It's not included in the BUILD.gn file and it defines methods that are
not declared in the header. Delete.
Bug: chromium:1381982
Change-Id: I0d8541e7b0e7d1d2b4f3ad7a4864d317d8799399
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328541
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41243}
The change adds dropped frame reporting for previously dropped frame
and also cleans up the colon list of the VSE.
Bug: None
Change-Id: Iad1c084739e5392ded4f100d940b45adf9b561ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327800
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41225}
Also make it more convenient to use uint8_t array view for
interfacing to the class.
Bug: webrtc:15665
Change-Id: Ib671b5add79a48004133a6ecd99429534f7de1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41212}
Add a StartShortCircuiting() callback to allow clients which have
configured Encoded Transforms when creating a PeerConnection to have
all frames skip the transform. This offers a zero cost path for streams
which don't need transforms.
This is preferable to uninstalling/not installing the transform to allow
implementing the behaviour in
https://w3c.github.io/webrtc-encoded-transform/#stream-creation -
giving web apps a chance to configure transforms within a short window
(before the next JS event loop run, so usually sub-millisecond) after stream creation, without any untransformed frames passing.
Usage in Chromium: crrev.com/c/5040731
Bug: chromium:1502781
Change-Id: I803477db1df51e80bdedf6c84d2d3695b088de83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41184}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
This reverts commit 8039cdbe48.
Reason for revert: remove functionality after measurement complete
Original change's description:
> Measure wall clock time of capture and encode processing.
>
> (NOTE: This and dependent CLs will be reverted in a few days after
> data collection from the field is complete.)
>
> This change introduces a new task queue concept, Voucher. They
> are associated with a currently running task tree. Whenever
> tasks are posted, the current voucher is inherited and set as
> current in the new task.
>
> The voucher exists for as long as there are direct and indirect
> tasks running that descend from the task where the voucher was
> created.
>
> Vouchers aggregate application-specific attachments, which perform
> logic unrelated to Voucher progression. This particular change adds
> an attachment that measures time from capture to all encode operations
> complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
>
> An accompanying Chrome change crrev.com/c/4992282 ensures survival of
> vouchers across certain Mojo IPC.
>
> Bug: chromium:1498378
> Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41061}
Bug: chromium:1498378
Change-Id: I9503575fbc52f1946ca26fc3c17b623ea75cd3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327023
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41135}
This is a reland of commit 3ea9fc4cd8
Original change's description:
> Make frame transformer MimeType pure virtual again
>
> after both audio and video have been implemented.
>
> BUG=webrtc:15579
>
> Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tony Herre <herre@google.com>
> Cr-Commit-Position: refs/heads/main@{#41114}
BUG=webrtc:15579
Change-Id: Ia020149cba3045022b539f290565d6c1d0e813ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326880
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41121}
after both audio and video have been implemented.
BUG=webrtc:15579
Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41114}
To replace CreateTimeControllerBasedCallFactory
Update webrtc tests to use this new function
Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
(NOTE: This and dependent CLs will be reverted in a few days after
data collection from the field is complete.)
This change introduces a new task queue concept, Voucher. They
are associated with a currently running task tree. Whenever
tasks are posted, the current voucher is inherited and set as
current in the new task.
The voucher exists for as long as there are direct and indirect
tasks running that descend from the task where the voucher was
created.
Vouchers aggregate application-specific attachments, which perform
logic unrelated to Voucher progression. This particular change adds
an attachment that measures time from capture to all encode operations
complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
An accompanying Chrome change crrev.com/c/4992282 ensures survival of
vouchers across certain Mojo IPC.
Bug: chromium:1498378
Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41061}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
instead of requiring to pass in call_factory and media_engine
webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies
Bug: webrtc:15574
Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41049}
This CL just moves the definition and adds a forward.
Actually using the new definition is left for later CLs.
Bug: webrtc:15622
Change-Id: I6d97ef45b98f9eb193c59dd7f8a89c99cfe0ba9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325381
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41048}
The new reference time contains a monotonically increasing clock time and represents the time when the frame was captured. Not all platforms provide the "true" sample capture time in |reference_time| but might instead use a somewhat delayed (by the time it took to capture the frame) version of it.
Bug: webrtc:15539
Change-Id: I95eff8b0f7bff8d3ae65798bf82046e1ac2b0cf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325261
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#41036}
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
Someone wondered why framesRendered has not been implemented. I had a
look, and discovered that a) we need to implement it, and b) our entire
inter-frame, pause, and freeze metrics are measured at the wrong time
because what WebRTC considered "OnRenderedFrame()" is not actually when
the frame was rendered.
So that we don't forget this again, I filed two crbugs and added TODOs
in the code for future reference to anyone interested in these metrics.
Bug: webrtc:15600, webrtc:15601
Change-Id: Id38df7874df715e9b9c0410efa4a9bc2af5d6232
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324306
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40987}
Goal is to initialize peerconnections in Chromium using this based
field trial config until a proper config that doesn't rely on the
global field trial string can be used (https://crrev.com/c/4936314).
Change-Id: I3d006e2445ccc4880b73b564c8ad4408242d3696
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323621
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40941}
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.
Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp
Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
Structs make api harder to evolve:
deprecated unused properties,
change how data is represented.
Classes with accessors allow more graduated and safer api evolution.
Bug: None
Change-Id: I8ebd5e072d51cf7f5800666cfdac523d0f9a937f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40714}
The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
Bug: b/295129711
Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40709}
since the spec and implementation took a different route
BUG=chromium:1354101
Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40665}
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame
We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.
Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
Remove code where integer -1 as delay is used to represent unset value.
Bug: webrtc:13756
Change-Id: I16a01e12c25a09ce21a971c9edabf47af5936662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316923
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40592}
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.
Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta
Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.
Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
which is present if a fec mechanism like FlexFEC is negotiated
spec change:
https://github.com/w3c/webrtc-stats/pull/765
BUG=webrtc:15250
Change-Id: I7d71d49fab0153d734f22831e6684d2acfc647fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314981
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@{#40514}
This patch adds a StunDictionary.
The dictionary has a reader and a writer.
A writer can update a reader by creating a delta.
The delta is applied by the reader, and the ack is applied by the
writer.
Using this mechanism, two ice agents can (in the future) communicate
properties w/o manually needing to add new code.
The delta and delta-ack attributes has been allocated at IANA.
Bug: webrtc:15392
Change-Id: Icdbaf157004258b26ffa0c1f922e083b1ed23899
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40513}
This is part of the long term plan to stop using pointer + length
to pass around buffers.
Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
since this may contain sensitive data, just like the address.
BUG=None
Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40473}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
The existing equality check did not always work since content_type
is sometimes overloaded with extra internal information such as simulcast layer index. Fix by using the videocontenttypehelpers::IsScreenshare helper method.
Bug: webrtc:15381
Change-Id: I2fe84e7f036ea2c223e4fa6dd58af1c4c0bcfbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40448}
This CL cleans up all local conversions, in favor of the common helper
function.
Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
This avoids the risk of a higher level owner approving something getting
added to the getStats() API without +1s from stats owners.
Bug: None
Change-Id: Iedd7133d0e943d1db6977dc0e1d406e5b545e31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311543
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40412}
Move the SetRTPTimestamp method from TransformableAudioFrameInterface
to the base class, so that RTPTimestamps can also be modified on encoded
video frames.
Bug: webrtc:14709
Change-Id: I355be527c2be201c9201e04c431394c962237140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310781
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40378}
Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API.
Once uses are removed from Chromium, I'll mark GetHeader() deprecated
and eventually remove it.
Bug: chromium:1456628
Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40344}
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.
BUG=webrtc:15250
Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
Scoped enums do not get automatically promoted to their underlying type,
so these uses have undefined behavior and Clang recently started warning
about it.
Bug: chromium:1456289
Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40321}
Introduces a field trial
WebRTC-IncreaseIceCandidatePriorityHostSrflx
that adjusts the priority of non-relay candidates such that the STUN priority attribute calculated as
(prflx-type-preference << 24) | (priority & 0x00FFFFFF)
as described in
https://www.rfc-editor.org/rfc/rfc5245#section-7.1.2.1
will satisfy the condition that the STUN priority of server-reflexive candidates will always be higher than the STUN priority of relay candidates.
Previously this was not the case because the TURN relay preference was added to the local_preference for relay candidates, making it higher than the local_preference of srflx candidates gathered from the same interface.
This led to cases where the resulting candidate pair priority of a srflx-relay pair was higher than the candidate pair priority of a srflx-srflx pair, i.e. using a TURN server in cases that work using a direct P2P connection.
Whether the field trial is active can be observed by checking that
priority-of-srflx-candidate&0x00FFFFFF
is greater than
priority-of-relay-candidate&0x00FFFFFF
BUG=webrtc:13195,webrtc:5813,webrtc:15020
Change-Id: Ib91708fbe7310b6454f93158a45c9d77da009091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292700
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40311}
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.
Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.
Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
This is no longer needed after downstream redefinitions are deleted.
Bug: webrtc:15241
Change-Id: Iea6839bff781fe7d0c56b4739f3d43398c70f2b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40306}
This is a reland of commit ccc87ea3c6
Downstream project has been updated.
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
Auto-Submit: 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@{#40305}
This reverts commit ccc87ea3c6.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
Also add a preprocessor definition to avoid redefinition in downstream projects.
Bug: webrtc:15241
Change-Id: Ic55d98c3d3a69b9b19195ee78f03af6e38fdd0e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40289}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
This can be done now as the function SetRTPTimestamp is now overriden
in blink MockTransformableAudioFrame.
Change-Id: I4fa4cb81d0282fea864818f0f2d9a5ed881a5d30
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40257}
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.
Want lgtm: asapersson
Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
This change will make it possible to let us modify timestamp in
RTCEncodedAudioFrame.
Change-Id: I97e9571c258fd718d6c211014f1476ca46c78097
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307501
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40238}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Currently FecController knows about network conditions, these information can be used to control RTX settings in-call.
Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada
Bug: webrtc:15167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40192}
Add an API to pass AudioFrameProcessor as a unique_ptr.
Bug: webrtc:15111
Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40187}
Whether a metric is to be exposed to JavaScript or not is a blink
implementation detail that the WebRTC repository does not need to be
concerned with.
This CL removes unused code and paves the way for the possibility of
making the one and only RTCStatsMember class be absl::optional<>-based
in the future.
Bug: webrtc:15162
Change-Id: I578715f48b8fcc3534b72b4c700fd6567f8d553e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40139}
NonStandardGroupId is no longer wired up to Chrome, but if we did want
to only expose certain metrics if a field trial was enabled then the
right place to do that would be in blink, where WebIDL lives.
This was only used prior to the WebRtcStatsReportIdl launch and
experiments haven't been active in several years so its dead code.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4514754
Bug: webrtc:15162
Change-Id: Ia41a4d21d7e5f029ddb121183fbd69ae7f98fac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304720
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40132}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
Change default value of is_active to false,
this means that VideoRenderer or other VideoSinks
added with default rtc::VideoSinkWants() does not
block usage of RequestedResolution, e.g JNI_VideoTrack_AddSink.
This problem occurs when attaching a VideoRenderer directly to
the sending VideoTrack (which is a great solution!). But the
VideoRenderer is "passive" and should not block adaptations
from RequestedResolution.
Bug: webrtc:14451
Change-Id: I2ab02596245c7b82bf94fe86f8788f458c7ea286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305024
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@{#40105}
Step 1: Make reading RTCStatsMember look the same as reading
absl::optional (this CL).
Step 2: Migrate uses of "is_defined()" to "has_value()".
Step 3: Delete "is_defined()".
Step 4: Make RTCStatsMember+Interface an implementation detail of
RTCStats::Members(), only used for abstract iteration ("for
each metric"). Lazy instantiate it upon Members().
Step 5: Replace RTCStatsMember with absl::optional for use in RTCStats
dictionaries (rtcstats_objects.h/cc).
Bug: webrtc:15164
Change-Id: I5a2c9fe56707e3c7d89e8ea62fb37171ae806a7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40048}
The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.
BUG=webrtc:9674
Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}
As pointed out in issue webrtc:15146 this Mac/iOS specific variable,
makes the SequenceChecker behave incorrectly on those platforms.
The variable was introduced in a CL that merged the previous checker
classes, ThreadChecker and SequencedTaskChecker, but curiously neither
one of them had such a variable. So I'm not exactly sure what problem
was being solved. Hence I'm wondering if we actually need it.
Reference: https://webrtc-review.googlesource.com/c/src/+/129721
Bug: webrtc:15146
Change-Id: Ia7a9eb17b993c4f8a1e8204c658bf0b3dbdaa1e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304401
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40019}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
* retransmittedBytesReceived
* retransmittedPacketsReceived
added to the specification in
https://github.com/w3c/webrtc-stats/pull/735
BUG=webrtc:15096
Change-Id: I6770e5d8d09ac1c2693c918fd943b0ab257ec7ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295260
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39959}
which gets assigned from a uint32_t VideoReceiverInfo::frames_received so should remain an unsigned type
BUG=None
Change-Id: I1db6a3f96c4ff49eee72dcce54eb6fff346c128c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302342
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39958}
Can be used to calculate the average delayed packet outage duration and
number of packet loss events by subtracting from concealment events.
Only used in simulations currently.
Bug: None
Change-Id: I03740a2bcb781af09e28a4d13d9e41c0f84bc506
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303600
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39957}
This creates the RtpCodec structure for the common fields
used in RtpCodecParameters and RtpCodecCapability.
Remove the unused fields from both that were defined from ORTC
and never implemented as well.
Bug: webrtc:15064
Change-Id: I37b4c83e2051a888fc99cc0d9f7aeb8d74f0421d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301182
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39862}
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.
This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.
Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
One problem with the existing Send() method is that it has a return
value that is problematic for a fully async implementation.
A second problem with Send() is that the return value is bool and not
RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError.
Also, start deprecating `bool Send()` in favor of `void SendAsync()` and
adding `network_safety_` flag for posting async operations to the
network thread. This flag also takes over from the
`connected_to_transport_` which can now be removed.
Bug: webrtc:11547, webrtc:13289
Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39817}
This is a partial reland of:
https://webrtc-review.googlesource.com/c/src/+/299142
This CL includes the interface change in DataChannelObserver but
not the code behind it. The point of landing this change first is
to be able to override this method in downstream implementations in
preparation for relanding the rest of the changes.
Bug: webrtc:11547
Change-Id: Ic3fe4fb8084908ef12bd4916b763df5a75604113
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300362
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39776}
Many STL containers define these type aliases, and they are easier to
work with than add_const_t<add_lvalue_reference_t<value_type>>.
In a followup, `WTF::Vector` in Blink's conversion constructor from
other containers will be SFINAE-guarded using these type aliases.
Bug: chromium:1408442
Change-Id: I7790e6f462a32e7e49bc6468afeda6b2e6d4b631
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300180
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Daniel Cheng <dcheng@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39771}
This reverts commit fe53fec24e.
Reason for revert: Speculative revert, may be breaking downstream project
Original change's description:
> [DataChannel] Send and receive packets on the network thread.
>
> This updates sctp channels, including work that happens between the
> data channel controller and the transport, to run on the network
> thread. Previously all network traffic related to data channels was
> routed through the signaling thread before going to either the network
> thread or the caller's thread (e.g. js thread in chrome). Now the
> calls can go straight from the network thread to the JS thread with
> enabling a special flag on the observer (see below) and similarly
> calls to send data, involve 2 threads instead of 3.
>
> * Custom data channel observer adapter implementation that
> maintains compatibility with existing observer implementations in
> that notifications are delivered on the signaling thread.
> The adapter can be explicitly disabled for implementations that
> want to optimize the callback path and promise to not block the
> network thread.
> * Remove the signaling thread copy of data channels in the controller.
> * Remove several PostTask operations that were needed to keep things
> in sync (but the need has gone away).
> * Update tests for the controller to consistently call
> TeardownDataChannelTransport_n to match with production.
> * Update stats collectors (current and legacy) to fetch the data
> channel stats on the network thread where they're maintained.
> * Remove the AsyncChannelCloseTeardown test since the async teardown
> step has gone away.
> * Remove `sid_s` in the channel code since we only need the network
> state now.
> * For the custom observer support (with and without data adapter) and
> maintain compatibility with existing implementations, added a new
> proxy macro that allows an implementation to selectively provide
> its own implementation without being proxied. This is used for
> registering/unregistering a data channel observer.
> * Update the data channel proxy to map most methods to the network
> thread, avoiding the interim jump to the signaling thread.
> * Update a plethora of thread checkers from signaling to network.
>
> Bug: webrtc:11547
> Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39760}
Bug: webrtc:11547
Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39764}
This updates sctp channels, including work that happens between the
data channel controller and the transport, to run on the network
thread. Previously all network traffic related to data channels was
routed through the signaling thread before going to either the network
thread or the caller's thread (e.g. js thread in chrome). Now the
calls can go straight from the network thread to the JS thread with
enabling a special flag on the observer (see below) and similarly
calls to send data, involve 2 threads instead of 3.
* Custom data channel observer adapter implementation that
maintains compatibility with existing observer implementations in
that notifications are delivered on the signaling thread.
The adapter can be explicitly disabled for implementations that
want to optimize the callback path and promise to not block the
network thread.
* Remove the signaling thread copy of data channels in the controller.
* Remove several PostTask operations that were needed to keep things
in sync (but the need has gone away).
* Update tests for the controller to consistently call
TeardownDataChannelTransport_n to match with production.
* Update stats collectors (current and legacy) to fetch the data
channel stats on the network thread where they're maintained.
* Remove the AsyncChannelCloseTeardown test since the async teardown
step has gone away.
* Remove `sid_s` in the channel code since we only need the network
state now.
* For the custom observer support (with and without data adapter) and
maintain compatibility with existing implementations, added a new
proxy macro that allows an implementation to selectively provide
its own implementation without being proxied. This is used for
registering/unregistering a data channel observer.
* Update the data channel proxy to map most methods to the network
thread, avoiding the interim jump to the signaling thread.
* Update a plethora of thread checkers from signaling to network.
Bug: webrtc:11547
Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39760}
The Mode is currently redundant with the optional input_file_name.
Change-Id: Ib4f0a363e86d925107d61867a7f743d6663e7071
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298743
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39754}
This is needed in order to be able to update the legacy stats
collector to fetch data channel stats from the network thread, which
is part of an upcoming change to data channels.
Bug: webrtc:11547
Change-Id: Ic205b0314b9f11a024d36d714c223cbddd0f3df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299462
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39732}
This is to be more robust to packet loss during DTX and paused streams.
Without it, we can wait to decode an available packet when in CNG or
PLC mode until more packets arrive, which for DTX and paused streams
can take a long time.
We already include the waiting time if the last packet in the buffer
is a DTX packet.
Bug: webrtc:13322
Change-Id: Iaf5b3894500140d6f83377ba2cd65b44e0cdac05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299009
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39667}
This CL is partly a test to see if there's an impact on binary size:
- Not a big difference for binaries (decrease): -776b to -4Kb
- For libraries (libwebrtc.a) it actually increases the size: +40Kb
Secondarily this CL is basically to introduce this pattern to the
code base. In terms of LOC, this makes things slightly more compact.
From:
class Foo {
public:
Foo() {
checker_.Detach();
}
private:
SequenceChecker checker_;
};
To:
class Foo {
public:
Foo() = default;
private:
SequenceChecker checker_{SequenceChecker::kDetached};
};
Bug: none
Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39664}
Various "if streams == 1" cases are updated to "if
IsSinglecastOrAllNonFirstLayersInactive()" in order not to cause subtle
differences between VP9 {active} and VP9 {active,inactive,inactive}.
This CL also affects a line that conditionally sets
`simulcastStream[0].active = codec_active` so it seemed fitting to
improve the test coverage of "if all streams are inactive, don't send".
Bug: webrtc:15028
Change-Id: I8872dc8be0f2dfc1d8914bdba5e6433f9ba8cbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298881
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39656}
This is to allow external tests to depend on it.
Bug: none
Change-Id: Ic8e2f864041d959f673e7f2c18eb563a13274dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298745
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39646}
Based on previous discussions I would have thought that this test would
fail, but it turns out that it passes. See referenced bug for context.
Bug: webrtc:15021
Change-Id: I845b48f688fb25942e3b770d50cafbf8a0bafe94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298562
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39625}
Also make it possible to pause an already paused stream by making it a no-op.
Change-Id: Id10f74a4c6464067ae63208162194f020c6470eb
Bug: b/271542055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298202
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39620}
The motivation for this is to not have to implement this pattern:
foo.h:
class Foo {
public:
Foo();
private:
SequenceChecker checker_;
};
foo.cc:
Foo::Foo() {
checker_.Detach();
}
And instead be able to do this inline in the .h file:
class Foo {
public:
Foo();
private:
SequenceChecker checker_{SequenceChecker::kDetached};
};
Bug: none
Change-Id: Idd7ca82d15c2f77f3aaccf26f1943a49f4b40661
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298445
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39616}
Initialization of Android HW codecs takes hundreds milliseconds. Exclude this time from frame processing time of first frame by initializing codecs before starting encoding/decoding.
Bug: b/261160916, webrtc:14852
Change-Id: I9ec84c6b12c1d9821b59965cf521170224066563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39613}
It seems like this is legacy and not useful. A comment mentions
transitioning between CNG and DTMF modes, but there is no way this can
happen currently.
Bug: webrtc:13322
Change-Id: I9e4706cb6ee145ee37a9e11e7cab6ea4ff697dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39590}
Because the adapter has a passthrough mode, it can already handle both
singlecast and simulcast cases, meaning the proxy is no longer providing
value. Let's delete.
Bug: webrtc:15001
Change-Id: I480eaba599448e9b82b8cf7f829dc35ad6ce0434
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39579}
The goal of the VP9 simulcast project is that when `scalability_mode`
is set, multiple encodings are always interpreted as simulcast, even
if VP9 or AV1 is used. This CL makes this so, but only if the flag
"WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us
to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast.
When we are ready to ship we will remove the need to use the field
trial, but before we ship this we'll want to revisit if
SvcRateAllocator can be updated to support simulcast. (Today if we use
SvcRateAllocator when VP9 simulcast is used, all encodings except the
first one get bitrate=0, causing the test to fail because media is not
flowing on all layers.) For now, a TODO is added.
Bug: webrtc:14884
Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39552}
payload type frequency is not communicated inside an RTP packet and
thus do not belong to the RTPHeader
Bug: None
Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39515}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39497}
following spec updates from
https://github.com/w3c/webrtc-extensions/pull/142
BUG=chromium:1051821
Change-Id: I1fd991a5024d38ac59ebe510ea1a48fd6f42d23b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39491}
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread.
This CL reaps 90% overhead savings in sent packet feedback
as measured with Perfetto under a 49p Meet call.
The identity of the posted calls was uncovered with WebRTC/Chrome's
new location-aware tracing.
TESTED=presubmit + local Meet calls.
Bug: chromium:1373439
Change-Id: I0c43aa4de884831838747d52b21c45fd360106e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295780
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39484}
making RTCOutboundRtpStreamStats inherit from RTCSentRtpStreamStats
as defined in
https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict*
This removes the duplicated definitions of packetsSent and bytesSent.
BUG=webrtc:14948
Change-Id: I184998b65d59dbd0d1288733d55d8a884e6de970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39481}
This will clone an encoded audio frame into a sender frame.
Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
Chromium uses have been migrated to Metadata(), so we should be clear.
Other projects can easily migrate similarly.
Bug: chromium:1420245
Change-Id: I150654812676dabd5c957cff00d40d4c95eaf5d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295481
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39455}
which is only used in tests.
BUG=None
Change-Id: If215ad84e6756af2ee90777a27376400f8f4d8e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39450}
Several files refer to symbols declared in headers not explicitly
included. This compiles now because libc++ tranitively includes these
headers via other libc++ headers; however, these transitive includes are
not guaranteed to exist and in Chrome, will no longer exist once libc++
is compiled with modules.
Bug: chromium:543704
Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762
Auto-Submit: Alan Zhao <ayzhao@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39448}
This CL completes migration to the new TaskQueueBase interface
permitting location tracing in Chrome.
Bug: chromium:1416199
Change-Id: Iff7ff5796752a1520384a3db0135a1d4b9438988
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39439}
This CL migrates unit tests to the new TaskQueueBase interface.
Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
It was marked deprecated on Feb 9th, ~3 weeks ago.
Bug: chromium:1414370
Change-Id: I251b91984ca9a958e221f6eaf01c63b05c5a7a48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295506
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39422}
This reverts commit 8bf3210629.
Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())
Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}
Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
Until now you only had the option to RTC_DCHECK for unregistered field
trials. This makes it possible to log a warning instead.
Bug: webrtc:14154
Change-Id: I8628054e3c9b5d690f241a93e61299126b732ed0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295300
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@{#39417}
This reverts commit 437bf78ed9.
Reason for revert: Breaks upstream project
Original change's description:
> operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
>
> Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
>
> Also default-initialized VideoFrameMetadata::ssrc_ to 0.
>
> Bug: webrtc:14708
> Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> Commit-Queue: Tove Petersson <tovep@google.com>
> Reviewed-by: Tony Herre <herre@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39411}
Bug: webrtc:14708
Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39413}
Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
Also default-initialized VideoFrameMetadata::ssrc_ to 0.
Bug: webrtc:14708
Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39411}
Add a method to TransformableVideoFrameInterface which returns a new
instance of VideoFrameMetadata which the caller can move and use as
they like.
This will replace the existing GetMetadata which returns a dangerous const ref to a field which might change if someone calls SetMetadata
etc. That method will be deprecated as soon as we've migrated Chromium
usages.
Bug: webrtc:14708
Change-Id: Id7c15f33d6ec28c4a975ce250cdc791d7a3087bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39403}
This prepares TaskQueueBase sub classes to be able to migrate to
the location and traits-based API. It re-introduces a Location class
into the webrtc namespace, which is meant to be overridden by Chromium.
Bug: chromium:1416199
Change-Id: I712c7806a71b3b99b2a2bf95e555b357c21c15ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39400}
This CL propagates capture_time_identifier introduced in
webrtc::VideoFrame and propagates it to EncodedImage. For use cases
involving EncodedTransforms, this identifier is further propagated to
TransformableVideoSenderFrame.
VideoEncoder::Encode function is overriden by each encoder. Each of
these overriden functions needs to be changed so that they can handle
this new identifier and propagate its value in the created EncodedImage.
Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39374}
This uses the field trial introduced is crbug.com/1406331 and
extends the usage to OpenH264. This simplifies experimentation
whether this change improves performance without requiring
multi-slice encoding.
BUG=webrtc:14368
Change-Id: I0031e59059f7113dd5453234869c957d46f311bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39371}
Replace the existing variable capture_time_identifier_ms_ with
capture_time_identifier_ in webrtc::VideoFrame and
webrtc::VideoFrame::Builder. This variable uses webrtc::Timestamp as its
type versus using int64_t which creates confusion about whether it is
being recorded in milliseconds/microseconds.
Change-Id: I0b83a6235fb1d5732f7afe2c14d7d6121f1e985b
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39369}
in favor of the variant with (or returning) a Timestamp object.
BUG=webrtc:14813,webrtc:13756
Change-Id: I7b40f48f640a8be40a134b380a7a1b99cc99913b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294287
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39366}
This is a reland of commit 8ad4924936
See diff between latest Patch Set and PS1. Fixes include:
- VideoStreamEncoder's call to bitrate_adjuster_->OnEncodedFrame()
is updated to take stream index (spatial or simulcast index) instead
of only looking at SpatialIndex().
- Migrate test-only helpers to use Spatial/SimulcastIndex correctly.
The fixes are to migrate
some test-only helpers that we had forgot to fix that are used by
external tests.
Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}
Bug: webrtc:14884
Change-Id: Ib966924efca1a040dae881599f0789a7f2ab24a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39358}
As requested by a CEF hosted application (https://crbug.com/1406331)
who want to be able to limit the number of threads in a controlled
environment, this CL adds a flag to control the max limit per encoder.
For plumbing-reasons, this is placed in VideoEncoder::Settings but
with a note that this is considered an experimental API with limited
support. For now only LibvpxVp8Encoder uses it and there are no plans
to roll this out.
I have manually confirmed this is working with printf debugging,
--force-fieldtrials=WebRTC-VideoEncoderSettings/encoder_thread_limit:2
and https://jsfiddle.net/henbos/2bd6m7Lt/
Bug: chromium:1406331
Change-Id: Ib02bd83e2071034874843d3aaa0d3b0adc5bbf46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293960
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39349}
This reverts commit 8ad4924936.
Reason for revert: Breaks downstream projects
Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}
Bug: webrtc:14884
Change-Id: Ibcb834a1519930336fa50e8e9d8d0137972e28e6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294282
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39347}
This CL removes the fallback logic to return the other index when the
one requested has not been set. This means we can remove the codec gates
that was previously needed because SpatialIndex() had multiple meanings,
resolving the TODOs previously added in
https://webrtc-review.googlesource.com/c/src/+/293343.
We have already migrated all known external dependencies from
SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
Bug: webrtc:14884
Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39343}
Allow mocking of GetDirection in tests.
Required for Chromium adoption of this mock:
https://chromium-review.googlesource.com/c/chromium/src/+/4236916
Bug: chromium:1414370
Change-Id: I2e7443a1bf24966cfcfaeadf47c5b29375e84f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293745
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39329}
In Meta we have our own audio encoder/decoder factories and we would like to exercise those in the peer connection e2e test framework.
Also, looks cleaner to have the APIs for both video and audio :)
Bug: webrtc:14910
Change-Id: Ibd1e0f39fc809882ef17b3de3154fdf4b567013b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293782
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39328}
As part of go/unblocking-vp9-simulcast (Step 1), EncodedImage is being
upgraded to be able to differentiate between what is a simulcast index
and what is a spatial index.
In order not to break existing code assuming that "if codec != VP9,
SpatialIndex() is the simulcast index", SimulcastIndex() has fallback
logic to return the value of spatial_index_ in the event that
SetSimulcastIndex() has not been called. This allows migrating external
code from (Set)SpatialIndex() to (Set)SimulcastIndex(). During this
intermediate time, codec gates are still necessary in some places of
the code, see TODOs added.
In a follow-up CL, after having fixed dependencies, we'll be able to
remove the fallback logic and rely on SimulcastIndex() and
SpatialIndex() actually being the advertised index and "if codec..."
hacks will be a thing of the past!
Bug: webrtc:14884
Change-Id: I70095c091d0ce2336640451150888a3c3841df80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293343
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39318}
Also move the frame_transformer_factory_unittest build target into the
if(rtc_include_tests) block, so it's not compiled without the mock.
Bug: chromium:1414370
Change-Id: I12653b173b419ec20bfad904e24a4d965e7e7830
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292863
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39288}
During code review, kFullFramerate was renamed kMaxFramerateFraction but
the uses of kFullFramerate in a comment were not updated. See
https://webrtc-review.googlesource.com/c/src/+/117900/3..4
Change-Id: I6b3c06b4c5b302e8ba40bde4ba722b94aab191eb
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292801
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39285}
This will be used by third_party/blink/renderer/platform/peerconnection/webrtc_video_track_source.cc to provide capture_time_identifier_ms_ from
media::VideoFrame.
This identifier would then be passed to webrtc::EncodedFrame and
webrtc::TransformableVideoSenderFrame (in the future CLs) to be used as
an identifier for encoded frames.
Bug: webrtc:14878
Change-Id: I1d8a27891323d86fdc2f014988a8da572df84119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39270}
This allows callers to modify an encoded video frame's SSRC via the
setMetadata() call, which we'd like to do from Chrome, to allow using
an encoded frame from one PC on a different one.
Bug: webrtc:14709
Change-Id: Ia6b33761a3f63038f6eabbcd848916877e24454b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292380
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39266}
Passing of ownership of codecs to tester is not strictly needed. We may need to continue using a codec after test. For example, to check codec state or to use the same codec instance in next test.
Bug: b/261160916, webrtc:14852
Change-Id: I179b262116d7de76b8171f0409f943ad6d87433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291802
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39256}
Step 1 of combining the sender and receiver types
Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.
Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters.
VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl.
Bug: b/261160916, webrtc:14852
Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39248}
