* 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 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.
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 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>
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>
* 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.
* 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>
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>
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.
* 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>
