This avoids creating unwieldy large packets, which is allowed but
does not seem to be a good idea
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some older compilers might have particular trouble with them,
and they do not really seem worth it to me.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Problem:
ffmpeg generated video file which had two audio packets with the same timestamp: last original audio packet and first padded audio packet.
Timestamp of first added audio packet by 'apad' fitler had the same value as last original audio packet. The problem was in 'aresample' fitler, which used next pts instead of current one.
As long as 'apad' and 'aresample' filters have separate mechanisms of timestamp calculation, they got the same values.
Command line:
ffmpeg -i <input_filename> -shortest -apad 512 -af asetnsamples=n=512 -b:a 1058400 -ac 1 -ar 44100 -async 0 -acodec pcm_s16le -sn -f matroska -y <output_file>
Fix:
Call swr_next_pts() function before swr_convert()
Tested:
FATE tests passed.
Fix has been tested in our Transcoder regression framework on ~10k test videos. It's about ~500k transcodes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
