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.
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>
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.
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>
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>
This code spews a multitude of warnings with glibc (unchecked
return values), some of them possibly warranted. Furthermore,
the deamonisation is not suitable for use with typical startup
scripts as it does not provide the PID of the daemon in any way.
Users wishing to run avserver as a daemon can still do so using
start-stop-daemon or equivalent tools.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Also retroactively add a changelog entry to the 9beta1 list
for general MSVC support, which was present there already.
Signed-off-by: Martin Storsjö <martin@martin.st>
It has not worked for anything other than fringe codecs (asv1/2, mdec,
mjpeg[b]) since about 2003 and nobody ever noticed or complained. This
sufficiently proves that there are no users of this option who have a
clue of what they are doing, so it is completely useless.
This muxer splits the output from the ismv muxer into individual
files, in realtime.
The same can also be done by the standalone tool ismindex, but this
muxer is needed for doing it in realtime (especially for live
streams that need extra handling for updating the lookahead fields
in the fragment headers).
Using this muxer, one can deliver live smooth streaming from a
normal static file web server. (Using ismindex, one can deliver
premade smooth streaming files from a static file web server,
or prepare files for serving with IIS.)
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the mandatory memcpy in vsrc_buffer has been eliminated, there
shouldn't be any significant reason to build without lavfi anymore.
This will make upcoming support for complex filtergraphs easier to do.
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Decodes 16-bit WMA Lossless encoded files. 24-bit is not supported yet.
Bitstream parser written by Andreas Öman with contributions from
Baptiste Coudurier and Ulion.
Includes a number of bug-fixes from Benjamin Larsson, Michael Niedermayer and
Konstantin Shishkov, shine and polish by Diego Biurrun.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This library does not fit into Libav as a whole and its code is just a
maintenance burden. Furthermore it is now available as an external project,
which completely obviates any reason to keep it around.
URL: http://git.videolan.org/?p=libpostproc.git
The WAVE demuxer returns packets with many blocks per frame, which needs to be
parsed into single blocks. This has a side-effect of fixing the timestamps.
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
The new decoder is much smaller and has better code quality.
Cleanup and fixes courtesy of Kostya Shishkov.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
It currently use the simple api and is using the latency information
provided only to offset the stream start.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
With the following additions:
* support to gray format
* support to yuva420p format
* parametric luma/chroma/alpha radius
* consistency check on the radius values, avoid crashes with invalid values
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This patch also introduces CODEC_ID_CELT.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Add an APIchanges entry for the av_pkt_dump2 and av_pkt_dump_log2
functions, and a changelog entry for the apple http live streaming
protocol handler.
Since neither of them got a minor bump at commit time, but were
applied before the jv demuxer, they all can be considered added
in this minor version.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
FILTERNAME=ARGS and FILTERNAME:ARGS syntax.
The same filter class will be used for managing all the libopencv
filtering functions.
Originally committed as revision 26079 to svn://svn.ffmpeg.org/ffmpeg/trunk
Galvão Póvoa <marspeoplester gmail com>, mentored by Robert Swain <robert
dot swain gmail com>.
Originally committed as revision 26051 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Nolan L nol888 <=> gmail >=< com.
See thread:
Subject: [FFmpeg-devel] [PATCH] Port gradfun to libavfilter (GCI)
Date: Mon, 29 Nov 2010 07:18:14 -0500
Originally committed as revision 25942 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
This works at least for some people testing it.
Patch by Anssi Hannula, anssi d hannula a iki fi
Originally committed as revision 25834 to svn://svn.ffmpeg.org/ffmpeg/trunk
Only works via HDMI.
Patch by Anssi Hannula (anssi d hannula a iki d fi), based on some work
by myself.
Originally committed as revision 25760 to svn://svn.ffmpeg.org/ffmpeg/trunk
The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }
Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
The demuxer inspects the payload type of a received RTP packet and
handles the cases where the content is fully described by the payload type.
Originally committed as revision 25527 to svn://svn.ffmpeg.org/ffmpeg/trunk
The option is useful to ensure that there is a seek point exactly at a
place the user will probably want to jump precisely sometime, the
major example would be the end of an opening and the beginning of a
chapter. The scene change detection system will often make it happen,
but not always for example if there is a fade-in.
See the thread:
Subject: [FFmpeg-devel] [PATCH] -force_key_frames option
Date: Tue, 12 Oct 2010 15:16:26 +0200
Patch by Nicolas George -mail nicolas,george,normalesup,org.
Originally committed as revision 25526 to svn://svn.ffmpeg.org/ffmpeg/trunk
See thread:
Subject: [FFmpeg-devel] [PATCH] Port MPlayer blackframe filter.
Date: Sun, 26 Sep 2010 01:10:40 +0200
Originally committed as revision 25214 to svn://svn.ffmpeg.org/ffmpeg/trunk
