since it has grown too large
BUG=None
Change-Id: I9dfffd6264db3206c0674a3446c857c139ba6fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267826
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37492}
Makes std::string version of rtc::RtcEventLogOutput::Write() no longer pure virtual while making the absl::string_view version pure virtual. Also removes unnecessary overloads in subclasses.
BUG=webrtc:13579
Change-Id: I8fb449560b795a1ef76fab27533d9042d0c34cd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268062
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37484}
to allow for downstream users to upgrade.
BUG=chromium:1338902
Change-Id: If6b56ab63f7859c13e9ebc70326e1088e5dfff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268141
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37475}
Lots of code call rtc::Thread directly instead of through TaskQueueBase
interface, thus to continue migration step by step rtc::Thread needs
to implement both old and new TaskQueueBase interfaces.
Bug: webrtc:14245
Change-Id: Ie7cac897a4c8a6227b8d467a39adb30aec6f1318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267984
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37474}
This reverts commit a17651f7d8.
Reason for revert: triggers failure in downstream test
Original change's description:
> Fix overflow due to rounding in AbsoluteSendTime::To24Bits
>
> Actual rounding is not important for this time as long it is consistent
> during the call: only difference between two absolute send time matter
> Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
>
> Bug: None
> Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37468}
Bug: None
Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37473}
This reverts commit dd32562f24.
Reason for revert: Updated the original change to dynamically load
the CoreMessaging.dll instead of statically linking with the .lib.
Original change's description:
> Revert "Wait for frames to arrive in WgcCapturer instead of returning nothing."
>
> 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}
Change-Id: I6cc2becd9ed363782ab2f326f58d9401bc8fb820
Bug: chromium:1314868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267902
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37470}
Actual rounding is not important for this time as long it is consistent
during the call: only difference between two absolute send time matter
Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
Bug: None
Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37468}
https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
due to which the min mic level override is always enforced, if specified
even if the user manually adjusts the mic level to zero.
This CL fixes that bug, the changes run behind a kill switch.
TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
Bug: chromium:1275566
Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37460}
This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
and always enables the code path behind that flag on Mac. In summary,
the analog AGC behaves as follows on Mac:
1. the minimum level is overridden to 20
2. the minimum is applied even when clipping is detected
3. when the level is manually adjusted to 0, the minimum level is
enforced - i.e., 20
Note that the 3rd property had been unintentionally added when the
changes were added behind the aforementioned field trial. This will
be fixed in a follow-up CL.
Bug: chromium:1275566
Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37459}
This adds the final piece, which makes the socket and the retransmission
queue generate the callbacks.
Bug: webrtc:5696
Change-Id: I1e28c98e9660bd018e817a3ba0fa6b03940fcd33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264125
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37455}
This allow to remove the testonly from the rtc_event_log_visualizer
binary and the implicit dependency on the path of the default
conversational speech file.
The binary size of event_log_visualizer passes from 2.1 MB to 4.0 MB.
Bug: b/237526033
Change-Id: I71cf647f039f26f30c792c49c752cff5c5b329a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267663
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37453}
This helper suppose to replace ToQueuedTask when calls to TaskQueueBase interfaces are converted to PostTask variants that take absl::AnyInvocable.
Bug: webrtc:14245
Change-Id: I590a6ca068cf5e682ffb34770bd54cf5ce37d826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267706
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37449}
For all messages where the last fragment was _not_ put on the wire, the
send queue is responsible for generating lifecycle events, but once all
fragments have been put on the wire, it's the retransmission queue that
is responsible. It does that by marking the final fragment of a message
with the lifecycle identifier, and once that message has been fully
acked by the cumulative ack TSN, it's clear that the peer has fully
received all fragments of the message, as the last fragment was acked.
For abandoned messages - where FORWARD-TSNs are sent, those will be
replied to by a SACK as well, and then we report abandoned messages
separately, to later trigger `OnLifecycleMessageExpired`.
This CL adds support in OutstandingData, which doesn't generate the
callbacks itself, but just reports them in the AckInfo.
Bug: webrtc:5696
Change-Id: I64092f13bcfda685443b7df9967b04d54aedd36a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264124
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37448}
This is a reland of commit 3afb8e2431
Patchset 1 is the original CL. Patchset 2 contains a fix:
Depending on call site, the number of spatial layers for VP9 might be
signalled in three different ways. One of them was afaict only used in
out perf tests, and resulted in the max bitrate being incorrectly
capped.
The fix now checks that field too.
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, webrtc:14234
Change-Id: Idcaf4321a20c917e4049522c577336ddcfc7ffbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267860
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37446}
Intended to be used in downstream code when deleting deleting this
attribute.
Bug: webrtc:11607
Change-Id: I39417997a2ec2e72d726da476b5bce88abe267b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267843
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37445}
