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}
This reverts commit 3afb8e2431.
Reason for revert: Causes some unexpected perf regressions.
Original change's description:
> When VP9 SVC is used, use SvcConfig to set max bitrate for the stream.
>
> Currently, a default max bitrate is determined within WebRtcVideoEngine,
> which maxes out at 2.5Mbps - and that limits the max bitrate deteremined
> by SvcConfig for resolutions above 720p.
>
> This does not affect simulcast, as WebRtcVideoEngine already knows to
> trust the rate allocation in simulcast.cc instead.
>
> Bug: webrtc:14017
> Change-Id: I0c310a6fd496e9e5a10eae45838900068aa1ae2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267160
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37370}
Bug: webrtc:14017
Change-Id: I1e45ee3f78deb50a9057d648146b1a6360782aa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267800
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37438}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
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}
We should have done this a long time ago.
Let's do the same for stats_types.h in a separate CL because that file
is part of the api/ folder and needs some special care (typedefs and
temporarily include helper to avoid breaking downstream projects).
Bug: webrtc:14180
Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37426}
In the modern getStats implementation, we currently do two
block-invokes when we trigger stats collection, once for
signaling -> worker and once for signaling -> network inside, both take
place inside the "prepare" method:
RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n.
For comparison, the legacy stats collector currently require 4 block
invokes to operate.
Bug: webrtc:14247
Change-Id: Ie739cbcf29d87041484183b520aeba520aafcaba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37424}
This trial has been unused for some time, time to clean it up.
Bug: webrtc:10144
Change-Id: I2b1bd9ff0335efdc07f47a361878915f1be383a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267410
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37421}
The send queue is responsible for generating lifecycle events for all
messages that are still in the queue. Because, if they are still in the
queue, that means that the last fragment of the message hasn't been sent
yet (because then it would have been in the retransmission queue
instead). And if the last fragment hasn't been sent, the send queue is
responsible for generating the
`OnLifecycleMessageExpired(/*maybe_sent=*/false)` event.
Bug: webrtc:5696
Change-Id: Icd5956d6aa0f392cae54f2a05bd20728d9f7f0a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264144
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37419}
RID is defined for multiple usages in RFC 8851, but we only support
usage with a=simulcast as specified in RFC 8853.
Bug: chromium:1341043
Change-Id: Ie72074c5b394bdc41865938a86ec9c7629e1f5e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267628
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37417}
According to Xcode 14 documentation [1]:
> Xcode no longer builds bitcode by default and generates a warning
> message if a project explicitly enables bitcode: “Building with
> bitcode is deprecated. Please update your project and/or target
> settings to disable bitcode.” The capability to build with bitcode
> will be removed in a future Xcode release. IPAs that contain bitcode
> will have the bitcode stripped before being submitted to the App
> Store. Debug symbols for past bitcode submissions remain available
> for download. (86118779)
[1]: https://developer.apple.com/documentation/Xcode-Release-Notes/xcode-14-release-notes
Bug: webrtc:14237
Change-Id: I39fb618409e1978f8e7b42aa71208e00ed69d85f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267407
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37415}
Instead of creating a TaskQueue from packet_sender, create a rtc::Thread
in test_controller so that test_controller instantiates a SocketServer,
eliminating the use of rtc::Thread::socketserver().
Also did various cleanups, such as adding threading annotations, and
ensuring that all network operations are done in dedicated threads.
Bug: webrtc:13145
Test: Unittest, and manually verified using Android clients and Linux servers
Change-Id: I05ebe5e29bd80f14a193c9ee8b0bf63a1b6b94d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263321
Commit-Queue: Daniel.l Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37411}
This reverts commit 93bb305149.
Reason for revert: It breaks a test while rolling into Chromium,
see https://webrtc-review.googlesource.com/c/src/+/261780/21#message-4a96e33bfb475f19a618be82bbe72951b23085ef for details.
Original change's description:
> Wait for frames to arrive in WgcCapturer instead of returning nothing.
>
> We're seeing a high instance of "first capture failed" in Chromium when
> using WGC. We can reduce this by waiting for frames to arrive if there
> are none in the frame pool instead of returning a temporary error.
>
> I've set the maximum time to wait for a frame to 50ms. If no frame
> arrives before 50ms has elapsed, we will return a temporary error.
> Added a new test, FirstCaptureSucceeds, to verify that this is working
> as expected.
>
> As part of this I updated the name of the `kCreateFreeThreadedFailed`
> enum value to `kCreateFramePoolFailed`. The value remains the same
> since they both report failures in frame pool creation.
>
> I also increased `kNumBuffers` from 1 to 2, so that the frame pool can
> store two frames. This should prevent us from having to wait on the
> event as frequently. This will increase the latency between capture
> and display, however. High frame rate applications should not be
> noticeably affected.
>
> Additionally, we uncovered a bug in the OS that prevents window capture
> when there are displays attached, but none of them are active. Added
> a new check to `IsWgcSupported` to cover this scenario.
>
> Finally, some issues with other WGC tests blocked moving the TryBots
> to a newer version of Windows. This CL fixes those issues and updates
> the TryBot configuration.
>
> bug: chromium:1314868
> Change-Id: Id9c4d5ee98621e682ef04864c3848d50e761cdb7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261780
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Austin Orion <auorion@microsoft.com>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#37404}
Change-Id: If237df4826fe20b6fe2ca4b57253623321bf33c5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267460
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37408}
We're seeing a high instance of "first capture failed" in Chromium when
using WGC. We can reduce this by waiting for frames to arrive if there
are none in the frame pool instead of returning a temporary error.
I've set the maximum time to wait for a frame to 50ms. If no frame
arrives before 50ms has elapsed, we will return a temporary error.
Added a new test, FirstCaptureSucceeds, to verify that this is working
as expected.
As part of this I updated the name of the `kCreateFreeThreadedFailed`
enum value to `kCreateFramePoolFailed`. The value remains the same
since they both report failures in frame pool creation.
I also increased `kNumBuffers` from 1 to 2, so that the frame pool can
store two frames. This should prevent us from having to wait on the
event as frequently. This will increase the latency between capture
and display, however. High frame rate applications should not be
noticeably affected.
Additionally, we uncovered a bug in the OS that prevents window capture
when there are displays attached, but none of them are active. Added
a new check to `IsWgcSupported` to cover this scenario.
Finally, some issues with other WGC tests blocked moving the TryBots
to a newer version of Windows. This CL fixes those issues and updates
the TryBot configuration.
bug: chromium:1314868
Change-Id: Id9c4d5ee98621e682ef04864c3848d50e761cdb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261780
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#37404}
Minor refactoring of the API, to put optional arguments last. Also
changed internal structures to reflect that order, for consistency.
Also reduced size of Item from 88 to 72 bytes, by packing fields better.
Bug: webrtc:5696
Change-Id: I1b9d50831a8e9a358224682d06a782a3269b8416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264123
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37403}
This flag has gone unused for a long time, time to clean it up.
While we're here, convert NackRequester to use unit types.
Bug: webrtc:8624
Change-Id: I1f314f9b5b6771d4f9c351a7a9a887130b86907c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267408
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37400}
Let the OutgoingStream reference the parent instead of passing
references to individual items it needs, as follow-up CLs will add even
more items.
No functional change - pure refactoring.
Bug: webrtc:5696
Change-Id: I914e590c0d90e898d7d230a16170cf4faff2338c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264142
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37398}
Return the bitrate estimate as DataRate type
Remove list of affected ssrcs as unused
Bug: None
Change-Id: Ie31dce591d861624736d834194f90eb6c93f70f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37397}
This CL adds the API to enable message lifecycle events to be generated.
Those can in turn be used to generate metrics, e.g. latency metrics
tracking the time to send a message, the time until it's acknowledged,
and metrics tracking how often messages are expired.
This will be used to validate that message interleaving really improves
latency for high priority data channels.
The actual implementation of the API will be provided in follow-up CLs.
Bug: webrtc:5696
Change-Id: Ic06f8244d1c79a336975e35479130521dff17519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37396}
For this I added a header called no_cfi_icall.h and use it.
Also, some files use the gio header, but if the //base dependency is
not used, compilation errors occur. So I added an explicit dependency
on gio.
Bug: webrtc:13662
Change-Id: If732ede202dd413be6702bf06bf024cd203fdae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267340
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37395}
