* commit 'e3b225a4fe0ff1e64a220b757c6f0a5cf9258521':
matroskaenc: add an option to put the index at the start of the file
Conflicts:
doc/muxers.texi
libavformat/matroskaenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a83c0da539fb07260310bc3b34056239d2b138b2':
avconv: make -t insert trim/atrim filters.
The filter insertion code is merged but disabled as it is buggy.
For example it fails in various ways when used with -s with some files.
Also the trimming is arguably less accurate than the default without
filters in some cases.
These issues should be fixed before auto inserting the filters,
until then the user can explicitly add a trim/atrim filter when one is
wanted.
Conflicts:
Changelog
ffmpeg.c
ffmpeg_filter.c
tests/ref/fate/bethsoft-vid
tests/ref/lavf/aiff
tests/ref/lavf/asf
tests/ref/lavf/au
tests/ref/lavf/avi
tests/ref/lavf/dpx
tests/ref/lavf/ffm
tests/ref/lavf/gxf
tests/ref/lavf/jpg
tests/ref/lavf/mkv
tests/ref/lavf/mmf
tests/ref/lavf/mov
tests/ref/lavf/mpg
tests/ref/lavf/nut
tests/ref/lavf/ogg
tests/ref/lavf/pcx
tests/ref/lavf/png
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
tests/ref/lavf/wav
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes -t sample-accurate for audio and will allow further
simplication in the future.
Most of the FATE changes are due to audio now being sample accurate. In
some cases a video frame was incorrectly passed with the old code, while
its was over the limit.
vidstabdetect and vidstabtransform common functions for interfacing
vid.stab are in libavfilter/vidstabutils.c
Signed-off-by: Georg Martius <martius@mis.mpg.de>
Based on the 2007 GSoC project from Kamil Nowosad <k.nowosad@students.mimuw.edu.pl>
Updated to current programming standards, style and many more small
fixes by Diego Biurrun <diego@biurrun.de>.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Currently, we have a AV_CODEC_ID_SSA, which matches the way the ASS/SSA
markup is muxed in a standalone .ass/.ssa file. This means the AVPacket
data starts with a "Dialogue:" string, followed by a timing information
(start and end of the event as string) and a trailing CRLF after each
line. One packet can contain several lines. We'll refer to this layout
as "SSA" or "SSA lines".
In matroska, this markup is not stored as such: it has no "Dialogue:"
prefix, it contains a ReadOrder field, the timing information is not in
the payload, and it doesn't contain the trailing CRLF. See [1] for more
info. We'll refer to this layout as "ASS".
Since we have only one common codec for both formats, the matroska
demuxer is constructing an AVPacket following the "SSA lines" format.
This causes several problems, so it was decided to change this into
clean ASS packets.
Some insight about what is changed or unchanged in this commit:
CODECS
------
- the decoding process still writes "SSA lines" markup inside the ass
fields of the subtitles rectangles (sub->rects[n]->ass), which is
still the current common way of representing decoded subtitles
markup. It is meant to change later.
- new ASS codec id: AV_CODEC_ID_ASS (which is different from the
legacy AV_CODEC_ID_SSA)
- lavc/assdec: the "ass" decoder is renamed into "ssa" (instead of
"ass") for consistency with the codec id and allows to add a real
ass decoder. This ass decoder receives clean ASS lines (so it starts
with a ReadOrder, is followed by the Layer, etc). We make sure this
is decoded properly in a new ass-line rectangle of the decoded
subtitles (the ssa decoder OTOH is doing a simple straightforward
copy). Using the packet timing instead of data string makes sure the
ass-line now contains the appropriate timing.
- lavc/assenc: just like the ass decoder, the "ssa" encoder is renamed
into "ssa" (instead of "ass") for consistency with the codec id, and
allows to add a real "ass" encoder.
One important thing about this encoder is that it only supports one
ass rectangle: we could have put several dialogue events in the
AVPacket (separated by a \0 for instance) but this would have cause
trouble for the muxer which needs not only the start time, but also
the duration: typically, you have merged events with the same start
time (stored in the AVPacket->pts) but a different duration. At the
moment, only the matroska do the merge with the SSA-line codec.
We will need to make sure all the decoders in the future can't add
more than one rectangle (and only one Dialogue line in it
obviously).
FORMATS
-------
- lavf/assenc: the .ass/.ssa muxer can take both SSA and ASS packets.
In the case of ASS packets as input, it adds the timing based on the
AVPacket pts and duration, and mux it with "Dialogue:", trailing
CRLF, etc.
- lavf/assdec: unchanged; it currently still only outputs SSA-lines
packets.
- lavf/mkv: the demuxer can now output ASS packets without the need of
any "SSA-lines" reconstruction hack. It will become the default at
next libavformat bump, and the SSA support will be dropped from the
demuxer. The muxer can take ASS packets since it's muxed normally,
and still supports the old SSA packets. All the SSA support and
hacks in Matroska code will be dropped at next lavf bump.
[1]: http://www.matroska.org/technical/specs/subtitles/ssa.html
* qatar/master:
FATE: add a test for the interlace filter
lavfi: new interlace filter
Conflicts:
Changelog
configure
doc/filters.texi
libavfilter/Makefile
libavfilter/allfilters.c
tests/fate/filter.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2e81acc687e64d15dd93c74793060bb5a233f44d':
x86inc: Fix number of operands for cmp* instructions
af_channelmap: fix uninitialized variable use introduced in ba8efac977
lavfi: add a bump and docs entries for the AVOptions switch
Conflicts:
Changelog
doc/APIchanges
libavfilter/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1ae44c87c924b69a0657256fbaa8ad140df2f27c':
lavfi/gradfun: remove rounding to match C and SSE code.
lavfi/gradfun: fix dithering in MMX code.
lavfi/gradfun: fix rounding in MMX code.
lavfi/gradfun: do not increment DC pointer for odd values.
fate: filter: Add dependencies
avconv: add options for reading filtergraphs from a file.
Conflicts:
Changelog
doc/ffmpeg.texi
doc/filters.texi
ffmpeg.h
ffmpeg_opt.c
libavfilter/vf_gradfun.c
tests/fate/filter.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '666fe5da47d127074be7f0e2bac93db6af8b4a30':
atomic: Exclude the unsupported implementation headers from checkheaders
avconv: do not silently ignore unused codec AVOptions.
Conflicts:
ffmpeg_opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Print an error and abort when the option is of the wrong type (decoding
for output file or vice versa), since this could never be correct for
any input or output configuration.
Print a warning and continue when the option is of the correct type,
just unused, so same commandlines can be reused for different kinds of
input or output files.
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.
If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '44e065d56c87d6a9d0effccec5f31517f72924ec':
vdpau: Add context and common helpers for hwaccel support
Conflicts:
Changelog
doc/APIchanges
libavcodec/vdpau.h
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is a port of virtual dub's histogram equalization filter by Donald
A. Graft. Based on the work by Jérémy Tran <tran.jeremy.av@gmail.com>,
done for SOCIS 2012.
This is a port of the kerndeint filter (libmpcodecs/vf_kerndeint) by
Donal A. Graft (original avisynth plugin author), and is based on the
work by Jérémy Tran <tran.jeremy.av@gmail.com> done for SOCIS 2012.
* qatar/master:
cmdutils: update copyright year to 2013
h264: check SPS entries directly to detect pixel format changes
forgotten changelogs for 9_beta2
Conflicts:
Changelog
cmdutils.c
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add support for limelight authentication
rtmp: Add support for adobe authentication
Conflicts:
Changelog
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Limelight is a not too uncommon CDN. The authentication scheme is
pretty similar to the adobe authentication, but is even closer to
normal http digest authentication (but not close enough to warrant
sharing code) than the adobe version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly used to authenticate the client when publishing.
Tested with wowza and akamai.
Some but not all servers support resending a new connect invoke
within the same connection, so always reconnect for sending a new
connection attempt. This matches what other applications do as well.
The authentication scheme is structurally pretty similar to http
digest authentication, but uses base64 instead of hex strings.
Signed-off-by: Martin Storsjö <martin@martin.st>
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
