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: (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>
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>
* 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:
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>
* 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>
* 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>
* 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>
This is required for letting applications to create and destroy
AVFilterInOut structs in a convenient way.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* 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>
* 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>
* 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>
* 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>
* 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>
* 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>
* 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>
* 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>
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
* 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>
