* qatar/master:
movenc: Don't write the 'wave' atom or its child 'enda' for lpcm audio.
imc: some cosmetics
rtmp: Pass the proper return code in rtmp_handshake
rtmp: Check return codes of net IO operations
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Note, if you want something mentioned in the release notes for 0.11
push it but be real quick ...
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There have been multiple user complaints about loosing this feature
while its not clear the 3% speedloss claims where real or fabricated.
My own testing indicates no statistically significant speed difference
both with mpeg2 and mpeg4, and if at all the code with lowres support
is a tiny bit faster than without.
This reverts commit 92ef4be4ab, reversing
changes made to 2e07f42957.
Conflicts:
cmdutils.c
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/mpegvideo.c
libavcodec/mpegvideo.h
libavutil/arm/Makefile
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Return a proper error code instead of -1
rtmp: Check malloc calls
rtmp: Check ff_rtmp_packet_create calls
lavfi: add audio mix filter
flvdec: Make sure sample_rate is set to the updated value
tqi: Pass errors from the MB decoder
Conflicts:
Changelog
doc/filters.texi
libavcodec/eatqi.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This lists git history, which is better kept in the history itself
that is both ours as this file as well as the actual history of the
ffmpeg-mt project.
If someone thinks this is not ok, drop me a mail and ill put it back!
Also note this file was not carried in our previous release with noone
complaining
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
git blame:
77 Stefano Sabatini
1 Michael Niedermayer
Initial commit:
commit 2f83681c79
Author: Stefano Sabatini <stefasab@gmail.com>
Date: Sat Mar 10 14:01:28 2012 +0100
lavfi: port libmpcodecs remove-logo filter
The code is based on the remove-logo filter in MPlayer/libmpcodecs, by
Robert Edele, relicensed to LGPL with consent of the author.
Address trac issue #249.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
git blame:
132 Stefano Sabatini
77 Vitor Sessak
49 Michael Niedermayer
24 Anton Khirnov
22 S.N. Hemanth Meenakshisundaram
13 Bobby Bingham
7 Luca Barbato
2 Nicolas George
2 Alex Converse
1 Diego Elio Pettenò
Initial commit not traced as this file was split out.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Only commit:
commit 54c5dd89e3
Author: Anton Khirnov <anton@khirnov.net>
Date: Wed May 9 14:08:21 2012 +0200
lavfi: Add fps filter.
Partially based on a patch by Robert Nagy <ronag89@gmail.com>
also see [FFmpeg-devel] [PATCH 07/10] vf_fps: fix copyright
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
git blame:
75 Anton Khirnov
72 Michael Niedermayer
39 Stefano Sabatini
23 S.N. Hemanth Meenakshisundaram
9 Vitor Sessak
6 Robert Nagy
2 Diego Biurrun
1 Andrey Utkin
Note:
commit ab165047a6
Author: Anton Khirnov <anton@khirnov.net>
Date: Fri Apr 27 17:27:40 2012 +0200
lavfi: add a function for copying properties from AVFilterBufferRef->AVFrame
Based on a commit by Stefano Sabatini <stefano.sabatini-lala@poste.it>
commit 4a1ac8c43f
Author: Anton Khirnov <anton@khirnov.net>
Date: Thu May 10 07:58:11 2012 +0200
lavfi: move buffer management function to a separate file.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
git blame:
45 Stefano Sabatini
23 Clément Bœsch
4 Michael Niedermayer
3 Robert Nagy
3 Nicolas George
2 Roger Pau Monné
Initial commit:
commit 566666caf3
Author: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Date: Sun May 1 14:47:05 2011 +0200
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Move AVPALETTE_SIZE and AVPALETTE_COUNT definition from
libavcodec/avcodec.h to libavutil/pixfmt.h.
The definition is more useful in libavutil, where it can be shared for
example by libavfilter and libswscale.
* qatar/master:
os_support: Define SHUT_RD, SHUT_WR and SHUT_RDWR on OS/2
http: Add support for reading http POST reply headers
http: Add http_shutdown() for ending writing of posts
tcp: Allow signalling end of reading/writing
avio: Add a function for signalling end of reading/writing
lavfi: fix comment, audio is supported now.
lavfi: fix incorrect comment.
lavfi: remove avfilter_null_* from public API on next bump.
lavfi: remove avfilter_default_* from public API on next bump.
lavfi: deprecate default config_props() callback and refactor avfilter_config_links()
avfiltergraph: smarter sample format selection.
avconv: rename transcode_audio/video to decode_audio/video.
asyncts: reset delta to 0 when it's not used.
x86: lavc: use %if HAVE_AVX guards around AVX functions in yasm code.
dwt: return errors from ff_slice_buffer_init()
Conflicts:
ffmpeg.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/version.h
libavfilter/vf_blackframe.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_format.c
libavfilter/vf_showinfo.c
libavfilter/video.c
libavfilter/video.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
dwt: check malloc calls
ppc: Drop unused header regs.h
af_resample: remove an extra space in the log output
Convert vector_fmul range of functions to YASM and add AVX versions
lavfi: add an audio split filter
lavfi: rename vf_split.c to split.c
Conflicts:
doc/filters.texi
libavcodec/ppc/regs.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/f_split.c
libavfilter/split.c
libavfilter/version.h
libavfilter/vf_split.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
tcp_shutdown() isn't needed at the moment, but is added for
consistency to explain how the function is supposed to be used.
Signed-off-by: Martin Storsjö <martin@martin.st>
Link properties have to be checked after config_props() is called to
make sure everything is sane, so the default config_props() for output
links was redundant.
Remove now empty defaults.c
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This fixes passing junk in stream.
It should not have any user vissible effect.
We are discarding the new data in the decoder as no case is known
where it is needed but it causes problems if used.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The special-case behavior may complicate parsing when the
-show_format_entry option is used programmatically in a script.
The option default=nk=1 achieves the same purpose, if the objective is to
skip printing the single field key.
FATE_SAMPLES is now used directly by the Makefiles, which induces the test
system to run a test with the value of the environment variable as name.
Renaming the environment variable to LIBAV_SAMPLES avoids this problem.
* qatar/master:
indeo: Make ivi_calc_band_checksum() static, it is only used in one file.
indeo: Drop unused debug function ivi_check_band().
avcodec/utils: cast a function argument to shut up a compiler warning
truemotion1: remove disabled code
fix typo in comment
fate: fix dependencies for non-SAMPLES avconv tests
indeo: check for invalid motion vectors
indeo: check that band output buffer exists
indeo: clear allocated band buffers
indeo: track tile macroblock size
indeo: check custom Huffman tables for errors
factor out common decoding code for Indeo 4 and Indeo 5
mp3: fix start band index for block type 2 in 8kHz audio
lavf: change some (de)muxer names to lowercase
lavf: make output format matching case insensitive
Conflicts:
libavcodec/indeo4.c
libavcodec/indeo5.c
libavcodec/ivi_common.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ffmpeg -i in.mxf -filter_complex "[0:0]fieldorder=tff" out.wav will
fail with an error message instead of crashing.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use codec aspect ratio for frame aspect ratio if AVFrame is NULL.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
the header in the sample provided for ticket #1306 is not parsed correctly and thus
ffmpeg tries to decode the sample instead of abording the decoding.
I tested it with two other exr samples I have - one float, one half float - and
they still decode correctly.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The encode/decode tests should all depend on avconv. Since
avconv requires libavfilter, there is no need to enable those
tests selectively.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The stream can be specified as "#129" or "#0x81".
It is especially useful for VOBs dumped from a DVD,
where the language-id mapping is available externally
and the probing can find the streams in a random order.
In hybrid frames long window part ends at 36 samples for most of the cases
but at 72 for 8kHz case. For some reason decoder assumed it's 48 or even 36
samples, which caused wrong bitstream decoding for such blocks.
l3_25207.mpg from conformance suite demonstrates it the best.
* commit '755cd4197d53946208e042f095b930dca18d9430':
mov: enable parsing for VC-1.
lavfi: Add fps filter.
lavfi: initialize pts to AV_NOPTS_VALUE when creating new buffer refs.
avconv: add support for audio in complex filtergraphs.
Conflicts:
ffmpeg.c
libavfilter/version.h
libavformat/mov.c
tests/ref/fate/vc1-ism
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous version checked the filter context name,
instead of checking the filter name.
The new version just uses the same type as the input.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Most of the code is moved to buffersrc.c
to help Git see the similarities.
src_buffer.c still contains the obsolete compatibility functions.
The format change detection code was removed as it has been
implemented directly in ffmpeg.
It can easily be brought back from the history.
* qatar/master:
doc: Replace some @file tags by more suitable markup.
fate: Set FUZZ factor of vorbis-13 test to 2.
fate: Set FUZZ factor of (e)ac3-encode test to 3.
fate: remove unused code from regressions-funcs.sh
rtmp: Don't assume path points to a string of nonzero length
avconv: fix behavior with -ss as an output option.
Conflicts:
doc/platform.texi
doc/protocols.texi
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The errors need to be defined before including functions depending on
them. See av_size_mult() for instance. stddef.h is included for the
prototype of av_sterror (use of size_t).
Invented timestamps for the h264 tests return to something resembling
sanity.
In the idroq-video-encode test when converting 25 fps -> 30 fps the
fifth frame gets duplicated instead of the sixth.
* cus/stable:
ffplay: put aspect ratio info to the VideoPicture struct
ffplay: use AVFrame::width and height instead of using codec or filter settings
ffplay: use stream sample_aspect_ratio if available in source frames
ffplay: fix video_thread when no frame is returned in get_video_frame
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
pcmenc: set correct bitrate value
avprobe: don't print format entry name when only one was requested
Conflicts:
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If using the new -rtmp_app and -rtmp_playpath parameters,
one can in many cases set the main url to just rtmp://server/.
If the trailing slash is omitted, path is a string of zero length,
and using path+1 will end up reading uninitialized data.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 'audio-filters' of https://github.com/ubitux/FFmpeg:
lavfi/pan: add supported sample rates to avoid a crash.
ffmpeg: do not warn when expecting EOF from lavfi.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some of the FATE changes are due to off-by-one different rounding being used
(lrintf vs av_rescale_q).
Some fate changes are due to 1 audio frame less being encoded (the new variant seems
matching what qatar does and according to ffprobe its closer to the requested duration)
the mapchan feature sadly is lost in this commit because it depends on resampling
being done in ffmpeg.c which is now moved completely into the av filter layer
-async is broken after this commit, this will be fixed in subsequent commits
the new filter reconfiguration system is flawed and will drop a frame on each
parameter change which is why the nelly moser checksums need updating.
Conflicts:
ffmpeg.c
tests/ref/fate/smjpeg
* qatar/master:
fate: Work around non-standard wc implementations at more places
fate: work around non-standard wc implementations
x86: rv40: Mark rv40_weight functions as MMX2; they use MMX2 instructions.
ac3dsp: simplify x86 versions of ac3_max_msb_abs_int16
fate: use standard diff options
tta: Fix comment about channel number; TTA supports >2 channels.
avfilter: Move ff_get_ref_perms_string() to where it is used.
build: Add 'check' target to run all compile and test targets.
indeo3: validate new frame size before resetting decoder
indeo3: when freeing buffers, set pointers referencing them to NULL as well
indeo3: initialise pixel planes on allocation
indeo3: ensure that decoded cell data is in 7-bit range as presumed by decoder
fate: rename psx-str-v3-mdec to mdec-v3
fate: convert psx-str to a demuxer test
lavf: add mdec to is_intra_only() list
Conflicts:
doc/developer.texi
libavcodec/indeo3.c
libavfilter/video.c
libavformat/utils.c
tests/fate/demux.mak
tests/fate/video.mak
tests/lavf-regression.sh
tests/ref/vsynth1/cljr
tests/ref/vsynth1/ffvhuff
tests/ref/vsynth2/cljr
tests/ref/vsynth2/ffvhuff
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The planar/packed switch and the packing_formats list is no longer
required, since the planar/packed information is now stored in the sample
format enum.
This is technically a major API break, possibly it should be not too
painful as we marked the audio filtering API as unstable.
On some systems, the wc command prints spaces before the first
number causing mismatches with the test references. Using the
output of wc as arguments to echo removes any extra whitespace.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (26 commits)
fate: use diff -b in oneline comparison
Add missing version bumps and APIchanges/Changelog entries.
lavfi: move buffer management function to a separate file.
lavfi: move formats-related functions from default.c to formats.c
lavfi: move video-related functions to a separate file.
fate: make smjpeg a demux test
fate: separate sierra-vmd audio and video tests
fate: separate smacker audio and video tests
libmp3lame: set supported channel layouts.
avconv: automatically insert asyncts when -async is used.
avconv: add support for audio filters.
lavfi: add asyncts filter.
lavfi: add aformat filter
lavfi: add an audio buffer sink.
lavfi: add an audio buffer source.
buffersrc: add av_buffersrc_write_frame().
buffersrc: fix invalid read in uninit if the fifo hasn't been allocated
lavfi: rename vsrc_buffer.c to buffersrc.c
avfiltergraph: reindent
lavfi: add channel layout/sample rate negotiation.
...
Conflicts:
Changelog
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffprobe.c
libavcodec/libmp3lame.c
libavfilter/Makefile
libavfilter/af_aformat.c
libavfilter/allfilters.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/buffersrc.c
libavfilter/defaults.c
libavfilter/formats.c
libavfilter/src_buffer.c
libavfilter/version.h
libavfilter/vf_yadif.c
libavfilter/vsrc_buffer.c
libavfilter/vsrc_buffer.h
libavutil/avutil.h
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
diff -w is not a standard option. This fixes the reference files
to match what the tests actually output and switches to using the
standard diff -b which is sufficient to handle different line ending
styles.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This happened when a frame was removed before any was added.
Fixes part of Ticket1208
Found-by: John Villamil, Piotr Bandurski and Carl Eugen Hoyos
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also use av_guess_sample_aspect_ratio for determining aspect ratio of the video
frame if not using avfilter.
Signed-off-by: Marton Balint <cus@passwd.hu>
Codec values may not reflect the actual frame size, and it also enables us to
simplify code in the avfilter enabled and the avfilter disabled case.
Signed-off-by: Marton Balint <cus@passwd.hu>
When we are using filter chains we have to set the aspect ratio of the source
to the best known value, we use the av_guess_sample_aspect_ratio function to
determine that.
Fixes ticket 1228.
Signed-off-by: Marton Balint <cus@passwd.hu>
Guesses the sample aspect ratio of a frame, based on both the stream and the
frame aspect ratio.
Since the frame aspect ratio is set by the codec but the stream aspect ratio
is set by the demuxer, these two may not be equal. This function tries to
return the value that you should use if you would like to display the frame.
Basic logic is to use the stream aspect ratio if it is set to something sane
otherwise use the frame aspect ratio. This way a container setting, which is
usually easy to modify can override the coded value in the frames.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's the same as av_vsrc_buffer_add_frame(), except it doesn't take pts
or pixel_aspect parameters. Those are read from AVFrame.
Deprecate av_vsrc_buffer_add_frame().
* qatar/master: (25 commits)
vcr1: Add vcr1_ prefixes to all static functions with generic names.
vcr1: Fix return type of common_init to match the function pointer signature.
vcr1enc: Replace obsolete get_bit_count by put_bits_count/flush_put_bits.
motion-test: remove disabled code
gxfenc: remove disabled half-implemented MJPEG tag
x86: use more standard construct for setting ASM functions in FFT code
fate: westwood-aud: disable decoding
fate: caf: disable decoding
fate: film-cvid: drop pcm audio and rename test
fate: d-cinema-demux: drop unnecessary flags
fate: split off dpcm-interplay from interplay-mve tests
fate: rename funcom-iss to adpcm-ima-iss
fate: rename cryo-apc to adpcm-ima-apc
fate: rename adpcm-psx-str-v3 to adpcm-xa
fate: split off adpcm-ms-mono test from dxa-feeble
fate: split off adpcm-ima-ws test from vqa-cc
fate: add adpcm-ima-smjpeg test
fate: split off adpcm-ima-amv from amv test
fate: separate bmv audio and video tests
fate: separate delphine-cin audio and video tests
...
Conflicts:
doc/platform.texi
libavcodec/vcr1.c
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/video.mak
tests/ref/fate/ea-mad-pcm-planar
tests/ref/fate/interplay-mve-16bit
tests/ref/fate/interplay-mve-8bit
tests/ref/fate/mtv
tests/ref/fate/qtrle-1bit
tests/ref/fate/qtrle-2bit
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
tests/ref/fate/vqa-cc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The codec (adpcm-ima-ws) is tested elsewhere. Using framecrc output
provides more information than a single md5 if something goes wrong.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fix some orthography, wording and grammar issues; update the SDL section
with more current instructions; simplify lib.exe example command line;
drop outdated comments about libnut.
The width of wmv1/2 video must be multiple of 2 or win32 codec will fail to decode it (WMP displays black screen).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: employ better names and add a convenient shorthand for vp6 tests
arm/neon: dsputil: use correct size specifiers on vld1/vst1
arm: dsputil: prettify some conditional instructions in put_pixels macros
vqavideo: change x/y loop counters to the usual pattern
avconv: use lrint() for rounding double timestamps
Conflicts:
tests/ref/fate/vc1-ism
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Change the size specifiers to match the actual element sizes
of the data. This makes no practical difference with strict
alignment checking disabled (the default) other than somewhat
documenting the code. With strict alignment checking on, it
avoids trapping the unaligned loads.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Converting the double to float for lrintf() loses precision when
the value is not exactly representable as a single-precision float.
Apart from being inaccurate, this causes discrepancies in some
configurations due to differences in rounding.
Note that the changed timestamp in the vc1-ism test is a bogus,
made-up value.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (25 commits)
rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC
ape: Use unsigned integer maths
arm: dsputil: fix overreads in put/avg_pixels functions
h264: K&R formatting cosmetics for header files (part II/II)
h264: K&R formatting cosmetics for header files (part I/II)
rtmp: Implement check bandwidth notification.
rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player.
rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin.
rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream.
cmdutils: Add fallback case to switch in check_stream_specifier().
sctp: be consistent with socket option level
configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags.
vcr1enc: drop pointless empty encode_init() wrapper function
vcr1: drop pointless write-only AVCodecContext member from VCR1Context
vcr1: group encoder code together to save #ifdefs
vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments
mov: make one comment slightly more specific
lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX
lavfi: move audio-related functions to a separate file.
lavfi: remove some audio-related function from public API.
...
Conflicts:
cmdutils.c
libavcodec/h264.h
libavcodec/h264_mvpred.h
libavcodec/vcr1.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/defaults.c
libavfilter/internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Its useful to support the official decoder for comparission and debugging.
This reverts commit f9def9ccc6.
Conflicts:
Changelog
configure
libavcodec/allcodecs.c
libavcodec/libvorbis.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Code mostly inspired by vp8's MC, however:
- its MMX2 horizontal filter is worse because it can't take advantage of
the coefficient redundancy
- that same coefficient redundancy allows better code for non-SSSE3 versions
Benchmark (rounded to tens of unit):
V8x8 H8x8 2D8x8 V16x16 H16x16 2D16x16
C 445 358 985 1785 1559 3280
MMX* 219 271 478 714 929 1443
SSE2 131 158 294 425 515 892
SSSE3 120 122 248 387 390 763
End result is overall around a 15% speedup for SSSE3 version (on 6 sequences);
all loop filter functions now take around 55% of decoding time, while luma MC
dsp functions are around 6%, chroma ones are 1.3% and biweight around 2.3%.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The vertically interpolating variants of these functions read
ahead one line to optimise the loop. On the last line processed,
this might be outside the buffer. Fix these invalid reads by
processing the last line outside the loop.
Signed-off-by: Mans Rullgard <mans@mansr.com>
According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
* qatar/master: (27 commits)
libxvid: Give more suitable names to libxvid-related files.
libxvid: Separate libxvid encoder from libxvid rate control code.
jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
fate: cosmetics: lowercase some comments
fate: Give more consistent names to some RealVideo/RealAudio tests.
lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
lavfi: add extended_data to AVFilterBuffer.
lavc: check that extended_data is properly set in avcodec_encode_audio2().
lavc: pad last audio frame with silence when needed.
samplefmt: add a function for filling a buffer with silence.
samplefmt: add a function for copying audio samples.
lavr: do not try to copy to uninitialized output audio data.
lavr: make avresample_read() with NULL output discard samples.
fate: split idroq audio and video into separate tests
fate: improve dependencies
fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
fate: split some combined tests into separate audio and video tests
fate: fix dependencies for probe tests
mips: intreadwrite: fix inline asm for gcc 4.8
mips: intreadwrite: remove unnecessary inline asm
...
Conflicts:
cmdutils.h
configure
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffplay.c
libavcodec/internal.h
libavcodec/jpeglsdec.c
libavcodec/libschroedingerdec.c
libavcodec/libxvid.c
libavcodec/libxvid_rc.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersink.h
tests/Makefile
tests/fate/aac.mak
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/image.mak
tests/fate/libavutil.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/qt.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/vqf.mak
tests/ref/fate/ea-mad
tests/ref/fate/ea-tqi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Those functions are only useful inside filters. It is better to not
support user filters until the API is more stable.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
There's no reason for it to be explicitly 32 bits. It's declared as a
plain int in all other places in Libav.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
The additional parameters are just complicating the function interface.
Assume that a requested samples buffer will *always* have the format
specified in the requested link.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Remove AVFilterBufferRefAudioProps.size, and use nb_samples in its place
everywhere.
This is required as the size in the audio buffer may be aligned, so it
may not contain a well defined number of samples.
Also remove the useless planar parameter, which can be deduced from the
sample format.
This is technically an API and ABI break, but since the audio part of
lavfi is not usable now, this should not be a problem in practice.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This makes only tests actually using avconv depend on it.
The remaining tests already depend on what they need.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Just like gcc 4.6 and later on ARM, gcc 4.8 on MIPS generates
inefficient code when a known-unaligned location is used as a
memory input operand. This applies the same fix as has been
previously done to the ARM version of the code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
GCC actually handles unaligned accesses correctly in all cases
except, absurdly, 32-bit loads on mips64. The remaining asm is
thus not needed, and removing it results in better code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Using release_buffer and get_buffer as currently might
not prefer the previous frame contents which the
decoder relies on.
This leads to horrible playback in players using direct
rendering like MPlayer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
rtmp: Support 'rtmp_live', an option which specifies if the media is a live stream.
av_samples_fill_array: Mark unmodified function argument as const.
lagarith: add YUY2 decoding support
Support decoding unaligned rgb24 lagarith.
dv: Split profile handling code into a separate file.
flvenc: use AVFormatContext, not AVCodecContext for logging.
mov: Remove write-only variable in mov_read_chan().
fate: Change the probe-format refs to match the final text format committed.
fate: Add avprobe as a make dependency
Add probe fate tests to test for regressions in detecting media types.
fate: Add oneline comparison method
qdm2: clip array indices returned by qdm2_get_vlc().
avplay: properly close/reopen AVAudioResampleContext on channel layout change
avcodec: do not needlessly set packet size to 0 in avcodec_encode_audio2()
avcodec: for audio encoding, reset output packet when it is not valid
avcodec: refactor avcodec_encode_audio2() to merge common branches
avcodec: remove fallbacks for AVCodec.encode() in avcodec_encode_audio2()
Conflicts:
ffplay.c
libavcodec/Makefile
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/qdm2.c
libavcodec/utils.c
libavformat/flvenc.c
libavformat/mov.c
tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavcodec/utils.c:274: warning: passing argument 3 of ‘av_samples_fill_arrays’ discards qualifiers from pointer target type
./libavutil/samplefmt.h:151: note: expected ‘uint8_t *’ but argument is of type ‘const uint8_t *’
Unlike other variants, for YUY2 we need to use different prediction:
* on line 0 for luma we should left predict starting from the second pixel
* on line 1 we should left predict first 4 pixels for luma and 2 for chroma
* median prediction employed here is taken directly from HuffYUV
Also update libav->ffmpeg as theres pretty much no code left from libav.
The new code is faster, requires fewer mallocs and less memory. Its
also half the number of lines of code.
This code is not 100% identical in behavior to the previous, but the
differences appear to be rather limitations of the previous design
than intended though i could be wrong of course.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libschroedinger: Switch to function names more in line with Libav style.
Move code shared between libdirac and libschroedinger to libschroedinger.
lavfi: uninline avfilter_copy_buffer_ref_props().
lavf: add missing '*' in a doxy.
h264: Remove a commented-out function pointer typedef.
txd: Remove write-only variable in txd_decode_frame().
mmvideo.c: Remove unused variable in mm_decode_pal().
build: cosmetics: Add missing end-of-line backslashes to item lists.
build: cosmetics: Split HEADERS/OBJS/PROGS lists into one entry per line.
libschroedinger: Move a function to avoid a forward declaration.
pthread: warn on high thread counts
vf_yadif: fix missing error handling for avfilter_poll_frame()
avprobe: allow showing only one container/stream property.
lavfi: support audio in avfilter_copy_frame_props().
lavfi: avfilter_merge_formats: handle case where inputs are same
lavc: add sample rate and channel layout to AVFrame.
zerocodec: check if the previous frame is missing
doc: clarify check for NULL pointer style
Conflicts:
doc/APIchanges
doc/developer.texi
ffprobe.c
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/libdirac_libschro.c
libavcodec/libdirac_libschro.h
libavcodec/mmvideo.c
libavcodec/txd.c
libavcodec/version.h
libavcodec/zerocodec.c
libavfilter/Makefile
libavfilter/avfilter.c
libavfilter/version.h
libavformat/Makefile
libavutil/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prevents subsequent overreads when these numbers are used as indices
in arrays.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Fixes the Xing tag identification string to be "Info" for MP3 files with
constant bitrate. The previous "Xing" caused some decoders to recognize the
file as VBR.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a double-free crash if lists are the same due to the two
merge_ref() calls at the end of the (useless) merging that happens.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
ZeroCodec relies on the keyframe flag being set in the container, and
prev is the previously decoded frame. A keyframe flags incorrectly set
will lead to this condition.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master: (28 commits)
dfa: use more meaningful return codes
eatgv: check vector_bits
eatgv: check motion vectors
Mark a number of variables only used in av_dlog() calls as av_unused.
dvdec: drop const qualifier from variable to eliminate a warning
avcodec: Improve comment for thread_safe_callbacks to avoid misinterpretation.
tests/utils: don't ignore the return value of fwrite()
lavfi/formats: use sizeof(var) instead of sizeof(type).
lavfi: remove avfilter_default_config_input_link() declaration
lavfi: always enable the scale filter and depend on sws.
vf_split: support user-specifiable number of outputs.
avconv: remove stray useless comment.
mpegmux: add stuffing to avoid incomplete PCM frames
rtsp: avoid const warnings from strtol() call
avserver: check return value of ftruncate()
lagarith: make offset array type unsigned
dfa: add some checks to ensure that decoder won't write past frame end
aacps: NEON optimisations
aacps: align some arrays
aacps: move some loops to function pointers
...
Conflicts:
configure
doc/filters.texi
libavcodec/dfa.c
libavcodec/eatgv.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/vf_split.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Istvan Sebok provided a sample where field pair -> two fields content
was being misdetected by the existing logic. I added an additional
test to check the input picture type as identified by our h.264
parser.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
tests: Refactor rotozoom/videogen common code into a separate file.
tests: Mark some file-internal symbols as static.
build: Drop leftover .exp pattern from LIBSUFFIXES list.
vsrc_buffer: return EAGAIN if no frame is available.
WMAL: Shift output samples by the specified number of padding zeroes.
WMAL: Restore removed code in mclms_predict()
rtpdec_h264: Remove a useless ifdef
rtpdec_h264: Remove outdated/useless/incorrect comments
rtpdec_h264: Remove useless memory corruption checks
rtpdec_h264: Return proper error codes
rtpdec_h264: Check the available data length before reading
rtpdec_h264: Add input size checks
png: check bit depth for PAL8/Y400A pixel formats.
ea: check chunk_size for validity.
celp filters: Do not read earlier than the start of the 'out' vector.
Conflicts:
libavcodec/pngdec.c
libavfilter/src_buffer.c
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add/fix spacing, split long lines, align assignments where suitable.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Split long comments, move long comments at the end of lines to
separate lines above, fix vertical alignment, fix up comment style
(unify trailing dots - comments had a mix of 2, 3 or 4 dots, where
it would be just as good without them at all).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is worth keeping instead of removing, in case reading this
bit becomes necessary at some later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Wrong bit depth can lead to invalid rowsize values, which crashes the
decoder further down.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
configure: add POWER[5-7] support
arm: intreadwrite: revert 16-bit load asm to old version for gcc < 4.6
vqavideo: return error if image size is not a multiple of block size
cosmetics: indentation
avformat: only fill-in interpolated timestamps if duration is non-zero
avformat: remove a workaround for broken timestamps
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the following usages:
FFMPEG_DATADIR=presets ./ffmpeg -f lavfi -i testsrc=d=5 -vcodec libx264 -vpre ipod640 -f null -
FFMPEG_DATADIR=presets ./ffmpeg -f lavfi -i testsrc=d=5 -vpre libx264-ipod640 -f null -
The second example was broken even if documented.
They are now replaced with presets/ directory. WIN32 still seems to use
a ffpresets/ directory, but it doesn't look like to be deployed at
install time.
This adds support for png image2pipe streaming
Update to latest git by: Eugene Ware <eugene@noblesamurai.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Commit adebad0 "arm: intreadwrite: fix inline asm constraints for gcc
4.6 and later" caused some older gcc versions to miscompile code.
This reverts to the old version of the code for these compilers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The decoder assumes in various places that the image size
is a multiple of the block size, and there is no obvious
way to support odd sizes. Bailing out early if the header
specifies a bad size avoids various errors later on.
Fixes CVE-2012-0947.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
mpeg12: fixed parsing in some mpeg2 streams
Add SMPTE240M transfer characteristics flag.
mpegts: Some additional HDMV types and reg descriptors for mpegts
motionpixels: Clip YUV values after applying a gradient.
jpeg: handle progressive in second field of interlaced.
ituh263dec: Implement enough of Annex O (scalability) to fix a FPE.
h263: more strictly forbid frame size changes with frame-mt.
h264: additional protection against unsupported size/bitdepth changes.
tta: prevents overflows for 32bit integers in header.
configure: remove malloc_aligned.
vp8: update frame size changes on thread context switches.
snowdsp: explicitily state instruction size.
wmall: fix reconstructing audio with uncoded channels
WMAL cosmetics: fix indentation
gitignore: add Win32 library suffixes
Conflicts:
configure
libavcodec/h263dec.c
libavcodec/h264.c
libavcodec/ituh263dec.c
libavcodec/mjpegdec.c
libavcodec/wmalosslessdec.c
libavcodec/x86/snowdsp_mmx.c
libavformat/mpegts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Compared to av_opt_ptr, accessors bring:
- better performance (negligible);
- compile-time type check;
- link-time existence check
(or at worst, a dynamic linker error instead of a NULL dereference).
The option is related to the timecode, the new name clearly specifies the
context. Also it allows to list the option close to the other timecode
options.
* qatar/master:
arm: intreadwrite: disable inline asm for gcc 4.7 and later
arm: intreadwrite: fix inline asm constraints for gcc 4.6 and later
indeo3: fix motion vector validation
pcm_bluray: set bits_per_raw_sample for > 16-bit
twinvq: fix out of bounds array access
lavr: use 8.8 instead of 10.6 as the 16-bit fixed-point mixing coeff type
Conflicts:
doc/APIchanges
libavcodec/indeo3.c
libavcodec/pcm-mpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Progressive data is allocated later in decode_sof(), not allocating
that data leads to NULL dereferences.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This prevents sample_rate/data_length from going negative, which
caused various crashes and undefined behaviour further down.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This properly synchronizes frame size changes between threads if
subsequent threads abort decoding before frame size is initialized, i.e.
it prevents the thread after that from ping-ponging back to the original
value.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Starting with version 4.7, gcc properly supports unaligned
memory accesses on ARM. Not using the inline asm with these
compilers results in better code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
With a dereferenced type-cast pointer as memory operand, gcc 4.6
and later will sometimes copy the data to a temporary location,
the address of which is used as the operand value, if it thinks
the target address might be misaligned. Using a pointer to a
packed struct type instead does the right thing.
The 16-bit case is special since the ldrh instruction addressing
modes are limited compared to ldr. The "Uq" constraint produces a
memory reference suitable for an ldrsb instruction, which supports
the same addressing modes as ldrh. However, the restrictions appear
to apply only when the operand addresses a single byte. The memory
reference must thus be split into two operands each targeting one
byte. Finally, the "Uq" constraint is only available in ARM mode.
The Thumb-2 ldrh instruction supports most addressing modes so the
normal "m" constraint can be used there.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Manually remove that flag again for formats that read an arbitrary
amount of data and thus truncation is not an error.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avplay: use libavresample for sample format conversion and channel mixing
Fix compilation with YASM/NASM without AVX support.
WMAL: do not output last frame again if nothing was decoded in current packet
WMAL: do not start decoding if frame does not end in current packet
adpcm-thp: fix invalid array indexing
ppc: add const where needed in scalarproduct_int16_altivec()
ppc: remove shift parameter from scalarproduct_int16_altivec()
ppc: dsputil: do unaligned block accesses correctly
dvenc: do not call dsputil functions with stride not a multiple of 16
APIchanges: fill in some dates and commit hashes
Conflicts:
doc/APIchanges
ffplay.c
libavcodec/adpcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ModeTab.fmode has only 3 elements, so indexing it with ftype
in the initialier for 'size' is invalid when ftype == FT_PPC.
This fixes crashes with gcc 4.8.
Signed-off-by: Mans Rullgard <mans@mansr.com>
SDL only supports s16 sample format and a limited number of channel layouts.
Some versions of SDL on some systems support 4-channel and 6-channel output,
but it's safer overall to downmix any layout with more than 2 channels to
stereo.
The shift parameter was removed from this interface in 7e1ce6a.
This updates the Altivec implementation to match.
Signed-off-by: Mans Rullgard <mans@mansr.com>
To load unaligned vector data in the usual way, explicit vec_ld()
should be used rather than dereferencing a pointer to a vector type.
When the VSX extension is enabled, gcc may compile vector pointer
dereferences using the VSX lxvw4x instruction instead of the lvx
instruction typically used with Altivec/VMX. As the behaviour of
these instructions with unaligned addresses differs, it is important
that only lvx is used here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Allowing dsputil functions to assume the stride is a multiple of 16
even for smaller block sizes can simplify their implementation.
This appears to be the only place this guarantee is not met.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Currently it always exits with an error when more than
one position is specified.
Fixes trac issue #1266.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This is useful for marking progressive video processed by the filter as
interlaced, avoiding the interlaced flag to switch back and forth at each
frame.
This new mode is useful for generating frames for interlaced video
displays. Typically interlaced video displays have no form of field
synchronisation. This new mode guarantees correct field order without
any requirement for field synchronisation.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
* qatar/master:
mkv: mark corrupted packets and return them
mkv: forward EMBL block data error
avcodec: introduce YCoCg colorspace
avcodec: cosmetic cleanup on header
aac sbr: align struct member by 32 byte.
Conflicts:
libavcodec/avcodec.h
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Non perceptual color model that aims to have an increase effectiveness
in compression like the normal YCbCr while having near-lossless/lossless
mapping to RGB.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
4xm: fix invalid array indexing
rv34dsp: factorize a multiplication in the noround inverse transform
rv40: perform bitwise checks in loop filter
rv34: remove inline keyword from rv34_decode_block().
rv40: change a logical test into a bitwise one.
rv34: remove constant parameter
rv40: don't always do the full prev_type search
dsputil x86: revert a test back to its previous value
rv34dsp x86: implement MMX2 inverse transform
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is possible that just extending the RMMuxContext.streams
array would avoid it.
It is also possible that two audio streams will fail to mux
correctly as well, though at least it should not crash for
this reason.
I do not feel like checking either of these.
This patch fixes trac issue #1022 (at least it makes it
exit with a proper error message instead of crashing).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also fixes an (incorrect) "control reaches end of non-void function"
warning with some compilers.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
h264: new assembly version of get_cabac for x86_64 with PIC
h264: use one table instead of several for cabac functions
h264: (trivial) remove unneeded macro argument in x86/cabac.h
libschroedingerdec: check malloc
segment: reorder seg_write_header allocation
avio: make avio_close(NULL) a no-op
mov: Parse EC3SpecificBox (dec3 atom).
Conflicts:
libavcodec/cabac.c
libavcodec/x86/cabac.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds a hand-optimized assembly version for get_cabac much like the
existing one, but it works if the table offsets are RIP-relative.
Compared to the non-RIP-relative version this adds 2 lea instructions
and it needs one extra register.
There is a surprisingly large performance improvement over the c version (more
so than the generated assembly seems to suggest) just in get_cabac, I measured
roughly 40% faster for get_cabac on a K8. However, overall the difference is
not that big, I measured roughly 5% on a test clip on a K8 and a Core2.
Hopefully it still compiles on x86 32bit...
Now that only one table is used, there's some chance even darwin as compiles
this (apparently the label arithmetic used previously doesn't work if it
involves symbols defined in a different file, thanks to Ronald S. Bultje for
helping me with this).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The reason is this is easier for PIC code (in particular on darwin...).
Keep the old names as pointers (static in cabac_functions.h so gcc
knows these are just immediate offsets) so the c code can nicely stay the same
(alternatively could use offsets directly in the functions needing the
tables). This should produce the same code as before with non-pic and better
code (confirmed) with pic.
The assembly uses the new table but still won't work for PIC case.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
not used outside the cabac test functions (which probably means it's
a bad test if it doesn't use the same tables as the real functions?)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
this hunk was merged in 8b97ae64 and cbf767a8 although the check was there a
few lines above since cdced09e. I removed the first check to reduce the differences
to libav.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This adds a hand-optimized assembly version for get_cabac much like the
existing one, but it works if the table offsets are RIP-relative.
Compared to the non-RIP-relative version this adds 2 lea instructions
and it needs one extra register. get_cabac() gets about 40% faster, for
an overall speedup of about 5%.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The reason is this is easier for PIC code (in particular on darwin...).
Keep the old names as pointers (static in cabac_functions.h so gcc
knows these are just immediate offsets) so the c code can nicely stay the same
(alternatively could use offsets directly in the functions needing the
tables). This should produce the same code as before with non-pic and better
code (confirmed) with pic.
The assembly uses the new table but still won't work for PIC case.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
mpegts: Make sure we don't return uninitialized packets
gitignore: replace library catch-all pattern by more specific patterns
Conflicts:
.gitignore
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Skip to parse fields for additional independent substreams and its
associated dependent substreams since libavcodec's E-AC-3 decoder does not
support them yet.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Also add internal function ff_null_start_frame_keep_ref().
Fix crash when a following filter (e.g. settb) will unref the reference
passed by start_frame(), and then the reference is accessed in
end_frame() through inlink->cur_buf.
This fixes crashes, where the demuxer could return 0 even
if the returned AVPacket isn't initialized at all. This
could happen if running into EOF or running out of probesize
with non-seekable sources.
Signed-off-by: Martin Storsjö <martin@martin.st>
Ignoring all files that start with the name of a library matches some
files that are not generated. So replace libfoo/libfoo* with patterns
for static and shared libraries, pkg-config and version files.
Omitting the seconds has not worked for a long time, if ever.
Omitting the minutes too is just nonsensical for a duration
(it is indistinguishable from just seconds).
This way the user can specify how many or few tests should run while
still providing good coverage over the whole parameter set.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This disables the warning "external declaration in primary source file"
which is issued when a prototype for an extern function is found in a
.c file rather than a header file. We have such prototypes for asm
functions where a separate header file would be pointless.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The assembler may fail to place literal pools close enough to
instructions referencing them. An explicit .ltorg directive
fixes this.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This simplifies handling by removing a special case.
Its also needed to make the next change possible.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avconv: fix a segfault on -c copy with -filter_complex.
isom: Support more DTS codec identifiers.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Note, 1.3 is not finalized and the bitstream will still change
do not use it yet. This option is just to make playing with it
easier, otherwise one would have to edit the source
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
This should fix the FATE test on ARM (not tested),
but it should also detect alpha values like 2^128
reliably as invalid which would be another out-of-range
case with implementation-dependant behaviour.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The old code had two bugs:
For audio filters, the format was not set.
For video filters, if several links reference the same format list,
the same format must be selected in the end. This is done by
setting formats->format_count to 1: the other links sharing
the reference will therefore have only one choice.
If the heuristic does not pick the first format, the selected format
must also be moved to the first position.
* qatar/master:
matroska: Clear prev_pkt between seeks.
avutil: change default buffer size alignment for sample buffer functions
audemux: Add a sanity check for the number of channels
Remove libdirac decoder.
matroska: Add incremental parsing of clusters.
avconv: fix off by one check in complex_filter
mpegts: Try seeking back even for nonseekable protocols
swscale: K&R formatting cosmetics (part III)
Conflicts:
configure
doc/general.texi
doc/platform.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdirac.h
libavcodec/libdiracdec.c
libavformat/au.c
libavformat/mpegts.c
libswscale/input.c
tests/ref/seek/lavf_mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This also avoids an issue with parallel make in some
cases never building asynth-16000-1.sw.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
quant_mats valid range depends on the block size.
This fixes a global array overread.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
One rule can be used to generate all asynth files.
Requires renaming the mapchan files though.
Also switch to using the .wav variants for mapchan
while changing the name anyway, this allows getting rid
of the explicitly specified format.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The new incremental parser doesn't always clear prev_pkt,
however the packet queue is cleared when seeking. Which leads
to a use-after-free.
Verified using Valgrind.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mpegts demuxer reads 5 KB at startup just for discovering
the packet size. Since the default avio buffer size is 32 KB,
the seek back to the start will in most cases be within the
avio buffer, and will in most cases succeed even if the actual
protocol isn't seekable.
This makes the demuxer startup faster/with less data when
reading data from a non-seekable input, by not skipping
the first few KB.
If it fails, don't warn if the protocol isn't seekable, making
it behave as before in the failure case.
Signed-off-by: Martin Storsjö <martin@martin.st>
The new lowres support is limited to decoders where lowres decoding
is possible in high quality.
I was not able to measure any speed difference, but if one is found
the 2-3 lines that might affect speed can be made compile time conditional
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ARM: allow runtime masking of CPU features
dsputil: remove unused functions
mov: Treat keyframe indexes as 1-origin if starting at non-zero.
mov: Take stps entries into consideration also about key_off.
Remove lowres video decoding
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/libopenjpegdec.c
libavcodec/mjpegdec.c
libavcodec/mpegvideo.c
libavcodec/utils.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This ended up corrupting data structures and may possibly
lead to a double free.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a null ptr dereference with attachments
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Before this the context could become inconsistent, this lead to a null ptr
dereference.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows masking CPU features with the -cpuflags avconv option
which is useful for testing different optimisations without rebuilding.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
avcodec: remove AVCodecContext.dsp_mask
avconv: fix a segfault when default encoder for a format doesn't exist.
utvideo: general cosmetics
aac: Handle HE-AACv2 when sniffing a channel order.
movenc: Support high sample rates in isomedia formats by setting the sample rate field in stsd to 0.
xxan: Remove write-only variable in xan_decode_frame_type0().
ivi_common: Initialize a variable at declaration in ff_ivi_decode_blocks().
Conflicts:
ffmpeg.c
libavcodec/utvideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This removes all references to AVCodecContext.dsp_mask and marks
it for eviction at the next version bump. It has been superseded
by av_set_cpu_flag_mask() which, unlike this field, works everywhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fix warning:
libavfilter/vf_setfield.c: In function ‘init’:
libavfilter/vf_setfield.c:64:20: warning: too many arguments for format [-Wformat-extra-args]
swr_convert is not properly buffering packed input audio when the
output is not large enough, and when resampling is not actually needed
(same samplerate and no SWR_FLAG_RESAMPLE).
buf_set() is only handling the first channel and leaving the others as-is.
Sample program to reproduce the problem is here https://gist.github.com/2431768
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
General cosmetics, such as keeping lines under 80 characters,
fixing a couple of typos (predition -> prediction) and a
general style fix that was pointed out by Derek when I was having
my sliced multithreading patch in review by him.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This adds a hand-optimized assembly version for get_cabac much like the
existing one, but it works if the table offsets are RIP-relative.
Compared to the non-RIP-relative version this adds 2 lea instructions
and it needs one extra register.
There is a surprisingly large performance improvement over the c version (more
so than the generated assembly seems to suggest) just in get_cabac, I measured
roughly 40% faster for get_cabac on a K8. However, overall the difference is
not that big, I measured roughly 5% on a test clip on a K8 and a Core2.
Hopefully it still compiles on x86 32bit...
v2: incorporated feedback from Loren Merritt to avoid rip-relative movs
for every table, and got rid of unnecessary @GOTPCREL.
v3: apply similar fixes to the the decode_significance functions, and use
same macro arguments for non-pic case.
v4: prettify inline asm arguments, add a non-fast-cmov version (as I expect
the c code to be faster otherwise since both cmov and sbb suck hard on a
Prescott, even can't construct the mask with a 64bit shift as that's just as
terrible - it's quite difficult to find usable instructions on that chip...).
This is tested to work but not on a P4, in theory it _should_ be fast there.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avcodec: add a cook parser to get subpacket duration
FATE: allow lavf tests to alter input parameters
FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
FATE: replace the acodec-g726 test with 4 new encode/decode tests
FATE: replace current g722 encoding tests with an encode/decode test
FATE: add a pattern rule for generating asynth wav files
FATE: optionally write a WAVE header in audiogen
avutil: add audio fifo buffer
Conflicts:
doc/APIchanges
libavcodec/version.h
libavutil/avutil.h
tests/Makefile
tests/codec-regression.sh
tests/fate/voice.mak
tests/lavf-regression.sh
tests/ref/acodec/g722
tests/ref/acodec/g726
tests/ref/acodec/pcm_s24daud
tests/ref/lavf/dv_fmt
tests/ref/lavf/gxf
tests/ref/lavf/mxf
tests/ref/lavf/mxf_d10
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
* qatar/master:
dv: Initialize encoder tables during encoder init.
dv: Replace some magic numbers by the appropriate #define.
FATE: pass the decoded output format and audio source file to enc_dec_pcm
FATE: specify the input format when decoding in enc_dec_pcm()
x86inc: support AVX abstraction for 2-operand instructions
configure: detect PGI compiler and set suitable flags
avconv: check for an incompatible changing channel layout
avio: make AVIOContext.av_class pointer to const
nutdec: add malloc check and fix const to non-const conversion warnings
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Generating warnings when casting const away leads to tight constraints
on the code if one wants to avoid warnings. This is especially true for
generic code that is supposed to work with both const and non const.
These tight constrains cause people to waste time trying to find a
way to write code so it doesnt generate any warning, while people
should rather spend their time thinking on how to write fast,
clean, maintainable and bug free code.
Removing this class of warnings fixes this issue.
Approved-by: Nicolas George <nicolas.george@normalesup.org>
Approved-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Yasm was fixed in its r2161 and yasm 0.8.0 (Apr 2010) contained this fix.
Nasm was fixed in 2.06 (Jun 2009):
https://groups.google.com/group/alt.lang.asm/browse_thread/thread/fcc85bbc3745d893
I tested with yasm 0.7.99 and yasm 1.2.0.7, where this works fine.
I also tested with nasm. The nasm shipping with Xcode is too old to understand
ffmpeg's assembly, before and after the patch. Nasm 2.10 fails to compile
fft_mmx.asm on trunk with
libavcodec/x86/fft_mmx.asm:88: panic: section ".text" has already been specified with alignment 32, conflicts with new alignment of 16
but builds fine if I change the two alignment "16"s in x86inc.asm to "32". With this patch,
nasm 2.10 fails with
libavcodec/x86/fft_mmx.asm:39: panic: section ".rodata" has already been specified with alignment 32, conflicts with new alignment of 16
instead, but again builds fine with s/16/32/.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In normal picture decoding this does not need to be checked but as
error concealment is run in the case of errors the availability of
references is less certain. This may be fixed differently at some
point so that all references are always filled in before the EC
code, in which case this should then be changed to an assert()
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This will allow decoding to md5 and doing a diff comparison to a reference
checksum instead of a fuzzy stddev or oneoff comparison.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The output format is not always the same as the file extension,
which is sometimes required for correct probing. We can avoid
probing by specifying the format since it is already known.
The decoder can change the layout and channel count during decoding,
but currently we only validate that the two are compatible when opening
the codec. This checks for incompatibilities after each decoded frame.
* hexene/stagefright:
libstagefright: avoid memory leak
libstagefright: support more output pixel formats
libstagefright: avoid potential deadlock on output MediaBuffer
libstagefright: explicitly set positive timestamps as stagefright expects them so
Merge branches 'stagefright' and 'stagefright-test' into stagefright-test
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix this warning:
libavformat/aviobuf.c:663:20: warning: assignment discards qualifiers from pointer target type
Although this is a public header, it should remain source and
binary compatible.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ppc: drop unused function dct_quantize_altivec()
mpegaudiodec: Do not discard mp_decode_frame() return value.
matroska: do not set invalid default duration if frame rate is zero
mkv: use av_reduce instead of av_d2q for framerate estimation
mkv: report average framerate as minimal as well
avcodec_string: Favor AVCodecContext.codec over the default codec.
cook: Make constants passed to AV_BE2NE32C() unsigned to avoid signed overflow.
Conflicts:
libavcodec/cook.c
libavcodec/ppc/mpegvideo_altivec.c
libavcodec/utils.c
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Maintain an output queue of AVFrames instead of MediaBuffers
so that the latter can be released early. This avoids a potential deadlock
between the stagefright decoder::read() and Stagefright_decode_frame()
This fixes crashes with frame threads caused by inconsistent context parameters.
Fixes Ticket1207
Found-by: John Villamil
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If a video track specifies a zero frame rate (invalid but occurs),
this results in a division by zero and subsequent undefined conversion
to integer. Setting the default duration from the frame rate only
if the latter is greater than zero avoids such problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
aacenc: Fix issues with huge values of bit_rate.
dv_tablegen: Drop unnecessary av_unused attribute from dv_vlc_map_tableinit().
proresenc: multithreaded quantiser search
riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header
avconv: only set the "channels" option when it exists for the specified input format
avplay: update get_buffer to be inline with avconv
aacdec: More robust output configuration.
faac: Fix multi-channel ordering
faac: Add .channel_layouts
rtmp: Support 'rtmp_playpath', an option which overrides the stream identifier
rtmp: Support 'rtmp_app', an option which overrides the name of application
avutil: add better documentation for AVSampleFormat
Conflicts:
libavcodec/aac.h
libavcodec/aacdec.c
libavcodec/aacenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Do not pointlessly call ff_alloc_packet multiple times,
and fix an infinite loop by clamping the maximum
number of bits to target in the algorithm that does
not use lambda.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Other cases are not supported and lead to inconsistencies which
can lead to out of array writes.
Reported-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
indeo3: add parens around some macro arguments
h264: use proper PROLOGUE statement for a function using 8 registers.
doc: Update sample Vim config with suitable (function) indentation settings.
dv: Merge dvquant.h into dvdata.c where all other DV tables reside.
dv: Move static tables only used in one place to where they are used.
graphparser: set next to NULL on an entry extracted from inputs list
doc/filters: update documentation.
avconv: flush decoders immediately after an EOF.
avconv: send EOF to vsrc_buffer.
avconv: reindent.
Conflicts:
doc/filters.texi
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3b266da3d35f3f7a61258b78384dfe920d875d29':
avconv: add support for complex filtergraphs.
avconv: make filtergraphs global.
avconv: move filtered_frame from InputStream to OutputStream.
avconv: don't set output width/height directly from input value.
avconv: move resample_{width,height,pix_fmt} to InputStream.
avconv: remove a useless variable from OutputStream.
avconv: get output pixel format from lavfi.
graphparser: fix the order in which unlabeled input links are returned.
avconv: change {input,output}_{streams,files} into arrays of pointers.
avconv: don't pass input/output streams to some functions.
Conflicts:
cmdutils.c
cmdutils.h
doc/ffmpeg.texi
ffmpeg.c
ffplay.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Save the old output configuration (if it has been used
successfully) when trying a new configuration. If the new configuration
fails to decode, restore the last successful configuration.
Overwriting the av_malloc etc. functions is not easily
possible anymore, even for systems that support overriding
symbols in shared libraries in principle.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This replaces the matroskadec one with the same name.
The advantage is not only easier reuse in other demuxers
but also that we can make the decisions after the parser.
This fixes seeking in files that mark the keyframes incorrectly,
for example the file in track ticket #1003.
The matroska variable is still kept to be able to complain
about such broken files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This option is the stream identifier to play or to publish.
Sometimes the URL parser cannot determine the correct
playpath automatically, so it must be given explicitly
using this option (ie. -rtmp_playpath).
Signed-off-by: Martin Storsjö <martin@martin.st>
This option is the name of application to connect on the RTMP server.
Sometimes the URL parser cannot determine the app name automatically,
so it must be given explicitly using this option (ie. -rtmp_app).
Signed-off-by: Martin Storsjö <martin@martin.st>
The new values lead to error messages when used
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This way we don't require a clearly defined corresponding input stream.
The result for the xwd test changes because rgb24 is now chosen instead
of bgra.
Right now, they are arrays of structs, reallocated when new
streams/files are added. This makes storing pointers to those structs
harder than necessary.
Based on a patch by Robert Nagy <ronag89@gmail.com>.
It makes a difference when the error code is immediately cast
into a larger integer, such as an int64_t.
This prevents writing into a too small array if some parameters changed
without the tile being reallocated.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This prevents writing into a too small array if some parameters changed
without the tile being reallocated.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Instead of allocating over the original, free first. MOVStreamContext
is zero initialized so no double free will occur. Same style as other
fixes for the same problem in this file.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
vsrc_buffer: fix check from 7ae7c41.
libxvid: Reorder functions to avoid forward declarations; make functions static.
libxvid: drop some pointless dead code
wmal: vertical alignment cosmetics
wmal: Warn about missing bitstream splicing feature and ask for sample.
wmal: Skip seekable_frame_in_packet.
wmal: Drop unused variable num_possible_block_size.
avfiltergraph: make the AVFilterInOut alloc/free API public
graphparser: allow specifying sws flags in the graph description.
graphparser: fix the order of connecting unlabeled links.
graphparser: add avfilter_graph_parse2().
vsrc_buffer: allow using a NULL buffer to signal EOF.
swscale: handle last pixel if lines have an odd width.
qdm2: fix a dubious pointer cast
WMAL: Do not try to read rawpcm coefficients if bits is invalid
mov: Fix detecting there is no sync sample.
tiffdec: K&R cosmetics
avf: has_duration does not check the global one
dsputil: fix optimized emu_edge function on Win64.
Conflicts:
doc/APIchanges
libavcodec/libxvid_rc.c
libavcodec/libxvidff.c
libavcodec/tiff.c
libavcodec/wmalosslessdec.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/version.h
libavfilter/vsrc_buffer.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
0-sized packets are used to implement variable fps.
However there seems to be a variation where these are not
even stored in the main file but as 0-size index entries
only.
This fixes the sample in trac issue #957, it now plays both
the same ways as in MPlayer and in a way that looks correct.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
They will only cause us to skip writing the Xing header,
not cause any serious breakage.
Related to trac issue #1027.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
There is no point in storing the value in a variable, since it is not
used anywhere else in the decoder.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The samples_per_frame check is ported from wmaprodec.c
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is required for letting applications to create and destroy
AVFilterInOut structs in a convenient way.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Right now, e.g. scale,[in]overlay would connect scale to the first
overlay input and [in] to the second, which goes against the
documentation and is unintuitive.
The bug happens because of the ordering mess in curr_inputs variable:
1) the unlabeled links from the previous filter are added to it in
correct order
2) input labels are parsed and inserted to the beginning one by one
(i.e. in reverse order)
3) curr_inputs is matched against filter inputs in reverse order
Fix the problem by always using proper ordering without trying to be
clever.
Unlike avfilter_graph_parse(), it returns unlinked inputs and outputs
to the caller, which allows parsing of graphs where inputs/outputs are
not known in advance.
This reworks a loop to get rid of an ugly pointer cast,
fixing errors seen with the PathScale ENZO compiler.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
swscale: K&R formatting cosmetics (part II)
tiffdec: Add a malloc check and refactor another.
faxcompr: Check malloc results and unify return path
configure: escape colons in values written to config.fate
ac3dsp: call femms/emms at the end of float_to_fixed24() for 3DNow and SSE
matroska: Fix leaking memory allocated for laces.
pthread: Fix crash due to fctx->delaying not being cleared.
vp3: Assert on invalid filter_limit values.
h264: fix 10bit biweight functions after recent x86inc.asm fixes.
ffv1: Fix size mismatch in encode_line.
movenc: Remove a dead initialization
git-howto: Explain how to avoid Windows line endings in git checkouts.
build: Move all arch OBJS declarations into arch subdirectory Makefiles.
Conflicts:
configure
libavcodec/vp3.c
libavformat/matroskadec.c
libavutil/Makefile
libswscale/Makefile
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Recent register allocation changes (x86inc.asm update) changed the
register order and thus opcodes for the inner loops. One of them became
>128bytes, which confuses other parts of this function where it jumps
to fixed-offset positions to extend the edge by fixed amounts. A simple
register change fixes this.
This allows simd optimized routines to work in steps of 8 pixels
without going over the linesize. (this matters for yuv->rgb24 for example)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The fields in config.fate are colon-separated so any colons
within the fields should be escaped to prevent confusion.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Its bad to free things without zeroing them.
This fixes a potential issue when mov_read_close() would be called twice.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If mov_read_header exits under error, the memory allocated is
not freed.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Convert key_off initialize to use the same sc->keyframe_count as
used elsewhere in the function.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Audio timestamps are passed through by default and when the input
doesnt contain clean timestamps this can lead to non monotonicity
errors. (rounding to a course timebase can cause this too)
Print a warning when the errors in the timestamps are large
Fixes Ticket1167 (regression since timestamps are passed through)
This is a generic workaround that is intended to handle
slightly incorrect input files. It is very possible that some
demuxers contain bugs that lead to wrong timestamps, these demuxers
should of course still be fixed even if this change happens to
hide the issue.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Partially based on the port by Niel van der Westhuizen
<nielkie@gmail.com>, done for GCI 2010. Same output as the original
filter and as fast.
See thread:
Subject: [FFmpeg-devel] [PATCH] Port MPlayer 2xSaI filter to libavfilter
Date: Thu, 25 Nov 2010 01:31:24 +1000
All code should use the local variable, the
AVCodecContext might not yet have the updated value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Fixes potential out-of-bounds writes.
This is mostly possible when muxing ALS files where from
an extradata size of about 1050 put_bits would write data
outside the buffer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
For the FATE test sample used, this only avoids a warning
message.
However for other samples like al05_44.mp4 the converted
file can be played only after this fix.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This will only work for DSEs that are first in a packet, but
that is enough to fix handling of the reference files in
fate-suite/aac (though most of them still have other issues).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
If channel residues are have not been decoded from bitstream, they should be
initialized to 0 instead of using values from previous subframe.
This causes bursts of noise in silent parts of some files.
This patch fixes bug #1055
Reviewed-by: Benjamin Larsson <benjamin@southpole.se>
Reviewed-by: Mashiat Sarker Shakkhar <mashiat.sarker@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise for muxers like e.g. latmenc that never call
avio_flush (and do not have a write_trailer function)
a part of the data will always be missing.
Also update references for the voc muxer, which was also
buggy before and did not write out all data.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avplay: Don't free video filters string until the end of decoding.
movenc: small refactor mov_write_packet
movenc: remove redundant check
interplayvideo: fix av_dlog parameter type mismatch
Drop some pointless #ifdefs.
Conflicts:
libavcodec/interplayvideo.c
libavcodec/libxvidff.c
libavcodec/snowenc.c
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously it would just silently write out incorrect data.
This also fixes a potential integer overflow in the allocation.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
"Fix" in so far as at least it will no longer overread and possibly
crash and makes somewhat sense, but no idea whether there is anything
that can play the resulting files (FFmpeg can't).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Add support for all x86-64 registers
Prefer caller-saved register over callee-saved on WIN64
Support up to 15 function arguments
Also (by Ronald S. Bultje)
Fix up our asm to work with new x86inc.asm.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Currently they end up twice in the binary, since both
encoder and decoder include the header and thus each gets
their own copy.
This is clearly nonsense for the const tables, but shouldn't
be necessary for the RLTable either.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
av_freep()ing inside configure_video_filters() leaves a dangling
reference in the calling code, and the filter string is needed again when
reconfiguring video filters for a size change.
Using swr_flags instead of plain flags will avoid conflicts that
arise with plain flags and multiple libs (which all have AVOption flags)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Share the formerly internal write_packet with the hinter and move the
fragment flush logic to the user facing one since it is not concerned
about movtrack-only streams.
Fixes bug #263
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (22 commits)
rv40dsp x86: use only one register, for both increment and loop counter
rv40dsp: implement prescaled versions for biweight.
avconv: use default channel layouts when they are unknown
avconv: parse channel layout string
nutdec: K&R formatting cosmetics
vda: Signal 4 byte NAL headers to the decoder regardless of what's in the extradata
mem: Consistently return NULL for av_malloc(0)
vf_overlay: implement poll_frame()
vf_scale: support named constants for sws flags.
lavc doxy: add all installed headers to doxy groups.
lavc doxy: add avfft to the main lavc group.
lavc doxy: add remaining avcodec.h functions to a misc doxygen group.
lavc doxy: add AVPicture functions to a doxy group.
lavc doxy: add resampling functions to a doxy group.
lavc doxy: replace \ with /
lavc doxy: add encoding functions to a doxy group.
lavc doxy: add decoding functions to a doxy group.
lavc doxy: fix formatting of AV_PKT_DATA_{PARAM_CHANGE,H263_MB_INFO}
lavc doxy: add AVPacket-related stuff to a separate doxy group.
lavc doxy: add core functions/definitions to a doxy group.
...
Conflicts:
ffmpeg.c
libavcodec/avcodec.h
libavcodec/vda.c
libavcodec/x86/rv40dsp.asm
libavfilter/vf_scale.c
libavformat/nutdec.c
libavutil/mem.c
tests/ref/acodec/pcm_s24daud
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some files contain a few additional, all-0 bits.
Check for that case and don't print incorrect "not supported"
message.
Fixes trac issue #836.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Quite often, the original weights are multiple of 512. By prescaling them
by 1/512 when they are computed (once per frame), no intermediate shifting
is needed, and no prescaling on each call either.
The x86 code already used that trick.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().
Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
This commit is dedicated to the audiophiles who can hear it when a
needle is dropped on the moon.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Plain POSIX malloc(0) is allowed to return either NULL or a
non-NULL pointer. The calling code should be ready to handle
a NULL return as a correct return (instead of a failure) if the size
to allocate was 0 - this makes sure the condition is handled
in a consistent way across platforms.
This also avoids calling posix_memalign(&ptr, 32, 0) on OS X,
which returns an invalid pointer (a non-NULL pointer that causes
crashes when passed to av_free).
Abort in debug mode, to help track down issues related to
incorrect handling of this case.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avconv: use default alignment for audio buffer
avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame()
avutil: use align == 0 for default alignment in audio sample buffer functions
avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc()
avconv: remove OutputStream.picref.
avconv: only set SAR once on the decoded frame.
avcodec: validate the channel layout vs. channel count for decoders
audioconvert: make av_get_channel_layout accept composite names.
avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()
Conflicts:
doc/APIchanges
ffmpeg.c
libavcodec/utils.c
libavcodec/version.h
libavutil/audioconvert.c
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Decode output must be converted to rgb24 to avoid CRC difference
due to palette being stored in machine endianness.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signal that it can output a frame when there are frames on the main
input and EOF on the overlay input, but a frame is buffered -- e.g.
single picture overlay.
This will allow a workaround for cases where input timestamps are invalid or
when decoder delay of 1 packet or more confuses avconv into using the wrong
timestamps as a sync reference.
Since those are pseudo-palette formats, swscale does not write
into data[1], swscale must initialize the palette properly itself.
This lead to frames that actually decoded as all-gray before.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes that the GIF encoder crashes with it because
it has no palette.
And the arguments for the pseudopalette apply to gray8 as
much as to RGB8 etc.
In addition the changes required in lavfi should be needed anyway
when adding support for RGB8 etc.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We can't use whether the input format is paletted to decide that
the output format has a palette in data[1], too, that makes no sense.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Resolution changes are usually only used to scale with
network bandwidth, the (full) resolution specified in the
RM header really is authoritative.
While I am not sure this is the best solution, it is a
conservative approach that still should fix the most
common cases.
In particular, it fixes aspect with the sample from trac
issue #785 (in MPlayer, ffplay seems to just ignore
sample aspect changes in mid-playback).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
When decoding LATM, this function will not process extradata
but a different buffer.
It seems this was forgotten to update when LATM support
was added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
rtsp: Don't use av_malloc(0) if there are no streams
rtsp: Don't use uninitialized data if there are no streams
vaapi: mpeg2: fix slice_vertical_position calculation.
hwaccel: mpeg2: decode first field, if requested.
cosmetics: Fix indentation
rtsp: Don't expose the MS-RTSP RTX data stream to the caller
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes it easy to allow people to run tests that are disabled
(e.g. because they are broken) without having to hack Makefiles
by adding the test name to FATE_TESTS-no.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This uses the same code as in decode_video also in decode_audio.
Should fix valgrind FATE failures for nellymoser encode test.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
VASliceParameterBufferMPEG2.slice_vertical_position shall express
the slice vertical position from the original bitstream. The HW
decoder will correctly decode to the right line computed from the
appropriate top_field_first and is_first_field flags.
This patch aligns with DXVA's definition, which is what most HW and
drivers expect. In particular, Intel PowerVR (Cedarview et al.) and
NVIDIA (through VA-to-VDPAU layer). Since it looks more complex to fix
binary drivers, I aligned the Intel Gen driver (Sandy Bridge et al.)
to this behaviour, while maintaining compatibility with codec layers
not providing this patch yet.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
If user opted to present fields as they come, then the first field
picture needs to be submitted to the HW for decoding. In particular,
this fixes MPEG-2 decoding of interlaced streams.
Tested on Intel Cedar Trail, Sandy Bridge and Ivy Bridge platforms.
Someone reported on the ffmpeg-devel@ list this also works on DXVA
(Windows) and other Linux platforms (NVIDIA, through the VA wrapper).
This also means a similar patch to non-hwaccel VDPAU may be necessary.
Note: I believe the SLICE_FLAG_ALLOW_FIELD is useless since the first
field shall always be submitted to the HW anyway. Nobody uses HW accels
(dxva, vaapi, vdpau, etc.) without that flag though.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since we cannot specify decode parameters (and also because
it is better in principle) the 1-channel reference file
needs to be enabled, too.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Remove it from two places where it is useless, do not apply
it to the encode command and make it apply to the output
instead of the input of the decode command.
Should fix the dpx test.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
asfdec: Add an option for not searching for the packet markers
cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
cosmetics: Align codec declarations
cosmetics: Convert mimic.c to utf-8
avconv: remove an unused function parameter.
avconv: remove now pointless variables.
avconv: drop support for building without libavfilter.
nellymoserenc: fix crash due to memsetting the wrong area.
libavformat: Only require first packet to be known for audio/video streams
avplay: Don't try to scale timestamps if the tb isn't set
Conflicts:
Changelog
configure
ffmpeg.c
libavcodec/aacenc.c
libavcodec/bmpenc.c
libavcodec/dnxhddec.c
libavcodec/dnxhdenc.c
libavcodec/ffv1.c
libavcodec/flacenc.c
libavcodec/fraps.c
libavcodec/huffyuv.c
libavcodec/libopenjpegdec.c
libavcodec/mpeg12enc.c
libavcodec/mpeg4videodec.c
libavcodec/pamenc.c
libavcodec/pgssubdec.c
libavcodec/pngenc.c
libavcodec/qtrleenc.c
libavcodec/rawdec.c
libavcodec/sgienc.c
libavcodec/tiffenc.c
libavcodec/v210dec.c
libavcodec/wmv2dec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The format is slightly proprietary.
DVDs use a format of
code byte (0x00, 0x01, 0xfe or 0xff), two data bytes
MOV uses instead
cdat/cdt2 atom, two data bytes
Auto-detecting and supporting both in one decoder is trivial,
so a single codec ID is used.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The code is based on the remove-logo filter in MPlayer/libmpcodecs, by
Robert Edele, relicensed to LGPL with consent of the author.
Address trac issue #249.
The PSNR values are of varying usefulness, though at least
the DTS and AAC ones are useful with the right shift value.
Note: due to usage of floats some of these may fail on other
architectures.
In that case they should be converted into a CMD = stddev
FATE test, but it seems useful to try this way first.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Do not pointlessly call ff_alloc_packet2 multiple times,
and fix an infinite loop by clamping the maximum
number of bits to target in the algorithm that does
not use lambda.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
As the encoder contained the same bug and has similar structure
to anatoliys encoder, it is possibly based on it.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Searching for packet markers doesn't make sense for this use case,
where packets are fed one at a time to the demuxer.
This fixes playing back streams that have packets not starting
with the 0x82, 0x00, 0x00 marker.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also break some long lines, remove codec function placeholder comments
and add spaces in sample/pixel format lists.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the mandatory memcpy in vsrc_buffer has been eliminated, there
shouldn't be any significant reason to build without lavfi anymore.
This will make upcoming support for complex filtergraphs easier to do.
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
If get_filtered_video_frame failed above, tb might not be
initialized at all, so don't scale using it.
This fixes cases where avplay could crash if aborting
avformat_find_stream_info with ctrl+c.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
* adapt examples to new syntax
* mention that glob chars need to be enabled by a preceding % char
* note that globbing will be performed if both a printf and globbing
pattern would be possible judging from the input pattern
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
The correct point that seperates ISO and MAC language codes is 0x400
according to the current QT spec. Old QT specs did not list where this
seperation is but apparently only defined the meaning of the first 137.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
h264: Factorize declaration of mb_sizes array.
vsrc_buffer: when no frame is available, return an error instead of segfaulting.
configure: add dl to frei0r extralibs.
dsputil x86: use SSE float instruction instead of SSE2 integer equivalent
dsputil x86: remove deprecated parameter from scalarproduct_int16 prototype
vp8dsp x86: perform rounding shift with a single instruction
fate: add BMP tests.
swscale: handle complete dimensions for monoblack/white.
aacenc: Mark deinterleave_input_samples argument as const.
vf_unsharp: Mark readonly variable as const.
h264: fix 4:2:2 PCM-macroblocks decoding
Conflicts:
configure
libavcodec/h264.h
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_unsharp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The handler name is stored as a pascal string in the QT specs (first
byte is the length of the string), thus leading to an invalid metadata
string export.
Also add a second length check based on the first character to avoid
overwriting an already specified handler_name (it happens with Youtube
videos for instance, the handler_name get masked), or specifying an
empty string metadata.
To reproduce the problem, using ffprobe:
./ffprobe -show_packets -print_format compact -fflags +genpts -i
fate_samples/mxf/C0023S01.mxf
You will notice that the last video frame does not have it's PTS being
set, even with using genpts.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Allows avoiding the buffer when using avio read, write and seek functions.
When using the ffmpeg executable -avioflags direct can be used to enable
this mode for input files, but has no effect on output files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes the warning:
libavcodec/aacenc.c:524: warning: passing argument 2 of ‘deinterleave_input_samples’ discards qualifiers from pointer target type
pthread_cond_wait is supposed to return an integer,
and indeed does sometimes. Fix its function declaration
to match its behavior and POSIX.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
pthread_cond_wait is supposed to return an integer,
and indeed does sometimes. Fix its function declaration
to match its behavior and POSIX.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
We choose the first encoder by default and libaccplus has a
quite limited set of supported bitrates/sample rates.
Thus leading to failure by default in many cases when it is
enabled at compile time.
Moving it down means that the other aac encoders are favored
by default which avoids this issue.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The reason for this is that such files have IndexTableSegments which when parsed
cover EditUnit ranges like this:
[0,1)
[249,250)
[249,377)
[0,249)
where each interval is [IndexStartPosition,IndexStartPosition+IndexDuration).
This would be reduced to a sparse index like:
[0,1), [249,250)
instead of the full range:
[0,249), [249,377)
See TimeCode_HD.mxf, UMID =
060a2b340101010101010410130000000004001aa0e59175025b2a5600da4101.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The audio codecs property is composed by all values except
SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010) which are
unused.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some applications use these combinations and to maintain ABI
compatibility with previous versions we should not suddenly
fail. Thus only display a warning for the newly detected cases
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vsrc_buffer: allow buffering arbitrary number of frames.
vf_scale: avoid a pointless memcpy in no-op conversion.
avfiltergraph: try to reduce format conversions in filters.
avfiltergraph: add an AVClass to AVFilterGraph on next major bump.
id3v2: fix skipping extended header in id3v2.4
Conflicts:
libavfilter/vf_scale.c
libavfilter/vsrc_buffer.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Current code, with a filterchain such as
(input - yuv411) -> (scale - any) -> (sink - any)
will result in yuv420 being chosen for the second link, which is clearly
not right.
This commit attempts to improve in the following way:
repeat until convergence:
loop over all filters
find input link with exactly one format
force this format on all output links of the same type (if possible)
This fixes a segfault where a video decoder was called
from avcodec_decode_audio*().
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
apedec: check bits <= 32.
cavs: Remove unused code.
oggenc: fix condition when not to flush due to keyframe granule.
oggenc: add pagesize option to set preferred page size
libspeexdec: set frame size in libspeex_decode_init()
smacker audio: sign-extend the initial 16-bit predicted value
Conflicts:
libavcodec/apedec.c
libavcodec/libspeexdec.c
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The square is always passed as 1 whenever the function is called and
thus the if block never gets executed.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes crashes when copying a data track as in trac
issue #236.
No proper timecode tracks will be written though.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #522.
The issue is that the mov demuxer insists on using its
calculated sample_size (which is nonsense for old-style tracks)
instead of the one encoded in the track.
The old raw audio code should be using the value in stsz, because
the size of a single sample never makes sense for the size of
a full audio packet, whereas the new code will multiply the
sample size by the chunk size, so it should use the calculated value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #733.
The issue is that the size of the trak elements is coded
too large, so that the next trak element would be parsed
as part of the first and truncated incorrectly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
h264: drop ff_h264_ prefix from static function ff_h264_decode_rbsp_trailing()
h264: Make ff_h264_decode_end() static, it is not used externally.
output-example: K&R formatting cosmetics, comment spelling fixes
avf: make the example output the proper message
avf: fix audio writing in the output-example
mov: don't overwrite existing indexes.
lzw: fix potential integer overflow.
truemotion: forbid invalid VLC bitsizes and token values.
truemotion2: handle out-of-frame motion vectors through edge extension.
configure: Check for a different SDL function
Conflicts:
configure
doc/examples/muxing.c
libavcodec/truemotion2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This ensures that they dont contain invalid values.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mp3dec: perform I/S and M/S only when frame mode is joint stereo.
id3v2: add another mimetype for JPEG image
lzw: prevent buffer overreads.
WMAL: Remove inaccurate and unnecessary doxy
h264: fix cabac-on-stack after safe cabac reader.
truemotion2: convert packet header reading to bytestream2.
Conflicts:
libavcodec/lzw.c
libavcodec/truemotion2.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
tilde expansion should/can be done by the shell
Reviewed-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This changes globbing support to only be used if the character
contains at least one glob meta character that is preceded by
an unescaped %. To escape a literal % one would use %% which is
identical to the way to match a % with image2 sequence generation
feature.
* Makes it possible to have patterns like %04d-[720p].jpg work
again with sequence number generation. Previously this would
always be detected as a glob pattern and was interpreted by
the image2 glob code instead.
* Makes it possible to use %*-[720p].jpg to match above pattern
without having to double escape it to be not interpreted by most
shells and not by the image2 glob code (previously one would
need to use \*-\\\[720p\\\].jpg to achieve the same)
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
SHOW_UBITS() is only defined up to n_bits is 25, therefore forbid
values larger than this in get_vlc2() (max_bits). tokens[][] can be
used as an index in deltas[], which has a size of 64, so ensure the
values are smaller than that.
This prevents crashes on corrupt bitstreams.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This one is available both in SDL 1.2 and in 1.3 (which is the current
version available e.g. in macports), while 1.3 doesn't contain
SDL_Linked_Version().
The current check for SDL_Linked_Version() (available since SDL 1.2.13)
was added in 8f1b06c8, because including the headers for SDL_Init()
redirects the main() function, requiring the main function signature
to match the one of SDL_main (including argc/argv).
Signed-off-by: Martin Storsjö <martin@martin.st>
Looks like some LAME versions produce dual stereo mode MP3s with
flags for intensity and middle stereo set. In this mode those flags
should be ignored like the reference decoder and derived ones do.
* git://github.com/mjbshaw/FFmpeg-OpenJPEG-J2K-Encoder:
Fixes ticket 1127. I'm still looking into why bpp is getting set to 0.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
dr1 seems to work fine with frame size changes but many filters
cant handle it yet. Simply disabling it forces the alternative
non dr1 code path which has been tested more completely and
is known to handle frame size changes in a wider varity of
cases.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
asf: only set index_read if the index contained entries.
cabac: add overread protection to BRANCHLESS_GET_CABAC().
cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC().
cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE().
cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC().
h264: add overread protection to get_cabac_bypass_sign_x86().
h264: reindent get_cabac_bypass_sign_x86().
h264: use struct offsets in get_cabac_bypass_sign_x86().
h264: fix overreads in cabac reader.
wmall: fix seeking.
lagarith: fix buffer overreads.
dvdec: drop unnecessary dv_tablegen.h #include
build: fix doc generation errors in parallel builds
Replace memset(0) by zero initializations.
faandct: Remove FAAN_POSTSCALE define and related code.
dvenc: print allowed profiles if the video doesn't conform to any of them.
avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size.
FATE: add a test for vp8 with changing frame size.
fate: add kgv1 fate test.
oggdec: calculate correct timestamps in Ogg/FLAC
Conflicts:
libavcodec/4xm.c
libavcodec/cook.c
libavcodec/dvdata.c
libavcodec/dvdsubdec.c
libavcodec/lagarith.c
libavcodec/lagarithrac.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A call to decode_packet() does not always decode a complete WMA packet.
Moreover, this is not the correct place to document calls that are part
of the public API.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Also use correct buffer sizes in calls to tm2_read_stream(). Together,
this prevents overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The $(dir) function used to construct OBJDIRS includes a trailing slash
in the names returned, which GNU make 3.82 does not match to the
slash-less 'doc' in the documentation dependencies, causing parallel
build to fail. Adding a slash fixes this and still works with make
3.81.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
westwood_vqa: fix SND0 chunk handling
westwood_vqa: set video stream duration
raw: forward avpicture_fill() error code in raw_decode().
build: Do not explicitly add the doc directory to the OBJDIRS list.
dv: Split off DV video decoder into its own file.
build: fix RALF decoder standalone compilation, which depends on Golomb code
configure: Drop stray duplicate entry for --disable-fft from help output.
Conflicts:
libavcodec/dv.c
libavcodec/rawdec.c
libavformat/westwood_vqa.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is not entirely clear that whilst for width and height only an
expression needs to be provided, for interlace the option must
also be given.
It is also unclear that the default is non interlaced aware scaling.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
make av_interleaved_write_frame() flush packets when pkt is NULL
mpegts: Fix dead error checks
vc1: Do not read from array if index is invalid.
targa: convert to bytestream2.
rv34: set mb_num_left to 0 after finishing a frame
Conflicts:
libavcodec/targa.c
libavcodec/vc1data.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Now that a documentation generator is built in the doc directory,
this is no longer necessary. Fixes the Make warning:
Makefile:188: target `doc' given more than once in the same rule.
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
Also remove some write-only variables or write-only variable
assignments, remove internal colorspace conversion to native
endianness (that can be done by swscale much more efficiently),
and some cosmetics.
This reverts commit cc5dd632ce.
The change was redundant, it has been fixed long ago (422e3a7)
Conflicts:
libavcodec/rawdec.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Prevents running error resilience on a previous frame which will write
to the pic->mb_type[] array of the previous image. The array might
already be re-used for a new image in a subsequent thread, thus cause
two threads to write to the same pic->mb_type[] array, causing a race
condition which can crash in rv34_decode_cbp(), called by
rv34_decode_inter_mb_header() (which accesses mb_type[] twice,
assuming values are maintained, which the race condition breaks).
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This way it catches all cases, and prevents later segfaults.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
build: ppc: drop stray leftover backslash
build: Only clean the architecture subdirectory we build for.
build: drop some unnecessary dependencies from the H.264 parser
build: prettyprinting cosmetics
libavutil: Remove pointless rational test program.
libavutil: Remove broken and pointless lzo test program.
lavf doxy: expand AVStream.codec doxy.
lavf doxy: improve AVStream.time_base doxy.
lavf doxy: add some basic documentation about reading from the demuxer.
lavf doxy: document passing options to demuxers.
lavf doxy: clarify that an AVPacket contains encoded data.
mpegtsenc: allow user triggered PES packet flushing
APIchanges: mark the place where 0.7 was cut.
APIchanges: mark the place where 0.8 was cut.
APIchanges: fill in missing dates and hashes.
smacker: convert palette and header reading to bytestream2.
alac: convert extradata reading to bytestream2.
Conflicts:
doc/APIchanges
libavcodec/smacker.c
libavcodec/x86/Makefile
libavfilter/Makefile
libavutil/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This depends on the proposed parser change for 0-size packets
in previous mail, otherwise video now plays far too fast.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Currently, the duration of those packets is just discarded
when enabling parsing, thus the output of the Metal Gear Solid
demuxer breaks completely when just setting AVSTREAM_PARSE_HEADERS.
The result will not be correct if a parser creates a delay even
with PARSER_FLAG_COMPLETE_FRAMES and there might be other cases
where it does not work correct, but just discarding them as it
is done currently seems worse.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
With the flag in place, it's hard to actually use the decoder, and
I'm happy with how it works, with the exception of DivX3 where I've
never found a sample that worked that I was confident actually
matched what the hardware claimed to support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This might even have prevented the compiler from some optimizations,
since both signed and unsigned types are used for the dezigzag tables/
table pointers, and if a branches uses both the compiler needs to
create more complex code.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
See trac issue #217.
Only the dsf field seems to be used to distinguish between PAL and NTSC.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avc: Add a function for converting mp4 style extradata to annex b
pthread: free progress if buffer allocation failed.
lavc/avconv: support changing frame sizes in codecs with frame mt.
libavformat: Document who sets the AVStream.id field
utvideo: mark output picture as keyframe.
sunrast: Add support for negative linesize.
vp8: fix update_lf_deltas in libavcodec/vp8.c
ralf: read Huffman code lengths without GetBitContext
Conflicts:
ffmpeg.c
libavcodec/sunrastenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This isn't exactly equivalent with the earlier code for codecs
other than H264 and VC1, but those are two only codecs supported
by this codepath anyway, and it simplifies it a bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
Before this, they were only added to the delayed release queue and not
freed until later. This could lead to unnecessary memory use or buffer
exhaustion.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This change should make no binary difference to the generated code.
the API version is just bumped for correctness sake, this is not
really a API or ABI change.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This prevents decoding happening on a half initialized context.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This prevents some variables from being changed in case of a
rejected resolution change.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
lf_delta.ref[i] and lf_delta.mode[i] were incorrectly reset to 0 if
specific delta value was not updated. Fixed to keep the previous value
if flag indicates that element in question is not updated.
Signed-off-by: Janne Salonen <jsalonen@google.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
rv34: error out on size changes with frame threading
aacsbr: Add a debug check to sbr_mapping.
aac: Reset some state variables when turning SBR off
aac: Reset PS parameters on header decode failure.
fate: add wmalossless test.
aacsbr: handle m_max values smaller than 4.
Conflicts:
libavcodec/aacsbr.c
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes following strange output:
DEV D libopenjpeg OpenJPEG based JPEG 2000 encoder
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Our decoder does not support changing w/h.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes race conditions that ultimately lead to memory corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure the reset flag gets set when SBR gets turned back on
and sets control variables for unguided mode back to their defaults.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
If the next header frame codes zero envelopes the previous frame's
values will be used. Consequently the invalid values must be cleared.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
rv34: Handle only complete frames in frame-mt.
MPV: set reference frame pointers to NULL when allocation of dummy pictures fails
configure: die if x11grab dependencies are unavailable
zerocodec: factorize loop
avconv: fix the resampling safety factors for output audio buffer allocation
avconv: move audio output buffer allocation to a separate function
avconv: make the async buffer global and free it in exit_program()
Conflicts:
ffmpeg.c
libavcodec/mpegvideo.c
libavcodec/rv34.c
libavcodec/zerocodec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prevents a signflip in the counter, and a subsequent crash because of
overreads/overwrites.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
They were moved into code under HAVE_YASM and most of them
even into completely disabled code with no reason given
for that in the commit message.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Correct handling of errors to prevent hags or crashes is very complex
otherwise.
The frame initializing is also moved from decode_slice() to
decode_frame() for clarity.
This prevents a null ptr dereference.
It could be checked differently but this way it should
be possible to return some data.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
xwma: Validate channels and bits_per_coded_sample.
mov: Do not read past the end of the ctts_data table.
mov: Add missing terminator to mov_ch_layout_map_1ch.
asf: reset side data elements on packet copy.
wmavoice: fix stack overread.
wmalossless: error out if a subframe is not used by any channel.
vqa: check palette chunk size before reading data.
wmalossless: reset sample pointer for each subframe.
wmalossless: error out on invalid values for order.
Conflicts:
libavcodec/vqavideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: Add ZeroCodec test
oggparseogm: fix order of arguments of avpriv_set_pts_info().
pngenc: better upper bound for encoded frame size.
aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C
aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF
aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C
aiffdec: do not set bit rate if block duration is unknown
wmall: output packet only if we have decoded some samples
Conflicts:
libavcodec/pngenc.c
tests/fate/lossless-video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allows working around issue #605.
Note: as a side effect this fixes that -vsync drop
as far as I could tell would not drop pts/dts values
when duplicating frames or when flushing encoder delay.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
It is currently only handled in the parser code for WMV/ASF style
header, but not the one used in the bytestream format used when
muxed into MPEG-TS as on e.g. BluRay.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The new API allows (optionally and on by default) using a internal buffer to encode, avoiding
the need to allocate large buffers or risking failure on too small buffers.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
ffplay: use frame count based queueing for audio queue
ffplay: reset audio_pkt_temp when opening audio
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is even potentially faster in this use-case.
Should fix AAC SBR decoding on machines with SSE but not
SSE2, fixing track issue #1041.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This new API stores useful data in a dedicated structure
and has clearly delimited init functions.
Hopefully, uses of the old API can be replaced quickly.
This is how it is defined in Amiga Developer CD from year 1992 and
is consistent with files created with ADPro.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
ISC doesn't contain this line, so remove it to
prevent confusion.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reduces the number of queued frames for audio data but also reduces the
amount of A-V difference after changing the audio stream (because less frames
are queued). Fixes bug #1035.
Signed-off-by: Marton Balint <cus@passwd.hu>
Otherwise we may use the remaining data of the last packet from the previous
audio stream. Fixes bug #951.
Signed-off-by: Marton Balint <cus@passwd.hu>
This change avoids accessing the segment map of the previous frame if
segmentation is not enabled for the current frame. The caller of
decode_mb_mode() only calls ff_thread_await_progress() on the reference
segmentation index array if segmentation is enabled, so Chromium's TSAN
will report a race when accessing this data while segmentation is not
enabled.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: make compare() function compatible with POSIX bc
Update Janne's email address.
APIchanges: Replace Subversion revision numbers by Git hashes.
bytestream: Eliminate one level of pointless macro indirection.
xwd: convert to bytestream2.
vqavideo: port to bytestream2 API
Read preset files with suffix .avpreset
prores: allow user to set fixed quantiser
lavf: remove some disabled code.
lavf: only set average frame rate for video.
lavf: remove a pointless check.
avcodec: add XBM encoder
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vqavideo.c
libavformat/img2enc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We can't do this in general since we could be reading a file with B-frames while
lacking an index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
resample: allocate a large enough output buffer
fate: fix enc_dec_pcm tests with remote target
wmaenc: remove bit-exact hack
FATE: remove WMA acodec tests
FATE: add WMAv1 and WMAv2 encode/decode tests with fuzzy comparison
FATE: add AC-3 and E-AC-3 encode/decode tests with fuzzy comparison
qtrle: Use bytestream2 functions to prevent buffer overreads.
vqavideo: check malloc return values
x11grab: fix a memory leak exposed by valgrind
threads: fix old frames returned after avcodec_flush_buffers()
MPV: always mark dummy frames as reference
h264: fix deadlocks on incomplete reference frame decoding.
mpeg4: report frame decoding completion at ff_MPV_frame_end().
mimic: don't use self as reference, and report completion at end of decode().
Conflicts:
libavcodec/h264.c
libavcodec/qtrle.c
libavcodec/resample.c
libavcodec/vqavideo.c
libavdevice/x11grab.c
tests/ref/seek/wmav1_asf
tests/ref/seek/wmav2_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It may have improved cross-platform stability, but wasn't the only place in
the encoder with bitexact issues. It is no longer needed because we have FATE
tests for float encoders using fuzzy comparison.
In non-blocking mode, lowest-level read protocols are
supposed block only for a short amount of time to let
retry_transfer_wrapper() check for interrupts.
Also, checking the interrupt_callback in the receiving thread is
wrong, as interrupt_callback is not guaranteed to be thread-safe
and the job is already done by retry_transfer_wrapper(). The error
code was also incorrect.
Bug reported by Andrey Utkin.
Also add bbox.h and bbox.c files, based on the remove-logo filter by
Robert Edele. These files are useful for sharing code with the pending
removelogo port.
* qatar/master:
h264: K&R formatting cosmetics
s3tc.h: Add missing #include to fix standalone header compilation.
FATE: add capability for audio encode/decode tests with fuzzy psnr comparison
FATE: allow a tolerance in the size comparison in do_tiny_psnr()
FATE: use absolute difference from a target value in do_tiny_psnr()
FATE: allow tests to set CMP_SHIFT to pass to tiny_psnr
FATE: use $fuzz directly in do_tiny_psnr() instead of passing it around
Conflicts:
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When using "-f x11grab -i :0.0" valgrind reports a definitely lost
memory block with this message:
==31544== 5 bytes in 1 blocks are definitely lost in loss record 1 of 2
==31544== at 0x4026E68: memalign (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so)
==31544== by 0x4026F17: posix_memalign (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so)
==31544== by 0x60D399A: av_malloc (in /usr/lib/x86_64-linux-gnu/libavutil.so.51.22.1)
==31544== by 0x60D3A70: av_strdup (in /usr/lib/x86_64-linux-gnu/libavutil.so.51.22.1)
==31544== by 0x4A2BE58: ??? (in /usr/lib/x86_64-linux-gnu/libavdevice.so.53.2.0)
==31544== by 0x506D29E: avformat_open_input (in /usr/lib/x86_64-linux-gnu/libavformat.so.53.21.0)
==31544== by 0x400A80: main (in /home/ao2/WIP/am7xxx-play/tests/a.out)
The 5 bytes lost are the ones from param = av_strdup(":0.0"), so let's
free param in the exit path.
Also check the av_strdup() return value.
Note: calling av_free(param) even when av_strdup() fails and param is
NULL is OK and keeps the code simpler without adding another label to
skip av_free().
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Calling avcodec_flush_buffers() and then avcodec_decode_video2() with
a 0-sized packet (to get remaining buffered frames) could incorrectly
return an old frame from before the avcodec_flush_buffers() call. Add
a loop in ff_thread_flush() to zero the got_frame field of each thread
to ensure the old frames will not be returned.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If the dummy frame are not created from a reference frame they could
be deleted untimely resulting in multithreaded decoder waiting on
the current frame to finish.
Noticed by Ronald S. Bultje in the RV34 decoder with a broken file.
If decoding a second complementary field, and the first was
decoded in our thread, mark decoding of that field as complete.
If decoding fails, mark the decoded field/frame as complete.
Do not allow switching between field modes or field/frame mode
between slices within the same field/frame. Ensure that two
subsequent fields cover top/bottom (rather than top/frame,
bottom/frame or such nonsense situations).
Fixes various deadlocks when decoding samples with errors in
reference frames.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Wolfram Gloger <wmglo@dent.med.uni-muenchen.de>
Modify the parser initialization so that parsers can
set pict_type themselves. Use this in the mpegvideo parser
so that initial frames are not unconditionally I frames.
I have had this in my tree for several years.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dxa: remove useless code
lavf: don't select an attached picture as default stream for seeking.
avconv: remove pointless checks.
avconv: check for get_filtered_frame() failure.
avconv: remove a pointless check.
swscale: convert hscale() to use named arguments.
x86inc: add *mp named argument support to DEFINE_ARGS.
swscale: convert hscale to cpuflags().
Conflicts:
ffmpeg.c
libswscale/x86/scale.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows for testing floating-point audio encoders across different
platforms where exact comparisons are unreliable due to float rounding
differences.
Note, this doesnt break compatibility with libav, as libav
has implemented a incompatible and more limited system under the same
-cpuflags command line option we used since some time.
The differences to libav for example are we can do things like
ffmpeg -cpuflags -sse+mmx -cpuflags +3dnow
Its also possible in our system to force flags that have not been
detected as available
And our -cpuflags works with all tools not just 1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids problems
where avio_tell() returns 0. I've updated all the checks against
cluster_pos
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Progressive images can have only 16 references, error out if there are
more, since the data is almost certainly corrupt, and the invalid value
will lead to random crashes or invalid writes later on.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Interlaced images can have 32 references (16 per field), so limiting the
array size to 16 leads to invalid writes.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The safe bitstream reader broke it since the buffer size was specified
in bytes instead of bits.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
CC: libav-stable@libav.org
* qatar/master:
doc/general: update supported devices table.
doc/general: add missing @tab to codecs table.
h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
avconv: reindent
avconv: link '-passlogfile' option to libx264 'stats' AVOption.
libx264: add 'stats' private option for setting 2pass stats filename.
libx264: fix help text for slice-max-size option.
http: Clear the auth state on redirects
http: Retry auth if it failed due to being stale
rtsp: Resend new keepalive commands if they used stale auth
rtsp: Retry authentication if failed due to being stale
httpauth: Parse the stale field in digest auth
dxva2_vc1: pass the overlap flag to the decoder
dxva2_vc1: fix decoding of BI frames
FATE: add shorthand to wavpack test
dfa: convert to bytestream2 API
anm decoder: move buffer allocation from decode_init() to decode_frame()
h264: improve parsing of broken AVC SPS
Conflicts:
ffmpeg.c
libavcodec/anm.c
libavcodec/dfa.c
libavcodec/h264.c
libavcodec/h264_direct.c
libavcodec/h264_ps.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This bug might have been exploitable (out of HEAP buffer writes)
Bug introduced by libav
commit a55d5bdc6e
Date: Tue Mar 6 15:15:42 2012 -0800
algmm: convert to bytestream2 API.
Allow up to 4 retries for normal requests, where both the
proxy and the target server might need to authenticate.
Signed-off-by: Martin Storsjö <martin@martin.st>
These commands are sent asynchronously, not waiting for the reply.
This reply is parsed later by ff_rtsp_tcp_read_packet or
udp_read_packet. If the reply indicates that we used stale
authentication and need to use a new nonce, resend a new keepalive
command immediately.
This is the only request sent asynchronously, so currently there's
no other command that needs to be resent in the same way.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
pcm-mpeg: convert to bytestream2 API
Revert "h264: clear trailing bits in partially parsed NAL units"
remove iwmmxt optimizations
mimic: do not continue if swap_buf_size is 0
mimic: convert to bytestream2 API
frwu: use MKTAG to check marker instead of AV_RL32
txd: port to bytestream2 API
c93: convert to bytestream2 API
iff: make .long_name more descriptive
FATE: add test for cdxl demuxer
rtsp: Fix a typo
Conflicts:
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/dsputil_iwmmxt_rnd_template.c
libavcodec/arm/mpegvideo_iwmmxt.c
libavcodec/c93.c
libavcodec/txd.c
libavutil/arm/cpu.c
tests/fate/demux.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Parsing the entire NAL as SPS fixes decoding of some AVC bitstreams
with broken escaping. Since the size of the NAL unit is known and
checked against the buffer end we can parse it entirely without buffer
overreads.
Fixes playback of
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This reverts commit 729ebb2f18.
There was an off-by-one error in the bit mask calculation clearing
actually the last valid bit and causing
http://bugzilla.libav.org/show_bug.cgi?id=227
The broken sample (Mr_MrsSmith-h264_aac.mp4) the commit was fixing
does not work after correcting the off-by-one error.
CC: libav-stable@libav.org
The were broken since August of 2010 without anyone noticing until
three weeks ago. Nobody cares about it anymore and hopefully Marvell
will support NEON like in the PXA978 from now on.
Yasm creates an implicit unaligned text section if "struc" is used
outside of any section:
http://tortall.lighthouseapp.com/projects/78676-yasm/tickets/247
Since yasm only honors the "align" annotation on the first declaration
of a section, this implicit text section causes all text section
alignments to be ignored. Also fixes a yasm warning about it agnoring
alignment.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
According to video_file_format_spec_v10_1.pdf flv stores AAC RAW
thanks to Baptiste Coudurier for pointing that out
thanks to Aℓex Converse for explaining:
This can't be at the start of a non-ADTS payload. 111 is the
EndOfFrame syntax element.
Together these proof that the check was correctly rejecting ADTS which
is not supposed to be in flv. Many players are able to play such ADTS
in flv though but its better if we conform to the spec as this should
ensure that not many but all players can play files generated by ffmpeg.
This reverts commit 3c9a86df0e.
* qatar/master:
cook: expand dither_tab[], and make sure indexes into it don't overflow.
xxan: reindent xan_unpack_luma().
xxan: protect against chroma LUT overreads.
xxan: convert to bytestream2 API.
xxan: don't read before start of buffer in av_memcpy_backptr().
vp8: convert mbedge loopfilter x86 assembly to use named arguments.
vp8: convert inner loopfilter x86 assembly to use named arguments.
Conflicts:
libavcodec/xxan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
MPC8 allows indices of mpc_CC up to -1, and mpc_SCF up to -6, thus pad
the tables by that much on the left end.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
mpjpeg video streamings would break and stop on Firefox after 1 - 30
seconds.
In order to fix this, two changes were made:
1. Replaced all occurrences of '\n' character in mjpeg metadata
with occurences of "\r\n".
2. Added "Content-length: <packet-size>" metadata entry for each
sent frame.
The change has been tested on Google Chrome 17.0.963.78 and Firefox 10.0.2
on lubuntu 11.10 and the streaming seems to work fine now.
The two samples both have stype 0.
Without this extra check, the code breaks 4:2:2 dvsd
(stype 4), since that has the same resolution.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
Fix a bunch of common typos.
build: Skip compiling xvmc.h under the correct condition.
configure: darwin: Change dylib install names to include major version.
mpegts: Always honor a registration descriptor if present and there is no other codec information.
aacdec: Fix SCE parity check.
aacdec: Fix out of array writes (stack).
rtsp: Only set the ttl parameter if the server actually gave a value
udp: Set ttl for read-write streams, too, not only for write-only ones
udp: Only bind to the multicast address if in read-only mode
udp: Clarify the comment about binding the multicast address
udp: Reorder comments
Conflicts:
libavcodec/aacdec.c
tools/patcheck
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This will cause linkers to link against the major lib names, instead of the
base names, allowing multiple major versions of the libraries to co-exist.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
An unpaired SCE preceding a CPE only makes sense for front SCEs
preceding the first CPE.
Split from FFmpeg commit a8d67efa53
Signed-off-by: Alex Converse <alex.converse@gmail.com>
Set the element to channel vector (e2c_vec) size to be the maximum
number of aac channel elements. This makes it slightly larger than it
needs to be because CCEs are never mapped to output channel locations.
Also add a check that all input tags (legal or not) will fit.
Split from FFmpeg commit a8d67efa53
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This fixes sending back RTCP RR packets if receiving RTP over
multicast.
If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.
This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes trac #1045.
Thanks to Peter Ross for his help with this patch.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Matroska demuxer needs to recreate tta header, so just display
crc error without aborting.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes some invalid memory access caused later in the function
by res_chan[] not being set for all channels. This happens when a
channel doesn't appear a submap. This change simply returns a
decoder error when this situation is detected.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The ogg decoder wasn't padding the input buffer with the appropriate
FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in
various pieces of parsing code when they thought they had more data than
they actually did.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
We slightly overread the input buffer, so we require
padding at the end of the buffer, as is documented in the
get_bits API. Without padding, we'll read uninitialized
data or beyond the end of the .rodata, which may crash.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Once fixed, the end_frame function does exactly what
avfilter_default_end_frame does; therefore, end_frame
can be removed to let avfilter_default_end_frame work.
Fixes ticket #1038.
* qatar/master: (29 commits)
sbrdsp.asm: convert all instructions to float/SSE ones.
dv: cosmetics.
dv: check buffer size before reading profile.
Revert "AAC SBR: group some writes."
udp: Print an error message if bind fails
cook: extend channel uncoupling tables so the full bit range is covered.
roqvideo: cosmetics.
roqvideo: convert to bytestream2 API.
dca: don't use av_clip_uintp2().
wmall: fix build with -DDEBUG enabled.
smc: port to bytestream2 API.
AAC SBR: group some writes.
dsputil: remove shift parameter from scalarproduct_int16
SBR DSP: unroll sum_square
rv34: remove dead code in intra availability check
rv34: clean a bit availability checks.
v4l2: update documentation
tgq: convert to bytestream2 API.
parser: remove forward declaration of MpegEncContext
dca: prevent accessing static arrays with invalid indexes.
...
Conflicts:
doc/indevs.texi
libavcodec/Makefile
libavcodec/dca.c
libavcodec/dvdata.c
libavcodec/eatgq.c
libavcodec/mmvideo.c
libavcodec/roqvideodec.c
libavcodec/smc.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The codec would keep returning the last decoded frame if the stream
contains B-frames, since it wouldn't clear that frame from the list of
frames to be returned to the user.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The "ECs != 1 -> OP1a" assumption was wrong. Luckily, the file that triggered
that behavior had two ECs, not zero. Hence distinguishing between them is
simple in this case.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes rare cases where OPAtom may be treated as OP1a, causing all essence
to be read into RAM.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since the values are floats, using the float operations
makes sense, improves performance on some CPUs and
makes the code SSE compatible instead of needing SSE2.
Based on suggestion by Jason.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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>
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.
Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
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.
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.
Also move the function higher in the file, since it will be called from
read_frame_internal().
The feedback factors for the timefilter are directly computed from
the expected period. This commit changes the init function to accept
the period itself and compute the feedback factors internally,
rather than having all client code duplicate the formulas.
This commit also actually fixes the formulas: the current code had
sqrt(2*o), but the correct formula, both theoretically and according
to experimental testing, is sqrt(2)*o.
Furthermore, it adds an exponential to feedback factors larger than
1 with large periods.
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.
This fixes issues when the bitrate is variable or inaccurate but the
frame size has not been determined yet.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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
commit 13f6917ca9 handles discards automatically,
but the ffm discard options are not fully parsed. Causing the input streams not
to be used, so no stream towards the ffserver after the initial probing.
Signed-off-by: Rick van der Zwet <info@rickvanderzwet.nl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
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>
This allows it to be used with get_bits without the thread of overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* 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>
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
This fixes duplicate timestamps on mp2 in ts with non seekable input.
It also fixed the fate pva demux timestamps.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes"
come first, "Frames" behind.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
* 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>
This makes the packetization spec compliant for cases where one single
GOB doesn't fit into an RTP packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
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.
This fixes cases where the user had specified one desired MTU
via an option, and the protocol indicates another one.
Signed-off-by: Martin Storsjö <martin@martin.st>
Frame sizes in ID3v2.3 are not synchsafe, they are simply 32be numbers.
In practice this bug is not noticeable unless the frame size takes more
than 7 bits (which is almost never for text frames).
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>
Seeking back on EOF will reset the EOF flag, causing us to re-enter
the loop to find the next marker in the ASF file, thus potentially
causing an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* 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>
The properties of the CDCI Descriptor are insufficient to specify
the pixel format for uncompressed picture data. SMPTE 377-1 and
RP224v10 have defined a set of picture coding labels to indicate what
formatting was used.
This patch uses 2 labels to detect UYVY422 or YUYV422 pixel formats.
It defaults to UYVY422 for 8-bit 4:2:2 pictures to support files
that were created before the coding labels were introduced ~2008
The codec pix_fmt default was changed from 0 (PIX_FMT_YUV420P) to
-1 (PIX_FMT_NONE)
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This supports detection of uncompressed picture in files that
didn't include a Picture Coding Label. The lables weren't
available until SMPTE 377-1 and RP224v10
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Must check all 16 bytes because there is a planar 10-bit format
label that has equal first 15 bytes
Review-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-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
Provide a way to wrap around the segment index so pseudostreaming
live through a web server and html5 browser is simpler.
Also ensure that 0 (disable) is a valid value across the options
providing wrap around.
* 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>
WavPack has a comprehensive test suite, and a bunch
of corner cases.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
By validating the index entries while reading, we don't need to
seek at startup to validate the entries. If the error in the
index entries is not pointing to (our definition of) the start
of packets, and there is an index entry pointing at some of the
first packets after the metadata, the invalid index can be discarded
almost immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.
Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.
Signed-off-by: Martin Storsjö <martin@martin.st>
For encoding, AVCodecContext.frame_size is the number of input samples to
send to the encoder and does not necessarily correspond directly to the
timestamps of the output packets.
* 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>
r_frame_rate should in theory have something to do with input framerate,
but in practice it is often made up from thin air by lavf. So unless we
are targeting a constant output framerate, it's better to just use input
stream timebase.
Brings back dropped frames in nuv and cscd tests introduced in
cd1ad18a65
* 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.
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* 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>
WavPack has a comprehensive test suite, and a bunch
of corner cases.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-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
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.
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>
This allows opting for a lower MTU than what the AVIOContext
indicated, and allows writing into outputs that don't indicate
an MTU at all (such as plain files, which is useful for testing).
This also allows querying for the MTU via the avoption.
Signed-off-by: Martin Storsjö <martin@martin.st>
This library does not fit into Libav as a whole and its code is just a
maintenance burden. Furthermore it is now available as an external project,
which completely obviates any reason to keep it around.
URL: http://git.videolan.org/?p=libpostproc.git
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
According to newer RFCs, this packetization scheme should only
be used for interfacing with legacy systems.
Implementing this packetization mode properly requires parsing
the full H263 bitstream to find macroblock boundaries (and knowing
their macroblock and gob numbers and motion vector predictors).
This implementation tries to look for GOB headers (which
can be inserted by using -ps <small number>), but if the GOBs
aren't small enough to fit into the MTU, the packetizer blindly
splits packets at any offset and claims it to be a GOB boundary
(by using Mode A from the RFC). While not correct, this seems
to work with some receivers.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current code, an automatically inserted aconvert necessary
for format change would usually convert to mono for no good reason.
The new code will not avoid all conversions, but at least will keep
them among the layouts common to both filters.
lavfi have optional filters that depends on some components:
it is necessary to test which one is enabled to set the correct
dependencies.
Signed-off-by: Nicolas George <nicolas.george@normalesup.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>
Specifically, prevent jumping back in the file for the next index, since
this can lead to infinite loops where we jump between indexes referring
to each other, and don't read indexes that don't fit in the file.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
It is not supposed to be done outside lavc.
This is basically a revert of 818062f2f3.
It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.
The wtv-demux test change is because the sample starts with a B-frame.
We read sub_packet_h / 2 packets per line of data (during deinterleaving),
which equals zero if sub_packet_h <= 1, thus causing us to not read any
data, leading to an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This allows writing QuickTime-compatible fragmented mp4 (with
a non-empty moov atom) to a non-seekable output.
This buffers the mdat for the initial fragment just as it does
for all normal fragments, too. Previously, the resulting
atom structure was mdat,moov, moof,mdat ..., while it now
is moov,mdat, moof,mdat.
Signed-off-by: Martin Storsjö <martin@martin.st>
In nonseekable files, we already stop parsing the toplevel atoms
after finding moov and one mdat. In large seekable files (or files
that are seekable, but slowly, e.g. http), reading all the fragments
at the start can take a considerable amount of time. This allows
opting out from this behaviour.
Signed-off-by: Martin Storsjö <martin@martin.st>
If parsing moov+mdat in a non-seekable file, we currently
abort parsing directly after parsing the header of the mdat
atom. If we want to continue parsing later (if looking to
parse later fragments), we need to skip past the content of the
mdat atom, otherwise we end up parsing the content of the mdat
atom as root level atoms.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
Theres no usefull or even remotely complete information on it currently.
Which just leads to confusion.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Mostly based on doc/examples/filtering.c. lavfi API is still limited to
"buffer feeding" instead of "frame feeding" at the moment, so this
example code sticks with it.
The AVOptions based default to threads auto in 2473a45c8
works only if avplay does not use custom option handling
for -threads.
CC: <libav-stable@libav.org>
Samples buffer ref is allocated and loaded with the uninitialized data
pointers:
av_asrc_buffer_add_buffer()
-> av_asrc_buffer_add_samples()
-> avfilter_get_audio_buffer_ref_from_arrays(data, ...)
...which leads to a crash with at least lavfi/ashowinfo in case of !NULL
(see the for loop while samplesref->data[plane]).
For video, mark the first sample in a trun which doesn't have the
sample-is-non-sync-sample flag set as a keyframe.
In particular, the "sample does not depend on other samples" flag
isn't enough to make it a keyframe, since later frames still can
reference frames prior to that one (the flag only says that that
particular frame doesn't depend on other frames).
This fixes bug 215.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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.
The parser was fixed so this workaround should no longer
be necessary.
This allows using stream-copy to fix files with keyframes
incorrectly marked.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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.
This avoids reading any old data in the AVIOContext buffer after
the seek, and indicates to the mpegts demuxer that we've seeked,
avoiding continuity check errors.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
Revert "swscale: update context offsets after removal of AlpMmxFilter."
(commit a95e3fa90b)
and
Revert "swscale: Remove some write-only variables related to alpha handling."
(commit 9d03cb9fc5).
They broke alpha handling - it's the evil inline asm that still uses that
variable, so it's not truely write-only.
Enhance seeking by demuxing until the requested timestamp is
reached within the segment selected by the seek code using the
playlist info.
Some mpegts streams don't have dts set for all packets though,
this seeking method doesn't work well for that case.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows globally forcing specific cpuflags (or lack thereof)
Useful for debugging and benchmarking
Signed-off-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>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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
This prevents failed assertions further down in the packet processing
where we require non-negative values for packet_size_left.
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
H263 in RTP can be packetized in two formats (RFC 2190, RFC
2429/4629). The former normally uses the static payload type 34,
while the latter normally uses dynamic payload types with the
SDP format names H263-1998 or H263-2000.
Look for packets that don't look like proper RFC 2190 packets and
switch to depacketizing them according to the new format if they
match some heuristic criteria.
Signed-off-by: Martin Storsjö <martin@martin.st>
In order to match Linux behaviour better our
Windows-specific open() replacement should disable
Windows default file locking.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This needs the extradata to be extracted.
The approach used is the one MPlayer uses, though it is
unclear whether the 4 bytes extradata that are skipped
should be skipped always or only for AAC.
The AAC parser must be disabled, too, otherwise playback
still does not work.
Fixes trac issue #547.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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
According to 14496-12, the duration should be all 1s if
the duration is unknown. This is the case if writing a moov
atom without any samples described in it (e.g. as in ismv files).
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
This allows handling matroska files with errors.
Fixes test4.mkv and test7.mkv from the official Matroska test suite.
These are also trac issues #544 and #545.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Muxing pcm audio in MOV using avcodec_encode_audio() was failing
because avcodec_encode_audio() returns an incorrect packet size of 4
bytes. This can be reproduced by modifying the sample
ffmpeg/doc/examples/muxing.c to encode PCM, see ML patch
muxing-test.diff
I git bisected and commit 89ddff92a3 is the one that broke this. In
mov_write_header() if st->codec->frame_size <= 1 it sets it to 1. Then
avcodec_encode_audio() sets frame->nb_samples = avctx->frame_size, and
frame->nb_samples of 1 is used to compute a packet size of 4 bytes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Disadvantage is that it no longer allows modifying brightness through
adjustment of the RGB lookup table. Advantage is that now monowhite/black
no longer need to be identified as a RGB format.
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>
Closes ticket #999
NO_DSHOW_STRSAFE asks dshow.h header to not use secure string function
replacements.
Using secure replacements would break mingw.org compatibility as they don't
declare/define those functions.
Signed-off-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>
According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.
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.
The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
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.)
Prefix the functions/tables brktimegm, pcm_read_seek,
dv_offset_reset, voc_get_packet, codec_movaudio_tags,
codec_movvideo_tags.
After this, lavf has no global symbols without the proper prefix.
Signed-off-by: Martin Storsjö <martin@martin.st>
This probably fixes some of the use of uninitialized issues valgrind shows in fate.
It might also fix other issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
Keep the old protocol name around for backwards compatibility
until the next bump.
Deprecate the method of implicitly assuming the nested protocol.
For applehttp://server/path, it might have felt logical, but
supporting hls://server/path isn't quite as intuitive. Therefore
only support hls+http://server/path from now on.
Using this protocol at all is discouraged, since the hls demuxer
is more complete and fits into the architecture better. There
have been cases where the protocol implementation worked better
than the demuxer, but this should no longer be the case.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this demuxer was created, there didn't seem to be any
consensus of a common short name for this protocol. Now
the consensus seems to be to call it hls.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is required when stream copying VC1 in ismv - there's one
global header in the moov atom, but keyframes have a separate
sequence header prepended.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows easily differentiating between both implementations within the build
system and combining the native implementation for plain RTMP with librtmp for
the RTMPE, RTMPS, RTMPT, RTMPTE protocol variants.
* 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>
They were introduced in an earlier commit that introduced use of named
arguments. One cause was a typo, a second cause appears to be a bug in
x264asm that I work around by not using named arguments.
If muxing into mpegts, 4 byte startcodes for the first NAL
of an access unit is required. Thus it is simplest for the
RTP depacketizer to just use 4 byte startcodes everywhere.
Signed-off-by: Martin Storsjö <martin@martin.st>
Seek beyond the end will now directly return an error instead
of claiming to succeed and then return EOF immediately on next read.
This change is because before 47e015e6f1
mkv seek incorrectly never failed.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
In particular, detect when the index is obviously broken.
This fixes the worst symptoms of trac issue #958 and makes
sense to allow seeking in files without index.
However it is possible that there still is an index parsing bug
with that file.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Otherwise when we run into levels beyond the max. allowed
playback will be permanently broken.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
When segmenting the output from the mpegts muxer, one can
now set this option when cutting to a new segment, to make sure
the next segment starts with PAT/PMT/SDT.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
To make seeking work correctly, we must write a new granule for
each keyframe.
Unfortunately we currently have no regression tests due to no
included Theora encoder.
A test based on -vcodec copy from a Theora FATE sample should
probably be added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Set output files duration to recording_time option, if given.
Rationale: to save duration into metadata for file that is written to
non-seekable output, for formats like FLV (with metadata at beginning).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* 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>
This is so that TS fragments produced by
http://code.google.com/p/httpsegmenter/
would be compatible with JW Player.
A new member variable prev_payload_key was added to MpegTSWriteStream
to help detect transition from non-key to key frame, so that
PAT/PMT would not be produced for every keyframe in intra-only videos.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
At very small dimensions, this calculation could lead to zero-sized
filters, which leads to uninitialized output, zero-sized allocations,
loop overflows in SIMD that uses do{..}while(i++<filtersize); instead
of for(i=0;i<filtersize;i++){..} and several other similar failures.
Therefore, require a minimum filtersize of 1.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* 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>
input_video_filter and output_video_filter can't be NULL at this point.
If they are, the current code would likely crash anyway (since
filtered_frame would be NULL and sent to do_video_out().
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>
The added tests are limited to the case where timestamp discontinuities
are not allowed. The default is 30 hours which is arbitrarily picked and
quite conservative.
This prevents a out of memory condition due to duplicating a frame
millions of times.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
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>
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.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
These are common with audio atoms. Without this the demuxer would read two
bytes at a time for a mono 16-bit file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This can happen if an index table segment has a very large IndexStartPosition.
zzuf3.mxf is an example of such a file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Only the OPAtom demuxing logic is guaranteed to have index tables,
meaning OP1a files that lack an index would cause SIGSEGV.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* 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.
This sets __OUTPUT_FORMAT__ to win64 instead of win32, even though both
(through -m amd64) produce 64-bit binary code.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Functions using INIT_MMX may still access XMM registers through direct
means (xmm0-15). Therefore, they still need to be marked for clobber
so they can be properly saved/restored.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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
It currently has different meanings at different times (dts of the last
read packet/pts of the last decoded frame). Reduce obfuscation by
storing pts of the decoded frame in the frame itself.
Conflicts:
ffmpeg.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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
next_dts is used for estimating the dts of the next packet if it's
missing. Therefore, it makes no sense to set it from the pts of the last
decoded frame. Also it should be estimated from the current packet
duration/ticks_per_frame always, not only when a frame was successfully
decoded.
It currently has different meanings at different times (dts of the last
read packet/pts of the last decoded frame). Reduce obfuscation by
storing pts of the decoded frame in the frame itself.
Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.
In several tests, one less frame is encoded, which is more correct.
In the idroq test one more frame is encoded, which is again more
correct.
Behavior with stream copy should be unchanged.
The actual number (1/1000) will probably require some
discussion/tweaking in the future, but should be good enough for now,
since the timestamps in AVSubtitle are in this timebase by definition.
The output is obviously not supposed to contain video (since only
-acodec copy is specified), but that only happens because of the way -t
handling is implemented currently.
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>
A lot of files do not mark keyframes correctly via
granule, so detect keyframe or not based on data
and complain if it mismatches.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Calculates based on total file size and wavetaillength from the header.
Falls back to multiplying finalframeblocks by 8 instead of 4 so that it will
at least be overestimating for 24-bit. Currently it can underestimate the
final packet size, leading to decoding errors.
Before, drawtext filter deliberately altered given text coordinates if
text didn't fully fit on the picture. This breaks the use case of
scrolling large text, e.g. movie closing credits.
Add 'fix_bounds', to make it usable in such cases (by setting its value to 0).
Default behavior is not changed, and non-fitting text coords are fixed.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It would never be called when the searched-for position
was already in the index.
In the other cases, the ogg_reset at the end of the
read_timestamp function handled it.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
In this case, the pts values will be delayed by one, but
at the same time pts values might only be supplied for e.g.
keyframes.
This results on only the frame after the keyframe having a
pts value.
As a hack, make read_timestamp return the keyframe position
together with the pts from a following frame when seeking
to a keyframe.
Fixes trac issue #438.
However it causes the read_timestamp function to return a
pos value that is actually before the packet with the
indicated pts.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
about twice as fast as before.
the not CONFIG_SMALL case is also droped as it is not faster than the
CONFIG_SMALL case.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
We can handle v4 just fine, the parts we currently use
are the same for v3 and v4.
v4 can in addition contain an index which we so far do
not use though.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Fixes trac issue #438.
Seeking in that sample would cause ogg_read_timestamp to fail
because ogg_packet would go into a state where all packets
of stream 1 would be discarded until the end of the stream.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Line sizes are only 8-byte aligned, so use unaliged loads
for add_bytes_l2 pointers.
Increasing the alignment requirement to 16 seemed a bit extreme
(png may be used for rather small sizes).
Also fix a mov that had its arguments swapped, leading
add_bytes_l2 being applied on up to 8 bytes too few.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Right now those muxers use the default timebase in all cases(1/90000).
This patch avoid unnecessary rescaling and makes the printed timestamps
more readable.
Also, extend the printed information to include the timebases and packet
pts/duration and align the columns.
Obviously changes the results of all fate tests which use those two
muxers.
* qatar/master:
libx264: fix indentation.
vorbis: fix overflows in floor1[] vector and inverse db table index.
win64: add a XMM clobber test configure option.
movdec: Parse the dvc1 atom
ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6
swscale: K&R formatting cosmetics for Blackfin code
frwu: lowercase the FRWU codec name
movdec: fix dts generation in fragmented files
fate: make acodec-ac3_fixed test output raw AC3
APIchanges: add missing commit hashes
swscale: implement MMX, SSE2 and AVX functions for RGB32 input.
ra144enc: drop pointless "encoder" from .long_name
bethsoftvideo: fix palette reading.
mpc7: use av_fast_padded_malloc()
mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes
doc: decoding Forward Uncompressed is supported
Fix a typo in the x86 asm version of ff_vector_clip_int32()
pcmenc: Do not set avpkt->size.
ff_alloc_packet: modify the size of the packet to match the requested size
Conflicts:
doc/APIchanges
libavcodec/libx264.c
libavcodec/mpc7.c
libavformat/isom.h
libswscale/Makefile
libswscale/bfin/yuv2rgb_bfin.c
tests/ref/fate/bethsoft-vid
tests/ref/seek/ac3_ac3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This will be useful to test more aggressively for failures to mark XMM
registers as clobbered in Win64 builds, and prevent regressions thereof.
Based on a patch by Ramiro Polla <ramiro.polla@gmail.com>
Normally, the actual payload data contains sequence headers, too,
and the parser can extract this and set it as extradata. However,
the data in the dvc1 atom is the "official" extradata for the file.
This is required for proper stream copy of vc1 from ismv to ismv.
Signed-off-by: Martin Storsjö <martin@martin.st>
Do not use AVStream's duration for dts generation since it contains in
some cases the duration of the whole file instead of duration of the
samples in the moov. This happens if the mdhd holds the duration of the
whole file but has no entries or a zero duration in its stts.
* qatar/master: (22 commits)
frwu: Employ more meaningful return values.
fraps: Use av_fast_padded_malloc() instead of av_realloc()
mjpegdec: use av_fast_padded_malloc()
eatqi: use av_fast_padded_malloc()
asv1: use av_fast_padded_malloc()
avcodec: Add av_fast_padded_malloc().
swscale: enable dithering in MMX functions.
swscale: make rgb24 function macros slightly smaller.
avcodec.h: Remove some disabled cruft.
swscale: remove obsolete comment.
swscale-test: Drop unused argc and argv arguments from main().
zmbv: Employ more meaningful return values.
zmbvenc: Employ more meaningful return values.
vc1: prevent null pointer dereference on broken files
zmbv: check av_realloc() return values and avoid memleaks on ENOMEM
truespeech: align buffer
ac3: Do not read past the end of ff_ac3_band_start_tab.
dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
dv: Fix null pointer dereference due to ach=0
dv: check stype
...
Conflicts:
doc/APIchanges
libavcodec/asv1.c
libavcodec/avcodec.h
libavcodec/eatqi.c
libavcodec/fraps.c
libavcodec/frwu.c
libavcodec/zmbv.c
libavformat/dv.c
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Return the correct number of consumed bytes and set *data_size = 0.
Returned size is 1 too small, leading to that 1 byte being read as the next
frame, which results in an extra blank frame at the beginning of the stream.
Avoids doing malloc/free for each frame.
Also fixes valgrind errors due to use of uninitialized padding bytes.
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Wrapper around av_fast_malloc() that keeps FF_INPUT_BUFFER_PADDING_SIZE
zero-padded bytes at the end of the used buffer.
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
The code in the file is written by vitor in be19d752 (2008)
thus cannot have originated in libav which did not exist at that
time
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"Copyright (c) 2001 Michael Niedermayer" and "part of Libav" is not likely
not only am i not a libav developer there also was no libav in 2001
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
dv: Fix null pointer dereference due to ach=0
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>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
dv: check stype
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>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
* qatar/master: (29 commits)
fate: add golomb-test
golomb-test: K&R formatting cosmetics
h264: Split h264-test off into a separate file - golomb-test.c.
h264-test: cleanup: drop timer invocations, commented out code and other cruft
h264-test: Remove unused DSP and AVCodec contexts and related init calls.
adpcm: Add missing stdint.h #include to fix standalone header compilation.
lavf: add functions for accessing the fourcc<->CodecID mapping tables.
lavc: set AVCodecContext.codec in avcodec_get_context_defaults3().
lavc: make avcodec_close() work properly on unopened codecs.
lavc: add avcodec_is_open().
lavf: rename AVInputFormat.value to raw_codec_id.
lavf: remove the pointless value field from flv and iv8
lavc/lavf: remove unnecessary symbols from the symbol version script.
lavc: reorder AVCodec fields.
lavf: reorder AVInput/OutputFormat fields.
mp3dec: Fix a heap-buffer-overflow
adpcmenc: remove some unneeded casts
adpcmenc: use int16_t and uint8_t instead of short and unsigned char.
adpcmenc: fix adpcm_ms extradata allocation
adpcmenc: return proper AVERROR codes instead of -1
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/adpcmenc.c
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/libavcodec.v
libavcodec/mpc7.c
libavcodec/mpegaudiodec.c
libavcodec/options.c
libavformat/Makefile
libavformat/avformat.h
libavformat/flvdec.c
libavformat/libavformat.v
Merged-by: Michael Niedermayer <michaelni@gmx.at>
get_ue_golomb_long() is only tested for values up to 2^15 - 2 since
we can not write larger values.
Silence the test on success and return a non-zero value on error.
Use an heap scratch buffer instead of large stack buffer.
Remove unneeded includes.
* shariman/wmall:
Cosmetics: Fix some whitespace errors and indentation
Use correct variable type for 32-bit samples buffer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().
It also allows to set codec private options using av_opt_set_* before
opening the codec.
It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().
* qatar/master: (26 commits)
avconv: deprecate the -deinterlace option
doc: Fix the name of the new function
aacenc: make sure to encode enough frames to cover all input samples.
aacenc: only use the number of input samples provided by the user.
wmadec: Verify bitstream size makes sense before calling init_get_bits.
kmvc: Log into a context at a log level constant.
mpeg12: Pad framerate tab to 16 entries.
kgv1dec: Increase offsets array size so it is large enough.
kmvc: Check palsize.
nsvdec: Propagate errors
nsvdec: Be more careful with av_malloc().
nsvdec: Fix use of uninitialized streams.
movenc: cosmetics: Get rid of camelCase identifiers
swscale: more generic check for planar destination formats with alpha
doc: Document mov/mp4 fragmentation options
build: Use order-only prerequisites for creating FATE reference file dirs.
x86 dsputil: provide SSE2/SSSE3 versions of bswap_buf
rtsp: Remove some unused variables from ff_rtsp_connect().
avutil: make intfloat api public
avformat_write_header(): detail error message
...
Conflicts:
doc/APIchanges
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/kmvc.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil_yasm.asm
libavcodec/x86/pngdsp-init.c
libavformat/movenc.c
libavformat/movenc.h
libavformat/mpegtsenc.c
libavformat/nsvdec.c
libavformat/utils.c
libavutil/avutil.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In some cases, what is left to read from ptr is smaller than EXTRABYTES.
Based on a patch by Thierry Foucu <tfoucu@gmail.com>.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
Provide MMX, SSE2 and SSSE3 versions, with a fast-path when the weights are
multiples of 512 (which is often the case when the values round up nicely).
*_TIMER report for the 16x16 and 8x8 cases:
C:
9015 decicycles in 16, 524257 runs, 31 skips
2656 decicycles in 8, 524271 runs, 17 skips
MMX:
4156 decicycles in 16, 262090 runs, 54 skips
1206 decicycles in 8, 262131 runs, 13 skips
MMX on fast-path:
2760 decicycles in 16, 524222 runs, 66 skips
995 decicycles in 8, 524252 runs, 36 skips
SSE2:
2163 decicycles in 16, 262131 runs, 13 skips
832 decicycles in 8, 262137 runs, 7 skips
SSE2 with fast path:
1783 decicycles in 16, 524276 runs, 12 skips
711 decicycles in 8, 524283 runs, 5 skips
SSSE3:
2117 decicycles in 16, 262136 runs, 8 skips
814 decicycles in 8, 262143 runs, 1 skips
SSSE3 with fast path:
1315 decicycles in 16, 524285 runs, 3 skips
578 decicycles in 8, 524286 runs, 2 skips
This means around a 4% speedup for some sequences.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The MP3 demuxer split the data in packets of 1024B which are later split
in MP3 frames by the MPEG audio parser. The last read is "truncated",
but this should not raise any error.
Solution-by: Michael Niedermayer
This makes the first packet of a track fragment run to get
the keyframe flag set properly if sample_degradation_priority
is nonzero.
This makes the keyframes flag be set properly for ismv files
created by Microsoft.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, we've only passed the key string on to the recursive
amf_parse_object for the mixedarray type, not for 'object'. By
passing the key string on, the recursive amf_parse_object can
store the amf objects as metadata.
This kind of data was seen in data from XSplit Broadcaster, received
over RTMP via Wowza. This patch allows reading this metadata.
Signed-off-by: Martin Storsjö <martin@martin.st>
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Currently, any samples in the final frame are not decoded because they are
only represented by one frame instead of two. So we encode two final frames to
cover both the analysis delay and the MDCT delay.
Check results for av_malloc() and fix an overflow in one call.
Related to CVE-2011-3940.
Based in part on work from Michael Niedermayer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
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>
(cherry picked from commit 5c011706bc)
Signed-off-by: Alex Converse <alex.converse@gmail.com>
I have no idea what the idea was behind the original code,
but the new code is equivalent to it.
In that loop that places the new node nodes[j] contains
always the data of the new node (since the steps are always
in order: FFSWAP copies node[j] to node[j-1], j is decremented).
Thus nodes[j].no == i and nodes[j].sym == HNODE.
make fate still passes and contains VP6 samples which use
FF_HUFFMAN_FLAG_HNODE_FIRST.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
While pshufb allows emulating bswap on XMM registers for SSSE3, more
shuffling is needed for SSE2. Alignment is critical, so specific codepaths
are provided for this case.
For the huffyuv sequence "angels_480-huffyuvcompress.avi":
C (using bswap instruction): ~ 55k cycles
SSE2: ~ 40k cycles
SSSE3 using unaligned loads: ~ 35k cycles
SSSE3 using aligned loads: ~ 30k cycles
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The functions are already av_ prefixed and intfloat header is already provided.
Install libavutil/intfloat.h
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
-vbsf doesn't exist anymore. It got renamed to -bsf somewhere along the
line. Update print statement accordingly.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Current demuxer recognizes several colorspace formats that begin with 'C420'
but does not yet recognize plain 'C420'. GStreamer's y4menc component
generates .y4m files with a 'C420' colorspace. This new comparison is
placed after the other 'C420' checks so that it doesn't interfere with
them.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
png: add missing #if HAVE_SSSE3 around function pointer assignment.
imdct36: mark SSE functions as using all 16 XMM registers.
png: move DSP functions to their own DSP context.
sunrast: Add a sample request for TIFF, IFF, and Experimental Rastfile formats.
sunrast: Cosmetics
sunrast: Remove if (unsigned int < 0) check.
sunrast: Replace magic number by a macro.
Conflicts:
libavcodec/dsputil.c
libavcodec/dsputil.h
libavcodec/pngdec.c
libavcodec/sunrast.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Offsets are relative to the end of the header, not the
start of the buffer, thus the buffer size needs to be subtracted.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Codec is too simple to gain much from it at lower resolutions,
but should help at very high resolutions, particularly for
v3 and v5 where a not too optimized pseudo-YUV to RGB
is done in the codec.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This reverts e6e7bfc1 and 365e1ec2.
The code may be incorrect both before and after the revert, but we
do not have any samples that were fixed by the original commits.
Fixes ticket #871.
With gcc 4.6 this part of the code is ca. 4x faster, resulting
in an overall speedup of around 5% for fate-fraps-v5 sample.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
On x86-64, it indeed uses all 16 registers (and on x86-32, this gets
clipped to 8). Not marking it properly causes callers of this function
to fail randomly because of XMM register clobbering.
Note: This fixes the following GCC warning :-
libavcodec/sunrast.c:94: warning: comparison of unsigned expression < 0 is always false.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Codec has only I- and skip-frames, so there is no
need for reget_buffer, change it so it works with
get_buffer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Causes FFmpeg to pass through the correct pts values,
instead of clobbering all to AV_NOPTS_VALUE (the av_init_packet
default) to then make up new ones based on only fps when muxing.
Included are also the related FATE ref changes, which all
some reasonable on quick investigation.
Also set all H.264 references to us -vsync drop to reduce the
diff for the ref files.
Otherwise almost all H.264 references need to change, mostly due
to now starting with negative pts values.
About 20 additional H.264 conformance tests needed -vsync
drop anyway because they create pts values that are out of
order and thus not possible to mux otherwise.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
aacenc: Fix LONG_START windowing.
aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
avplay: use the correct array size for stride.
lavc: extend doxy for avcodec_alloc_context3().
APIchanges: mention avcodec_alloc_context()/2/3
avcodec_align_dimensions2: set only 4 linesizes, not AV_NUM_DATA_POINTERS.
aacsbr: ARM NEON optimised sbrdsp functions
aacsbr: align some arrays
aacsbr: move some simdable loops to function pointers
cosmetics: Remove extra newlines at EOF
Conflicts:
libavcodec/utils.c
libavfilter/formats.c
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move libgsm_encode_close before its first use and call it
with the correct number of arguments.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
10l: Forgot to adjust deinterleave for new location of incoming samples in 7946a5a.
This produced incorrect, but surprisingly listenable results.
Thanks to Justin Ruggles for the report.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This allows to work around any non-monotonic time-stamp errors
by just discarding all time stamps.
This will be necessary to allow H.264 conformance tests to pass
after fixing time stamps to be passed through rawenc.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We may or may not be able to play the latter parts
but not demuxing at all seems like the worst possible behaviour.
Fixes playback of e.g.
http://playlist.yahoo.com/makeplaylist.dll?sid=128114687&sdm=web&pt=rd
As a proper solution either multiple video streams should
be exported or side data should be used to update extradata
if necessary.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Since it is set for e.g. webm muxer we should make it possible
to test such streams with framecrc, too.
Though the primary reason is that this allows the H.264 tests
to not run into this check when fixing raw video encode to
pass pts values on.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The tests work fine without it, and it will cause issues when the
rawvideo decoder is changed to properly handle pts values.
The H.264 conformance tests however are still broken, usually losing
the first frames without it.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes the video frame pts (off by one for each MVIh)
and makes the "key frames" decode stand-alone (MVIh
contains only palette, such a palette-only frame being
marked as key frame is not really correct).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Previously the decoder would raise an error.
The end result is the same, the time stamps only change
because regression tests create time stamps incorrectly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This prepares for assembly optimisations by moving the most
time-consuming loops to functions called through pointers
in a new context.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In this mode, no seeks will be done except for within moov/moof
fragments, which should fit within the AVIOContext buffer.
This allows pushing live smooth streaming format data to
a live publishing point on IIS over http.
Signed-off-by: Martin Storsjö <martin@martin.st>
Earlier, calling avcodec_encode_audio worked fine even if time_base
wasn't set. Now it crashes due to trying to scale the output pts to
the codec context time base. This affects e.g. VLC.
If no time_base is set for audio codecs, set it to the sample
rate.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
MVDATA may or may not be transmitted. If it is not, both
dmv_x and dmv_y is to be assumed zero.
This may not trigger wrong picture in all systems, but
it's a bug nevertheless. Fixes SA10116.vc1 on my 64-bit
Windows 7.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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