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}
Also call out the places where it happens explicitly - these are places
that need to be redesigned.
Bug: chromium:1177125
Change-Id: I3237d028dbb22380e8fbf7cedb03e965d1fcf2aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279022
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38384}
Recent WebRTC stats spec changes have added restrictions on what stats
are available to JavaScript. This is done to reduce that fingerprinting
surface of WebRTC getStats. For example, stats exposing hardware
capabilities have requirements that must be met by the browser. See [1]
for more details.
This CL adds the types and the enumerations. Stats with these
restrictions should not be added until Chromium has implemented
filtering based on the stat type.
[1] https://w3c.github.io/webrtc-stats/#limiting-exposure-of-hardware-capabilities
Bug: webrtc:14546
Change-Id: I6dae5d4921c7a2bc828a4fc8f7d68e0c59f3be82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279043
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38381}
Only use the network thread for sending and receiving packets.
The one and only network thread is used as a worker thread in all
PeerConnections. Pacing when sending packets is done on the worker thread.
Bug: webrtc:14502
Change-Id: Ib373315688ae4d810ae1e4421101a859fca93b31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278621
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38354}
RTPHeader is not exported, so the TransformableIncomingAudioFrame can't be mocked in chrome tests, using a getter instead.
Bug: chromium:1247260
Change-Id: I2af4e6a88b3f4772b3bb50ee0ae9d5c80fed3ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278785
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38352}
This also tests the UMA stats newly added to it.
Bug: chromium:1177125
Change-Id: I89bb17c1897565cd91ea5bbd92062018317738ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278600
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38345}
The one to use is StunMessage::ValidateMessageIntegrity(password).
Bug: chromium:1177125
Change-Id: I345f4d6b60090651bc23c3aa6358d4fb24723f9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278601
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38338}
This will allow us to see if bad integrity ever occurs, and where integrity
is not applied.
Bug: chromium:1177125
Change-Id: I7abdaba93088e4eef8121205e7dd76b21204cae8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38330}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
This patch
1) modifies VideoAdapter to use requested_resolution
instead on OnOutputFormatRequest, iff there are no active encoders
that is not using requested_resolution (i.e all "old" encoder(s) are
not active).
2) modifies VideoBroadcaster to not broadcast wants from
encoders that are not active (iff there is an active encoder
using requested_resolution).
3) fixes a bug in encoder_stream_factor in that the
requested_resolution was not propagated to return value
(must have been lost in merge?).
Bug: webrtc:14451
Change-Id: I00e0907f0fe9329141ed169576fa46cdc5384886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38323}
They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.
To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.
Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
tkchin and deadbeef are not working on webrtc on a daily basis at the
moment, so non-urgent approvals should not go to them.
Not mentioning this has led to misunderstandings.
Bug: chromium:1371843
Change-Id: I91e99249d32e52d6083de9c2b1bfebfc4693acac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278201
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38314}
These metrics were recently standardized. Part of the standardization
effort was to move them from obsolete "track" stats (on track for
deprecation and removal: https://crbug.com/webrtc/14175) into the
"inbound-rtp" stats which are not deprecated.
To ease transition for downstream projects, the metrics are temporarily
duplicated in both the old and new locations. In a follow-up CL, they
will be deleted from "track".
Bug: webrtc:14521
Change-Id: I0d9036472607a8c717ec823a458a79a49ddb80c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38308}
Now that SDES is (largely) removed, this is no longer useful.
Bug: chromium:1365484
Change-Id: I3e626a7d5d83130a70958851de3df0aa27616bd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38278}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
This updates VideoReceiveStream2::Stop() to symmetrically tear down
state that's built up in VideoReceiveStream2::Start().
Bug: webrtc:11993, webrtc:14486
Change-Id: I41f4feea5584e5baaeed2143432136f8b9761321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272537
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38244}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
This cl/ changes so that the EncoderStreamFactory is
not created inside WebRtcVideoSendStream (webrtc_video_engine).
The benifit of this is that the VideoStreamEncoder can then
amend the EncoderStreamFactory with state (and types)
w/o exposing it in VideoEncoderConfig.
I.e as an alternative to changes done inside
https://webrtc-review.googlesource.com/c/src/+/276742.
The fake_webrtc_call is modified to (if needed) create
it's own EncoderStreamFactory if needed.
Note: this cl/ will have to be merged with with
https://webrtc-review.googlesource.com/c/src/+/277002.
Bug: webrtc:14451
Change-Id: I3d896b227d39725ba6409622e8d09d14bd45d5fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277160
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38237}
ExpectationToString is used to explain why RTC_DCHECK_RUN_ON is
triggered.
Unfortunately, the current implementation only generates verbose strings
when SequenceCheckerImpl is passed as an argument.
Modify ExpectationToString to generate detailed messages even for
derived classes of SequenceCheckerImpl.
Bug: None
Change-Id: I55f76d44ad59dbe6f21cee7d7d8e19188e0f3088
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276061
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38211}
Polymorphic comparison operators doesn't work in C++20.
(-Wambiguous-reversed-operator)
Fix this issue by using the non-virtual interface pattern.
Bug: chromium:1284275
Change-Id: I79e2bbcd3ae2f3b089183146f7e7c775c493e3f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38210}
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}
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}
This is a reland of commit 6326c9c201
Original change's description:
> Add an active ICE controller that wraps a legacy controller (#7/n)
>
> The wrapping ICE controller will allow existing ICE controller implementations to migrate to the active interface, and eventually deprecate the legacy interface.
>
> Follow-up CL has unit tests for P2PTransportChannel using the new wrapping controller.
>
> Bug: webrtc:14367, webrtc:14131
> Change-Id: I6c517449ff1e503e8268a7ef91afda793723fdeb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275302
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#38130}
Bug: webrtc:14367, webrtc:14131
Change-Id: I5662595db1df8c06b3acac9b39749f236906fa7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276044
Auto-Submit: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38149}
Rename C++ API units to match new proto format units in metric.proto
Bug: b/246095034
Change-Id: Ice5d388a080c474f275ef597f98fac1bcb98cf49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38137}
This reverts commit 6326c9c201.
Reason for revert: breaks upstream project
Original change's description:
> Add an active ICE controller that wraps a legacy controller (#7/n)
>
> The wrapping ICE controller will allow existing ICE controller implementations to migrate to the active interface, and eventually deprecate the legacy interface.
>
> Follow-up CL has unit tests for P2PTransportChannel using the new wrapping controller.
>
> Bug: webrtc:14367, webrtc:14131
> Change-Id: I6c517449ff1e503e8268a7ef91afda793723fdeb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275302
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#38130}
Bug: webrtc:14367, webrtc:14131
Change-Id: I61dd98de62657852068c7566b55f19f662df9ff4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276043
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Sameer Vijaykar <samvi@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38131}
The wrapping ICE controller will allow existing ICE controller implementations to migrate to the active interface, and eventually deprecate the legacy interface.
Follow-up CL has unit tests for P2PTransportChannel using the new wrapping controller.
Bug: webrtc:14367, webrtc:14131
Change-Id: I6c517449ff1e503e8268a7ef91afda793723fdeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275302
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38130}
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
existing field that is related to the abs-capture-timestamp
header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`
Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
Introduce main API for new test performance metrics logging system.
Bug: b/246095034
Change-Id: I9b33740bfe69158c2d7f3f73e18442d1683aa8d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274901
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38116}
Controlled by a field trial, P2PTransportChannel can now use an active ICE controller instead of a legacy ICE controller.
P2PTransportChannel unit tests need non-trivial changes to exercise the refactored code path, so the testing changes are added in a follow-up CL.
Bug: webrtc:14367, webrtc:14131
Change-Id: I00d4930a5692c7d6d331ea9d6c2a2199304e363c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274701
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38114}
P2PTransportChannel can then use either of the ICE controller factories configured with field trials.
Bug: webrtc:14367, webrtc:14131
Change-Id: I09ab99673d6ef81f56abe88987f5b67d84c24cb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271292
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38076}
Plan B having been deleted from Chrome, there is no need to collect UMAs
relating to Plan B vs Unified Plan setups.
Bug: chromium:1357994
Change-Id: Icb5d16823ea9d849798583cd1c82683014b8a15c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275309
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38069}
This is unused at the moment and webrtc::SimulatedNetwork is going
through a refactoring, to keep things simple and well tested this CL
removes CoDel but nothing blocks us from re-implementing it when needed.
No-Try: True
Bug: webrtc:14426
Change-Id: Ie7d40d20a66d3939fc7d3251c47e4f13f3869a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274407
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38032}
apart from the certificate stats which need to update the
reference to the previous certificate stats in the chain.
BUG=None
Change-Id: I27f58084b849fd9afe236e5b57139bedb8eb1811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274175
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38026}
which may allow caching some relatively persistent statistics
such as codec statistics that only change during renegotiation.
BUG=webrtc:8693
Change-Id: Ifd68c9d666d9f328d0efecb64e4201d003788ca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273324
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37981}
Those never existed, were likely a copy-paste error in the spec
that we somehow inherited.
Bug: webrtc:11607
Change-Id: Ib4a038f061123e879f1099656273f6392f092213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273485
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37953}
This reverts commit 021512b76a.
Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
Original change's description:
> rtpsender interface: make pure virtual again
>
> after providing default implementations in Chromium tests
>
> BUG=None
>
> Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37941}
Bug: None
Change-Id: I40f27c36819365fadae32032521f7e11184bee62
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37947}
This reverts commit 83db78e854.
Reason for revert: Some tests in Chromium's blink no longer compile because of the change in the signature of the CreateDefaultTaskQueueFactory() function.
Original change's description:
> Add TaskQueueStdlib experiment.
>
> Bug: webrtc:14389
> Change-Id: I23c6e0ae675748ec35a99c334104dd2654995a33
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265802
> Commit-Queue: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37888}
Bug: webrtc:14389
Change-Id: If3e63d6b4ab9e838dc5020b88076a73fd29916e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272920
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37902}
The input SocketAddress for STUN host lookup is constructed with just
the hostname, so the family is AF_UNSPEC. So added an overload with a
target family to distinguish this from the family of the input addr.
Bug: webrtc:14319, webrtc:14131
Change-Id: Ia5ac5aa2e894e0c4dfb4417e3e8a76a6cec3ea71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270624
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#37750}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Calling InitFieldTrialsFromString modifies a global variable so we must
make sure that state is reset between test runs.
Bug: webrtc:10335, webrtc:14336
Change-Id: Ia9839dd16a330ed3220ed470c28c541fc1cc0678
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271022
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37731}
clangd ignores ASSERT_EXCLUSIVE_LOCK macro attached to an inline function in header, thus IDEs relying on clangd issue false positive warnings about members acceesses without the check of the current sequence.
Attaching assert attribute to an inlined lambda function seems to solve that issue
Bug: None
Change-Id: I6199fee26061aa4223f2e3ea7b7b14bb5820c0bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270480
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37678}
This renames the tests to also capture the expected outcome of the test
along with some minor code cleanups. Some tests have also been added or
extended to tests more invariants.
Bug: None
Change-Id: I0bc733026118eb90646929b164bfc148665556a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267169
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37673}
This CL propose a new API for video dumps in PCLF also removing
differences between p2p and multipeer usage of API.
Bug: b/240540206
Change-Id: Id4d32cc98250500949b3f9e2cf2e86c4bdce7efb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37665}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
The VideoDecoderFactoryTemplate takes decoder implementations as template arguments, making it possible to easily implement a VideoDecoderFactory only using the implementations required for the particular application. This will replace the BuiltinVideoDecoderFactory.
Bug: webrtc:13573
Change-Id: I0213acd20b69dacf06fc6934851b73bd19b1afc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268470
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37523}
Since https://webrtc-review.googlesource.com/c/src/+/267780 supported scalability modes are also used to compare for equality between SdpVideoFormats(?).
Bug: webrtc:14267, webrtc:13573
Change-Id: I2f3c2fca93bac6fadd222f776f672c9bd3f1de0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268304
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37510}
Makes std::string version of rtc::RtcEventLogOutput::Write() no longer pure virtual while making the absl::string_view version pure virtual. Also removes unnecessary overloads in subclasses.
BUG=webrtc:13579
Change-Id: I8fb449560b795a1ef76fab27533d9042d0c34cd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268062
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37484}
This helper suppose to replace ToQueuedTask when calls to TaskQueueBase interfaces are converted to PostTask variants that take absl::AnyInvocable.
Bug: webrtc:14245
Change-Id: I590a6ca068cf5e682ffb34770bd54cf5ce37d826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267706
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37449}
Intended to be used in downstream code when deleting deleting this
attribute.
Bug: webrtc:11607
Change-Id: I39417997a2ec2e72d726da476b5bce88abe267b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267843
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37445}
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.
Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
This CL also removes the existing non-standard implementation of the metric.
Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
The default values are zero, for consistency with the memset of VideoCodec. Except for numberOfTemporalLayers; This cl sets
numberOfTemporalLayers to 1 by default. The intention is to be able to
delete exlpicit setting of .numberOfTemporalLayers = 1 in downstream
code, to ease replacing it with a scalability mode.
Bug: webrtc:11607
Change-Id: I9de442f1893d474ea360f9b33364a00627f6c3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267662
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37430}
Look for first echo (and not only the strongest one) on the same matched
filter.
This change is bit exact with previous version when `pre_echo` is false.
Author: Jesús de Vicente Peña <devicentepena@webrtc.org>
Bug: webrtc:14205
Change-Id: I6782eaa1d690b0df78d00f6d425a85c951b2ca9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266321
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37360}
The tests check that the various scalability mode are supported
and the frames are marked properly by the encoder with their
spatial and temporal index.
The same information is then checked on the receiving side.
A new member is added on EncodedImage to store the temporal index,
and is filled by the encoders and retreived by the ref finder
objects on the decoding side.
Bug: webrtc:11607
Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37303}
This CL removes the last "nogncheck" comment that was related to a
known build cycle. The remaining ones are because of conditional
dependencies.
Bug: webrtc:8733
Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37302}
This is a reland of commit 905c3a6c73
Change from previous attempt is between ps#1 and ps#2: Use PeerConnectionFactoryInterface::Options to clear the `network_ignore_mask`.
Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}
Bug: webrtc:7447, webrtc:14204
Change-Id: Ic78ebec2e88a8c44699015c8c7a44e137f44253a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265982
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37290}
This reverts commit 905c3a6c73.
Reason for revert: New test fails internal tests, with a similar problem as the failed android test: No networks are detected on the test bot.
Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}
Bug: webrtc:7447, webrtc:14204
Change-Id: Ib412d09699a48d8f5db27e2960e365b536ab3aa8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266146
Owners-Override: Niels Moller <nisse@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37273}
Injection via PeerConnectionDependecies was broken, in not accepting
ownership of the injected object.
Bug: webrtc:7447, webrtc:14204
Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37270}
First step of the process to remove the last cycle in the build graph.
Bug: webrtc:8733
Change-Id: I5a0c987ce3d602d1cb30991b73b68a389f13cfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265874
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37261}
422 10 bit format is called I210 in the code and implemented in I210Buffer, and 420 10-bit format format is using is using the already existing I010 format and implemented in I010Buffer.
Bug: webrtc:13826
Change-Id: I6b6ed65b9fbb295386ea20f751bd0badc49ef21b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256964
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37252}
This is what allowed us to remove "transceiver" stats from the spec.
Bug: webrtc:14191
Change-Id: I687a2dd97de016832005cb4271f6e1a0e0560cd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37247}
This should allow standard stats users not to have to rely on the
obsolete "track" stats.
Bug: webrtc:14174
Change-Id: I24e5e1478ee47c73c12fcdecf7314f41fcc76bc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37246}
This is a reland of commit 626f87d905
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: Ib12488fb8510fb3430e92bcd72d88c7879ecb0ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265861
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37226}
This is the first CL out of three to make the low-latency stream signaling
explicit. At the moment this is done by setting the render time to 0.
There's a dependency between Chromium and WebRTC which is why this is
split into three CLs to not break any existing functionality.
Bug: chromium:1327251
Change-Id: Ie6b268746d587a99334485db77181fb2c6e9b567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264502
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37225}
This is cleaner than checking the size before and after, as is currently
done in FrameBufferProxy
Bug: webrtc:14168
Change-Id: Iac896ddf7b1b0b8513159451de7cd8a10668a49a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265663
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37222}
This is a reland of commit 626f87d905
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: Ifdc37e7a48fea516c727c06d2f510780386cb204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37215}
This reverts commit 626f87d905.
Reason for revert: Breaks one downstream project, will re-land after the dependency stops referencing an unimplemented RTT metric
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: I7e9ac60eb474b44fab678d4c08ddcae846ce456c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265800
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37206}
In preparation for the spec moving closer to PR, let's not have
placeholder metrics not implemented.
Bug: webrtc:14167
Change-Id: If4688ef85b57f88154d490186b306b30414874e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37205}
This cl adds a forwarding header, a build target, and migrates headers
in api/ to use it.
Moving actual implementation, will follow, in
https://webrtc-review.googlesource.com/c/src/+/265390.
Bug: webrtc:12701
Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37200}
I've added a basic AV1 impl to Chrome Remote Desktop and am looking into
what is needed to test with I444 (Profile-1) in our platform. This CL
adds a few helper functions, constants, and enums that can be used to
configure the SDP with different AV1 profiles. More work is still needed
but I wanted to get this in place first so I can build on it in the CRD
host code.
Change-Id: I1af9ebf31f833138e8c36e0c0a30e32289e7b58e
Bug: chromium:1329660
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264000
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#37182}
The same information can be found in `AudioFrame.packet_infos_`.
Bug: none
Change-Id: Ib63bc41ffb896677a445d875afce0a98acea6999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37153}
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
Removes all remaining usage of SetType and marks the method as
deprecated.
Bug: none
Change-Id: I98dc613978ffe7ad8a4ffd951dd974d56ed92983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265100
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37137}
This moves the construction of StunMessage instances for
ConnectionRequest, outside of the Prepare() method.
Following this, removing Construct()+Prepare() is relatively
straight forward.
Bug: none
Change-Id: Ibcf0510cef30a6e648005b43602c7ae1fb06729e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264558
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37122}
* Add ctors for providing the type and transaction id at construction.
* Update tests to use them instead of SetType+SetTransactionID
* Make sure stun message enum types are based on uint16_t
* Mark SetTransactionID as deprecated.
* Mark SetStunMagicCookie as deprecated (unused in webrtc).
* Add SetTransactionIdForTest for the one test that uses it (might not
actually need it)
* Make StunRequest::Construct() protected.
* Add a TODO to follow up on this since construction of StunRequest
goes through an unnecessarily complex 3-step process involving
other classes and a virtual method.
Bug: none
Change-Id: Ib013e58f28e7b2b4fcb3b3e1034da31dfc93e9d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264546
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37079}
Currently has the same contents as struct SpatialLayer. Intention is
to add a ScalabilityMode member, which isn't appropriate for a spatial
layer.
Bug: webrtc:11607
Change-Id: I75c9e9b39407e3f24ec117bb17dc37830076b26f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262255
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37062}
This cl/ creates a constructor for a FieldTrials object that is
not backed by the global string. Use with care!
Bug: webrtc:10335
Change-Id: I8c48d1e8bb52fef78524d890cc90473355be617f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264461
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37057}
This currently only exists as a goog legacy stat and has no spec
equivalent according to
https://docs.google.com/document/d/1z-D4SngG36WPiMuRvWeTMN7mWQXrf1XKZwVl3Nf1BIE/edit
Yet it is useful to debug issues sometimes. Exposing it as a
nonstandard stat will make it show up in chrome://webrtc-internals,
removing a need to switch to the legacy stats API there.
BUG=webrtc:14118
Change-Id: I506357ad54ff33df3ba46fb81558aa32187ac8e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37055}
Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.
Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36946}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
Now only using the complexity from the main VideoCodec settings.
Bug: webrtc:13694
Change-Id: I5a29df0fac0c0686bf5ea0b677f8946d23ef9888
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262762
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36912}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
VideoSimulcastConfig::encoding_params and VideoConfig::min/max_encode_bitrate_bps
were replaced by VideoConfig::encoding_params.
All usage of the previous options has been updated to the new option.
Bug: webrtc:11607
Change-Id: I52cd9efa6e640929485da7aa1e61d15a1a693b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261949
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36865}
Similar mocks are used internally, there should only be one in WebRTC.
Bug: None
Change-Id: Ic5163ae0c554c602344a0d25d17c3f0d46fc2e95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261955
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36864}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36742}
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.
Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36731}
media_kind is the old name (that is kept around since we can't deprecate)
BUG=None
Change-Id: I445441a54bb4ff408502d1aba6834cdac874324b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259766
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36625}
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.
Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
Move VideoSubscription::Resolution to the fixture class and rename it to
the VideoResolution. It should be then integrated into other video
related classes.
Bug: b/213863770
Change-Id: Ifd391f840ef8de43bbac66d23df3ecf7258b3943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259523
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36594}
Currently only implemented for codec internal CNG (Opus).
Bug: webrtc:13322
Change-Id: I00622f2967f066dba64a792e26081038ae0cb0d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259200
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36590}
This cl/ adds the feature actually injecting a FieldTrialsView into
PeerConnectionFactory, or into a PeerConnection or both.
The field trials used for a PeerConnection is those specified in
PeerConnectionDependencies. Otherwise will those from
PeerConnectionFactoryDependencies be used (and until we're finished with
this conversion, the global string fallback is used as last resort).
Note that it is currently not possible to create 2 FieldTrials
objects concurrently...due to global string,
so this cl/ is mostly (but entirely) for show, i.e one _can_
realistically inject them into a PeerConnectionFactory.
Bug: webrtc:10335
Change-Id: Id2e60525f48a1f8293c1dd0be771e3ed03790963
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258134
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36578}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
DSCP is controlled by the spec-compliant API
RTCRtpEncodingParameters.networkPriority[1]. It already has a default
value that is the same as when DSCP is disabled.
- If you want non-default DSCP default values, you need to set
networkPriority and shouldn't need to set a non-standard googDscp flag
for it to have an effect.
- If you want the default DSCP value, you wouldn't change
networkPriority and so you don't care if enable_dscp is true... you'll
get the default regardless.
Drive-by: This CL also adds crbug references to other goog flags.
[1] https://w3c.github.io/webrtc-priority/#dom-rtcrtpencodingparameters-networkpriority
Bug: chromium:1315574
Change-Id: I15a0470fa04f55e2534cee0d240eeb03446c2de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36550}
Even if playout audio is only very briefly stereo, the AEC will enter stereo processing mode. To save CPU and improve AEC performance, this CL adds a hysteresis period before treating playout as stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I29116ab2e7823e25a02aa3b66a1c619f1d966d9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258479
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36503}
If playout audio is temporarily stereo, the AEC will currently enter stereo processing mode indefinitely. To save CPU and improve AEC performance, this CL adds support for falling back to mono after a period of no stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I690b5b22f8407f950bf41f3bcaa9ca0138452157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258421
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36502}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
This CL sets speed 9 for all resolutions when two or less cores are
available, as a heuristic for a "slow" machine.
This gives a large speed bost at a relatively small quality loss.
A field-trial kill-switch is available to override this behavior.
Bug: webrtc:13888
Change-Id: I24278a45de000ad7984d0525c47d9eb6b9ab6b60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257421
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36466}
This CL replaces those references with the smallest set of targets
that can satisfy the linker dependencies revealed by building the
"all" target.
Bug: webrtc:13634
Change-Id: Ia778630b18e1164138c41d245c3c8effed67f8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257282
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36445}
The fallback implementation currently returns "...(fallback from
unknown)" since ImplemenationName() is deprecated. Fix this by
using GetDecoderInfo() to determine the implementation name.
Bug: webrtc:12271
Change-Id: Ifa1d97678cd1bf05d9b5a10b73da23c4d54a1e05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257901
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36440}
Create FieldTrials class for setting the field trials from a string
in a program. We can later e.g add a builder class where one
can add key/value pairs.
This class is supposed to replace
webrtc::field_trial::InitFieldTrialsFromString.
No-Try: True
Bug: webrtc:10335
Change-Id: I17f45e401102fddda50ca7c4a04bea2f1cb87788
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256973
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36376}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
The VideoEncoderFactoryTemplate takes encoder implementations as template arguments, making it possible to easily implement a VideoEncoderFactory only using the implementations required for the particular application. This will replace the BuiltinVideoEncoderFactory.
Change-Id: Ifb0e93d0d4491664fb7f7acf085190d8a90ddc0e
Bug: webrtc:13573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251904
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36216}
This patch just refactors creation of P2P transport channel,
pushing down the IceTransportInit object rather than decomposing
it going down.
The IceTransportInit object will in subsequent patches be
extended with a field trial container.
Reason for splitting patch into this and subsequent is
to allow changes to internal factories.
Bug: webrtc:10335
Change-Id: Icc8b6e4142744b64d134bcb2d4a56777745db62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36215}
Add field trials to audio api.
It is added as a pointer with nullptr as default.
It is not (yet) used anywhere.
Usage of field trials comes in subsequent patches.
Bug: webrtc:10335
Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36213}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.
Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.
Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.
Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
Adds new class DecodeSynchronizer that will coalesce the decoding
of received streams on the metronome. This feature is experimental and
is backed by a field trial WebRTC-FrameBuffer3.
This experiment now has 3 arms to it,
"WebRTC-FrameBuffer3/arm:FrameBuffer2/": Default, uses old frame buffer.
"WebRTC-FrameBuffer3/arm:FrameBuffer3/": Uses new frame buffer.
"WebRTC-FrameBuffer3/arm:SyncDecoding/": Uses new frame buffer with
frame scheduled on the metronome.
The SyncDecoding arm will not work until it is wired up in the follow-up
CL.
This change also makes the following modifications,
* Adds FakeMetronome utilities for tests using a metronome.
* Makes FrameDecodeScheduler an interface. The default implementation is
TaskQueueFrameDecodeScheduler.
* FrameDecodeScheduler now has a Stop() method, which must be called
before destruction.
TBR=philipel@webrtc.org
Change-Id: I58a306bb883604b0be3eb2a04b3d07dbdf185c71
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250665
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35988}
This is important for writing tests that affect the DTLS role.
Bug: webrtc:13668
Change-Id: I5d9a93eca7996a8b74cdcfe412f59a85892e1ec1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251389
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35971}
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340
Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
This reverts commit 7b370b935e.
Reason for revert: Breaking WebRTC in Chrome rolls. Roll can be found here https://chromium-review.googlesource.com/c/chromium/src/+/3436384/. Example failed build https://ci.chromium.org/ui/p/chromium/builders/try/chromeos-amd64-generic-rel-compilator/65973/overview. Failures seem to be in ChromeOS with the nearby library:
error: no viable conversion from 'rtc::RefCountedObject<CreateSessionDescriptionObserverImpl> *' to 'rtc::scoped_refptr<CreateSessionDescriptionObserverImpl>'
Original change's description:
> Delete implicit conversion from raw pointer to scoped_ref_ptr
>
> Followup to https://webrtc-review.googlesource.com/c/src/+/242363
>
> Bug: webrtc:13464
> Change-Id: I44358e8cfedeea92aac4ef47c540aff9a4865cdc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247362
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35897}
TBR=mbonadei@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Ib0beb44421519c8393131c55564c62c9b4d91504
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250621
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35905}
This will enable Chrome to inject its metronome for use in WebRTC for
tasks like synchronized decoding.
Bug: webrtc:13560, chromium:1253787
Change-Id: I2488d746f57152a32d3adf92a3cdfdfdb8000c06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35853}
This reverts commit 3babb8af23.
Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.
This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.
Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35684}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
Context: The timer precision of PostDelayedTask() is about to be lowered
to include up to 17 ms leeway. In order not to break use cases that
require high precision timers, PostDelayedHighPrecisionTask() will
continue to have the same precision that PostDelayedTask() has today.
webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which
precision to use when calling PostDelayedTaskWithPrecision().
See go/postdelayedtask-precision-in-webrtc for motivation and a table of
delayed task use cases in WebRTC that are "high" or "low" precision.
Most timers in DCSCTP are believed to only be needing low precision (see
table), but the delayed_ack_timer_ of DataTracker[1] is an example of a
use case that is likely to break if the timer precision is lowered (if
ACK is sent too late, retransmissions may occur). So this is considered
a high precision use case.
This CL makes it possible to specify the precision of dcsctp::Timer.
In a follow-up CL we will update delayed_ack_timer_ to kHigh precision.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340
Bug: webrtc:13604
Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35809}
Just applied a short sed script. See bug description for
the motiviation for this change.
This is the command that was used to generate the changes:
$ find . -type f \( -iname '*.cc' -o -iname '*.h' \) -print0 | \
xargs -0 sed -i -e 's/(const override)/(const, override)/'
Bug: webrtc:13090
Change-Id: Iec7d280f9d55263a972dbb3bd644ebfcd2eb38cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249088
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35801}
See go/postdelayedtask-precision-in-webrtc for context of which use
cases are considered "high" or "low". Most use cases are "low" which
is the default, but this CL allows opting in to "high".
Will be used by FrameBuffer2.
Bug: webrtc:13604
Change-Id: Iebf6eea44779873e78746da749a39e1101b92819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248861
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35776}
After the refactoring, the test fixture is only used for creating the
object under test and dependencies. This leads to more readable code and
allows more flexibility when creating the object under test.
Bug: none
Change-Id: I643330290da17efe0a02fe5dc6b884136705de0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35770}
AsString is constexpr, but RTC_CHECK_NOTREACHED is not. Using some gcc
compile rules, having a constexpr make use of RTC_CHECK_NOTREACHED does
not compile.
See internal issue number 215785261. We could either remove constexpr
or remove the RTC_CHECK_NOTREACHED. This CL does the latter.
Bug: None
Change-Id: I7ea84b345e9abdba60a7620e1d92c3159c0d7974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248167
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35768}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
As per go/postdelayedtask-precision-in-webrtc we want to reduce the
precision of PostDelayedTask() in order to schedule work on the CPU
more efficiently. In order not to break "high precision" use cases, a
new API is added to allow opting in to high precision.
PostDelayedHighPrecisionTask() has the same precision that
PostDelayedTask() has today, but by changing the interface's
requirements on PostDelayedTask(), adding the high precision version
of it will unblock making the old PostDelayedTask() API lower
precision.
This CL does not update implementations to support low precision so
until those are updated, both PostDelayedTask() and
PostDelayedHighPrecisionTask() have the same precision (=high).
This CL also adds TODOs to make some rtc::Thread-specific versions
of PostTask/PostDelayedTask obsolete, see
https://crbug.com/webrtc/13582 for more info.
Bug: webrtc:13583, webrtc:13582
Change-Id: I4c6d53d22bb299c49893ce9f3ef73a40d8c75de1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247367
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35748}
Minor clean up of BUILD file.
Add explicit events for begin and end of log.
Add a helper function to populate timestamps.
Add a GroupKey method that will be used for grouping events by for example SSRC in additon to event type.
Bug: webrtc:11933
Change-Id: Ie3c5f5a5582c89805a0273f4b27978f47ed0fb4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234260
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35725}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency
Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
This can cause issues on Android S if this initialization happens when
the app doesn't have permission to access the microphone.
Bug: b/197461765
Change-Id: Iebccff9d15f5bb12a7b2c78e1c373e379b37a127
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246104
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35689}
Added Nutanix Inc. to the AUTHORS file.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
Bug: chromium:1251096
Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35684}
Update RtpPacketInfos internals to use rtc::make_ref_counted, and a
Data class with no virtual methods.
Bug: webrtc:13464, webrtc:12701
Change-Id: I03f6bee69a9f060dcf287284fc779268d5eb433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244505
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35660}
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.
Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.
If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.
Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
With these changes, we now often have 0 invokes and at most 1 when
calling SetLocalContent on a channel. Before we had at least 1 and
typically 2.
Summary of changes.
* Updating RtpExtension::DeduplicateHeaderExtensions to return a sorted
vector (+test) for easy detection of changes.
* Before updating the transport on the network thread, detect if
actual changes to the demuxer criteria or changes to the rtp header
extensions have been made.
* Consolidate both transport updates to a single call instead of two.
* Added DCHECK guards to catch regressions in number of invokes.
A possible upcoming improvement is to update the transport
asynchronously.
Bug: webrtc:13536
Change-Id: I71ef7b181635a796ffa1e3a02a0f661d28a4870c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35638}
This also removes all internal usage of RemoveTrack, and changes
the replacement function to RemoveTrackOrError rather than RemoveTrackNew.
Bug: webrtc:9534
Change-Id: Idf7bb17495686de77c70428dcbfb12278328ce59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244094
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35624}
Under zero-hertz mode, provided that a frame arrived to the
VideoStreamEncoder, the receiver may experience up to a second
between incoming frames. This results in key frame requests getting
serviced with that delay, which is undesired.
What's worse is also the fact that if no frame ever arrived to the
VideoStreamEncoder, it will not service the keyframe requests at all
until the first frame comes.
This change introduces VideoSourceInterface::RequestRefreshFrame
which results in a refresh frame being sent from complying sources.
The method is used under zero-hertz mode from the VideoStreamEncoder
when frames didn't arrive to it yet (with changes to the zero-hertz
adapter).
With this change, when the frame adapter has received at least one
frame, it will conditionally repeat the last frame in response to the
key frame request.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I6f97813b3a938747357d45e5dda54f759129b44d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242361
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35562}
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
The frame cadence adapter previously resulted in unconditional
frame repeating at max FPS. Change this to slow down to an idle
rate (1 Hz) when quality convergence in all configured spatial
layers has been achieved.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Ifa593dbf8a61aa29da20ac250da332734ae82791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35547}
Fix a use-after-move issue in RTCErrorOr, as found by clang-tidy:
api/rtc_error.h:247:
'error' used after it was moved
Bug: chromium:1122844
Change-Id: I9e826023618067ba37c2567b5e194c46db1dbd23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241200
Auto-Submit: Maksim Ivanov <emaxx@chromium.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35535}
So applications don't need to create and inject their own instance of
BasicPortAllocator, just to change these settings.
Bug: webrtc:13145
Change-Id: I08ac8658b4c0ef87019fa579be9195a8a6b50feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239643
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35476}
This patch adds reporting of relay protocol,
i.e how a client connect to the turn server.
This is added in the old stats api...cause there
are clients still using it.
Bug: none
Change-Id: Iac7fe3e3de0ba42d5897c304ebbae368edf498fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35469}
This method is no longer useful after a previous refactoring, but it was
not removed from the interface.
Bug: webrtc:13444
Change-Id: I9c4761e8503acdec06c16cc37c2a804d4913eac8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239366
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35439}
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.
Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35422}
this variant was deprecated 6 month ago in
https://webrtc-review.googlesource.com/c/src/+/219081
with a trivial replacement.
Bug: None
Change-Id: Ib9cd686280edf36da5f39e8e22b6073530837147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238983
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35421}
This allows to differentiate and test codecs of the same type but
different implementations/settings.
Bug: none
Change-Id: I74f799b36411e63387513133ffc19a7f0c45d550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238165
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35396}
This is a partial revert of commit f9e502d935.
Reason for revert: Functionality turns out to be needed by some partners for some months more.
Original change's description:
> Remove enable_dtls_srtp option
>
> This is part of the removal of support for SDES.
>
> Bug: webrtc:11066
> Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35262}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11066, chromium:1271469
Change-Id: I79a90f025e53816789b391bc52a0e896b9be87e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238170
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35378}
The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values. However, the audio code has limitations that
prevent a high number of channels from working well in practice.
Bug: chromium:1265806
Change-Id: I6f6c3f68a3791bb189a614eece6bd0ed7874f252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237807
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35359}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
