associated IMA ADPCM audio decoder.
mpegaudiodec: optimized iMDCT transform
mpegaudiodec: change imdct window arrangment for better pointer alignment
mpegaudiodec: move imdct and windowing function to mpegaudiodsp
mpegaudiodec: interleave iMDCT buffer to simplify future SIMD implementations
swscale: convert yuy2/uyvy/nv12/nv21ToY/UV from inline asm to yasm.
FATE: test to exercise WTV demuxer.
mjpegdec: K&R formatting cosmetics
swscale: K&R formatting cosmetics for code examples
swscale: K&R reformatting cosmetics for header files
FATE test: cvid-grayscale; ensures that the grayscale Cinepak variant is exercised.
Conflicts:
libavcodec/cabac.c
libavcodec/mjpegdec.c
libavcodec/mpegaudiodec.c
libavcodec/mpegaudiodsp.c
libavcodec/mpegaudiodsp.h
libavcodec/mpegaudiodsp_template.c
libavcodec/x86/Makefile
libavcodec/x86/imdct36_sse.asm
libavcodec/x86/mpegaudiodec_mmx.c
libswscale/swscale-test.c
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/fate/demux.mak
tests/fate/microsoft.mak
tests/fate/video.mak
tests/fate/wma.mak
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows to get out of ffplay if it or SDL got stuck.
This for example happens when the audio driver is playing something
else and doesnt support mixing multiple sources.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
The functions are not used in any part of Libav, therefore testing them in the
cabac-test is unnecessary. Since this makes them unused, remove the functions.
Print a "\n" at the end of each section, also print the section name in
the section print function, print the chapter name only in case the
chapter contains multiple entries.
Increase textual output readability - different sections can be
distinguished more easily.
Using an always_inline function makes the memcpy length a constant,
any reasonable compiler will replace it by a single mov instruction
without us having to duplicate the actual code.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes reads of uninitialized data by the parser when running
FATE sample h264-conformance/SL1_SVA_B.264.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The new version returns AVERROR(EINVAL) is the specified paramters are invalid,
and also creates the resampler if none was used so far.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(Does not attempt to decode percetual audio data inside.)
Code coverage: libavformat/xwma.c: 3% -> 75%
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(Don't attempt to decode JPEG data.)
Code coverage: libavformat/smjpeg.c: 0% -> 69%
libavcodec/adpcm.c: 0% -> 10% (fresh run); 92.4% -> 93% following a FATE run
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
>> time ./avconv -i file.avi -f null -
Before : real 0m7.784s
After : real 0m3.662s
Tested on a Intel Core i3 Processor (2 cores, 4 threads).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (29 commits)
cabac: Move code only used within the CABAC test program into the test program.
vp56: Drop unnecessary cabac.h #include.
h264-test: Initialize AVCodecContext.av_class.
build: Skip compiling network.h and rtsp.h if networking is not enabled.
cosmetics: drop some pointless parentheses
Disable annoying warning without changing behavior
faq: Solutions for common problems with sample paths when running FATE.
avcodec: attempt to clarify the CODEC_CAP_DELAY documentation
avcodec: fix avcodec_encode_audio() documentation.
FATE: xmv-demux test; exercise the XMV demuxer without decoding the perceptual codecs inside.
vqf: recognize more metadata chunks
FATE test: BMV demuxer and associated video and audio decoders.
FATE: indeo4 video decoder test.
FATE: update xxan-wc4 test to a sample with more code coverage.
Change the recent h264_mp4toannexb bitstream filter test to output to an elementary stream rather than a program stream.
g722enc: validate AVCodecContext.trellis
g722enc: set frame_size, and also handle an odd number of input samples
g722enc: split encoding into separate functions for trellis vs. no trellis
mpegaudiodec: Use clearer pointer math
tta: Fix returned error code at EOF
...
Conflicts:
libavcodec/h264.c
libavcodec/indeo3.c
libavcodec/interplayvideo.c
libavcodec/ivi_common.c
libavcodec/libxvidff.c
libavcodec/mpegvideo.c
libavcodec/ppc/mpegvideo_altivec.c
libavcodec/tta.c
libavcodec/utils.c
libavfilter/vsrc_buffer.c
libavformat/Makefile
tests/fate/indeo.mak
tests/ref/acodec/g722
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add -show_frames option to ffprobe.
Partially based on the work of Thomas Kuehnel <kuehnelth@googlemail.com>
for SOCIS 2011.
The wicked idea of creating a special "packets_and_frames" container for
structured formats (JSON and XML) comes from Clément.
rtsp.h relies on network.h and the latter conditionally defines fallback OS
structures that rely on configure tests, which are only run if networking
is enabled.
This fixes various problems with getting stream info. For example playback of the
file of Ticket88. Multithreaded find_stream_info should be reenabled
once it works correctly
This partly reverts 212fd3a1f1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The previous sample used for this test only contained type 0 frames.
Replace it with a sample that also features type 1 frames.
Code coverage:
libavcodec/xxan.c: 72% -> 89%
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
flicvideo: fix invalid reads
vorbis: Avoid some out-of-bounds reads
vqf: add more known extensions
cabac: remove unused function renorm_cabac_decoder
h264: Only use symbols from the SVQ3 decoder under proper conditionals.
add bytestream2_tell() and bytestream2_seek() functions
parsers: initialize MpegEncContext.slice_context_count to 1
spdifenc: use special alignment for DTS-HD length_code
Conflicts:
libavcodec/flicvideo.c
libavcodec/h264.c
libavcodec/mpeg4video_parser.c
libavcodec/vorbis.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The decoders should not only be flushed on EOF or error, but also when
e.g. probe size was reached.
It is best to just always flush by default and only disable it
explicitly when we know that we have everything we need.
Fixes trac ticket #879.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (21 commits)
ipmovie: do not read audio packets before the codec is known
truemotion2: check size before GetBitContext initialisation
avio: Only do implicit network initialization for network protocols
avio: Add an URLProtocol flag for indicating that a protocol uses network
adpcm: ADPCM Electronic Arts has always two channels
matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
fate: Add missing reference file from 9b4767e4.
mov: Support MOV_CH_LAYOUT_USE_DESCRIPTIONS for labeled descriptions.
4xm: Prevent buffer overreads.
mjpegdec: parse RSTn to prevent skipping other data in mjpeg_decode_scan
vp3: add fate test for non-zero last coefficient
vp3: fix streams with non-zero last coefficient
swscale: remove unused U/V arguments from yuv2rgb_write().
timer: K&R formatting cosmetics
lavf: cosmetics, reformat av_read_frame().
lavf: refactor av_read_frame() to make it easier to understand.
Report an error if pitch_lag is zero in AMR-NB decoder.
Revert "4xm: Prevent buffer overreads."
4xm: Prevent buffer overreads.
4xm: pass the correct remaining buffer size to decode_i2_frame().
...
Conflicts:
libavcodec/4xm.c
libavcodec/mjpegdec.c
libavcodec/truemotion2.c
libavformat/ipmovie.c
libavformat/mov_chan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The mpeg4 video, H264 and VC-1 parser hold (directly or indirectly)
a MpegEncContext in their private context. Since they do not call the
common mpegvideo init function slice_context_count has explicitly set
to 1.
Prevents a null pointer dereference in the h264 parser and fixes
bug 193.
Author: Mans Rullgard <mans@mansr.com>
Date: Sun Dec 11 21:41:59 2011 +0000
x86: cabac: replace explicit memory references with "m" operands
This replaces the explicit offset(reg) memory references with
"m" operands for the same locations. As a result, one fewer
register operand is needed for these inline asm statements.
This change appears to have broken compilation on darwin, and subsequent
fixes by martin (which did not fix compilation) removed the register
advantage, thus this change seems not a good idea to keep.
See: http://fate.ffmpeg.org/log.cgi?time=20120103122446&log=compile&slot=i386-darwin-llvm-gcc-4.2.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The implicit network initialization is set to be removed in the
future, but is kept for compatibility. By not doing the implicit
initialization for non-network protocols, we avoid the warning
about avformat_network_init() not being called for these, where
it really doesn't make much sense.
Signed-off-by: Martin Storsjö <martin@martin.st>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
This issue was discovered while decoding the FATE sample vqa/ws_snd.vqa.
For some unknown reason only audio decoding is tested by FATE for that file,
but not video.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Not all applications (e.g. MPlayer) set block_align, and
when using a different demuxer it might not even be
easily available.
So fall back to selecting mode based on bit rate as before
if block_align has not useful value.
It can't be worse than failing to decode completely.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* cus/stable:
ffplay: fix invalid wanted_channel_layout calculation
ffplay: honor SDL_AUDIO_CHANNELS and make sure to use SDL supported number of audio channels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Check explicitly if enough bits are left to prevent an infinite loop
when the bitstream buffer is not followed by zero-padding.
Based on patches by Michael Niedermayer <michaelni@gmx.at>.
frame_size is the number of bytes left in the packet, so if we are passing
buf-4 we can safely read frame_size+4 bytes.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (46 commits)
mtv: Make sure audio_subsegments is not 0
v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
avconv: add symbolic names for -vsync parameters
flvdec: Fix compiler warning for uninitialized variables
rtsp: Fix compiler warning for uninitialized variable
ulti: convert to new bytestream API.
swscale: Use standard multiple inclusion guards in ppc/ header files.
Place some START_TIMER invocations in separate blocks.
v4l2: list available formats
v4l2: set the proper codec_tag
v4l2: refactor device_open
v4l2: simplify away io_method
v4l2: cosmetics
v4l2: uniform and format options
v4l2: do not force interlaced mode
avio: exit early in fill_buffer without read_packet
vc1dec: fix invalid memory access for small video dimensions
rv34: fix invalid memory access for small video dimensions
rv34: joint coefficient decoding and dequantization
avplay: Don't call avio_set_interrupt_cb(NULL)
...
Conflicts:
Changelog
avconv.c
doc/APIchanges
doc/indevs.texi
libavcodec/adxenc.c
libavcodec/dnxhdenc.c
libavcodec/h264.c
libavdevice/v4l2.c
libavformat/flvdec.c
libavformat/mtv.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Thu Nov 3 22:38:10 2011 +0100
lavf: fix null pointer dereference in rdt
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is no longer needed and causes various problems with RTSP
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes compilation failures related to START_TIMER/STOP_TIMER macros and
-Werror=declaration-after-statement. START_TIMER declares variables and thus
may not be placed after statements outside of a new block.
Fixes an invalid free() with ass in avi. The sample in bug 98 passes
parts of AVPacket.data as buffer for the AVIOContext. Since the packet
is quite large fill_buffer tries to reallocate the buffer before doing
nothing. Fixes bug 98.
For small video dimensions, these calculations of the upper bound
for pixel access may have a negative result. Using an unsigned
comparison to bound a potentially negative value only works if
the greater operand is non-negative. Fixed by doing edge emulation
when the upper bound is probably negative, everywhere that this
pattern appears.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
For small video dimensions calculations of the upper bound for pixel
access may result in negative value. Using an unsigned comparison
works only if the greater operand is non-negative. This is fixed by
doing edge emulation explicitly for such conditions.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Perform dequantization while decoding coefficients instead of performing it
on the entire coefficients buffer.
Since quantized coefficients are very sparse, this usually causes a small
speedup. Speedup of around 1% on Panda board compared to the removed here
neon code. Global speedup is probably around 3%.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Since we don't use avio_set_interrupt_cb for interrupt callbacks,
we don't need to call it to reset the interrupt cb either.
This avoids a warning about use of deprecated functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes compilation failures related to START_TIMER/STOP_TIMER macros and
-Werror=declaration-after-statement. START_TIMER declares variables and thus
may not be placed after statements outside of a new block.
* qatar/master:
fate: add dxtory test
adx_parser: rewrite.
adxdec: Validate channel count to fix a division by zero.
adxdec: Do not require extradata.
cmdutils: K&R reformatting cosmetics
alacdec: implement the 2-pass prediction type.
alacenc: implement the 2-pass prediction type.
alacenc: do not generate invalid multi-channel ALAC files
alacdec: fill in missing or guessed info about the extradata format.
utvideo: proper median prediction for interlaced videos
lavu: bump lavu minor for av_popcount64
dca: K&R formatting cosmetics
dct: K&R formatting cosmetics
lavf: flush decoders in avformat_find_stream_info().
win32: detect number of CPUs using affinity
Add av_popcount64
snow: Restore three mistakenly removed casts.
Conflicts:
cmdutils.c
doc/APIchanges
libavcodec/adx_parser.c
libavcodec/adxdec.c
libavcodec/alacenc.c
libavutil/avutil.h
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
Fix audio output
Suppress dumping of residues buffer
Use quantizer value read from bitstream
Cosmetics: Remove two empty lines and realign some code
Reset acfilter_prevvalues buffer in clear_codec_buffers()
Fix AC filter buffers and AC filter reversion
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code ended in multiple different infinite
loops. See stl_ten_1_big.sfd as example with and without zzuf
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
The original code wasn't taking into account the fact that linesize may not equal the frame's width. This is to correct that.
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegenc: use avctx->slices as number of slices
v410enc: fix undefined signed left shift caused by integer promotion
Release notes: mention cleaned up header includes
fix Changelog file
Fix a bunch of typos.
Drop some pointless void* return value casts from av_malloc() invocations.
wavpack: fix typos in previous cosmetic clean-up commit
wavpack: cosmetics: K&R pretty-printing
avconv: remove the 'codec framerate is different from stream' warning
wavpack: determine sample_fmt before requesting a buffer
bmv audio: implement new audio decoding API
mpegaudiodec: skip all channels when skipping granules
mpegenc: simplify muxrate calculation
Conflicts:
Changelog
avconv.c
doc/RELEASE_NOTES
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpegaudiodec.c
libavcodec/mpegvideo.c
libavformat/mpegenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Up until now, the decoder didn't output anything
in the data buffer. This fixes all the issues
related to sample format, removes leftover code
and actually outputs some audio to data buffer.
With this, the only sample we have can be played.
Seeking is still broken though.
Adds a new member to MpegEncContext to hold the number of used slice
contexts. Fixes segfaults with '-threads 17 -thread_type slice' and
fate-vsynth{1,2}-mpeg{2,4}thread{,_ilace} with --disable-pthreads.
* qatar/master:
avconv: make -frames work for all types of streams, not just video.
bfi: K&R cosmetics
bgmc: K&R cleanup
rawdec: Set start_time to 0 for raw audio files.
Detect 'yuv2' as rawvideo also in avi.
rawdec: propagate pict_type information to the output frame
rawdec: Support more QT 1bpp rawvideo files.
avconv: free bitstream filters
threads: limit the number of automatic threads to MAX_AUTO_THREADS
avplay: K&R cleanup
fate: use rgb24 as output format for dfa tests
threads: set thread_count to 1 when thread support is disabled
threads: introduce CODEC_CAP_AUTO_THREADS and add it to libx264
Conflicts:
ffplay.c
libavcodec/avcodec.h
libavcodec/pthread.c
libavcodec/version.h
tests/ref/fate/dfa1
tests/ref/fate/dfa10
tests/ref/fate/dfa11
tests/ref/fate/dfa2
tests/ref/fate/dfa3
tests/ref/fate/dfa4
tests/ref/fate/dfa5
tests/ref/fate/dfa6
tests/ref/fate/dfa7
tests/ref/fate/dfa8
tests/ref/fate/dfa9
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The fate-h264-bsf-mp4toannexb failures were caused by an integer
overflow of the unneeded multiplication.
Inspired by patch by: Michael Niedermayer <michaelni@gmx.at>
The extra thread added in {frame_}*thread_init was not taken into
account. Explicitly sets thread_count to 1 if only one CPU core was
detected. Also fixes two typos in comments.
Palette is as supposed in native endianness. Converting the pal8 output
to rgb24 is thus necessary for identical CRCs on big and little endian
systems.
Some external codecs have their own code to determine the best number
of threads. This number is not necessary the number of cpu cores.
Thread_count will be only 0 if the codec has CODEC_CAP_AUTO_THREADS.
* qatar/master:
FATE: add tests for dfa
mpegaudiodec: fix seeking.
mpegaudiodec: fix compilation when testing the unchecked bitstream reader
threads: add sysconf based number of CPUs detection
threads: always include necessary headers for number of CPUs detection
threads: default to automatic thread count detection
Changelog: restore version <next> header
cook: K&R formatting cosmetics
Conflicts:
Changelog
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The safe bitstream reader does not allow using skip_bits_long() to seek to a
point before the start of the buffer, which was needed by the mp3 decoder.
This change instead calculates the start point of the first valid granule and
skips to that position.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Since the conditions for the actual usage are more specific a less
preferred method can be used. This would cause compilation errors
because necessary headers are not included.
This works around issues arising from inputs that claim to have a
filesize of 0.
Reported-by: buzz_
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Port MPlayer tinterlace filter from MPlayer, with some ideas taken
from the FFmbc/libavfilter port, with the following main differences:
* added support for full-scale YUVJ formats
* added support for YUVA420P
* request_frame() on the filter is forced to return a frame
* some code factorization (related to the copy_picture_fields() function)
* fixed black padding values for mode 3
Rationale: avfilter_copy_frame_props() was already defined in
libavfilter/avcodec.h, and keeping the lavc/lavfi API glue localized in a
specific file should ease maintainance and help the ones which use
libavfilter without depending on libavcodec.
This has been fixed differently and its revert avoids error messages
when decoding xx.flv. This also reduces the difference to qatar.
This reverts commit 5a2b3f3a52.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: whitespace cosmetics
fate: split off video codec FATE tests into their own file
fate: split off audio codec FATE tests into their own file
fate: split off Electronic Arts codec FATE tests into their own file
fate: split off QuickTime codec FATE tests into their own file
fate: split off voice codec FATE tests into their own file
fate: split off demuxer FATE tests into their own file
cosmetics: Drop unnecessary parentheses around return values.
fate: drop pointless _audio and _video suffixes from xan tests
qt-faststart: K&R reformatting; fix comment typos
FATE: Add test for H.264 MP4->annex.B bitstream filter.
Conflicts:
ffplay.c
tests/fate.mak
tests/fate/h264.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/qtrle.mak
tests/fate/real.mak
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This only happens for a "back" value of 0 which is invalid anyway,
but lcldec does not properly validate input.
Also extend the documentation to specify valid values.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
fate: split off DPCM codec FATE tests into their own file
fate: split off PCM codec FATE tests into their own file
libvorbis: K&R reformatting cosmetics
libmp3lame: K&R formatting cosmetics
fate: Add a video test for xxan decoder
mpegvideo_enc: K&R cosmetics (line 1000-2000).
avconv: K&R cosmetics
qt-faststart: Fix up indentation
indeo4: remove two unused variables
doxygen: cleanup style to support older doxy
fate: add more tests for VC-1 decoder
applehttpproto: Apply the same reload interval changes as for the demuxer
applehttp: Use half the target duration as interval if the playlist didn't update
applehttp: Use the last segment duration as reload interval
lagarith: add decode support for arith rgb24 mode
Conflicts:
avconv.c
libavcodec/libmp3lame.c
libavcodec/mpegvideo_enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 9d6b2077b2.
This error is annoying while debugging, and if someone disables it for
convenience, it raises the odds of leaving ffmpeg unbuildable by default.
According to draft-pantos-http-live-streaming-07, 6.3.4,
the duration of the last media segment in the playlist
should be used as initial minimum reload delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current default PES packet size, and very small audio bitrates,
audio packet duration gets too long. For players, which wait for a whole
audio packet (or more) it takes a very long time to start playing sound.
For 24kbps audio, one PES packet is about 1 second long. On Motorola STBs,
we observe about 3 second delay before the playback starts with the
default setting.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Do not assume the audio packets being always smaller than
DEFAULT_PES_PAYLOAD_SIZE.
Signed-off-by: Jindřich Makovička <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
mpegvideo_enc: K&R cosmetics
doxygen: remove unreplaced variables from custom header and footer
threads: test for sys/param.h and include it for sysctl on OpenBSD
v4l2: remove unneded linux specific asm/types.h include
x86: Fix constraints for decode_significance*_x86
Conflicts:
libavcodec/mpegvideo_enc.c
libavdevice/v4l2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing code expected a palette buffer holding 256 uint32_t's allocated in the data[1] field of the AVFrame structure, but data[1] was NULL. The bug is fixed by using a fixed local array (palette256) to hold the palette instead.
This solves http://ffmpeg.org/trac/ffmpeg/ticket/833
Signed-off-by: Frank Vernaillen <fr_ve@hotmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_samples_alloc() behavior changed in bbb46f3ec, resulting in random
data filling the data[] and linesize[] arrays of the returned
AVFilterBufferRef, which was resulting in wrong behavior in case of code
checking on data[i] nullity.
In particular fixes crash in avfilter_filter_samples():
for (i = 0; samplesref->data[i]; i++)
memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]);
and correctly fills the linesize[] array for planar data.
Originally, prior to 8742a4ff8, the caller code was compiled
within this condition:
ARCH_X86 && HAVE_7REGS && HAVE_EBX_AVAILABLE && !defined(BROKEN_RELOCATIONS)
Since HAVE_7REGS is defined as
(ARCH_X86_64 || (HAVE_EBX_AVAILABLE && HAVE_EBP_AVAILABLE))
the subcondition HAVE_7REGS && HAVE_EBX_AVAILABLE is equal
to HAVE_7REGS (for 32 bit at least). The correct simplification
of the original condition thus is HAVE_7REGS, not
HAVE_EBX_AVAILABLE.
This fixes compilation in some cases where HAVE_EBP_AVAILABLE = 0
and HAVE_EBX_AVAILABLE = 1.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
fate: split off vqf/twinvq FATE tests into their own file
fate: split off mpc FATE tests into their own file
fate: split off libavcodec FATE tests into their own file
fate: split off Microsoft codec FATE tests into their own file
fate: group all VP* codec FATE tests together in one file
swscale: prevent invalid writes in packed_16bpc_bswap
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ID3v2.4 allows for zlib compressed tags, but libavformat skips them.
Implement code to inflate compressed tags.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avs: call release_buffer() at the end.
Add minor bumps and APIchanges entries for lavc/lavfi changes.
mpegvideo.c: K&R formatting and cosmetics.
avconv: avoid memcpy in vsrc_buffer when possible.
avconv: implement get_buffer()/release_buffer().
lavfi: add a new function av_buffersrc_buffer().
lavfi: add avfilter_copy_frame_props()
lavc: add format field to AVFrame
lavc: add width and height fields to AVFrame
lavc: add a sample_aspect_ratio field to AVFrame
doxy: add website-alike style to the html output
FAQ: add an entry for common error when using -profile
Conflicts:
avconv.c
cmdutils.c
doc/APIchanges
libavcodec/avcodec.h
libavcodec/mpegvideo.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/Makefile
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/src_movie.c
libavfilter/vsrc_buffer.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The format is a per-frame property, having it in AVFrame simplify the
operation of extraction of that information, since avoids the need to
access the codec/stream context.
width and height are per-frame properties, setting these values in
AVFrame simplify the operation of extraction of that information,
since avoids the need to check the codec/stream context.
The sample aspect ratio is a per-frame property, so it makes sense to
define it in AVFrame rather than in the codec/stream context.
Simplify application-level sample aspect ratio information extraction,
and allow further simplifications.
When generating the .dep files for .texi sources, verbatim includes
(@verbatiminclude) should also be taken into account.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The FATE documentation depends on the mentioned file. But that
did break out of tree builds because the file was not found.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
Perform dequantization of channel coefficients
Perform inverse inter-channel decorrelation and ac-filter
Implement revert_inter_ch_decorr() and revert_acfilter()
Enable inverse-MCLMS filter
Fix inverse-MCLMS filtering routines
Do not update buffers in case no speed change is necessary
Use int for channel_coeffs instead of int16_t
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vp3dec: Check coefficient index in vp3_dequant()
svq1dec: call avcodec_set_dimensions() after dimensions changed.
Prepare for 0.8_beta1 snapshot release
threads: check defines before using them in automatic thread detection
pthread: include sys/types.h before sys/sysctl.h
4xm: remove unused variables.
h264: Fix a possible overread in decode_nal_units()
allfilters: fix type of avfilter_vsrc_buffer.
w32thread: call ResetEvent() in pthread_cond_broadcast().
Conflicts:
Changelog
RELEASE
doc/RELEASE_NOTES
libavcodec/pthread.c
libavcodec/vp3.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Based on a patch by Michael Niedermayer <michaelni@gmx.at>
Fixes NGS00145, CVE-2011-4352
Found-by: Phillip Langlois
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master:
build: fix standalone compilation of OMA muxer
build: fix standalone compilation of Microsoft XMV demuxer
build: fix standalone compilation of Core Audio Format demuxer
kvmc: fix invalid reads
4xm: Add a check in decode_i_frame to prevent buffer overreads
adpcm: fix IMA SMJPEG decoding
options: set minimum for "threads" to zero
bsd: use number of logical CPUs as automatic thread count
windows: use number of CPUs as automatic thread count
linux: use number of CPUs as automatic thread count
pthreads: reset active_thread_type when slice thread_init returrns early
v410dec: include correct headers
Drop ALT_ prefix from BITSTREAM_READER_LE name.
lavfi: always build vsrc_buffer.
ra144enc: zero the reflection coeffs if the filter is unstable
sws: readd PAL8 to isPacked()
mov: Don't stick the QuickTime field ordering atom in extradata.
truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
Conflicts:
configure
libavcodec/4xm.c
libavcodec/avcodec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavformat/Makefile
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When no data was available both the buffer thread as well as
the main thread would block in select(), when data becomes
available both should move forward and as data is read in the
buffer thread the main thread would block in select() later
the read data was put in the fifo but the main thread still
would be blocked in select() until either the timeout or
another packet would come in.
This is solved in this commit by using a mutex and a condition
variable
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Prior to this a X bytes write could be seen as less than X bytes being
available if the check was done at an unfortunate moment.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use sched_getaffinity to determine the number of logical CPUs.
Limits the number of threads to 16 since slice threading of H.264
seems to be buggy with more than 16 threads.
This file does not use anything from get_bits.h but needs
intreadwrite.h.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* tjoppen/fuzz_fixes:
mxfdec: Don't crash in mxf_packet_timestamps() if current_edit_unit overflows
mxfdec: Zero nb_ptses in mxf_compute_ptses_fake_index()
mxfdec: Sanity check PreviousPartition
mxfdec: Never seek back in local sets and KLVs
mxfdec: Move the current_partition check inside mxf_read_header()
mxfdec: Fix infinite loop in mxf_packet_timestamps()
mxfdec: Check url_feof() in mxf_read_local_tags()
mxfdec: Check for NULL component
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A list of formats may have been dynamically created by the calling code,
and thus should not be referenced by the sink buffer context.
Avoid possible invalid data reference.
The 'fiel' atoms can be found in H.264 tracks clobbering the extradata.
MJPEG supports non field based extradata, and this data should be
preserved when copying.
This way ffmpeg can be distinguished from the fork by a user
application or a encoded file by a decoder.
The highest value micro had, in the past, that i could find, was 6
thus 100 should be safe.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
asfdec: add side data to ASFStream packet instead of output packet.
idroqdec: set AVFMTCTX_NOHEADER and create streams as they occur.
nellymoserdec: Indicate that the decoder can handle changed parameters
libavcodec: Apply parameter change side data when decoding audio
flvdec: Add param change side data if the sample rate or channels have changed
libavformat: Add a utility function for adding parameter change side data
libavcodec: Define a side data type for parameter changes
aacdec: Handle new extradata passed as side data
flvdec: Export new AAC/H.264 extradata as side data on the next packet
libavcodec: Define a side data type for new extradata
flacdec: skip all track indices at once instead of looping.
mxf: Add PictureEssenceCoding UL for V210.
mxfdec: consider QuantizationBits between 17 and 24 to be pcm_s24*
mxfenc: Add support for MPEG-2 MP@HL-14 in mxf container.
mxf: H.264/MPEG-4 AVC Intra support
configure: Show whether the safe bitstream reader is enabled
x86: Tighten register constraints for decode_significance*_x86.
Replace Subversion revisions in comments by Git hashes.
h264_cabac: synchronize decode_significance_*_x86 conditionals
w32threads: wait for the waked thread in pthread_cond_signal.
...
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/flvdec.c
libavformat/utils.c
tests/ref/lavfi/pixdesc
tests/ref/lavfi/pixfmts_copy
tests/ref/lavfi/pixfmts_null
tests/ref/lavfi/pixfmts_scale
tests/ref/lavfi/pixfmts_vflip
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes demuxing of file where the first packet is not audio. Such files
are generated by our idroq muxer. It also fixes demuxing of audio only
idroq files.
Also define a codec capability for codecs that can handle
parameters changed externally between decoded packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Compared to just overwriting the old extradata, this has the
advantage of letting the decoder know exactly when the
extradata changed (otherwise it is changed immediately when the
new extradata packet is demuxed, even if there's old queued packets
awaiting to be decoded). This makes it easier for decoders to
actually react to the change, so they won't have to inspect
the extradata for each packet to see if it might have changed.
This works when sequentially playing a file with sample rate
changes, but if seeking past a new extradata packet in the
file, it obviously doesn't work properly. That case doesn't
work in flash player either, so it's probably ok not to handle
it.
Signed-off-by: Martin Storsjö <martin@martin.st>
Support Main Profile at High 1440 Level in MXF container,
using essence coding label from SMPTE RDD 9, table 6.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* tjoppen/mxf_fixes_20111220:
mxfdec: Sanity-check SampleRate
mxfdec: Make sure mxf->nb_index_tables > 0 in mxf_packet_timestamps()
mxfdec: Remove unused variables
mxfdec: Make sure x < index_table->nb_ptses
mxfdec: Ignore the last entry in Avid's index table segments
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specially crafted files can lead the parsing code to take too long.
We fix a lot of these problems by not allowing local tags to extend past the
end of the set and not allowing other KLVs to be read past the end of
themselves.
On 32-bit OS X with gcc 4.0/4.2 and shared libraries enabled, the ebx register
is not available, but required to assemble the functions.
This reverts commit 8742a4f to a simplified version of the original constraints.
This fixes a deadlock VLC triggered with multithreaded decoding. The
wait forces one of the current waiters to wake and not the thread
which calls pthread_cond_signal() itself.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
lavc: always align height by 32 pixel
raw: add 10bit YUV definitions
nut: support 10bit YUV
mpegvideo_enc: separate declarations and statements
oma: make header compile standalone
vp3: Reorder some functions to fix VP3 build with Theora disabled.
build: fix standalone compilation of ADX encoder
build: fix standalone compilation of ADPCM decoders
build: fix standalone compilation of mpc7/mpc8 decoders
4xm: Use bytestream2 functions to prevent overreads
bytestream: add a new set of bytestream functions with overread checking
mpegts: Suppress invalid timebase warnings on DMB streams.
mpegts: Fix typo in handling sections in the PMT.
vc1dec: Use the right pointer type for the tmp pointer
Conflicts:
libavcodec/4xm.c
libavcodec/utils.c
libavcodec/vc1dec.c
libavcodec/vp3.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some libavifilter tests use NUT as output even if the produced
files were not decodable. The support for 10bit introduced in
432f0e5b7d and 91b1e6f0c changed the hashes.
This moves declarations without initialisers or with constant
initialisers to the start of a block, and adds do {} while(0)
around some macros, thus allowing declarations within them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Only the OPAtom demuxing logic is guaranteed to have index tables, meaning OP1a
files that lack an index would cause SIGSEGV.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Interlaced content for most codec requires it.
This patch is a stop-gap pending a serious rework to support
codecs with non 16 pixel macroblocks.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Also create a plain text (.txt) file from fate.texi if the makeinfo
program is available.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The fate.txt file is ported to texinfo format. Therefore the
fate.txt is renamed to fate.texi. The contents of the already
existing fate.texi file are discarded.
However there should be no loss of information. If you find
anything missing, please report.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When generating the .dep files for .texi sources, verbatim includes
(@verbatiminclude) should also be taken into account.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The last entry is the total size of the essence container.
Previously a TemporalOffset error would be logged, even though segments like
these are expected.
* qatar/master:
h264: clear trailing bits in partially parsed NAL units
vc1: Handle WVC1 interlaced stream
xl: Fix overreads
mpegts: rename payload_index to payload_size
segment: introduce segmented chain muxer
lavu: add AVERROR_BUG error value
avplay: clear pkt_temp when pkt is freed.
qcelpdec: K&R formatting cosmetics
qcelpdec: cosmetics: drop some pointless parentheses
x86: conditionally compile dnxhd encoder optimizations
Revert "h264: skip start code search if the size of the nal unit is known"
swscale: fix formatting and indentation of unscaled conversion routines.
h264: skip start code search if the size of the nal unit is known
cljr: fix buf_size sanity check
cljr: Check if width and height are positive integers
Conflicts:
libavcodec/cljr.c
libavcodec/vc1dec.c
libavformat/Makefile
libavformat/mpegtsenc.c
libavformat/segment.c
libswscale/swscale_unscaled.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Trailing bits are likely to be non-zero if the NAL unit is truncated.
Clearing the bits make overreads of the bitstream less likely in this
case. Fixes playback of
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4 which
has a forbidden byte sequence of 0x00 0x00 0x00 in it SPS.
* qatar/master:
APIchanges: fill in revision for AVFrame.age deprecation
avcodec: deprecate AVFrame.age
4xm: remove unneeded check for remaining unused data.
lavf: force threads to 1 in avformat_find_stream_info()
swscale: fix overflows in vertical scaling at top/bottom edges.
lavf: add OpenMG audio muxer.
omadec: split data that will be used in the muxer to a separate file.
lavf: rename oma.c -> omadec.c
tmv decoder: set correct pix_fmt
Conflicts:
Changelog
doc/APIchanges
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/oma.c
libavformat/version.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Start code emulation prevention is only required in Annex B bytestream
packed NAL units. For other coding formats the size is already known.
Looking for a start code prefix can result in false positives like in
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4
which has a false positive in the SPS.
Width and height might get passed as 0 and would cause floating point
exceptions in decode_frame.
Fixes bugzilla #149
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This was intended as an optimisation for skipped blocks in MPEG2
P-frames and never used elsewhere. Removing this "optimisation"
speeds up MPEG2 decoding by 1-2% (ARM Cortex-A9).
Signed-off-by: Mans Rullgard <mans@mansr.com>
* tjoppen/proper_mxf_track_linking:
mxfdec: Don't parse slices or DeltaEntryArrays
mxfdec: Remove dead/useless code
mxfdec: Hybrid demuxing/seeking solution
mxfdec: Add mxf_edit_unit_absolute_offset()
mxfdec: Replace zero IndexDurations with st->duration
mxfdec: Add "fake" index to MXFIndexTable to assist seeking
mxfdec: Add MXFIndexTables
mxfdec: Move mxf_read_packet*() near the bottom of the file
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code ended in multiple different infinite
loops. See stl_ten_1_big.sfd as example with and without zzuf
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes integer multiplication overflows in RGB48 output
(vertical) scaling as detected by IOC. What happens is that for
certain types of filters (lanczos, spline, bicubic), the
intermediate sum of coefficients in the middle of a filter can
be larger than the fixed-point equivalent of 1.0, even if the
final sum is 1.0. This is fine and we support that.
However, at frame edges, initFilter() will merge the coefficients
for the off-screen pixels into the top or bottom pixel, such as
to emulate edge extension. This means that suddenly, a single
coefficient can be larger than the fixed-point equivalent of
1.0, which the vertical scaling routines do not support.
Therefore, remove the merging of coefficients for edges for
the vertical scaling filter, and instead add edge detection
to the scaler itself so that it copies the pointers (not data)
for the edges (i.e. it uses line[0] for line[-1] as well), so
that a single coefficient is never larger than the fixed-point
equivalent of 1.0.
Previously the decoder only worked if the user had set avctx->pix_fmt
manually. For some reason the libavformat tmv demuxer sets this, so
the problem was not visible in avplay etc.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This should be replaced by a more appropriate error code of course but
we should not leave compilation broken until that is decided.
Found-by: jb
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
that way qatar maintains the code for me and i dont need to resolve conflicts.
If someone wants the a32 reader back, only thing you need to do is maintain
it, i would be happy to have it back, iam just not volunteering to maintain
it due to lack of time.
Based on: a1e98f198e by Mans Rullgard.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Many of the test programs directly access internal symbols not
exported from the shared libraries. This allows tests to run
when configured with shared libraries.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes the same overflow as in the RGB48/16-bit YUV scaling;
some filters can overflow both negatively and positively (e.g.
spline/lanczos), so we bias a signed integer so it's "half signed"
and "half unsigned", and can cover overflows in both directions
while maintaining full 31-bit depth.
Signed-off-by: Mans Rullgard <mans@mansr.com>
No difference in PSNR or bitrate in the printed precission with the matrix lobby scene at 322x242
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
We're shifting individual components (8-bit, unsigned) left by 24,
so making them unsigned should give the same results without the
overflow.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
For certain types of filters where the intermediate sum of coefficients
can go above the fixed-point equivalent of 1.0 in the middle of a filter,
the sum of a 31-bit calculation can overflow in both directions and can
thus not be represented in a 32-bit signed or unsigned integer. To work
around this, we subtract 0x40000000 from a signed integer base, so that
we're halfway signed/unsigned, which makes it fit even if it overflows.
After the filter finishes, we add the scaled bias back after a shift.
We use the same trick for 16-bit bpc YUV output routines.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The buffer splicing relies on the bitstream reader over-reading
the end of the buffer as declared in init_get_bits(), although
more data is actually present. Manually moving the bitstream
boundary after init_get_bits() allows this to work as expected.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The sample has an incomplete last frame. Decoding it is pointless.
The garbage produced was changed by the bitstream reader now
protecting against over-reads.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When turned on, H264/CAVLC gets ~15% (CVPCMNL1_SVA_C.264) slower for
ultra-high-bitrate files, or ~2.5% (CVFI1_SVA_C.264) for lower-bitrate
files. Other codecs are affected to a lesser extent because they are
less optimized; e.g., VC-1 slows down by less than 1% (all on x86).
The patch generated 3 extra instructions (cmp, cmovae and mov) per
call to get_bits().
The performance penalty on ARM is within the error margin for most
files, up to 4% in extreme cases such as CVPCMNL1_SVA_C.264.
Based on work (for GCI) by Aneesh Dogra <lionaneesh@gmail.com>, and
inspired by patch in Chromium by Chris Evans <cevans@chromium.org>.
* qatar/master:
get_bits: remove A32 variant
avconv: support stream specifiers in -metadata and -map_metadata
wavpack: Fix 32-bit clipping
wavpack: Clip samples after shifting
h264: don't drop B-frames after next keyframe on POC reset.
get_bits: remove useless pointer casts
configure: refactor lists of tests and components into variables
rv40: NEON optimised weak loop filter
mpegts: replace some magic numbers with the existing define
swscale: add unscaled packed 16 bit per component endianess conversion
Conflicts:
libavcodec/get_bits.h
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The A32 bitstream reader variant is only used on ARMv5 and for
Prores due to the larger bit cache this decoder requires.
In benchmarks on ARMv5 (Marvell Sheeva) with gcc 4.6, the only
statistically significant difference between ALT and A32 is
a 4% advantage for ALT in FLAC decoding. There is thus no (longer)
any reason to keep the A32 reader from this point of view.
This patch adds an option to the ALT reader increasing the bit
cache to 32 bits as required by the Prores decoder. Benchmarking
shows no significant change in speed on Intel i7. Again, the
A32 reader fails to justify its existence.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the case that (frame_flags & 0x03) == 3, hybrid_maxclip
may have had a signed integer overflow.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It doesn't make much sense to clip pre-shift,
nor is it correct for proper decoding.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The keyframe after a POC reset may not be the first to be returned to
the user. Therefore, don't reset the expected next POC once we return
a keyframe to the user, but once we know that the next frame in the
return-queue is a keyframe.
width and height might get passed as 0 and would cause floating point
exceptions in decode_frame.
Fixes bugzilla #149
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we don't
use delta entries or slices, only StreamOffsets.
OPAtom seeking basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
This fixes ticket #746.
This changes mxf_compute_ptses() to be used for MXFIndexTable, and also adds
code for computing the fake index to it.
This also temporarily disables PTS computation. A future patch will restore it.
XBMC's configure script checks for this function in installed
libavcodec.so to determine VDPAU support.
Fixes ticket #762 reported by Christian Marillat
* qatar/master:
movenc: Rudimentary IODs support.
v410enc: fix output buffer size check
v410enc: include correct headers
fate: add -pix_fmt rgb48le to r210 test
flvenc: Support muxing 16 kHz nellymoser
configure: refactor list of programs into a variable
fate: add r210 decoder test
fate: split off Indeo FATE tests into their own file
fate: split off ATRAC FATE tests into their own file
fate: Add FATE tests for v410 encoder and decoder
ARM: fix external symbol refs in rv40 asm
westwood: Make sure audio header info is present when parsing audio packets
libgsm: Reset the MS mode of GSM in the flush function
libgsm: Set options on the right object
ARM: dca: disable optimised decode_blockcodes() for old gcc
Conflicts:
configure
libavformat/movenc.c
libavformat/movenc.h
tests/fate2.mak
tests/ref/acodec/alac
tests/ref/vsynth1/mpeg4
tests/ref/vsynth2/mpeg4
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Audio header information might get scrambled and would not parse,
yet wsqva_read_packet would try to parse audio packets causing
segfaults such as floating point exception.
Fixes bugzilla #141.
Signed-off-by: Martin Storsjö <martin@martin.st>
The mode is set in libgsm_decode_init, but the decoder
object is simply destroyed and recreated in the flush
function - therefore the mode has to be set again.
This fixes playback using the libgsm_ms decoder in avplay.
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts commit 4f820131fa.
It is better to abort() than to have remotly exploitable arbitrary code
execution bugs. Even more so that this abort has never been triggered
by any input people threw at it.
If after more extensive testing its removial is found safe we can remove
the abort() later.
* qatar/master: (23 commits)
applehttp: Properly clean up if unable to probe a segment
applehttp: Avoid reading uninitialized memory
fate: Replace misleading "aac" in the name of an ADTS test with "adts".
fate: Drop pointless "-an" from pictor test command.
fate: split off image codec FATE tests into their own file
fate: split off WMA codec FATE tests into their own file
fate: split off lossless video and audio FATE tests into their own files
fate: split off qtrle codec FATE tests into their own file
fate: split off Ut Video codec FATE tests into their own file
fate: split off screen codec FATE tests into their own file
fate: split off Real Inc. codec FATE tests into their own file
fate: split off AC-3 codec FATE tests into their own file
mpegvideo: remove abort() in ff_find_unused_picture()
rv40: NEON optimised loop filter strength selection
rv40: rearrange loop filter functions
configure: cosmetics: sort some lists where appropriate
swscale_mmx: drop no longer required parameters from VSCALEX macros
swscale: Mark yuv2planeX_8_mmx as MMX2; it contains MMX2 instructions.
build: conditionally compile x86 H.264 chroma optimizations
v410 encoder and decoder
...
Conflicts:
Changelog
configure
doc/developer.texi
doc/general.texi
libavcodec/arm/asm.S
libavcodec/avcodec.h
libavcodec/v410dec.c
libavcodec/v410enc.c
libavcodec/version.h
libavcodec/x86/Makefile
libavcodec/x86/dsputil_mmx.c
libswscale/x86/swscale_mmx.c
tests/Makefile
tests/fate2.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
ffplay: clear pkt_temp when pkt is freed.
ffplay: Fix got_frame type.
ffplay: add 10 minute seek support to ffplay
ffplay: force setting video mode on fullscreen toggle
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids a segfault if the probe function wasn't able to
determine the format.
The bug was found by Panagiotis H.M. Issaris.
Signed-off-by: Martin Storsjö <martin@martin.st>
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
A user who wishes to use default error concealment cannot set the
AV_EF_CRCCHECK flag because not every CRC in every format is a
reliable indicator of bitstream damage. In some formats crcrs
can be nonsensical in absence of any damage. We thus add the
AV_EF_CAREFUL flag in addition to the AV_EF_CRCCHECK flag to
allow a user to enable this reliable CRC check without having to
enable all CRC checks in all formats.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
ulti: Fix invalid reads
lavf: dealloc private options in av_write_trailer
yadif: support 10bit YUV
vc1: mark with ER_MB_ERROR bits overconsumption
lavc: introduce ER_MB_END and ER_MB_ERROR
error_resilience: use the ER_ namespace
build: move inclusion of subdir.mak to main subdir loop
rv34: NEON optimised 4x4 dequant
rv34: move 4x4 dequant to RV34DSPContext
aacdec: Use intfloat.h rather than local punning union.
Conflicts:
libavcodec/h264.c
libavcodec/vc1dec.c
libavfilter/vf_yadif.c
libavformat/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch is a generalization of what Michael Niedermayer
fixed in a single case.
The wmv8-drm fate test had been updated accordingly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
As far as I could see the only change is increased pos values,
which is as expected with additional metadata in the files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
It sets the supplied AVFormatContext pointer to NULL after freeing it,
which is safer and its name is consistent with other lavf functions.
Also deprecate av_close_input_file().
These indexes duplicate every entry and have the total size of the essence
container as the last entry.
This patch also computes the size of the packets when unknown.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
I thought it had to do with file offsets, but's actually the offset inside
the essence container.
In other words, unbreak multiple EditUnitByteCounts.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The change in 599b4c6ef didn't turn out to work properly on
i386 on OS X, where it broke building with PIC enabled.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit f1dba9e498)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The change in 599b4c6ef didn't turn out to work properly on
i386 on OS X, where it broke building with PIC enabled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Firstly, this test never worked as intended, always reporting
success. Secondly, bswap is available from 486 onward and can
thus be assumed present.
Signed-off-by: Mans Rullgard <mans@mansr.com>
With these changes, gcc 4.5 and later recognise it as a bswap
and use the proper instructions on ARM and x86. On x86, the
16-bit bswap is recognised from gcc 4.1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the function prototype match the argument of
AVCodecCntext.execute() and remove the cast hiding
this mismatch.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This replaces the explicit offset(reg) memory references with
"m" operands for the same locations. As a result, one fewer
register operand is needed for these inline asm statements.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When the buf and last pointers are equal, the FFSWAP() results
in an invalid call to memcpy() with same source and destination
on some targets. Although assigning a struct to itself is valid
C99, gcc does not check for this before calling memcpy().
See http://gcc.gnu.org/bugzilla/show_bug.cgi?id=32667
Signed-off-by: Mans Rullgard <mans@mansr.com>
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If the sdp is generated before the rtp muxer is initialized
(e.g. as when called from the rtsp muxer), this has to be done,
otherwise the rtp muxer doesn't know that the input really is
in mp4 format.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an annex b bitstream is muxed into mov, the actual written
sample is reformatted to mp4 syntax before writing.
Currently, the RTP hints that copy data from the normal video
track, where the payload data might be offset compared to the
original sample that the RTP hinting used (when 3 byte
annex b startcodes have been converted into 4 byte mp4 format
startcodes).
Signed-off-by: Martin Storsjö <martin@martin.st>
av_log(NULL,AV_LOG_ERROR,"CABAC failure at %d\n",i);
STOP_TIMER("get_cabac")
}
#if 0
for(i=0;i<SIZE;i++){
START_TIMER
if(r[i]!=get_cabac_u(&c,state,(i&1)?6:7,3,i&1))
av_log(NULL,AV_LOG_ERROR,"CABAC unary (truncated) binarization failure at %d\n",i);
STOP_TIMER("get_cabac_u")
}
for(i=0;i<SIZE;i++){
START_TIMER
if(r[i]!=get_cabac_ueg(&c,state,3,0,1,2))
av_log(NULL,AV_LOG_ERROR,"CABAC unary (truncated) binarization failure at %d\n",i);
STOP_TIMER("get_cabac_ueg")
}
#endif
if(!get_cabac_terminate(&c))
av_log(NULL,AV_LOG_ERROR,"where's the Terminator?\n");
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