Current method for constraining fragment position drift suffers from
round-off error build up.
Instead of calculating cumulative drift as a sum of input fragment
position corrections, it is more accurate to calculate drift as the
difference between current fragment position and the ideal position
specified by the tempo scale factor.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
valgrind reported uninitialized memory access which was caused by
incorrect number of samples being passed to push_samples(..)
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly automated global search and replace
The deprecated aconvert filter is disabled, if it still has users
it should be updated
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Memory obtained from av_realloc is not aligned enough for AVX.
The other similar allocations are changed too because they use
the same macro. The buffers were cleared afterwards anyway.
Fix trac ticket #1692.
* 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>
* commit 'a5e8c41c28f907d98d2a739db08f7aef4cbfcf3a':
lavfi: remove 'opaque' parameter from AVFilter.init()
mov: do not try to read total disc/track number if data atom is too short.
avconv: fix -force_key_frames
dxva2_h264: fix signaling of mbaff frames
x86: fft: elf64: fix PIC build
Conflicts:
ffmpeg.c
libavcodec/v210dec.h
libavfilter/asrc_anullsrc.c
libavfilter/buffersrc.c
libavfilter/src_movie.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_overlay.c
libavfilter/vsrc_color.c
libavfilter/vsrc_testsrc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavfilter API was designed in order to be clarly distinguished from the
libavcodec API, including avcodec.h in avfilter.h is not going to help to
stick to this principle.
The inclusion of libavutil/audioconvert.h in many files was required
because avcodec.h includes audioconvert.h.
libavfilter/avcodec.h is where the lavc/lavfi interface should be
entirely placed.
* qatar/master:
x86: Only use optimizations with cmov if the CPU supports the instruction
x86: Add CPU flag for the i686 cmov instruction
x86: remove unused inline asm macros from dsputil_mmx.h
x86: move some inline asm macros to the only places they are used
lavfi: Add the af_channelmap audio channel mapping filter.
lavfi: add join audio filter.
lavfi: allow audio filters to request a given number of samples.
lavfi: support automatically inserting the fifo filter when needed.
lavfi/audio: eliminate ff_default_filter_samples().
Conflicts:
Changelog
libavcodec/x86/h264dsp_mmx.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/version.h
libavutil/x86/cpu.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Reduce audio fragment alignment jitter by penalizing alignment
correction offsets that deviate too much from the target offset.
This is accomplished by multiplying the cross correlation search
window with a quadratic function.
Signed-off-by: Pavel Koshevoy <pavel@homestead.aragog.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add atempo audio filter for adjusting audio tempo without affecting
pitch. This filter implements WSOLA algorithm with fast cross
correlation calculation in frequency domain.
Signed-off-by: Pavel Koshevoy <pavel@homestead.aragog.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Pavel Koshevoy
Nicolas George
Michael Niedermayer
Paul B Mahol
Nicolas George
Clément Bœsch
Stefano Sabatini