Unlike the cache of the entire stats report which is time limited, this
certificate cache is valid for an unlimited amount of time, but is
cleared at ClearCachedStatsReport() which is already called on each
SLD/SRD call. Since certificates can only change by negotiation, this
cache is ensured to always be invalidated when certificates change.
Since ClearCachedStatsReport() can happen for other reasons than
certificates changing we may clear the cache more often then is
necessary, but arguably this is seldom enough that we don't have to
create a separate "ClearCertificateStats()" method. Keep it simple?
The cache specifically avoids rtc::SSLCertChain::GetStats which
trigger rtc::SSLCertificate::GetStats and rtc::Base64::EncodeFromArray.
Bug: webrtc:14458
Change-Id: I5f95a4a5eb51cc4462147270fdae7bb9fb7bc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276602
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38205}
In non-BUNDLE use cases, it is possible for multiple RTP streams to have
the same SSRC (as long as the SSRC is unique within the same transport).
This CL adds support for "outbound-rtp" and "inbound-rtp" stream stats
to have the same SSRC on different transports by adding the transport to
the stats ID. This avoids multiple RTP stream stats having the same
stats ID and fixes the problem. It's a stupid use case, but it should
work.
There could still be a stats ID collision in the event of multiple
"remote-inbound-rtp" or "remote-outbound-rtp" reference the same SSRC
but on separate transports for the same reason, and would require the
same fix... but one bug at a time. Not addressed in this CL.
Bug: webrtc:14443
Change-Id: I1a2ffd79fc67c2765e6dbd1ccc6828d4e91c4589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38201}
Error message:
Traceback (most recent call last):
File "/b/s/w/ir/cache/builder/src/tools_webrtc/autoroller/roll_deps.py", line 778, in <module>
sys.exit(main())
File "/b/s/w/ir/cache/builder/src/tools_webrtc/autoroller/roll_deps.py", line 748, in main
raise RollError('WebRTC DEPS entries are missing from Chromium: %s.\n'
__main__.RollError: WebRTC DEPS entries are missing from Chromium: ['src/third_party/fuchsia-sdk/sdk'].
Remove them or add them to either WEBRTC_ONLY_DEPS or DONT_AUTOROLL_THESE.
src/third_party/fuchsia-sdk/sdk has been removed from
https://crrev.com/c/3914609.
Bug: None
Change-Id: Ic2b7b39ffd1a3e5fb9bb73ddb1318a5e36b0bc30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276720
Reviewed-by: Andrey Logvin <landrey@google.com>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38199}
To properly handle SSRC collisions in non-BUNDLE we need to change how
RTP stats IDs are generated, but that is a riskier change to be dealt
with in a separate CL.
For now, we just make sure that crashing is not a possibility during
SSRC collisions as a mitigation for https://crbug.com/1361612. This is
achieved by adding a TryAddStats() method to RTCStatsReport returning
whether successful.
Bug: chromium:1361612
Change-Id: I8577ae4c84a7c1eb3c7527e9efd8d1b0254269a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275766
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38197}
This CL wraps the |Dav1dPicture| data directly for |VideoFrame| using
instead of copy data out to new buffer.
Bug: None
Change-Id: I21ceffb5cac7dda4a44eafbd0ed221974b8d45ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276526
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38194}
Shared screencast stream is tied to desktop capturer options,
which may outlive capturer itself. This leads to a case where
one may attempt to restart the stream in the capturer. This
causes the previous pipewire objects to leak (as observed
in `pw-top` output) and seems to appear as frozen screen for
clients. This CL ensures that the shared screen cast stream,
which is started in this capturer, is also stopped when the
capturer is destroyed.
Bug: chromium:1291247
Change-Id: I5f2b22e54e916549a5280ec457cd76360e42e48a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276640
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38187}
Chrome Remote Desktop will support both X11 and Wayland desktop
capturers in the near future and we'd like to differentiate between
the two in our video frame stats and telemetry. I beleive other
products are in a similar position so I would like to add a capturer
ID to the frames generated by the capturer classes.
Bug: chromium:1366062
Change-Id: If27c35ad6ef89b6396120982edc4dd0cf2a1e51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276081
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38185}
Creates the EmulatedSFUConfig that will receive the parameters for
controlling the virtual SFU used in the call.
Its current only field is the previous target_spatial_index from
VideoSimulcastConfig.
This allow to filter out the bottom layers for SVC S mode tests
and enable them.
Bug: webrtc:11607
Change-Id: Id4f3a96b3a03b9be7155796c3bafefce01f32b7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274162
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38182}
With rtc::Thread::Clear removed, there are no longer calls to external code while holding the mutex and thus it doesn't need to be recursive.
Bug: webrtc:11567
Change-Id: If350684be0bfcbc33806019bd986138905aec66d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276542
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38179}
Make it possible to use both APIs inside same test and have consistent
export results to the Chrome Perf Dashboard and stdout.
Bug: b/246095034
Change-Id: I924088a2ddcb04981e56bbeb4544ac317833fb98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276540
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38177}
Command [1] on openssl 1.1.1m and newer generates output
containing "unsigned char the_(subject_name|public_key|certificate)"
records, making it incompatible with current version of the script
that relies on "unsigned char XXX_".
This patch handles both cases by using regular expression so as
to match strings and provide an adequate replacement.
[1] - openssl x509 -in <path-to-cacert.pem> -noout -C
Bug: webrtc:11710
Change-Id: I46b87d2980ec2dd26660b93fcf9019254950ce12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38173}
Reporting timeouts is useful for native hw backed codec implementations.
The value is in sync with VideoCodecStatus.java in the Android sdk.
Bug: b/185740707
Change-Id: I9a08a1303586c677be53aaa4f39455f42e519996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38168}
This converts all P2PTransportChannel unit tests to parameterized tests, with a string parameter for the field_trials which is used to enable the refactor. This adds a variation of each existing test using the refactored code path.
Tests are initialized twice, once for legacy and refactored path each, to strike a balance between file name length and descriptiveness.
Bug: webrtc:14367, webrtc:14131
Change-Id: I0469550d571ed389804eb486fe5bd22504e59373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275303
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38162}
If we don't pass the frames to OnFrameEncoded, we can't see the frames
being sent in the SVC tests. We want to check the frames even if the
SFU would discard them later.
Bug: webrtc:11607
Change-Id: I5b9c6a86c0966047efa7be088f90e83e01f7900b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273350
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38159}
