This allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame's expected pts, so no padding or
trimming is done.
* 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>
The voice register functions return the same voice structure
upon multiple registration. It causes us two problems:
If we delete a voice without deregistering it, it leaves
a dangling pointer inside the library.
If we delete or unregister a voice at uninit, it may still
be in use by another instance of the filter.
The second problem is solved by keeping an usage counter inside
asrc_flite. This is not thread-safe, but neither is flite itself.
Since the recent changes, movie and amovie are able to deal with more
than one type of stream, so they should be categorized as "multimedia
sources" rather than audio/video sources.
* 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>
Allow substitution of channel pairs in the input for nearby channel pairs in
the output in order to get a closer match. Also weigh LFE channel mismatch
differently to favor matching the same layout without LFE over one less
channel with LFE.
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
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>
The video frame reference is passed along the filterchain, and is not
possessed anymore by the filter. If out_buf is not set to NULL, it will
be freed by ff_end_frame() causing a crash.
* 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>
* qatar/master:
lavfi: unref AVFilterLink.out_buf in ff_end_frame().
lavfi: unref AVFilterLink.cur_buf in ff_end_frame().
vsrc_testsrc: avoid an unnecessary avfilter_ref_buffer().
vf_slicify: clear AVFilterLink.cur_buf in start_frame().
vf_settb: simplify start_frame().
vf_fieldorder: don't give up its own reference to the output buffer.
vf_pad: don't give up its own reference to the output buffer.
vf_overlay: don't access a buffer reference that's been given away.
vf_drawtext: don't give up its own reference to the input buffer.
vf_gradfun: don't store two pointers to one AVFilterBufferRef.
vf_delogo: don't store two pointers to one AVFilterBufferRef.
vf_aspect: clear AVFilterLink.cur_buf in start_frame().
lavfi: add avfilter_unref_bufferp()
Conflicts:
doc/APIchanges
libavfilter/avfilter.h
libavfilter/buffer.c
libavfilter/vf_aspect.c
libavfilter/vf_drawtext.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Directly forwarding the input buffer to the next filter means that
drawtext no longer owns any references to it and thus shouldn't refer to
it in any way.
This one is tricky. Consider a graph with two sink links, A and B.
request_oldest() requests on A, and A returns EOF. This EOF causes
a filter to flush pending frames to B; they are unrequested.
Then request_oldest() moves on to B, and B returns EOF.
movie currently forwards EOF (or begins looping) immediately upon
reaching the last packet in the input stream, which can leave frames in
the decoder. This change first tries to read any remaining packets from
the decoder before forwarding EOF.
Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Use the init_opaque callback for the purpose.
Fix regression introduced in a5e8c41c28. In particular, fix lavfi
device in case a conversion to the supported (packed) formats is needed.
This will be used by filters which require an opaque field to be passed
in input. Should be required only for filters which imply a programmatic
use.
This is possibly a temporary solution, to be removed when we'll have a
clearer and better agreememnt of how/if to pass binary data for
initializing a filter.
See thread:
Subject: [FFmpeg-devel][PATCH] lavfi: add init2 callback
Date: Fri, 6 Jul 2012 01:22:17 +0200
* 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>
* qatar/master:
configure: add functions for testing code fragments
af_amix: avoid spurious EAGAIN.
af_amix: return AVERROR(EAGAIN) when request_frame didn't produce output.
af_amix: only consider negative return codes as errors.
avconv: use only meaningful timestamps in start time check.
avconv: fix the check for -ss as an output option.
mss3: add forgotten 'static' qualifier for private table
lavc: options: add planar names for request_sample_fmt
flacdec: add planar output support
flvdec: Treat all nellymoser versions as the same codec
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Input on/off state can change in request_samples(), which can result in
a state where only the first input is active. get_available_samples()
will then return 0, and request_frame() will fail with EAGAIN even
though there is data on the single active input.
Take this into account and check the number of active inputs again after
calling request_samples().
