UsePadding - signals to GoogCC that padding should be used to fill up to
BWE while BWE is ramping up.
Bug: webrtc:12707
Change-Id: I7b4922dff3a83da370c50c567050bfa748190b40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324160
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40979}
Ensure acked bitrate is not used for lower loss based estimate if
estimate improve.
Ensure LossBasedBweV2 is in state DelayBased if reached max rate.
Bug: webrtc:12707
Change-Id: I20230b99e0c2b530570e2f2de8ea88179f795c50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324140
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40977}
Expect OnNetworkAvailabability to be invoked when the transport becomes writable.
Before this change, ProbeController in GoogCC was expected to be created when the transport is writable or explicitly notifed after creation that network is not writable.
Bug: None
Change-Id: I623b1c34e40a82e912f85b92fea49629e7e72d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323463
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40975}
Sometimes OpenH264 returns a key frame even though we have not
requested one. However, SVC controller does not know about this
and will not reset its state. Since we are comparing expected tid
from SVC controller with actual tid from OpenH264, and drop frames
if they do not match, that causes a missing frame.
This CL resets the SVC controller state on key frames, ensuring
that it accurately maintains its state and does not drop frames.
Also, changes the message of the error log to be more descriptive.
Now, it will print the expected tid and actual tid.
Bug: webrtc:14877
Change-Id: I6c9e7532b2478773f03e5707bf7a1ca56e4f7b99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324001
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40972}
Without this change a FrameDropped sample will be added to
WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult at the
current capture rate as long as a captured window is minimized.
Bug: webrtc:1314868
Change-Id: I9b68675486642e7ca25674df689c207ac94a206e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323882
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40969}
The state should be computed from the actual estimate rather than the best estimate candidate. The fix is NOT under field trial.
And some other cleanup:
1. Loss based result will be computed in UpdateBandwidthEstimate method. Currently it is re-computed in GetLossBasedResult.
2. Rename current_estimate to current_best_estimate to avoid misunderstanding that current_estimate is the `final estimate`. The final estimate is computed by applying lower and upper bound on current_best_estimate
3. Remove current_state_. The state is stored directly in loss_based_result_.
Bug: webrtc:12707
Change-Id: Ie612845f907b9e6333fbd8249ddc9b93ad9f8042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324022
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40964}
This will be used in an experiment to ramp up BWE when BWE is reduced
due to loss.
Bug: webrtc:12707
Change-Id: I3b78f9dd3fe8ef9f94a9616640ffb8b2225e161e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324042
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40961}
This cl/ makes sure that the estimate cannot go lower than a factor of acked bitrate. The current flag LowerBoundByAckedRateFactor is set to 0, means we dont use it.
Bug: webrtc:12707
Change-Id: I75d5881f0b85a374af3f7039b82c71aee97fb7b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323881
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40958}
This ensure probe results can not be lower than 85% percentage of the
acked bitrate.
Bug: webrtc:11498
Change-Id: I501eeb84f7a049140c45c89e7de7e8080c13f94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324040
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40957}
MinNumObservations is set to 3 per default as loss based BWE should not be ready if it has few feedbacks. We use a flag, rather than a const since we want to customize it for our unit tests, which often have 1-2 packet feedbacks only, and customize it later in prod if necessary.
Bug: webrtc:12707
Change-Id: Id1cd21aaf6137996de2e51cb5e33fc2a4bb07d8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40952}
By producing new message only when new max is 10% larger than the previous max.
Bug: b/305042040
Change-Id: Id85784939f944de8115b881471b02214c34b3043
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323841
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40950}
Cleanup field trials for not probing when BWE limited due to high RTT,
loss.
Bug: webrtc:14754, webrtc:12707
Change-Id: Ib664784e321d9284d842ea42a0dd1d8361000f20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323640
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40949}
If the tests are run in a different order, the test might fail.
We fix this by resetting the histogram data at the start of the test.
Change-Id: I6fb349609842b55f416cf2ec8cd93d0b4328960e
Bug: chromium:1430806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323801
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Fredrik Hernqvist <fhernqvist@google.com>
Cr-Commit-Position: refs/heads/main@{#40946}
These features are not in use.
Bug: webrtc:12707
Change-Id: Ibe9fcae5e3fd7cb7ca289af80dad8480288c9ba3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323601
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40938}
Replacing RTC_DCHECK code with EXPECT_TRUE in the remote ntp time estimator unittest code.
This to prevent test failures when building and testing in non-debug mode.
Bug: webrtc:15572
Change-Id: I372fcd6ee29a4ddc07d6b27ddd492dcea13d399f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323181
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40936}
Makes it more clear that a certain API is only supported in Windows 11.
Bug: webrtc:15451
Change-Id: Ic3abfb2cbf0e30f9cb722ac843876f41279bf200
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323161
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40931}
The purpose is to ensure send socket buffers are not overfilled at high
pacing rates.
Bug: chromium:1354491
Change-Id: Ic6f473080292f84a2a099b85fb5817f7e14e7355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323000
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40911}
since the extension can be too large to fit the 16 bytes available
to one-byte extensions
https://www.rfc-editor.org/rfc/rfc8285#section-4.2
when including the width and height fields.
Also document when those fields are sent.
BUG=webrtc:12000
Change-Id: If17f57d40c0bde9b060f223c548e407d6c124b82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40910}
Convert most field trials used in PCLF tests.
Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40909}
This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string.
Bug: webrtc:7101, webrtc:10335
Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40897}
Do not automatically remove all tokens once we attempt to use them. This
mitigates an issue with Google Meet where an additional instance of a
DesktopCapturer is created and destroyed right away, taking away the
token we would use otherwise. Also save the token under same SourceId
once we get a new (but could be same) token from the restored session.
Bug: webrtc:15544
Change-Id: I565b22f5bf6a4d8a3b7d6d757f9c1046c7a0557d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322621
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40892}
ARM-specific settings were intended to be used on mobile ARM devices which may not be powerful enough. But the settings were also applied to ARM-based Macs. This changes restricts ARM-specific settings to Android and iOS platforms.
Bug: none
Change-Id: I68764b4c0679db07399bba5923f4a6be89c5ad80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321861
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40884}
This is a reland of commit 0d4b350006eae7dfeeb8c67f16f51b1c62351cee
Patchset 1 is the original CL. Patchset 2 contains a small tweak of the target bitrate in the unit test, in order to make in less susceptible to flakiness on runtime environments running a slightly different libvpx.
Original change's description:
> Add mitigation for very long frame drop gaps with vp8
>
> Bug: webrtc:15530
> Change-Id: I11f5e3f31f71301700dbff3fc9285236160bee45
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322320
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Auto-Submit: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40866}
Bug: webrtc:15530
Change-Id: I096b7d952286f7f53852d1ca70aea398b2747784
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322540
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40874}
PacketBuffer is not designed to store wide range of the rtp sequence numbers
Bug: webrtc:15508
Change-Id: I62b19ba2896a667d795a41c38a60f55ee3f60566
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321845
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@google.com>
Cr-Commit-Position: refs/heads/main@{#40839}
Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.)
With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates.
Bug: webrtc:13402,b/298636540
Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40827}
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}
NSApplicationActivateIgnoringOtherApps is about to be deprecated.
The default behavior is good enough.
Tested on Chrome using https://wicg.github.io/conditional-focus/demo/
Bug: webrtc:15511
Change-Id: I1f59aea3d4e7c4942d17ee5c4f1b6c2d398016ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321080
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40795}
A follow up cl/ is to remove passing upper link capacity from goog_cc to loss_based_bwe_v2.
Bug: webrtc:12707
Change-Id: I45af8ca6e8ba185700d0b7eb57004d2b61edeb9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40780}
This reverts commit 0f87b3853554ee5d4e92e487a5165b57771b6742.
This is not needed with the macOS 14 SDK, which has the fix, and which
was landed in https://crrev.com/c/4875713.
Bug: chromium:1484363, chromium:1431897
Change-Id: I1e019ce71b90333d5d1333a3cf8bb510a3dbd212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320820
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Auto-Submit: Avi Drissman <avi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40777}
When the specified device was not found in GetDevicesInfo,
SetPlayoutDevice/SetRecordingDevice will never return a (-1) error.
Bug: None
Change-Id: I9ac71cf72f7876c1c54ee593f184aa4007dba22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320500
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40768}
The purpose is to not allow an initial low link capacity estimate to reduce the current estimate.
Only delay overuse detection , low probe results or a loss event can
reduce the estimate.
Bug: webrtc:14392
Change-Id: Ib1618347f2c7681e3bd65d85ee687dec3cd67c97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320380
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40751}
