* commit '051aadeed104ecbe8ee4850ec2d7e5394f5e1ccd':
ogg: Provide aliases for Speex, Opus and audio-only ogg
Conflicts:
Changelog
libavformat/oggenc.c
libavformat/version.h
See: 2ccc6ff03a
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '194be1f43ea391eb986732707435176e579265aa':
lavf: switch to AVStream.time_base as the hint for the muxer timebase
Conflicts:
doc/APIchanges
libavformat/filmstripenc.c
libavformat/movenc.c
libavformat/mxfenc.c
libavformat/oggenc.c
libavformat/swf.h
libavformat/version.h
tests/ref/lavf/mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
On big endian machines, the default value set via the faulty
AVOption ended up as 2^32 times too big.
This fixes the fate-lavf-ogg test which currently is broken on
big endian machines, broken since 3831362. Since that commit,
a final zero-sized packet is written to the ogg muxer in that test,
which caused different flushing behaviour on little and big endian
depending on whether the pref_duration option was handled as it
should or not.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '95b7fa1729b93bbb3f4fb85a5c0cb53cf970c3c7':
oggenc: Support flushing the muxer
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the caller to write all buffered data to disk, allowing
the caller to know at what byte position in the file a certain
packet starts (any packet written after the flush will be located
after that byte position).
Signed-off-by: Martin Storsjö <martin@martin.st>
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'efcde917af407a6031ecff68edd51fce7b83d104':
vorbiscomment: simplify API by using av_dict_count()
Conflicts:
libavformat/flacenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it instead of checking CODEC_FLAG_BITEXACT in the first stream's
codec context.
Using codec options inside lavf is fragile and can easily break when the
muxing codec context is not the encoding context.
Since 2007, the Xiph.org Foundation recommends that .ogg only be used
for Ogg Vorbis audio files.
Source: http://wiki.xiph.org/index.php/MIME_Types_and_File_Extensions
However we only do it if we have libvorbis available because the
built in vorbis encoder is not as good.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit 'c2cb01d418dd18e1cf997c038d37378d773121be':
lavf: introduce AVFMT_TS_NEGATIVE
Conflicts:
libavformat/avformat.h
libavformat/mux.c
tests/ref/lavf/asf
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/ts
tests/ref/seek/lavf-asf
tests/ref/seek/lavf-mkv
tests/ref/seek/lavf-mpg
tests/ref/seek/lavf-ts
This commit does not change the default ts behaviour, such
change will, if its done, be done separately.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most formats do not support negative timestamps, shift them to avoid
unexpected behaviour and a number of bad crashes.
CC:libav-stable@libav.org
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This uses page duration instead of byte size to determine when to buffer
the page. Also, it tries to avoid continued pages by buffering the current
page if there are already packets in the page and adding the next packet
would require it to be continued on a new page. This can improve seeking
performance.
The default page duration is 1 second, which is much saner than filling
all page segments by default.
* 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>
* qatar/master:
apedec: check bits <= 32.
cavs: Remove unused code.
oggenc: fix condition when not to flush due to keyframe granule.
oggenc: add pagesize option to set preferred page size
libspeexdec: set frame size in libspeex_decode_init()
smacker audio: sign-extend the initial 16-bit predicted value
Conflicts:
libavcodec/apedec.c
libavcodec/libspeexdec.c
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
docs: use -bsf:[vas] instead of -[vas]bsf.
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
lavf: move the packet keyframe setting code.
oggenc: free comment header for all codecs
lcl: error out if uncompressed input buffer is smaller than framesize.
mjpeg: abort decoding if packet is too large.
golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
get_bits: add HAVE_BITS_REMAINING macro.
lavf/output-example: use new audio encoding API correctly.
lavf/output-example: more proper usage of the new API.
tiff: Prevent overreads in the type_sizes array.
tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
apetag: do not leak memory if avio_read() fails
apetag: propagate errors.
SBR DSP x86: implement SSE sbr_hf_g_filt
SBR DSP x86: implement SSE sbr_sum_square_sse
SBR DSP: use intptr_t for the ixh parameter.
Conflicts:
doc/bitstream_filters.texi
doc/examples/muxing.c
doc/ffmpeg.texi
libavcodec/golomb.h
libavcodec/x86/Makefile
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
To make seeking work correctly, we must write a new granule for
each keyframe.
Unfortunately we currently have no regression tests due to no
included Theora encoder.
A test based on -vcodec copy from a Theora FATE sample should
probably be added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Michael Niedermayer
Luca Barbato
Anton Khirnov
Martin Storsjö
Andrew Kelley
James Almer
Carl Eugen Hoyos
Diego Biurrun
Justin Ruggles
Nathan Caldwell
Reimar Döffinger
Andres Gonzalez