Commit Graph

354 Commits

Author SHA1 Message Date
Martin Storsjö
895678f823 rtsp: Specify unicast for TCP interleaved streams, too
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.

According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-03-21 20:58:33 +01:00
Mans Rullgard
2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Nicolas George
c76374c6db Use AVERROR_EXIT with url_interrupt_cb.
Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.

This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).

Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-03-15 08:09:19 -04:00
Anton Khirnov
22a3212e32 avio: rename url_fopen/fclose -> avio_open/close.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 10:18:55 -05:00
Martin Storsjö
28c4741a66 libavformat: Remove FF_NETERRNO()
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.

This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).

This fixes roundup issue 2614, unbreaking blocking network IO on
windows.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 07:21:31 -05:00
Anton Khirnov
b7effd4e83 avio: avio_ prefixes for get_* functions
In the name of consistency:
get_byte           -> avio_r8
get_<type>         -> avio_r<type>
get_buffer         -> avio_read

get_partial_buffer will be made private later

get_strz is left out becase I want to change it later to return
something useful.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-21 11:23:22 -05:00
Anton Khirnov
e731b8d872 avio: move init_put_byte() to a new private header and rename it
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:31 -05:00
Anton Khirnov
ae628ec1fd avio: rename ByteIOContext to AVIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:15 -05:00
Anton Khirnov
9fcae9735e Replace remaining uses of parse_date with av_parse_time.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-16 23:39:57 +00:00
Martin Storsjö
2c35a6bde9 rtsp: udp_read_packet returning 0 doesn't mean success
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).

This fixes issue 2612.
2011-02-17 00:37:00 +01:00
Martin Storsjö
b2dd842d21 rtsp/rdt: Assign the RTSPStream index to AVStream->id
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.

Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-11 16:58:19 -05:00
Martin Storsjö
b22dbb291d Use avformat_free_context for cleaning up muxers
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:39:55 -05:00
Martin Storsjö
1338dc0823 libavformat: Use avcodec_copy_context for chained muxers
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.

This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:28:07 -05:00
Martin Storsjö
ce41c51b0c Free AVStream->info in chained muxers
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 01:03:31 +01:00
Martin Storsjö
d9c0510e22 rtsp: Don't store RTSPStream in AVStream->priv_data
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.

This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.

Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 00:49:15 +01:00
Luca Barbato
ea7f080749 Free the RTSPStreams in ff_rtsp_close_streams
This plugs a small memory leak

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-02-01 20:40:16 +01:00
Luca Barbato
dfd2a005eb Replace dprintf with av_dlog
dprintf clashes with POSIX.1-2008
2011-01-29 23:55:37 +01:00
Luca Barbato
f81c7ac70a rtsp: make ff_sdp_parse return value forwarded
the sdp demuxer did not forward it at all while the rtsp demuxer assumed
a single kind of error
2011-01-28 15:45:19 +01:00
Luca Barbato
a8475bbdb6 os: replace select with poll
Select has limitations on the fd values it could accept and silently
breaks when it is reached.
2011-01-28 15:45:19 +01:00
Diego Elio Pettenò
c6610a216e Prefix all _demuxer, _muxer, _protocol from libavformat and libavdevice.
This also lists the objects from those two libraries as internal (by adding
the ff_ prefix) so that they can then be hidden via linker scripts.
2011-01-26 22:10:09 +00:00
Diego Elio Pettenò
57c4d01ec9 Make ff_rtsp_send_cmd_with_content_async static to rtsp.c.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-25 22:10:36 +01:00
Martin Storsjö
2762a7a28b rtspdec: Retry with TCP if UDP failed
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:49:36 +01:00
Martin Storsjo
aeb2de1c82 rtsp: Use ff_rtsp_undo_setup in the cleanup code in ff_rtsp_make_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:43 +01:00
Martin Storsjo
93e7490ee0 rtsp: Split out a function undoing the setup made by ff_rtsp_make_setup_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:39 +01:00
Martin Storsjo
fef5649a82 rtsp: Make make_setup_request a nonstatic function
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:36 +01:00
Martin Storsjö
a3b058b7ba rtsp: Properly fail if unable to open an input RTP port
Originally committed as revision 26285 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-09 10:47:53 +00:00
Martin Storsjö
a92c30d76e rtsp: Allow requesting of filtering of source packets
If filtered, only packets from the right source address and port
are received.

To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.

If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.

Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-06 15:22:58 +00:00
Martin Storsjö
29db7c3af4 rtsp: Parse RTP-Info headers
Originally committed as revision 26236 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-05 21:23:42 +00:00
Martin Storsjö
d2995eb910 rtsp: Store the Content-Base header value straight to the target
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.

Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:11:12 +00:00
Martin Storsjö
77223c5388 rtsp: Pass the method name to ff_rtsp_parse_line
Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:10:12 +00:00
Martin Storsjö
acc9ed1450 rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthState
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.

Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:07:56 +00:00
Martin Storsjö
3df54c6bf2 rtsp: Add a method parameter to ff_rtsp_read_reply
Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:06:21 +00:00
Martin Storsjö
3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-01 22:27:16 +00:00
Martin Storsjö
9e99f84f7d rtsp: Check if the rtp stream actually has an RTPDemuxContext
For example MS-RTSP doesn't have RTPDemuxContexts for all streams.

This fixes issue 2448.

Originally committed as revision 26107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-27 09:56:19 +00:00
Martin Storsjö
8c579c1c60 rtsp: Require the transport reply from the server to match the request
This fixes a crash if we requested TCP interleaved transport, but the
server replies with transport data for UDP. According to the RFC, the
server isn't allowed to respond with another transport type than the
one requested.

Originally committed as revision 26077 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-23 15:05:24 +00:00
Martin Storsjö
bbd8f5477d rtsp: Don't set the RTP time base from the sample rate if no sample rate is set
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.

The stream that triggered the fix in 26016 still works after this commit.

Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-15 21:06:25 +00:00
Martin Storsjö
86b6e387cc rtsp/rtpdec: Set the AVStream time_base directly in rtsp when it is known
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).

Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.

Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:29:44 +00:00
Martin Storsjö
bb776f3b00 rtsp: Parse RealRTSP sample rate declarations from the SDP
The RTP time base can be different from the actual content sample rate.

Originally committed as revision 25907 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:28:45 +00:00
Martin Storsjö
6a7e31a901 rtsp: Look for RTP payload handlers for static payload types, too
Originally committed as revision 25893 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:44 +00:00
Martin Storsjö
003eb64217 rtsp: Factorize code for initializing the rtp payload handler
Originally committed as revision 25892 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:09 +00:00
Martin Storsjö
0b6a7ff4b4 rtsp: Do a forgotten reindenting
Originally committed as revision 25839 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-28 21:17:39 +00:00
Martin Storsjö
dd22cfb101 rtsp: Parse and use the Content-Base reply header, if present
This fixes playing RTSP urls with query parameters.

Originally committed as revision 25755 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-15 15:08:53 +00:00
Martin Storsjö
0526c6f7c7 rtsp: Split out the RTSP demuxer functions to a separate, new file
Originally committed as revision 25601 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-29 08:43:57 +00:00
Martin Storsjö
c2688f3ac8 rtsp: Move rtsp_setup_output_streams into rtspenc.c
Originally committed as revision 25600 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-29 08:41:49 +00:00
Martin Storsjö
47bfe49c64 rtsp: Add stub declarations of the setup_in/output_streams functions
This may be needed to avoid calls to implicitly defined functions
(that will be removed by dead code elimination later anyway).

Originally committed as revision 25585 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-27 00:42:35 +00:00
Aurelien Jacobs
a5cea13202 drop rtsp_default_protocols which is not part of public API and not used anymore
Originally committed as revision 25557 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-23 16:22:36 +00:00
Aurelien Jacobs
67f34aaa97 use rtp_get_local_rtp_port() instead of the deprecated rtp_get_local_port()
Originally committed as revision 25554 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-23 16:19:53 +00:00
Martin Storsjö
eced8fa02e rtsp: Move the rtsp_probe function to the demuxer code block
This function is only used by the RTSP demuxer.

Originally committed as revision 25537 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-21 12:25:12 +00:00
Martin Storsjö
44b70ce563 rtsp: Untangle the dependencies between the RTSP/SDP demuxers and RTSP muxer
This allows compilation of one of them without requiring the others'
dependencies to be present.

Originally committed as revision 25535 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-21 12:18:48 +00:00
Martin Storsjö
8bf0f96954 rtsp: Reorder functions
Originally committed as revision 25534 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-21 12:13:02 +00:00