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>
