Commit Graph

496 Commits

Author SHA1 Message Date
Martin Storsjö
e02dcdf6bb rtsp: Allow $ as interleaved packet indicator before a complete response header
Some RTSP servers ("HiIpcam/V100R003 VodServer/1.0.0") respond to
our keepalive GET_PARAMETER request by a truncated RTSP header
(lacking the final empty line to indicate a complete response
header). Prior to 764ec70149, this worked just fine since we
reacted to the $ as interleaved packet indicator anywhere.

Since $ is a valid character within the response header lines,
764ec70149 changed it to be ignored there. But to keep
compatibility with such broken servers, we need to at least
allow reacting to it at the start of lines.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-10-23 10:31:55 +03:00
Eloi BAIL
1ec611a102 rtsp: warning when max_delay reached
packets are queued due to packet reordering until the queue reach its
maximal size or max delay is reached.
This commit adds a warning trace when max delay is reached.

Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-09-16 09:58:12 +03:00
Martin Storsjö
764ec70149 rtsp: Only interpret $ as interleaved packet indicator at the start of replies
Allow $ as character anywhere within normal RTSP replies - both
within the lines, and as the first character of RTSP header lines.
(The existing old comment indicated that an inline packet could
start at any line within a RTSP reply header, but that doesn't
sound valid to me, and I'm not sure if the existing code
handled that correctly either.)

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-07-30 16:59:36 +03:00
Martin Storsjö
b90adb0aba rtsp: Make sure we don't write too many transport entries into a fixed-size array
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-04-24 16:07:14 +03:00
Martin Storsjö
eb7ddb5066 rtsp: Don't warn about unparsed time ranges
This removes the error logging added in 4e54432164.

This avoids warnings about "Invalid interval start specification 'now'"
for live rtsp streams.

We only try to parse some of the many valid values for time ranges
in RTSP - the other ones are fully valid but not interesting for the
use case in rtsp.c, so we shouldn't warn about them.

(Parsing the time ranges is needed to allow seeking, but e.g. setting
the current realtime clock for the start time doesn't make sense.
av_parse_time has got a different mode for parsing absolute times
as well, which can handle the special case "now", but that doesn't
make much sense for this particular use in rtsp.c.)

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-04-23 23:20:02 +03:00
Vittorio Giovara
1a3eb042c7 Replace av_dlog with normal av_log at trace level
This applies to every library where performance is not critical.
2015-04-19 12:41:59 +01:00
Himangi Saraogi
4e54432164 rtsp: Fix unchecked return value
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-04-12 23:08:14 +03:00
Himangi Saraogi
0b72a47f55 rtsp: Check a malloc return value
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-04-12 23:08:13 +03:00
Luca Barbato
e3ec6fe7bb rtsp: Add a buffer_size option
And forward it to rtp and udp.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2015-04-01 14:26:35 +02:00
Martin Storsjö
d594dbecce rtpdec: Rename the free method to close
Many of these functions were named foo_free_context, and since
the functions no longer should free the context itself, only
allocated elements within it, the previous naming was slightly
misleading.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 23:07:50 +02:00
Martin Storsjö
8e32b1f096 libavformat: Use ffio_free_dyn_buf where applicable
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 23:07:47 +02:00
Martin Storsjö
078d43e23a rtpdec: Free depacketizers if the init function failed
This is different from how it is handled in codecs/demuxers/muxers
though (where the close function isn't called if the open function
failed), but since the number of depacketizers that have an .init
function is quite limited, this is easy to change.

The main point is that if the init function failed, we shouldn't
try to use that depacketizer at all - this makes sure that the
parse function doesn't need to check for the things that were
initialized in the init function.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 23:07:39 +02:00
Martin Storsjö
bb4a310bb8 rtpdec: Don't free the payload context in the .free function
This makes it more consistent with depacketizers that don't have any
.free function at all, where the payload context is freed by the
surrounding framework. Always free the context in the surrounding
framework, having the individual depacketizers only free any data
they've specifically allocated themselves.

This is similar to how this works for demuxer/muxers/codecs - a
component shouldn't free the priv_data that the framework has
allocated for it.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 23:07:35 +02:00
Martin Storsjö
f4b59334bd rtpdec: Remove the now unused .alloc field
Always use the .priv_data_size field instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 23:07:32 +02:00
Martin Storsjö
e72605f80b rtpdec: Allow allocating and freeing the private data without explicit functions
This can reduce the amount of boilerplate in simple depacketizers.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 16:23:37 +02:00
Martin Storsjö
b7a4c319fd rtpdec: Allow setting the need_parsing field in RTPDynamicProtocolHandler
This allows getting rid of quite a bit of boilerplate in depacketizers.

The default value (initializing need_parsing to 0, aka
AVSTREAM_PARSE_NONE) is the same as it is initialized to by default
in AVStream.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 16:22:21 +02:00
Martin Storsjö
74d318f138 rtsp: Fix the indentation of a linewrapped statement
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 16:20:10 +02:00
Martin Storsjö
26524e3581 rtsp: Interpret the text media type as AVMEDIA_TYPE_DATA
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-24 16:19:37 +02:00
Gilles Chanteperdrix
cdcc370293 rtsp: punch holes again after pause
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.

Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.

Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-23 16:55:56 +02:00
Gilles Chanteperdrix
4438d1c6ed rtsp: parse lang attribute in SDP
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-21 23:37:24 +02:00
Vittorio Giovara
76ccf114a6 rtsp: check ffurl_get_file_handle() return value
CC: libav-stable@libav.org
Bug-Id: CID 717844
2014-12-18 23:27:14 +01:00
Vittorio Giovara
604c9b1196 rtsp: move the CONFIG_ macros to the beginning of the check
With --disable-optimizations, the DCE of some compilers does not remove
such unused code, causing linking failure.
2014-12-02 15:28:22 +00:00
Martin Storsjö
6df9d9b55d lavf: Use av_gettime_relative
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.

Signed-off-by: Martin Storsjö <martin@martin.st>
2014-10-24 09:53:45 +03:00
Michael Lynch
460b509a34 rtsp: Check a memory allocation
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2014-10-16 09:19:03 +03:00
Luca Barbato
8b2e9636c5 rtsp: Support tls-encapsulated RTSP 2014-10-10 16:29:06 +02:00
Martin Storsjö
2f172f1ae9 rtsp: Clear the session id on redirects
This fixes handling redirects in case the server provided a session
id within the redirect reply.

Signed-off-by: Martin Storsjö <martin@martin.st>
2014-09-24 23:34:02 +03:00
Joshua Kordani
c385313d25 rtsp: Support misordered fmtp rtpmap
Issue present in Avigilon IP cameras.
2014-06-18 22:03:40 +02:00
Luca Barbato
c9c1e00f95 rtsp: Factor out fmtp parsing 2014-06-18 22:03:40 +02:00
Martin Storsjö
f797b134ca rtpenc_chain: Don't copy the time base to the source stream by default
Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.

This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.

This reverts abb810db but achieves the same effect for the other
muxers.

Signed-off-by: Martin Storsjö <martin@martin.st>
2014-05-31 12:02:23 +03:00
Vittorio Giovara
46c0cbd5dc rtsp: suppress a incompatible pointer types warning 2013-12-12 16:10:14 +01:00
Martin Storsjö
50aef03b24 rtspenc: Make sure BYE packets are sent before TEARDOWN
Also make sure the BYE packets are sent at all when using
TCP interleaved transport.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-11-01 09:57:06 +02:00
Martin Storsjö
b56fc18b20 sdp: Add an option for sending RTCP packets to the source of the last packets
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.

With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-14 11:21:33 +03:00
Martin Storsjö
f4d371b973 rtsp: Don't include the listen flag in the SDP demuxer flags
It's only relevant for the RTSP demuxer. Similarly, the custom_io
flag is only present in the SDP demuxer options list.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-07 13:16:44 +03:00
Martin Storsjö
fd8f91e3f4 rtsp: Simplify code for forming the remote peer url
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-01 12:11:17 +03:00
Martin Storsjö
b7e6da988b rtpproto: Move rtpproto specific function declarations to a separate header
Mixing these with the rtp depacketizer functions in rtpdec.h is
no good.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Ed Torbett
1f57d60129 rtsp: Support RFC4570 (source specific multicast) more properly.
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:56 +03:00
Ed Torbett
36fb0d02a1 rtsp: Support multicast source filters (RFC 4570)
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-19 12:02:13 +03:00
Martin Storsjö
1dd1b2332e rtsp: Include an User-Agent header field in all requests
Some rtsp servers like the IP Cam IcyBox IB-CAM2002 need it.

Based on a patch by Carl Eugen Hoyos.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-14 20:53:04 +03:00
Diego Biurrun
2832ea26f3 Remove commented-out debug #define cruft 2013-05-16 00:23:30 +02:00
Diego Biurrun
e0f8be6413 avformat: Add AVPROBE_SCORE_EXTENSION define and use where appropriate 2013-05-04 21:43:06 +02:00
Diego Biurrun
e926b5ceb1 avformat: Drop unnecessary ff_ name prefixes from static functions 2013-04-30 16:16:23 +02:00
Andrew Van Til
350ad50bf4 lavf: Use RTP_MAX_PACKET_LENGTH instead of 1500
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-04-09 10:36:43 +03:00
Martin Storsjö
2326558d52 rtpdec: Split mpegts parsing to a normal depacketizer
This gets rid of a number of special cases from the common rtpdec
code.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-20 18:17:17 +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
54cb096ee4 rtsp: Remove an outdated comment
It is unclear what the bug exactly was and if it ever was fixed,
and we don't even support decoding via faad any longer. The
comment has been present since d0deedcb in 2006.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-12 00:02:11 +02:00
Martin Storsjö
3900d53fb1 rtsp: Remove references to weirdly named variables in other files
One of them is renamed now, but mentioning it by name serves
no purpose here.  The other table mentioned ceased to exist
under that name in 4934884a1 in 2006.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-12 00:02:04 +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
f811cd2d47 rtsp: Respect max_delay for the reordering queue when using custom IO
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-08 11:22:43 +02:00
Martin Storsjö
8729698d50 rtsp: Recheck the reordering queue if getting a new packet
If we timed out and consumed a packet from the reordering queue,
but didn't return a packet to the caller, recheck the queue status.
Otherwise, we could end up in an infinite loop, trying to consume
a queued packet that has already been consumed.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-08 11:22:37 +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