commit 20e88d8618
Fix avui stream-copy.
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Convert them to zigzag order, as the rest of them are.
When I was adding support for 10-bit DNxHD, I just copy-pasted the
missing quant matrices from the spec. Now it turns out the existing
matrices in dnxhddata.c were in zigzag order. This resulted in wrong
quantization for 10-bit DNxHD. The attached patch fixes the problem by
converting 10-bit quant matrices to zigzag order.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This change introduces a basic decoder for 3GPP Timed Text subtitles,
also known as TX3G, Quicktime subtitles, or "movtext" in the existing
code.
This initial change doesn't attempt to parse styling information,
and just reads the plain text of the subtitles. I intend to add
support for styles eventually, but it's challenging due to a lack
of existing players that support them.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master: (29 commits)
lavfi: reclassify showfiltfmts as a TESTPROG
graph2dot: fix printf format specifier
swscale: yuv2planeX 8bit >=sse2 functions need aligned stack on x86-32.
vp8: loopfilter >=sse2 functions need aligned stack on x86-32.
amr: remove shift out of the AMR_BIT() macro.
dsputilenc: group yasm and inline asm function pointer assignment.
mov: use forward declaration of a function instead of a table.
Clarify Doxygen comment for FF_API_* #defines.
configure: simplify get_version()
Create version.h headers for libraries that lack them
gitignore: Use full path instead of relative path to specify patterns
mpegvideo: remove VLAs
Add XTEA encryption support in libavutil
Add Blowfish encryption support in libavutil
eval: Add the isinf() function and tests for it
flacdec: move lpc filter to flacdsp
flacdec: split off channel decorrelation as flacdsp
avplay: Add an option for not limiting the input buffer size
FATE: add a test for WMA cover art.
FATE: add a test for apetag cover art
...
Conflicts:
.gitignore
configure
ffplay.c
libavcodec/Makefile
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavcodec/ratecontrol.c
libavdevice/avdevice.h
libavfilter/Makefile
libavfilter/filtfmts.c
libavfilter/version.h
libavformat/mov.c
libavformat/version.h
libavutil/Makefile
libavutil/avutil.h
libavutil/version.h
libswscale/swscale.h
libswscale/x86/swscale_mmx.c
tests/fate/libavutil.mak
tests/lavfi-regression.sh
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is based on libav's implementation and
makes sure to compare output timestamps together.
It also reduces the differences with avconv.
The changes to the test reference files are caused
by an additional packet at the end, the timestamp
of the frame encoded by this packet is always
strictly below the limit stated by the -t option.
The "Default" style written in the header is ignored unless you explicit
it in the Dialogue events (it was valid, just ignored). This requires an
update of the SubRip test since the ASS output obviously changes.
* qatar/master:
flv: add support for G.711
doc: git: Add checklist with test steps to perform before pushing
flvenc: K&R formatting cosmetics
movenc: Add channel layouts for PCM.
Conflicts:
libavformat/flvenc.c
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
af_resample: fix format modifier in debug string for FF_API_SAMPLERATE64
segment: remove unnecessary <strings.h> include
fate: add snow hpel tests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Write chan atom for all audio tracks in mov mode movies.
mpegtsenc: use avio_open_dyn_buf(), zero pointers after freeing
doc/avconv: add some details about the transcoding process.
avidec: make scale and rate unsigned.
avconv: check output stream recording time before each frame returned from filters
avconv: split selecting input file out of transcode().
avconv: split checking for active outputs out of transcode().
avfiltergraph: make some functions static.
Conflicts:
ffmpeg.c
libavfilter/avfiltergraph.c
libavfilter/internal.h
libavformat/mpegtsenc.c
tests/ref/fate/acodec-alac
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ra144 uses floats so bitexactness cannot be guranteed
This should fix a long standing issue with icc
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avconv: extend -r to work on any input stream.
doc/avconv: expand documentation for the -s option.
avconv: don't print filters inserted by avconv in stream mappings.
avconv: merge configuration code for complex and simple filters
avconv: split configuring input filters out of configure_complex_filter
Conflicts:
configure
doc/ffmpeg.texi
ffmpeg.c
tests/ref/fate/idroq-video-encode
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f919cc7df6ab844bc12f89fe7bef4fb915a47725':
fate: fix acodec/vsynth tests for make 3.81
pcm_mpeg: fix number of consumed bytes to include the header.
avfilter: include required header file avfilter.h in video.h
x86: Avoid movs on BUTTERFLYPS when in AVX mode
x86: use new schema for ASM macros
fate: convert codec-regression.sh to makefile rules
fate: allow tests to specify unit size for psnr comparison
fate: teach videogen/rotozoom to output a single raw video stream
http: Add support for reusing the http socket for subsequent requests
http: Add support for using persistent connections
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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.
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>
* 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>
* 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>
* 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>
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: (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>
* 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:
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>
* 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>
* 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>
* 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>
* 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>
* 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: (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>
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.
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.
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>
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
* 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>
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>
* 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>
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
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>
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>
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>
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.
* 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>
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.
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.
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.
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 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.
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>
* 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.
* 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.
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>
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>
* 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>
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:
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>
* 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>
* 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>
(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>
* 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>
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: (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>
* 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>
* 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>
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.
