Commit graph

50 commits

Author SHA1 Message Date
Mathieu Duponchelle
7563a68ec8 rtpbasepayload: do not forget delayed segment when forwarding gaps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/797>
2020-09-08 23:01:46 +00:00
Matthew Waters
a1e9f4e37b rtpbasepayload: place twcc-ext-id behind environment variable
Adding properties for each and every rtp header extension is not
scalable and a new interface will be implemented for the general case
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/777).

Set the environment variable "GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY"
to any value to reenable the short-lived twcc-ext-id property.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/761

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/756>
2020-07-21 11:57:55 +00:00
Tobias Ronge
f1b3ed37c6 gstrtpbasepayloader: Add property for scaling RTP timestamp
This patch introduces a property which, if set to FALSE, prevents RTP
basepayloader from scaling the RTP time when a segment's rate is not
equal to 1.0. The specification is ambiguous on this subject and some
clients expect the timestamps not to be scaled.
2020-03-16 10:25:44 +00:00
Håvard Graff
85e201fe30 rtpbasepayload: add property for embedding twcc sequencenumbers
By setting the extension-ID for TWCC (Transport Wide Congestion Control),
the payloader will embed sequencenumbers as a RTP header-extension
according to https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-2

The negotiation of this being enabled with downstream elements
is done with caps reflecting the way this is communicated using SDP.
2020-02-14 09:40:59 +00:00
Kristofer Björkström
4152b0c840 rtpbasepayload: timestamp bug, if rate control=no
With commit "basepayload: Expose onvif-no-rate-control property" the rtp
timestamp changed behaviour when rate control is disabled.

When disabling rate control, we must take care of the stream time to
avoid the timestamps to begin from zero again.
2020-02-11 12:30:49 +00:00
Havard Graff
f7408f9418 rtpbasepayload: don't use GINT_TO_POINTER with GType
GType can (and will) be 64bit. GINT_TO_POINTER is not.
This will result in the api-type checked for being a different one than
it actually is...
2019-06-12 12:38:26 +00:00
Mathieu Duponchelle
3c4bef46b7 basepayload: Expose onvif-no-rate-control property
The ONVIF spec mandates that when Rate-Control=no, the RTP timestamps
match the original sampling times, as opposed to the intended playback
time.
2019-04-05 16:42:55 +00:00
Antonio Ospite
1eb9c5b309 rtpbasepayload: print list size in log output instead of -1
It is weird to see "Preparing to push packet with size 4294967295" in
the logs, so print the list length in case of a buffer list.
2019-03-15 17:38:58 +01:00
Linus Svensson
72ecbe2aef rtpbasepayload: Update current seqnum for buffer lists
The current sequence number will be the one from the first RTP buffer
when a buffer list is pushed, but should be the last one.

Fixes #495
2018-11-14 12:30:06 +00:00
Stian Selnes
f766b85b96 rtpbasepayload: rtpbasedepayload: Add source-info property
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947
2018-10-10 14:38:01 -04:00
Tim-Philipp Müller
7f9730ecf4 rtp: Update for g_type_class_add_private() deprecation in recent GLib
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
2018-06-23 22:22:22 +02:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
2017-03-10 18:19:17 -03:00
Sebastian Dröge
b9f59fd999 rtpbasepayload: Ensure to set the RECONFIGURE flag again if reconfiguration failed
https://bugzilla.gnome.org/show_bug.cgi?id=774623
2016-11-18 12:04:27 +02:00
Sebastian Dröge
2c29f09da8 rtpbasepayload: Handle gst_pad_get_current_caps() returning NULL gracefully 2016-02-23 18:23:45 +02:00
Sebastian Rasmussen
042e71a117 rtpbasepayload: Implement video SDP attributes
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472
2015-10-02 17:44:14 +03:00
Sebastian Dröge
3113e341ea rtpbasepayload: Always prefer downstream's ssrc suggestion if any
Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
are not possible anymore. rtpsession was now patched to only suggest an ssrc
if it makes sense to do so.

In 2.0 we should get rid of all the properties that are also negotiated via
caps, the code and behaviour is too confusing otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=749581
2015-06-05 16:44:08 +02:00
Sebastian Dröge
faafaaec56 rtpbasepayload: Try harder to reuse previously configured caps values and give more preference to anything set as properties
This affects the pt, ssrc, seqnum-offset and timestamp-offset properties. If
they were set from a property, or we configured caps before, we try to use
that value for them. Even if the first structure of the downstream caps
specifies a different value, we check if the value is supported by other
structures.
Only if all this fails, we use the values given by downstream in the first
structure, i.e. if no properties were set and these are the first caps we
negotiate or downstream does not support our values.

By doing this we ensure that we don't spuriously change ssrcs or other fields
in the middle of the stream (and also consider property values more). Ssrc
changes would currently happen after sending an RTX packet (thus creating a
new internal source inside the rtpsession), and then renegotiating the
payloader (which then gets the RTX ssrc from rtpsession).

https://bugzilla.gnome.org/show_bug.cgi?id=749581
2015-05-19 16:59:45 +03:00
Olivier Crête
bdf8ce286d rtpbasepayload: Implement reconfigure event & renegotiation without subclass
Implement the reconfigure event, also do correct downstream caps negotiation
if the subclass doesn't implementy set_caps.

https://bugzilla.gnome.org/show_bug.cgi?id=725361
2014-05-03 10:21:04 +02:00
Olivier Crête
deb27bddf6 rtpbasepayload: Save the PT after fixating 2014-05-02 18:30:23 -04:00
John Bassett
0fd60ac858 rtpbasepayload: restrict initial random sequence number to be <= 32767
In order to prevent SRTP roll over counter issues the initial sequence
number is restricted to <= 32767. This is recommended by RFC 4568 section 6.4.
2014-05-01 17:00:47 -04:00
Wim Taymans
314eee6dd1 rtpbasepayload: update docs 2014-04-12 06:43:24 +02:00
Wim Taymans
f0348d7005 rtpbasepayload: add current timestamp and seqnum offset to stats
Expose the current timestamp and seqnum offset in the stats

See https://bugzilla.gnome.org/show_bug.cgi?id=646577
2014-04-12 06:27:36 +02:00
Wim Taymans
bf4079277d rtpbasepayload: add pt and ssrc to stats 2014-03-20 09:19:46 +01:00
Sebastian Rasmussen
d6dc1b6c46 rtpbasepayload: Let caps event also configure seqnum-offset
Previously the sequence number kept track of by GstRTPBasePayload would
only be set when going from READY to PAUSED state. This meant that a
downstream element that attempted to configure a basepayloader by
setting seqnum-offset e.g. in its sinkpad's caps template would have
trouble configuring the basepayloader. The reason was that the caps
event which arrives with the desired value for seqnum-offset did not
arrive at the basepayloader until caps negotiation took place,
significantly later than the transition from READY to PAUSED.

The result after this patch is that the default value for the
seqnum-offset property, or later set values for this property, will take
effect when going from READY to PAUSED like before. In addition the an
arriving caps event will also affect the basepayloaders configured
sequence number as the event arrives.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen
638d069c91 rtpbasepayload: Fix payload type property boundary value
The payload type field in an RTP packet header is 7 bits wide, hence the
boundary values ought to be 0x00 and 0x7f, not the previously stated
values 0x00 and 0x80.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen
125b9c19c0 rtpbasepayload: Do cosmetic changes to rtptime calculations
* Change running time type to guint64
 * Use GST_CLOCK_TIME_NONE() to check for invalid timestamps
 * Name variables so ns-based and hz-based timestamps are evident

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen
0142cd5e35 rtpbasepayload: Expose running-time of payloaded stream
https://bugzilla.gnome.org/show_bug.cgi?id=719415
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen
865a5d1c8f rtpbasepayload: Improve documentation for perfect-rtptime
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen
713dfe0d70 rtpbasepayload: Fix typos in documentation for properties
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen
fa393e5d60 rtpbasepayload: Add statistics property
This property allows for an atomically retrieved set of properties that
can e.g. be used to generate RTP-Info headers.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719415
2014-01-27 15:11:09 +01:00
George Kiagiadakis
6108407db1 gstrtpbasepayload: use the session's suggested ssrc after a collision, if the session provides one
Conflicts:
	gst-libs/gst/rtp/gstrtpbasepayload.c
2013-12-30 13:13:35 +01:00
Julien Isorce
71788c1432 rtpbasepayload: change SSRC on GstRTPCollision event
Change our SSRC and update the caps when we receive a GstRTPCollision
event from downstream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-12-12 13:44:15 +01:00
Julien Isorce
6f614e1225 rtpbasepayload: implement src_event function
Add a srcpad event handler and call the src_event vmethod.
2013-12-12 13:16:01 +01:00
Ognyan Tonchev
25fdde908a rtpbasepayload: Do not leak the event when segment is delayed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119
2013-06-26 15:45:30 +02:00
Nicolas Dufresne
94b7ae7767 rtpbasepayload: Delay segment event after caps
https://bugzilla.gnome.org/show_bug.cgi?id=700222
2013-05-14 09:50:22 +02:00
Tim-Philipp Müller
21c61586ad rtpbasepayload: error out if no CAPS event was received before buffers
Most payloaders set/send their own output format from the setcaps
function, so if we don't get input caps, things probably wont' work
right, even if the input format is fixed (as in the case of the mpeg-ts
payloader for example).

https://bugzilla.gnome.org/show_bug.cgi?id=683428
2012-09-06 18:23:22 +01:00
Tim-Philipp Müller
3d006f6d2a rtpbasepayload: assume input caps are accepted if subclass has no set_caps vfunc
Not that anyone should ascribe too much meaning to these return
values in the age of sticky caps.
2012-09-06 17:47:01 +01:00
Wim Taymans
11a494d5c9 rtp: Add support for multiple memory blocks in RTP
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
2012-07-17 16:41:36 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Mark Nauwelaerts
6039266c43 rtpbasepayload: plug caps leak 2012-03-29 17:15:43 +02:00
Wim Taymans
37e940df83 rtpbasepay: add support for DTS and PTS 2012-03-13 18:15:04 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Olivier Crête
1a592199e9 rtpbasepayload: Port to group-less GstBufferList 2012-01-25 11:55:02 +01:00
Edward Hervey
94230af7a3 rtp: Don't forget to initialize GstRTPBuffer 2011-12-05 18:30:37 +01:00
Wim Taymans
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
026ec68f75 _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
2011-11-15 18:04:16 +01:00
Wim Taymans
107d5a3d05 rtp: fix headers
indent, add padding, remove old abidata
2011-11-11 19:21:09 +01:00
Wim Taymans
5f1312b5d8 rename files to match object names 2011-11-11 12:32:23 +01:00
Renamed from gst-libs/gst/rtp/gstbasertppayload.c (Browse further)