* 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>
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
* qatar/master:
matroska: Clear prev_pkt between seeks.
avutil: change default buffer size alignment for sample buffer functions
audemux: Add a sanity check for the number of channels
Remove libdirac decoder.
matroska: Add incremental parsing of clusters.
avconv: fix off by one check in complex_filter
mpegts: Try seeking back even for nonseekable protocols
swscale: K&R formatting cosmetics (part III)
Conflicts:
configure
doc/general.texi
doc/platform.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdirac.h
libavcodec/libdiracdec.c
libavformat/au.c
libavformat/mpegts.c
libswscale/input.c
tests/ref/seek/lavf_mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* 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.
* commit '3b266da3d35f3f7a61258b78384dfe920d875d29':
avconv: add support for complex filtergraphs.
avconv: make filtergraphs global.
avconv: move filtered_frame from InputStream to OutputStream.
avconv: don't set output width/height directly from input value.
avconv: move resample_{width,height,pix_fmt} to InputStream.
avconv: remove a useless variable from OutputStream.
avconv: get output pixel format from lavfi.
graphparser: fix the order in which unlabeled input links are returned.
avconv: change {input,output}_{streams,files} into arrays of pointers.
avconv: don't pass input/output streams to some functions.
Conflicts:
cmdutils.c
cmdutils.h
doc/ffmpeg.texi
ffmpeg.c
ffplay.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way we don't require a clearly defined corresponding input stream.
The result for the xwd test changes because rgb24 is now chosen instead
of bgra.
For the FATE test sample used, this only avoids a warning
message.
However for other samples like al05_44.mp4 the converted
file can be played only after this fix.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This will only work for DSEs that are first in a packet, but
that is enough to fix handling of the reference files in
fate-suite/aac (though most of them still have other issues).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Otherwise for muxers like e.g. latmenc that never call
avio_flush (and do not have a write_trailer function)
a part of the data will always be missing.
Also update references for the voc muxer, which was also
buggy before and did not write out all data.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (22 commits)
rv40dsp x86: use only one register, for both increment and loop counter
rv40dsp: implement prescaled versions for biweight.
avconv: use default channel layouts when they are unknown
avconv: parse channel layout string
nutdec: K&R formatting cosmetics
vda: Signal 4 byte NAL headers to the decoder regardless of what's in the extradata
mem: Consistently return NULL for av_malloc(0)
vf_overlay: implement poll_frame()
vf_scale: support named constants for sws flags.
lavc doxy: add all installed headers to doxy groups.
lavc doxy: add avfft to the main lavc group.
lavc doxy: add remaining avcodec.h functions to a misc doxygen group.
lavc doxy: add AVPicture functions to a doxy group.
lavc doxy: add resampling functions to a doxy group.
lavc doxy: replace \ with /
lavc doxy: add encoding functions to a doxy group.
lavc doxy: add decoding functions to a doxy group.
lavc doxy: fix formatting of AV_PKT_DATA_{PARAM_CHANGE,H263_MB_INFO}
lavc doxy: add AVPacket-related stuff to a separate doxy group.
lavc doxy: add core functions/definitions to a doxy group.
...
Conflicts:
ffmpeg.c
libavcodec/avcodec.h
libavcodec/vda.c
libavcodec/x86/rv40dsp.asm
libavfilter/vf_scale.c
libavformat/nutdec.c
libavutil/mem.c
tests/ref/acodec/pcm_s24daud
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().
Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
Decode output must be converted to rgb24 to avoid CRC difference
due to palette being stored in machine endianness.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Since those are pseudo-palette formats, swscale does not write
into data[1], swscale must initialize the palette properly itself.
This lead to frames that actually decoded as all-gray before.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Since we cannot specify decode parameters (and also because
it is better in principle) the 1-channel reference file
needs to be enabled, too.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The PSNR values are of varying usefulness, though at least
the DTS and AAC ones are useful with the right shift value.
Note: due to usage of floats some of these may fail on other
architectures.
In that case they should be converted into a CMD = stddev
FATE test, but it seems useful to try this way first.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
resample: allocate a large enough output buffer
fate: fix enc_dec_pcm tests with remote target
wmaenc: remove bit-exact hack
FATE: remove WMA acodec tests
FATE: add WMAv1 and WMAv2 encode/decode tests with fuzzy comparison
FATE: add AC-3 and E-AC-3 encode/decode tests with fuzzy comparison
qtrle: Use bytestream2 functions to prevent buffer overreads.
vqavideo: check malloc return values
x11grab: fix a memory leak exposed by valgrind
threads: fix old frames returned after avcodec_flush_buffers()
MPV: always mark dummy frames as reference
h264: fix deadlocks on incomplete reference frame decoding.
mpeg4: report frame decoding completion at ff_MPV_frame_end().
mimic: don't use self as reference, and report completion at end of decode().
Conflicts:
libavcodec/h264.c
libavcodec/qtrle.c
libavcodec/resample.c
libavcodec/vqavideo.c
libavdevice/x11grab.c
tests/ref/seek/wmav1_asf
tests/ref/seek/wmav2_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Wolfram Gloger <wmglo@dent.med.uni-muenchen.de>
Modify the parser initialization so that parsers can
set pict_type themselves. Use this in the mpegvideo parser
so that initial frames are not unconditionally I frames.
I have had this in my tree for several years.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-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.
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
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.
* 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>
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>
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
CC:libav-stable@libav.org
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:
mpegvideo_enc: only allocate output packet when we know there will be output
Add names for more channel layouts to the channel layout map.
sunrast: Add a sample request for RMP_RAW colormap.
avcodec: do not override pts or duration from the audio encoder
Add prores regression test.
Enable already existing rso regression test.
Add regression test for "sox" format muxer/demuxer.
Add dpx encoding regression test.
swscale: K&R formatting cosmetics for PowerPC code (part I/II)
img2: Use ff_guess_image2_codec(filename) shorthand where appropriate.
Clarify licensing information about files borrowed from libjpeg.
Mark mutable static data const where appropriate.
avplay: fix -threads option
dvbsubdec: avoid undefined signed left shift in RGBA macro
mlpdec: use av_log_ask_for_sample()
gif: K&R formatting cosmetics
png: make .long_name more descriptive
movdec: Adjust keyframe flagging in fragmented files
rv34: change most "int stride" into "ptrdiff_t stride".
Conflicts:
avprobe.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/mpegvideo_enc.c
libavcodec/pngenc.c
libavcodec/x86/v210-init.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_drawtext.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_setpts.c
libavfilter/vf_settb.c
libavformat/img2.c
libavutil/audioconvert.c
tests/codec-regression.sh
tests/lavf-regression.sh
tests/ref/lavf/dpx
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
mov: Use defines for sample flags in fragments
mov: Use defines for trun flags
mov: Use defines for tfhd flags
proresenc: force bitrate not to exceed given limit
vc1parse: call vc1_init_common().
wma: don't return 0 on invalid packets.
asf: prevent packet_size_left from going negative if hdrlen > pktlen.
mjpegb: don't return 0 at the end of frame decoding.
rtpdec: Identify incorrectly signalled H263
vp8dsp: split long line.
aiff: don't skip block_align==0 check on COMM-after-SSND files.
dpcm: ignore extra unpaired bytes in stereo streams.
mp3on4: require a minimum framesize.
mpc7: assign an error level + context to av_log() msg.
huffyuv: error out on bit overrun.
dct-test: Add the missing ff_ prefix to the altivec functions
dct-test: Remove a stray declaration of a nonexistent function
movenc: Write the unknown duration as 64 bit fields in ismv
movenc: Write track durations with all bits set if duration is unknown
Conflicts:
libavcodec/dct-test.c
libavcodec/wmadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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.
Seek beyond the end will now directly return an error instead
of claiming to succeed and then return EOF immediately on next read.
This change is because before 47e015e6f1
mkv seek incorrectly never failed.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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>
This is so that TS fragments produced by
http://code.google.com/p/httpsegmenter/
would be compatible with JW Player.
A new member variable prev_payload_key was added to MpegTSWriteStream
to help detect transition from non-key to key frame, so that
PAT/PMT would not be produced for every keyframe in intra-only videos.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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:
smacker: Sanity check huffman tables found in the headers.
smacker: remove dead store
qdm2: Check data block size for bytes to bits overflow.
mxfdec: Fix files with essence containers larger than 2 GiB.
mxfdec: Employ correct printf conversion specifiers for POSIX int types.
vc1: always read the bfraction element for interlaced fields
fate: add XWD image regression test
lavf: prevent infinite loops while flushing in avformat_find_stream_info
matroskadec: Pad AAC extradata.
ismindex: Fix build on mingw
Conflicts:
libavformat/mxfdec.c
libavformat/utils.c
tests/lavf-regression.sh
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>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
* 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: (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:
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: (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:
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.
* 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:
fate: whitespace cosmetics
fate: split off video codec FATE tests into their own file
fate: split off audio codec FATE tests into their own file
fate: split off Electronic Arts codec FATE tests into their own file
fate: split off QuickTime codec FATE tests into their own file
fate: split off voice codec FATE tests into their own file
fate: split off demuxer FATE tests into their own file
cosmetics: Drop unnecessary parentheses around return values.
fate: drop pointless _audio and _video suffixes from xan tests
qt-faststart: K&R reformatting; fix comment typos
FATE: Add test for H.264 MP4->annex.B bitstream filter.
Conflicts:
ffplay.c
tests/fate.mak
tests/fate/h264.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/qtrle.mak
tests/fate/real.mak
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: split off DPCM codec FATE tests into their own file
fate: split off PCM codec FATE tests into their own file
libvorbis: K&R reformatting cosmetics
libmp3lame: K&R formatting cosmetics
fate: Add a video test for xxan decoder
mpegvideo_enc: K&R cosmetics (line 1000-2000).
avconv: K&R cosmetics
qt-faststart: Fix up indentation
indeo4: remove two unused variables
doxygen: cleanup style to support older doxy
fate: add more tests for VC-1 decoder
applehttpproto: Apply the same reload interval changes as for the demuxer
applehttp: Use half the target duration as interval if the playlist didn't update
applehttp: Use the last segment duration as reload interval
lagarith: add decode support for arith rgb24 mode
Conflicts:
avconv.c
libavcodec/libmp3lame.c
libavcodec/mpegvideo_enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: always align height by 32 pixel
raw: add 10bit YUV definitions
nut: support 10bit YUV
mpegvideo_enc: separate declarations and statements
oma: make header compile standalone
vp3: Reorder some functions to fix VP3 build with Theora disabled.
build: fix standalone compilation of ADX encoder
build: fix standalone compilation of ADPCM decoders
build: fix standalone compilation of mpc7/mpc8 decoders
4xm: Use bytestream2 functions to prevent overreads
bytestream: add a new set of bytestream functions with overread checking
mpegts: Suppress invalid timebase warnings on DMB streams.
mpegts: Fix typo in handling sections in the PMT.
vc1dec: Use the right pointer type for the tmp pointer
Conflicts:
libavcodec/4xm.c
libavcodec/utils.c
libavcodec/vc1dec.c
libavcodec/vp3.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some libavifilter tests use NUT as output even if the produced
files were not decodable. The support for 10bit introduced in
432f0e5b7d and 91b1e6f0c changed the hashes.
* tjoppen/proper_mxf_track_linking:
mxfdec: Don't parse slices or DeltaEntryArrays
mxfdec: Remove dead/useless code
mxfdec: Hybrid demuxing/seeking solution
mxfdec: Add mxf_edit_unit_absolute_offset()
mxfdec: Replace zero IndexDurations with st->duration
mxfdec: Add "fake" index to MXFIndexTable to assist seeking
mxfdec: Add MXFIndexTables
mxfdec: Move mxf_read_packet*() near the bottom of the file
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The sample has an incomplete last frame. Decoding it is pointless.
The garbage produced was changed by the bitstream reader now
protecting against over-reads.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we don't
use delta entries or slices, only StreamOffsets.
OPAtom seeking basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
This fixes ticket #746.
This changes mxf_compute_ptses() to be used for MXFIndexTable, and also adds
code for computing the fake index to it.
This also temporarily disables PTS computation. A future patch will restore it.
* 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: (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>
* qatar/master:
ulti: Fix invalid reads
lavf: dealloc private options in av_write_trailer
yadif: support 10bit YUV
vc1: mark with ER_MB_ERROR bits overconsumption
lavc: introduce ER_MB_END and ER_MB_ERROR
error_resilience: use the ER_ namespace
build: move inclusion of subdir.mak to main subdir loop
rv34: NEON optimised 4x4 dequant
rv34: move 4x4 dequant to RV34DSPContext
aacdec: Use intfloat.h rather than local punning union.
Conflicts:
libavcodec/h264.c
libavcodec/vc1dec.c
libavfilter/vf_yadif.c
libavformat/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch is a generalization of what Michael Niedermayer
fixed in a single case.
The wmv8-drm fate test had been updated accordingly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
As far as I could see the only change is increased pos values,
which is as expected with additional metadata in the files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
I am not entirely sure the seek functionality is working correctly,
there are some strange cases like successful seek but no dts value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (21 commits)
Warn about avserver being broken.
avconv: drop code for special handling of avserver streams.
rawdec: don't set codec timebase.
lavf doxy: add muxing stuff to lavf_encoding group
lavf doxy: add demuxing stuff to lavf_decoding group
lavf doxy: expand/reword metadata API doxy.
lavf doxy: add installed headers to groups.
lavf doxy: add avio groups into the lavf_io group.
lavf doxy: rename lavf I/O group to lavf_io.
lavf doxy: add metadata docs to the main lavf group
ttadec: check channel count as read from extradata.
Add CLJR encoding and decoding regression tests
cljr: remove unused code
flacdec: Support for tracks in cuesheet metadata block
ptx: fix inverted check for sufficient data
flac muxer: fix writing of file header and STREAMINFO header from extradata
ptx: emit a warning on insufficient picture data
utvideo: add fate tests covering all codec variants
doc: update to refer to avconv
doc: remove some stale entries from the faq
...
Conflicts:
Changelog
avconv.c
doc/avconv.texi
doc/faq.texi
doc/ffplay.texi
doc/ffprobe.texi
doc/ffserver.texi
libavcodec/avcodec.h
libavcodec/cljr.c
libavformat/avformat.h
libavformat/riff.c
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.
* tjoppen/opatom_demuxing_and_seeking:
mxfdec: Index table driven demuxing and seeking
mxfdec: Compute packet offsets properly
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack
mxfdec: Parse more values in PartitionPack
mxfdec: Parse TemporalOffsets
mxfdec: av_dlog():ify 'no corresponding source package found'
mxfdec: Compute essence container offsets and lengths into mxf->partitions
mxfdec: Make mxf->partitions sorted by offset
mxfdec: Parse ThisPartition
mxfdec: Speed up metadata and index parsing
mxfdec: Make sure DataDefinition is consistent between material track and source track
mxfdec: Add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In 1bpp mode, interpret skip&0x80 as "start a new line" instead of "go to next line", this is almost the same except for the first line which was always skipped before and caused to try to write an extra line at the end of the frame (ticket #226).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The default of 9 gives different results on different FATE systems.
However the zlib test using compression level 6 works, so
try this instead.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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>
More specifically, PNG, v210, zlib and zmbv codecs.
zmbv needs vf_scale to be able to produce PAL8.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
While quality is bad, PAL8 support is needed to allow testing some
encoders that only support PAL8 input.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
h264: fix frame reordering code.
fate: Add a test for the VBLE decoder
doc: break some long lines in developer.texi
drawtext: make x and y parametric
drawtext: manage memory allocation better
drawtext: refactor draw_text
doc: remove space between variable and post increment in example code
Conflicts:
doc/developer.texi
doc/filters.texi
libavcodec/h264.c
libavfilter/vf_drawtext.c
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>
The qatar implementation makes no sense.
a muxer without timestamps is constant fps thus needs vsync.
the crc/mp5 are special cases that have timestamps yet allow any
nonsensical timestamps.
raw (yuv/rgb) video is constant fps thus needs vsync too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
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:
id3v2: fix doxy comment - 'machine byte order' makes no sense on char arrays
VC1: restore mistakenly removed code
twinvq: check output buffer size before decoding
twinvq: return an error when the packet size is too small
lavf: export some forgotten symbols with non-av prefixes.
swscale: update altivec yuv2planeX asm to new per-plane API.
swscale: make yuv2yuvX_10_sse2/avx 8/9/16-bits aware.
yuv2planeX10 SIMD
swscale: decide whether to use yuv2plane1/X on a per-plane basis.
swscale: reintroduce full precision in 16-bit output.
Split up yuv2yuvX functions
Split out yuv2yuv1 luma and chroma in order to make them generic DSP functions
lavc: replace references to deprecated AVCodecContext.error_recognition to use AVCodecContext.err_recognition
lavc: translate non-flag-based er options into flag-based ef options at codec open
add -err_filter AVOptions to access flag-based error recognition
h264_weight: initialize "height" function argument properly.
presets: spelling error in libvpx 1080p50_60
avplay: fix fullscreen behaviour with SDL 1.2.14 on Mac OS X
Conflicts:
ffplay.c
libavformat/libavformat.v
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
flvdec: Do not call parse_keyframes_index with a NULL stream
libspeexdec: include system headers before local headers
libspeexdec: return meaningful error codes
libspeexdec: cosmetics: reindent
libspeexdec: decode one frame at a time.
swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
Move timefilter code from lavf to lavd.
mov: add support for hdvd and pgapmetadata atoms
mov: rename function _stik, some indentation cosmetics
mov: rename function _int8 to remove ambiguity, some indentation cosmetics
mov: parse the gnre atom
mp3on4: check for allocation failures in decode_init_mp3on4()
mp3on4: create a separate flush function for MP3onMP4.
mp3on4: ensure that the frame channel count does not exceed the codec channel count.
mp3on4: set channel layout
mp3on4: fix the output channel order
mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
mp3on4: copy MPADSPContext from first context to all contexts.
fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
...
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_ps.c
libavcodec/h264dsp_template.c
libavcodec/h264idct_template.c
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavcodec/x86/h264dsp_mmx.c
libavdevice/Makefile
libavdevice/jack_audio.c
libavformat/Makefile
libavformat/flvdec.c
libavformat/flvenc.c
libavutil/pixfmt.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (47 commits)
lavc: hide private symbols.
lavc: deprecate img_get_alpha_info().
lavc: use avpriv_ prefix for ff_toupper4.
lavc: use avpriv_ prefix for ff_copy_bits and align_put_bits.
lavc: use avpriv_ prefix for ff_ac3_parse_header.
lavc: use avpriv_ prefix for ff_frame_rate_tab.
lavc: rename ff_find_start_code to avpriv_mpv_find_start_code
lavc: use avpriv_ prefix for ff_split_xiph_headers.
lavc: use avpriv_ prefix for ff_dirac_parse_sequence_header.
lavc: use avpriv_ prefix for some dv symbols used in lavf.
lavc: use avpriv_ prefix for some flac symbols used in lavf.
lavc: use avpriv_ prefix for some mpeg4audio symbols used in lavf.
lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
lavc: use avpriv_ prefix for ff_aac_parse_header().
lavf: hide private symbols.
lavf: use avpriv_ prefix for some dv functions.
lavf: use avpriv_ prefix for ff_new_chapter().
avcodec: add CODEC_CAP_DELAY note to avcodec_decode_audio3() documentation
avcodec: clarify the CODEC_CAP_DELAY note in avcodec_decode_video2()
avcodec: clarify documentation of CODEC_CAP_DELAY
...
Conflicts:
configure
doc/general.texi
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dv.c
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/libspeexenc.c
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/avidec.c
libavformat/dv.c
libavformat/dv.h
libavformat/flvenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/oggparsespeex.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
enable CODEC_CAP_DELAY to flush any remaining frames in the buffer.
Stop decoding when the FN_QUIT command is found so that a trailing seek table
isn't decoded as a normal frame.
decode all channels in the same call to avcodec_decode_audio3() so that
decoding will not stop after the first channel of the last frame.
Updated FATE reference. More valid audio is now decoded.
* qatar/master:
sunrast: Check for out of bounds reads
lavc: rename AV_ER_* options to AV_EF_* and rename AGGRESSIVE to BUFFER
lavc: replace API-bump-triggered AVCodecContext field change with shorter, non-conflicting name
Add libvpx presets.
doc/avtools: add forgotten part to stream specifiers description
swscale: prevent overflow during initialization
g722: Add a fate test for the encoder
fate: Add a target for creating a 16000 Hz mono synthetic audio file
macosx: use the default surface on newer sdl
Conflicts:
ffplay.c
libavcodec/avcodec.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
prores: get correct size for coded V plane if alpha is present
prores: do not set pixel format on codec init
pthread: prevent updating AVCodecContext from itself in frame_thread_free
pthread: copy coded frame dimensions in update_context_from_thread
vp8: prevent read from uninitialized memory in decode_mvs
vp8: force reallocation in update_thread_context after frame size change
vp8: fix return value if update_dimensions fails
matroskadec: fix out of bounds write
adpcmdec: calculate actual number of output samples for each decoder.
adpcmdec: check remaining buffer size before decoding next block in the ADPCM IMA WAV decoder.
adpcmdec: do not terminate early in ADPCM IMA Duck DK3 decoder.
adpcmdec: remove unneeded buf_size==0 check.
adpcmdec: remove unneeded zeroing of *data_size
dnxhdenc: fixed signed multiplication overflow
Conflicts:
tests/ref/fate/prores-alpha
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The pixel format is not known until the frame header is parsed.
Guessing it here only causes trouble for the caller if the guess
turns out to be wrong (and actually causes very wrong output by
avconv/avplay).
Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
* qatar/master: (22 commits)
prores: add FATE tests
id3v2: reduce the scope of some non-globally-used symbols/structures
id3v2: cosmetics: move some declarations before the places they are used
shorten: remove the flush function.
shn: do not allow seeking in the raw shn demuxer.
avformat: add AVInputFormat flag AVFMT_NO_BYTE_SEEK.
avformat: update AVInputFormat allowed flags
avformat: don't unconditionally call ff_read_frame_flush() when trying to seek.
truespeech: use sizeof() instead of hardcoded sizes
truespeech: remove unneeded variable, 'consumed'
truespeech: simplify truespeech_read_frame() by using get_bits()
truespeech: decode directly to output buffer instead of a temp buffer
truespeech: check to make sure channels == 1
truespeech: check for large enough output buffer rather than truncating output
truespeech: remove unneeded zero-size packet check.
mlpdec: return meaningful error codes instead of -1
mlpdec: remove unnecessary wrapper function
mlpdec: only calculate output size once
mlpdec: validate that the reported channel count matches the actual output channel count
pcm: reduce pointer type casting
...
Conflicts:
libavformat/avformat.h
libavformat/id3v2.c
libavformat/id3v2.h
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (24 commits)
mpegps: Use av_get_packet() instead of poorly emulating it.
motionpixels: decode only the 111 complete frames for fate
mpc8: Check out of bound bands limit
xan: Prevent NULL dereference with missing palette
xan: Check for out of bound reads in xan_huffman_decode()
xan: Fixed out of bound accesses in xan_unpack()
motionpixels: Prevent calling init_vlc() with invalid parameters
shorten: Fix out of bound writes in fix_bitshift()
dsicinav: Check for out of bounds writes
tiertexseqv: Check for out of bound reads
quickdraw: Check for out of bound reads
dsicinav: Check for out of bounds reads
motionpixels: Fix the size of workspace buffers
motionpixels: Clear FF_INPUT_BUFFER_PADDING_SIZE bytes at the end of the temporary buffer
wmavoice: Check for corrupted extra data
wmavoice: Check for out of bound writes
xan: Prevent NULL dereferences with missing reference frame
bink: Prevent NULL dereferences with missing reference frame
wavpack: Reset internal state on corrupted blocks
wmapro: Validate the number of audio channels before using it
...
Conflicts:
libavcodec/h264.c
libavcodec/xan.c
tests/ref/fate/motionpixels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Makes the code less obfuscated and fixes encoding one video stream to
several outputs.
Also use avcodec_alloc_frame() instead of allocating AVFrame on stack.
Breaks me_threshold in avconv, as motion vectors aren't passed through
lavfi. They could be copied manually, but I don't think this misfeature
is useful enough to justify ugly hacks.
* qatar/master:
rtp: factorize dynamic payload type fallback
flvdec: Ignore the index if it's from a creator known to be different
cmdutils: move grow_array out of #if CONFIG_AVFILTER
avconv: actually set InputFile.rate_emu
ratecontrol: update last_qscale_for sooner
Fix unnecessary shift with 9/10bit vertical scaling
prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
prores: return more meaningful error values
prores: improve error message wording
prores: cosmetics: prettyprinting, drop useless parentheses
prores: lowercase AVCodec name entry
Conflicts:
cmdutils.c
libavcodec/proresdec_lgpl.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Add LATM demuxer
avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
8svx/iff: fix decoding of compressed stereo 8svx files.
8svx: log an error message if output buffer is too small
8svx: check packet size before reading the initial sample value.
8svx: output 8-bit samples instead of 16-bit.
8svx: split delta decoding into a separate function.
mp4: Don't read an empty Decoder Config Descriptor
fate.sh: Ignore errors from rm command during cleanup.
fate.sh: Run git-pull in quiet mode to avoid console spam.
Apple ProRes decoder
rtmp: Make the input FLV parser handle data cut at any point
rv34: Check for invalid slices offsets
eval: test isnan(sqrt(-1)) instead of just sqrt(-1)
Conflicts:
Changelog
libavcodec/8svx.c
libavcodec/proresdec.c
libavcodec/version.h
libavformat/iff.c
libavformat/version.h
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
This uses the RIFF header stored size to figure out the expected AVI file size, instead
of the actual file. To work fully it requires handling failed avio_seek() instead
of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which previously
was not output during demuxing. This frame is now output to encoder, thus fate
diff update.
* 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:
fifo: add FIFO API test program, and fate test
fifo: add av_fifo_peek2(), and deprecate av_fifo_peek()
postprocess.c: filter name needs to be double 0 terminated
doxygen: fix wrong comment syntax, //< vs. ///<
doxygen: drop pointless star from pointer variable names
Replace deprecated av_find_stream_info() by avformat_find_stream_info().
xmv: eliminate superfluous zeroing of zero data
configure: fix typo in avconv dependency list
Conflicts:
configure
doc/APIchanges
libavutil/Makefile
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
h264: hide reference frame errors unless requested
swscale: split hScale() function pointer into h[cy]Scale().
Move clipd macros to x86util.asm.
avconv: reindent.
avconv: rescue poor abused start_time global.
avconv: rescue poor abused recording_time global.
avconv: merge two loops in output_packet().
avconv: fix broken indentation.
avconv: get rid of the arbitrary MAX_FILES limit.
avconv: get rid of the output_streams_for_file vs. ost_table schizophrenia
avconv: add a wrapper for output AVFormatContexts and merge output_opts into it
avconv: make itsscale syntax consistent with other options.
avconv: factor out adding input streams.
avconv: Factorize combining auto vsync with format.
avconv: Factorize video resampling.
avconv: Don't unnecessarily convert ipts to a double.
ffmpeg: remove unsed variable nopts
RV3/4 parser: remove unused variable 'off'
add XMV demuxer
rmdec: parse FPS in RealMedia properly
...
Conflicts:
avconv.c
libavformat/version.h
libswscale/swscale.c
tests/ref/fate/lmlm4-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
First, container stores only DTS and not PTS as it was believed.
Second, multiple frames in a packet store timestamp instead of position
after the frame length.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
swscale: add dithering to yuv2yuvX_altivec_real
rv34: free+allocate buffer instead of reallocating it to preserve alignment
h264: add missing brackets.
swscale: use 15-bit intermediates for 9/10-bit scaling.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b2c087871dafc7d030b2d48457ddff597dfd4925':
Move x86util.asm from libavcodec/ to libavutil/.
Move x86inc.asm to libavutil/.
APIchanges: note error_recognition in lavf
lavf: add support for error_recognition, use it in avidec, and bump minor API version
avconv: change semantics of -map
avconv: get rid of new* options.
cmdutils: allow precisely specifying a stream for AVOptions.
configure: add missing CFLAGS to fix building on the HURD
libx264: Include hint for possible values for configuring libx264
cmdutils: allow ':'-separated modifiers in option names.
avconv: make -map_metadata work consistently with the other options
avconv: remove deprecated options.
avconv: make -map_chapters accept only the input file index.
Make a copy of ffmpeg under a new name -- avconv.
ffmpeg: add a warning stating that the program is deprecated.
Add weighted motion compensation for RV40 B-frames
RV3/4: calculate B-frame motion weights once per frame
Move RV3/4-specific DSP functions into their own context
mjpeg: propagate decode errors from ff_mjpeg_decode_sos and ff_mjpeg_decode_dqt
h264: notice memory allocation failure
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/ffplay.texi
doc/ffprobe.texi
doc/ffserver.texi
libavcodec/libx264.c
libavformat/avformat.h
libavformat/avidec.c
libavformat/version.h
tests/lavf-regression.sh
tests/lavfi-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Revert "swscale: use 15-bit intermediates for 9/10-bit scaling."
swscale: use 15-bit intermediates for 9/10-bit scaling.
dct32: Add SSE2 ASM optimizations
Correct chroma vector calculation for RealVideo 3.
lavf: Add an option to discard corrupted frames
mpegts: Mark wrongly-sized packets as corrupted
mpegts: Move scan test to handle_packets
mpegts: Mark corrupted packets
mpegts: Reset continuity counter on seek
mpegts: Fix for continuity counter
mpegts: Silence "can't seek" warning on unseekable
apichange: add an entry for AV_PKT_FLAG_CORRUPT
avpacket: signal possibly corrupted packets
mpeg4videodec: remove dead code that would have detected erroneous encoding
aac: Remove some suspicious illegal memcpy()s from LTP.
bink: Eliminate unnecessary shadow declaration.
Conflicts:
doc/APIchanges
libavcodec/version.h
libavformat/avformat.h
libavformat/options.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Old version divided it wrong, which resulted in chroma drift (visible on FATE
sample too as dirty trails left by clouds).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
dnxhddec: optimise dnxhd_decode_dct_block()
rtp: remove disabled code
eac3enc: use different numbers of blocks per frame to allow higher bitrates
dnxhd: add regression test for 10-bit
dnxhd: 10-bit support
dsputil: update per-arch init funcs for non-h264 high bit depth
dsputil: template get_pixels() for different bit depths
dsputil: create 16/32-bit dctcoef versions of some functions
jfdctint: add 10-bit version
mov: add clcp type track as Subtitle stream.
mpeg4: add Mpeg4 Profiles names.
mpeg4: decode Level Profile for MPEG4 Part 2.
ffprobe: display bitstream level.
imgconvert: remove unused glue and xglue macros
Conflicts:
libavcodec/dsputil_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ffmpeg: fix some indentation
ffmpeg: fix operation with --disable-avfilter
simple_idct: remove disabled code
motion_est: remove disabled code
vc1: remove disabled code
fate: separate lavf-mxf_d10 test from lavf-mxf
cabac: Move code only used in the cabac test program to cabac.c.
ffplay: warn that -pix_fmt is no longer working, suggest alternative
ffplay: warn that -s is no longer working, suggest alternative
lavf: rename enc variable in utils.c:has_codec_parameters()
lavf: use designated initialisers for all (de)muxers.
wav: remove a use of deprecated AV_METADATA_ macro
rmdec: remove useless ap parameter from rm_read_header_old()
dct-test: remove write-only variable
des: fix #if conditional around P_shuffle
Use LOCAL_ALIGNED in ff_check_alignment()
Conflicts:
ffmpeg.c
libavformat/avidec.c
libavformat/matroskaenc.c
libavformat/mp3enc.c
libavformat/oggenc.c
libavformat/utils.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b5849f77095439e994b11c25e6063d443b36c228': (21 commits)
ac3enc: merge AC3MDCTContext with AC3EncodeContext.
ac3enc: prefer passing AC3EncodeContext rather than AVCodecContext
ac3enc: fix memleak
mpeg1video: add CODEC_CAP_SLICE_THREADS.
lavf: fix segfault in av_open_input_stream()
mpegtsenc: set Random Access indicator on keyframe start packets
lavf: Cleanup try_decode_frame() logic.
Replace some gotos that lead to single return statements by direct return.
build: move tests/seek_test.c to libavformat and reuse generic build rules
mxfenc: include needed header for ff_iso8601_to_unix_time() prototype
Add a check for strptime().
lavf: factor out conversion of ISO8601 string to unix time
wav: parse 'bext' metadata
wav: keep parsing until EOF if the input is seekable and we know the size of the data tag
wav: Refactor the tag checking into a switch statement
wav: make sure neither data_size nor sample_count is negative.
wav: refactor the 'fmt ' tag search and parsing.
wav: add an option for writing BEXT chunk
ffmpeg: get rid of a pointless limit on number of streams.
ffmpeg: remove an unused define.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* sws_32bit_integration:
regtests/sws: update checksums for recent changes
sws: dont mess with XInc when the code needing it isnt used
sws: Fix chroma init for 32bit buffers.
swscale: error dithering for 16/9/10-bit to 8-bit.
swscale: fix overflow in 16-bit vertical scaling.
swscale: fix crash in 8-bpc bilinear output without alpha.
swscale: fix 16-bit scaling when output is 8-bits.
sws: fix non native endian 9-15 bit input with 16bit out
sws: disable scale16 when int32 is used
sws: fix rgb -> 16bit
sws: fix uv overwrite in 32bt
sws: fix gray16_1
sws:ix yuv2rgb48_1_c_template()
sws: fix 16/32 bug from merge
swscale: for >8bit scaling, read in native bit-depth.
swscale: fix another yuv range conversion overflow in 16bit scaling. (cherry picked from commit 81cc7d0bd1)
swscale: fix yuv range correction when using 16-bit scaling. (cherry picked from commit e0b8fff6c7)
swscale: implement >8bit scaling support.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (28 commits)
mp3enc: write a xing frame containing number of frames in the file
lavf: update AVStream.nb_frames when muxing.
ffmpeg: remove unused variables from InputStream.
doc: update ffmpeg -ar and -ac documentation to reflect reality.
ffmpeg: remove pointless if (nb_input_files)
ffmpeg: merge input_files_ts_offset into input_files.
ffmpeg: merge input_codecs into input_streams.
ffmpeg: drop AV prefixes from struct names.
ffmpeg: deprecate loop_input and loop_output options
gif: add loop private option.
img2: add loop private option.
AVOptions: in av_opt_find() don't return named constants unless unit is specified.
x11grab: replace undocumented nomouse hackery with a private option.
dict: extend documentation.
lls: whitespace cosmetics
docs: Use proper markup for a literal command line option
docs: Remove a remark that isn't relevant any longer
docs: Explain how to regenerate import libraries with MSVC tools
docs: Mention that libraries for MSVC can be built with a cross compiler
docs: Remove old docs that mention setting up a build environment with lib.exe
...
Conflicts:
doc/ffmpeg.texi
doc/general.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/dnxhddata.c
libavformat/mp3enc.c
libavformat/utils.c
libavutil/Makefile
tests/copycooker.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We operated on 31-bits, but with e.g. lanczos scaling, values can
add up to beyond 0x80000000, thus leading to output of zeroes. Drop
one bit of precision fixes this.
* qatar/master:
ffserver: remove unused variable.
Remove unused and outdated TODO file.
gitignore: Drop individual .d ignore; it is already covered by a wildcard.
lavf: deprecate AVStream.quality.
bink: pass Bink version to audio decoder through extradata instead of codec_tag.
libpostproc: Remove disabled code.
flashsv: improve some comments and fix some wrong ones
flashsv: Eliminate redundant variable indirection.
flashsv: set reference frame type to full frame
flashsv: replace bitstream description by a link to the specification
flashsv: convert a debug av_log into av_dlog
flashsv: simplify condition
flashsv: return more meaningful error values
flashsv: cosmetics: break some overly long lines
flashsv: cosmetics: drop some unnecessary parentheses
swscale: amend documentation to mention use of native depth for scaling.
eval: add missing comma to tests.
eval: fix memleak.
H.264: make loopfilter bS const where applicable
Conflicts:
libavcodec/binkaudio.c
libavformat/bink.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
ARM: allow unaligned buffer in fixed-point NEON FFT4
fate: test more FFT etc sizes
dca: set AVCodecContext frame_size for DTS audio
YASM: Shut up unused variable compiler warning with --disable-yasm.
x86_32: Fix build on x86_32 with --disable-yasm.
iirfilter: add fate test
doxygen: Add qmul docs.
ogg: propagate return values and return more meaningful error values
H.264: fix overreads of qscale_table
Remove unused static tables and static inline functions.
eval: clear Parser instances before using
dct-test: remove 'ref' function pointer from tables
build: Remove deleted 'check' target from .PHONY list.
oggdec: Abort Ogg header parsing when encountering a data packet.
Add LGPL license boilerplate to files lacking it.
mxfenc: small typo fix
doxygen: Fix documentation for some VP8 functions.
sha: use AV_RB32() instead of assuming buffer can be cast to uint32_t*
des: allow unaligned input and output buffers
aes: allow unaligned input and output buffers
...
Conflicts:
libavcodec/dct-test.c
libavcodec/libvpxenc.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/h264_qpel_mmx.c
libavfilter/x86/gradfun.c
libavformat/oggdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (40 commits)
H.264: template left MB handling
H.264: faster fill_decode_caches
H.264: faster write_back_*
H.264: faster fill_filter_caches
H.264: make filter_mb_fast support the case of unavailable top mb
Do not include log.h in avutil.h
Do not include pixfmt.h in avutil.h
Do not include rational.h in avutil.h
Do not include mathematics.h in avutil.h
Do not include intfloat_readwrite.h in avutil.h
Remove return statements following infinite loops without break
RTSP: Doxygen comment cleanup
doxygen: Escape '\' in Doxygen documentation.
md5: cosmetics
md5: use AV_WL32 to write result
md5: add fate test
md5: include correct headers
md5: fix test program
doxygen: Drop array size declarations from Doxygen parameter names.
doxygen: Fix parameter names to match the function prototypes.
...
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/flvenc.c
libavformat/oggenc.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
cosmetics: fix some then/than typos
doxygen: Include libavcodec and libavformat examples into the documentation
avutil: elaborate documentation for av_get_random_seed
Add support for aac streams in mp4/mov without extradata.
aes: whitespace cosmetics
adler32: whitespace cosmetics
swscale: fix another yuv range conversion overflow in 16bit scaling.
Fix cpu flags test program
opt-test: Add missing braces to silence compiler warnings.
build: Eliminate obsolete test targets.
udp: Fix a compilation warning
swscale: Unbreak build with --enable-small
base64: add fate test
aes: improve test program and add fate test
adler32: make test program more useful and add fate test
swscale: fix yuv range correction when using 16-bit scaling.
aacenc: Make chan_map const correct
Conflicts:
Makefile
doc/examples/muxing-example.c
libavformat/udp.c
libavutil/random_seed.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
swscale: Add Doxygen for hyscale_fast/hScale.
fate: enable lavfi-pixmt tests on big endian systems
PPC: swscale: disable altivec functions for unsupported formats
fate: merge identical pixdesc_be/le tests
swscale: Add Doxygen for yuv2planar*/yuv2packed* functions.
build: call texi2pod.pl with full path instead of symlink
build: include sub-makefiles using full path instead of symlinks
swscale: update big endian reference values after dff5a835.
wavpack: skip blocks with no samples
cosmetics: remove outdated comment that is no longer true
build: replace some addprefix/addsuffix with substitution refs
avutil: Remove unused arbitrary precision integer code.
configure: Drop check for availability of ten assembler operands.
aacenc: Save channel configuration for later use.
aacenc: Fix codebook trellising for zeroed bands.
swscale: change prototypes of scaled YUV output functions.
swscale: re-add support for non-native endianness.
swscale: disentangle yuv2rgbX_c_full() into small functions.
swscale: split yuv2packed[12X]_c() remainders into small functions.
swscale: split yuv2packedX_altivec in smaller functions.
...
Conflicts:
Makefile
configure
libavcodec/x86/dsputil_mmx.c
libavfilter/Makefile
libavformat/Makefile
libavutil/integer.c
libavutil/integer.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: Add -U__STRICT_ANSI__ to CPPFLAGS on Cygwin and DOS.
aacdec: fix typo in scalefactor clipping check
fate: fix fate-h264-conformance-frext-pph10i4-panasonic-a crcs.
fate: update 9/10bit refs.
h264: Properly set coded_{width, height} when parsing H.264.
x86 asm: Add SECTION_TEXT to dct32_sse.asm.
Fix 9/10 bit in swscale.
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>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
Also remove code that overwrites the C versions of functions in
sws_init_swScale_altivec(), so that it uses the C functions of files
if no altivec-optimized version exists.
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix handling of input if not in native endianness, and add support for
9/10-bit output. This allows us to force endianness of YUV420P 9/10bit
in the H264/10bit fate tests, which should fix them on big-endian
systems.
framecrc returns different values when one swiches endianness,
this apparently has been missed by "the fork" who added the 10bit fate
tests. Sorry for missing this during the merge.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (30 commits)
AVOptions: make default_val a union, as proposed in AVOption2.
arm/h264pred: add missing argument type.
h264dsp_mmx: place bracket outside #if/#endif block.
lavf/utils: fix ff_interleave_compare_dts corner case.
fate: add 10-bit H264 tests.
h264: do not print "too many references" warning for intra-only.
Enable decoding of high bit depth h264.
Adds 8-, 9- and 10-bit versions of some of the functions used by the h264 decoder.
Add support for higher QP values in h264.
Add the notion of pixel size in h264 related functions.
Make the h264 loop filter bit depth aware.
Template dsputil_template.c with respect to pixel size, etc.
Template h264idct_template.c with respect to pixel size, etc.
Preparatory patch for high bit depth h264 decoding support.
Move some functions in dsputil.c into a new file dsputil_template.c.
Move the functions in h264idct into a new file h264idct_template.c.
Move the functions in h264pred.c into a new file h264pred_template.c.
Preparatory patch for high bit depth h264 decoding support.
Add pixel formats for 9- and 10-bit yuv420p.
Choose h264 chroma dc dequant function dynamically.
...
Conflicts:
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/alpha/dsputil_alpha.c
libavcodec/arm/dsputil_init_arm.c
libavcodec/arm/dsputil_init_armv6.c
libavcodec/arm/dsputil_init_neon.c
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/h264pred_init_arm.c
libavcodec/bfin/dsputil_bfin.c
libavcodec/dsputil.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_loopfilter.c
libavcodec/h264_ps.c
libavcodec/h264_refs.c
libavcodec/h264dsp.c
libavcodec/h264idct.c
libavcodec/h264pred.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/options.c
libavcodec/ppc/dsputil_altivec.c
libavcodec/ppc/dsputil_ppc.c
libavcodec/ppc/h264_altivec.c
libavcodec/ps2/dsputil_mmi.c
libavcodec/sh4/dsputil_align.c
libavcodec/sh4/dsputil_sh4.c
libavcodec/sparc/dsputil_vis.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/options.c
libavformat/utils.c
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/swscale_template.c
tests/ref/seek/lavf_avi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Looks better for some cases, worse for others, overall not much difference.
Its more correct though.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (23 commits)
doc: Check standalone compilation before submitting new components.
Fix standalone compilation of pipe protocol.
Fix standalone compilation of ac3_fixed encoder.
Fix standalone compilation of binkaudio_dct / binkaudio_rdft decoders.
Fix standalone compilation of IMC decoder.
Fix standalone compilation of WTV demuxer.
Fix standalone compilation of MXPEG decoder.
flashsv: K&R cosmetics
matroskaenc: fix memory leak
vc1: make overlap filter for I-frames bit-exact.
vc1dec: use s->start/end_mb_y instead of passing them as function args.
Revert "VC1: merge idct8x8, coeff adjustments and put_pixels."
Replace strncpy() with av_strlcpy().
indeo3: Eliminate use of long.
get_bits: make cache unsigned to eliminate undefined signed overflow.
asfdec: fix assert failure on invalid files
avfilter: check malloc return values.
Not pulled as reason for reindent is not pulled: mpegvideo: reindent.
nutenc: check malloc return values.
Not pulled due to much simpler solution in ffmpeg *: don't av_malloc(0).
...
Conflicts:
doc/developer.texi
libavcodec/Makefile
libavcodec/get_bits.h
libavcodec/mpegvideo.c
libavformat/Makefile
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9)
Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master:
ac3dec: fix processing of delta bit allocation information.
vc1: fix fate-vc1 after previous commit.
wmv3dec: fix playback of complex WMV3 files using simple_idct.
make av_dup_packet() more cautious on allocation failures
make containers pass palette change in AVPacket
introduce side information for AVPacket
Politic commits that have not been pulled:
Update regtest checksums after revision 6001dad.
Replace more FFmpeg references by Libav.
Replace references to ffmpeg-devel with libav-devel; fix roundup URL.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (32 commits)
libopencore-amr, libvo-amrwbenc: Allow enabling DTX via private AVOptions
libopencore-amr, libvo-amrwbenc: Only check the bitrate when changed
libopencore-amr, libvo-amrwbenc: Find the closest matching bitrate
libvo-*: Fix up the long codec names
libavcodec: Mark AVCodec->priv_class const
swscale: Factorize FAST_BGR2YV12 definition.
libvo-aacenc: Only produce extradata if the global header flag is set
lavf: postpone removal of public metadata conversion API
lavc: postpone removal of request_channels
lavc: postpone removal of audioconvert and sample_fmt wrappers
lavf: postpone removal of deprecated avio functions
libopencore-amr: Cosmetics: Rewrap and align
libopencore-amr, libvo-amrbwenc: Rename variables and functions
libopencore-amr: Convert commented out debug logging into av_dlog
libopencore-amr: Remove an unused state variable
libvo-amrwbenc: Don't explicitly store bitrate modes in the bitrate table
libopencore-amr: Remove a useless local variable
libopencore-amr, libvo-amrwbenc: Make the bitrate/mode mapping array static const
libopencore-amr, libvo-amrwbenc: Return proper error codes in most places
libopencore-amr: Don't print carriage returns in log messages
...
Conflicts:
doc/developer.texi
libavcodec/avcodec.h
libavcodec/libvo-aacenc.c
libavcodec/libvo-amrwbenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proto: include os_support.h in network.h
matroskaenc: don't write an empty Cues element.
lavc: add a FF_API_REQUEST_CHANNELS deprecation macro
avio: move extern url_interrupt_cb declaration from avio.h to url.h
avio: make av_register_protocol2 internal.
avio: avio_ prefix for url_set_interrupt_cb.
avio: AVIO_ prefixes for URL_ open flags.
proto: introduce listen option in tcp
doc: clarify configure features
proto: factor ff_network_wait_fd and use it on udp
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
dsputil: allow to skip drawing of top/bottom edges.
Split fate-psx-str-v3 into a video-only and audio-only test.
Conflicts:
libavcodec/dsputil.c
libavcodec/mpegvideo.c
libavcodec/snow.c
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This updates the seek test reference to match de11ee9. Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned. With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.
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>
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)
Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
The rm demuxer has timestamp bugs, so this test is sensitive to changes in
timestamp correction. The previous commit did not make output any better or worse
on this test, just different.
See https://roundup.ffmpeg.org/issue2288 for details.
Originally committed as revision 25432 to svn://svn.ffmpeg.org/ffmpeg/trunk
and add a test for regular GSM as fate-gsm.
Fixes a 8kHz sample from issue 113.
Originally committed as revision 25313 to svn://svn.ffmpeg.org/ffmpeg/trunk
Increase readability and robustness, as the test result is not going
to differ if the order of the pixfmts codes changes.
Originally committed as revision 24665 to svn://svn.ffmpeg.org/ffmpeg/trunk
The corresponding lavfi-pixfmts BE tests are not yet added, as there
are some bugs in the scaler (scaling rgba, argb, bgra, abgr, yuva420p)
which result in differences with the LE reference, and I cannot
visually check the generated files on BE.
Originally committed as revision 24657 to svn://svn.ffmpeg.org/ffmpeg/trunk
