There is only one caller, which does not need the shifting. Other use cases
are situations where different roundings would be needed.
The x86 and neon versions are modified accordingly.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The length is even, so some unrolling can be performed. Timings are for x86:
- 32bits: 102c -> 82c
- 64bits: 82c -> 69c
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This was an incorrect copy-and-paste to a code not needing the original code.
Spotted by Jason in a previous review but forgotten in the commit.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
movq from SSE register _to_ memory is an SSE2 instruction.
Use the SSE movlps function instead that does the same thing.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes an issue in the code to check the size that will
be written to match the actual code writing. In the long
term it would make sense to change this so the counting and
writing code are the same so they dont need to be kept in sync.
It also increases the array size, which was too small either way
and adds a redudnant saftey check.
This issue does not affect any FFmpeg release as it has been
introduced Jan 31 which is narrowly after our last release.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Results of IDCT can by far outreach the range of ff_cropTbl[], leading
to overreads and potentially crashes.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This fixes some arith decoder overreads and a potential infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/h264_ps.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All colorspaces are supported.
Renamed libutvideo.cpp to libutvideodec.cpp.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is so the forthcoming encoder wrapper can share
them.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This only returns bits per sample when it is exactly correct. That is, the
codec contains only raw samples with no frame headers or padding. This applies
to basically all PCM codecs and a small subset of ADPCM codecs.
This condition cannot happen, if it can it is a bug that MUST be fixed.
And i very happily volunteer to fix it if someone reports a case to
me that fails.
This reverts commit 5d652e063b.
This splits ff_dsputil_init_mmx() into multiple functions, one for
each MMX/SSE level, somewhat simplifying the nested conditions.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This fixes some global out of array reads and wrong cliping.
No speed difference meassurable under clang on i5
also all important code paths on all important platforms should
use SIMD.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a out of global array read in the cplscale* tables.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
cmdutils: use new avcodec_is_decoder/encoder() functions.
lavc: make codec_is_decoder/encoder() public.
lavc: deprecate AVCodecContext.sub_id.
libcdio: add a forgotten AVClass to the private context.
swscale: remove "cpu flags" from -sws_flags description.
proresenc: give user a possibility to alter some encoding parameters
vorbisenc: add output buffer overwrite protection
libopencore-amrnbenc: fix end-of-stream handling
ra144enc: fix end-of-stream handling
nellymoserenc: zero any leftover packet bytes
nellymoserenc: use proper MDCT overlap delay
qpeg: Use bytestream2 functions to prevent buffer overreads.
swscale: make %rep unconditional.
vp8: convert simple loopfilter x86 assembly to use named arguments.
vp8: convert idct x86 assembly to use named arguments.
vp8: convert mc x86 assembly to use named arguments.
vp8: convert loopfilter x86 assembly to use cpuflags().
vp8: convert idct/mc x86 assembly to use cpuflags().
swscale: remove now unnecessary hack.
x86inc: don't "bake" stack_offset in named arguments.
...
Conflicts:
cmdutils.c
doc/APIchanges
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/qpeg.c
libavcodec/utils.c
libavcodec/version.h
libavdevice/libcdio.c
tests/lavf-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In most places where it's used, it's as a pointless write-only field.
Only rv10 decoder actually reads from it, but it stores some internal
version info in it. There is no reason for it to be in a public field.
Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.
Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
CC:libav-stable@libav.org
Currently we have an assert() that prevents the frame from being too large,
but it is more user-friendly to give an error message instead of aborting on
assert(). This condition is quite unlikely due to the minimum bit rate check
in encode_init(), but it is still worth having.
The maximum theoretical frame size is around 17000 bytes. Although in
practice it will generally be much smaller, we require a larger buffer
just to be safe.
CC: libav-stable@libav.org
ff_wma_init() allows up to 50kHz, but this generates an exponent band
size table that requires 65 bands. The code assumes 25 bands in many
places, and using sample rates higher than 48kHz will lead to buffer
overwrites.
CC:libav-stable@libav.org
This is near the theoretical limit for wma frame size and is the most that
our decoder can handle. Allowing higher bit rates will just end up padding
each frame with empty bytes.
Fixes invalid writes for avconv when using very high bit rates.
CC:libav-stable@libav.org
The code only supports 16 and 24 bps currently, 20bps causes
out of array reads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When reading sequentially, we are using the actual flag from the previous
frame, but when seeking we do not know what the previous window flag was, so
we need to read it from the bitstream.
This fixes some out of global array accesses of dither_tab.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Benjamin Larsson <benjamin@southpole.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
qpeg should probably be changed to use the checked bytestream reader.
But for now this fixes it and is significantly less work.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (29 commits)
amrwb: remove duplicate arguments from extrapolate_isf().
amrwb: error out early if mode is invalid.
h264: change underread for 10bit QPEL to overread.
matroska: check buffer size for RM-style byte reordering.
vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
vp8: change int stride to ptrdiff_t stride.
wma: fix invalid buffer size assumptions causing random overreads.
Windows Media Audio Lossless decoder
rv10/20: Fix slice overflow with checked bitstream reader.
h263dec: Disallow width/height changing with frame threads.
rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
rmdec: Honor .RMF tag size rather than assuming 18.
g722: Fix the QMF scaling
r3d: don't set codec timebase.
electronicarts: set timebase for tgv video.
electronicarts: parse the framerate for cmv video.
ogg: don't set codec timebase
electronicarts: don't set codec timebase
avs: don't set codec timebase
wavpack: Fix an integer overflow
...
Conflicts:
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/fraps.c
libavcodec/h264.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/msmpeg4.c
libavcodec/pnmdec.c
libavcodec/qpeg.c
libavcodec/rawenc.c
libavcodec/ulti.c
libavcodec/vcr1.c
libavcodec/version.h
libavcodec/wmalosslessdec.c
libavformat/electronicarts.c
libswscale/ppc/yuv2rgb_altivec.c
tests/ref/acodec/g722
tests/ref/fate/ea-cmv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prevents warnings because the dst and src overlap (are the same) in the
memcpy() inside the function.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Prevents using the invalid mode as an index in a static array, which
would generate invalid reads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Decodes 16-bit WMA Lossless encoded files. 24-bit is not supported yet.
Bitstream parser written by Andreas Öman with contributions from
Baptiste Coudurier and Ulion.
Includes a number of bug-fixes from Benjamin Larsson, Michael Niedermayer and
Konstantin Shishkov, shine and polish by Diego Biurrun.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Fixes out of bounds read.
Checked against SMPTE 421M-2006
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Signed-off-by: Martin Storsjö <martin@martin.st>
Integer Overflow Checker detected an integer
overflow while FATE was running.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
h264: error out on invalid bitdepth.
aacsbr: use a swap index for the Y matrix rather than copy buffers.
huffyuv: do not abort on unknown pix_fmt; instead, return an error.
lcl: return negative error codes on decode_init() errors.
rtpenc: Use MB info side data for splitting H263 packets for RFC 2190
h263enc: Add an option for outputting info about MBs as side data
avpacket: Add a function for shrinking already allocated side data
nellymoserdec: Saner and faster IMDCT windowing
Conflicts:
doc/APIchanges
libavcodec/avpacket.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes invalid reads while initializing the dequant tables, which uses
the bit depth to determine the QP table size.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This is based on the reference implementation and fixes
a global out of array read.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a out of global array read.
This change is based on the reference mpc imlementation.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It appears there are corner cases with damaged input that can lead
to small overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Code ported from qatar/master, please see there for per line authorship.
Main authors AFAIK are Ronald and Justin. I have no authorship on this.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Its not clear from the spec what to do with values larger than 127
so iam opting for the safe side and ask for a sample.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With the encode2 API, encoders allocate huge packets to be
sure they have enough room (a typical case is mpeg4, which
allocs ~10M for 1280x768 yuv420p) but only actually use a
very small part of the buffer.
ProRes codes chroma blocks in 444 mode in different order than luma blocks,
so make both decoder and encoder read/write chroma blocks in right order.
Reported by Phil Barrett
[alex.converse@mgail.com]
Move code to get_che()
Update for AAC new channel configuration interface
Only set chan_config if output_configure succeeds.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This is somewhat redundant as no decoder should call get_buffer() with such argument.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
ALS spec:
11.6.3.1.1 Quantization and encoding of parcor coefficients
...
In all cases the resulting quantized values ak are restricted to the range [-64,63].
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
adpcm: Clip step_index values read from the bitstream at the beginning of each frame.
oma: don't read beyond end of leaf_table.
doxygen: Remove documentation for non-existing parameters; misc small fixes.
Indeo3: fix crashes on corrupt bitstreams.
msmpeg4: Replace forward declaration by proper #include.
segment: implement wrap around
avf: reorder AVStream and AVFormatContext
aacdec: Remove erroneous reference to global gain from the out of bounds scalefactor error message.
Conflicts:
libavcodec/indeo3.c
libavformat/avformat.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Splits at borders of cells are invalid, since it leaves one of the
cells with a width/height of zero. Also, propagate errors on buffer
allocation failures, so we don't continue decoding (which crashes).
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
avcodec_default_reget_buffer(): fix compilation in DEBUG mode
fate: Overhaul WavPack coverage
h264: fix mmxext chroma deblock to use correct TC values.
flvdec: Remove the now redundant check for known broken metadata creator
flvdec: Validate index entries added from metadata while reading
rtsp: Handle requests from server to client
movenc: use timestamps instead of frame_size for samples-per-packet
movenc: use the first cluster duration as the tfhd default duration
movenc: factorize calculation of cluster duration into a separate function
doc/APIchanges: fill in missing dates and hashes.
lavc: reorder AVCodecContext fields.
lavc: reorder AVFrame fields.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavformat/flvdec.c
libavformat/movenc.c
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: don't guess r_frame_rate from either stream or codec timebase.
avconv: set discard on input streams automatically.
Fix parser not to clobber has_b_frames when extradata is set.
lavf: don't set codec timebase in avformat_find_stream_info().
avconv: saner output video timebase.
rawdec: set timebase to 1/fps.
avconv: refactor vsync code.
FATE: remove a bunch of useless -vsync 0
cdxl: bit line plane arrangement support
cdxl: remove early check for bpp
cdxl: set pix_fmt PAL8 only if palette is available
Conflicts:
ffmpeg.c
libavcodec/h264_parser.c
libavformat/rawdec.c
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/h264.mak
tests/fate/prores.mak
tests/fate/video.mak
tests/ref/fate/bethsoft-vid
tests/ref/fate/creatureshock-avs
tests/ref/fate/ea-cmv
tests/ref/fate/interplay-mve-16bit
tests/ref/fate/interplay-mve-8bit
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/fate/qtrle-16bit
tests/ref/fate/qtrle-1bit
tests/ref/fate/real-rv40
tests/ref/fate/rpza
tests/ref/fate/wmv8-drm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Because in contrast to the decoder, the parser does not setup low_delay.
The code in parse_nal_units would always end up setting has_b_frames
to "1", except when stream is explicitly marked as low delay.
Since the parser itself would create 'extradata', simply reopening
the parser would cause this.
This happens for instance in estimate_timings_from_pts(), which causes the
parser to be reopened on the same stream.
This fixes Libav #22 and FFmpeg (trac) #360
CC: libav-stable@libav.org
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(commit 31ac0ac29b)
Comments and description adapted by Reinhard Tartler.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master: (34 commits)
mlp_parser: fix the channel mask value used for the top surround channel
vorbisenc: check all allocations for failure
roqaudioenc: return AVERROR codes instead of -1
roqaudioenc: set correct bit rate
roqaudioenc: use AVCodecContext.frame_size correctly.
roqaudioenc: remove unneeded sample_fmt check
ra144enc: use int16_t* for input samples rather than void*
ra144enc: set AVCodecContext.coded_frame
ra144enc: remove unneeded sample_fmt check
nellymoserenc: set AVCodecContext.coded_frame
nellymoserenc: improve error checking in encode_init()
nellymoserenc: return AVERROR codes instead of -1
libvorbis: improve error checking in oggvorbis_encode_init()
mpegaudioenc: return AVERROR codes instead of -1
libfaac: improve error checking and handling in Faac_encode_init()
avutil: add AVERROR_UNKNOWN
check for coded_frame allocation failure in several audio encoders
audio encoders: do not set coded_frame->key_frame.
g722enc: check for trellis data allocation error
libspeexenc: export encoder delay through AVCodecContext.delay
...
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/fraps.c
libavcodec/kgv1dec.c
libavcodec/libfaac.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/mlp_parser.c
libavcodec/roqaudioenc.c
libavcodec/vorbisenc.c
libavutil/avutil.h
libavutil/error.c
libavutil/error.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is not allowed to change mid-stream like it does currently. Instead we need
to buffer the first 8 frames before returning them as a single packet, then
only return single frame packets after that.
* qatar/master:
docs: use -bsf:[vas] instead of -[vas]bsf.
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
lavf: move the packet keyframe setting code.
oggenc: free comment header for all codecs
lcl: error out if uncompressed input buffer is smaller than framesize.
mjpeg: abort decoding if packet is too large.
golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
get_bits: add HAVE_BITS_REMAINING macro.
lavf/output-example: use new audio encoding API correctly.
lavf/output-example: more proper usage of the new API.
tiff: Prevent overreads in the type_sizes array.
tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
apetag: do not leak memory if avio_read() fails
apetag: propagate errors.
SBR DSP x86: implement SSE sbr_hf_g_filt
SBR DSP x86: implement SSE sbr_sum_square_sse
SBR DSP: use intptr_t for the ixh parameter.
Conflicts:
doc/bitstream_filters.texi
doc/examples/muxing.c
doc/ffmpeg.texi
libavcodec/golomb.h
libavcodec/x86/Makefile
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prevents crash when trying to copy from a non-existing plane in e.g.
a RGB32 reference image to a YUV420P target image
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Instead of clipping extrasize based on EXTRABYTES, clip based on the
amount of buffer actually left. Without this fix, there are warbles
and other distortions in the test case below.
http://kevincennis.com/mix/assets/sounds/1901_voxfx.mp3
This prevents crashes when trying to read beyond the end of the buffer
while decoding frame data.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master: (40 commits)
swf: check return values for av_get/new_packet().
wavpack: Don't shift minclip/maxclip
rtpenc: Expose the max packet size via an avoption
rtpenc: Move max_packet_size to a context variable
rtpenc: Add an option for not sending RTCP packets
lavc: drop encode() support for video.
snowenc: switch to encode2().
snowenc: don't abuse input picture for storing information.
a64multienc: switch to encode2().
a64multienc: don't write into output buffer when there's no output.
libxvid: switch to encode2().
tiffenc: switch to encode2().
tiffenc: properly forward error codes in encode_frame().
lavc: drop libdirac encoder.
gifenc: switch to encode2().
libvpxenc: switch to encode2().
flashsvenc: switch to encode2().
Remove libpostproc.
lcl: don't overwrite input memory.
swscale: take first/lastline over/underflows into account for MMX.
...
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/APIchanges
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdiracenc.c
libavcodec/libxvidff.c
libavcodec/qtrleenc.c
libavcodec/tiffenc.c
libavcodec/utils.c
libavformat/mov.c
libavformat/movenc.c
libpostproc/Makefile
libpostproc/postprocess.c
libpostproc/postprocess.h
libpostproc/postprocess_altivec_template.c
libpostproc/postprocess_internal.h
libpostproc/postprocess_template.c
libswscale/swscale.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Unrolling the main loop to process, instead of 4 elements:
- 8: minor gain of 2 cycles (not worth the extra object size)
- 2: loss of 8 cycles.
Assigning STEP to a register is a loss. Output address (Y) is almost always
unaligned.
Timings:
- C (32/64 bits): 117/109 cycles
- SSE: 57 cycles
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The 32bits targets have been compiled with -mfpmath=sse for proper reference.
sbr_sum_square C /32bits: 82c (unrolled)/102c
C /64bits: 69c (unrolled)/82c
SSE/32bits: 42c
SSE/64bits: 31c
Use of SSE4.1 dpps to perform the final sum is slower.
Not unrolling to perform 8 operations in a loop yields 10 more cycles.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Since we are clipping before we shift the values to
16 or 32 bits, we should not shift the min/max clip
values to compensate.
Fixes 8 and 24 bit lossy decoding.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If the PNG filter is enabled, a PNG-style filter will run over the
input buffer, writing into the buffer. Therefore, if no zlib compression
was used, ensure that we copy into a temporary buffer, otherwise we
overwrite user-provided input data.
This prevents crashers and errors further down when reading nodes in the
empty tree.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
dxva2: don't check for DXVA_PictureParameters->wDecodedPictureIndex
img2: split muxer and demuxer into separate files
rm: prevent infinite loops for index parsing.
aac: fix infinite loop on end-of-frame with sequence of 1-bits.
mov: Add more HDV and XDCAM FourCCs.
lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields().
rmdec: when using INT4 deinterleaving, error out if sub_packet_h <= 1.
cdxl: correctly synchronize video timestamps to audio
mlpdec_parser: fix a few channel layouts.
Add channel names to channel_names[] array for channels added in b2890f5
movenc: Buffer the mdat for the initial moov fragment, too
flvdec: Ignore the index if the ignidx flag is set
flvdec: Fix indentation
movdec: Don't parse all fragments if ignidx is set
movdec: Restart parsing root-level atoms at the right spot
prores: use natural integer type for the codebook index
mov: Add support for MPEG2 HDV 720p24 (hdv4)
swscale: K&R formatting cosmetics (part I)
swscale: variable declaration and placement cosmetics
Conflicts:
configure
libavcodec/aacdec.c
libavcodec/mlp_parser.c
libavformat/flvdec.c
libavformat/img2.c
libavformat/isom.h
libavformat/mov.c
libavformat/movenc.c
libswscale/rgb2rgb.c
libswscale/rgb2rgb_template.c
libswscale/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The operations that use it require it to be promoted to a larger (natural)
type and thus perform sign extension on it.
While an optimal compiler may account for this, gcc 4.6 (for x86 Windows)
fails. Using the natural integer type provides a 2% speedup for Win64
and 1% for Win32.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
mpegvideo_enc: only allocate output packet when we know there will be output
Add names for more channel layouts to the channel layout map.
sunrast: Add a sample request for RMP_RAW colormap.
avcodec: do not override pts or duration from the audio encoder
Add prores regression test.
Enable already existing rso regression test.
Add regression test for "sox" format muxer/demuxer.
Add dpx encoding regression test.
swscale: K&R formatting cosmetics for PowerPC code (part I/II)
img2: Use ff_guess_image2_codec(filename) shorthand where appropriate.
Clarify licensing information about files borrowed from libjpeg.
Mark mutable static data const where appropriate.
avplay: fix -threads option
dvbsubdec: avoid undefined signed left shift in RGBA macro
mlpdec: use av_log_ask_for_sample()
gif: K&R formatting cosmetics
png: make .long_name more descriptive
movdec: Adjust keyframe flagging in fragmented files
rv34: change most "int stride" into "ptrdiff_t stride".
Conflicts:
avprobe.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/mpegvideo_enc.c
libavcodec/pngenc.c
libavcodec/x86/v210-init.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_drawtext.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_setpts.c
libavfilter/vf_settb.c
libavformat/img2.c
libavutil/audioconvert.c
tests/codec-regression.sh
tests/lavf-regression.sh
tests/ref/lavf/dpx
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
adpcmenc: Use correct frame_size for Yamaha ADPCM.
avcodec: add ff_samples_to_time_base() convenience function to internal.h
adx parser: set duration
mlp parser: set duration instead of frame_size
gsm parser: set duration
mpegaudio parser: set duration instead of frame_size
(e)ac3 parser: set duration instead of frame_size
flac parser: set duration instead of frame_size
avcodec: add duration field to AVCodecParserContext
avutil: add av_rescale_q_rnd() to allow different rounding
pnmdec: remove useless .pix_fmts
libmp3lame: support float and s32 sample formats
libmp3lame: renaming, rearrangement, alignment, and comments
libmp3lame: use the LAME default bit rate
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
libmp3lame: cosmetics: remove some pointless comments
libmp3lame: convert some debugging code to av_dlog()
libmp3lame: remove outdated comment.
libmp3lame: do not set coded_frame->key_frame.
libmp3lame: improve error handling in MP3lame_encode_init()
...
Conflicts:
doc/APIchanges
libavcodec/libmp3lame.c
libavcodec/pcxenc.c
libavcodec/pnmdec.c
libavcodec/pnmenc.c
libavcodec/sgienc.c
libavcodec/utils.c
libavformat/hls.c
libavutil/avutil.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This prevents having to sign-extend on 64-bit systems with 32-bit ints,
such as x86-64. Also fixes crashes on systems where we don't do it and
arguments are not in registers, such as Win64 for all weight functions.
We now require at least libmp3lame 3.98.3.
lame_encode_buffer_interleaved() still doesn't work for mono, but it does not
"die"; it just expects a stereo interleaved buffer.
* qatar/master:
doxy: remove reference to removed api
examples: unbreak compilation
ttadec: cosmetics: reindent
sunrast: use RLE trigger macro inplace of the hard coded value.
sunrastenc: set keyframe flag for the output packet.
mpegvideo_enc: switch to encode2().
mpegvideo_enc: force encoding delay of at least 1 frame when low_delay=0
Conflicts:
doc/examples/muxing.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
Perform inter-channel decorr. only if both channels are coded
Use fixed-length array in revert_mclms()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the following commit to extrapolate better dts for the first
frame. Pts difference between the first two frames is reused as the
difference between pts and dts of the first frame.
This zeros all the memory once and avoids valgrind warnings.
alternatively the warnings could be suppressed.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov: Use defines for sample flags in fragments
mov: Use defines for trun flags
mov: Use defines for tfhd flags
proresenc: force bitrate not to exceed given limit
vc1parse: call vc1_init_common().
wma: don't return 0 on invalid packets.
asf: prevent packet_size_left from going negative if hdrlen > pktlen.
mjpegb: don't return 0 at the end of frame decoding.
rtpdec: Identify incorrectly signalled H263
vp8dsp: split long line.
aiff: don't skip block_align==0 check on COMM-after-SSND files.
dpcm: ignore extra unpaired bytes in stereo streams.
mp3on4: require a minimum framesize.
mpc7: assign an error level + context to av_log() msg.
huffyuv: error out on bit overrun.
dct-test: Add the missing ff_ prefix to the altivec functions
dct-test: Remove a stray declaration of a nonexistent function
movenc: Write the unknown duration as 64 bit fields in ismv
movenc: Write track durations with all bits set if duration is unknown
Conflicts:
libavcodec/dct-test.c
libavcodec/wmadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Apple ProRes Format Specifications mentions target data size for every frame,
so make sure frame meets it. This also allows encoder to demand much smaller
packet sizes for output.
The parser uses VLC tables initialized in vc1_common_init(), therefore
we should call this function on parser init also.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Return 0 means "please return the same data again", i.e. it causes an
infinite loop. Instead, return an error.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Return 0 indicates "please return the same data again", i.e. it causes
an infinite loop. Instead, return that we consumed the buffer if we
finished decoding succesfully, or return an error if an error occurred.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master: (22 commits)
als: prevent infinite loop in zero_remaining().
cook: prevent div-by-zero if channels is zero.
pamenc: switch to encode2().
svq1enc: switch to encode2().
dvenc: switch to encode2().
dpxenc: switch to encode2().
pngenc: switch to encode2().
v210enc: switch to encode2().
xwdenc: switch to encode2().
ttadec: use branchless unsigned-to-signed unfolding
avcodec: add a Sun Rasterfile encoder
sunrast: Move common defines to a new header file.
cdxl: fix video decoding for some files
cdxl: fix audio for some samples
apetag: add proper support for binary tags
ttadec: remove dead code
swscale: make access to filter data conditional on filter type.
swscale: update context offsets after removal of AlpMmxFilter.
prores: initialise encoder and decoder parts only when needed
swscale: make monowhite/black RGB-independent.
...
Conflicts:
Changelog
libavcodec/alsdec.c
libavcodec/dpxenc.c
libavcodec/golomb.h
libavcodec/pamenc.c
libavcodec/pngenc.c
libavformat/img2.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If bufsize < headersize, init_get_bits() will be called with a negative
number, causing it to fail and any subsequent call to get_bits() will
crash because it reads from a NULL pointer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
On EOF, get_bits() will continuously return 0, causing an infinite
loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The unused code being removed is for encoding only and therefore is not needed
by the decoder.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
WMApro actually support 13-bits block sizes (potentially even up to 14),
and thus we should support that also. If we get block sizes beyond what
the decoder can handle (14 is possible depending on s->decode_flags),
error out instead of crashing.
* qatar/master:
shorten: Use separate pointers for the allocated memory for decoded samples.
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples counts.
movenc: Don't set a default sample duration when creating ismv
rtp: Factorize the check for distinguishing RTCP packets from RTP
golomb: avoid infinite loop on all-zero input (or end of buffer).
bethsoftvid: synchronize video timestamps with audio sample rate
bethsoftvid: add audio stream only after getting the first audio packet
bethsoftvid: Set video packet duration instead of accumulating pts.
bethsoftvid: set packet key frame flag for audio and I-frame video packets.
bethsoftvid: fix read_packet() return codes.
bethsoftvid: pass palette in side data instead of in a separate packet.
sdp: Ignore RTCP packets when autodetecting RTP streams
proresenc: initialise 'sign' variable
mpegaudio: replace memcpy by SIMD code
vc1: prevent using last_frame as a reference for I/P first frame.
Conflicts:
libavcodec/atrac3.c
libavcodec/golomb.h
libavcodec/shorten.c
libavcodec/ws-snd1.c
tests/ref/fate/bethsoft-vid
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes invalid free() if any of the buffers are not allocated due to either
not decoding a header or an error prior to allocating all buffers.
Fixes CVE-2012-0858
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Add a check to avoid writing past the end of the channel_unit.components[]
array.
Bug Found by: cosminamironesei
Fixes CVE-2012-0853
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This makes the check that avoids overwrite of the samples array actually
work properly.
fixes CVE-2012-0848
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
By replacing memcpy with an unrolled loop using the alignment knowledge
it has, some speedup can be obtained.
Before (gcc 4.6.1): ~400 cycles
After: ~370 cycles
Overall, around 2% speed increase when decoding a 2400s mp3 to f32le.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* shariman/wmall:
Do not try to read residue if ave_mean <= 1
Move some variable declarations to comply with C90
Cosmetics: fix some whitespace errors
Support 24-bit decoding
wmall: remove ;;
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise, we end up with with log(0) or log(1). av_ceil_log2 simply
assumes the argument is non-zero and returns wrong result when it is.
(Not that there is a proper way of returning an undefined value.)
Since quantisation matrices are stored in context, decoding slices with
different quantisers in parallel leads to unpredictable content of
aforementioned matrices and wrong output picture thereof.
* qatar/master: (21 commits)
CDXL demuxer and decoder
hls: Re-add legacy applehttp name to preserve interface compatibility.
hlsproto: Rename the functions and context
hlsproto: Encourage users to try the hls demuxer instead of the proto
doc: Move the hls protocol section into the right place
libavformat: Rename the applehttp protocol to hls
hls: Rename the functions and context
libavformat: Rename the applehttp demuxer to hls
rtpdec: Support H263 in RFC 2190 format
rv30: check block type validity
ttadec: CRC checking
movenc: Support muxing VC1
avconv: Don't split out inline sequence headers when stream copying VC1
rv34: handle size changes during frame multithreading
rv40: prevent undefined signed overflow in rv40_loop_filter()
rv34: use AVERROR return values in ff_rv34_decode_frame()
rv34: use uint16_t for RV34DecContext.deblock_coefs
librtmp: Add "lib" prefix to librtmp URLProtocol declarations.
movenc: Use defines instead of hardcoded numbers for RTCP types
smjpegdec: implement seeking
...
Conflicts:
Changelog
doc/general.texi
libavcodec/avcodec.h
libavcodec/rv30.c
libavcodec/tta.c
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec: Use 4 byte startcodes for H.264
matroskadec: Mark variable as av_unused.
Move some conditionally used variables into the block where they are used.
Drop some completely unnecessary av_unused attributes.
swscale: Remove unused variable alpMmxFilter.
Drop unnecessary av_uninit attributes from some variable declarations.
movenc: Support muxing wmapro in ismv/isma
mpegtsenc: Add an AVOption for forcing a new PAT/PMT/SDT to be written
swscale: move YUV2PACKED16WRAPPER() macro down to where it is used.
swscale: handle gray16 as a "planar" YUV format (Y-only, of course).
swscale: use yuv2packed1() functions for unscaled chroma also.
swscale: fix incorrect chroma bias in yuv2rgb48_1_c().
swscale: fix invalid memory accesses in yuvpacked1() functions.
Move PS2 MMI code below the mips subdirectory, where it belongs.
mips: Move MMI function declarations to a header.
build: Set correct dependencies for rtmp* protocols implemented by librtmp.
Conflicts:
libavcodec/ac3enc_template.c
libavformat/mpegtsenc.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The wrong variable was passed into decode_ham_plane32()
Fixes: Ticket922
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
swscale: convert yuv2yuvX() to using named arguments.
swscale: rename "dstw" to "w" to prevent name collisions.
swscale: use named registers in yuv2yuv1_plane() place.
lavf: fix aspect ratio mismatch message.
avconv: set AVFormatContext.duration from '-t'
cljr: implement encode2.
cljr: set the properties of the coded_frame, not input frame.
dnxhdenc: switch to encode2.
bmpenc: switch to encode2().
Conflicts:
libavcodec/bmpenc.c
libavcodec/cljr.c
libavformat/utils.c
tests/ref/vsynth1/cljr
tests/ref/vsynth2/cljr
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cleanup is only done now when
a picture is returned (assuming that it has to be done when its returned)
a error is returned (assuming that there will be no further progress on the frame)
the codec is not h264 (this is still needed due to some deadlocks in realvideo)
This fixes a decoding regression with 00017.MTS
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (38 commits)
v210enc: remove redundant check for pix_fmt
wavpack: allow user to disable CRC checking
v210enc: Use Bytestream2 functions
cafdec: Check return value of avio_seek and avoid modifying state if it fails
yop: Check return value of avio_seek and avoid modifying state if it fails
tta: Check return value of avio_seek and avoid modifying state if it fails
tmv: Check return value of avio_seek and avoid modifying state if it fails
r3d: Check return value of avio_seek and avoid modifying state if it fails
nsvdec: Check return value of avio_seek and avoid modifying state if it fails
mpc8: Check return value of avio_seek and avoid modifying state if it fails
jvdec: Check return value of avio_seek and avoid modifying state if it fails
filmstripdec: Check return value of avio_seek and avoid modifying state if it fails
ffmdec: Check return value of avio_seek and avoid modifying state if it fails
dv: Check return value of avio_seek and avoid modifying state if it fails
bink: Check return value of avio_seek and avoid modifying state if it fails
Check AVCodec.pix_fmts in avcodec_open2()
svq3: Prevent illegal reads while parsing extradata.
remove ParseContext1
vc1: use ff_parse_close
mpegvideo parser: move specific fields into private context
...
Conflicts:
libavcodec/4xm.c
libavcodec/aacdec.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/mpeg4video_parser.c
libavcodec/svq3.c
libavcodec/v210enc.c
libavformat/cafdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The DC coefficient should be included, too.
This probably was missed because DC quantizer is always
even for MPEG-1/2 but this function is also used for MPEG-4.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Conversion of the luma intra prediction mode to one of the constrained
("alzheimer") ones can happen by crafting special bitstreams, causing
a crash because we'll call a NULL function pointer for 16x16 block intra
prediction, since constrained intra prediction functions are only
implemented for chroma (8x8 blocks).
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
in , else (1) { if (!1) } the if conditional will never evaluate to be true.
So as making the check useless.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
We need to do unsigned saturation in order to cover the corner case when the
absolute coefficient value is 16777215 (the maximum value).
Fixes Bug #216
That way all mix levels as exported by avpriv_ac3_parse_header()
will have the same meaning.
Previously the 3-bit center mix level for E-AC-3 was used to index in a
4-entry table, leading to out-of-array reads.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
* qatar/master: (26 commits)
eac3dec: replace undefined 1<<31 with INT32_MIN in noise generation
yadif: specify array size outside DECLARE_ALIGNED
prores: specify array size outside DECLARE_ALIGNED brackets.
WavPack demuxer: set packet duration
tta: use skip_bits_long()
mxfdec: Ignore the last entry in Avid's index table segments
mxfdec: Sanity-check SampleRate
mxfdec: Handle small EditUnitByteCount
mxfdec: Consider OPAtom files that do not have exactly one EC to be OP1a
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 eof_reached in mxf_read_local_tags()
mxfdec: Check for NULL component
mxfdec: Make sure mxf->nb_index_tables > 0 in mxf_packet_timestamps()
mxfdec: Make sure x < index_table->nb_ptses
build: Add missing directories to DIRS declarations.
...
Conflicts:
doc/build_system.txt
doc/fate.texi
libavfilter/x86/yadif_template.c
libavformat/mxfdec.c
libavutil/Makefile
tests/fate/audio.mak
tests/fate/prores.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/ref/fate/bethsoft-vid
tests/ref/fate/cscd
tests/ref/fate/dfa4
tests/ref/fate/nuv
tests/ref/fate/vp8-sign-bias
tests/ref/fate/wmv8-drm
tests/ref/lavf/gxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Comment-by-michael: iam commiting this as the code cannot work without it and likely works with it.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes crashes on exit when closing a bitstream filter that
hasn't allocated any private data, on OS X.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
pixdesc: mark pseudopaletted formats with a special flag.
avconv: switch to avcodec_encode_video2().
libx264: implement encode2().
libx264: split extradata writing out of encode_nals().
lavc: add avcodec_encode_video2() that encodes from an AVFrame -> AVPacket
cmdutils: update copyright year to 2012.
swscale: sign-extend integer function argument to qword on x86-64.
x86inc: support yasm -f win64 flag also.
h264: manually save/restore XMM registers for functions using INIT_MMX.
x86inc: allow manual use of WIN64_SPILL_XMM.
aacdec: Use correct speaker order for 7.1.
aacdec: Remove incorrect comment.
aacdec: Simplify output configuration.
Remove Sun medialib glue code.
dsputil: set STRIDE_ALIGN to 16 for x86 also.
pngdsp: swap argument inversion.
Conflicts:
cmdutils.c
configure
doc/APIchanges
ffmpeg.c
libavcodec/aacdec.c
libavcodec/dsputil.h
libavcodec/libx264.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/utils.c
libavfilter/vf_scale.c
libavutil/avutil.h
libswscale/mlib/yuv2rgb_mlib.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This changes a number of FATE results, since before this commit, the
timestamps in all tests using rawenc were made up by lavf.
In most cases, the previous timestamps were completely bogus.
In some other cases -- raw formats, mostly h264 -- the new timestamps
are bogus as well. The only difference is that timestamps invented by
the muxer are replaced by timestamps invented by the demuxer.
cscd -- avconv sets output codec timebase from r_frame_rate
and r_frame_rate is in this case some guessed number 31.42 (377/12),
which is not accurate enough to represent all timestamps. This results
in some frames having duplicate pts. Therefore, vsync 0 needs to be
changed to vsync 2 and avconv drops two frames. A proper fix in the
future would be to set output timebase to something saner in avconv.
nuv -- previous timestamps for video were wrong AND the cscd
comment applies, one frame is dropped.
vp8-signbias -- the file contains two frames with identical timestamps,
so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one
frame.
vc1-ism -- apparrently either the demuxer lies about timestamps or the
file is broken, since dts == pts on all packets, but reordering clearly
takes place.
The spec says the following speaker mapping is default:
center front speaker
left, right center front speakers,
left, right outside front speakers,
left surround, right surround rear speakers,
front low frequency effects speaker
* qatar/master:
swscale: make yuv2yuv1 use named registers.
h264: mark h264_idct_add8_10 with number of XMM registers.
swscale: fix V plane memory location in bilinear/unscaled RGB/YUYV case.
vp8: always update next_framep[] before returning from decode_frame().
avconv: estimate next_dts from framerate if it is set.
avconv: better next_dts usage.
avconv: rename InputStream.pts to last_dts.
avconv: reduce overloading for InputStream.pts.
avconv: rename InputStream.next_pts to next_dts.
avconv: rework -t handling for encoding.
avconv: set encoder timebase for subtitles.
pva-demux test: add -vn
swscale: K&R formatting cosmetics for SPARC code
apedec: allow the user to set the maximum number of output samples per call
apedec: do not unnecessarily zero output samples for mono frames
apedec: allocate a single flat buffer for decoded samples
apedec: use sizeof(field) instead of sizeof(type)
swscale: split C output functions into separate file.
swscale: Split C input functions into separate file.
bytestream: Add bytestream2 writing API.
The avconv changes are due to massive regressions and bugs not merged yet.
Conflicts:
ffmpeg.c
libavcodec/vp8.c
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/fate/demux.mak
tests/ref/lavf/asf
tests/ref/lavf/avi
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/nut
tests/ref/lavf/ogg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_avi
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_rm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes crashes in e.g. PNG decoding with SSE2 enabled. In fact, many
x86 optimizations for codecs assume that our buffer strides are 16-byte
aligned.
Also slightly move around code not allocate a new frame if we won't
decode it. This prevents us from putting undecoded frames in frame
pointers, which (in mt decoding) other threads will use and wait on
as references, causing a deadlock (if we skipped decoding) or a crash
(if we didn't initialized next_framep[] at all).
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
It makes sense in some cases to split up the output packet to save on memory
usage (ape frames can be very large), but the current/default size is
arbitrary. Allowing the user to configure this gives more flexibility and
requires minimal additional code.
* qatar/master:
Revert "v210enc: use FFALIGN()"
doxygen: Do not include license boilerplates in Doxygen comment blocks.
avplay: reset decoder flush state when seeking
ape: skip packets with invalid size
ape: calculate final packet size instead of guessing
ape: stop reading after the last frame has been read
ape: return AVERROR_EOF instead of AVERROR(EIO) when demuxing is finished
ape: return error if seeking to the current packet fails in ape_read_packet()
avcodec: Clarify AVFrame member documentation.
v210dec: check for coded_frame allocation failure
v210enc: use stride as it is already calculated
v210enc: use FFALIGN()
v210enc: return proper AVERROR codes instead of -1
v210enc: do not set coded_frame->key_frame
v210enc: check for coded_frame allocation failure
drawtext: add 'fix_bounds' option on coords fixing
drawtext: fix text_{w, h} expression vars
drawtext: add missing braces around an if() block.
Conflicts:
libavcodec/arm/vp8.h
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/v210dec.c
libavfilter/vf_drawtext.c
libavformat/ape.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
FFALIGN doesn't work with non-powers-of-2.
This reverts commit 7ad1b612c8.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
