* 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>
* qatar/master:
smacker: Sanity check huffman tables found in the headers.
smacker: remove dead store
qdm2: Check data block size for bytes to bits overflow.
mxfdec: Fix files with essence containers larger than 2 GiB.
mxfdec: Employ correct printf conversion specifiers for POSIX int types.
vc1: always read the bfraction element for interlaced fields
fate: add XWD image regression test
lavf: prevent infinite loops while flushing in avformat_find_stream_info
matroskadec: Pad AAC extradata.
ismindex: Fix build on mingw
Conflicts:
libavformat/mxfdec.c
libavformat/utils.c
tests/lavf-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes not yet fixed parts of CVE-2011-3946.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously, it would not be read if refdist_flag was not set, however
according to the spec and the reference decoder, it should always be read.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master: (22 commits)
wma: Clip WMA1 and WMA2 frame length to 11 bits.
movenc: Don't require frame_size to be set for modes other than mov
doc: Update APIchanges with info on muxer flushing
movenc: Reindent a block
tools: Remove some unnecessary #undefs.
rv20: prevent calling ff_h263_decode_mba() with unset height/width
tools: K&R reformatting cosmetics
Ignore generated aviocat and ismindex tools.
build: Automatically include architecture-specific library Makefile snippets.
indeo5: prevent null pointer dereference on broken files
pktdumper: Use usleep instead of sleep
cosmetics: Remove some unnecessary block braces.
Drop unnecessary prefix from *sink* variable and struct names.
Add a tool for creating smooth streaming manifests
movdec: Calculate an average bit rate for fragmented streams, too
movenc: Write the sample rate instead of time scale in the stsd atom
movenc: Add a separate ismv/isma (smooth streaming) muxer
movenc: Allow the caller to decide on fragmentation
libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
movenc: Add support for writing fragmented mov files
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
ffmpeg.c
ffplay.c
libavfilter/Makefile
libavformat/Makefile
libavformat/avformat.h
libavformat/movenc.c
libavformat/movenc.h
libavformat/version.h
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If no data was seen for a stream decoder are returning 0 when fed with
empty packets for flushing. We can stop flushing when the decoder does
not return delayed delayed frames anymore. Changes try_decode_frame()
return value to got_picture or negative error.
CC: libav-stable@libav.org
The MDCT buffers in the decoder are only sized for up to 11 bits. The
reverse engineered documentation for WMA1/2 headers say that that for
all samplerates above 32kHz 11 bits are used. 12 and 13 bit support
were added for WMAPro. I was unable to make any Microsoft tools generate
a test file at a samplerate above 48kHz.
Discovered by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The field frame_size isn't written to the output anywhere except
than in mov.
This facilitates stream copy from formats that don't set frame_size.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add some space around operators and wrap a comment
that extends past the 80 char "limit"/guideline.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes a double release of the current frame on deinit.
Fixes CVE-2011-3934
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The index of the motion vector has to be checked before being
multiplied by 2 for the array index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes part2 of CVE-2011-3929
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes part1 of CVE-2011-3929
Possibly fixes part of CVE-2011-3936
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
MinGW doesn't have sleep, only _sleep (which is deprecated),
Sleep (which is defined in winbase.h and not in the standard
C headers) and usleep.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can also optionally split the file into individual fragments,
which allows it to be served from any web server without any
server side support.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes calculation of trackDuration if the MOVIentry array
is cleared. This is required by the fragmentation support in the
next patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit makes the check specific to the case that needs it.
Regression was introduced by
commit 62adc60b97
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Fri Dec 16 06:13:04 2011 +0100
avidec: Check that the header chunks fit in the available filesize.
Fixes Ticket771
Bug found by: Diana Elena Muscalu
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Although it has been deprecated for a long time, its intended
replacement (request_channel_layout) is not actually used anywhere, so
request_channels is currently the only way to access that functionality.
Allows our users to still build against a libpostproc with the old
API/ABI. Distributions can use this option to defer the soname bump.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously this was just checked in case of slice threads,
but frame threads do not support this either currently.
Making them support this is of course the long term goal
Fixes bug155
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reduces the delay when opening the video with quicktime.
Idea-by: Maksym Veremeyenko <verem@m1stereo.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The previous implementation assumed that a new picture would always
supersede the previous picture. Similarly, presentation segments
were assumed to pertain to the most-recently-read picture.
However, each presentation segment may refer to 0 or more pictures
by their ID. Picture IDs may repeat, and a repeated picture ID
indicates that the old picture for that ID is no longer needed
and may be discarded.
The new implementation allocates a buffer with one slot for each
possible picture ID (the picture ID is a 16-bit field) and
properly decodes presentation segments so that all relevant
pictures are output upon encountering a display segment.
Given that most PGS streams are unlikely to use more than a small
fraction of the available picture IDs, it would probably be better
to use a more memory-efficient data structure. I'm lazy though, so
I leave this to a more motivated individual.
I've tested the code with MKV files in VLC (a recent revision from
their git repo) and with HandBrake (a version that I hacked up to
use ffmpeg's PGS subtitle decoder).
Review-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes memory corruption when seeking in broken streams.
a random mpeg4 in nut file was used to debug.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This matches the spec as well as the reference decoder, and fixes a bug
with interlaced frame decoding.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master: (25 commits)
riff: fix invalid av_freep() calls on EOF in ff_read_riff_info
pam: Fix a typo that broke writing and reading PAM files.
mxfdec: fix memleak on av_realloc failures
mxfdec: Do not parse slices or DeltaEntryArrays.
mxfdec: hybrid demuxing/seeking solution
mxfdec: Add Avid's essence element key.
mfxdec: Separate mxf_essence_container_uls for audio and video.
mxfdec: Compute packet offsets properly.
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack.
mxfdec: use av_dlog() for 'no corresponding source package found'
mxfdec: Make mxf->partitions sorted by offset.
mxfdec: parse ThisPartition
mxfdec: Speed up metadata and index parsing.
mxfdec: Make sure DataDefinition is consistent between material track and source track.
mxfdec: add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
mxfdec: Add hack that adjusts the n_delta calculation when system items are present.
mxfdec: Parse IndexTableSegments and convert them into AVIndexEntry arrays.
mxfdec: Move FooterPartition to MXFContext and make sure it is never zero.
mxfdec: check return value of avio_seek
mxfdec: skip to end of structural sets
...
Conflicts:
configure
libavcodec/pnm.c
libavformat/mxfdec.c
libavformat/riff.c
libavformat/rtsp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This was a regression that came in when I switched to using the
h.264 annex b filter all the time. As the filter modifies extradata,
its use violates the statelessness assumption that exists in the
'ffmpeg' command line tool, and maybe elsewhere. It assumes that
a docoder can be reinitalised and pointed to an existing stream and
get the same results.
For now, the only way to meet this requirement is to backup the
extradata.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Makes it possible to select the name/path of the tool for compiling
the non-inline assembly code.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
It is a really bad idea to assign a video codec id
when we have set codec_type to audio and vice versa.
Prevents detection of mp2 in mxf as mpeg2video.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers are not needlessly
parsed. This speeds up parsing large files a lot.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes 0001GL.MXF.V1.mxf_opatom.mxf and 0001GL00.MXF.A1.mxf_opatom.mxf
getting two streams each due to both using the same SourcePackageID.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes reading of partition packs. The code stops reading after the
operational pattern and should skip the array of essence container
labels that follow.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Now that 0.8 is out we can reapply this commit. It breaks shared
avserver builds due to avserver using internal libavformat symbols,
which are now hidden, so this commit also disables avserver with
--enable-shared.
the written length was off by 2 causing aac decoders to fail with the data.
lucky the encoder was marked as experimental and not used much
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some applications use the j2c extension for jpeg2000 codestream files.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.
This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.
Signed-off-by: Martin Storsjö <martin@martin.st>
This check isn't relevant in the way the code currently works.
Also change a case of if (x == 0) into if (!x).
Signed-off-by: Martin Storsjö <martin@martin.st>
Original commit:
commit 2473a45c85
Author: Janne Grunau <janne-libav@jannau.net>
Date: Wed Jan 18 10:53:41 2012 +0100
threads: change the default for threads back to 1
Using threaded decoding by default breaks backward compatibility if
AVHWAccel is used or if an appliction sets threadunsafe callbacks.
Avconv and avplay still use -threads auto if not specified.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec: Use our own SSRC in the SDES field when sending RRs
Finalize changelog for 0.8 Release
Prepare for 0.8 Release
threads: change the default for threads back to 1
threads: update slice_count and slice_offset from user context
aviocat: Remove useless includes
doc/APIChanges: fill in missing dates and hashes
Revert "avserver: fix build after the next bump."
mpegaudiodec: switch error detection check to AV_EF_BUFFER
lavf: rename fer option and document resulting (f_)err_detect options
lavc: rename err_filter option to err_detect and document it
mpegvideo: fix invalid memory access for small video dimensions
movenc: Reorder entries in the MOVIentry struct, for tigheter packing
rtsp: Remove extern declarations for variables that don't exist
aviocat: Flush the output before closing
Conflicts:
Changelog
RELEASE
libavcodec/mpegaudiodec.c
libavcodec/pthread.c
libavformat/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using threaded decoding by default breaks backward compatibility if
AVHWAccel is used or if an appliction sets threadunsafe callbacks.
Avconv and avplay still use -threads auto if not specified.
This temporarily (until 0.8 is released) reverts commit
8e1340abc3. That commit breaks shared
builds because of symbol hiding. Reverting it will enable shared builds
for 0.8
When either video dimension is only one macroblock, subtractions
based on v_edge_pos and the macroblock size may be negative. In
that situation, an unsigned comparison isn't sufficent to test for
MV overruns, because a limit of (unsigned)-1 will let any other
value pass.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
That way all mix levels as exported by the parser
will have the same meaning.
Previously the 3bit center mix level for eac3 was
used to index in a 4 entry table leading to out of array reads.
this change removes the table and offsets the ac3 variable by 4
so it matches the meanings for eac3 except the reserved case.
The reserved case is then explicitly handled.
Idea-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Sometimes the scan finds nothing that qualifies for addition to
the array and pos is zero after the loops. The code forces pos to
1 and the array is then processed as if it had one valid element in it,
producing some amusing but not very useful results.
I don't see the rationale for this. If pos is zero coming out of the
loops, the only appropriate thing to do is set t->angle to zero. The
attached patch does that. It's worked properly in several tests so far.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
As the test is run during fate and the benchmark is useless for fate
this very slightly speeds up fate. Its also consistent with the other
tests.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Add a tool that uses avio to read and write, doing a plain copy of data
ARM: fix build with FFT enabled and MDCT disabled
lavf: force single-threaded decoding in avformat_find_stream_info
avidec: migrate last of lavf from FF_ER_* to AV_EF_*
avserver: fix build after the next bump.
Conflicts:
libavformat/Makefile
libavformat/avidec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Originally, sizeof(struct MOVIentry) was 48, after the reordering,
it is 40 in my build configuration.
When writing really long mov/mp4 files, this can make a difference
- this saves a bit over 2 MB of memory per hour of video (down to
10.3 MB per hour from 12.3 MB per hour initially) for a video with
75 packets per second - 25 fps + 50 audio packets (which is the
case for AMR audio).
Signed-off-by: Martin Storsjö <martin@martin.st>
The H.264 decoder needs SPS and PPS for initialization during
multi-threaded decoding. When probed single-threaded SPS and PPS are
copied to extradata and are available for proper initialization of
the decoder before the first frame is decoded.
* qatar/master:
mpeg12: check for available bits to avoid an infinite loop
fate: add some shorthands to run groups of tests
fate: Give some tests more sensible names.
cosmetics: Rename ffsink to avsink.
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/dpcm.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/pcm.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/wma.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This one was missed in the previous fraps fix, the
allocation is exactly the same in both cases.
Fixes fraps-v5 under valgrind.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The ABI differences are negligible, but its easier for all if
all distros have libpostproc HEAD under the same soname and
debian bumped soname without consulting upstream, so as silly as
it is following this is probably the least pain for all.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is needed in case the get_buffer() callback doesnt set
width/height.
Ideally all decoders would make calls through some wraper
to the callbacks and that wraper would call ff_init_buffer_info()
But until thats done, the default reget buffer must call this
itself as it needs the values for the changed size check later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tta: cast output data pointer to the correct type
avconv: fix -frames for video encoders with delay.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
There is no guarantee that the casted double which is assigned to the
variable will be contained in an int (also if it is almost sure for most
non-alien architectures).
Use long long int to contain such values instead of an int, which is
required to contain at least 64 bits, so it is guaranteed to contain also
int64_t values, which are used by some fields.
In particular, should fix trac ticket #921.
This fixes parallel FATE (make fate -j4) failing under valgrind with:
Syscall param ioctl(TCSET{S,SW,SF}) points to uninitialised byte(s)
at 0x5D98B23: tcsetattr (tcsetattr.c:88)
by 0x43D66C: term_init (ffmpeg.c:734)
by 0x43CD8D: main (ffmpeg.c:5071)
Address 0x7fefffdd0 is on thread 1's stack
Uninitialised value was created by a stack allocation
at 0x43D5B0: term_init (ffmpeg.c:716)
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The same as av_fast_malloc but uses av_mallocz and keeps extra
always-0 padding.
This does not mean the memory will be 0-initialized after each call,
but actually only after each growth of the buffer.
However this makes sure that
a) all data anywhere in the buffer is always initialized
b) the padding is always 0
c) the user does not have to bother with adding the padding themselves
Fixes another valgrind warning about use of uninitialized data,
this time with fate-vsynth1-jpegls.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This combination is quite odd and almost certainly a bug if
it happens.
Reviewed-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The tool is useful for testing the internal arithmetic evaluation engine
(indeed I plan to use it in FATE), and provides a handy calculator when
you can't rely on bc ;-).
They allow to implement the if/then/else logic, which cannot be
implemented otherwise.
For example the expression:
A*B + not(A)*C
always evaluates to NaN if B is NaN, even in the case where A is 0.
* qatar/master:
rv34: add NEON rv34_idct_add
rv34: 1-pass inter MB reconstruction
add SMJPEG muxer
avformat: split out common SMJPEG code
pictordec: Use bytestream2 functions
avconv: use avcodec_encode_audio2()
pcmenc: use AVCodec.encode2()
avcodec: bump minor version and add APIChanges for the new audio encoding API
avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
avcodec: add a public function, avcodec_fill_audio_frame().
rv34: Intra 16x16 handling
rv34: Inter/intra MB code split
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/pictordec.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/rv34dsp.asm
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Overall almost 4% faster, idct_add down from 350 to 85 cycles, idct_dc_add
down from 83 to 30 cycles.
squash: rv34 idct rearrange partial register loads
This reworks the frame skipping code such that the reference
buffers are still updated according to the header.
However it also ensures that the current frame will not end
up in any reference buffer.
Also fixes a hang with frame-multithreading, probably because
get_buffer was already called and would have reset the progress,
however the frame could remain in framep due to the missing update
(or it could be assigned to next_framep and a skip_frame skip would
then write it into framep - there might be even more failure modes).
Sample might become available at samples/nsv/vp8.nsv
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Enhances seeking by demuxing until the requested timestamp is reached within
the segment selected by the seek code using the playlist info.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is required by the spec and fixes video-1frag.ogg.48.ogg. (FPE)
Based on the debuging work of Oana Stratulat and ubitux.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
mov: cosmetics - move a line to a better position and add a comment
Oana Andreea Stratulat submitted a similar patch to trac, but forgot
to notify the ML about it.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.
This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.
* qatar/master:
fate: split ADPCM and DPCM test references into separate files.
mov, mxfdec: Employ more meaningful return values.
lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer()
wavpack: fix clipping for 32-bit lossy mode
vb: Use bytestream2 functions
Conflicts:
libavcodec/utils.c
libavcodec/vb.c
libavformat/mxfdec.c
tests/fate/dpcm.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Extract processing of intra 16x16 blocks from intra macroblock
processing.
Also implement a function performing inverse transform and block
reconstruction for DC-only blocks in 1 pass instead of 2.
Split inter/intra macroblock handling code. This will allow further
optimizations such as performing inverse transform and block reconstruction
in a single pass as well as specialize code.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Do not fail audio decoding with avcodec_decode_audio3 if user has set a
custom get_buffer. Strictly speaking, this was never allowed by the API,
but it seems that some software packages did so anyways. In order to
unbreak applications (cf. http://bugs.debian.org/655890), this change
clarifies the API and overrides the custom get_buffer() with the defaults.
This change is inspired by a similar
commit (c3846e3eba) in FFmpeg.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Current code would just return uninitialized data with no way
to detect this condition.
Instead, fill the whole GUID with 0 in that case.
Fixes valgrind uninitialized data errors in fate-seek-lavf_asf.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reference decoder clips data before shifting it to final range and also
forces 32-bit lossy mode to be actually 24-bit lossy mode in order to be
able to perform proper clipping.
This is not a real error and memsetting always even when the
size did not change is overkill, but it still should be
an acceptable trade-off.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
sgidec: Use bytestream2 functions to prevent buffer overreads.
cosmetics: Move static and inline attributes to more standard places.
configure: provide libavfilter/version.h header to get_version()
swscale: change yuv2yuvX code to use cpuflag().
libx264: Don't leave max_b_frames as -1 if the user didn't set it
FATE: convert output to rgba for the targa tests which currently output pal8
fate: add missing reference files for targa tests in 9c2f9b0e2
FATE: enable the 2 remaining targa conformance suite tests
targa: add support for rgb555 palette
FATE: fix targa tests on big-endian systems
Conflicts:
libavcodec/sgidec.c
libavcodec/targa.c
libswscale/x86/output.asm
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At the very least this should fix warnings about unused static
functions if one or more of these is not defined.
However even compilation might be broken if the compiler does
not optimize the function away completely.
This actually happens in case of the AVX function, since the
function pointer is used in an assignment that is not under
an #if and thus probably only optimized away after the function
was already marked as used.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We do this for all other codec_tag checks in mpegvideo*/h26*
doing it here too makes the code more consistent.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While we correctly "register" the side data when we split it,
the application (in this case FFmpeg) might not update the
AVPacket pool it uses to finally free the packet, thus
causing a leak.
This also makes the av_dup_packet unnecessary which could
cause an even worse leak in this situation.
Also change the code to not modify the user-provide AVPacket at all.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* cus/stable:
ffplay: silence buffer size must be a multiple of frame size
ffplay: use swr_set_compensation for audio synchronization
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: add tests for targa
ARM: fix Thumb-mode simple_idct_arm
ARM: 4-byte align start of all asm functions
rgb2rgb: rgb12to15()
swscale-test: fix stack overread.
swscale: fix invalid conversions and memory problems.
cabac: split cabac.h into declarations and function definitions
cabac: Mark ff_h264_mps_state array as static, it is only used within cabac.c.
cabac: Remove ff_h264_lps_state array.
Conflicts:
libswscale/rgb2rgb.h
libswscale/swscale_unscaled.c
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some of these encoders may produce invalid bitstreams, which should not
be done without the user knowing.
Some of these decoders may be unfinished and may contain security issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
max_b_frames is initialized to -1 for libx264, to allow
distinguishing between an explicit user set 0 and a default not
touched 0 (see bb73cda2).
If max_b_frames is left as -1, this affects dts generation (where
expressions like max_b_frames != 0 are used), so make sure it is
left at the default 0 after the libx264 init function returns.
This avoids unnecessarily producing dts != pts when using
profile=baseline.
Signed-off-by: Martin Storsjö <martin@martin.st>
The alignment directive must obviously precede the label.
This was never noticed in ARM mode since the location is
already aligned there.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Due to apprent bugs in the GNU assembler and/or linker, relocations
can be incorrectly processed if the alignment of a Thumb instruction
is changed in the output file compared to the input object.
This fixes crashes in h264 decoding with Thumb enabled. No effect in
ARM mode since everything is 4-byte aligned there.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes a segmentation fault when doing a transcoding and a stream
copy of the same input stream at the same time, e.g.:
ffmpeg -i in.mkv -c:v mpeg2video transcode.m2v -c copy copy.ts
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With the added benefit that allowing -segment_list_size 0 makes it
possible to keep all segment entries in the list file.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes problems where rgbToRgbWrapper() is called even though it doesn't
support this particular conversion (e.g. converting from RGB444 to
anything). Thirdly, fixes issues where rgbToRgbWrapper() is called for
non-native endiannness conversions (e.g. RGB555BE on a LE system).
Fourthly, fixes crashes when converting from e.g. monowhite to
monowhite, which calls planarCopyWrapper() and overwrites/reads because
n_bytes != n_pixels.
* qatar/master: (21 commits)
utils: Check for extradata size overflows.
ARM: rv34: fix asm syntax in dc transform functions
avio: Fix the value of the deprecated URL_FLAG_NONBLOCK
rv34: fix and optimise frame dependency checking
rv34: NEON optimised dc only inverse transform
avprobe: use avio_size() instead of deprecated AVFormatContext.file_size.
ffmenc: remove references to deprecated AVFormatContext.timestamp.
lavf: undeprecate read_seek().
avserver: remove code using deprecated CODEC_CAP_PARSE_ONLY.
lavc: replace some remaining FF_I_TYPE with AV_PICTURE_TYPE_I
lavc: ifdef out parse_only AVOption
nellymoserdec: SAMPLE_FMT -> AV_SAMPLE_FMT
mpegvideo_enc: ifdef out/replace references to deprecated codec flags.
riff: remove references to sonic codec ids
indeo4: add some missing static and const qualifiers
rv34: DC-only inverse transform
avconv: use AVFrame.width/height/format instead of corresponding AVCodecContext fields
lavfi: move version macros to a new installed header version.h
vsrc_buffer: release the buffer on uninit.
rgb2rgb: rgb12tobgr12()
...
Conflicts:
avconv.c
doc/APIchanges
ffprobe.c
libavfilter/Makefile
libavfilter/avfilter.h
libswscale/rgb2rgb.c
libswscale/rgb2rgb.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes standalone compilation of some decoders with --disable-optimizations.
cabac.h defines some inline functions that use symbols from cabac.c. Without
optimizations these inline functions are not eliminated and linking fails with
references to non-existing symbols.
Splitting the inline functions off into their own header and only #including
it in the places where the inline functions are used allows #including cabac.h
from anywhere without ill effects.
This isn't used in practice anywhere within libav at the moment,
but change it for consistency until it is removed.
URL_RDONLY/WRONLY were fixed in commit 5b81e29593 (after the
values that actually were used were changed at the major bump,
in commit cbea3ac8), but this flag was unintentionally left unfixed.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using the double variant causes several pointless conversions between
double and int.
Worse, one of the conversions is in an inner loop together with a
function using MMX, resulting in undefined behaviour.
Based on debugging by Ray Simard.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Tested-by: Ray Simard <rhs.ffmpeg@sylvan-glade.com>
The sporadic threading errors during fate-rv30 were caused by calling
ff_thread_await_progress with mb row -1 as argument. That returns
immediately since progress is initialized to -1. Not yet computed
motion vectors from the reference could be used for the first
macroblocks.
This fixes dithering for rgb555le, it appears gcc had moved the
setup of the variables after the asm or something like that.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_destruct_packet() always frees the packet data even when the demuxer
is going to re-use it, thus causing crashes when decoding audio
frames (as implemented in a pending patch).
av_free_packet() is used instead, as it allows each demuxer to set the
right packet data releasing mechanism through the pkt->destruct callback.
When decoding coefficients, detect whether the block is DC-only, and take
advantage of this knowledge to perform DC-only inverse transform.
This is achieved by:
- first, changing the 108x4 element modulo_three_table into a 108 element
table (kind of base4), and accessing each value using mask and shifts.
- then, checking low bits for 0 (as they represent the presence of higher
frequency coefficients)
Also provide x86 SIMD code for the DC-only inverse transform.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master:
fft: init functions with INIT_XMM/YMM.
pcmenc: set frame_size to 0.
gsm demuxer: use generic seeking instead of a gsm-specific function.
gsm demuxer: return packets with only 1 gsm block at a time.
avcodec: add GSM parser
doc: Replace ffmpeg references in avserver config file by avconv.
doc: Fix names of av_log color environment variables.
Fix a bunch of platform name and other typos.
Add some missing changelog entries and release 0.8_beta2
No longer build libpostproc by default
wtv: fix memleaks during normal operation
threads: add CODEC_CAP_AUTO_THREADS for libvpx and xavs
Conflicts:
Changelog
RELEASE
cmdutils.c
configure
doc/ffserver.conf
doc/platform.texi
ffplay.c
libavcodec/Makefile
libavcodec/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also change synchronize_audio to only calculate the wanted number of samples
instead of doing the actual synchronization, and make swr_convert handle the
sample addition or deletion.
This new method replaces the old buggy way which simply deleted or
multiplied samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
This is required to handle clobbering of XMM registers on Win64
correctly. Fixes FFT and all tests depending on FFT on Win64.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes some cases where the clipping was entirely missing.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Thanks (for the comments and review) -to: Reimar, beastd, Ronald
This reverts commit 0efd48dfd1.
Reason for the revert is that the code seems based on some
misunderstanding on how the code works.
Conflicts:
libavdevice/v4l2.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The WAVE demuxer returns packets with many blocks per frame, which needs to be
parsed into single blocks. This has a side-effect of fixing the timestamps.
* qatar/master: (22 commits)
rv34: frame-level multi-threading
mpegvideo: claim ownership of referenced pictures
aacsbr: prevent out of bounds memcpy().
ipmovie: fix pts for CODEC_ID_INTERPLAY_DPCM
sierravmd: fix audio pts
bethsoftvideo: Use bytestream2 functions to prevent buffer overreads.
bmpenc: support for PIX_FMT_RGB444
swscale: fix crash in fast_bilinear code when compiled with -mred-zone.
swscale: specify register type.
rv34: use get_bits_left()
avconv: reinitialize the filtergraph on resolution change.
vsrc_buffer: error on changing frame parameters.
avconv: fix -copyinkf.
fate: Update file checksums after the mov muxer change in a78dbada55
movenc: Don't store a nonzero creation time if nothing was set by the caller
bmpdec: support for rgb444 with bitfields compression
rgb2rgb: allow conversion for <15 bpp
doc: fix stray reference to FFmpeg
v4l2: use C99 struct initializer
v4l2: poll the file descriptor
...
Conflicts:
avconv.c
libavcodec/aacsbr.c
libavcodec/bethsoftvideo.c
libavcodec/kmvc.c
libavdevice/v4l2.c
libavfilter/vsrc_buffer.c
libswscale/swscale_unscaled.c
libswscale/x86/input.asm
tests/ref/acodec/alac
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
tests/ref/vsynth1/dnxhd_1080i
tests/ref/vsynth1/mpeg4
tests/ref/vsynth1/qtrle
tests/ref/vsynth1/svq1
tests/ref/vsynth2/dnxhd_1080i
tests/ref/vsynth2/mpeg4
tests/ref/vsynth2/qtrle
tests/ref/vsynth2/svq1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* richardpl/sws:
rgb2rgb: remove unused bgr8torgb8()
rgb2rgb: rgb12tobgr12()
rgb2rgb: allow conversion for <15 bpp
bmpenc: support for PIX_FMT_RGB444
bmpdec: support for rgb444 with bitfields compression
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Statistics for bourne.rmvb -an -f null
1 thread: 37.12s user 0.03s system 99% cpu 37.174 total
2 threads: 47.63s user 0.24s system 185% cpu 25.807 total
4 threads: 41.21s user 0.30s system 327% cpu 12.674 total
Under certain conditions pictures could be released before they were
returned with frame-threading. Broken mv computation in the upcoming
rv34 frame-threading patch was caused by this.
To prevent contexts from running out of available pictures the loop
releasing "unused" pictures has to be run for B frames too.
Fixes Libav Bug 195.
This doesn't make the code handle sample rate or upsample/downsample
change properly but this is still a good sanity check.
Based on change by Michael Niedermayer.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This reverts commit 77d88b872d.
The revert fixes actual overflows and a segfault as the variables
are signed and can be negative.
Conflicts:
libswscale/swscale.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If the creation time is stored in the file as a zero, the
mov demuxer skips exporting the creation time. Currently,
files muxed without a creation time get demuxed with a
Jan 1st 1970 creation timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
This test does not work on all platforms and until it does
it just hides new failures, which is really bad.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: Add tests for more AAC features.
aacps: Add missing newline in error message.
fate: Add tests for vc1/wmapro in ism.
aacdec: Add a fate test for 5.1 channel SBR.
aacdec: Turn off PS for multichannel files that use PCE based configs.
cabac: remove put_cabac_u/ueg from cabac-test.
swscale: RGB4444 and BGR444 input
FATE: add test for xWMA demuxer.
FATE: add test for SMJPEG demuxer and associated IMA ADPCM audio decoder.
mpegaudiodec: optimized iMDCT transform
mpegaudiodec: change imdct window arrangment for better pointer alignment
mpegaudiodec: move imdct and windowing function to mpegaudiodsp
mpegaudiodec: interleave iMDCT buffer to simplify future SIMD implementations
swscale: convert yuy2/uyvy/nv12/nv21ToY/UV from inline asm to yasm.
FATE: test to exercise WTV demuxer.
mjpegdec: K&R formatting cosmetics
swscale: K&R formatting cosmetics for code examples
swscale: K&R reformatting cosmetics for header files
FATE test: cvid-grayscale; ensures that the grayscale Cinepak variant is exercised.
Conflicts:
libavcodec/cabac.c
libavcodec/mjpegdec.c
libavcodec/mpegaudiodec.c
libavcodec/mpegaudiodsp.c
libavcodec/mpegaudiodsp.h
libavcodec/mpegaudiodsp_template.c
libavcodec/x86/Makefile
libavcodec/x86/imdct36_sse.asm
libavcodec/x86/mpegaudiodec_mmx.c
libswscale/swscale-test.c
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/fate/demux.mak
tests/fate/microsoft.mak
tests/fate/video.mak
tests/fate/wma.mak
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows to get out of ffplay if it or SDL got stuck.
This for example happens when the audio driver is playing something
else and doesnt support mixing multiple sources.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
The functions are not used in any part of Libav, therefore testing them in the
cabac-test is unnecessary. Since this makes them unused, remove the functions.
Print a "\n" at the end of each section, also print the section name in
the section print function, print the chapter name only in case the
chapter contains multiple entries.
Increase textual output readability - different sections can be
distinguished more easily.
Using an always_inline function makes the memcpy length a constant,
any reasonable compiler will replace it by a single mov instruction
without us having to duplicate the actual code.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes reads of uninitialized data by the parser when running
FATE sample h264-conformance/SL1_SVA_B.264.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The new version returns AVERROR(EINVAL) is the specified paramters are invalid,
and also creates the resampler if none was used so far.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(Does not attempt to decode percetual audio data inside.)
Code coverage: libavformat/xwma.c: 3% -> 75%
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(Don't attempt to decode JPEG data.)
Code coverage: libavformat/smjpeg.c: 0% -> 69%
libavcodec/adpcm.c: 0% -> 10% (fresh run); 92.4% -> 93% following a FATE run
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
>> time ./avconv -i file.avi -f null -
Before : real 0m7.784s
After : real 0m3.662s
Tested on a Intel Core i3 Processor (2 cores, 4 threads).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (29 commits)
cabac: Move code only used within the CABAC test program into the test program.
vp56: Drop unnecessary cabac.h #include.
h264-test: Initialize AVCodecContext.av_class.
build: Skip compiling network.h and rtsp.h if networking is not enabled.
cosmetics: drop some pointless parentheses
Disable annoying warning without changing behavior
faq: Solutions for common problems with sample paths when running FATE.
avcodec: attempt to clarify the CODEC_CAP_DELAY documentation
avcodec: fix avcodec_encode_audio() documentation.
FATE: xmv-demux test; exercise the XMV demuxer without decoding the perceptual codecs inside.
vqf: recognize more metadata chunks
FATE test: BMV demuxer and associated video and audio decoders.
FATE: indeo4 video decoder test.
FATE: update xxan-wc4 test to a sample with more code coverage.
Change the recent h264_mp4toannexb bitstream filter test to output to an elementary stream rather than a program stream.
g722enc: validate AVCodecContext.trellis
g722enc: set frame_size, and also handle an odd number of input samples
g722enc: split encoding into separate functions for trellis vs. no trellis
mpegaudiodec: Use clearer pointer math
tta: Fix returned error code at EOF
...
Conflicts:
libavcodec/h264.c
libavcodec/indeo3.c
libavcodec/interplayvideo.c
libavcodec/ivi_common.c
libavcodec/libxvidff.c
libavcodec/mpegvideo.c
libavcodec/ppc/mpegvideo_altivec.c
libavcodec/tta.c
libavcodec/utils.c
libavfilter/vsrc_buffer.c
libavformat/Makefile
tests/fate/indeo.mak
tests/ref/acodec/g722
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add -show_frames option to ffprobe.
Partially based on the work of Thomas Kuehnel <kuehnelth@googlemail.com>
for SOCIS 2011.
The wicked idea of creating a special "packets_and_frames" container for
structured formats (JSON and XML) comes from Clément.
rtsp.h relies on network.h and the latter conditionally defines fallback OS
structures that rely on configure tests, which are only run if networking
is enabled.
This fixes various problems with getting stream info. For example playback of the
file of Ticket88. Multithreaded find_stream_info should be reenabled
once it works correctly
This partly reverts 212fd3a1f1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The previous sample used for this test only contained type 0 frames.
Replace it with a sample that also features type 1 frames.
Code coverage:
libavcodec/xxan.c: 72% -> 89%
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
flicvideo: fix invalid reads
vorbis: Avoid some out-of-bounds reads
vqf: add more known extensions
cabac: remove unused function renorm_cabac_decoder
h264: Only use symbols from the SVQ3 decoder under proper conditionals.
add bytestream2_tell() and bytestream2_seek() functions
parsers: initialize MpegEncContext.slice_context_count to 1
spdifenc: use special alignment for DTS-HD length_code
Conflicts:
libavcodec/flicvideo.c
libavcodec/h264.c
libavcodec/mpeg4video_parser.c
libavcodec/vorbis.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The decoders should not only be flushed on EOF or error, but also when
e.g. probe size was reached.
It is best to just always flush by default and only disable it
explicitly when we know that we have everything we need.
Fixes trac ticket #879.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (21 commits)
ipmovie: do not read audio packets before the codec is known
truemotion2: check size before GetBitContext initialisation
avio: Only do implicit network initialization for network protocols
avio: Add an URLProtocol flag for indicating that a protocol uses network
adpcm: ADPCM Electronic Arts has always two channels
matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
fate: Add missing reference file from 9b4767e4.
mov: Support MOV_CH_LAYOUT_USE_DESCRIPTIONS for labeled descriptions.
4xm: Prevent buffer overreads.
mjpegdec: parse RSTn to prevent skipping other data in mjpeg_decode_scan
vp3: add fate test for non-zero last coefficient
vp3: fix streams with non-zero last coefficient
swscale: remove unused U/V arguments from yuv2rgb_write().
timer: K&R formatting cosmetics
lavf: cosmetics, reformat av_read_frame().
lavf: refactor av_read_frame() to make it easier to understand.
Report an error if pitch_lag is zero in AMR-NB decoder.
Revert "4xm: Prevent buffer overreads."
4xm: Prevent buffer overreads.
4xm: pass the correct remaining buffer size to decode_i2_frame().
...
Conflicts:
libavcodec/4xm.c
libavcodec/mjpegdec.c
libavcodec/truemotion2.c
libavformat/ipmovie.c
libavformat/mov_chan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The mpeg4 video, H264 and VC-1 parser hold (directly or indirectly)
a MpegEncContext in their private context. Since they do not call the
common mpegvideo init function slice_context_count has explicitly set
to 1.
Prevents a null pointer dereference in the h264 parser and fixes
bug 193.
Author: Mans Rullgard <mans@mansr.com>
Date: Sun Dec 11 21:41:59 2011 +0000
x86: cabac: replace explicit memory references with "m" operands
This replaces the explicit offset(reg) memory references with
"m" operands for the same locations. As a result, one fewer
register operand is needed for these inline asm statements.
This change appears to have broken compilation on darwin, and subsequent
fixes by martin (which did not fix compilation) removed the register
advantage, thus this change seems not a good idea to keep.
See: http://fate.ffmpeg.org/log.cgi?time=20120103122446&log=compile&slot=i386-darwin-llvm-gcc-4.2.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The implicit network initialization is set to be removed in the
future, but is kept for compatibility. By not doing the implicit
initialization for non-network protocols, we avoid the warning
about avformat_network_init() not being called for these, where
it really doesn't make much sense.
Signed-off-by: Martin Storsjö <martin@martin.st>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
This issue was discovered while decoding the FATE sample vqa/ws_snd.vqa.
For some unknown reason only audio decoding is tested by FATE for that file,
but not video.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Not all applications (e.g. MPlayer) set block_align, and
when using a different demuxer it might not even be
easily available.
So fall back to selecting mode based on bit rate as before
if block_align has not useful value.
It can't be worse than failing to decode completely.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* cus/stable:
ffplay: fix invalid wanted_channel_layout calculation
ffplay: honor SDL_AUDIO_CHANNELS and make sure to use SDL supported number of audio channels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Check explicitly if enough bits are left to prevent an infinite loop
when the bitstream buffer is not followed by zero-padding.
Based on patches by Michael Niedermayer <michaelni@gmx.at>.
frame_size is the number of bytes left in the packet, so if we are passing
buf-4 we can safely read frame_size+4 bytes.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (46 commits)
mtv: Make sure audio_subsegments is not 0
v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
avconv: add symbolic names for -vsync parameters
flvdec: Fix compiler warning for uninitialized variables
rtsp: Fix compiler warning for uninitialized variable
ulti: convert to new bytestream API.
swscale: Use standard multiple inclusion guards in ppc/ header files.
Place some START_TIMER invocations in separate blocks.
v4l2: list available formats
v4l2: set the proper codec_tag
v4l2: refactor device_open
v4l2: simplify away io_method
v4l2: cosmetics
v4l2: uniform and format options
v4l2: do not force interlaced mode
avio: exit early in fill_buffer without read_packet
vc1dec: fix invalid memory access for small video dimensions
rv34: fix invalid memory access for small video dimensions
rv34: joint coefficient decoding and dequantization
avplay: Don't call avio_set_interrupt_cb(NULL)
...
Conflicts:
Changelog
avconv.c
doc/APIchanges
doc/indevs.texi
libavcodec/adxenc.c
libavcodec/dnxhdenc.c
libavcodec/h264.c
libavdevice/v4l2.c
libavformat/flvdec.c
libavformat/mtv.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Thu Nov 3 22:38:10 2011 +0100
lavf: fix null pointer dereference in rdt
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is no longer needed and causes various problems with RTSP
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes compilation failures related to START_TIMER/STOP_TIMER macros and
-Werror=declaration-after-statement. START_TIMER declares variables and thus
may not be placed after statements outside of a new block.
Fixes an invalid free() with ass in avi. The sample in bug 98 passes
parts of AVPacket.data as buffer for the AVIOContext. Since the packet
is quite large fill_buffer tries to reallocate the buffer before doing
nothing. Fixes bug 98.
For small video dimensions, these calculations of the upper bound
for pixel access may have a negative result. Using an unsigned
comparison to bound a potentially negative value only works if
the greater operand is non-negative. Fixed by doing edge emulation
when the upper bound is probably negative, everywhere that this
pattern appears.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
For small video dimensions calculations of the upper bound for pixel
access may result in negative value. Using an unsigned comparison
works only if the greater operand is non-negative. This is fixed by
doing edge emulation explicitly for such conditions.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Perform dequantization while decoding coefficients instead of performing it
on the entire coefficients buffer.
Since quantized coefficients are very sparse, this usually causes a small
speedup. Speedup of around 1% on Panda board compared to the removed here
neon code. Global speedup is probably around 3%.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Since we don't use avio_set_interrupt_cb for interrupt callbacks,
we don't need to call it to reset the interrupt cb either.
This avoids a warning about use of deprecated functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes compilation failures related to START_TIMER/STOP_TIMER macros and
-Werror=declaration-after-statement. START_TIMER declares variables and thus
may not be placed after statements outside of a new block.
* qatar/master:
fate: add dxtory test
adx_parser: rewrite.
adxdec: Validate channel count to fix a division by zero.
adxdec: Do not require extradata.
cmdutils: K&R reformatting cosmetics
alacdec: implement the 2-pass prediction type.
alacenc: implement the 2-pass prediction type.
alacenc: do not generate invalid multi-channel ALAC files
alacdec: fill in missing or guessed info about the extradata format.
utvideo: proper median prediction for interlaced videos
lavu: bump lavu minor for av_popcount64
dca: K&R formatting cosmetics
dct: K&R formatting cosmetics
lavf: flush decoders in avformat_find_stream_info().
win32: detect number of CPUs using affinity
Add av_popcount64
snow: Restore three mistakenly removed casts.
Conflicts:
cmdutils.c
doc/APIchanges
libavcodec/adx_parser.c
libavcodec/adxdec.c
libavcodec/alacenc.c
libavutil/avutil.h
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
Fix audio output
Suppress dumping of residues buffer
Use quantizer value read from bitstream
Cosmetics: Remove two empty lines and realign some code
Reset acfilter_prevvalues buffer in clear_codec_buffers()
Fix AC filter buffers and AC filter reversion
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code ended in multiple different infinite
loops. See stl_ten_1_big.sfd as example with and without zzuf
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
The original code wasn't taking into account the fact that linesize may not equal the frame's width. This is to correct that.
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegenc: use avctx->slices as number of slices
v410enc: fix undefined signed left shift caused by integer promotion
Release notes: mention cleaned up header includes
fix Changelog file
Fix a bunch of typos.
Drop some pointless void* return value casts from av_malloc() invocations.
wavpack: fix typos in previous cosmetic clean-up commit
wavpack: cosmetics: K&R pretty-printing
avconv: remove the 'codec framerate is different from stream' warning
wavpack: determine sample_fmt before requesting a buffer
bmv audio: implement new audio decoding API
mpegaudiodec: skip all channels when skipping granules
mpegenc: simplify muxrate calculation
Conflicts:
Changelog
avconv.c
doc/RELEASE_NOTES
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpegaudiodec.c
libavcodec/mpegvideo.c
libavformat/mpegenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Up until now, the decoder didn't output anything
in the data buffer. This fixes all the issues
related to sample format, removes leftover code
and actually outputs some audio to data buffer.
With this, the only sample we have can be played.
Seeking is still broken though.
Adds a new member to MpegEncContext to hold the number of used slice
contexts. Fixes segfaults with '-threads 17 -thread_type slice' and
fate-vsynth{1,2}-mpeg{2,4}thread{,_ilace} with --disable-pthreads.
* qatar/master:
avconv: make -frames work for all types of streams, not just video.
bfi: K&R cosmetics
bgmc: K&R cleanup
rawdec: Set start_time to 0 for raw audio files.
Detect 'yuv2' as rawvideo also in avi.
rawdec: propagate pict_type information to the output frame
rawdec: Support more QT 1bpp rawvideo files.
avconv: free bitstream filters
threads: limit the number of automatic threads to MAX_AUTO_THREADS
avplay: K&R cleanup
fate: use rgb24 as output format for dfa tests
threads: set thread_count to 1 when thread support is disabled
threads: introduce CODEC_CAP_AUTO_THREADS and add it to libx264
Conflicts:
ffplay.c
libavcodec/avcodec.h
libavcodec/pthread.c
libavcodec/version.h
tests/ref/fate/dfa1
tests/ref/fate/dfa10
tests/ref/fate/dfa11
tests/ref/fate/dfa2
tests/ref/fate/dfa3
tests/ref/fate/dfa4
tests/ref/fate/dfa5
tests/ref/fate/dfa6
tests/ref/fate/dfa7
tests/ref/fate/dfa8
tests/ref/fate/dfa9
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The fate-h264-bsf-mp4toannexb failures were caused by an integer
overflow of the unneeded multiplication.
Inspired by patch by: Michael Niedermayer <michaelni@gmx.at>
The extra thread added in {frame_}*thread_init was not taken into
account. Explicitly sets thread_count to 1 if only one CPU core was
detected. Also fixes two typos in comments.
Palette is as supposed in native endianness. Converting the pal8 output
to rgb24 is thus necessary for identical CRCs on big and little endian
systems.
Some external codecs have their own code to determine the best number
of threads. This number is not necessary the number of cpu cores.
Thread_count will be only 0 if the codec has CODEC_CAP_AUTO_THREADS.
* qatar/master:
FATE: add tests for dfa
mpegaudiodec: fix seeking.
mpegaudiodec: fix compilation when testing the unchecked bitstream reader
threads: add sysconf based number of CPUs detection
threads: always include necessary headers for number of CPUs detection
threads: default to automatic thread count detection
Changelog: restore version <next> header
cook: K&R formatting cosmetics
Conflicts:
Changelog
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The safe bitstream reader does not allow using skip_bits_long() to seek to a
point before the start of the buffer, which was needed by the mp3 decoder.
This change instead calculates the start point of the first valid granule and
skips to that position.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Since the conditions for the actual usage are more specific a less
preferred method can be used. This would cause compilation errors
because necessary headers are not included.
This works around issues arising from inputs that claim to have a
filesize of 0.
Reported-by: buzz_
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Port MPlayer tinterlace filter from MPlayer, with some ideas taken
from the FFmbc/libavfilter port, with the following main differences:
* added support for full-scale YUVJ formats
* added support for YUVA420P
* request_frame() on the filter is forced to return a frame
* some code factorization (related to the copy_picture_fields() function)
* fixed black padding values for mode 3
Rationale: avfilter_copy_frame_props() was already defined in
libavfilter/avcodec.h, and keeping the lavc/lavfi API glue localized in a
specific file should ease maintainance and help the ones which use
libavfilter without depending on libavcodec.
This has been fixed differently and its revert avoids error messages
when decoding xx.flv. This also reduces the difference to qatar.
This reverts commit 5a2b3f3a52.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: whitespace cosmetics
fate: split off video codec FATE tests into their own file
fate: split off audio codec FATE tests into their own file
fate: split off Electronic Arts codec FATE tests into their own file
fate: split off QuickTime codec FATE tests into their own file
fate: split off voice codec FATE tests into their own file
fate: split off demuxer FATE tests into their own file
cosmetics: Drop unnecessary parentheses around return values.
fate: drop pointless _audio and _video suffixes from xan tests
qt-faststart: K&R reformatting; fix comment typos
FATE: Add test for H.264 MP4->annex.B bitstream filter.
Conflicts:
ffplay.c
tests/fate.mak
tests/fate/h264.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/qtrle.mak
tests/fate/real.mak
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This only happens for a "back" value of 0 which is invalid anyway,
but lcldec does not properly validate input.
Also extend the documentation to specify valid values.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
fate: split off DPCM codec FATE tests into their own file
fate: split off PCM codec FATE tests into their own file
libvorbis: K&R reformatting cosmetics
libmp3lame: K&R formatting cosmetics
fate: Add a video test for xxan decoder
mpegvideo_enc: K&R cosmetics (line 1000-2000).
avconv: K&R cosmetics
qt-faststart: Fix up indentation
indeo4: remove two unused variables
doxygen: cleanup style to support older doxy
fate: add more tests for VC-1 decoder
applehttpproto: Apply the same reload interval changes as for the demuxer
applehttp: Use half the target duration as interval if the playlist didn't update
applehttp: Use the last segment duration as reload interval
lagarith: add decode support for arith rgb24 mode
Conflicts:
avconv.c
libavcodec/libmp3lame.c
libavcodec/mpegvideo_enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 9d6b2077b2.
This error is annoying while debugging, and if someone disables it for
convenience, it raises the odds of leaving ffmpeg unbuildable by default.
According to draft-pantos-http-live-streaming-07, 6.3.4,
the duration of the last media segment in the playlist
should be used as initial minimum reload delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current default PES packet size, and very small audio bitrates,
audio packet duration gets too long. For players, which wait for a whole
audio packet (or more) it takes a very long time to start playing sound.
For 24kbps audio, one PES packet is about 1 second long. On Motorola STBs,
we observe about 3 second delay before the playback starts with the
default setting.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Do not assume the audio packets being always smaller than
DEFAULT_PES_PAYLOAD_SIZE.
Signed-off-by: Jindřich Makovička <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
mpegvideo_enc: K&R cosmetics
doxygen: remove unreplaced variables from custom header and footer
threads: test for sys/param.h and include it for sysctl on OpenBSD
v4l2: remove unneded linux specific asm/types.h include
x86: Fix constraints for decode_significance*_x86
Conflicts:
libavcodec/mpegvideo_enc.c
libavdevice/v4l2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing code expected a palette buffer holding 256 uint32_t's allocated in the data[1] field of the AVFrame structure, but data[1] was NULL. The bug is fixed by using a fixed local array (palette256) to hold the palette instead.
This solves http://ffmpeg.org/trac/ffmpeg/ticket/833
Signed-off-by: Frank Vernaillen <fr_ve@hotmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_samples_alloc() behavior changed in bbb46f3ec, resulting in random
data filling the data[] and linesize[] arrays of the returned
AVFilterBufferRef, which was resulting in wrong behavior in case of code
checking on data[i] nullity.
In particular fixes crash in avfilter_filter_samples():
for (i = 0; samplesref->data[i]; i++)
memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]);
and correctly fills the linesize[] array for planar data.
Originally, prior to 8742a4ff8, the caller code was compiled
within this condition:
ARCH_X86 && HAVE_7REGS && HAVE_EBX_AVAILABLE && !defined(BROKEN_RELOCATIONS)
Since HAVE_7REGS is defined as
(ARCH_X86_64 || (HAVE_EBX_AVAILABLE && HAVE_EBP_AVAILABLE))
the subcondition HAVE_7REGS && HAVE_EBX_AVAILABLE is equal
to HAVE_7REGS (for 32 bit at least). The correct simplification
of the original condition thus is HAVE_7REGS, not
HAVE_EBX_AVAILABLE.
This fixes compilation in some cases where HAVE_EBP_AVAILABLE = 0
and HAVE_EBX_AVAILABLE = 1.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
fate: split off vqf/twinvq FATE tests into their own file
fate: split off mpc FATE tests into their own file
fate: split off libavcodec FATE tests into their own file
fate: split off Microsoft codec FATE tests into their own file
fate: group all VP* codec FATE tests together in one file
swscale: prevent invalid writes in packed_16bpc_bswap
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ID3v2.4 allows for zlib compressed tags, but libavformat skips them.
Implement code to inflate compressed tags.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avs: call release_buffer() at the end.
Add minor bumps and APIchanges entries for lavc/lavfi changes.
mpegvideo.c: K&R formatting and cosmetics.
avconv: avoid memcpy in vsrc_buffer when possible.
avconv: implement get_buffer()/release_buffer().
lavfi: add a new function av_buffersrc_buffer().
lavfi: add avfilter_copy_frame_props()
lavc: add format field to AVFrame
lavc: add width and height fields to AVFrame
lavc: add a sample_aspect_ratio field to AVFrame
doxy: add website-alike style to the html output
FAQ: add an entry for common error when using -profile
Conflicts:
avconv.c
cmdutils.c
doc/APIchanges
libavcodec/avcodec.h
libavcodec/mpegvideo.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/Makefile
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/src_movie.c
libavfilter/vsrc_buffer.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The format is a per-frame property, having it in AVFrame simplify the
operation of extraction of that information, since avoids the need to
access the codec/stream context.
width and height are per-frame properties, setting these values in
AVFrame simplify the operation of extraction of that information,
since avoids the need to check the codec/stream context.
The sample aspect ratio is a per-frame property, so it makes sense to
define it in AVFrame rather than in the codec/stream context.
Simplify application-level sample aspect ratio information extraction,
and allow further simplifications.
When generating the .dep files for .texi sources, verbatim includes
(@verbatiminclude) should also be taken into account.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The FATE documentation depends on the mentioned file. But that
did break out of tree builds because the file was not found.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
Perform dequantization of channel coefficients
Perform inverse inter-channel decorrelation and ac-filter
Implement revert_inter_ch_decorr() and revert_acfilter()
Enable inverse-MCLMS filter
Fix inverse-MCLMS filtering routines
Do not update buffers in case no speed change is necessary
Use int for channel_coeffs instead of int16_t
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vp3dec: Check coefficient index in vp3_dequant()
svq1dec: call avcodec_set_dimensions() after dimensions changed.
Prepare for 0.8_beta1 snapshot release
threads: check defines before using them in automatic thread detection
pthread: include sys/types.h before sys/sysctl.h
4xm: remove unused variables.
h264: Fix a possible overread in decode_nal_units()
allfilters: fix type of avfilter_vsrc_buffer.
w32thread: call ResetEvent() in pthread_cond_broadcast().
Conflicts:
Changelog
RELEASE
doc/RELEASE_NOTES
libavcodec/pthread.c
libavcodec/vp3.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Based on a patch by Michael Niedermayer <michaelni@gmx.at>
Fixes NGS00145, CVE-2011-4352
Found-by: Phillip Langlois
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master:
build: fix standalone compilation of OMA muxer
build: fix standalone compilation of Microsoft XMV demuxer
build: fix standalone compilation of Core Audio Format demuxer
kvmc: fix invalid reads
4xm: Add a check in decode_i_frame to prevent buffer overreads
adpcm: fix IMA SMJPEG decoding
options: set minimum for "threads" to zero
bsd: use number of logical CPUs as automatic thread count
windows: use number of CPUs as automatic thread count
linux: use number of CPUs as automatic thread count
pthreads: reset active_thread_type when slice thread_init returrns early
v410dec: include correct headers
Drop ALT_ prefix from BITSTREAM_READER_LE name.
lavfi: always build vsrc_buffer.
ra144enc: zero the reflection coeffs if the filter is unstable
sws: readd PAL8 to isPacked()
mov: Don't stick the QuickTime field ordering atom in extradata.
truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
Conflicts:
configure
libavcodec/4xm.c
libavcodec/avcodec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavformat/Makefile
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When no data was available both the buffer thread as well as
the main thread would block in select(), when data becomes
available both should move forward and as data is read in the
buffer thread the main thread would block in select() later
the read data was put in the fifo but the main thread still
would be blocked in select() until either the timeout or
another packet would come in.
This is solved in this commit by using a mutex and a condition
variable
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Prior to this a X bytes write could be seen as less than X bytes being
available if the check was done at an unfortunate moment.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use sched_getaffinity to determine the number of logical CPUs.
Limits the number of threads to 16 since slice threading of H.264
seems to be buggy with more than 16 threads.
This file does not use anything from get_bits.h but needs
intreadwrite.h.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* tjoppen/fuzz_fixes:
mxfdec: Don't crash in mxf_packet_timestamps() if current_edit_unit overflows
mxfdec: Zero nb_ptses in mxf_compute_ptses_fake_index()
mxfdec: Sanity check PreviousPartition
mxfdec: Never seek back in local sets and KLVs
mxfdec: Move the current_partition check inside mxf_read_header()
mxfdec: Fix infinite loop in mxf_packet_timestamps()
mxfdec: Check url_feof() in mxf_read_local_tags()
mxfdec: Check for NULL component
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A list of formats may have been dynamically created by the calling code,
and thus should not be referenced by the sink buffer context.
Avoid possible invalid data reference.
The 'fiel' atoms can be found in H.264 tracks clobbering the extradata.
MJPEG supports non field based extradata, and this data should be
preserved when copying.
This way ffmpeg can be distinguished from the fork by a user
application or a encoded file by a decoder.
The highest value micro had, in the past, that i could find, was 6
thus 100 should be safe.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
asfdec: add side data to ASFStream packet instead of output packet.
idroqdec: set AVFMTCTX_NOHEADER and create streams as they occur.
nellymoserdec: Indicate that the decoder can handle changed parameters
libavcodec: Apply parameter change side data when decoding audio
flvdec: Add param change side data if the sample rate or channels have changed
libavformat: Add a utility function for adding parameter change side data
libavcodec: Define a side data type for parameter changes
aacdec: Handle new extradata passed as side data
flvdec: Export new AAC/H.264 extradata as side data on the next packet
libavcodec: Define a side data type for new extradata
flacdec: skip all track indices at once instead of looping.
mxf: Add PictureEssenceCoding UL for V210.
mxfdec: consider QuantizationBits between 17 and 24 to be pcm_s24*
mxfenc: Add support for MPEG-2 MP@HL-14 in mxf container.
mxf: H.264/MPEG-4 AVC Intra support
configure: Show whether the safe bitstream reader is enabled
x86: Tighten register constraints for decode_significance*_x86.
Replace Subversion revisions in comments by Git hashes.
h264_cabac: synchronize decode_significance_*_x86 conditionals
w32threads: wait for the waked thread in pthread_cond_signal.
...
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/flvdec.c
libavformat/utils.c
tests/ref/lavfi/pixdesc
tests/ref/lavfi/pixfmts_copy
tests/ref/lavfi/pixfmts_null
tests/ref/lavfi/pixfmts_scale
tests/ref/lavfi/pixfmts_vflip
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes demuxing of file where the first packet is not audio. Such files
are generated by our idroq muxer. It also fixes demuxing of audio only
idroq files.
Also define a codec capability for codecs that can handle
parameters changed externally between decoded packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Compared to just overwriting the old extradata, this has the
advantage of letting the decoder know exactly when the
extradata changed (otherwise it is changed immediately when the
new extradata packet is demuxed, even if there's old queued packets
awaiting to be decoded). This makes it easier for decoders to
actually react to the change, so they won't have to inspect
the extradata for each packet to see if it might have changed.
This works when sequentially playing a file with sample rate
changes, but if seeking past a new extradata packet in the
file, it obviously doesn't work properly. That case doesn't
work in flash player either, so it's probably ok not to handle
it.
Signed-off-by: Martin Storsjö <martin@martin.st>
Support Main Profile at High 1440 Level in MXF container,
using essence coding label from SMPTE RDD 9, table 6.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* tjoppen/mxf_fixes_20111220:
mxfdec: Sanity-check SampleRate
mxfdec: Make sure mxf->nb_index_tables > 0 in mxf_packet_timestamps()
mxfdec: Remove unused variables
mxfdec: Make sure x < index_table->nb_ptses
mxfdec: Ignore the last entry in Avid's index table segments
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specially crafted files can lead the parsing code to take too long.
We fix a lot of these problems by not allowing local tags to extend past the
end of the set and not allowing other KLVs to be read past the end of
themselves.
On 32-bit OS X with gcc 4.0/4.2 and shared libraries enabled, the ebx register
is not available, but required to assemble the functions.
This reverts commit 8742a4f to a simplified version of the original constraints.
This fixes a deadlock VLC triggered with multithreaded decoding. The
wait forces one of the current waiters to wake and not the thread
which calls pthread_cond_signal() itself.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
lavc: always align height by 32 pixel
raw: add 10bit YUV definitions
nut: support 10bit YUV
mpegvideo_enc: separate declarations and statements
oma: make header compile standalone
vp3: Reorder some functions to fix VP3 build with Theora disabled.
build: fix standalone compilation of ADX encoder
build: fix standalone compilation of ADPCM decoders
build: fix standalone compilation of mpc7/mpc8 decoders
4xm: Use bytestream2 functions to prevent overreads
bytestream: add a new set of bytestream functions with overread checking
mpegts: Suppress invalid timebase warnings on DMB streams.
mpegts: Fix typo in handling sections in the PMT.
vc1dec: Use the right pointer type for the tmp pointer
Conflicts:
libavcodec/4xm.c
libavcodec/utils.c
libavcodec/vc1dec.c
libavcodec/vp3.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some libavifilter tests use NUT as output even if the produced
files were not decodable. The support for 10bit introduced in
432f0e5b7d and 91b1e6f0c changed the hashes.
This moves declarations without initialisers or with constant
initialisers to the start of a block, and adds do {} while(0)
around some macros, thus allowing declarations within them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Only the OPAtom demuxing logic is guaranteed to have index tables, meaning OP1a
files that lack an index would cause SIGSEGV.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Interlaced content for most codec requires it.
This patch is a stop-gap pending a serious rework to support
codecs with non 16 pixel macroblocks.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Also create a plain text (.txt) file from fate.texi if the makeinfo
program is available.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The fate.txt file is ported to texinfo format. Therefore the
fate.txt is renamed to fate.texi. The contents of the already
existing fate.texi file are discarded.
However there should be no loss of information. If you find
anything missing, please report.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When generating the .dep files for .texi sources, verbatim includes
(@verbatiminclude) should also be taken into account.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The last entry is the total size of the essence container.
Previously a TemporalOffset error would be logged, even though segments like
these are expected.
* qatar/master:
h264: clear trailing bits in partially parsed NAL units
vc1: Handle WVC1 interlaced stream
xl: Fix overreads
mpegts: rename payload_index to payload_size
segment: introduce segmented chain muxer
lavu: add AVERROR_BUG error value
avplay: clear pkt_temp when pkt is freed.
qcelpdec: K&R formatting cosmetics
qcelpdec: cosmetics: drop some pointless parentheses
x86: conditionally compile dnxhd encoder optimizations
Revert "h264: skip start code search if the size of the nal unit is known"
swscale: fix formatting and indentation of unscaled conversion routines.
h264: skip start code search if the size of the nal unit is known
cljr: fix buf_size sanity check
cljr: Check if width and height are positive integers
Conflicts:
libavcodec/cljr.c
libavcodec/vc1dec.c
libavformat/Makefile
libavformat/mpegtsenc.c
libavformat/segment.c
libswscale/swscale_unscaled.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Trailing bits are likely to be non-zero if the NAL unit is truncated.
Clearing the bits make overreads of the bitstream less likely in this
case. Fixes playback of
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4 which
has a forbidden byte sequence of 0x00 0x00 0x00 in it SPS.
* qatar/master:
APIchanges: fill in revision for AVFrame.age deprecation
avcodec: deprecate AVFrame.age
4xm: remove unneeded check for remaining unused data.
lavf: force threads to 1 in avformat_find_stream_info()
swscale: fix overflows in vertical scaling at top/bottom edges.
lavf: add OpenMG audio muxer.
omadec: split data that will be used in the muxer to a separate file.
lavf: rename oma.c -> omadec.c
tmv decoder: set correct pix_fmt
Conflicts:
Changelog
doc/APIchanges
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/oma.c
libavformat/version.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Start code emulation prevention is only required in Annex B bytestream
packed NAL units. For other coding formats the size is already known.
Looking for a start code prefix can result in false positives like in
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4
which has a false positive in the SPS.
Width and height might get passed as 0 and would cause floating point
exceptions in decode_frame.
Fixes bugzilla #149
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This was intended as an optimisation for skipped blocks in MPEG2
P-frames and never used elsewhere. Removing this "optimisation"
speeds up MPEG2 decoding by 1-2% (ARM Cortex-A9).
Signed-off-by: Mans Rullgard <mans@mansr.com>
* tjoppen/proper_mxf_track_linking:
mxfdec: Don't parse slices or DeltaEntryArrays
mxfdec: Remove dead/useless code
mxfdec: Hybrid demuxing/seeking solution
mxfdec: Add mxf_edit_unit_absolute_offset()
mxfdec: Replace zero IndexDurations with st->duration
mxfdec: Add "fake" index to MXFIndexTable to assist seeking
mxfdec: Add MXFIndexTables
mxfdec: Move mxf_read_packet*() near the bottom of the file
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code ended in multiple different infinite
loops. See stl_ten_1_big.sfd as example with and without zzuf
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes integer multiplication overflows in RGB48 output
(vertical) scaling as detected by IOC. What happens is that for
certain types of filters (lanczos, spline, bicubic), the
intermediate sum of coefficients in the middle of a filter can
be larger than the fixed-point equivalent of 1.0, even if the
final sum is 1.0. This is fine and we support that.
However, at frame edges, initFilter() will merge the coefficients
for the off-screen pixels into the top or bottom pixel, such as
to emulate edge extension. This means that suddenly, a single
coefficient can be larger than the fixed-point equivalent of
1.0, which the vertical scaling routines do not support.
Therefore, remove the merging of coefficients for edges for
the vertical scaling filter, and instead add edge detection
to the scaler itself so that it copies the pointers (not data)
for the edges (i.e. it uses line[0] for line[-1] as well), so
that a single coefficient is never larger than the fixed-point
equivalent of 1.0.
Previously the decoder only worked if the user had set avctx->pix_fmt
manually. For some reason the libavformat tmv demuxer sets this, so
the problem was not visible in avplay etc.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This should be replaced by a more appropriate error code of course but
we should not leave compilation broken until that is decided.
Found-by: jb
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
that way qatar maintains the code for me and i dont need to resolve conflicts.
If someone wants the a32 reader back, only thing you need to do is maintain
it, i would be happy to have it back, iam just not volunteering to maintain
it due to lack of time.
Based on: a1e98f198e by Mans Rullgard.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Many of the test programs directly access internal symbols not
exported from the shared libraries. This allows tests to run
when configured with shared libraries.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes the same overflow as in the RGB48/16-bit YUV scaling;
some filters can overflow both negatively and positively (e.g.
spline/lanczos), so we bias a signed integer so it's "half signed"
and "half unsigned", and can cover overflows in both directions
while maintaining full 31-bit depth.
Signed-off-by: Mans Rullgard <mans@mansr.com>
No difference in PSNR or bitrate in the printed precission with the matrix lobby scene at 322x242
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
We're shifting individual components (8-bit, unsigned) left by 24,
so making them unsigned should give the same results without the
overflow.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
For certain types of filters where the intermediate sum of coefficients
can go above the fixed-point equivalent of 1.0 in the middle of a filter,
the sum of a 31-bit calculation can overflow in both directions and can
thus not be represented in a 32-bit signed or unsigned integer. To work
around this, we subtract 0x40000000 from a signed integer base, so that
we're halfway signed/unsigned, which makes it fit even if it overflows.
After the filter finishes, we add the scaled bias back after a shift.
We use the same trick for 16-bit bpc YUV output routines.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The buffer splicing relies on the bitstream reader over-reading
the end of the buffer as declared in init_get_bits(), although
more data is actually present. Manually moving the bitstream
boundary after init_get_bits() allows this to work as expected.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The sample has an incomplete last frame. Decoding it is pointless.
The garbage produced was changed by the bitstream reader now
protecting against over-reads.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When turned on, H264/CAVLC gets ~15% (CVPCMNL1_SVA_C.264) slower for
ultra-high-bitrate files, or ~2.5% (CVFI1_SVA_C.264) for lower-bitrate
files. Other codecs are affected to a lesser extent because they are
less optimized; e.g., VC-1 slows down by less than 1% (all on x86).
The patch generated 3 extra instructions (cmp, cmovae and mov) per
call to get_bits().
The performance penalty on ARM is within the error margin for most
files, up to 4% in extreme cases such as CVPCMNL1_SVA_C.264.
Based on work (for GCI) by Aneesh Dogra <lionaneesh@gmail.com>, and
inspired by patch in Chromium by Chris Evans <cevans@chromium.org>.
* qatar/master:
get_bits: remove A32 variant
avconv: support stream specifiers in -metadata and -map_metadata
wavpack: Fix 32-bit clipping
wavpack: Clip samples after shifting
h264: don't drop B-frames after next keyframe on POC reset.
get_bits: remove useless pointer casts
configure: refactor lists of tests and components into variables
rv40: NEON optimised weak loop filter
mpegts: replace some magic numbers with the existing define
swscale: add unscaled packed 16 bit per component endianess conversion
Conflicts:
libavcodec/get_bits.h
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The A32 bitstream reader variant is only used on ARMv5 and for
Prores due to the larger bit cache this decoder requires.
In benchmarks on ARMv5 (Marvell Sheeva) with gcc 4.6, the only
statistically significant difference between ALT and A32 is
a 4% advantage for ALT in FLAC decoding. There is thus no (longer)
any reason to keep the A32 reader from this point of view.
This patch adds an option to the ALT reader increasing the bit
cache to 32 bits as required by the Prores decoder. Benchmarking
shows no significant change in speed on Intel i7. Again, the
A32 reader fails to justify its existence.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the case that (frame_flags & 0x03) == 3, hybrid_maxclip
may have had a signed integer overflow.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It doesn't make much sense to clip pre-shift,
nor is it correct for proper decoding.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The keyframe after a POC reset may not be the first to be returned to
the user. Therefore, don't reset the expected next POC once we return
a keyframe to the user, but once we know that the next frame in the
return-queue is a keyframe.
width and height might get passed as 0 and would cause floating point
exceptions in decode_frame.
Fixes bugzilla #149
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we don't
use delta entries or slices, only StreamOffsets.
OPAtom seeking basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
This fixes ticket #746.
This changes mxf_compute_ptses() to be used for MXFIndexTable, and also adds
code for computing the fake index to it.
This also temporarily disables PTS computation. A future patch will restore it.
XBMC's configure script checks for this function in installed
libavcodec.so to determine VDPAU support.
Fixes ticket #762 reported by Christian Marillat
* qatar/master:
movenc: Rudimentary IODs support.
v410enc: fix output buffer size check
v410enc: include correct headers
fate: add -pix_fmt rgb48le to r210 test
flvenc: Support muxing 16 kHz nellymoser
configure: refactor list of programs into a variable
fate: add r210 decoder test
fate: split off Indeo FATE tests into their own file
fate: split off ATRAC FATE tests into their own file
fate: Add FATE tests for v410 encoder and decoder
ARM: fix external symbol refs in rv40 asm
westwood: Make sure audio header info is present when parsing audio packets
libgsm: Reset the MS mode of GSM in the flush function
libgsm: Set options on the right object
ARM: dca: disable optimised decode_blockcodes() for old gcc
Conflicts:
configure
libavformat/movenc.c
libavformat/movenc.h
tests/fate2.mak
tests/ref/acodec/alac
tests/ref/vsynth1/mpeg4
tests/ref/vsynth2/mpeg4
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Audio header information might get scrambled and would not parse,
yet wsqva_read_packet would try to parse audio packets causing
segfaults such as floating point exception.
Fixes bugzilla #141.
Signed-off-by: Martin Storsjö <martin@martin.st>
The mode is set in libgsm_decode_init, but the decoder
object is simply destroyed and recreated in the flush
function - therefore the mode has to be set again.
This fixes playback using the libgsm_ms decoder in avplay.
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts commit 4f820131fa.
It is better to abort() than to have remotly exploitable arbitrary code
execution bugs. Even more so that this abort has never been triggered
by any input people threw at it.
If after more extensive testing its removial is found safe we can remove
the abort() later.
* qatar/master: (23 commits)
applehttp: Properly clean up if unable to probe a segment
applehttp: Avoid reading uninitialized memory
fate: Replace misleading "aac" in the name of an ADTS test with "adts".
fate: Drop pointless "-an" from pictor test command.
fate: split off image codec FATE tests into their own file
fate: split off WMA codec FATE tests into their own file
fate: split off lossless video and audio FATE tests into their own files
fate: split off qtrle codec FATE tests into their own file
fate: split off Ut Video codec FATE tests into their own file
fate: split off screen codec FATE tests into their own file
fate: split off Real Inc. codec FATE tests into their own file
fate: split off AC-3 codec FATE tests into their own file
mpegvideo: remove abort() in ff_find_unused_picture()
rv40: NEON optimised loop filter strength selection
rv40: rearrange loop filter functions
configure: cosmetics: sort some lists where appropriate
swscale_mmx: drop no longer required parameters from VSCALEX macros
swscale: Mark yuv2planeX_8_mmx as MMX2; it contains MMX2 instructions.
build: conditionally compile x86 H.264 chroma optimizations
v410 encoder and decoder
...
Conflicts:
Changelog
configure
doc/developer.texi
doc/general.texi
libavcodec/arm/asm.S
libavcodec/avcodec.h
libavcodec/v410dec.c
libavcodec/v410enc.c
libavcodec/version.h
libavcodec/x86/Makefile
libavcodec/x86/dsputil_mmx.c
libswscale/x86/swscale_mmx.c
tests/Makefile
tests/fate2.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
ffplay: clear pkt_temp when pkt is freed.
ffplay: Fix got_frame type.
ffplay: add 10 minute seek support to ffplay
ffplay: force setting video mode on fullscreen toggle
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids a segfault if the probe function wasn't able to
determine the format.
The bug was found by Panagiotis H.M. Issaris.
Signed-off-by: Martin Storsjö <martin@martin.st>
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
A user who wishes to use default error concealment cannot set the
AV_EF_CRCCHECK flag because not every CRC in every format is a
reliable indicator of bitstream damage. In some formats crcrs
can be nonsensical in absence of any damage. We thus add the
AV_EF_CAREFUL flag in addition to the AV_EF_CRCCHECK flag to
allow a user to enable this reliable CRC check without having to
enable all CRC checks in all formats.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
ulti: Fix invalid reads
lavf: dealloc private options in av_write_trailer
yadif: support 10bit YUV
vc1: mark with ER_MB_ERROR bits overconsumption
lavc: introduce ER_MB_END and ER_MB_ERROR
error_resilience: use the ER_ namespace
build: move inclusion of subdir.mak to main subdir loop
rv34: NEON optimised 4x4 dequant
rv34: move 4x4 dequant to RV34DSPContext
aacdec: Use intfloat.h rather than local punning union.
Conflicts:
libavcodec/h264.c
libavcodec/vc1dec.c
libavfilter/vf_yadif.c
libavformat/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch is a generalization of what Michael Niedermayer
fixed in a single case.
The wmv8-drm fate test had been updated accordingly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
As far as I could see the only change is increased pos values,
which is as expected with additional metadata in the files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
It sets the supplied AVFormatContext pointer to NULL after freeing it,
which is safer and its name is consistent with other lavf functions.
Also deprecate av_close_input_file().
These indexes duplicate every entry and have the total size of the essence
container as the last entry.
This patch also computes the size of the packets when unknown.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
I thought it had to do with file offsets, but's actually the offset inside
the essence container.
In other words, unbreak multiple EditUnitByteCounts.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The change in 599b4c6ef didn't turn out to work properly on
i386 on OS X, where it broke building with PIC enabled.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit f1dba9e498)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The change in 599b4c6ef didn't turn out to work properly on
i386 on OS X, where it broke building with PIC enabled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Firstly, this test never worked as intended, always reporting
success. Secondly, bswap is available from 486 onward and can
thus be assumed present.
Signed-off-by: Mans Rullgard <mans@mansr.com>
With these changes, gcc 4.5 and later recognise it as a bswap
and use the proper instructions on ARM and x86. On x86, the
16-bit bswap is recognised from gcc 4.1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the function prototype match the argument of
AVCodecCntext.execute() and remove the cast hiding
this mismatch.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This replaces the explicit offset(reg) memory references with
"m" operands for the same locations. As a result, one fewer
register operand is needed for these inline asm statements.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When the buf and last pointers are equal, the FFSWAP() results
in an invalid call to memcpy() with same source and destination
on some targets. Although assigning a struct to itself is valid
C99, gcc does not check for this before calling memcpy().
See http://gcc.gnu.org/bugzilla/show_bug.cgi?id=32667
Signed-off-by: Mans Rullgard <mans@mansr.com>
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
I am not entirely sure the seek functionality is working correctly,
there are some strange cases like successful seek but no dts value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (21 commits)
Warn about avserver being broken.
avconv: drop code for special handling of avserver streams.
rawdec: don't set codec timebase.
lavf doxy: add muxing stuff to lavf_encoding group
lavf doxy: add demuxing stuff to lavf_decoding group
lavf doxy: expand/reword metadata API doxy.
lavf doxy: add installed headers to groups.
lavf doxy: add avio groups into the lavf_io group.
lavf doxy: rename lavf I/O group to lavf_io.
lavf doxy: add metadata docs to the main lavf group
ttadec: check channel count as read from extradata.
Add CLJR encoding and decoding regression tests
cljr: remove unused code
flacdec: Support for tracks in cuesheet metadata block
ptx: fix inverted check for sufficient data
flac muxer: fix writing of file header and STREAMINFO header from extradata
ptx: emit a warning on insufficient picture data
utvideo: add fate tests covering all codec variants
doc: update to refer to avconv
doc: remove some stale entries from the faq
...
Conflicts:
Changelog
avconv.c
doc/avconv.texi
doc/faq.texi
doc/ffplay.texi
doc/ffprobe.texi
doc/ffserver.texi
libavcodec/avcodec.h
libavcodec/cljr.c
libavformat/avformat.h
libavformat/riff.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If the sdp is generated before the rtp muxer is initialized
(e.g. as when called from the rtsp muxer), this has to be done,
otherwise the rtp muxer doesn't know that the input really is
in mp4 format.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an annex b bitstream is muxed into mov, the actual written
sample is reformatted to mp4 syntax before writing.
Currently, the RTP hints that copy data from the normal video
track, where the payload data might be offset compared to the
original sample that the RTP hinting used (when 3 byte
annex b startcodes have been converted into 4 byte mp4 format
startcodes).
Signed-off-by: Martin Storsjö <martin@martin.st>
It's broken and doesn't work anyway.
This patch means that avconv will ignore encoding options from the ffm
file and will instead use whatever is provided on the commandline as for
normal output.
It had become dead code when code was added to avoid
exporting audio and video codec id as metadata.
Untested due to lack of sample.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Return the whole packet as consumed in this case and not the size the
packet should have had. Move the insufficient data check into the for
condition to fix a ISO C90 error on bigendian.
* qatar/master:
isom: sort and pretty-print codec_movaudio_tags[]
isom: remove pointless comments in codec_movaudio_tags[]
isom: remove commented-out tag for vorbis
movenc: write 'chan' tag for AC-3 in MOV
mov: add support for reading and writing the 'chan' tag
audioconvert: add some additional channel and channel layout macros
audioconvert: change 7.1 "wide" layout to use side surround channels
movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
doc: update documentation to use avconv
doc: update demuxers section
doc: extend external library coverage
doc: split platform specific information
doc: port the git-howto to texinfo
doc: provide fallback css and customize @float
doc: document fate in a texinfo
doxy: change hue value to match our green
Conflicts:
doc/fate.txt
doc/ffserver.texi
doc/general.texi
doc/muxers.texi
doc/protocols.texi
doc/t2h.init
libavformat/isom.c
libavformat/mov.c
libavutil/avutil.h
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This implements reading the tag in the demuxer and adds support for writing it
in the muxer. Some example channel layout tables for muxing are included for
ac3, aac, and alac, but they are not utilized yet.
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
* qatar/master:
cljr: K&R cosmetics
cljr: return a more sensible value when encountering invalid headers
cljr: drop unnecessary emms_c() calls without MMX code
cljr: remove useless casts
cljr: group encode/decode parts under single ifdefs
cljr: remove stray semicolon
cljr: add missing return statement in decode_end()
doc: add pulseaudio to the input list
avconv: remove unsubstantiated comment
shorten: avoid abort() on unknown audio types
cljr: add encoder
build: merge lists of HTML documentation targets
tests/examples: Mark some variables only used within their files as static.
tests/tools/examples: Replace direct exit() calls by return.
x86 cpuid: set vendor union members separately
cljr: release picture at end of decoding
rv40: NEON optimised rv40 qpel motion compensation
Conflicts:
doc/examples/muxing.c
libavcodec/cljr.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* tjoppen/opatom_demuxing_and_seeking:
mxfdec: Index table driven demuxing and seeking
mxfdec: Compute packet offsets properly
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack
mxfdec: Parse more values in PartitionPack
mxfdec: Parse TemporalOffsets
mxfdec: av_dlog():ify 'no corresponding source package found'
mxfdec: Compute essence container offsets and lengths into mxf->partitions
mxfdec: Make mxf->partitions sorted by offset
mxfdec: Parse ThisPartition
mxfdec: Speed up metadata and index parsing
mxfdec: Make sure DataDefinition is consistent between material track and source track
mxfdec: Add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the case that the buffer sinks fifo contains frames which
would not show up on the poll from its input
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These values include KAGSize, HeaderByteCount and IndexByteCount.
The length of the pack itself is also stored, and KAGSize is sanity checked.
The FATE sample has KAGSize == 0, which is adjusted to 512.
Other bad KAGSizes are set to 1.
Thus we can check for 4 being available.
If the next block is skiped we need 4 bits to encode the skip. If the
next block is not skiped then we need 1 bit for the skip code and
at least 3 bits for the block.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This groups the encode/decode parts under single ifdefs and
eliminates the encode_init() function as it merely calls
common_init(). Also fix whitespace in moved code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The information is relevant, but under normal circumstances
it raises far too many false alarms.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Solaris Studio (suncc) has difficulty with filling in
members of a union. Instead, let's retrieve and store the
cpuid() results separately. This is still a compiler bug,
however this fix does not cause a regression on other platforms.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master:
drawtext: remove typo
pcm-mpeg: implement new audio decoding api
w32thread: port fixes to pthread_cond_broadcast() from x264.
doc: add editor configuration section with Vim and Emacs settings
dxva2.h: include d3d9.h to define LPDIRECT3DSURFACE9
avformat/utils: Drop unused goto label.
doxygen: Replace '\' by '@' in Doxygen markup tags.
cosmetics: drop some completely pointless parentheses
cljr: simplify CLJRContext
drawtext: introduce rand(min, max)
drawtext: introduce explicit draw/hide variable
rtmp: Use nb_invokes for all invoke commands
Conflicts:
libavcodec/mpegvideo.c
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In 1bpp mode, interpret skip&0x80 as "start a new line" instead of "go to next line", this is almost the same except for the first line which was always skipped before and caused to try to write an extra line at the end of the frame (ticket #226).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers aren't needlessly
parsed. This speeds up parsing large files a lot.
704af3e29c broke publishing
of rtmp streams, at least publishing to Wowza servers.
This changes all invoke commands to use nb_invokes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* shariman/wmall:
Fix lms_update()
Move num_lms reading out of a loop
Use correct value for range
Fix some int / int16_t / int32_t confusion
Implement revert_mclms() and associated functions
Fix two more int16_t vs. int confusion
Init s->cdlms[][].recent to order - 1
Add a size argument to dump_int_buffer()
Get rid of logging that are not required anymore
Fix some int vs. int16_t confusion
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The default of 9 gives different results on different FATE systems.
However the zlib test using compression level 6 works, so
try this instead.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Its checked a few lines below too.
The only difference is that empty atoms with size=0 will now get parsed too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The computed size doesn't contain the header size because it's already
skipped by incrementing total_size, but then it's skipped again in the
last line. The atom comes out 8 bytes short and the function
mov_read_chan() aborts the whole parsing process. I think the computed
size should be atom.size - total_size + 8.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov: Don't av_malloc(0).
avconv: only allocate 1 AVFrame per input stream
avconv: fix memleaks due to not freeing the AVFrame for audio
h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
misc Doxygen markup improvements
doxygen: eliminate Qt-style doxygen syntax
g722: Add a regression test for muxing/demuxing in wav
g722: Change bits per sample to 4
g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
api-example: update to use avcodec_decode_audio4()
avplay: use avcodec_decode_audio4()
avplay: use a separate buffer for playing silence
avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
mov: Allow empty stts atom.
doc: document preferred Doxygen syntax and make patcheck detect it
Conflicts:
avconv.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/version.h
libavformat/mov.c
tests/codec-regression.sh
tests/fate/h264.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
More specifically, PNG, v210, zlib and zmbv codecs.
zmbv needs vf_scale to be able to produce PAL8.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
While quality is bad, PAL8 support is needed to allow testing some
encoders that only support PAL8 input.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
For some of the code e.g. doing timing measurements there is no
real point in running regression testing on it, thus it should
not be counted against coverage.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Adds --enable-coverage to configure and a "coverage-html" make target.
The dependency stuff in the Makefile is a bit questionable, but the
best I could think of so far.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
malloc() is allowed to return NULL when zero is the argument. This
causes us to think malloc has failed and return AVERROR(ENOMEM). In
addition OS X malloc() returns an unfreeable non-NULL pointer for size
zero when alignment is greater than 16.
Earlier, bits per sample was defined as 8, since
bits_per_coded_sample was used to indicate whether to ignore
the lower bits of the codeword, having values 6, 7 or 8.
g722 encodes 2 samples into one byte codeword, therefore the
bits per sample is 4. By changing this, the generated timestamps
for streams encoded with g722 become correct.
This makes timestamp generation for g722 data correct (both when
encoding and when demuxing from raw g722 files).
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids using bits_per_coded_sample for this information.
bits_per_coded_sample should be 4 for this codec normally,
since two samples are encoded into one 8 bit codeword.
In principle, this might be info that needs to be passed from
a demuxer, and in that case, a private AVOption isn't the best
choice, but no such samples are available at the moment, so
that use case is purely theoretical at the moment.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
h264: fix frame reordering code.
fate: Add a test for the VBLE decoder
doc: break some long lines in developer.texi
drawtext: make x and y parametric
drawtext: manage memory allocation better
drawtext: refactor draw_text
doc: remove space between variable and post increment in example code
Conflicts:
doc/developer.texi
doc/filters.texi
libavcodec/h264.c
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When decoding lossy WavPack samples, they are supposed
to be clipped, in order to be decoded correctly.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
instead of when the 2nd stream has been found.
This isnt ideal as we will likely still like before miss a data stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is the usual weirdness with const/non-const pointers through a function.
See the prototype of strchr for a standard example.
The function had to be moved down, below the point where stdint.h is
indirectly included.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Introduce variables "E", "PHI", "PI", "main_w"/"W", "main_h"/"H",
"text_w"/"w", "text_h"/"h", "x", "y", "n" and "t" in line with
vf_overlay and refactor the code accordingly.
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
swresample_test.c:123:21: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement]
swresample_test.c:127:25: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/swscale.c:2744:40: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2745:41: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2746:41: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2747:78: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/x86/swscale_mmx.c:131:36: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/x86/swscale_mmx.c:132:37: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/x86/swscale_mmx.c:133:74: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add AV_NUM_DATA_POINTERS to simplify the bump transition.
This will allow for supporting more planar audio channels without having to
allocate separate pointer arrays.
Commit 035af99 made avconv always call an encoder when using the
null muxer. While useful for 2-pass encodes, it inadvertently
caused an extra memcpy of raw frames when decoding only.
This hack restores the old behaviour when only decoding while
allowing use of the null muxer with encoded streams as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change was dependent on a different patch that
never actually made it into FFmpeg, and it actually
ended up breaking builds.
This reverts commit 70cf7bb958.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When skipping over the extended header, take into account
that the size field has already been read. The extended header
also takes up space, so adjust total header length accordingly.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
For the standardized 8 kHz sample rate, this works exactly the same.
For nonstandard sample rates, the different predefined G726
names (G726-16, G726-24, G726-32, G726-40) are interpreted as an
indication of the bits per coded sample, even though their
actual bitrates aren't what the name specifies.
This feels more sane than using free-form names for nonstandard
sample rate/bitrate combinations, e.g like G726-22, G726-33
for 11025 Hz.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids overflow if frame_size is over 2147, since both
frame_size and AV_TIME_BASE are plain integers.
Signed-off-by: Martin Storsjö <martin@martin.st>
* cus/stable:
ffplay: Copy audio side data too. This fixes handling of some rare nellymoser files that change the sample rate mid stream (sample file at: http://trac.videolan.org/vlc/ticket/5586)
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This simplifies the open functions by avoiding one function
call that needs error checking, reducing the amount of
extra bulk code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This string will be passed to ff_http_auth_create_response
even if no proxy is used, resulting in reading uninitialized
memory. The other auth string is always initialized by
av_url_split.
Signed-off-by: Martin Storsjö <martin@martin.st>
There's no reason to use two arrays for this.
Based off commit 2fea60c600
to FFmpeg by Michael Niedermayer.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>
I.e. on streamcopy set output codec timebase from input stream timebase
(as opposed to input codec timebase). This should be more sane, because
since the stream is not decoded, the input codec tb has no relation to
the timestamps of the copied packets.
The next call to decode() will update from an invalid index, which will
either lead to a memcpy() where dest==src (2 threads), or lead to a
crash (>2 threads).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
adtsenc: Check frame size.
txd: Fix order of operations.
APIchanges: fill in some blanks
timer: fix misspelling of "decicycles"
Eliminate pointless 0/NULL initializers in AVCodec and similar declarations.
indeo3: cosmetics
md5proto: Fix order of operations.
dca: Replace oversized unused get_bits() with skip_bits_long().
Conflicts:
doc/APIchanges
libavformat/mmsh.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vc1: use an enum for Frame Coding Mode
doc: cleanup filter section
indeo3: error out if no motion vector is set.
x86inc: Flag shufps as an floating-point instruction for the AVX emulation code.
mpegaudio: do not use init_static_data() for initializing tables.
musepack: fix signed shift overflow in mpc_read_packet()
mov: Make format string match variable type.
wmavoice: Make format string match variable type.
vc1: select interlaced scan table by FCM element
Generalize RIFF INFO tag support; support reading INFO tag in wav
pthread: track thread existence in a separate variable.
Conflicts:
doc/filters.texi
libavcodec/pthread.c
libavformat/avi.c
libavformat/riff.c
libavformat/riff.h
libavformat/wav.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using an unsigned variable avoids problems with overflows.
There is further no need for a 64-bit intermediate here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Interlaced videos can contain progressive frames too and now wrong scantable
is selected for them.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes a compile error on mingw32 when using p->thread
directly (as if it were a pointer) to track thread existence,
because the type is opaque and may be a non-pointer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Interlaced videos can contain progressive frames too and now wrong scantable
is selected for them.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (42 commits)
swscale: fix signed overflow in yuv2mono_X_c_template
snow: fix integer overflows
svq1enc: remove stale altivec-related hack
snow: fix signed overflow in byte to 32-bit replication
adx: rename ff_adx_decode_header() to avpriv_adx_decode_header()
avformat: add CRI ADX format demuxer
adx: add an ADX parser.
adx: move header decoding to ADX common code
adx: calculate the number of blocks in a packet
adx: define and use 2 new macro constants BLOCK_SIZE and BLOCK_SAMPLES
adx: check for unsupported ADX formats
adx: simplify encoding by using put_sbits()
adx: calculate correct LPC coeffs
adx: use 12-bit coefficients instead of 14-bit to avoid integer overflow
adx: simplify adx_decode() by using get_sbits() to read residual samples
adx: fix the data offset parsing in adx_decode_header()
adx: remove unneeded post-decode channel interleaving
adx: validate header values
adx: cosmetics: general pretty-printing and comment clean-up
adx: remove useless comments
...
Conflicts:
Changelog
libavcodec/cook.c
libavcodec/fraps.c
libavcodec/nuv.c
libavcodec/pthread.c
libavcodec/version.h
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
As old bits are shifted out of the accumulator, they cause signed
overflows when they reach the end. Making the variable unsigned fixes
this.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The way these values are used, they should have an unsigned type.
A similar change was made for mpegvideo in cb66847.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This simplifies the decoder so it doesn't have to process an in-packet header
or handle arbitrary-sized packets. It also fixes decoding of files with large
headers.
Instead of using fixed coefficients, the correct way is to calculate the
coefficients using the highpass cutoff frequency from the ADX stream header
and the sample rate.
This multiplication can overflow the signed range but not the
unsigned. After right-shifting it will thus fit in the signed
range again.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This was removed erroneously in
046f081b46. This define still is
necessary for getting MAP_ANONYMOUS defined on linux/glibc,
despite the define reshuffling done in that commit.
Without MAP_ANONYMOUS defined, the mprotect calls for setting the
generated mmx2 scaler code pages executable are left out, causing
crashes if that codepath is chosen.
This patch fixes scaling from 192x144 to 320x240 with
-sws_flags fast_bilinear, which crashes on linux at the
moment.
Signed-off-by: Martin Storsjö <martin@martin.st>
A value shifted left by >31 needs to have a 64-bit type.
As there are no defined channels in this range, the fix
is purely theoretical at this point, although it does
avoid some invalid shifts triggering the overflow
checker.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It makes more sense for a bit mask to use an unsigned type.
The change should be source and binary compatible on all
supported systems, hence micro version bump.
Fixes a few invalid shifts.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is a hand-tuned version of the code with impossible parts of
the FASTDIV function ommitted.
2-5% faster overall on Cortex-A8.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add all seven test bitstreams of Coding Technologies "aacPlus Decoder
Check Package". The streams cover different ways to signal SBR and PS
in different formats.
Casting the left-most byte to unsigned avoids an undefined
result of the shift by 24 if bit 7 is set. This affects
the rm demuxer.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Also reduce verbosity for the unsupported stream message, use
an AVFormatContext for av_log and and print the tag of the
unknown stream.
Improves ticket #672.
The number of used channels need to be reset to zero when
swr_alloc_set_opts() is called successive times.
This is a regression introduced in c8136ebd.
The qatar implementation makes no sense.
a muxer without timestamps is constant fps thus needs vsync.
the crc/mp5 are special cases that have timestamps yet allow any
nonsensical timestamps.
raw (yuv/rgb) video is constant fps thus needs vsync too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At EOF it makes no sense to modify avpkt.{data,size} in output_packet
since no data is consumed. Frame threading with more than 1 threads
hits the segfault.
I.e. if the packet was only partially consumed, pass the rest of it into
the decoder again.
Also simplify the code so it's the same for video/audio/subs.
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framemd5, as it prints dts.
-vsync 0 for the vp8 test is needed because with vsync 2 the timestamp
guessing code gets confused by an altref frame that is never displayed
and drops a frame later.
Note, this is a theoretical fix, iam not aware of a case where this function receives
duplicates in its input.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall:
call revert_cdlms()
Fix some loop conditions to prevent overreads
Initialize pred in lms_predict()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The initial values are not checked against the number of block sizes.
Initializing them to frame_len_bits will result in a block size index of 0
in these cases instead of something that might be out-of-range.
Fixes Bug 81.
This prevents build errors when compiler and assembler default
targets differ. Ideally each file would declare the highest
level it requires. This is however not easily possible as it
complicates assembling pre-armv6t2 code in Thumb-2 mode.
HAVE_NEON is used as indicator for ARMv7-A since no other
symbol exists for this and NEON is only available in this
variant.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Adding the thread count in frame level multithreading to has_b_frames
as an additional delay causes more problems than it solves.
For example inconsistent behaviour during timestamp calculation in
libavformat.
Thread count and frame level multithreading are both set by the user.
If the additional delay caused by frame level multithreading needs
to be considered in the calling code it has all information to take
it into account.
Should it become necessary to calculate a maximum delay inside
libavcodec it should be exported as its own field and not reusing
an existing field.
Based on a patch by Michael Niedermayer.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
FATE samples paths can not be contain a '~' character. This has bitten me
on multiple occasions and I have heard I'm not the only victim.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* mjbshaw/master:
Applied patches from Michael Niedermayer to clean up the code
Added support for J2K encoding with libopenjpeg
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes NGS00144
Thanks to Phillip for suggestions to improve the patch.
Found-by: Phillip Langlois
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
A user who wishes to use default error concealment cannot set the
AV_EF_CRCCHECK flag because not every CRC in every format is a
reliable indicator of bitstream damage. In some formats crcrs
can be nonsensical in absence of any damage. We thus add the
AV_EF_CAREFUL flag in addition to the AV_EF_CRCCHECK flag to
allow a user to enable this reliable CRC check without having to
enable all CRC checks in all formats.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
git.videolan.org is confusing to some users as videolan is a seperate project from ffmpeg.
the new name points to git.videolan.org but avoids the confusion.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall: (24 commits)
Clean-up
dump_int_buffer() to dump samples from a buffer
Implement revert_cdlms()
Doxy for reset_codec()
Store transient state and position of transient area
Implement use_high_update_speed() and use_normal_update_speed()
Initialize num_logged_tiles and remove unnecessary codes
Log index for each line of output
Log tile size
Output decoded residues
Replace placeholders with actual calls to clear_codec_buffers() and reset_codec()
Implement lms_update()
Implement lms_predict()
Implement reset_codec()
Add missing syntax elements to WmallDecodeCtx
Add .recent syntax element to cdlms struct
Implement clear_codec_buffers()
Add buffers to context necessary for reverting cdmls and mclms filter
Use avpriv_copy_bits() instead of ff_copy_bits()
Cosmetics
...
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Bug introduced in:
commit 7bb1807c2d
Author: Anton Khirnov <anton@khirnov.net>
Date: Tue Aug 23 12:58:49 2011 +0200
AVOptions: refactor set_number/write_number
write_number() does the actual writing of the supplied
number to destination. Move finding the option and choosing destination
address out of it.
YASM does not support %ifempty even though it is in the official
NASM documentation and supported since NASM 2.02.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This issue is actually already fixed in NASM, however it will be a
bit before it arrives everywhere.
This should work as a temporary work-around.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Although gcc guarantees 16 byte stack alignment, threads under WinXP
don't appear to be guaranteed to start stack aligned. So fix the
alignment.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
A new field, AVCodecContext.internal is used to hold a new struct
AVCodecInternal, which has private fields that are not codec-specific and are
used by general libavcodec functions.
Moved internal_buffer, internal_buffer_count, and is_copy.
libavcodec/options.c:583: warning: assignment discards qualifiers from pointer
target type
libavcodec/options.c:589: warning: initialization discards qualifiers from
pointer target type
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/swscale_unscaled.c:915:9: warning: new qualifiers in middle of multi-level non-const cast are unsafe
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/swscale_unscaled.c:805:5: warning: passing argument 1 of ‘check_image_pointers’ from incompatible pointer type
libswscale/swscale_unscaled.c:774:12: note: expected ‘uint8_t **’ but argument is of type ‘const uint8_t * const*’
libswscale/swscale_unscaled.c:809:5: warning: passing argument 1 of ‘check_image_pointers’ discards qualifiers from pointer target type
libswscale/swscale_unscaled.c:774:12: note: expected ‘uint8_t **’ but argument is of type ‘uint8_t * const*’
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/utils.c:941:13: warning: passing argument 5 of ‘initMMX2HScaler’ from incompatible pointer type
libswscale/utils.c:524:12: note: expected ‘int32_t *’ but argument is of type ‘int16_t *’
libswscale/utils.c:942:13: warning: passing argument 5 of ‘initMMX2HScaler’ from incompatible pointer type
libswscale/utils.c:524:12: note: expected ‘int32_t *’ but argument is of type ‘int16_t *’
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The filled in refs cause corruptions in the video frame
for a long time after it should have recovered.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Note: FCPublish/FCUnpublish are adobe server specific and not described
in the rtmp specification. Some servers might not cope with them at
all.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The Apple HTTP Live Streaming demuxer's implementation of
seeking searches for the MPEG TS segment which contains the
requested timestamp. In its current implementation it assumes
that the first segment will start from 0.
But, MPEG TS streams do not necessarily start with timestamp
(near) 0, causing seeking to fail for those streams.
This also occurs when using live streaming of HTTP Live Streams.
In this case sliding playlists may be used, which means that in
that case only the last x encoded segments are stored, the earlier
segments get deleted from disk and removed from the playlist.
Because of this, when starting playback of a stream in the middle
of such a broadcast, the initial segment fetched after parsing
the m3u8 playlist will not start from timestamp (near) 0, causing
(the admittedly limited live) seeking to fail.
This patch changes this demuxers seeking implementation to use
the initial DTS as an offset for searching the segments containing
the requested timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tls protocol handles connections via proxies internally.
With TLS/SSL, the peer verification requires that the client
speaks directly with the server, since the proxy doesn't have
the remote server's private key.
Signed-off-by: Martin Storsjö <martin@martin.st>
This opens a plain TCP connection through the proxy via the
CONNECT HTTP method. Normally, this is allowed for connections
on port 443, but can in general be used to allow connections
to any port (depending on proxy configuration), and could thus
be used to tunnel any TCP connection via a HTTP proxy.
Signed-off-by: Martin Storsjö <martin@martin.st>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
Note: Some of the previous patches have had two bugs that have been fixed
in this patch.
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Our ac3 code chain can handle it fine.
More ideal would be to write a demuxer that actually extracts what can be from the additional
headers and uses it for whatever it can be used for.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
TLSv1 is compatible with SSLv3, so this doesn't change much
in terms of compatibility. By explicitly using TLSv1, OpenSSL
sends the server name indication (SNI) header, which we
already set using SSL_set_tlsext_host_name (earlier, this
didn't have any effect).
SNI allows servers to serve SSL content for different host
names with separate certificates on one single port (vhosts).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpegaudiodec: Don't use a nonexistent log context for av_dlog
avformat: Accept the ISO8601 separate format as input, too
avformat: Interpret times in ff_iso8601_to_unix_time as UTC
avutil: Add av_timegm as a public function
cinepak: Add another special case so that it can handle the following file:
lagarith: add some RGBA decoding support
lagarith: Add correct line prediction for RGB
Conflicts:
doc/APIchanges
libavcodec/cinepak.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the function accept the format of creation_time
as output by demuxers (e.g. the mov demuxer), making the
creation timestamp stay intact if transcoding.
Signed-off-by: Martin Storsjö <martin@martin.st>
This function is used in muxers for parsing the 'creation_time'
metadata key, for converting it to a time value.
This makes it match the behaviour of the exported 'creation_time'
metadata from demuxers, where it is in UTC, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is useful, since the normal timegm function isn't a standard
function (requiring _BSD_SOURCE or _SVID_SOURCE on glibc to
be visible, and not available on e.g. windows). The widely available
function mktime uses the local time zone, which requires ugly
workarounds to handle UTC time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Converting to double before the multiplication rather than after
avoids an integer overflow in some cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add flags field to Writer, and define the
WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS flag which forces the display of
optional fields, for example non-available or invalid fields.
Also set the flag in the default writer.
This change is required as for some writers it is preferable to show all
fields (for example for avoiding a variable number of fields in CSV
output), while for other formats (e.g. JSON, XML) it is better to leave
invalid/unavailable fields unspecified and thus simplify the parsing
stage on the deserializer side.
The Apple HTTP Live Streaming demuxer's implementation of seeking searches for
the MPEG TS segment which contains the requested timestamp. In its current
implementation it assumes that the first segment will start from 0.
But, MPEG TS streams do not necessarily start with timestamp (near) 0, causing
seeking to fail for those streams.
This also occurs when using live streaming of HTTP Live Streams. In this case
sliding playlists may be used, which means that in that case only the last x
encoded segments are stored, the earlier segments get deleted from disk and
removed from the playlist. Because of this, when starting playback of a stream
in the middle of such a broadcast, the initial segment fetched after parsing
the m3u8 playlist will not start from timestamp (near) 0, causing (the
admittedly limited live) seeking to fail.
This patch changes this demuxers seeking implementation to use the initial DTS
as an offset for searching the segments containing the requested timestamp.
* qatar/master:
binkvideo: simplify and remove invalid shifts
pulse: compute frame_duration once and fix it
lavf: simplify format_child_class_next()
hwaccel: OS X Video Decoder Acceleration (VDA) support.
doc: add support for an optional navigation bar in texi2html pages
Conflicts:
configure
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/vda.c
libavcodec/vda.h
libavcodec/vda_h264.c
libavcodec/vda_internal.h
libavcodec/version.h
libavformat/options.c
libavutil/avutil.h
libavutil/pixfmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The frame duration was calculated without taking in account the
bytes per sample.
Thanks to Lorenzo Pistone <blaffablaffa@gmail.com> for pointing
the issue and providing an initial fix.
The navigation header for the web pages lives in a different repository.
Read it during documentation regeneration to use the same navigation bar
on all pages.
* qatar/master:
lavf: pass options from AVFormatContext to avio.
avformat: Use avio_open2, pass the AVFormatContext interrupt_callback onwards
avio: add avio_open2, taking an interrupt callback and options
avio: add support for passing options to protocols.
avio: add and use ffurl_protocol_next().
avformat: Pass the interrupt callback on to chained muxers/demuxers
avio: Add an AVIOInterruptCB parameter to ffurl_open/ffurl_alloc
avformat: Use ff_check_interrupt
avio: Add an internal utility function for checking the new interrupt callback
avio: Add AVIOInterruptCB
texi2html: remove stray \n
doc: prettyfy the texi2html documentation
swscale: handle unaligned buffers in yuv2plane1
Conflicts:
libavformat/avformat.h
libavformat/avio.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use log(sqrt(mb->bailout)) instead of log(mb->bailout) because mb->bailout represent
the bailout radius squared, and then simplify the two sqrt().
This is also slightly faster.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
previous calculated pixel, this is more accurate due to the last
calculated often being farther away.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The interrupt callback has to be passed in during opening (setting it
after opening isn't enough), since a blocking open couldn't be
interrupted otherwise.
Options are passed down to procotols and also need to be available
during open() in most cases.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This is a better io interrupt callback function, which has an
opaque parameter, which is given to the interrupt callback.
This allows callers to precisely cancel IO for one single
AVFormatContext, without interrupt other ones in the same
process.
Note, it's not needed in AVIOContext, at the moment.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
vble: remove vble_error_close
VBLE Decoder
tta: use an integer instead of a pointer to iterate output samples
shorten: do not modify samples pointer when interleaving
mpc7: only support stereo input.
dpcm: do not try to decode empty packets
dpcm: remove unneeded buf_size==0 check.
twinvq: add SSE/AVX optimized sum/difference stereo interleaving
vqf/twinvq: pass vqf COMM chunk info in extradata
vqf: do not set bits_per_coded_sample for TwinVQ.
twinvq: check for allocation failure in init_mdct_win()
swscale: add padding to conversion buffer.
rtpdec: Simplify finalize_packet
http: Handle proxy authentication
http: Print an error message for Authorization Required, too
AVOptions: don't return an invalid option when option list is empty
AIFF: add 'twos' FourCC for the mux/demuxer (big endian PCM audio)
Conflicts:
libavcodec/avcodec.h
libavcodec/tta.c
libavcodec/vble.c
libavcodec/version.h
libavutil/opt.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It does not make much sense to factor the error handling to its own
av_always_inline function. Fixes "format not a string literal and no
format arguments" warning in the av_log.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It does not make much sense to factor the error handling to its own
av_always_inline function. Fixes "format not a string literal and no
format arguments" warning in the av_log.
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This is needed because the twinvq decoder cannot rely on bit_rate to be set.
The API documentation says that bit_rate is set by libavcodec, not by the
user.
Altivec does unaligned reads from this buffer in
hscale_altivec_real(), and can thus read up to 16 bytes beyond
the end of the buffer. Therefore, add an extra 16 bytes of
padding at the end of the conversion buffer.
This fixes fate-lavfi-pixfmts_scale on AltiVec-enabled builds
under valgrind.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Tested with both Basic and Digest authentication, and tested with
both proxy authentication and authentication for the requested
resource at the same time.
Signed-off-by: Martin Storsjö <martin@martin.st>
The error was hidden before, to avoid showing an error on the
first request where no auth has been provided, when the server
indicates which authentication method to use.
Now the error is printed if an authentication method was used,
but failed.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (29 commits)
doc: update libavfilter documentation
tls: Use the URLContext as logging context
aes: Avoid illegal read and don't generate more key than we use.
mpc7: Fix memset call in mpc7_decode_frame function
atrac1: use correct context for av_log()
apedec: consume the whole packet when copying to the decoder buffer.
apedec: do not needlessly copy s->samples to nblocks.
apedec: check output buffer size after calculating actual output size
apedec: remove unneeded entropy decoder normalization.
truespeech: use memmove() in truespeech_update_filters()
vorbisdec: remove AVCODEC_MAX_AUDIO_FRAME_SIZE check
vorbisdec: remove unneeded buf_size==0 check
vorbisdec: return proper error codes instead of made-up ones
http: Don't add a Range: bytes=0- header for POST
sunrast: Check for invalid/corrupted bitstream
http: Change the chunksize AVOption into chunked_post
http: Add encoding/decoding flags to the AVOptions
avconv: remove some codec-specific hacks
crypto: add decoding flag to options.
tls: use AVIO_FLAG_NONBLOCK instead of deprecated URL_FLAG_NONBLOCK
...
Conflicts:
doc/libavfilter.texi
libavcodec/atrac1.c
libavcodec/sunrast.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Update the reference to the conversion tool to use avconv and
make sure the example line works as supposed.
Remove the paragraph pointing to the svn repo
The decoder already skips data at the end of the packet without this.
Also remove 2 APEContext fields that were only used for the end-of-frame
normalization.
The chunksize internal variable has two different uses - for
reading, it's the amount of data left of the current chunk
(or -1 if the server doesn't send data in chunked mode), where
it's only an internal state variable. For writing, it's used
to decide whether to enable chunked encoding (by default), by
using the value 0, or disable chunked encoding (value -1).
This, while consistent, doesn't make much sense to expose
as an AVOption. This splits the usage of the internal variable
into two variables, chunksize which is used for reading (as
before), and chunked_post which is the user-settable option,
with the values 0 and 1, where 1 is default.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avcodec: add support for planar signed 8-bit PCM.
ra144enc: add sample_fmts list to ff_ra_144_encoder
smackaud: use uint8_t* for 8-bit output buffer type
smackaud: clip output samples
smackaud: use sign_extend() for difference value instead of casting
sipr: use a function pointer to select the decode_frame function
sipr: set mode based on block_align instead of bit_rate
sipr: do not needlessly set *data_size to 0 when returning an error
ra288: fix formatting of LOCAL_ALIGNED_16
udp: Allow specifying the local IP address
VC1: Add bottom field offset to block_index[] to avoid rewriting (+10L)
vc1dec: move an if() block.
vc1dec: use correct hybrid prediction threshold.
vc1dec: Partial rewrite of vc1_pred_mv()
vc1dec: take ME precision into account while scaling MV predictors.
lavf: don't leak corrupted packets
Conflicts:
libavcodec/8svx.c
libavcodec/ra288.c
libavcodec/version.h
libavformat/iff.c
libavformat/udp.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is found in some 8svx files (e.g. ones created by SoX).
Currently the decoder reuses the 8svx functions because we already have
handling of a single large planar packet for the compressed 8svx codecs.
8556406d38 (lavc: lowercase the FRWU codec name)
got partially reverted by ec6402b7c5
(lavc: use designated initialisers for all codecs.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
not to mention race conditions and that its used for stream copy, used to determine IPB type by
applications and other things.
Fixes various frame drop/timestamp issues with frame multithreading.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The current code is a generalization of the earlier progressive MV
prediction code. This was supposed to predict MVs for both interlaced
and progressive pictures. But the interlaced MV prediction is buggy
and works mostly by luck.
This partially fixes interlaced MV prediction.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
tls: Use ERR_get_error() in do_tls_poll
indeo3: Fix a fencepost error.
mxfdec: Fix comparison of unsigned expression < 0.
mpegts: set stream id on just created stream, not an unrelated variable
ra288: return error if input buffer is too small
ra288: utilize DSPContext.vector_fmul()
ra288: use memcpy() to copy decoded samples to output
mace: only calculate output buffer size once
Remove redundant filename self-references inside files.
indeo3data: add missing config.h #include for HAVE_BIGENDIAN
x86: drop pointless ARCH_X86 #ifdef from files in x86 subdirectory
avplay: reset rdft when closing stream.
doc/git-howto: expand format-patch and send-email notes.
lavf: expand doxy for some AVFormatContext fields.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The codec uses all previous frames as reference frames, so they
certainly must be preserved and readable.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The return value ret isn't an error code that can be passed
to ERR_error_string().
This makes the error messages printed actually contain useful
information.
Signed-off-by: Martin Storsjö <martin@martin.st>
Changing the lowres value is risky because the user application may have a
local copy and not read back into it, or not undo some lowres dependant things.
A patch implementing this in ffplay is already on ffmpeg-dev, so this feature
should be back soon.
This reverts commit 125ea3ee06.
* qatar/master:
avformat: Avoid a warning about mixed declarations and code
BMV demuxer and decoder
matroskaenc: Make sure the seekhead struct is freed even on seek failure
mpeg12enc: Remove write-only variables.
mpeg12enc: Don't set up run-level info for level 0.
msmpeg4: Don't set up run-level info for level 0.
avformat: Warn about using network functions without calling avformat_network_init
avformat: Revise wording
rdt: Set AVFMT_NOFILE on ff_rdt_demuxer
rdt: Check the return value of avformat_open
rtsp: Discard the dynamic handler, if it has an alloc function which failed
dsputil: use cpuflags in x86 versions of vector_clip_int32()
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Actually check the return value of DecodeBegin, to make
sure that it has encountered no errors.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Remove the CODEC_CAP_LOSSLESS flag, as it doesn't make
any sense for a decoder to use it.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since buf_size is only used in this one function, there
is no reason for it to be part of UtVideoContext.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The caller expects the seekhead struct to be freed when calling
matroska_write_seekhead. Currently, the structure is leaked if the
seek fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
run: The number of zero coefficients preceding a non-zero coefficient,
in the scan order. The absolute value of the non-zero coefficient is
called "level".
The run-level code makes illegal reads when trying to set up tables for
nonsense level 0.
run: The number of zero coefficients preceding a non-zero coefficient,
in the scan order. The absolute value of the non-zero coefficient is
called "level".
The run-level code makes illegal reads when trying to set up tables for
nonsense level 0.
Width and height, as used in utvideo_decode_frame, do not
need to be unsigned, and it could cause sign-compare
warnings later on.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The frameinfo size member of the Ut Video extradata
was erroneously thought to be the number of stripes.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is to make developers aware of the fact that they will
start using the new init function at some point.
Signed-off-by: Martin Storsjö <martin@martin.st>
It might make sense not to make the function completely mandatory
immediately at the next bump, which might be quite soon after
the function was introduced.
Signed-off-by: Martin Storsjö <martin@martin.st>
The original format field in Ut Video's extradata
should not be used to determine the output format.
Cases can occur where the original format differs
from the actual current format, and thus should
not be used as the output format. Instead, rely
solely on the FOURCC.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new design allows to specialize the format for continuos time values
and discrete values. In particular, allows to print integer values like
"12" rather than "12.000".
Prettify, simplify parsing for integer values.
This makes rdt work again, which has been broken since
603b8bc2a1. This commit made
opening a demuxer without a file (or in this case, with a filename
which can't be opened) fail, unless the demuxer actually declared
AVFMT_NOFILE.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (23 commits)
x86inc: use sse versions of common macros instead of sse2 when applicable
doc/APIchanges: add missing dates and hashes
lavf: don't return from void av_update_cur_dts()
Changelog: add more entries.
Changelog: update ffmpeg/avconv incompatibility list.
avconv: remove some redundant temporary variables.
avconv: fix broken indentation
avconv: move copy_initial_nonkeyframes to the options context.
avconv: use file:stream instead of file.stream in log messages.
doc/avconv: elaborate on basic functionality.
doc/avconv: -sample_fmts, not -help sample_fmts prints the sample formats
openssl: Only use CRYPTO_set_id_callback on OpenSSL < 1.0.0
Call avformat_network_init/deinit in the programs
Remove leftover includes of strings.h
avutil: Don't allow using strcasecmp/strncasecmp
Replace all usage of strcasecmp/strncasecmp
avstring: Add locale independent implementations of strcasecmp/strncasecmp
avstring: Add locale independent implementations of toupper/tolower
cosmetics: insert some spaces in explicit enum value assignments
move 8SVX audio codecs to the audio codec list part on the next bump
...
Conflicts:
avprobe.c
doc/APIchanges
ffplay.c
ffserver.c
libavcodec/avcodec.h
libavdevice/bktr.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavformat/matroskaenc.c
libavformat/wtv.c
libavutil/avstring.c
libavutil/avstring.h
libavutil/avutil.h
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Since 1.0.0, this function is deprecated. A new function,
CRYPTO_THREADID_set_callback is available, but if not set at all,
it uses the address of errno as thread id, which should be
sufficient for most systems.
On windows, it never was necessary to use this function even
before 1.0.0, it used the right win32 API function for this
by default.
Signed-off-by: Martin Storsjö <martin@martin.st>
Calling the init function will become mandatory at some later
point. By calling it, more heavy network init (such as SSL/TLS
library init) can be done once at startup, instead of implicitly
when used (which could lead to it being done a number of times).
Signed-off-by: Martin Storsjö <martin@martin.st>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Martin Storsjö <martin@martin.st>
p cannot be calculated before av_dup_packet since that one
might change avpkt->data, causing invalid reads and a
non-working range check.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We keep INIT_AVX (for backwards compatibility). 3arg AVX ops with
a memory arg can only have it in src2, whereas SSE emulation of
3arg prefers to have it in src1 (i.e. the mov). So, if the op is
symmetric and the wrong one is memory, swap them.
* qatar/master:
http: Remove the custom function for disabling chunked posts
rtsp: Disable chunked http post through AVOptions
movdec: Set frame_size for AMR
h264_weight: remove duplication functions.
swscale: align vertical filtersize by 2 on x86.
libavfilter: reindent.
matroskadec: empty blocks are in fact valid.
avfilter: don't abort() on zero-size allocations.
h264: improve calculation of codec delay.
movenc: Set a correct packet size for AMR-NB mode 15, "no data"
avformat: Add functions for doing global network initialization
avformat: Add the https protocol
avformat: Add the tls protocol, using OpenSSL or gnutls
avformat: Initialize gnutls in ff_tls_init()
w32threads: Wrap the mutex functions in inline functions returning int
configure: Allow linking to the gnutls library
avformat: Add ff_tls_init()/deinit() that initialize OpenSSL
configure: Allow linking to openssl
avcodec: Allow locking and unlocking an avformat specific mutex
avformat: Split out functions from network.h to a new file, network.c
Conflicts:
Changelog
configure
doc/APIchanges
libavcodec/internal.h
libavcodec/version.h
libavfilter/formats.c
libavformat/matroskadec.c
libavformat/mov.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
avctx->width/height remain right/bottom cropped as previous behaviour.
Hardware decoders need to know the uncropped data to allocate surfaces
of correct height. Some hardware is picky and fails to decoder properly
if a surface larger than needed is used during decode, so just aligning
up is not enough.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When constructing a channel layout string from individual channel names,
+ is a more practical separator than |, because it is usually not a shell
special character.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There were multiple issues, for example might we have to re-run
the decompression when the size of the buffer increased,
we should always use a decompression buffer large enough for
the header (so we do not get stuck when the size is too small).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Treating them like 0 is safest, current code would invoke
undefined pointer arithmetic behaviour in this case.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
I am not sure these new values are correct, not am I sure
the semantics are a good idea since we do not seem to make any
use of them but they caused a lot of confusion, but this
seems to make things closer to matching the documentation.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Earlier, sc->samples_per_frame was used for setting the frame size,
but all files don't have that set properly. The frame size is a
known constant for these codecs.
If frame_size isn't set, the mov/3gp muxer refuses to mux it.
This fixes stream copy of audio from
https://roundup.libav.org/file1248/Video_With_AMR-NB_Audio.3gp
to another 3gp file (roundup issue 2468).
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes the following conformance suite samples:
HCBP1_HHI_A.264, HCBP2_HHI_A.264, HCMP1_HHI_A.264 (main)
HCHP1_HHI_B.264, HCHP2_HHI_A.264, HCHP3_HHI_A.264 (frext)
These packets are valid packets, and consist of 1 byte (which
contains the mode bits).
This had been analyzed and reported by Igor Levin, igor d levin comverse com.
Signed-off-by: Martin Storsjö <martin@martin.st>
Saves a bit on space and relocations.
Also makes the (very hackish) lossless conversion check
in ffmpeg.c work reliably.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Note, this protocol doesn't yet check verify the server
certificate against a local database of trusted CA root
certificates.
Signed-off-by: Martin Storsjö <martin@martin.st>
This extends the lock manager in avcodec to manage two separate
mutexes via the user-specified lock functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
Issue an error and return a more meaningful error code in case the
maximum number of registered filters has been reached.
This is better than silently ignore the error, improve feedback.
* qatar/master:
binkaudio: expand quant_table to accommodate all possible values
libx264: Set the default of the rc_lookahead option to -1
avcodec: Set flags2 default value depending on availability
configure: declare dependency of h264_vaapi_hwaccel on h264_decoder
Conflicts:
configure
libavcodec/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Change the libutvideo wrapper name to "libutvideo", as
it currently conflicts with the native utvideo decoder's
name of "utvideo".
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows it to use the defaults specified by preset/tune,
without overwriting it with the default value from the
AVCodecContext field.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
vp6: Fix illegal read.
avfilter: Don't copy garbage from the stack when setting up video pictures.
avcodec: Make sure codec_type is set by avcodec_get_context_defaults2
avcodec: Remove a misplaced and useless attribute_deprecated
avconv: add -dump_attachment option.
avconv: add -attach option.
avconv: make negative mappings disable only streams from the specified file
fmtconvert: fix int32_to_float_fmul_scalar() for windows x86_64
Conflicts:
libavcodec/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This function used to set codec_type. With the current fallback
implementation based on avcodec_get_context_defaults3, codec_type
won't be set to the value passed in, but will be set to
AVMEDIA_TYPE_UNKNOWN. Legacy callers of this function might expect
this field to be set to the value passed in.
Signed-off-by: Martin Storsjö <martin@martin.st>
If attribute_deprecated is used in an enum declaration, it
should follow the 'enum' keyword, otherwise it's ignored
silently. This is the only case of attribute_deprecated for
enum declarations currently.
Currently, this attribute_deprecated doesn't have any effect.
If moved to the right place, it emits a warning every single
time avcodec.h is included, like this:
avcodec.h:2827: warning: ‘AVLPCType’ is deprecated (declared at avcodec.h:543)
There is already a working attribute_deprecated for the
corresponding field in AVCodecContext, so therefore this
one shouldn't be needed.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (44 commits)
replacement Indeo 3 decoder
gsm demuxer: do not allocate packet twice.
flvenc: use first packet delay as global delay.
ac3enc: doxygen update.
imc: return error codes instead of 0 for error conditions.
imc: return meaningful error codes instead of -1
imc: do not set channel layout for stereo
imc: validate channel count
imc: check for ff_fft_init() failure
imc: check output buffer size before decoding
imc: use DSPContext.bswap16_buf() to byte-swap packet data
rtsp: add allowed_media_types option
libgsm: add flush function to reset the decoder state when seeking
libgsm: simplify decoding by using a loop
gsm: log error message when packet is too small
libgsmdec: do not needlessly set *data_size to 0
gsmdec: do not needlessly set *data_size to 0
gsmdec: add flush function to reset the decoder state when seeking
libgsmdec: check output buffer size before decoding
gsmdec: log error message when output buffer is too small.
...
Conflicts:
Changelog
ffplay.c
libavcodec/indeo3.c
libavcodec/mjpeg_parser.c
libavcodec/vp3.c
libavformat/cutils.c
libavformat/id3v2.c
libavutil/parseutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The new decoder is much smaller and has better code quality.
Cleanup and fixes courtesy of Kostya Shishkov.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Add some parameters to existing function documentation.
Remove some unneeded documentation.
Convert some static function documentation to non-doxygen style.
Current implementation fades to 0. This implementation fades to 16 for
YUV formats that contain CCIR601/709 video levels. RGB and YUVJ
formats are not altered.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adds support for year (TYER) and day/month (TDAT) tags when writing
id3v2 version 3 metadata by splitting the "date" tag. The date tag
should have a format of "YYYY-MM-DD" or "YYYY".
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is an easy alternative to setting bit_rate. This patch also selects the
closest bit_rate to the requested one rather than requiring an exact value.
This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.
This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
This code was already added by Yuvi in c82cbea68273c6f08c4d0e94fc9fd50bfdea4e2b
It was subsequently lost somehow
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Based on the work of Mark Himsley <mark@mdsh.com>.
See thread:
Subject: [FFmpeg-devel] libavfilter: extending overlay filter
Date: Sun, 13 Mar 2011 14:18:42 +0000
* qatar/master:
libdirac/libschroedinger: Drop unnecessary symbol prefixes.
cmdutils: check fread() return value
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
tune, fastfirstpass and profile are not available through x264opts.
Since some FFmpeg options don't match the names used in libx264, add
libx264 option names as a reference when needed. And last thing, x264
don't have a manpage ATM, so prefer the x264 --fullhelp over the
unexistant manpage reference.
* qatar/master: (51 commits)
cin audio: use sign_extend() instead of casting to int16_t
cin audio: restructure decoding loop to avoid a separate counter variable
cin audio: use local variable for delta value
cin audio: remove unneeded cast from void*
cin audio: validate the channel count
cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
dsicin: fix several audio-related fields in the CIN demuxer
flacdec: use av_get_bytes_per_sample() to get sample size
dca: handle errors from dca_decode_block()
dca: return error if the frame header is invalid
dca: return proper error codes instead of -1
utvideo: handle empty Huffman trees
binkaudio: change short to int16_t
binkaudio: only decode one block at a time.
binkaudio: store interleaved overlap samples in BinkAudioContext.
binkaudio: pre-calculate quantization factors
binkaudio: add some buffer overread checks.
atrac3: support float or int16 output using request_sample_fmt
atrac3: add CODEC_CAP_SUBFRAMES capability
atrac3: return appropriate error codes instead of -1
...
Conflicts:
libavcodec/atrac1.c
libavcodec/dca.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
bits_per_coded_sample should be 8.
block_align is calculated incorrectly, but it is not needed anyway.
packet pts should be calculated in samples.
packet duration can be set.
If the frame is filled with the same colour, encoder may produce no data
and the fill value is indicated by zero code length (the rest of symbols
will have 0xFF for code length, meaning invalid). So such Huffman trees
should be treated specially.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Bug introduced by:
commit f12f40b31a
Author: Anton Khirnov <anton@khirnov.net>
Date: Tue Aug 30 04:05:20 2011 +0200
ffmpeg: get rid of new* options.
They are confusing, irregular and redundant -- -map already contains all
the information. Stream maps can now be parsed in opt_output_file().
Add a more user-friendly default behavior in case no maps are present.
Breaks -programid for now, but it never worked properly anyway. A better
solution will be written soon.
Bug has been introduced by:
commit 1cede1d011
Author: Anton Khirnov <anton@khirnov.net>
Date: Sat Jun 25 07:43:49 2011 +0200
ffmpeg: factor common code from new_a/v/s/d_stream to new_output_stream()
This reverts commit 5dd514af93.
Silently ignoring errors allows some broken files to simply be played, instead of failing.
(cherry picked from commit 7804b0693375c1a7ba1046f7a3579e9f63c2b15a)
The intended goal (as confirmed with its author) of fixing a crash has been
fixed differently prior to the application of this patch and this patch does
notsucessfully propagate parse errors either.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Luma slices in YUV420 colourspace should have height in multiple of two since
they have the same line of chrominance data corresponding to pair of them.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes false positives of has_codec_delay_been_guessed() for
streams where not every input picture generates an output picture,
such as interlaced H264.
* qatar/master: (53 commits)
probe: Restore identification of files with very large id3 tags and no extension.
probe: Remove id3 tag presence as a criteria to do file extension checking.
mpegts: MP4 SL support
mpegts: MP4 OD support
mpegts: Add support for Sections in PMT
mpegts: Replace the MP4 descriptor parser with a recursive parser.
mpegts: Add support for multiple mp4 descriptors
mpegts: Parse mpeg2 SL descriptors.
isom: Add MPEG4SYSTEMS dummy object type indication.
aacdec: allow output reconfiguration on channel changes
nellymoserenc: take float input samples instead of int16
nellymoserdec: use dsp functions for overlap and windowing
nellymoserdec: do not fail if there is extra data in the packet
nellymoserdec: fail if output buffer is too small
nellymoserdec: remove pointless buffer size check.
lavf: add init_put_byte() to the list of visible symbols.
seek-test: free options dictionary after use
snow: do not draw_edge if emu_edge is set
tools/pktdumper: update to recent avformat api
seek-test: update to recent avformat api
...
Conflicts:
doc/APIchanges
libavcodec/mpegaudiodec.c
libavcodec/nellymoserdec.c
libavcodec/snow.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/avformat.h
libavformat/mpegts.c
libavformat/mxfdec.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Locking the decoder against channel config changes in
parse_adts_frame_header() seems to be unnecessary and
streams with channel config changes are reported.
The sample in http://roundup.libav.org/issue999 still works.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
buffer during entropy decoding.
The pointer address could overflow, which would likely segfault. Instead set
the context error flag to indicate that the decoder tried to read past the
end of the packet data.
The dca parser needs to check that the framesize is actually of a complete
frame, and not of a partial DTS-HD frame, which is not constant size, and
thus the check would fail.
(cherry picked from commit ebc0ccb9af59b78732e82c09f8c90b1d46b478e0)
Review-by: Benjamin Larsson
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The documentation for CODEC_CAP_PARSE_ONLY and AVCodecContext.parse_only
indicates that they are utilized through avcodec_parse_frame(), which was
never actually implemented.
Its functionality was removed several years ago, so it doesn't do anything.
AVCodecContext.frame_number could serve the same purpose if someone
wants to debug the frame count.
On frame decoding failure, return an error if the frame is the same size as
the whole packet, otherwise just log an error message and return the number
of bytes consumed.
Prevent error condition in case sample_rate is unset or set to a negative
value. In particular, fix divide-by-zero error occurring in ffmpeg due to
sample_rate set to 0 in output_packet(), in code:
ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
ist->st->codec->sample_rate;
Fix trac ticket #324.
This atom typically is used for a track title.
(cherry picked from commit a356137816b4ea20a892d1fb203b11dbfedbc543)
Reviewed-by: Baptiste Coudurier
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
they wont conflict when they get added to the fork with randomly changed order and values.
Also add duplicate codec_ids to allow maintaining compatibility.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It currently use the simple api and is using the latency information
provided only to offset the stream start.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master: (35 commits)
libopencore-amr: check output buffer size before decoding
libopencore-amr: remove unneeded buf_size==0 check.
libopencore-amr: remove unneeded frame_count field.
aac_latm: remove unneeded check for zero-size packet.
pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
pcmdec: return buf_size instead of src-buf.
avcodec: remove the Zork PCM encoder.
pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
pcmenc: remove unneeded sample_fmt check.
pcmdec: move number of channels check to pcm_decode_init()
pcmdec: remove unnecessary check for sample_fmt change
pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
pcmdec: do not needlessly set *data_size to 0
alacdec: remove unneeded NULL or zero-size packet checks.
alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
alacdec: ask for a sample for unsupported sample depths.
alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
alacdec: move some declarations to the top of the function
alacdec: always use get_sbits_long() for uncompressed samples
...
Conflicts:
libavcodec/pcm.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
* qatar/master: (22 commits)
g722dec: check output buffer size before decoding
g722dec: cosmetics: reindent/linewrap
g722dec: remove the use of lowres for half-rate decoding.
tta: check for extradata allocation failure in tta demuxer
tta: check for allocation failure of decode_buffer
tta: use correct frame_length calculation.
tta: add support for decoding 24-bit sample format
cosmetics: indentation
tta: remove pointless braces
tta: check output buffer size after adjusting frame length for last frame
tta: fix reading of format in TTA header.
tta: remove useless commented-out lines
tta: check remaining bitstream size while reading unary value
lavf: deprecate AVStream.stream_copy
avconc: split choose_codec() to choose_decoder/choose_encoder.
lavf: simplify by using FFMAX/FFMIN.
mpegenc: add preload private option.
cosmetics: simplify latm_decode_init
latm: avoid unnecessary reinit of the aac decoder
aacdec: initialize sbr context only in new channel elements
...
Conflicts:
avconv.c
libavcodec/resample.c
libavcodec/tta.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Without this patch each displayed second is incremented 1 frame early,
second 0 is only 24 frames long where as every other second is 25 frames
long.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is broken because an AVCodecContext can be opened/closed multiple
times, and sample_rate is getting divided by 2 each time that happens.
This removes the only use of lowres for audio.
Note that this will not work in most cases with avconv and avplay due to the
AVCODEC_MAX_AUDIO_FRAME_SIZE limit, but it will decode correctly if given a
large enough output buffer.
The situation was not clear when support was added but it is now:
CELT and Opus are really two different codecs.
The current code supports CELT via libcelt, not Opus.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
ffplay: add frame drop statistics
ffplay: consider estimated time of filter in early frame drop
ffplay: reimplement early frame drop
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch reimplements early frame drop, it is now based on the current
difference between the master clock and the video clock, and the pts of the
current and the last displayed (or skipped) frame. If the frame to be added to
the queue is late after decoding, then we drop it early because later we would
drop it anyway (unless it is the only frame in the picture queue).
The current approach has only one downside that I know of, it does not handle
well when the filters are changing significantly the pts of the frames, because
we compare pts values from filtered and unfiltered frames.
We also start using the pictq_mutex to ensure consistent video_current_pts,
video_current_pts_drift, frame_last_pts, frame_last_dropped_pts and
frame_last_dropped_pos values.
Signed-off-by: Marton Balint <cus@passwd.hu>
Factorize code, extend the functionality of the filter, and make it
return empty buffers. This is useful for filters which ignore the input
frames content.
This is also changing the syntax of the nullsrc source, and dropping the
framerate expression evaluation, which does not look particularly useful.
During slice threading only MpegEncContext is passed to
mpeg_decode_slice, remove a wrong cast and change the function
definition to take MpegEncContext pointer.
This reverts commit 3fc0830432.
The revert is done because the reasoning behind the commit was
flawed, closed_gop is not mpeg1/2 specific.
Conflicts:
libavcodec/mpeg12.c
libavcodec/mpeg12.h
* qatar/master:
Move id3v2 tag writing to a separate file.
swscale: add missing colons to x86 assembly yuv2planeX.
g722: split decoder and encoder into separate files
cosmetics: remove extra spaces before end-of-statement semi-colons
vorbisdec: check output buffer size before writing output
wavpack: calculate bpp using av_get_bytes_per_sample()
ac3enc: Set max value for mode options correctly
lavc: move get_b_cbp() from h263.h to mpeg4videoenc.c
mpeg12: move closed_gop from MpegEncContext to Mpeg1Context
mpeg12: move full_pel from MpegEncContext to Mpeg1Context
mpeg12: move Mpeg1Context from mpeg12.c to mpeg12.h
mpegvideo: remove some unused variables from MpegEncContext.
Conflicts:
libavcodec/mpeg12.c
libavformat/mp3enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit da22ba7df4 since it
broke slice threading. Slice threading just duplicates MpegEncContext
so every value used during mpeg_decode_slice has to be in it.
A second patch will fix the illusion that Mpeg1Context is available
in mpeg_decode_slice.
for example:
Attempting to set ac3 dsur_mode to "on" results in:
[AC-3 Encoder @ 0x7f508f0a3280] Value 2.000000 for parameter 'dsur_mode' out of range
[AC-3 Encoder @ 0x7f508f0a3280] Error setting option dsur_mode to value on.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
id3v2: fix doxy comment - 'machine byte order' makes no sense on char arrays
VC1: restore mistakenly removed code
twinvq: check output buffer size before decoding
twinvq: return an error when the packet size is too small
lavf: export some forgotten symbols with non-av prefixes.
swscale: update altivec yuv2planeX asm to new per-plane API.
swscale: make yuv2yuvX_10_sse2/avx 8/9/16-bits aware.
yuv2planeX10 SIMD
swscale: decide whether to use yuv2plane1/X on a per-plane basis.
swscale: reintroduce full precision in 16-bit output.
Split up yuv2yuvX functions
Split out yuv2yuv1 luma and chroma in order to make them generic DSP functions
lavc: replace references to deprecated AVCodecContext.error_recognition to use AVCodecContext.err_recognition
lavc: translate non-flag-based er options into flag-based ef options at codec open
add -err_filter AVOptions to access flag-based error recognition
h264_weight: initialize "height" function argument properly.
presets: spelling error in libvpx 1080p50_60
avplay: fix fullscreen behaviour with SDL 1.2.14 on Mac OS X
Conflicts:
ffplay.c
libavformat/libavformat.v
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The code was mistakenly removed in cad16562c8.
It stored some motion vector data for future use in B-pictures.
This fixes Bugzilla bug #57.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This allows commands of the form "make foo.s", which is useful
for inspecting the compiler output for debugging purposes.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids an indirection in the table. Data size is slightly
smaller on 64-bit systems, 8 bytes larger on 32-bit.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 214a85202b2eedbef06357e40278c2f56aa2c6e3)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 08e3dea3f7f69309574dafc0af6671615e909720)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes failures when codec IDs are defined conditionally,
for example when scheduling for a major bump.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These are available via -preset or -vprofile and the addition of
-qp 0 or -crf 0 if lossless is desired.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
flvdec: Do not call parse_keyframes_index with a NULL stream
libspeexdec: include system headers before local headers
libspeexdec: return meaningful error codes
libspeexdec: cosmetics: reindent
libspeexdec: decode one frame at a time.
swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
Move timefilter code from lavf to lavd.
mov: add support for hdvd and pgapmetadata atoms
mov: rename function _stik, some indentation cosmetics
mov: rename function _int8 to remove ambiguity, some indentation cosmetics
mov: parse the gnre atom
mp3on4: check for allocation failures in decode_init_mp3on4()
mp3on4: create a separate flush function for MP3onMP4.
mp3on4: ensure that the frame channel count does not exceed the codec channel count.
mp3on4: set channel layout
mp3on4: fix the output channel order
mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
mp3on4: copy MPADSPContext from first context to all contexts.
fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
...
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_ps.c
libavcodec/h264dsp_template.c
libavcodec/h264idct_template.c
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavcodec/x86/h264dsp_mmx.c
libavdevice/Makefile
libavdevice/jack_audio.c
libavformat/Makefile
libavformat/flvdec.c
libavformat/flvenc.c
libavutil/pixfmt.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows for knowing the output size before decoding even when there is no
header (e.g. FLV). Otherwise we would have to do a preliminary full frame
decode to determine the number of frames-per-packet.
Enable multithreding when decoding with libopenjpeg
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Associate segmentation_map[] with reference frame, rather than
decoding instance. This fixes cases where the map would be free()'ed
on e.g. a size change in one thread, whereas the other thread was
still accessing it. Also, it fixes cases where threads overwrite data
that is still being referenced by the previous thread, who thinks that
it's part of the frame previously decoded by the next thread.
* qatar/master: (47 commits)
lavc: hide private symbols.
lavc: deprecate img_get_alpha_info().
lavc: use avpriv_ prefix for ff_toupper4.
lavc: use avpriv_ prefix for ff_copy_bits and align_put_bits.
lavc: use avpriv_ prefix for ff_ac3_parse_header.
lavc: use avpriv_ prefix for ff_frame_rate_tab.
lavc: rename ff_find_start_code to avpriv_mpv_find_start_code
lavc: use avpriv_ prefix for ff_split_xiph_headers.
lavc: use avpriv_ prefix for ff_dirac_parse_sequence_header.
lavc: use avpriv_ prefix for some dv symbols used in lavf.
lavc: use avpriv_ prefix for some flac symbols used in lavf.
lavc: use avpriv_ prefix for some mpeg4audio symbols used in lavf.
lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
lavc: use avpriv_ prefix for ff_aac_parse_header().
lavf: hide private symbols.
lavf: use avpriv_ prefix for some dv functions.
lavf: use avpriv_ prefix for ff_new_chapter().
avcodec: add CODEC_CAP_DELAY note to avcodec_decode_audio3() documentation
avcodec: clarify the CODEC_CAP_DELAY note in avcodec_decode_video2()
avcodec: clarify documentation of CODEC_CAP_DELAY
...
Conflicts:
configure
doc/general.texi
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dv.c
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/libspeexenc.c
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/avidec.c
libavformat/dv.c
libavformat/dv.h
libavformat/flvenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/oggparsespeex.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes missing blue channel when switching from/to fullscren on OSX and libsdl
1.2.14. Fixes issue 548. Thanks for Jean First for the original patch and
for testing.
Signed-off-by: Marton Balint <cus@passwd.hu>
It's never been touched or used in any way since it was added in 2003
and only works for PIX_FMT_PAL8, which makes me conclude that nobody's
ever used it.
enable CODEC_CAP_DELAY to flush any remaining frames in the buffer.
Stop decoding when the FN_QUIT command is found so that a trailing seek table
isn't decoded as a normal frame.
decode all channels in the same call to avcodec_decode_audio3() so that
decoding will not stop after the first channel of the last frame.
Updated FATE reference. More valid audio is now decoded.
* qatar/master:
presets: rename presets directory
lavc: make avcodec_get_context_defaults3 "officially" public
lavf: replace av_new_stream->avformat_new_stream part II.
lavf,lavd: replace av_new_stream->avformat_new_stream part I.
lavf: add avformat_new_stream as a replacement for av_new_stream.
Use correct scaling table for bwd-pred MVs in second B-field
Ut Video decoder
Makefile: change presets extension to .avpreset
lavfi: add rgbtestsrc source, ported from MPlayer libmpcodecs
lavfi: add testsrc source
AVOptions: add documentation.
presets: update libx264 ffpresets
Conflicts:
Changelog
doc/APIchanges
doc/ffmpeg.texi
ffpresets/libx264-ipod320.ffpreset
ffpresets/libx264-ipod640.ffpreset
ffserver.c
libavcodec/avcodec.h
libavcodec/options.c
libavcodec/version.h
libavdevice/libdc1394.c
libavfilter/avfilter.h
libavfilter/vsrc_testsrc.c
libavformat/flvdec.c
libavformat/riff.c
libavformat/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
avpicture_get_size() returns the size of buffer required for avpicture_layout.
For pseudo-paletted formats (gray8...) this size does not include the palette.
However, avpicture_layout doesn't know this and still writes the palette. Consequently,
avpicture_layout writes passed the length of the buffer. This fixes it
by fixing avpicture_layout so that it doesn't write the palette for these formats.
Signed-off-by: Matthew Einhorn <moiein2000@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Seems to fix trac issue #569.
Sample is unfortunately not available, but it might be caused by
an index existing for non-existing audio stream (?).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
When scaling backward predicted MVs in second B-field, the scaling table is
opposite of that for P field pictures; i.e. first field P table will be used as
second field B table and second field P table will be used as first field B
table. This is not documented in the spec, but exists in the ref. decoder.
This fixes SA10139.
Signed-off-by: Martin Storsjö <martin@martin.st>
The licence was changed from GPL to LGPL with explicit approval from
the original author.
See thread:
Subject: [FFmpeg-devel] [PATCH] lavfi: add rgbtestsrc source,
ported from MPlayer libmpcodecs
Date: Mon, 11 Jul 2011 16:32:41 +0200
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
sunrast: Check for out of bounds reads
lavc: rename AV_ER_* options to AV_EF_* and rename AGGRESSIVE to BUFFER
lavc: replace API-bump-triggered AVCodecContext field change with shorter, non-conflicting name
Add libvpx presets.
doc/avtools: add forgotten part to stream specifiers description
swscale: prevent overflow during initialization
g722: Add a fate test for the encoder
fate: Add a target for creating a 16000 Hz mono synthetic audio file
macosx: use the default surface on newer sdl
Conflicts:
ffplay.c
libavcodec/avcodec.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The function strtok_r() is part of the POSIX.1 specification, but is not
available on some platforms. We provide an internal implementation, so we
do not need to rely on a platform implementation.
The new code avoids to call av_malloc/av_free when escaping is needed
(possibly faster), avoids an integer overflow in case of a huge
string, and provides feedback when a string cannot be escaped.
When a string cannot be escaped, a special string is printed instead.
* qatar/master:
avconv: add presets
rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
rtsp: Make the rtsp flags avoptions set via a define
rtpenc: Set a default video codec
avoptions: Fix av_opt_flag_is_set
rtp: Fix ff_rtp_get_payload_type
doc: Update the documentation on setting options for RTSP
rtsp: Remove the separate filter_source variable
rtsp: Accept options via private avoptions instead of URL options
rtsp: Simplify AVOption definitions
rtsp: Merge the AVOption lists
lavfi: port libmpcodecs delogo filter
lavfi: port boxblur filter from libmpcodecs
lavfi: add negate filter
lavfi: add LUT (LookUp Table) generic filters
AVOptions: don't segfault on NULL parameter in av_set_options_string()
avio: Check for invalid buffer length.
mpegenc/mpegtsenc: add muxrate private options.
lavf: deprecate AVFormatContext.file_size
mov: add support for TV metadata atoms tves, tvsn and stik
Conflicts:
Changelog
doc/filters.texi
doc/protocols.texi
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/internal.h
libavfilter/vf_boxblur.c
libavfilter/vf_delogo.c
libavfilter/vf_lut.c
libavformat/mpegtsenc.c
libavformat/utils.c
libavformat/version.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).
Signed-off-by: Martin Storsjö <martin@martin.st>
avconv doesn't map video streams to a muxer without specifying a
manual stream mapping if the default video codec is CODEC_ID_NONE.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the changes in 3b3ea34655,
"Remove all uses of deprecated AVOptions API", av_opt_flag_is_set
was broken, since it now uses av_opt_find, which doesn't return
named constants unless a unit to look for the constant in is given.
This broke enabling LATM encapsulated AAC output in RTP.
Signed-off-by: Martin Storsjö <martin@martin.st>
It was broken in 3b3ea34655
"Remove all uses of deprecated AVOptions API", where any
presence of a payload_type AVOption caused its value to
be returned, even if it wasn't set (and thus had the default
-1 value).
This caused the RTP muxer to be broken.
Signed-off-by: Martin Storsjö <martin@martin.st>
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.
This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use defines for shortening common parts, omit the .dbl named
initializer (since it's the first element in the union).
Signed-off-by: Martin Storsjö <martin@martin.st>
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the following additions:
* support to gray format
* support to yuva420p format
* parametric luma/chroma/alpha radius
* consistency check on the radius values, avoid crashes with invalid values
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* cus/stable:
ffplay: avoid window resize crash on osx with libsdl 1.2.14
ffplay: add delay multiple times to frame_timer if it is less than current time
ffplay: remove early frame drop functionality
ffplay: calculate target clock dynamically, make code more readable
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Replace our incomplete w32threads implementation with x264's pthreads
w32threads wrapper.
Relicensed to LGPL with kind permission by Pegasys Inc.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
If the picture queue is empty, or when the calculated delay is 0, frame_timer
is not increased but we are still displaying the old frame. When we eventually
get a frame, where the computed delay is positive, so we don't need to drop any
more frames, then it is best to update frame_timer to be as near as the current
time as it can.
This way we dont't have to wait several frames to add the necesarry delays to
frame_timer to reach current time, therefore there are no extra frame drops
after reaching a positive delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
The current impementation of early frame drops (dropping frames before adding
them to the picture queue) has multiple problems:
Even after gettin A-V sync, the frame droping continues until
VideoState->skip_frames reaches 1, which can take a lot of time causing useless
additional frame drops and bad AV-sync. This issue can be easily triggered with
for example changing the audio stream.
Also video_refresh currenly does not handle early skipped frames in every case,
for example if we skip a frame, then the last frame duration calculation will
compute the duration of the sum of the skipped frame and the duration of the
frame before that, and in compute_target_delay we may multiply this unusually
big delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since target clock is based on the current A-V delay, it is better calculate it
when we actually need it rather than when we put a picture in the picture
queue.
The patch also makes a code a bit more readable by renaming some delay
variables to duration, and converting compute_target_time to a delay
calculating function which does not modify the state. Factoring out the
iteration of the pictq to standalone function is also done in this patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
* qatar/master:
prores: get correct size for coded V plane if alpha is present
prores: do not set pixel format on codec init
pthread: prevent updating AVCodecContext from itself in frame_thread_free
pthread: copy coded frame dimensions in update_context_from_thread
vp8: prevent read from uninitialized memory in decode_mvs
vp8: force reallocation in update_thread_context after frame size change
vp8: fix return value if update_dimensions fails
matroskadec: fix out of bounds write
adpcmdec: calculate actual number of output samples for each decoder.
adpcmdec: check remaining buffer size before decoding next block in the ADPCM IMA WAV decoder.
adpcmdec: do not terminate early in ADPCM IMA Duck DK3 decoder.
adpcmdec: remove unneeded buf_size==0 check.
adpcmdec: remove unneeded zeroing of *data_size
dnxhdenc: fixed signed multiplication overflow
Conflicts:
tests/ref/fate/prores-alpha
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In particular: set output timebase to 1/output_sample_rate, fix output
PTS computation, and do not forget to copy properties values from the
input buffer.
The pixel format is not known until the frame header is parsed.
Guessing it here only causes trouble for the caller if the guess
turns out to be wrong (and actually causes very wrong output by
avconv/avplay).
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The low 32 bits of a multiplication are the same for signed
and unsigned operands. Casting to unsigned before multiplying
is thus equivalent while avoiding signed overflow, which is
undefined by the C99 standard.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master: (22 commits)
prores: add FATE tests
id3v2: reduce the scope of some non-globally-used symbols/structures
id3v2: cosmetics: move some declarations before the places they are used
shorten: remove the flush function.
shn: do not allow seeking in the raw shn demuxer.
avformat: add AVInputFormat flag AVFMT_NO_BYTE_SEEK.
avformat: update AVInputFormat allowed flags
avformat: don't unconditionally call ff_read_frame_flush() when trying to seek.
truespeech: use sizeof() instead of hardcoded sizes
truespeech: remove unneeded variable, 'consumed'
truespeech: simplify truespeech_read_frame() by using get_bits()
truespeech: decode directly to output buffer instead of a temp buffer
truespeech: check to make sure channels == 1
truespeech: check for large enough output buffer rather than truncating output
truespeech: remove unneeded zero-size packet check.
mlpdec: return meaningful error codes instead of -1
mlpdec: remove unnecessary wrapper function
mlpdec: only calculate output size once
mlpdec: validate that the reported channel count matches the actual output channel count
pcm: reduce pointer type casting
...
Conflicts:
libavformat/avformat.h
libavformat/id3v2.c
libavformat/id3v2.h
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The demuxer does not read the seektable, a parser is not possible without a
full decode, and no shorten decoder can handle random seeking because it needs
side info from the seektable.
Making 'samples' a pointer to uint8_t simplifies the DECODE
macro and reduces the amount of type casting overall.
This also fixes some signed overflows on left shift.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* hexene/stagefright:
tools/build_libstagefright: fetch android system headers and libraries
libstagefright: reindent after previous commit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This function intentionally overflows the signed range on
the left shift. Using this type-punning avoids errors from
the overflow checker without disabling this test globally.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Both v->fcm and v->field_mode are used in common code, now they won't be
reset for progressive frame after interlaced one causing writing past the
frame end for example.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Previous code could read 4 bytes past the end of the buffer on a RTCP_SR
packet or offset a pointer by an unchecked external value (payload_len),
though neither will reliably cause a crash or other misbehavior beyond
garbage timestamps.
Additionally, unknown RTCP packet types, even in compounded packets, are
now ignored as per RFC 3550 section 6.1, page 22, though currently this
only has any practical effect if a sender puts an unrecognized type
before RTCP_BYE in a compounded packet, or (incorrectly) does not put
RTCP_SR first.
Signed-off-by: John Brooks <john.brooks@bluecherry.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
'struct AVClass' is used in the code since
641c7afe3c, but AVClass is typedeffed as
an anonymous struct.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This will allow the caller to enumerate child contexts in a generic way
and since the API is recursive, it also allows for deeper nesting (e.g.
AVFormatContext->AVIOContext->URLContext)
This will also allow the new setting/reading API to transparently apply
to children contexts.
No application rely on this count being correct as far as
I know, but if we write a nonzero count value, it might just
as well be the right one.
Signed-off-by: Martin Storsjö <martin@martin.st>
DSS enables this automatically if streaming VOD over TCP. If
enabled, the server feeds packets faster than realtime, screwing
up RTCP NTP based timestamps.
Also, DSS doesn't indicate that this was indicated, if it was
enabled automatically (although if it was requested to be enabled,
a header saying that it was enabled is added, but this isn't
added if it is enabled automatically), making it even harder
to detect and work around properly without explicitly asking
for it to be disabled(/enabled, if we were able to support it).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.
Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.
Signed-off-by: Martin Storsjö <martin@martin.st>
This code relies on the result wrapping as for unsigned
values, and the sign is not used. Thus an unsigned type
is proper here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This avoids writing these entries doubly if transcoding from
flv to flv, since the muxer blindly writes any and all metadata
keys set, in addition to the fixed fields that the muxer
always writes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (23 commits)
fix AC3ENC_OPT_MODE_ON/OFF
h264: fix HRD parameters parsing
prores: implement multithreading.
prores: idct sse2/sse4 optimizations.
swscale: use aligned move for storage into temporary buffer.
prores: extract idct into its own dspcontext and merge with put_pixels.
h264: fix invalid shifts in init_cavlc_level_tab()
intfloat_readwrite: fix signed addition overflows
mov: do not misreport empty stts
mov: cosmetics, fix for and if spacing
id3v2: fix NULL pointer dereference
mov: read album_artist atom
mov: fix disc/track numbers and totals
doc: fix references to obsolete presets directories for avconv/ffmpeg
flashsv: return more meaningful error value
flashsv: fix typo in av_log() message
smacker: validate channels and sample format.
smacker: check buffer size before reading output size
smacker: validate number of channels
smacker: Separate audio flags from sample rates in smacker demuxer.
...
Conflicts:
cmdutils.h
doc/ffmpeg.texi
libavcodec/Makefile
libavcodec/motion_est_template.c
libavformat/id3v2.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The bit_rate_value_minus1 and cpb_size_value_minus1 elements
allow a wider range than get_ue_golomb() supports. This
adds a get_ue_golomb_long() function supporting up to 31
leading zeros, which is the maximum for these syntax
elements, and uses it in decode_hrd_parameters().
Signed-off-by: Mans Rullgard <mans@mansr.com>
The level_code expression includes a shift which is invalid in
those cases where the value is not used. Moving the calculation
to the branch where the result is used avoids these.
Signed-off-by: Mans Rullgard <mans@mansr.com>
These additions might overflow the signed range for large
input values. Converting to unsigned before the addition
rather than after avoids such undefined behaviour. The
result under normal two's complement wraparound remains
unchanged.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The PPS may contain a few trailing elements whose presence is
only signalled by data remaining after the the mandatory part
has been parsed. The current code fails to take into account
the rbsp_trailing_bits() when deciding whether to parse these
optional elements. Assuming no unnecessary padding bytes are
passed to this function, the optional elements are present if
either more than 8 extra bits remain or the remaining bits do
not form a valid rbsp_trailing_bits() after the mandatory PPS
elements have been parsed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The way this value is used, it should be an unsigned type.
While the numerical value has no meaning, unsigned wraparound
is relied upon.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (34 commits)
h264: reset h->ref_count in case of errors in ff_h264_decode_ref_pic_list_reordering()
error_resilience: fix the check for missing references in ff_er_frame_end() for H264
4xm: prevent NULL dereference with invalid huffman table
4xmdemux: prevent use of uninitialized memory
4xm: clear FF_INPUT_BUFFER_PADDING_SIZE bytes in temporary buffers
ptx: check for out of bound reads
tiffdec: fix out of bound reads/writes
eacmv: check for out of bound reads
eacmv: fix potential pointer arithmetic overflows
adpcm: fix out of bound reads due to integer overflow
anm: prevent infinite loop
avsdemux: check for out of bound writes
avs: check for out of bound reads
avsdemux: check for corrupted data
AVOptions: refactor set_number/write_number
AVOptions: cosmetics, rename static av_set_number2() to write_number().
AVOptions: cosmetics, move and rename static av_set_number().
AVOptions: split av_set_string3 into opt type-specific functions
avidec: fix signed overflow in avi_sync()
mxfdec: Fix some buffer overreads caused by the misuse of AVPacket related functions.
...
Conflicts:
Changelog
configure
libavcodec/ptx.c
libavcodec/ra144.c
libavcodec/vaapi_vc1.c
libavcodec/vc1.c
libavcodec/version.h
libavformat/4xm.c
libavformat/avidec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Keeping byte values read from the file as unsigned is consistent
with how they are subsequently used and avoids an undefined left
shift by 24 when bit 7 is set.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
mpeg4dec: use unsigned type for startcode in ff_mpeg4_decode_picture_header
mpeg124: use sign_extend() function
ac3dec: use get_sbits() instead of manually sign-extending
4xm: fix signed overflow
wmavoice: fix a signed overflow
mpegvideo_enc: fix a signed overflow
crc: fix signed overflow
fate: run avconv with -nostats flag
avtools: add -v as alias for -loglevel
avconv: always print stats with AV_LOG_INFO
doc/avconv: add forgotten output/per-stream info to -filter
avconv: add -stats option to enable/disable printing encoding progress
avconv: in output_packet() don't set decoded_data_size for video.
avconv: remove pointless always true condition
avconv: factorize common code in transcode_init()
zmbv: remove memcpy() of decoded frame
mpeg12enc: use sign_extend() function
h264pred: use unsigned types for pixel values, fix signed overflows
h264: fix signed overflows in x*0x01010101 expressions
h264pred: remove unused variables
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a signed overflow from i << 24 when i == 255 by
making i unsigned. The result of the shift is already
assigned to an variable of unsigned type.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It's now possible to disable printing stats during encoding with
-nostats, so there's no reason to print them differently depending on
whether it's last report or not.
* qatar/master:
lavf: fix signed overflow in avformat_find_stream_info()
vp8: fix signed overflows
motion_est: fix some signed overflows
dca: fix signed overflow in shift
aacdec: fix undefined shifts
bink: Check for various out of bound writes
bink: Check for out of bound writes when building tree
put_bits: fix invalid shift by 32 in flush_put_bits()
Conflicts:
libavcodec/bink.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
On the first iteration through this code, last_dts is always
INT64_MIN (AV_NOPTS_VALUE) and the subtraction overflows in
an invalid manner. Although the result is only used if the
input values are valid, performing the subtraction is still
not allowed in a strict environment.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In addition to avoiding undefined behaviour, an unsigned type
makes more sense for packing multiple 8-bit values.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Bug introduced by:
commit e3bf4e292c
Author: Anton Khirnov <anton@khirnov.net>
Date: Tue Aug 30 04:10:54 2011 +0200
ffmpeg: replace -vcodec/-acodec/-scodec with a better system.
The new option doesn't depend on its placement wrt -new* options (which
don't exist anymore) and works in a similar way as per-stream AVOptions.
-[vas]codec remain as aliases to -codec:[vas]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If flush_put_bits() is called when the 32-bit buffer is empty,
e.g. after writing a multiple of 32 bits, and invalid shift by
32 is performed. Since flush_put_bits() is called infrequently,
this additional check should have negligible performance impact.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (24 commits)
mpegps: Use av_get_packet() instead of poorly emulating it.
motionpixels: decode only the 111 complete frames for fate
mpc8: Check out of bound bands limit
xan: Prevent NULL dereference with missing palette
xan: Check for out of bound reads in xan_huffman_decode()
xan: Fixed out of bound accesses in xan_unpack()
motionpixels: Prevent calling init_vlc() with invalid parameters
shorten: Fix out of bound writes in fix_bitshift()
dsicinav: Check for out of bounds writes
tiertexseqv: Check for out of bound reads
quickdraw: Check for out of bound reads
dsicinav: Check for out of bounds reads
motionpixels: Fix the size of workspace buffers
motionpixels: Clear FF_INPUT_BUFFER_PADDING_SIZE bytes at the end of the temporary buffer
wmavoice: Check for corrupted extra data
wmavoice: Check for out of bound writes
xan: Prevent NULL dereferences with missing reference frame
bink: Prevent NULL dereferences with missing reference frame
wavpack: Reset internal state on corrupted blocks
wmapro: Validate the number of audio channels before using it
...
Conflicts:
libavcodec/h264.c
libavcodec/xan.c
tests/ref/fate/motionpixels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
wavpack_decode_block() supposes that it is called back with the exact
same buffer unless it has returned with an error. With multi-channels
files, wavpack_decode_frame() was breaking this assumption.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* hexene/stagefright:
libstagefright: start decode_thread() only after decode_frame() is called at least once.
libstagefright: mark the dummy frame as keyframe.
libstagefright: limit the output queue size
libstagefright: return EOS if CustomSource::read() is called after decode_thread() returns
libstagefright: set the correct frame size
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows concurrent decoding of the last field/frame, rather than
only the last slice, of data packets with multiple NAL units packed
together.
This will fix the slowdown reported in e.g. bug 52.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
vp6: Reset the internal state when aborting key frames header parsing
vp56: Release old pictures after a resolution changes
vp6: Check for huffman tree build errors
vp56: Check for missing reference frame data
cinepak: Fix invalid read access on extra data
vmd: fix segfaults on corruped streams
cook: Fix js_vlc_bits value validation for joint stereo
segafilm: Check for memory allocation failures in segafilm demuxer.
segafilm: Fix potential division by 0 on corrupted streams in the demuxer
Fixed segfault on corrupted sega streams in the demuxer.
Fixed deference of NULL pointer in motionpixels decoder.
libx264: support 9- and 10-bit output.
h264: correct implicit_weight for field-interlaced pictures.
mpegvideo: set correct offset for edge emulation buffer.
mpegvideo: fix position of bottom edge.
Conflicts:
libavcodec/motionpixels.c
libavcodec/mpegvideo.c
libavcodec/version.h
libavcodec/vmdav.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Well, probably an ABI breakage at least for 32bit, but hopefully it will go
unnoticed because the fix happened realtively soon.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix 'heigth' vs. 'height' typos.
lavc/lavf: use unique private classes.
lavc: use designated initializers for av_codec_context_class
Conflicts:
libavdevice/fbdev.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Now that the option was moved in the per-stream context,
the parsing is done before the time_base for the stream is decided.
This patch does the parsing in AV_TIME_BASE units and rescales the
timestamps later when the correct time base is known.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In case of av_new_packet() error, a correct return error code is raised,
the data memcpy is avoided, and pkt dts/pts are not assigned anymore
(since the defaults are good).
* qatar/master:
qcelpdec: cosmetics: do not add line break before opening bracket in 'for', 'while', 'if/else', and 'switch' statements.
qcelp: check output buffer size before decoding
qcelpdec: fix the return value of qcelp_decode_frame().
sipr: fix the output data size check and only calculate it once.
Synchronize various 4CCs and codec tags from FFmpeg.
qdm2: check output buffer size before decoding
Fix out of bound reads in the QDM2 decoder.
Check for out of bound writes in the QDM2 decoder.
ogg/celt: do not set sample_fmt in the demuxer
Conflicts:
libavcodec/avcodec.h
libavcodec/qdm2.c
libavformat/oggparsecelt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
id3v2: remove pointless casts
id3v2: read TXXX frames with two calls to decode_str() instead of one.
id3v2: don't discard the whole tag when encountering empty frames.
libvpx: fix build with older libvpx versions.
ARM: check for inline asm 'y' operand modifier support
Conflicts:
libavcodec/libvpxenc.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies useage for users who are used to vpxenc.
We continue to support the previous options with the exception of the flags
for which it would be slightly messy.
This fix is not ideal as it still limits the multithreading on field pictures
to the 2nd field only.
Ill try to fix it properly to allow both fields to decode concurrently but this
needs more work.
This bug exists since and was caused by:
commit ea6331f8bb
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Mon Jun 20 10:24:33 2011 -0400
h264-mt: fix deadlock in packets with multiple slices (e.g. MP4).
VPX_ERROR_RESILIENT_DEFAULT and VPX_ERROR_RESILIENT_PARTITIONS weren't
defined before 4cb0ebe5b27d35ccc2a78c1d16f2622ddef21f74 (CommitDate: Tue
Jun 28 11:10:17 2011)
The inline asm added in bf5d46d uses the 'y' modifier which
is only supported from gcc 4.5. This check allows building
with older compilers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
From 52.503s (~40fps) to 27.973sec (~80fps) decoding of 480p sintel
trailer, i.e. a ~2x speedup overall, on a Nexus S.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpc8: check output buffer size before decoding
mpc7: return error if packet is too small.
mpc7: check output buffer size before decoding
nellymoserdec: allocate float_buf only when decoding to int16
nellymoserdec: use NELLY_BUF_LEN instead of 128
nellymoserdec: use NELLY_BLOCK_LEN instead of 64 when appropriate.
nellymoserdec: allow user to request SAMPLE_FMT_FLT for output samples.
nellymoser: check output buffer size before decoding
win32: improve threading algorithm warning
Conflicts:
libavcodec/nellymoserdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
ffplay: use libswresample instead of av_audio_convert
audioconvert: add av_get_default_channel_layout public function
ffplay: use avctx->channels and avctx->freq before avcodec_open2 consistently
ffplay: remove now unnecessary request_channels, we set it now with options
ffplay: set request_channels to 2
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously ffplay expected SDL_AudioOpen to provide the requested sample rate
and channel number. This is no longer a requirement because this patch replaces
the audio convert function with libswresample's swr_convert which is capable of
handling different sample formats, sample rates and different number of
channels and different channel layouts.
The patch also removes the hardcoded 16bit samples assumption and uses
av_get_bytes_per_sample almost everywhere. The only exceptions are
the update_sample_display and video_audio_display functions, it
seemed too much of a headache to make them generic.
We also fix a tiny bug in sdl_audio_callback, we ensure that the number of
bytes when we put silence in the buffer is a multiple of the frame size.
This patch set the limit to 16.
For information, thoses previous commits:
41f7e2d11d5cbb0e70a0
assumed it was either 30 or 32.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: add video/audio/encoding flags to global_quality option
libvpxenc: use libvpx's own defaults for some parameters
vpxenc: add private options
Conflicts:
libavcodec/libvpxenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (34 commits)
dpcm: return error if packet is too small
dpcm: use smaller data types for static tables
dpcm: use sol_table_16 directly instead of through the DPCMContext.
dpcm: replace short with int16_t
dpcm: check to make sure channels is 1 or 2.
dpcm: misc pretty-printing
dpcm: remove unnecessary variable by using bytestream functions.
dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
dpcm: consistently use the variable name 'n' for the next input byte.
dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
dpcm: calculate and check actual output data size prior to decoding.
dpcm: factor out the stereo flag calculation
dpcm: cosmetics: rename channel_number to ch
avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
lavf: Avoid using av_malloc(0) in av_dump_format
dxva2_h264: pass the correct 8x8 scaling lists
dca: NEON optimised high freq VQ decoding
avcodec: reject audio packets with NULL data and non-zero size
dxva: Add ability to enable workaround for older ATI cards
latmenc: Set latmBufferFullness to largest value to indicate it is not used
...
Conflicts:
libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Copy the Inter 8x8 scaling list as second 8x8 matrix into DXVA2's
quantization matrix data structure instead of a potentially unset
Intra chroma scaling matrix.
Fix dxva2 decoding for some H264 samples.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
There is no valid reason the user should ever send such packets in the
first place, but the documentation for CODEC_CAP_DELAY states that the
codec is guaranteed not to get a NULL packet unless that capability is
set. That isn't true without preventing this case.
The workaround needs to be enabled per PCI ID which cannot be detected inside
libavcodec. So add a flag to manually enable the alternate behavior.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master:
dca: clear inactive subbands only once in qmf_32_subbands()
vf_unsharp: set default chroma size value to 5x5
vf_unsharp: fix out-of-buffer read
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Extradata should only be parsed from the avss, fiel, jp2h and alac atoms for
AVS, MJPEG, Motion JPEG 2000 and ALAC respectively.
This also fixes the mov demuxer coming up with bogus extradata for some
AVC-Intra samples due to the presence of fiel atoms.
Add Multimedia Wiki link.
Mark dead links with [dead]. Some can still be accessed through archive.org.
Update URLs for pages which have moved.
Replace duplicated links in adpcmenc.c with a note to see the ADPCM decoder
reference documents.
Writing zeros to the high entries in the array need only be
done once as the cutoff position is constant throughout the
loop.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In apply_unsharp(), when y is >= height, prevent out-of-buffer reading
from src, read from the last buffer line in src2 instead.
The check was implemented in the original unsharp libmpcodecs code and
lost in the port.
This also fixes output discrepancy between the two filters.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (23 commits)
avconv: Reformat s16 volume adjustment.
ARM: NEON optimised vector_fmac_scalar()
dca: use vector_fmac_scalar from dsputil
dsputil: add vector_fmac_scalar()
latmenc: Fix private options
vf_unsharp: store hsub/vsub in the filter context
vf_unsharp: adopt a more natural order of params in apply_unsharp()
vf_unsharp: rename method "unsharpen" to "apply_unsharp"
vf_scale: apply the same transform to the aspect during init that is applied per frame
vf_pad: fix "vsub" variable value computation
vf_scale: add a "sar" variable
lavfi: fix realloc size computation in avfilter_add_format()
vsrc_color: use internal timebase
lavfi: fix signature for avfilter_graph_parse() and avfilter_graph_config()
graphparser: prefer void * over AVClass * for log contexts
avfiltergraph: use meaningful error codes
avconv: Initialize return value for codec copy path.
fate: use 'run' helper for seek-test
fate: remove seek-mpeg2reuse test
Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
...
Conflicts:
doc/filters.texi
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/vf_scale.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also mark with a visible comment "FIXME realloc failure" places where
av_realloc seems to lack a proper test for failure.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_realloc_f helps avoiding memory-leaks in typical uses of realloc.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
av_size_mult helps checking for overflow when computing the size of a memory
area.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Replace sizeof((*avff)->formats)
with sizeof(*(*avff)->formats)
as the size of the array element is given by the pointed element
rather than by its pointer.
In particular fix computation with the pending patch when
sizeof(int64_t) != sizeof(int64_t *).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
avconv: use different variables for decoded and filtered frame.
avconv: add support for copying attachments.
matroskaenc: write attachments.
matroskadec: export mimetype of attachments as metadata.
avconv: factorize common code from new_*_stream()
doc/avconv: expand documentation for some options.
doc/avconv: document -timelimit.
Conflicts:
avconv.c
cmdutils.c
tests/codec-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Whitespace of the patch cleaned up by Aurel
Some of the issues have been reported by Steve Manzuik / Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 956c901c68)
Further suggestions from Kostya <kostya.shishkov@gmail.com> have been
implemented by Reinhard Tartler <siretart@tauware.de>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Makes the code less obfuscated and fixes encoding one video stream to
several outputs.
Also use avcodec_alloc_frame() instead of allocating AVFrame on stack.
Breaks me_threshold in avconv, as motion vectors aren't passed through
lavfi. They could be copied manually, but I don't think this misfeature
is useful enough to justify ugly hacks.
* qatar/master: (21 commits)
fate: allow testing with libavfilter disabled
x86: XOP/FMA4 CPU detection support
ws_snd: misc cosmetic clean-ups
ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
ws_snd: use memcpy() and memset() instead of loops
ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
ws_snd: make sure number of channels is 1
ws_snd: add some checks to prevent buffer overread or overwrite.
ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
flacdec: fix buffer size checking in get_metadata_size()
rtp: Simplify ff_rtp_get_payload_type
rtpenc: Add a payload type private option
rtp: Correct ff_rtp_get_payload_type documentation
avconv: replace all fprintf() by av_log().
avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
cmdutils: replace fprintf() by av_log()
avtools: parse loglevel before all the other options.
oggdec: add support for Xiph's CELT codec
sol: return error if av_get_packet() fails.
cosmetics: reindent and pretty-print
...
Conflicts:
avconv.c
cmdutils.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/oggparsecelt.c
libavformat/utils.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
wavpack_decode_block() supposes that it is called back with the exact
same buffer unless it has returned with an error. With multi-channels
files, wavpack_decode_frame() was breaking this assumption.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is simpler and more consistent to use "-preset $speed -qp 0" for
lossless. Also, the old presets encode slower.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Adds an additional check before reading the next block header and avoids a
potential integer overflow when checking the metadata size against the
remaining buffer size.
Specifying the payload type is useful when the type number has
already been negotiated before creating the stream, for example
in SIP protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
This way, all messages can be easily disabled with -loglevel.
Behavior change -- configuration is printed with AV_LOG_VERBOSE, so it's
not shown by default.
* qatar/master:
ppc: fix some pointer to integer casts
ppc: fix 32-bit PIC build
vmdaudio: fix decoding of 16-bit audio format.
lavf: do not set codec_tag for rawvideo
h264: check for out of bounds reads in ff_h264_decode_extradata().
flvdec: Check for overflow before allocating arrays
avconv: use correct output stream index when checking max_frames
avconv: remove fake coded_frame on streamcopy hack
Conflicts:
avconv.c
libavcodec/h264.c
libavcodec/ppc/asm.S
libavcodec/vmdav.c
libavformat/flvdec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavcodec/g729postfilter.c: In function ‘long_term_filter’:
libavcodec/g729postfilter.c:114: warning: unused variable ‘n’
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libavcodec/g729dec.c: In function ‘decode_frame’:
libavcodec/g729dec.c:662: warning: passing argument 3 of ‘ff_g729_postfilter’ from incompatible pointer type
libavcodec/g729postfilter.h:96: note: expected ‘int16_t *’ but argument is of type ‘int *’
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use uintptr_t instead of plain int. Without this change, the
comparisons will come out wrong for pointers in certain ranges.
Fixes random failures on ppc64. Also fixes some compiler warnings.
Signed-off-by: Mans Rullgard <mans@mansr.com>
On 32-bit ppc, the GOT pointer must be loaded manually.
This adds a "get_got" assembler macro to compute the
GOT address. The "movrel" macro is updated to take an
additional parameter containing the GOT address since
no register is reserved for this purpose on ppc32.
These changes have no effect on ppc64 builds.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The ff_h264_decode_extradata routine now checks for the buffer size
and pointer internally. This makes it possible to remove the external
checks in ff_h264_decode_init.
In decode_frame there was a size check missing because the buffer
gets tested prior to the invocation of ff_h264_decode_extradata().
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
The buffer size and pointer were not checked prior to testing the first
byte of the buffer. These were sometimes checked before calling, but it is
better to add it inside the function as it takes buf and size arguments.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Replaces a very hackish hack to fix the same issue (call instruction
overwriting stack variables).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
If the demuxer did not set a codec_tag, there is none and
inventing one makes no sense. This change stops the rawvideo
"decoder" over-writing user-supplied pixfmt with one derived
from the codec_tag. The pixfmt-codec_tag-pixfmt round-trip
is lossy since several pixfmts map to the same codec_tag.
This fixes fate-lavf-pixfmt with avfilter disabled.
Signed-off-by: Mans Rullgard <mans@mansr.com>
On allocation, the array length is multiplied by sizeof(int64_t),
this prevents the multiplication from overflowing.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
flvdec: Fix invalid pointer deferences when parsing index
configure: disable hardware capabilities ELF section with suncc on Solaris x86
Use explicit struct initializers for AVCodec declarations.
Use explicit struct initializers for AVOutputFormat/AVInputFormat declarations.
adpcmenc: Set bits_per_coded_sample
adpcmenc: fix QT IMA ADPCM encoder
adpcmdec: Fix QT IMA ADPCM decoder
permit decoding of multichannel ADPCM_EA_XAS
Fix input buffer size check in adpcm_ea decoder.
fft: avoid a signed overflow
mpegps: Handle buffer exhaustion when reading packets.
Conflicts:
libavcodec/adpcm.c
libavcodec/adpcmenc.c
libavdevice/alsa-audio-enc.c
libavformat/flvdec.c
libavformat/mpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When using suncc to build, the Solaris linker will mark
an executable with each instruction set encountered by
the Solaris assembler. As our libraries contain their own
guards for processor-specific code, instead suppress
generation of the HWCAPS ELF section on Solaris x86 only.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Or in other word, do not go to the next line when the text cannot be
rendered within the frame.
The rationale is that wrapping is hard to manage, as it depends on the
position of the text, thus making hard/impossible to compute the size
occupied by the rendered text, and makes the filter behavior
unpredictible with moving text (implemented in a pending patch).
Also the current wrapping code is very limited (for example it splits
words in the middle), and correctly handling wrapping requires much
more complexity (alignment, gravity, word parsing etc.), while
drawtext is meant to be only a thin wrapper around libfreetype, so
such advanced features don't belong here.
Unfortunately the output buffer size check assumes that the
input buffer is never over-consumed, thus this actually
also allowed to write outside the output buffer if "lucky".
Based on:
git.videolan.org/ffmpeg.git
commit 701d0eb185
libpostproc/postprocess_template.c: In function 'dering_MMX2':
libpostproc/postprocess_template.c:1045:5: error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'
libpostproc/postprocess_template.c:1045:5: error: 'asm' operand has impossible constraints
* qatar/master:
rtp: factorize dynamic payload type fallback
flvdec: Ignore the index if it's from a creator known to be different
cmdutils: move grow_array out of #if CONFIG_AVFILTER
avconv: actually set InputFile.rate_emu
ratecontrol: update last_qscale_for sooner
Fix unnecessary shift with 9/10bit vertical scaling
prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
prores: return more meaningful error values
prores: improve error message wording
prores: cosmetics: prettyprinting, drop useless parentheses
prores: lowercase AVCodec name entry
Conflicts:
cmdutils.c
libavcodec/proresdec_lgpl.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move the identical code in rtp_write_header() and
ff_sdp_write_media() inside ff_rtp_get_payload_type()
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes changing channels
It possibly might cause regressions but i cant avoid this without having a test
case that needs the locking code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Add LATM demuxer
avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
8svx/iff: fix decoding of compressed stereo 8svx files.
8svx: log an error message if output buffer is too small
8svx: check packet size before reading the initial sample value.
8svx: output 8-bit samples instead of 16-bit.
8svx: split delta decoding into a separate function.
mp4: Don't read an empty Decoder Config Descriptor
fate.sh: Ignore errors from rm command during cleanup.
fate.sh: Run git-pull in quiet mode to avoid console spam.
Apple ProRes decoder
rtmp: Make the input FLV parser handle data cut at any point
rv34: Check for invalid slices offsets
eval: test isnan(sqrt(-1)) instead of just sqrt(-1)
Conflicts:
Changelog
libavcodec/8svx.c
libavcodec/proresdec.c
libavcodec/version.h
libavformat/iff.c
libavformat/version.h
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change fixes a bug where seeking doesn't work properly for
matroska files that have the CUES element before the first cluster.
This bug was accidentally introduced a few months ago by my deferred CUES
loading patch<http://git.videolan.org/?p=ffmpeg.git;a=commit;h=31ad14c21e0735387ba8082c6e3436241f7ccfc8>
.
When the CUES element appears before the first cluster in the file, the data
is parsed and placed in matroska->index but that data is never added to the
seek index. Currently the transfer from matroska->index to the seek index
only happens when matroska_parse_cues() is called.
Matroska_parse_cues() only gets called on a seek if cues_parsing_deferred is
set. Cues_parsing_deferred only gets set if parsing the CUES requires
seeking past the first cluster. There is no code to handle the case where
CUES is before the first cluster.
This fix essentially restores the matroska->index processing that was
happening at the end of matroska_read_header() before I made my CUES
deferral change. In the case where CUES is before the first
cluster, matroska->index will have data and the seek index will be updated.
In the case where CUES is later in the file, matroska->index will be empty
and cues_parsing_deferred will be set so loading will happen later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.
Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da2a
The install directory being deleted might not be present if the build failed.
This can lead to annoying error output if the FATE client is run as a cronjob.
This makes the RTMP writing code able to handle FLV data
fed in arbitrarily small or large chunks, with multiple
consecutive packets in one write call, or having the FLV
packet header split over numerous write calls.
When used in conjunction with the flv muxer, the AVIO buffer
size still needs to be large enough to fit the initial metadata
packet though, since the size of that packet is written with a
seekback.
Signed-off-by: Martin Storsjö <martin@martin.st>
drawbox() expects a color specified in RGBA format, while the code was
providing a color with the components specified in the same order of
the input pixel format.
In particular, fix box coloring with AXXX RGB formats.
* qatar/master:
swfdec: Add support for sample_rate_code 0 (5512 Hz)
dct-test: factor out some common code and do whas was likely intended
doc: library versions need to be bumped in version.h
Revert "ffmpeg: get rid of useless AVInputStream.nb_streams."
Remove some forgotten AVCodecContext.palctrl usage.
lavc/utils: move avcodec_init() higher in the file.
lavc: replace some deprecated FF_*_TYPE with AV_PICTURE_TYPE_*
ac3dec: actually use drc_scale private option
lavc: undeprecate AVPALETTE_SIZE and AVPALETTE_COUNT macros
alsa: add missing header
msmpeg4: remove leftover unused debug variable declaration
Fix assert() calls that need updates after FF_COMMON_FRAME macro elimination.
Fix av_dlog invocations with wrong or missing logging context.
vf_yadif: add support to yuva420p
vf_yadif: correct documentation on the parity parameter
vf_yadif: copy buffer properties like aspect for second frame as well
oma: support for encrypted files
id3v2: add support for non-text and GEOB type tag frames
des: add possibility to calculate DES-CBC-MAC with small buffer
Conflicts:
ffmpeg.c
libavcodec/dct-test.c
libavformat/mpegts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes some nonsensical code by moving some should-be-shared
code to separate functions and using these.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This reverts commit 2cf8355f98.
AVInputStream.nb_streams tracks number of streams found at the
beginning, new streams may appear that ffmpeg doesn't know about. Fixes
crash in this case.
This extends the ID3v2 parser to allow for reading of non-text (i.e.
other than T***) meta tag frames providing a ff_id3v2_read_all()
function. An additional data structure 'ID3v2ExtraMeta' is introduced
for these tags since AVDictionary is string oriented and unsuitable
for binary data.
A parser for tag frames of type GEOB is implemented, which is needed
to extract keyring information from encrypted OMA files. GEOB data
is parsed into 'ID3v2ExtraMetaGEOB' data structures.
The routine to decode characters from different encodings to UTF-8,
formerly part of the read_ttag() function, is moved to its own
function. Because some tag frames contain subparts of unknown length,
the function is now also able to read until a null character is found.
In addition, the function now takes care of allocating a buffer long
enough to hold the decoded characters.
Signed-off-by: David Goldwich <david.goldwich@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This patch adds the possibility to calculate the DES-CBC-MAC of a
source buffer (i.e. the last block of the buffer encrypted in CBC
mode) without having to allocate a destination buffer that is as
long as the complete source buffer, but instead only 8 bytes
for the MAC.
Signed-off-by: David Goldwich <david.goldwich@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
rtmp: Don't blindly skip the 4 trailer bytes from the FLV packets
rtmp: Handle FLV packets written in more than one write call
rv34: Check for invalid slice offsets
Conflicts:
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(A minimal RTP packet is 12 bytes, but a minimal RTCP packet can be
much smaller, at least as small as 8 bytes.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
LOCAL_ALIGNED should work for all compilers/systems whereas
DECLARE_ALIGNED does not work on some (do not remember which though).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
gcc 4.6 no longer decrements esp to account for local variables.
Thus using call will end up overwriting some local variable.
So add an extra one it can safely clobber.
This is a huge hack because it's basically pure chance it works,
no idea how this is supposed to be done.
Fixes trac ticket #397.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
If not enough bytes are available, keep track of them and skip
them on next call.
In practice, if these trailer bytes are written in a separate
call, there is no other data written in this call, making it
fall into the "FLV packet too small" case currently - working,
but not as intended.
This patch makes the code more robust, handling all cases
except for having the FLV packet header split over multiple
write calls.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the FLV packet is larger than the AVIO buffer, a partial
FLV packet will be flushed to the RTMP protocol.
This commit handles the most common cases of FLV packets
being written in more than one call.
Signed-off-by: Martin Storsjö <martin@martin.st>
(code from mplayer see their svn for detailed authorship)
parameters must be integers as we do not have mplayers parser
also the stuff passed to it through non constant globals is not supported
this should be fixed in mplayer first.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: fix NULL reference in mov_write_tkhd_tag
rmdec: Reject invalid deinterleaving parameters
rv34: Fix potential overreads
rv34: Fix buffer size used for MC of B frames after a resolution change
rv34: Avoid NULL dereference on corrupted bitstream
rv10: Reject slices that does not have the same type as the first one
vf_yadif: add an option to enable/disable deinterlacing based on src frame "interlaced" flag
vsrc_color: set output pos values to -1
vsrc_color: add @file doxy
vsrc_buffer: remove duplicated file description
eval: implement not() expression
eval: add sqrt function for computing the square root
rmdec: use the deinterleaving mode and not the codec when creating audio packets.
lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
Conflicts:
doc/eval.texi
doc/filters.texi
libavcodec/rv10.c
libavfilter/vsrc_color.c
libavformat/rmdec.c
libavutil/avutil.h
libavutil/eval.c
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
recovery points that have non-zero recovery_frame_cnt resulted in
flagging incomplete frames as keyframes. This delays setting the
keyframe flag till the frame has been fully constructed.
The new syntax is more extensible and more user-friendly.
This breaks the previous syntax, should not be an issue as possibly
no-one is already using anullsrc.
Original code had the assumption of only one FLV packet per RTMP packet. But that assumption is incorrect for higher bit rates. Made changes to the code to accommodate more than one FLV packet per RTMP
+packet.
* qatar/master:
lavfi: add select filter
oggdec: fix out of bound write in the ogg demuxer
movenc: create an alternate group for each media type
lavd: add libcdio-paranoia input device for audio CD grabbing
rawdec: refactor private option for raw video demuxers
pcmdec: use unique classes for all pcm demuxers.
rawdec: g722 is always 1 channel/16kHz
Conflicts:
Changelog
configure
doc/filters.texi
libavdevice/avdevice.h
libavfilter/avfilter.h
libavfilter/vf_select.c
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes Ticket472
This may (or may not) cause problems with files that have no keyframes.
Plese open a bugreport or mail me if you have a file for which this fails.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pixel_format/video_size only apply to 'rawvideo' (==uncompressed) demuxer
and make no sense for the other raw (== containerless) demuxers. Keep
only the framerate option for those.
Also use unique classes for all raw video demuxers
Behaviour has been changed to fail negotiation if fields have been
left empty.
Patch was originally based on a patch by Hemanth.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Add aconvert filter to perform sample format, channel layout, and
packing format conversion.
The aconvert code depends on audio conversion code in libavcodec, so
this requires a dependency on libavcodec.
Based on previous work by S.N. Hemanth Meenakshisundaram and Mina Nagy
Zaki, performed for the GSoC 2010 and 2011.
This list is mainly useful to be directly used as a parameter by the
audio sink buffer, when there is no need to force a specific channel
layout.
The list of formats is generated programmatically, and avoids channel
layouts not supported by libavfilter (with more than 8 channels).
We preferred to avoid compile-time generation in order to avoid some
unnecessary complexity (build hacks, generating code maintainance
etc.).
A more compact way to express this list through the use of macros
should be possible when libavfilter will support more than 8 channels,
as there will be no need to filter out channel layouts with more than
8 channels.
See thread:
Subject: [FFmpeg-devel] [PATCH] lavfi: add ff_all_channel_layouts internal symbol
Date: Sat, 20 Aug 2011 15:33:16 +0200
A function name should tell what the function does rather than what
the function returns. This also avoids possible conflicts
(e.g. between a function and a public list of all supported formats),
and clarifies the code.
Breaks audio API/ABI, add a backward compatibility layer for video
API/ABI.
* qatar/master:
swscale: fix byte overreads in SSE-optimized hscale().
matroskadec: fix typo.
matroskadec: bail on parsing of incorrect seek index segments
Merged-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>
SSE-optimized hScale() scales up to 4 pixels at once, so we need to
allocate up to 3 padding pixels to prevent overreads. This fixes
valgrind errors in various swscale-tests on fate.
* qatar/master:
lavf utils: Remove write-only variable
lavf utils: Rename shadowing variable
smacker: fix a few off by 1 errors
Check for invalid VLC value in smacker decoder.
Check and propagate errors when VLC trees cannot be built in smacker decoder.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* hexene/scratchpad:
libstagefright: Explicitly free smart pointer objects
Changelog: Explain why C++ support is added, and that its optional
tools/build_libstagefright: rename build/libav to something neutral
Conflicts:
Changelog
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
sws: implement MMX/SSE2/SSSE3/SSE4 versions for horizontal scaling.
include stdint.h in adpcm_data.h
mpeg12: reorder functions to avoid ugly forward declarations
Fixed off by one packet size allocation in the smacker demuxer.
Check for invalid packet size in the smacker demuxer.
ape demuxer: fix segfault on memory allocation failure.
xan: Add some buffer checks
xan: Remove extra trailing newline
Fixed size given to init_get_bits() in xan decoder.
Conflicts:
libavcodec/mpeg12.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Having a string callback is much more simpler than a variable args
one for writers to deal with, especially when dealing with escaping.
This patch also introduces a local fast_asprintf() function which is
similar to av_asprintf() but avoids reallocating at each print (leading
to a performance issue).
Speed: from 3.9x to 9.6x speed improvement over C, and some small
(up to 15%) speed improvements over existing MMX code (particularly
for bigger filters).
I dont know on which system the prior check failed but an advanced
linker on some platform might have optimized func() out as unreachable.
Patch taken from http://floss.freebox.fr
Author: unknown
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoid retrying to read ASF index in files for every
attempt to seek. This makes a big difference to protocols
with slow seeking (for example http)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These changes were forgotten in commits:
lavfi: simplify signature for avfilter_get_audio_buffer() and friends
(commit 2c0317419b)
and
lavfi: consistently use int for sample_rate in AVFilterLink and AVFilterBufferRefAudioProps
(commit 4381bddc9f)
* qatar/master:
adpcm: split ADPCM encoders and decoders into separate files.
doc/avconv: fix typo.
rv34: check that subsequent slices have the same type as first one.
smacker demuxer: handle possible av_realloc() failure.
lavfi: add split filter from soc.
lavfi: add showinfo filter
libxavs: add private options corresponding to deprecated global options
Conflicts:
Changelog
libavcodec/adpcm.c
libavfilter/avfilter.h
libavfilter/vf_showinfo.c
libavfilter/vf_split.c
libavformat/smacker.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The additional parameters were never used and are complicating the
function interface. Also, they were inconsistent with the way the
video API works.
So this assumes that a requested samples buffer will have *always* the
format specified in the requested link.
This breaks audio filtering API and ABI.
This prevents some crashes when corrupted bitstream reports e.g. P-type
slice in I-frame. Official RealVideo decoder demands all slices to be
of the same type too.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This reduces problems when underlying protocol is not
seekable even if marked as such or if the file has been
cut short.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The code path using for mpegts over rtp doesn't open the demuxer using
mpegts_read_header,
so it never starts listening for PAT/SDT, only uses auto_guess
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the RIFF header stored size to figure out the expected AVI file size, instead
of the actual file. To work fully it requires handling failed avio_seek() instead
of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which previously
was not output during demuxing. This frame is now output to encoder, thus fate
diff update.
Merge video_qscale (set by -qscale) and audio_qscale (set by -aq) into
one 'qscale' field in the options context. Add a shortcut -q for -qscale
and make -aq an alias for -q:a.
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: Michael Niedermayer <michaelni@gmx.at>
The libavfilter SOC SVN repo is badly outdated, and some filters
supports a syntax different from the current code, so it is better do
not misled users into using it and just drop its reference.
* qatar/master:
Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
avconv: move audio_channels to the options context.
avconv: move *_disable to options context.
avconv: remove -[vas]lang options.
avconv: move codec tags to options context.
cljr: init_get_bits size in bits instead of bytes
indeo2: fail if input buffer too small
indeo2: init_get_bits size in bits instead of bytes
ffv1: Fixed size given to init_get_bits() in decoder.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Employ FF_ARRAY_ELEMS instead of manually calculating array length.
Fixed invalid access in wavpack decoder on corrupted bitstream.
Fixed invalid writes in wavpack decoder on corrupted bitstreams.
Fixed invalid access in wavpack decoder on corrupted extra bits sub-blocks.
rtpdec_asf: Fix integer underflow that could allow remote code execution
Conflicts:
libavformat/rtpdec_asf.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is a workaround to avoid links in the ffmpeg.c documentation randomly pointing to
different code in avconv.c
This should be removed once doxygen is fixed and we upgraded
doxygen 1.7.3 is the latests i tested and that still has this issue.
Fixed Ticket421
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If there is only 1 stream in an flv avformat_find_stream_info will continually
read until probesize is reached. This should stop it reading if the metadata
also claims there to be 1 stream.
Fixes MSVR-11-0088.
Credit: Jeong Wook Oh of Microsoft and Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes MSVR-11-0088
Credit: Jeong Wook Oh of Microsoft and Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's a hack which was created to allow for multiple options with
different defaults to refer to same field (e.g. 'b' vs 'ab'). There is
no need for it anymore.
Specifically, trellis, nr, me_range, me_method, subq, b_strategy,
keyint_min, chroma_me and coder.
Change their defaults to -1 and apply them after
x264_param_default_preset() if the user explicitly set them.
Specifically: gop_size, max_b_frames, scenechange_threshold, qmin, qmax,
max_qdiff, qblur, qcompress and refs.
Change their default values to -1 and only use them if the user
explicitly set them. Otherwise x264 defaults are used.
Move setting those options after x264_param_default_preset(), so they
don't get overwritten by it.
This is done in order to clarify the non-video-specific nature of the
buffersink code, as the result of the video/audio API unification of
the previous commit, and for improving overall consistency.
The new API is more generic (no distinction between audio/video for
pulling frames), and avoids code duplication.
A backward compatibility layer is kept for avoiding tools ABI breaks
(only for the video binary interface, audio interface was never used
in the tools).
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Author of the fix is ronald, the enabling & commit message are mine.
This fixes
commit 4e3e333a79
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Tue Jul 5 12:49:11 2011 -0700
swscale: error dithering for 16/9/10-bit to 8-bit.
Based on a somewhat similar idea in FFmpeg's swscale copy.
The Fix was originally commited in: (and i missed it due to the commit message)
commit 5c391a161a
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Fri Jul 8 14:39:04 2011 -0700
swscale: rename uv_off/uv_off2 to uv_off_px/byte.
This function should return number of samples decoded, not number of bytes
decoded.
Spotted by Uoti Urpala.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes improper flushing in the cases when the same frame is decoded in
several iterations (for being too large to fit into output buffer) and flush is
called mid-decoding and it also resets context in case of decoding errors.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
lavc: fix type for thread_type option
avconv: move format to options context
avconv: move limit_filesize to options context
avconv: move start_time, recording_time and input_ts_offset to options context
avconv: add a context for options.
cmdutils: allow storing per-stream/chapter/.... options in a generic way
cmdutils: split per-option code out of parse_options().
cmdutils: add support for caller-provided option context.
cmdutils: declare only one pointer type in OptionDef
cmdutils: move grow_array() from avconv to cmdutils.
cmdutils: move exit_program() declaration to cmdutils from avconv
http: Consider the stream as seekable if the reply contains Accept-Ranges: bytes
nutenc: add namespace to the api facing functions
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
ffserver.c
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The initial request contains "Range: 0-", which servers normally
have responded with "HTTP/1.1 206 Partial Content" reply with
a Content-Range header, which was used as indicator for seekability.
Apache, since 2.2.20, responds with "HTTP/1.1 200 OK" for these
requests, which is more friendly to caches and proxies, but the
seekability still is indicated via the Accept-Ranges: bytes header.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
AVOptions: fix av_set_string3() doxy to match reality.
cmdutils: get rid of dummy contexts for examining AVOptions.
lavf,lavc,sws: add {avcodec,avformat,sws}_get_class() functions.
AVOptions: add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find().
cpu detection: avoid a signed overflow
Conflicts:
avconv.c
cmdutils.c
doc/APIchanges
ffmpeg.c
libavcodec/options.c
libavcodec/version.h
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
currently libavformat only allows seeking if a request with "Range:
0-" results in a 206 reply from the HTTP server which includes a
Content-Range header. But according to RFC 2616, the server may also
reply with a normal 200 reply (which is more efficient for a request
for the whole file). In fact Apache HTTPD 2.2.20 has changed the
behaviour in this way and it looks like this change will be kept in
future versions. The fix for libavformat is easy: Also look at the
Accept-Ranges header.
Transforms the loose enumeration of issues that was to far below
in the text into a more verbose paragraph at the top. The new text
also mentions the possibility to propose new types of issues to
track on the development mailing list.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
This change makes the first part of the file easier to read quickly.
It makes it also easier to spot the URLs which might be one of the main
interests of many readers.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Correct a few grammar and spelling errors. Also improve consistency
by always referring to substatus (not substate).
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
This is the same code as for x86_64.
This is necessary because uname returns PPC64 if the hardware
is 64 bit, however the userland can still be fully 32 bit.
In that case FFmpeg fails to compile because some macros in the
asm code are set up incorrectly.
For details see https://bugs.gentoo.org/show_bug.cgi?id=341235
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reset it when we actually process an EOI, not when we actually
decide to ignore it for bug-workarounds, and reset it on
SOF to make it more reliable in case of missing EOI.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avconv: fix some bugs introduced in 630902a1e1
libmp3lame: fix typo
AVOptions: drop av_ prefix from static av_get_number().
libx264: use X264_THREADS_AUTO constant instead of 0.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.
These caused scrambled error messages to be printed and
floating point exceptions.
For example when there was no decoder available for a
stream.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
commit 630902a1e1
Author: Anton Khirnov <anton@khirnov.net>
Date: Wed Jul 6 08:49:07 2011 +0200
avconv: factor out initializing input streams.
These caused scrambled error messages to be printed and
floating point exceptions.
For example when there was no decoder available for a
stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It was mistakenly removed in 2cf8355f98,
not taking into account that new streams might appear in av_read_frame()
that avconv doesn't know about.
Fixes bug 24.
The new option doesn't depend on its placement wrt -new* options (which
don't exist anymore) and works in a similar way as per-stream AVOptions.
-[vas]codec remain as aliases to -codec:[vas]
New syntax contains an optional stream type, allowing to refer to n-th
stream of specific type.
Omitting stream number now maps all streams of the given type.
They are confusing, irregular and redundant -- -map already contains all
the information. Stream maps can now be parsed in opt_output_file().
Add a more user-friendly default behavior in case no maps are present.
Breaks -programid for now, but it never worked properly anyway. A better
solution will be written soon.
Before, it took an input and output file index, now it only takes an
input file and applies to the next output file.
Stream/chapter/program specification is now part of the option name and
the delimiter was changed from ',' to ':' to be consistent with the
similar feature for AVOptions.
Since RV40 is based on H.264, it uses the same interpolation filters and
functions for halfpel interpolation in one direction can be reused.
Unfortunately other cases are implemented differently and produce slightly
different output because of rounding.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The Makito encoder sets stream type 0x11 for AAC.
This patch should be reverted if it breaks decoding valid streams (and
the problem can't be fixed in the probe function).
This fixes ticket #348 for the few users who have a playback application
that supports Sisvel 3D. The other 99% have to wait until FFmpeg
supports arbitrary cropping.
* qatar/master:
Revert "h264: Properly set coded_{width, height} when parsing H.264."
isom: add missing AVC-Intra tags, rearrange list and update comments
avconv: remove stubs of crop* and pad* options
avconv: re-add nb_streams to InputFile.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It was mistakenly removed in 2cf8355f98,
not taking into account that new streams might appear in av_read_frame()
that avconv doesn't know about.
Fixes bug 24.
* qatar/master:
Use deinterleavers for demangling audio packets in RealMedia.
vf_scale: don't leak SWS context.
doxygen: drop another pointless star from pointer variable name
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes trac issue #343.
Carl Eugen Hoyos actually made a patch first, but I missed it because
trac does not send notification emails when an attachment is added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also extend the functionality to use the last found program to start the search
after that program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Unlike other containers RealMedia stores its audio packets in scrambled form,
with interleaver ID preceeding audio codec ID. Currently deinterleaving
decision is tied to the codec while it's possible to have non-default
deinterleaver with audio codec (like Int0 deinterleaver instead of specific
one for Sipro).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Similar to avcodec_find_best_pix_fmt(), but instead only compares two destination
pix fmts and selects one of these fmts as the best dest format when converting from
a given source format. Also, as opposed to avcodec_find_best_pix_fmt() which supports
only the first 64 defined pixel formats, this supports as input any of the defined
pixel formats.
Also replaces the old function to rely on the new function.
Signed-off-by: Matthew Einhorn <moiein2000@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fifo: add FIFO API test program, and fate test
fifo: add av_fifo_peek2(), and deprecate av_fifo_peek()
postprocess.c: filter name needs to be double 0 terminated
doxygen: fix wrong comment syntax, //< vs. ///<
doxygen: drop pointless star from pointer variable names
Replace deprecated av_find_stream_info() by avformat_find_stream_info().
xmv: eliminate superfluous zeroing of zero data
configure: fix typo in avconv dependency list
Conflicts:
configure
doc/APIchanges
libavutil/Makefile
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The new function provides a more generic interface than av_fifo_peek()
for peeking at a FIFO buffer data.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If the video queue is aborted, we have to pop the pending ALLOC event or wait
for the allocation to complete, because the current code assumes that
VideoState->pictq_windex does not change until the allocation is complete.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
mpeg12: propagate chunk decode errors and fix conditional indentation
vc1: fix VC-1 Pulldown handling.
VC1: Fix first/last row checks with slices
mp4: Handle non-trivial ES Descriptors.
vc1: properly zero coded_block[] edges on new slice entry.
Conflicts:
libavcodec/vc1dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Pulldown flags are being set incorrectly and AVFrame->repeat_pict is not
being set. Also, skipped frames exit header parsing too early and do not
set pulldown flags appropriately. Ticks_per_frame needs to be set and
time_base adjusted so player can extend frame duration by a field time.
This fixes problems encountered when attempting to transcode HD-DVD EVOB
files with HandBrake. Also makes these files play smoothly in avplay.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In some places 0/mb_height were used in place of start_mb_y/end_mb_y.
Fixes SA00049, SA00058, SA10091, SA10097, SA10131, SA20021, SA30030
Improves PSNR in SA00054, SA00059, SA00060, SA10096, SA10098, SA20022,
SA30031, SA30032, SA40012, SA40013
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Altough ffplay is working pretty well without using a lock manager, it is still
a multithreaded application calling libavcodec functions from multiple threads,
so using a lock manager is probably a good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
this fixes accuracy with normal ogg files while keeping support for ogg files
starting at times different from 0
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Deprecate corresponding global option.
Ideally all x264 private options should be generated automatically, but
x264 doesn't provide the API for this yet.
* qatar/master:
avconv: print the codecs names in the stream mapping.
avconv: move the avcodec_find_decoder() call to add_input_streams().
Windows Media Image decoder (WMVP/WVP2)
ac3enc: remove outdated TODO comment for apply_channel_coupling()
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/vc1dec.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libx264: only use ABR mode when the user explicitly set bitrate.
libx264: use medium preset by default.
mp2 encoder: make 128k the default bitrate.
movenc: use libx264 by default when possible for mov, mp4 and psp
avienc: saner default audio codec.
matroskaenc: saner default codecs.
avplay: add examples of how to specify size/pixel format through private options
lavc: add A|E|D flags to "ac" and "ar" options
Conflicts:
doc/ffplay.texi
libavcodec/libx264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The table is automatically generated from the definition of enum CodecID in
avcodec.h and contains the name of all known codecs, even those for which no
encoder nor decoder exists or is enabled.
The table is queried using the avcodec_get_name function.
If CONFIG_SMALL is true, the table is not compiled in; the avcodec_get_name
looks for names in the list of available decoders and encoders.
Clarify with an example what the user should use instead of the
deprecated (and no longer working) -s/-pix_fmt options.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Getting rid of globals are generally a good thing. The patch also makes
toggle_pause and step_to_next_frame use a function parameter instead of
the global cur_stream variable.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix setting of the corresponding AVCodecContext fields via
cmdutils.c:filter_codec_opts().
In particular, fix ffplay -ac and -ar options.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
WavPack demuxer: do not rely on index when timestamp is not in indexed range.
WavPack demuxer: store position of the first block in index.
WavPack decoder: implement flush function
avconv: Separate initialization from the main transcode loop.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the situation when there are not enough entries in the index
(e.g. on initial seek there's only one index entry in the index) and index
search returns just the last known entry. That causes seeking function just to
seek there instead of trying harder to get at the requested position.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Currently for multichannel audio position for the last block position is
stored in index (and used for seeking), which is obviously not correct.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Fix GCC warning:
asrc_abuffer.c: In function ‘init’:
asrc_abuffer.c:258: warning: passing argument 1 of ‘strtok_r’ discards qualifiers from pointer target type
Due to it in contrast to the decoder not setting up 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.
estimate_timings_from_pts would cause the parser to be reopened
on the same stream.
This fixes trac issue #360.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avconv: Set error code before before jumping to fail.
h264: fix PCM intra-coded blocks in monochrome case
Conflicts:
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
h264: correct implicit weight table computation for long ref pics
h264: correct the check for invalid long term frame index in MMCO decode
Conflicts:
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Correct computation of implicit weight tables when referencing pictures
that are marked for long reference.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The current check on MMCO parameters prohibits a "max long term frame index
plus 1" of 16 (frame idx of 15) for the "set max long term frame index" MMCO.
Fix this off-by-one error to allow the full range of legal values.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
avconv: Fix spelling errors.
aac: Only set sample rate and object type from ADTS if output hasn't been configured.
aac: Set SBR and PS to unsignalled during headerless and ADTS initialization.
aac: Only output configure if audio was found.
avconv: save two levels of indentation in flush_encoders()
avconv: factor flushing encoders out of output_packet().
avconv: factor out initializing input streams.
avconv: remove -intra option.
avconv: reset streamid_map between output files.
avconv: make timer_start a local var in transcode().
avconv: cosmetics, move OutputStream.
avconv: remove two unused macros.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Long term it would be nice to support error resilient reconfiguration
but right now setting this every frame does more harm than help.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master: (23 commits)
h264: hide reference frame errors unless requested
swscale: split hScale() function pointer into h[cy]Scale().
Move clipd macros to x86util.asm.
avconv: reindent.
avconv: rescue poor abused start_time global.
avconv: rescue poor abused recording_time global.
avconv: merge two loops in output_packet().
avconv: fix broken indentation.
avconv: get rid of the arbitrary MAX_FILES limit.
avconv: get rid of the output_streams_for_file vs. ost_table schizophrenia
avconv: add a wrapper for output AVFormatContexts and merge output_opts into it
avconv: make itsscale syntax consistent with other options.
avconv: factor out adding input streams.
avconv: Factorize combining auto vsync with format.
avconv: Factorize video resampling.
avconv: Don't unnecessarily convert ipts to a double.
ffmpeg: remove unsed variable nopts
RV3/4 parser: remove unused variable 'off'
add XMV demuxer
rmdec: parse FPS in RealMedia properly
...
Conflicts:
avconv.c
libavformat/version.h
libswscale/swscale.c
tests/ref/fate/lmlm4-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows using more specific implementations for chroma/luma, e.g.
we can make assumptions on filterSize being constant, thus avoiding
that test at runtime.
A pix_fmt_mask is obtained by bitwise or-ing two pixel format bits,
rather than doing a logical or.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
This fixed the problem where if there's more than 64 pixel formats
defined avcodec_find_best_pix_fmt() returns the wrong pixel format.
Signed-off-by: Matthew Einhorn <moiein2000@gmail.com>
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
First, container stores only DTS and not PTS as it was believed.
Second, multiple frames in a packet store timestamp instead of position
after the frame length.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
Revert "avconv: use stream copy by default when possible."
avconv: print stream copy information.
avconv: use stream copy by default when possible.
matroskaenc: vertical alignment.
matroskaenc: implement query_codec()
lavf: add avformat_query_codec().
lavc: add avcodec_get_type() for mapping codec_id -> type.
flvenc: use int64_t to store offsets
avconv: don't segfault on 0 input files.
Do not write ID3v1 tags by default
mpegts: log into an AVFormatContext rather than MpegTSContext.
Conflicts:
doc/APIchanges
libavcodec/version.h
libavformat/avformat.h
libavformat/mp3enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
this more than 4000 line file has less than 600 lines that are not in ffmpeg.c
and they are largely cosmetic changes from ffmpeg.c.
Thus the copyright statement from ffmpeg.c should be used.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This way the content of "vfilters" can be reused.
For example when the frame size changes, the filterchain is
reconfigured reusing again the vfilters value.
Use av_log(AV_LOG_LEVEL...) rather than av_dlog, the log is useful
even for "normal" debugging, and consistent with what is done in
ffmpeg.
Also change the message to achieve better consistency with the
corresponding ffmpeg message.
Use the value specified in the codec context for setting the
filterchain sample aspect ratio, when it is not specified in the
stream context.
Consistent with the ffmpeg behavior.
Fix trac issue #398.
avcodec_set_dimensions should be used for size changes to ensure
compatibility with future changes.
avctx->width/avctx->height may not be set to display-only dimensions.
Even more so since vc1dec.c would later set coded_width/height based
on this.
coded_width/coded_height should be used instead of width/height for
decoder setup.
This fixes playback of e.g. zz-mcr-nsqr.vc1 sample (containing
display width/height settings) in MPlayer and should fix a crash
with MPC: http://forum.doom9.org/showthread.php?t=162221.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Metadata currently is written only at the start of the file in normal
cases, when transcoding from a rtmp source metadata could be
written later and the offset recorded can exceed 32bit.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
ID3v1 are legacy tags with several limitations; furthermore
avconv/ffmpeg writes the tags in UTF-8 which probably has near-0
software support.
Add a -write_id3v1 option to be able to turn it on; disabled by default.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Its only reason for its existence was ffmpeg's inability to properly assign
AVOptions to streams. Now this is not a problem anymore, so 'ab' should
go.
The new option doesn't depend on its placement wrt -new* options (which
don't exist anymore) and works in a similar way as per-stream AVOptions.
-[vas]codec remain as aliases to -codec:[vas]
* qatar/master:
Fix NASM include directive
dsputil_mmx: Honor HAVE_AMD3DNOW
lavf,lavd: remove all usage of AVFormatParameters from demuxers.
jack: add 'channels' private option.
VC-1: fix reading of custom PAR.
Remove redundant and dubious video codec detection by its extradata
mpeg12: remove repeat-field code disabled since May 2002
patch checklist: suggest fate instead of regression tests
Turn on resampling on sudden size change instead of bailing out during recode.
avtools: reinitialise filter chain when input video stream changes dimensions
Conflicts:
Makefile
avconv.c
doc/developer.texi
ffplay.c
libavcodec/x86/dsputil_mmx.c
libavdevice/libdc1394.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFormatParameters are converted into corresponding private options in
av_open_input_file/stream() compat wrappers, so accessing them from
demuxers is redundant.
Since SDL has no audio buffer fullness info, one can get a much precise audio
clock based on the last time of the audio callback and the elapsed time since.
To achieve this I introduced the audio_current_pts and audio_current_pts_drift
variables (similar to video_current_pts and video_current_pts_drift) and
calculate them in the end of the audio callback, when VideoState->audio_clock
is already updated. The reference time I use is from the start of the audio
callback, because this way the amount of time used for audio decoding is not
interfereing with calculation.
I also replaced the audio_write_get_buf_size function with a calculated
variable because when the audio frame decoding is in progress audio_buf_size
and audio_buf_index are not stable, so using them from other threads are not a
good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aacenc: Fix number of coefficients used in a LFE channel.
aacenc: Fix a segfault with grouped psymodel.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In apply_unsharp(), when y is >= height, prevent out-of-buffer reading
from src, read from the last buffer line in src2 instead.
The check was implemented in the original unsharp libmpcodecs code and
lost in the port.
This also fixes output discrepancy between the two filters.
Allow to cache more than one frame (e.g. for filters which return
more than one frame when avfilter_request_frame() is called on them),
and do not discard previously cached frames when a new one is added.
In lavfi_read_header(), use the pad index designated in the inout for
linking an output to a sink, rather than always 0. Fix link creation
for filters with more than one output (e.g. the split filter).
FFmpeg writes data_size as AU_UNKNOWN_SIZE, make demuxer not
fail when data_size is set to this value.
Should fix trac issue #394.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
swscale: add dithering to yuv2yuvX_altivec_real
rv34: free+allocate buffer instead of reallocating it to preserve alignment
h264: add missing brackets.
swscale: use 15-bit intermediates for 9/10-bit scaling.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b2c087871dafc7d030b2d48457ddff597dfd4925':
Move x86util.asm from libavcodec/ to libavutil/.
Move x86inc.asm to libavutil/.
APIchanges: note error_recognition in lavf
lavf: add support for error_recognition, use it in avidec, and bump minor API version
avconv: change semantics of -map
avconv: get rid of new* options.
cmdutils: allow precisely specifying a stream for AVOptions.
configure: add missing CFLAGS to fix building on the HURD
libx264: Include hint for possible values for configuring libx264
cmdutils: allow ':'-separated modifiers in option names.
avconv: make -map_metadata work consistently with the other options
avconv: remove deprecated options.
avconv: make -map_chapters accept only the input file index.
Make a copy of ffmpeg under a new name -- avconv.
ffmpeg: add a warning stating that the program is deprecated.
Add weighted motion compensation for RV40 B-frames
RV3/4: calculate B-frame motion weights once per frame
Move RV3/4-specific DSP functions into their own context
mjpeg: propagate decode errors from ff_mjpeg_decode_sos and ff_mjpeg_decode_dqt
h264: notice memory allocation failure
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/ffplay.texi
doc/ffprobe.texi
doc/ffserver.texi
libavcodec/libx264.c
libavformat/avformat.h
libavformat/avidec.c
libavformat/version.h
tests/lavf-regression.sh
tests/lavfi-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It just does that part in scalar form, I doubt using a vector store
over 2 array would speed it up particularly.
The function should be written to not use a scratch buffer.
New syntax contains an optional stream type, allowing to refer to n-th
stream of specific type.
Omitting stream number now maps all streams of the given type.
They are confusing, irregular and redundant -- -map already contains all
the information. Stream maps can now be parsed in opt_output_file().
Add a more user-friendly default behavior in case no maps are present.
Breaks -programid for now, but it never worked properly anyway. A better
solution will be written soon.
Before, it took an input and output file index, now it only takes an
input file and applies to the next output file.
Stream/chapter/program specification is now part of the option name and
the delimiter was changed from ',' to ':' to be consistent with the
similar feature for AVOptions.
* qatar/master:
ac3enc: add macros for option names to make them more understandable.
ac3enc: fix 'channel_coupling' option to actually allow 'auto'.
cavs: fix some crashes with invalid bitstreams
ac3enc: clip coupling coordinates during calculation
Remove incorrect info in documentation of AVCodecContext.bits_per_raw_sample.
lavc: fix parentheses placement in avcodec_open2().
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is faster than using the clip_coefficients() function. Also,
clipping the coordinates before determining reuse is better since it
will compare coordinates that are closer to their final encoded values.
This fixes the two following warnings:
libavdevice/alsa-audio-dec.c:62:26: warning: unused variable ‘sw_params’ [-Wunused-variable]
libavdevice/alsa-audio-dec.c:109:15: warning: unused variable ‘st’ [-Wunused-variable]
This is consistent, allows custom handlers to print more info
(since they probably know about the AVFormatContext class
but not a demuxer-specific one) and also avoids issues due
to the class pointer being NULL for non-raw mpegts.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
lavc: make avcodec_init() static on next bump.
ac3enc: remove unneeded #include
ac3enc: restructure coupling coordinate reuse calculation
ac3enc: allow new coupling coordinates to be sent independently for each channel.
ac3enc: separate exponent bit counting from exponent grouping.
h264: propagate error return values for AV_LOG_ERROR-triggering events
aac: Don't attempt to output configure an invalid channel configuration.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4d58e4cb4ca8e04317fecea432ea3272a445877c':
Rename ffserver to avserver.
Rename ffprobe to avprobe.
Rename ffplay to avplay.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move bit counting to the bit allocation function. Move exponent grouping to
after bit allocation. This will allow for adjustment of bandwidth parameters
during bit allocation without having to do exponent grouping multiple times.
Previously the target clock of the next frame was calculated by using video_clock
which is modified from another thread...
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Old version was based on lots of guesswork which had different hacks based on
32-bit version field instead of splitting it into major/minor/micro version.
RV2 picture header parser also had a few places where it deviated from binary
decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
libx264: do not set pic quality if no frame is output
libx264: add preset, tune, profile and fastfirstpass private options
Conflicts:
libavcodec/libx264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The gradfun filter was already integrated natively in libavfilter.
Both filters issue the same output, and have a comparable performance.
See thread:
Subject: [FFmpeg-devel] [PATCH] lavfi: remove gradfun mp wrapper
Date: Fri, 5 Aug 2011 00:19:04 +0200
Previously both the buffer and buffersink elements were associated to
the "BUFFER" symbol in allfilters.c, so it was not possible to enable
one without the other.
void* pointer arithmetic semantics is undefined, use uint8_t* instead.
Also fix GCC warnings:
defaults.c: In function ‘set_common_formats’:
defaults.c:184: warning: pointer of type ‘void *’ used in arithmetic
defaults.c:189: warning: pointer of type ‘void *’ used in arithmetic
Update/extend documentation related to the link fields to set, and fix
a pads->filters braino.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
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.
Also remove channel_layout check as it is now negotiated.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
* qatar/master:
lsws: remove optimization debug logs in sws_init_context()
lsws: use array for storing the supported in/out information
Conflicts:
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The logged information is possibly false, and it tends to be outdated
after each change since the logging code needs to be manually updated.
Simplify and prevent confusing wrong debug messages.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Also remove the unnecessary isSupportedIn/Out macros.
Make the code more compact/readable, and simplify the access to
lsws-specific pixel format information.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
These tables contain only a 1-bit flag each. Combining them reduces
the data size and saves some instructions in the block decode loop.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is the value actually used by the decoder in speed-critical code.
The encoder uses these tables only in init code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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