Commit Graph

175 Commits

Author SHA1 Message Date
Anton Khirnov
2d2d6a4883 lavf: add a raw WavPack muxer. 2013-05-28 18:19:19 +02:00
Anton Khirnov
e3b225a4fe matroskaenc: add an option to put the index at the start of the file 2013-05-03 08:32:35 +02:00
Anton Khirnov
85a5bc054c lavf: remove disabled FF_API_R_FRAME_RATE cruft 2013-03-11 18:23:50 +01:00
Anton Khirnov
7b486ab13b lavf: remove disabled FF_API_AV_GETTIME cruft 2013-03-11 18:23:18 +01:00
Anton Khirnov
32e5194969 lavf: remove disabled FF_API_INTERLEAVE_PACKET cruft 2013-03-11 18:23:10 +01:00
Anton Khirnov
435c2a31ad lavf: remove disabled FF_API_READ_PACKET cruft 2013-03-11 18:23:02 +01:00
Anton Khirnov
c7e044c61b lavf: remove disabled FF_API_APPLEHTTP_PROTO cruft 2013-03-11 18:22:54 +01:00
Anton Khirnov
0a7c4daf46 lavf: remove disabled FF_API_CLOSE_INPUT_FILE cruft 2013-03-11 18:22:45 +01:00
Anton Khirnov
d8b31be6ca Add the bumps and APIchanges entries for reference counted buffers changes. 2013-03-08 07:41:49 +01:00
Martin Storsjö
de9cd1b173 lavf: Handle the environment variable no_proxy more properly
The handling of the environment variable no_proxy, present since
one of the initial commits (de6d9b6404), is inconsistent with
how many other applications and libraries interpret this
variable. Its bare presence does not indicate that the use of
proxies should be skipped, but it is some sort of pattern for
hosts that does not need using a proxy (e.g. for a local network).

As investigated by Rudolf Polzer, different libraries handle this
in different ways, some supporting IP address masks, some supporting
arbitrary globbing using *, some just checking that the pattern matches
the end of the hostname without regard for whether it actually is
the right domain or a domain that ends in the same string.

This simple logic should be pretty similar to the logic used by
lynx and curl.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-02-27 21:32:13 +02:00
Martin Storsjö
ab587f39b2 rtpenc: Start the sequence numbers from a random offset
Expose the current sequence number via an AVOption - this can
be used both for setting the initial sequence number, or for
querying the current number.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-22 00:25:38 +02:00
Martin Storsjö
e1d0b3d875 srtp: Add support for a few DTLS-SRTP related crypto suites
The main difference to the existing suites from RFC 4568 is
that the version with a 32 bit HMAC still uses 80 bit HMAC
for RTCP packets.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-21 00:13:35 +02:00
Martin Storsjö
2f3bada63e lavf: Add a protocol for SRTP encryption/decryption
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-15 11:55:10 +02:00
Martin Storsjö
424da30830 rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
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>
2013-01-15 11:54:40 +02:00
Martin Storsjö
86d9181cf4 rtpdec: Support sending RTCP feedback packets
This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.

This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).

The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.

The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-08 17:48:14 +02:00
Martin Storsjö
e96406eda4 rtsp: Add support for depacketizing RTP data via custom IO
To use this, set sdpflags=custom_io to the sdp demuxer. During
the avformat_open_input call, the SDP is read from the AVFormatContext
AVIOContext (ctx->pb) - after the avformat_open_input call,
during the av_read_frame() calls, the same ctx->pb is used for reading
packets (and sending back RTCP RR packets).

Normally, one would use this with a read-only AVIOContext for the
SDP during the avformat_open_input call, then close that one and
replace it with a read-write one for the packets after the
avformat_open_input call has returned.

This allows using the RTP depacketizers as "pure" demuxers, without
having them tied to the libavformat network IO.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-03 15:15:27 +02:00
Martin Storsjö
c1ea44c54d rtmp: Add support for limelight authentication
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>
2012-12-31 13:39:09 +02:00
Martin Storsjö
08225d0126 rtmp: Add support for adobe authentication
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>
2012-12-31 13:39:08 +02:00
Paul B Mahol
57231e4d5b tak: demuxer, parser, and decoder
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-12-07 16:15:02 -05:00
Victor Vasiliev
58b619c8a2 wav muxer: write metadata
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-10-16 18:51:56 +02:00
Luca Barbato
b522000e9b avio: introduce avio_closep 2012-10-10 18:56:55 +02:00
Nathan Caldwell
bcc1f7caeb Add Opus support to the Ogg muxer.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2012-09-27 10:48:35 +02:00
Dmitry Samonenko
b6bf1490da rtpdec: Support depacketizing speex
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-09-26 19:05:10 +03:00
Dmitry Samonenko
490ae95aa8 rtpenc: Add support for packetizing speex
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>
2012-09-26 19:04:57 +03:00
Samuel Pitoiset
cee1950bbb rtp: Packetization of JPEG (RFC 2435) 2012-09-23 21:58:41 +03:00
Stefano Sabatini
5d1203f063 avio: flush the internal buffer in avio_close()
This is consistent with stdio, and thus what people would naturally
expect.
2012-09-15 18:24:49 +02:00
Martin Storsjö
62c9ae11a7 Add a smooth streaming segmenter muxer
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>
2012-09-12 23:42:16 +03:00
Samuel Pitoiset
3c19815416 rtp: Depacketization of JPEG (RFC 2435)
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-09-09 22:22:21 +03:00
Jordi Ortiz
e5f2731c73 rtmp: Add support for receiving incoming streams
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-16 18:13:41 +03:00
Samuel Pitoiset
93f257db6b rtmp: Automatically compute the hash for SWFVerification
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-15 22:05:36 +03:00
Samuel Pitoiset
635ac8e1be rtmp: Add support for SWFVerification
Specifies how the server verifies client SWF files before allowing the
files to connect to an application. Verifying SWF files is a security
measure that prevents someone from creating their own SWF files that can
attempt to stream your resources.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-15 10:37:34 +03:00
Martin Storsjö
1243c72251 rtsp: Support mpegts in raw udp packets
This is basically the same way as mpegts packets are parsed in
rtpdec.c.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-09 00:25:57 +03:00
Samuel Pitoiset
00cb52c65c rtmp: Add a new option 'rtmp_subscribe'
This option specifies the name of live stream to subscribe.
Defaults to rtmp_playpath.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-07 23:35:39 +03:00
Anton Khirnov
aba232cfa9 lavf: deprecate r_frame_rate.
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.

Replace it with the average framerate where it makes sense.

FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.

In some other tests lavf starts making up frame durations from different
frame.
2012-07-29 08:06:30 +02:00
Luca Barbato
681ed00099 avf: introduce nobuffer option
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.

An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.

Additional changes by Josh Allmann <joshua.allmann@gmail.com>

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-07-29 07:58:00 +02:00
Antti Seppälä
5423e908c9 Support urlencoded http authentication credentials
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.

Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-26 00:18:32 +03:00
Samuel Pitoiset
08cd95e8a3 RTMPTE protocol support
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-23 16:32:09 +03:00
Samuel Pitoiset
acd554c103 RTMPE protocol support
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>
2012-07-23 16:32:07 +03:00
Kostya Shishkov
1470ce21ce Bump libavcodec and libavformat minor versions for G.723.1 decoder and demuxer 2012-07-22 08:43:12 +02:00
Samuel Pitoiset
86991ce2dd RTMPTS protocol support
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-17 14:02:55 +03:00
Samuel Pitoiset
6aedabc9b6 RTMPS protocol support
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-17 13:53:33 +03:00
Jordi Ortiz
a8ad6ffafe rtsp: Add listen mode
This makes the RTSP demuxer act as a server, listening for an
incoming connection.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 22:00:28 +03:00
Diego Biurrun
2047e40e6e Clarify Doxygen comment for FF_API_* #defines. 2012-07-04 15:10:10 +02:00
Diego Biurrun
09f211987c misc typo and wording fixes 2012-07-03 17:35:11 +02:00
Mans Rullgard
dc7e336cae lavf, lavu: version bumps and APIchanges for av_gettime() move
Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-06-21 11:45:28 +01:00
Mans Rullgard
ae0a301668 Move av_gettime() to libavutil
Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-06-20 17:09:03 +01:00
Martin Storsjö
579fd87b46 rtpenc: Support packetizing iLBC
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:01:04 +03:00
Martin Storsjö
89c3960544 rtpdec: Add a depacketizer for iLBC
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:01:04 +03:00
Martin Storsjö
a2b251a05e Implement the iLBC storage file format
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:01:02 +03:00
Samuel Pitoiset
46743a859c rtmp: Don't send every flv packet in a separate HTTP request in RTMPT
Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.

This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:00:31 +03:00