* 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>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: fix the asyncts test
build: Drop gcc-specific warning flag from header compilation rule
FATE: add a test for the asyncts audio filter.
matroskadec: return more correct error code on read error.
buffersrc: check ff_get_audio_buffer() for errors.
lavfi: check all ff_get_video_buffer() calls for errors.
lavfi: check all avfilter_ref_buffer() calls for errors.
vf_select: avoid an unnecessary avfilter_ref_buffer().
buffersrc: avoid creating unnecessary buffer reference
lavfi: use avfilter_unref_bufferp() where appropriate.
vf_fps: add more error checks.
vf_fps: fix a memleak on malloc failure.
lavfi: check all ff_start_frame/draw_slice/end_frame calls for errors
lavfi: add error handling to end_frame().
lavfi: add error handling to draw_slice().
lavfi: add error handling to start_frame().
Conflicts:
Makefile
ffplay.c
libavfilter/buffersrc.c
libavfilter/vf_boxblur.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_frei0r.c
libavfilter/vf_hflip.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/video.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These filters are designed for storing and transmitting video sequences
with alpha using higher-efficiency codecs such as x264 which don't
natively support an alpha channel. 'alphaextract' takes an input stream
with an alpha channel and returns a video containing just the alpha
component as a grayscale value; 'alphamerge' takes an RGB or YUV stream
and adds an alpha channel recovered from a second grayscale stream.
Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Convert them to zigzag order, as the rest of them are.
When I was adding support for 10-bit DNxHD, I just copy-pasted the
missing quant matrices from the spec. Now it turns out the existing
matrices in dnxhddata.c were in zigzag order. This resulted in wrong
quantization for 10-bit DNxHD. The attached patch fixes the problem by
converting 10-bit quant matrices to zigzag order.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This change introduces a basic decoder for 3GPP Timed Text subtitles,
also known as TX3G, Quicktime subtitles, or "movtext" in the existing
code.
This initial change doesn't attempt to parse styling information,
and just reads 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.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
mss3: use standard zigzag table
mss3: split DSP functions that are used in MTS2(MSS4) into separate file
motion-test: do not use getopt()
tcp: add initial timeout limit for incoming connections
configure: Change the rdtsc check to a linker check
avconv: propagate fatal errors from lavfi.
lavfi: add error handling to filter_samples().
fate-run: make avconv() properly deal with multiple inputs.
asplit: don't leak the input buffer.
af_resample: fix request_frame() behavior.
af_asyncts: fix request_frame() behavior.
libx264: support aspect ratio switching
matroskadec: honor error_recognition when encountering unknown elements.
lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
lavr: resampling: add filter type and Kaiser window beta to AVOptions
lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
lavr: mix: validate internal sample format in ff_audio_mix_init()
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/libx264.c
libavfilter/audio.c
libavfilter/split.c
libavformat/tcp.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
