Commit graph

18 commits

Author SHA1 Message Date
Philippe Normand
d317379287 webrtcstats: Properly report IceCandidate type
strcmp returns a positive value if s1 is greater than s2, while we actually
needed to check equality here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4952>
2023-07-03 03:51:53 +00:00
Martin Nordholts
85e3f31740 webrtc: Track stats for data channels opened and closed
Track data channel stats for `dataChannelsOpened` and
`dataChannelsClosed` in `RTCPeerConnectionStats` as specified by
https://www.w3.org/TR/webrtc-stats/#dictionary-rtcpeerconnectionstats-members

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4638>
2023-05-18 04:31:16 +00:00
Jan Schmidt
621604aa3e webrtc: Calculate the jitter for remote-inbound-rtp stats
Populate the clock-rate in the internal stats structure, so
it can be used by the _get_stats_from_remote_rtp_source_stats()
method to calculate remote receivers' jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3900>
2023-02-07 04:58:04 +11:00
Jan Schmidt
615a019457 webrtcbin: Report full codec-stats for source pads
Use the current caps for webrtcbin srcpads, as received_caps
are only stored for sink pads based on incoming caps events.

Makes it so that webrtcbin stats reports contain fuller
codec information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3900>
2023-02-07 04:49:34 +11:00
Philippe Normand
72884f141c webrtcbin: Support for setting kind attribute on RTCRtpStreamStats
The attribute maps the `kind` property of the associated transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3630>
2022-12-22 21:35:51 +00:00
Edward Hervey
a100f36b69 webrtcbin: Don't duplicate enum string values
Some were leaked when debugging was enabled. Instead just directly use the
static strings as-is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3347>
2022-11-07 11:21:00 +00:00
yatinmaan
2c1e61ea16 webrtc: Split WebRTCICE into base classes and implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-07-26 13:51:11 +00:00
Philippe Normand
c19319c777 webrtc: Refactor ICECandidateStats freeing logic to a dedicated function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Sherrill Lin
3e7fb83393 webrtcstats: Improve selected candidate pair stats by adding ICE candidate info
The implementation follows w3.org specs:
* https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*
* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*

Corresponding unit tests are also added.

Rebased and updated from
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462

Fixes #1207

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Sangchul Lee
a801d6dd63 webrtcstats: Unify 'packets-lost' data type to int64
Previously, 'packets-lost' member of RTCReceivedRtpStreamStats had
a value of G_TYPE_INT from rtpsource or a value of G_TYPE_UINT64
from rtpjitterbuffer.
Because of the negative value of estimated amount of packets lost
in rtpsource as well as the description in
https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats
it is fixed to set this value with G_TYPE_INT64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2049>
2022-03-31 05:37:39 +00:00
Matthew Waters
041eee6c2e webrtc: produce stats for all relevant streams
Instead of only using the last ssrc that was pushed into a sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
2377f8b3f2 webrtcbin: initial support for sending and receiving simulcast streams
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
  - mid
  - stream-id
  - repaired-stream-id

Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Philippe Normand
43856a0735 webrtcstats: Fix null pointer dereference
If there is no jitterbuffer stats we should not attempt to store them in the
global stats structure.

Also add a g_return_if_fail in _gst_structure_take_structure() about this
because it is a programmer error to pass an invalid pointer address there.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1479>
2021-12-29 15:55:57 +00:00
Olivier Crête
818a185b5d webrtcstats: Fall back to last packet ssrc if caps dont provide it
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête
4e32d6bf3e webrtcstats: Use our own caps instead of the sticky event
The sticky event seems to get cleared sometimes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête
fc7e7f5ccc webrtc stats: Remove duplicate structure get
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête
f35435f1f7 webrtc stats: Add more details about codecs into the stats
This makes the output a little closer to what the upstream stats are.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Thibault Saunier
019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00
Renamed from ext/webrtc/gstwebrtcstats.c (Browse further)