| Commit message (Collapse) | Author | Age | Files | Lines |
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This fixes at least issues with empty lines, and also allows CRLF lines
(in case a user makes its own preset on a MS plateform).
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Fixes Ticket1260, Ticket1267
Signed-off-by: Michael Niedermayer <[email protected]>
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This allows non integer scales.
Signed-off-by: Michael Niedermayer <[email protected]>
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Currently it always exits with an error when more than
one position is specified.
Fixes trac issue #1266.
Signed-off-by: Reimar Döffinger <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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* qatar/master:
FATE: use updated reference for aac-latm_stereo_to_51
avconv: use libavresample
Add libavresample
FATE: avoid channel mixing in lavf-dv_fmt
Conflicts:
Changelog
Makefile
cmdutils.c
configure
doc/APIchanges
ffmpeg.c
tests/lavf-regression.sh
tests/ref/lavf/dv_fmt
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <[email protected]>
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* qatar/master:
avconv: fix a segfault on -c copy with -filter_complex.
isom: Support more DTS codec identifiers.
Merged-by: Michael Niedermayer <[email protected]>
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Fixes Ticket1138
Based on a patch by: Eelco Lempsink <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Not all processing goes through filters,
so processing can continue even when
all filtergraphs have reached EOF on all sinks.
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It allows ffmpeg filtering logic to work with filters
that do not implement the poll_frame method,
such as split or tile.
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Also fix a typo in the next line.
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* qatar/master:
avcodec: remove AVCodecContext.dsp_mask
avconv: fix a segfault when default encoder for a format doesn't exist.
utvideo: general cosmetics
aac: Handle HE-AACv2 when sniffing a channel order.
movenc: Support high sample rates in isomedia formats by setting the sample rate field in stsd to 0.
xxan: Remove write-only variable in xan_decode_frame_type0().
ivi_common: Initialize a variable at declaration in ff_ivi_decode_blocks().
Conflicts:
ffmpeg.c
libavcodec/utvideo.c
Merged-by: Michael Niedermayer <[email protected]>
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Fixes a segfault with Ogg output, libtheora not compiled in
and no codec specified.
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* qatar/master:
dv: Initialize encoder tables during encoder init.
dv: Replace some magic numbers by the appropriate #define.
FATE: pass the decoded output format and audio source file to enc_dec_pcm
FATE: specify the input format when decoding in enc_dec_pcm()
x86inc: support AVX abstraction for 2-operand instructions
configure: detect PGI compiler and set suitable flags
avconv: check for an incompatible changing channel layout
avio: make AVIOContext.av_class pointer to const
nutdec: add malloc check and fix const to non-const conversion warnings
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <[email protected]>
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* qatar/master:
aacenc: Fix issues with huge values of bit_rate.
dv_tablegen: Drop unnecessary av_unused attribute from dv_vlc_map_tableinit().
proresenc: multithreaded quantiser search
riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header
avconv: only set the "channels" option when it exists for the specified input format
avplay: update get_buffer to be inline with avconv
aacdec: More robust output configuration.
faac: Fix multi-channel ordering
faac: Add .channel_layouts
rtmp: Support 'rtmp_playpath', an option which overrides the stream identifier
rtmp: Support 'rtmp_app', an option which overrides the name of application
avutil: add better documentation for AVSampleFormat
Conflicts:
libavcodec/aac.h
libavcodec/aacdec.c
libavcodec/aacenc.c
Merged-by: Michael Niedermayer <[email protected]>
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* qatar/master:
indeo3: add parens around some macro arguments
h264: use proper PROLOGUE statement for a function using 8 registers.
doc: Update sample Vim config with suitable (function) indentation settings.
dv: Merge dvquant.h into dvdata.c where all other DV tables reside.
dv: Move static tables only used in one place to where they are used.
graphparser: set next to NULL on an entry extracted from inputs list
doc/filters: update documentation.
avconv: flush decoders immediately after an EOF.
avconv: send EOF to vsrc_buffer.
avconv: reindent.
Conflicts:
doc/filters.texi
ffmpeg.c
Merged-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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* 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 <[email protected]>
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Necessary to make it possible to copy teletext streams
(trac issue #531).
Signed-off-by: Reimar Döffinger <[email protected]>
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Audio timestamps are passed through by default and when the input
doesnt contain clean timestamps this can lead to non monotonicity
errors. (rounding to a course timebase can cause this too)
Print a warning when the errors in the timestamps are large
Fixes Ticket1167 (regression since timestamps are passed through)
This is a generic workaround that is intended to handle
slightly incorrect input files. It is very possible that some
demuxers contain bugs that lead to wrong timestamps, these demuxers
should of course still be fixed even if this change happens to
hide the issue.
Signed-off-by: Michael Niedermayer <[email protected]>
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Should not make any difference, but its more correct.
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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* 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 <[email protected]>
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* qatar/master:
avconv: use default alignment for audio buffer
avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame()
avutil: use align == 0 for default alignment in audio sample buffer functions
avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc()
avconv: remove OutputStream.picref.
avconv: only set SAR once on the decoded frame.
avcodec: validate the channel layout vs. channel count for decoders
audioconvert: make av_get_channel_layout accept composite names.
avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()
Conflicts:
doc/APIchanges
ffmpeg.c
libavcodec/utils.c
libavcodec/version.h
libavutil/audioconvert.c
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
Merged-by: Michael Niedermayer <[email protected]>
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* qatar/master:
rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
asfdec: Add an option for not searching for the packet markers
cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
cosmetics: Align codec declarations
cosmetics: Convert mimic.c to utf-8
avconv: remove an unused function parameter.
avconv: remove now pointless variables.
avconv: drop support for building without libavfilter.
nellymoserenc: fix crash due to memsetting the wrong area.
libavformat: Only require first packet to be known for audio/video streams
avplay: Don't try to scale timestamps if the tb isn't set
Conflicts:
Changelog
configure
ffmpeg.c
libavcodec/aacenc.c
libavcodec/bmpenc.c
libavcodec/dnxhddec.c
libavcodec/dnxhdenc.c
libavcodec/ffv1.c
libavcodec/flacenc.c
libavcodec/fraps.c
libavcodec/huffyuv.c
libavcodec/libopenjpegdec.c
libavcodec/mpeg12enc.c
libavcodec/mpeg4videodec.c
libavcodec/pamenc.c
libavcodec/pgssubdec.c
libavcodec/pngenc.c
libavcodec/qtrleenc.c
libavcodec/rawdec.c
libavcodec/sgienc.c
libavcodec/tiffenc.c
libavcodec/v210dec.c
libavcodec/wmv2dec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <[email protected]>
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This struct was forgotten after
e5df74dc56494068042f094d43a5d774c257ae69.
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This is unused since a7b5e841ffe4b0f8423288965b8d069bd2a7a792.
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Fix map_channel when mapped audio stream index is greater than 0.
Signed-off-by: Michael Niedermayer <[email protected]>
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Fixes Ticket387
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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This heuristic is less likely to overshoot and cause encoder failure.
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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codec_get_buffer().
Signed-off-by: Michael Niedermayer <[email protected]>
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dr1 seems to work fine with frame size changes but many filters
cant handle it yet. Simply disabling it forces the alternative
non dr1 code path which has been tested more completely and
is known to handle frame size changes in a wider varity of
cases.
Signed-off-by: Michael Niedermayer <[email protected]>
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Address trac ticket #1120.
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Fixes Ticket1131
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Output now lists the available sample rates with commands like
ffmpeg -f lavfi -i aevalsrc=0 -ar 20000 -y /tmp/out.mp3
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