This new class implements the existing FieldTrialsView interface,
extending it with the verification functionality. For now, the
verification will only be performed if the rtc_strict_field_trials GN
arg is set.
Most classes extending FieldTrialsView today have been converted to
extend from FieldTrialsRegistry instead to automatically perform
verification.
Bug: webrtc:14154
Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38453}
Remove useless comments and properly test frame values. Also rename the
FakeScreenCastStream to TestScreenCastStreamProvider.
Bug: webrtc:13429
Change-Id: I9b1943f0903101a1d9228cded541d3766879d84f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279740
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38450}
We already communicated SPA_META_VideoDamage before, but we never used
these metadata. This change checks whether SPA_META_VideoDamage metadata
are available and construct a damage rect combined from all sent damage
regions.
Bug: webrtc:13429
Change-Id: I326109b4bacf51855904e53345c671640d670323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278820
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38449}
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.
In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.
[1] https://crbug.com/webrtc/10335
Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.
Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}
which is a more modern way to represent something that either has a value or is not set
BUG=webrtc:14544
Change-Id: I0a06b30b1c7f802247eb1f60e69271594b94a6f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278421
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38443}
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats
There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.
In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.
# Unrelated infra failures
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
Investigation showed that a function is revalidating STUN packets
against the wrong password.
This CL also allows absl/strings/escape.h as #include.
Bug: chromium:1177125
Change-Id: Ie068d4c076a5462f2922a012f5e1de23aa6c0b06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279560
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38438}
This patch adds variant of PeerConnectionInterface::AddTrack
that takes an initial_send_encodings.
This allows for setting/modifying encoding parameters before sdp
negotiation is performed/complete (e.g requested_resolution).
This is already available if using RtpTransciverInit and AddTransceiver,
but was not added to AddTrack because of concerns that it complicated matching with existing transceivers. This CL sidesteps that by never matching to a preexisting transceiver if initial_send_encodings are specified.
Note:
1) The patch adds a new method rather than an extra (e.g optional)
argument to existing AddTrack. This is to avoid problems with downstream mocks.
2) chromium "problems" was fixed in https://chromium-review.googlesource.com/c/chromium/src/+/3952684 and https://chromium-review.googlesource.com/c/chromium/src/+/3956060
Bug: webrtc:14451
Change-Id: I19b5a03872730280fbf868ca5d3a2f46443359f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38437}
Typically, remote candidates come from signalling and are deserialized
into C++ objects. The network_type field of these candidates is
always ADAPTER_TYPE_UNKNOWN.
However, in tests it is common to pass SDP and remote candidates as C++
objects. In this case, the network_type property of remote candidates
is preserved, so DCHECK might be triggered when GetStats is called.
Clearing fields that are not suitable as remote candidates fixes
this issue.
Bug: None
Change-Id: Ida01b0224bce5cf3e87bcad1ddaca35c9f4fffe7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38436}
ScalabilityMode should be validated against the currently
allowed codecs or the currently used codec.
Bug: webrtc:11607
Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38433}
This also swithces lifetime management of entries to using
std::unique_ptr.
Bug: chromium:1374310
Change-Id: I5a2c89e9b3bcf7bceec2a4e5347750540ee21e1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279521
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38432}
This test created another PipeWire stream we can connect to with
SharedScreenCastStream and recieve frames from there. This is an
initial version, where I test whether we can successfuly connect
and disconnect, receive frames and it also tests DesktopFrameQueue.
In the future I will add tests to test mouse cursor and try to
come up with some corner cases and possible scenarios.
Bug: webrtc:13429
Change-Id: Ib2a749207085c6324ffe3d5cc8f2f9c631fa6459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256267
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38431}
Track which connection instances are associated with with TurnEntry
instances. This fixes a bug whereby removing an entry by address that
more than one Connection instances had in common, caused operations
such as SendTo and HandleConnectionDestroyed to fail because no
entry could be found.
Also, as requested: A ham sandwich walks into a bar and orders a beer, bartender says “sorry, we don’t serve food here.”
Bug: chromium:1374310
Change-Id: Ie45a99346f8015b10a212d80fbd32255d1544669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279264
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38429}
Currently if you want to obtain the stats for a specific sender/receiver
in Android, you need to call peerConnection.getStats() and filter
manually the result by sender.
pc.getStats(receiver/sender) exists in c++ and ios but was not exposed
in Android
Bug: webrtc:14547
Change-Id: I9954434880f0f93821fcd2e2de24a875e8d136ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275880
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38428}
The tool will generate a C++ header with all field trials in
REGISTERED_FIELD_TRIALS. This registry will later be used while looking
up field trials from native code to ensure they have been properly
registered in accordance with the policy.
Bug: webrtc:14154
Change-Id: I29bf880735121034585c541c46ef19f617d0afb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276268
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38426}
Tested: Chromium built with this change; verified that the
behavior at the beginning of the call has not changed with
both low (< 12) and high (> 12) input volumes.
Bug: webrtc:7494
Change-Id: Ie184c994d46bf6fd1cb209873383b911beb766e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278787
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38420}
Ownership does not need to cross the interface boundary, so ArrayView can be safely accepted in ForgetStateForConnections and PruneConnections.
Bug: webrtc:14131
Change-Id: I18a739aea1dc47976d17925e9bca3461225bf803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278629
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38418}
This reverts commit c1d5fda22c8ae456950c5549d22d099b478c67e2.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
This reverts commit 332810ab5d41862b8f85ef30e84dbec4241f8b21.
Reason for revert: This commit chain seems to cause problems in LossBasedBwe.
Original change's description:
> Probing integration in loss based bwe 2.
>
> - Loss based bwe has 3 states: increasing (increasing when loss limited), decreasing (decreasing when loss limited), or delay based bwe (the same as delay based estimate).
> - When bandwidth is loss limited and decreasing, and probe result is available, GetLossBasedResult = min(estimate, probe result).
> - When bandwidth is loss limited and increasing, and the estimate is bounded by acked bitrate * a factor.
> - When bandwidth is loss limited and probe result is available, use probe bitrate as the current estimate, and reset probe bitrate.
>
> Bug: webrtc:12707
> Change-Id: I53cb82aa16397941c0cfaf1035116f775bdce72b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277400
> Commit-Queue: Diep Bui <diepbp@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38382}
Bug: webrtc:12707
Change-Id: Ied86323b0ce94b87ac503a2ee34753cebef5f53d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38412}
This reverts commit aa71259b06b72ba0fc5a28c0ffa4891c69c09441.
Reason for revert: This commit chain seems to cause problems in LossBasedBwe.
Original change's description:
> Probe when network is loss limited.
>
> Trigger probes next process intervals if the loss based current state is either increasing or decreasing. 0/ first probe at the loss based estimate. 1/ if increasing: allow further probing. 2/ if decreasing: not allow further probing.
>
>
> Bug: webrtc:12707
> Change-Id: I4e99edcbe4e2c315e8498ffb7fb2e589cdb4e666
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279041
> Commit-Queue: Diep Bui <diepbp@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38395}
Bug: webrtc:12707
Change-Id: I1fb61337148faf6faaa0056dc25f14536a19a462
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38410}
While looking into chromium:1374310 I noticed that the function was
returning 0 in a particular case. 0 isn't a valid return value as per
this shortened snippet from connection.cc [1] specifically meant to
catch this:
int sent = port_->SendTo(...);
if (sent <= 0) {
RTC_DCHECK(sent < 0);
error_ = port_->GetError();
...
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/p2p/base/connection.cc;l=1687
Also propagating the socket error value in case of failure.
Bug: chromium:1374310
Change-Id: Ie00f60388d53d4127c1d419ab0352e0574044485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38408}
