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>
* 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>
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:
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>
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:
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>
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>
* 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>
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>
Also define a codec capability for codecs that can handle
parameters changed externally between decoded packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
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>
* 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>
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>
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:
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>
* 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>
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>
* 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>
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.
