Replace mock implmentation with manual noop implementaion.
libvpx interface is called a lot, and mock implementation of it adds
noticable overhead.
Bug: chromium:1281020
Change-Id: I7fe5cbfd08d5056a14d75e009acff368700c26a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269214
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37656}
This replaces the former WriteFatalLogAndAbort function with the two new
WriteFatalLog functions that're already submitted as overrides in
Chromium.
The default implementations of these are not defined under
WEBRTC_CHROMIUM_BUILD.
Bug: chromium:1216177
Change-Id: I207e1f96f14094d742a51849f4fa6b4f1022333e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Boström <pbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37652}
This adds a file,line version of this function (not yet committed) as
Chromium logging uses LogMessage(file, line, severity) and needs this
information to give better logs.
The two versions of this method will be implemented in webrtc_overrides/
and then committed to Chromium. At this point checks.cc will move its
anonymous-namespace version of this function (and be renamed) to
match this definition, but only define it when not building with
Chromium.
At this point WriteFatalLog will be using LogMessage(LOG_FATAL) to crash
in Chromium allowing us to upload better crash dumps and stacks to crash
reporting.
Bug: chromium:1216177
Change-Id: I3fd6a84cdfbb2552a5e628d46257bd7a00c9e6dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269288
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Boström <pbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37646}
This reduces the number of times we recreate video receive streams
and prepares for not having to do that for flexfec streams and avoid
having to recreate a video receive stream when flexfec config changes.
Bug: webrtc:11993
Change-Id: I11134b6a72eb162623ebbc12521d409da8616107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261941
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37641}
This removes the unused field trials
`WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` and
`WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` as well as the
`video_hysteresis` and `screenshare_hysteresis` parameters in
`WebRTC-VideoRateControl`.
The hysteresis parameters in `WebRTC-StableTargetRate` are currently
left, their future is unclear...
Bug: webrtc:9734
Change-Id: I9e6bbe4b630a0501d365bf69e87e65164c500122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269207
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37635}
From when callTest's send_transport_ is deleted and until the test is
completely ended, there is a possibility that the background task
webrtc::ModuleRtpRtcpImpl2::MaybeSendRtcpAtOrAfterTimestamp
will call send_transport_ which has already been deleted.
Fix this by deleting rtp_rtcp_ before send_transport_ is deleted.
Bug: webrtc:14202
Change-Id: Ief96c4712875beb55ef232a8bce990d1e9e9efe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37633}
Before this change the full screen application handler was failing to
detect PowerPoint going into presentation mode, resulting in the editor
window continuing to be shared rather than the intended behavior of
sharing the presentation itself.
Fix this by always looking for the PowerPoint full screen presentation
window, regardless of whether the editor window is still open. In
the current version of PowerPoint, the editor stays open during
presentation.
Bug: chromium:1231437
Change-Id: I1b21e263d25320cc236d127d22d4d64bb52fcbda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269560
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37632}
This is the first step of migrating
AudioProcessing::CreateAndAttachAecDump() from using std::string to
absl::string_view.
Bug: webrtc:13579
Change-Id: I8fc373e7ac55fd8e96bb0b01d1a30e28883ac9a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269400
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37631}
Optional better describes "optionality" so let's do it for the sake of
style. But a side-effect of switching to optional may be better memory
locality than std::unique_ptr<>. (Anecdotally I saw a pprof suggesting a
significant amount of time being spent allocating/reading these maps.
This CL is unlikely to make the difference but it can't hurt.)
Bug: webrtc:14289
Change-Id: I7dcea9625b95c2f1a23e7d9595d27b58883570e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269404
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37624}
This CL removes the old WebRTC-FakeNetwork(Send|Receive)[Param] field
trials. Instead, use the paremterized WebRTC-FakeNetworkSendConfig and
WebRTC-FakeNetworkSendConfig.
As an example, instead of WebRTC-FakeNetworkSendLossPercent/5/ you can
use WebRTC-FakeNetworkSendConfig/loss_percent:5/
Bug: webrtc:14238
Change-Id: Ib0021c5425e59688a5043a6285d9baccf2234a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269405
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37623}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
In WebRtcVideoChannelBaseTest.EncoderSelectorSwitchCodec a mock encoder
selecter or stack allocated and then registered with the channel.
Since this test uses real-time clocks/threads, there is a chance that
the selector callback will be called after the mock goes out of scope,
but before the test had time to be torn down.
This CL fixes that by simply de-registering the callback before the
end of the test.
Bug: b/239855550
Change-Id: Ibb38a914933494fd04c963b9a13f2cc4aee160d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269402
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37618}
Replace helper functions with the constant
Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.
Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
Fix call in FrameBuffer3Proxy to OnDecodableFrameTimeout to match with
FrameBuffer2Proxy and how OnEncodedFrame is called.
Bug: none
Change-Id: I6f5f18bd9a425aa55f5826654b24c2c0ee5af5f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269302
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37610}
Inserting packet with zero arrival time may trigger inconsistent state in the internal map where packet sometimes treated as received, but sometimes treated as not received.
Bug: chromium:1346959
Change-Id: I0809e41a873103dcd62528358e64794c1d3cb28f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37609}
1. Add loss threshold for high bandwidth preference. If the average loss ratio is less than the threshold, then the model prefers higher bandwidth candidates. Otherwise, it prefers lower bandwidth candidates. Before, it always prefers higher bandwidth candidate. The default value is 0.99, means it always prefers high bandwidth candidates.
2. Only increase the estimate if the inherent loss (random loss) is equal to/greater than the average loss. If the inherent loss is less than the average loss, then it is oversending, thus should not increase the estimate.
Bug: webrtc:12707
Change-Id: I37eb536679ca29e017a4a47703b417efd4d103ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269101
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37608}
