* qatar/master:
libopus: Remap channels using libopus' internal remapping.
Opus decoder using libopus
avcodec: document the use of AVCodecContext.delay for audio decoding
vc1dec: add flush function for WMV9 and VC-1 decoders
http: Increase buffer sizes to cope with longer URIs
nutenc: const correctness for ff_put_v_trace/put_s_trace function arguments
h264_refs: Fix debug tprintf argument types
golomb: const correctness for get_ue()/get_se() function arguments
get_bits: const correctness for get_bits_trace()/get_xbits_trace() arguments
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/version.h
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way we can directly remap channels from Opus' channel order to
libav's internal channel order, instead of mapping channels from
Opus' order to Vorbis' order then to libav's order.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
dwt: Drop unused functions spatial_compose{53|97}i()
nutdec: Remove unused and broken debug function stub
avcodec: Drop long-deprecated imgconvert.h header
Add Opus support to the Ogg muxer.
Add Opus codec id and codec description.
avformat: Identify anonymous AVIO typedef structs.
Conflicts:
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/imgconvert.h
libavcodec/version.h
libavformat/oggenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
I was sadly unable to find a non fuzzed mp3 that uses the
feature that contained the bug (and i searched hard ...), thus
while this fixes the security issue. It may or may not fix
mixed blocks in 8khz mp3s, i cant say due to lack of samples to test.
Security issue exists since: b37d945dd4
Reported-by: Dale Curtis <dalecurtis@google.com>
(Probably) Found-by: inferno@chromium.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use the MAX_URL_SIZE define where applicable. Increase buffer
sizes for all buffers that need to fit a long pathname - buffers
that need to fit only the hostname (and other short strings, but
not the pathname - such as "headers" in http_connect) are kept
at 1024 bytes for now.
Also increase the max line length in http_read_header, since it
might need to contain a full url for Location: redirects.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes some DNXHD files generated by AVID TM, where codec UL was set to A-law
meanwhile the real audio codec was PCM S16. According to SMPTE RP 224, A-law is
the default value for sound essence parameters therefore we should handle it
specially.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Implement av_buffersink_read() and av_buffersink_read_samples()
for ffmpeg's version of buffersink.
With this change, avconv linked against ffmpeg's libraries passes
the same number of tests whether it uses ffbuffersink or
buffersink_old.
* qatar/master:
nutdec: const correctness for get_v_trace/get_s_trace function arguments
truemotion2: Request samples for old TM2 headers
rtpdec: Remove a useless ff_ prefix from a static symbol
rtpdec: Support depacketizing speex
rtpenc: Add support for packetizing speex
Conflicts:
libavformat/rtpdec.c
libavformat/sdp.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Anonymous typedef structs prevent forward declaration, this
change gives the AVIOContext and AVIOInterruptCB structures
a name. These structures are now in line with other common
structures such as AVFormatContext and AVCodecContext.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Also factorize the common options for the different mov-based tests.
Since the header is now on top in the last generated file, the data
offset in the seek test needed some updates as well.
At the moment, the moov header is written at the end of the file, so we
can use the current offset (which focus on the end of the mdat already
written) to guess if 64-bits offset will be required or not.
Though, the next commits will make possible the writing of this table at
the beginning, so this heuristic can't work. As a consequence, we check
all the values within the potential offset table for any value >
32-bits.
Normally we discard things prior to the intended start
for stream copy this is not always possible, and its not done by default
this option allows discarding to be enabled
this is primarely usefull when transcoding a video and stream copying an
audio stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously we had ignored the past dts and just filled in from the
point where we have had sufficient information.
This should fix Ticket1734
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This packetization scheme simply places the full packets into the
RTP packet without any extra header bytes.
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts commit d25f87f517.
This breaks decoding of some h264 files
I have tested the original patch with fate but by mistake have
forgotten to specify the fate samples so testing was limited to
the internal regression tests.
