Unsurprisingly, if a timing-less subrip decoder is desireable, an
encoder is as well. With this in place, we can move on to remove
the use of the old encoder/decoder with embedded timing and move
all timing handling the (de)muxer where they belong.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
lavf: Detect discontinuities in timestamps for framerate/analyzeduration calculation
lavf: Initialize the stream info timestamps in avformat_new_stream
id3v2: Match PIC mimetype/format case-insensitively
configure: Rename check_asm() to more fitting check_inline_asm()
fate: Only test enabled filters
avresample: De-doxygenize some comments where Doxygen is not appropriate
rtmp: split chunk_size var into in_chunk_size and out_chunk_size
rtmp: Factorize the code by adding find_tracked_method
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes Ticket1627
The fate change is due to ffmpeg no longer pushing audio timestamps
aggressively up (which is what caused the AV sync issues in the ticket)
but leaving them as they are.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
g723.1: fix addition overflow
g723.1: simplify and fix multiplication overflow
g723.1: deobfuscate an expression
g723.1: remove unused #includes
ARM: add missing "cc" clobber in av_clipl_int32_arm()
rtmp: Factorize the code by adding handle_invoke_error
rtmp: Factorize the code by adding handle_invoke_status
rtmp: Factorize the code by adding handle_invoke_result
libavutil: remove unused av_abort() macro
ffmenc: replace if/abort with assert()
libavutil: drop offsetof() fallback definition
libavutil: drop fallback definitions of INTxx_MIN/MAX
configure: Check for a sctp struct instead of just the header
configure: suncc: Add -xc99 to dependency flags, required on Solaris
doxygen: Fix function parameter names to match the code
doc: Drop obsolete shared libs cflags hint to workaround Cygwin gcc bugs
swf: Move shared table out of the header file
swf: Move swf_audio_codec_tags table to the only place it is used
fate: add G.723.1 decoder tests
Conflicts:
configure
doc/platform.texi
libavformat/Makefile
libavutil/arm/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous code dependent on the input buffer matching the
buffer that has been provided by yadifs get_buffer.
The API does in now way gurantee this though its often true.
This fixes some out of array reads.
The regression test checksums change due to "out of picture" values
being initialized differently.
There should be no visual difference in the filters output
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavr: fix handling of custom mix matrices
fate: force pix_fmt in lagarith-rgb32 test
fate: add tests for lagarith lossless video codec.
ARMv6: vp8: fix stack allocation with Apple's assembler
ARM: vp56: allow inline asm to build with clang
fft: 3dnow: fix register name typo in DECL_IMDCT macro
x86: dct32: port to cpuflags
x86: build: replace mmx2 by mmxext
Revert "wmapro: prevent division by zero when sample rate is unspecified"
wmapro: prevent division by zero when sample rate is unspecified
lagarith: fix color plane inversion for YUY2 output.
lagarith: pad RGB buffer by 1 byte.
dsputil: make add_hfyu_left_prediction_sse4() support unaligned src.
Conflicts:
doc/APIchanges
libavcodec/lagarith.c
libavfilter/x86/gradfun.c
libavutil/cpu.h
libavutil/version.h
libswscale/utils.c
libswscale/version.h
libswscale/x86/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change introduces a basic encoder for 3GPP Timed Text subtitles,
also known as TX3G, Quicktime subtitles, or "movtext" in the existing
code.
This initial change doesn't attempt to write styling information,
and just writes the plain text of the subtitles. I intend to add
support for styles eventually, but it's challenging due to a lack
of existing players that support them.
Note that an additional change is required to the mov/mp4 muxer to
write empty subtitle packets to indicate subtitle duration.
Signed-off-by: Philip Langdale <philipl@overt.org>
Restore functionality to set the samples directory via the
FATE_SAMPLES environment variable . This is broken since commit
63dcd16 was merged.
Additionally the name FATE_EXTERN is more suited as the current
FATE_SAMPLES make file variable does not carry the name of the
FATE samples or the name of the directory they are stored in, but
does contain the names of the FATE targets that need external
samples. That is samples that are not in the repository and are
not generated on the fly.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
* qatar/master:
vc1dec: Remove separate scaling function for interlaced field MVs
vc1dec: Invoke edge_emulation regardless of MV precision
x86: Use consistent 3dnowext function and macro name suffixes
g723_1: scale output as supposed for the case with postfilter disabled
g723_1: increase excitation storage by 4
g723_1: fix upper bound parameter from inverse maximum autocorrelation
g723_1: make scale_vector() behave like the reference
g723_1: fix off-by-one error in normalize_bits()
g723_1: save/restore excitation with offset to store LPC history
wmapro: prevent division by zero when sample rate is unspecified
x86: proresdsp: improve SIGNEXTEND macro comments
x86: h264dsp: K&R formatting cosmetics
LICENSE: Document all GPL files
Conflicts:
libavcodec/g723_1.c
libavcodec/wmaprodec.c
libavcodec/x86/h264dsp_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avformat: Drop pointless "format" from container long names
swscale: bury one more piece of inline asm under HAVE_INLINE_ASM.
wv: K&R formatting cosmetics
configure: Add missing descriptions to help output
h264_ps: declare array of colorspace strings on its own line.
fate: amix: specify f32 sample format for comparison
tiny_psnr: support 32-bit float samples
eamad/eatgq/eatqi: call special EA IDCT directly
eamad: remove use of MpegEncContext
mpegvideo: remove unnecessary inclusions of faandct.h
af_asyncts: avoid overflow in out_size with large delta values
af_asyncts: add first_pts option
Conflicts:
configure
libavcodec/eamad.c
libavcodec/h264_ps.c
libavformat/crcenc.c
libavformat/ffmdec.c
libavformat/ffmenc.c
libavformat/framecrcenc.c
libavformat/md5enc.c
libavformat/nutdec.c
libavformat/rawenc.c
libavformat/yuv4mpeg.c
tests/tiny_psnr.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.
The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
MMX-enabled systems by default use some dsputil functions differing
from the C versions. Adding these flags ensures accurate ones are
used everywhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
proresdsp: port x86 assembly to cpuflags.
lavr: x86: improve non-SSE4 version of S16_TO_S32_SX macro
lavfi: better channel layout negotiation
alac: check for truncated packets
alac: reverse lpc coeff order, simplify filter
lavr: add x86-optimized mixing functions
x86: add support for fmaddps fma4 instruction with abstraction to avx/sse
tscc2: fix typo in array index
build: use COMPILE template for HOSTOBJS
build: do full flag handling for all compiler-type tools
eval: fix printing of NaN in eval fate test.
build: Rename aandct component to more descriptive aandcttables
mpegaudio: bury inline asm under HAVE_INLINE_ASM.
x86inc: automatically insert vzeroupper for YMM functions.
rtmp: Check the buffer length of ping packets
rtmp: Allow having more unknown data at the end of a chunk size packet without failing
rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets
Conflicts:
Makefile
configure
libavcodec/x86/proresdsp.asm
libavutil/eval.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
there are some technical problems with fate.ffmpeg.org
thus split the subdomain between fate-suite and fate
fate-suite is now (temporary) provided by our main server
until fate-suite.ffmpeg.org is setup to point somewhere
we use fate-suite.avcodec.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
commit 20e88d8618
Fix avui stream-copy.
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
