Commit graph

619 commits

Author SHA1 Message Date
Patrick Radizi
3de0244532 rtpbin: add option for sanity checking timestamp offset
Timestamp offsets needs to be checked to detect unrealistic values
caused for example by NTP clocks not in sync. The new parameter
max-ts-offset lets the user decide an upper offset limit. There
are two different cases for checking the offset based on if
ntp-sync is used or not:
1) ntp-sync enabled
   Only negative offsest are allowed since a positive offset would
   mean that the sender and receiver clocks are not in sync.
   Default vaule of max-ts-offset = 0 (disabled)
2) ntp-sync disabled
   Both positive and negative offsets are allowed.
   Default vaule of max-ts-offset = 3000000000
The reason for different default values is to be backwards
compatible.

https://bugzilla.gnome.org/show_bug.cgi?id=785733
2017-09-15 13:33:14 +03:00
Patrick Radizi
23f7739ba4 rtpbin: add option for increasing ts_offset gradually
Instant large changes to ts_offset may cause timestamps to move
backwards and also cause visible effects in media playback. The new
option max-ts-offset-adjustment lets the application control the rate to
apply changes to ts_offset.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2017-09-14 13:15:56 +03:00
Sebastian Dröge
cd17c71dce rtspsrc: Create send/recv mutexes once, not on every connect()
Also fixes a crash caused by freeing an uninitialized mutex in an error
case.

https://bugzilla.gnome.org//show_bug.cgi?id=784282
2017-06-29 18:59:58 +03:00
Sebastian Dröge
c438545dc9 rtspsrc: Actually use the receive lock when receiving, not the send lock 2017-06-22 11:39:16 +03:00
Mathieu Duponchelle
0da5679c6f rtspsrc: do not checksum the stream id
https://bugzilla.gnome.org/show_bug.cgi?id=783307
2017-06-16 17:30:11 +02:00
Sebastian Dröge
a722f6e832 rtspsrc: Use a mutex for protecting against concurrent send/receives
We currently send data to the RTSP connection from multiple threads:
whenever a command is to be handled and whenever RTCP is generated. This
can cause data corruption or worse if both happen at the same time.

As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
calls with a mutex. While this means that we hold a mutex during the IO
operation, this is not actually a problem as the IO operation can be
interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
itself anyway.
2017-06-15 15:25:23 +03:00
Mathieu Duponchelle
f6283b082e rtspsrc: uniquify stream ids
https://bugzilla.gnome.org/show_bug.cgi?id=783307
2017-06-07 23:30:05 +02:00
Nicolas Dufresne
b68d936ae0 Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-16 14:41:19 -04:00
Sebastian Dröge
c99f7579f3 rtspsrc: Chain up to the parent class' provide_clock() implementation
If no clock was provided directly by rtspsrc. This behaviour was removed
by f8013487c9 and results in rtspsrc not
providing the system clock via the rtpjitterbuffer.

As a result, if another element like an audio sink, provides a clock,
the pipeline would select that (when going to PAUSED/PLAYING again later).
Audio clocks usually don't progress in PAUSED, and thus our live source
won't be able to use the clock to produce data, making the sink never
preroll and everything is stuck.
2017-04-21 19:14:09 +01:00
Edward Hervey
474819cc3d rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS
Allows the application to know the exact status code that was returned
by the server in a programmatic fashion.

https://bugzilla.gnome.org/show_bug.cgi?id=781304
2017-04-17 13:10:06 +02:00
Edward Hervey
49002fa8a7 rtspsrc: Properly notify missing elements
If the srtp elements are not present, post a message on the bus
informing about the missing plugins.
2017-02-13 11:17:25 +01:00
Matt Staples
a8eb0515f1 rtspsrc: find_stream_by_channel should ignore unconfigured streams
https://bugzilla.gnome.org/show_bug.cgi?id=777101
2017-01-26 15:31:47 +02:00
Tim-Philipp Müller
d7b2820b73 Fix indentation 2017-01-09 19:05:10 +00:00
Matt Staples
bc2adef550 rtspsrc: Fix session cleanup when handling redirect on PLAY
Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
removing code from gst_rtspsrc_send that changed the state varable upon
encountering a redirect. Better to let the redirect handlers in
gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
state-dependent cleanup.

https://bugzilla.gnome.org/show_bug.cgi?id=775543
2016-12-05 18:23:31 +02:00
Aleix Conchillo Flaque
bc9f06f37b rtspsrc: always send teardown request
Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE
and ignore CMD_WAIT while closing.

https://bugzilla.gnome.org/show_bug.cgi?id=748360
2016-12-05 11:32:43 +02:00
Sebastian Dröge
058ab0df1d rtspsrc: Move to new helper function to parse authentication responses
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-21 09:40:08 +02:00
Matt Staples
cd71e3a8e8 rtspsrc: Also handle redirect on PLAY
https://bugzilla.gnome.org/show_bug.cgi?id=772610
2016-11-01 20:14:35 +02:00
Petr Kulhavy
5cdf66d5d2 rtspsrc: allow missing control attribute in case of a single stream
Improve RFC2326 - chapter C.3 compatibility:
In case just a single stream is specified in SDP and the control attribute
is missing do not drop the stream but rather assume "a=control:*"

https://bugzilla.gnome.org/show_bug.cgi?id=770568
2016-11-01 20:13:49 +02:00
Mark Nauwelaerts
735924236e rtspsrc: reset connection info to non-flushing when closing
This solves a hanging mainloop in following scenario:
* connect to source
* network/server drops
* pipeline set to NULL (and connection to flushing as part)
* pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded)
* [connecting still not possible]
* pipeline set to NULL => mainloop hangs (since no actual flushing is done)
2016-10-26 12:30:39 +02:00
Sebastian Dröge
eaae016884 rtspsrc: Use new bin suppressed flags API for managing the element flags 2016-09-15 18:20:30 +02:00
Thibault Saunier
150edef830 Use the new API to post flow ERROR messages on the bus
https://bugzilla.gnome.org/show_bug.cgi?id=770158
2016-08-26 19:23:26 -03:00
Nirbheek Chauhan
b09f478e80 Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:21:12 +01:00
Sebastian Dröge
0b0a042781 rtspsrc: Allow mimetypes with properties as long as they're application/sdp
Some servers add properties like charset, e.g.
  application/sdp; charset=utf8

Ideally we should also parse the charset and do conversion of all messages,
but that's for a later time.
2016-08-17 09:49:04 +03:00
Steven Hoving
ec59291b2e rtspsrc: Fix error messages to first convert to doubles before division 2016-07-06 11:22:53 +03:00
Sebastian Dröge
b9532527ec rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
There's a small window for a race condition otherwise.
2016-07-05 21:11:35 +03:00
Brad Lackey
6d3071f200 rtspsrc: Don't disable UDP protocols on redirecting
https://bugzilla.gnome.org/show_bug.cgi?id=768232
2016-07-01 12:21:43 +02:00
Sebastian Dröge
c18b609c06 rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
If we consider the RTSP state, what can happen is that it is PLAYING but the
element already asynchronously tried to PAUSE and it just did not happen yet.

We would then override this setting to PAUSED (while the element actually is
in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
to produce packets while the sinks are all PAUSED, piling up thousands of
packets in the rtpjitterbuffer and other elements and finally failing.
2016-06-28 11:01:24 +03:00
Aleix Conchillo Flaqué
12eb5d6912 rtspsrc: make all srtp encoder properties explicit
The Session Data Protocol doesn't allow specifying a cipher for the
SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
"aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.

https://bugzilla.gnome.org/show_bug.cgi?id=767799
2016-06-20 09:53:24 +02:00
Guillaume Desmottes
a6c4763b42 rtpdec: fix clock leak
gst_system_clock_obtain() returns a new ref.

https://bugzilla.gnome.org/show_bug.cgi?id=766521
2016-05-17 09:59:08 +03:00
Sebastian Dröge
7c728db1f3 rtspsrc: Update caps for TCP whenever they change
We only changed them for UDP so far, which caused the wrong seqnum-base and
other information to be passed to rtpjitterbuffer/etc when seeking. This
usually wasn't that much of a problem as the code there is robust enough, but
every now and then it causes us to drop up to 32756 packets before we
continue doing anything meaningful.

https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-04-27 20:52:32 +03:00
Sebastian Dröge
d24e68719b rtspsrc: Don't propagate spurious state change returns from internal elements further
We handle them inside rtspsrc and override them in all other cases anyway, so
do the same for "internal" state changes like PAUSED->PAUSED and
PLAYING->PLAYING.

This keeps unexpected NO_PREROLL to confuse state changes in GstBin.

See also https://bugzilla.gnome.org/show_bug.cgi?id=760532

https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-04-27 20:52:15 +03:00
Aleix Conchillo Flaqué
c36930535d rtspsrc: add srtp rollover counters from mikey crypto sessions
The server can send multiple crypto sessions, one for each SSRC with its
own rollover counter. We parse this information and pass it to the SRTP
decoder via the "request-key" signal.

https://bugzilla.gnome.org/show_bug.cgi?id=730540
2016-04-15 18:12:06 +02:00
Sebastian Dröge
df247f091c rtpjitterbuffer: Add RFC7273 media clock handling
https://bugzilla.gnome.org/show_bug.cgi?id=762259
2016-04-03 11:24:34 +03:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763076
2016-03-24 14:32:20 +02:00
Jihae Yi
da5c8a954c rtspsrc: avoid potentially overflowing expression
https://bugzilla.gnome.org/show_bug.cgi?id=757569
2016-03-24 14:28:50 +02:00
Sebastian Dröge
bf5a72a6dd rtspsrc: Properly error out if binding the UDP sockets fails
udpsrc is not returning us a socket in that case.
2016-02-28 13:01:34 +02:00
Aleix Conchillo Flaqué
665d14a2a0 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
We check the stream profile and use the proper RTCP caps:
application/x-srtcp if we are using a secure profile and
application/x-rtcp otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=760556
2016-01-18 11:29:25 +02:00
Hyunjun Ko
3300039513 rtspsrc: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:12:09 +02:00
Evan Callaway
4718870959 rtspsrc: Retry connection if tunneling needs authentication
Leverage response from gst_rtsp_connection_connect_with_response to
determine if the connection should be retried using authentication.  If
so, add the appropriate authentication headers based upon the response
and retry the connection.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:42:27 +01:00
Luis de Bethencourt
4735d2a9a5 rtspsrc: check port-range format
The string could exist but with a wrong format, in that case we still want
to reset the values of client_port_range.min and max like we do if there is
no string.

CID 1139593
2015-12-14 14:53:57 +00:00
Reynaldo H. Verdejo Pinochet
a4c8ec8bd7 rtspsrc: remove unnecessary NULL checks before g_free() 2015-11-15 01:43:08 -08:00
Sebastian Dröge
01c0f8723f rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
https://bugzilla.gnome.org/show_bug.cgi?id=755125
2015-09-25 23:55:05 +02:00
Martin Kelly
00a938f134 rtsp: fix small README typo
https://bugzilla.gnome.org/show_bug.cgi?id=754807
2015-09-10 08:43:20 +01:00
Jan Schmidt
b38e24995b rtspsrc: Don't parse key data more than needed.
When an auxilliary streams are present in the SDP media,
there's no need to re-parse the SDP attributes multiple
times.
2015-09-05 23:44:51 +10:00
Jan Schmidt
fe4ed1d1df rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).

When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.

Fix a memory leak handling Mikey data.

When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
2015-09-05 23:44:51 +10:00
Vineeth TM
ba8cda54f4 rtspsrc: Trivial fix to check correct condition
When checking for describe method, because of missing parentheses, wrong
condition is being checked, which will result in wrong behavior.

https://bugzilla.gnome.org/show_bug.cgi?id=753912
2015-08-21 11:06:57 +03:00
George Kiagiadakis
e2f2f087ec rtspsrc: send the User-Agent header
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.

This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"

https://bugzilla.gnome.org/show_bug.cgi?id=750101
2015-08-14 15:59:06 +02:00
George Kiagiadakis
af03341e26 rtspsrc: wrap gst_rtsp_message_init_request in a local function
This will allow adding common request initialization, like the
user agent string, in just one place.
2015-08-14 15:59:06 +02:00
Vineeth TM
cf19525d5c rtspsrc: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-30 15:51:25 +03:00
Sebastian Dröge
39a90710b7 rtspsrc: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 14:31:49 +01:00