The FATE sample contains some pixels with value 0, but the palette
stored in the file contains only values from 16 up. Because the default
and cmdutils get_buffer() initialize the data to 0x80, they appear as
gray dots.
After this commit they change to black dots, which is probably still
incorrect but less visible and doesn't rely on get_buffer() initializing
the data.
We have to make some symetric changes elsewhere as this increases
the precission with which samples are stored.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '69c25c9284645cf5189af2ede42d6f53828f3b45':
dnxhdenc: fix invalid reads in dnxhd_mb_var_thread().
x86: h264qpel: Move stray comment to the right spot and clarify it
atrac3: use correct loop variable in add_tonal_components()
Conflicts:
tests/ref/vsynth/vsynth1-dnxhd-1080i
tests/ref/vsynth/vsynth2-dnxhd-1080i
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e6bc38fd49c94726b45d5d5cc2b756ad8ec49ee0':
wmv2: move IDCT to its own DSP context.
Conflicts:
libavcodec/dsputil.h
tests/ref/seek/vsynth2-wmv2
tests/ref/vsynth/vsynth1-wmv2
tests/ref/vsynth/vsynth2-wmv2
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8a4f26206d7914eaf2903954ce97cb7686933382':
dsputil: remove butterflies_float_interleave.
srtp: Move a variable to a local scope
srtp: Add tests for the crypto suite with 32/80 bit HMAC
Conflicts:
libavcodec/x86/dsputil.asm
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows us to remove FF_IDCT_WMV2, which serves no practical purpose
other than to be able to select the WMV2 IDCT for MPEG (or vice versa)
and get corrupt output.
Fate tests for all wmv2-related tests change, because (for some obscure
reason) they forced use of the MPEG IDCT. You would get the same changes
previously by not using -idct simple in the fate test (or replacing it
with -idct auto).
* commit '0eecafc948b74c247ebbc59f18f508db5d590d0b':
configure: Make the new srtp protocol depend on the rtp protocol
lavf: Add a fate test for the SRTP functions
lavu: Add a fate test for the HMAC API
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'abae27ed3acd0a7c54f11760c5be2d2653c4edf8':
rtpdec: Fix the calculation of expected number of packets
fate: vp3: Fix fate-vp3-coeff-level64 test dependencies
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ec86ba57312745fd7ad9771e3121e79c6aacba30':
vp3: Fix double free in vp3_decode_end()
fate: Split fate-siff test into demuxer and VB video decoder test
Conflicts:
tests/fate/demux.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add some additional checks for EOF and print error messages on an incomplete
header or packet.
FATE reference updated for id-cin-video due to the demuxer no longer
returning a partial video packet at EOF.
This is a port of virtual dub's histogram equalization filter by Donald
A. Graft. Based on the work by Jérémy Tran <tran.jeremy.av@gmail.com>,
done for SOCIS 2012.
This is a port of the kerndeint filter (libmpcodecs/vf_kerndeint) by
Donal A. Graft (original avisynth plugin author), and is based on the
work by Jérémy Tran <tran.jeremy.av@gmail.com> done for SOCIS 2012.
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.
The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.
Thanks to Daniel for helping out with the listening tests.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
The data does not contain timing or trailing line breaks anymore. In
addition to being less idiotic, it is consistent with other codecs and
thus allows more switches between formats and codecs. It also fixes the
issue of the trailing line returns being simple \n instead of CRLF in
the ASS rectangle dialogue (this is the reason of the FATE update).
* commit '5e6ee38bd3cef0dd05f1dd7977c71f3479eb6d01':
FATE: add cavs test
cavsdec: export picture type in the output frame
Conflicts:
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Current MicroDVD AVPackets contain timing information and trailing line
breaks. The data is now only composed of the markup data. Doing this
consistently between text subtitles decoders allows to use different
codec for various formats. For instance, MicroDVD markup is sometimes
found in some VPlayer files. Also, generally speaking, the subtitles
text decoders have no use of these timings (and they must not use them
since it would break any user timing adjustment).
Technically, this is a major ABI break. In practice, a mismatching
lavf/lavc will now error out for MicroDVD decoding. Supporting both
formats requires unnecessary complex and fragile code.
FATE needs update because line breaks in the ASS file were "\n" (because
that's what is used in the original file). ASS format expect "\r\n" line
breaks; this commit fixes this issue. Also note that this "\r\n"
trailing need to be moved at some point from the decoders to the ASS
muxer.
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
The SRT muxer is reponsible for separating events with two line breaks,
there is no need to add more than necessary. Similarly, other muxers
(such as Matroska) are not supposed to add line breaks at the end of the
payload.
Since 83cab07 audio stream time bases are based on SampleRate, not EditRate.
This fixes trac ticket #2029 and a few seeking issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Do not overwrite linesize set by get_buffer().
The last frame in the FATE test is not decoded anymore, since the file
is cut and a part of it is missing.
The initial testing of the VFW binary codec was flawed,
likely due to an AviSynth bug.
Re-testing using VirtualDub and various professional editing
applications has revealed it should have been flipped.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '9d5c62ba5b586c80af508b5914934b1c439f6652':
lavu/opt: do not filter out the initial sign character except for flags
eval: treat dB as decibels instead of decibytes
float_dsp: add vector_dmul_scalar() to multiply a vector of doubles
Conflicts:
libavutil/eval.c
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is broken, and results will be messed up when seeking.
This also fix duration displayed for streams when using -c copy.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
* commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6':
fate-seek: remove use of gnu make 3.82 only private modifier
fate: move vsynth reference files to their own directory
fate: move fate-acodec reference files to their own dir
configure: avplay now depends on avresample
fate: split dependencies for fate-seek tests
Conflicts:
configure
tests/fate/seek.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Each fate-seek test depends now only on the corresponding fate-acodec,
fate-vsynth2 or fate-lavf test which creates the file seek-tests
operates on. The tests and references are renamed to match the test they
depend on.
Without this exception files with ".gif" extension by default
recognized as input suitable for image2 demuxer rather than gif.
In order to pass image through gif demuxer it was necessary
to use -f gif option.
This change affected 'make fate' test results because previously
image2 demuxer and gif decoder took only first frame of multiframe
test data, which is no longer true with gif demuxer.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
The new code reads the input frame when its ready, the previous
code did read the input frame during start_frame at which point it
may not yet be available.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The following commit will make it useless.
The crop_scale_vflip FATE test changes because of off-by-one differences
in output when vflipped slices are passed to sws.
This fixes correctly storing and identifying PCM in nut.
Based on patch by Luca Barbato
Found-by: durandal_1707
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: add a bink version 'b' test
FATE: add a bink version 'i' test
libswscale: remove unnecessary direct #if LIBSWSCALE_VERSION_MAJOR
Conflicts:
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently FFM files generated with one versions of ffmpeg generally
cannot be read by another.
By spliting data into chunks, more fields can saftely be appended to
chunks as well as new chunks added.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise during scaling it will try to interpret input in the wrong way and
that leads to the test results disagreeing on different platforms and with
different optimizations.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
pixfmt: support more yuva formats
swscale: support gray to 9bit and 10bit formats
configure: rewrite print_config() function using awk
FATE: fix (AD)PCM test dependencies broken in e519990
Use ptrdiff_t instead of int for intra pred "stride" function parameter.
x86: use PRED4x4/8x8/8x8L/16x16 macros to declare intrapred prototypes.
Conflicts:
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavutil/pixfmt.h
libswscale/swscale_unscaled.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>
Previously if frame decoding failed it would be
silently reported as valid frame.
The fate ref is updated because sample have
truncated last video packet.
While here return meaningful error codes.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This fixes playback in some circumstances (like webm in firefox).
Regression after 2c34367b.
It is also matching the Matroska specifications:
http://matroska.org/technical/specs/notes.html, "The quick eye will
notice that if a Cluster's Timecode is set to zero, it is possible to
have Blocks with a negative Raw Timecode. Blocks with a negative Raw
Timecode are not valid."
* commit '0a7005bebd23ade7bb852bce0401af1a8fdbb723':
rtpdec_xiph: fix function return type
smjpeg: fix type of 'ret' variable in smjpeg_read_packet()
mpegvideo: remove write-only variable
Use proper return values in case of missing features
fate: add avstring test
rangecoder-test: Set error message log level to error, instead of debug
Conflicts:
libavcodec/aacdec.c
libavcodec/amrnbdec.c
libavcodec/mpegvideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b522000e9b2ca36fe5b2751096b9a5f5ed8f87e6':
avio: introduce avio_closep
mpegtsenc: set muxing type notification to verbose
vc1dec: Use correct spelling of "opposite"
a64multienc: change mc_frame_counter to unsigned
arm: call arm-specific rv34dsp init functions under if (ARCH_ARM)
svq1: Drop a bunch of useless parentheses
parseutils-test: do not print numerical error codes
svq1: K&R formatting cosmetics
Conflicts:
doc/APIchanges
libavcodec/svq1dec.c
libavcodec/svq1enc.c
libavformat/version.h
libavutil/parseutils.c
tests/ref/fate/parseutils
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This test only make sure that the idet filter does not crash.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f':
adxdec: use planar sample format
adpcmdec: use planar sample format for adpcm_thp
adpcmdec: use planar sample format for adpcm_ea_xas
adpcmdec: use planar sample format for adpcm_ea_r1/r2/r3
adpcmdec: use planar sample format for adpcm_xa
adpcmdec: use planar sample format for adpcm_ima_ws for vqa version 3
adpcmdec: use planar sample format for adpcm_4xm
adpcmdec: use planar sample format for adpcm_ima_wav
adpcmdec: use planar sample format for adpcm_ima_qt
pcmdec: use planar sample format for pcm_lxf
mace: use planar sample format
atrac1: use planar sample format
build: non-x86: Only compile mpegvideo optimizations when necessary
rtpdec_mpeg4: au_headers is a single array, simple av_free is enough
avcodec: free extended_data instead address of it
fate: Add tests of the ff_make_absolute_url function
url: Handle relative urls starting with two slashes
url: Handle relative urls being just a new query string
url: Don't treat slashes in query parameters as directory separators
Conflicts:
libavcodec/adxdec.c
libavcodec/mips/Makefile
libavcodec/pcm.c
libavcodec/utils.c
libavformat/Makefile
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The timebases before where only guranteed to be 1/fps precisse
and could cause AV sync errors on low fps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new fields are only printed when they differ from their defaults
this way only few fate refs change
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The fate checksum change is due to the header size having been wrong.
Credit&Authorship for the code belongs to Justin Ruggles
Blame for bugs in this merging of the code belong to the Commiter
Commit message by Commiter
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise, the last byte of each stream is left uninitialized.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While a 25 fps stream can in general store frame durations in 1/25
units, this is not true for the timestamps. For example a 25fps
and a 25000/1001 fps stream when they are stored together might have
a matching 0 timestamp point but when for example a chapter from
this is cut the new start is no longer aligned. The issue gets
MUCH worse when the streams are lower fps, like 1 or 2 fps.
This commit thus makes the muxer choose a multiple of the
framerate as timebase that is at least about 20 micro seconds precise
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this, when we use a finer timebase than neccessary to store
durations the demuxer still knows what the original timebase was.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This generalizes the previous work on disposition printing.
Disposition flags are shown in a dedicated section, which should improve
output intellegibility, extensibility and filtering operations.
This breaks output syntax with the recently introduced disposition
printing.
Also factorize the common options for the different mov-based tests.
Since the header is now on top in the last generated file, the data
offset in the seek test needed some updates as well.
The YUV channels of VP6 are encoded in a highly linear fashion which does
not have any slice-like concept to thread. The alpha channel of VP6A is
fairly independent of the YUV and comprises 40% of the work. This patch
uses the THREAD_SLICE capability to split the YUV and A decodes into
separate threads.
Two bugs are fixed by splitting YUV and alpha state:
- qscale_table from VP6A decode was for alpha channel instead of YUV
- alpha channel filtering settings were overwritten by YUV header parse
Signed-off-by: Ben Jackson <ben@ben.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
A flashsv2 block may have a "diff block" to indicate which scan lines of
the block are actually encoded. However, this diff block need not be
used when the entire block is coded.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In vp6 Advanced Profile, deblock filtering is conditionally enabled in
each frame header. In Simple Profile it should always be off. vp6 was
inheriting the wrong default from ff_vp56_init.
Signed-off-by: Ben Jackson <ben@ben.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously, the value given to put_bits was 10 bits long for positive
predictors, even though 9 bits were to be written. The extra bit could
in some cases overwrite existing bits in the bitstream writer cache.
This fixes a failed assert in put_bits.h, when running a version
built with -DDEBUG.
The fate test result gets slightly improved, thanks to getting rid
of the overwritten bits in the bitstream writer cache.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'ec36aa69448f20a78d8c4588265022e0b2272ab5':
x86: Fix linking with some or all of yasm, mmx, optimizations disabled
configure: Add more fine-grained SSE CPU capabilities flags
avfilter: x86: Use more precise compile template names
x86: cosmetics: Comment some #endifs for better readability
g723_1: add comfort noise generation
utvideoenc: Switch to dsputils' median prediction
utvideoenc: Avoid writing into the input picture
avtools: remove the distinction between func_arg and func2_arg.
avconv: make the -passlogfile option per-stream.
avconv: make the -pass option per-stream.
cmdutils: make -codecs print lossy/lossless flags.
lavc: add lossy/lossless codec properties.
Conflicts:
Changelog
cmdutils.c
configure
doc/APIchanges
ffmpeg.h
ffmpeg_opt.c
ffprobe.c
libavcodec/codec_desc.c
libavcodec/g723_1.c
libavcodec/utvideoenc.c
libavcodec/version.h
libavcodec/x86/mpegaudiodec.c
libavcodec/x86/rv40dsp_init.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Rewrite 10 bit dpx decoder to decode into GBRP10 color space
instead of converting to RGB48.
Add 12 bit decoder to decode into GBRP12 color space.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
audio_frame_queue: Clean up ff_af_queue_log_state debug function
dwt: Remove unused code.
cavs: convert cavsdata.h to a .c file
cavs: Move inline functions only used in one file out of the header
cavs: Move data tables used in only one place to that file
fate: Add a single symbol Ut Video decoder test
vf_hqdn3d: x86 asm
vf_hqdn3d: support 16bit colordepth
avconv: prefer user-forced input framerate when choosing output framerate
Conflicts:
ffmpeg.c
libavcodec/audio_frame_queue.c
libavcodec/dwt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
19% faster
smaller files
this may also fix possible integer overflows due to previous 32bit useage
Tested with libutvideo and our utvideo decoder, this patch does not change
decoder output in the test
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The failures on various architectures and compilers on the RGB(A)
tests seem to have been because of one-off YCbCr->RGB conversion
results. This should make the conversion results match on most if
not all code paths.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The failures on various architectures and compilers on the RGB(A)
tests seem to have been because of one-off YCbCr->RGB conversion
results. This should make the conversion results match on most if
not all code paths.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
fate: Add FATE tests for the Ut Video encoder
lavc: add Ut Video encoder
mpegvideo_enc: remove stray duplicate line from 7f9aaa4
swscale: x86: fix #endif comments in rgb2rgb template file
avconv: mark more options as expert.
avconv: split printing "main options" into global and per-file.
avconv: refactor help printing.
Conflicts:
Changelog
ffmpeg_opt.c
ffserver.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Unsurprisingly, if a timing-less subrip decoder is desireable, an
encoder is as well. With this in place, we can move on to remove
the use of the old encoder/decoder with embedded timing and move
all timing handling the (de)muxer where they belong.
Signed-off-by: Philip Langdale <philipl@overt.org>
Fixes Ticket1627
The fate change is due to ffmpeg no longer pushing audio timestamps
aggressively up (which is what caused the AV sync issues in the ticket)
but leaving them as they are.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
g723.1: fix addition overflow
g723.1: simplify and fix multiplication overflow
g723.1: deobfuscate an expression
g723.1: remove unused #includes
ARM: add missing "cc" clobber in av_clipl_int32_arm()
rtmp: Factorize the code by adding handle_invoke_error
rtmp: Factorize the code by adding handle_invoke_status
rtmp: Factorize the code by adding handle_invoke_result
libavutil: remove unused av_abort() macro
ffmenc: replace if/abort with assert()
libavutil: drop offsetof() fallback definition
libavutil: drop fallback definitions of INTxx_MIN/MAX
configure: Check for a sctp struct instead of just the header
configure: suncc: Add -xc99 to dependency flags, required on Solaris
doxygen: Fix function parameter names to match the code
doc: Drop obsolete shared libs cflags hint to workaround Cygwin gcc bugs
swf: Move shared table out of the header file
swf: Move swf_audio_codec_tags table to the only place it is used
fate: add G.723.1 decoder tests
Conflicts:
configure
doc/platform.texi
libavformat/Makefile
libavutil/arm/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code dependent on the input buffer matching the
buffer that has been provided by yadifs get_buffer.
The API does in now way gurantee this though its often true.
This fixes some out of array reads.
The regression test checksums change due to "out of picture" values
being initialized differently.
There should be no visual difference in the filters output
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavr: fix handling of custom mix matrices
fate: force pix_fmt in lagarith-rgb32 test
fate: add tests for lagarith lossless video codec.
ARMv6: vp8: fix stack allocation with Apple's assembler
ARM: vp56: allow inline asm to build with clang
fft: 3dnow: fix register name typo in DECL_IMDCT macro
x86: dct32: port to cpuflags
x86: build: replace mmx2 by mmxext
Revert "wmapro: prevent division by zero when sample rate is unspecified"
wmapro: prevent division by zero when sample rate is unspecified
lagarith: fix color plane inversion for YUY2 output.
lagarith: pad RGB buffer by 1 byte.
dsputil: make add_hfyu_left_prediction_sse4() support unaligned src.
Conflicts:
doc/APIchanges
libavcodec/lagarith.c
libavfilter/x86/gradfun.c
libavutil/cpu.h
libavutil/version.h
libswscale/utils.c
libswscale/version.h
libswscale/x86/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change introduces a basic encoder for 3GPP Timed Text subtitles,
also known as TX3G, Quicktime subtitles, or "movtext" in the existing
code.
This initial change doesn't attempt to write styling information,
and just writes the plain text of the subtitles. I intend to add
support for styles eventually, but it's challenging due to a lack
of existing players that support them.
Note that an additional change is required to the mov/mp4 muxer to
write empty subtitle packets to indicate subtitle duration.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
vc1dec: Remove separate scaling function for interlaced field MVs
vc1dec: Invoke edge_emulation regardless of MV precision
x86: Use consistent 3dnowext function and macro name suffixes
g723_1: scale output as supposed for the case with postfilter disabled
g723_1: increase excitation storage by 4
g723_1: fix upper bound parameter from inverse maximum autocorrelation
g723_1: make scale_vector() behave like the reference
g723_1: fix off-by-one error in normalize_bits()
g723_1: save/restore excitation with offset to store LPC history
wmapro: prevent division by zero when sample rate is unspecified
x86: proresdsp: improve SIGNEXTEND macro comments
x86: h264dsp: K&R formatting cosmetics
LICENSE: Document all GPL files
Conflicts:
libavcodec/g723_1.c
libavcodec/wmaprodec.c
libavcodec/x86/h264dsp_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
MMX-enabled systems by default use some dsputil functions differing
from the C versions. Adding these flags ensures accurate ones are
used everywhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
commit 20e88d8618
Fix avui stream-copy.
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These filters are designed for storing and transmitting video sequences
with alpha using higher-efficiency codecs such as x264 which don't
natively support an alpha channel. 'alphaextract' takes an input stream
with an alpha channel and returns a video containing just the alpha
component as a grayscale value; 'alphamerge' takes an RGB or YUV stream
and adds an alpha channel recovered from a second grayscale stream.
Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Convert them to zigzag order, as the rest of them are.
When I was adding support for 10-bit DNxHD, I just copy-pasted the
missing quant matrices from the spec. Now it turns out the existing
matrices in dnxhddata.c were in zigzag order. This resulted in wrong
quantization for 10-bit DNxHD. The attached patch fixes the problem by
converting 10-bit quant matrices to zigzag order.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
