* 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libmp3lame: remove outdated comment.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>
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libx264: fix indentation.
vorbis: fix overflows in floor1[] vector and inverse db table index.
win64: add a XMM clobber test configure option.
movdec: Parse the dvc1 atom
ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6
swscale: K&R formatting cosmetics for Blackfin code
frwu: lowercase the FRWU codec name
movdec: fix dts generation in fragmented files
fate: make acodec-ac3_fixed test output raw AC3
APIchanges: add missing commit hashes
swscale: implement MMX, SSE2 and AVX functions for RGB32 input.
ra144enc: drop pointless "encoder" from .long_name
bethsoftvideo: fix palette reading.
mpc7: use av_fast_padded_malloc()
mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes
doc: decoding Forward Uncompressed is supported
Fix a typo in the x86 asm version of ff_vector_clip_int32()
pcmenc: Do not set avpkt->size.
ff_alloc_packet: modify the size of the packet to match the requested size
Conflicts:
doc/APIchanges
libavcodec/libx264.c
libavcodec/mpc7.c
libavformat/isom.h
libswscale/Makefile
libswscale/bfin/yuv2rgb_bfin.c
tests/ref/fate/bethsoft-vid
tests/ref/seek/ac3_ac3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reduces the delay when opening the video with quicktime.
Idea-by: Maksym Veremeyenko <verem@m1stereo.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
rv34: frame-level multi-threading
mpegvideo: claim ownership of referenced pictures
aacsbr: prevent out of bounds memcpy().
ipmovie: fix pts for CODEC_ID_INTERPLAY_DPCM
sierravmd: fix audio pts
bethsoftvideo: Use bytestream2 functions to prevent buffer overreads.
bmpenc: support for PIX_FMT_RGB444
swscale: fix crash in fast_bilinear code when compiled with -mred-zone.
swscale: specify register type.
rv34: use get_bits_left()
avconv: reinitialize the filtergraph on resolution change.
vsrc_buffer: error on changing frame parameters.
avconv: fix -copyinkf.
fate: Update file checksums after the mov muxer change in a78dbada55
movenc: Don't store a nonzero creation time if nothing was set by the caller
bmpdec: support for rgb444 with bitfields compression
rgb2rgb: allow conversion for <15 bpp
doc: fix stray reference to FFmpeg
v4l2: use C99 struct initializer
v4l2: poll the file descriptor
...
Conflicts:
avconv.c
libavcodec/aacsbr.c
libavcodec/bethsoftvideo.c
libavcodec/kmvc.c
libavdevice/v4l2.c
libavfilter/vsrc_buffer.c
libswscale/swscale_unscaled.c
libswscale/x86/input.asm
tests/ref/acodec/alac
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
tests/ref/vsynth1/dnxhd_1080i
tests/ref/vsynth1/mpeg4
tests/ref/vsynth1/qtrle
tests/ref/vsynth1/svq1
tests/ref/vsynth2/dnxhd_1080i
tests/ref/vsynth2/mpeg4
tests/ref/vsynth2/qtrle
tests/ref/vsynth2/svq1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* qatar/master: (46 commits)
mtv: Make sure audio_subsegments is not 0
v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
avconv: add symbolic names for -vsync parameters
flvdec: Fix compiler warning for uninitialized variables
rtsp: Fix compiler warning for uninitialized variable
ulti: convert to new bytestream API.
swscale: Use standard multiple inclusion guards in ppc/ header files.
Place some START_TIMER invocations in separate blocks.
v4l2: list available formats
v4l2: set the proper codec_tag
v4l2: refactor device_open
v4l2: simplify away io_method
v4l2: cosmetics
v4l2: uniform and format options
v4l2: do not force interlaced mode
avio: exit early in fill_buffer without read_packet
vc1dec: fix invalid memory access for small video dimensions
rv34: fix invalid memory access for small video dimensions
rv34: joint coefficient decoding and dequantization
avplay: Don't call avio_set_interrupt_cb(NULL)
...
Conflicts:
Changelog
avconv.c
doc/APIchanges
doc/indevs.texi
libavcodec/adxenc.c
libavcodec/dnxhdenc.c
libavcodec/h264.c
libavdevice/v4l2.c
libavformat/flvdec.c
libavformat/mtv.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Rudimentary IODs support.
v410enc: fix output buffer size check
v410enc: include correct headers
fate: add -pix_fmt rgb48le to r210 test
flvenc: Support muxing 16 kHz nellymoser
configure: refactor list of programs into a variable
fate: add r210 decoder test
fate: split off Indeo FATE tests into their own file
fate: split off ATRAC FATE tests into their own file
fate: Add FATE tests for v410 encoder and decoder
ARM: fix external symbol refs in rv40 asm
westwood: Make sure audio header info is present when parsing audio packets
libgsm: Reset the MS mode of GSM in the flush function
libgsm: Set options on the right object
ARM: dca: disable optimised decode_blockcodes() for old gcc
Conflicts:
configure
libavformat/movenc.c
libavformat/movenc.h
tests/fate2.mak
tests/ref/acodec/alac
tests/ref/vsynth1/mpeg4
tests/ref/vsynth2/mpeg4
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
isom: sort and pretty-print codec_movaudio_tags[]
isom: remove pointless comments in codec_movaudio_tags[]
isom: remove commented-out tag for vorbis
movenc: write 'chan' tag for AC-3 in MOV
mov: add support for reading and writing the 'chan' tag
audioconvert: add some additional channel and channel layout macros
audioconvert: change 7.1 "wide" layout to use side surround channels
movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
doc: update documentation to use avconv
doc: update demuxers section
doc: extend external library coverage
doc: split platform specific information
doc: port the git-howto to texinfo
doc: provide fallback css and customize @float
doc: document fate in a texinfo
doxy: change hue value to match our green
Conflicts:
doc/fate.txt
doc/ffserver.texi
doc/general.texi
doc/muxers.texi
doc/protocols.texi
doc/t2h.init
libavformat/isom.c
libavformat/mov.c
libavutil/avutil.h
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
* qatar/master:
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>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
libopencore-amr: check output buffer size before decoding
libopencore-amr: remove unneeded buf_size==0 check.
libopencore-amr: remove unneeded frame_count field.
aac_latm: remove unneeded check for zero-size packet.
pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
pcmdec: return buf_size instead of src-buf.
avcodec: remove the Zork PCM encoder.
pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
pcmenc: remove unneeded sample_fmt check.
pcmdec: move number of channels check to pcm_decode_init()
pcmdec: remove unnecessary check for sample_fmt change
pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
pcmdec: do not needlessly set *data_size to 0
alacdec: remove unneeded NULL or zero-size packet checks.
alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
alacdec: ask for a sample for unsupported sample depths.
alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
alacdec: move some declarations to the top of the function
alacdec: always use get_sbits_long() for uncompressed samples
...
Conflicts:
libavcodec/pcm.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavfi: add select filter
oggdec: fix out of bound write in the ogg demuxer
movenc: create an alternate group for each media type
lavd: add libcdio-paranoia input device for audio CD grabbing
rawdec: refactor private option for raw video demuxers
pcmdec: use unique classes for all pcm demuxers.
rawdec: g722 is always 1 channel/16kHz
Conflicts:
Changelog
configure
doc/filters.texi
libavdevice/avdevice.h
libavfilter/avfilter.h
libavfilter/vf_select.c
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
APIchanges: fill in date and commit for request_sample_fmt
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
Add support for request_sample_format in ffmpeg and ffplay.
Add APIchanges entry for request_sample_fmt.
Add request_sample_fmt field to AVCodecContext.
Add float_interleave() to FmtConvertContext with x86-optimized versions.
Remove unused make variable SEEK_REFFILE
fate: remove redundant aref and vref references
fate: remove do_ffmpeg_nocheck function
fate: do not collect -benchmark output
mpegaudiodec: remove decode_end() function
fate: run aref and vref as regular tests
mpegaudio: sanitise compute_antialias_* names
mpeg12: add slice-threading checks to slice-threading initializers.
h264: copy pixel_shift between slice threading contexts.
mdec: enable frame-level multithreading.
mdec.c: fix overread.
Conflicts:
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/dca.c
libavcodec/h264.c
libavcodec/mdec.c
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/version.h
libavcodec/vorbisdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These tests create reference files used for psnr calculation in
the other codec tests. Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation. The checksum changes are due to
different rounding in the MDCT.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
