This moves steps from the sdp code for pc state over to the PC class
and slightly simplifies the contract between the two classes.
Moving forward it's easier to consolidate those steps in the PC
class with other grouped operations e.g. during teardown.
Also removing GetDataMid() method in favor of the sctp_mid() property.
Bug: none
Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40981}
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}
In some very high-bandwidth application there have been observations of
packet loss in the socket implementation (not on the network itself) due
to large bursts of packets arriving. Allocating too big buffers can of
course lead to issue as well, so this flag is intended to find a good
tradeoff.
Bug: webrtc:15585
Change-Id: I63eccb1a9f34d852d80c286fc27bffd17818f0ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324021
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40963}
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}
Required logic to query the min value of a SampleCounter along with some
additions to the existing test cases
Bug: webrtc:15580
Change-Id: I46afb30ad130f17f9e68ebc794b6935187bb2479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323900
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40956}
LegacyAudioDeviceModule depends on
org.webrtc.voiceengine.WebRtcAudioTrack and
org.webrtc.voiceengine.WebRtcAudioRecord, which were removed in
https://webrtc.googlesource.com/src/+/6fc700ec3d1f86d06e203011aa8f375f32b39d9e.
Including LegacyAudioDeviceModule results in build failures like:
> /builds/worker/checkouts/gecko/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java:13: error: package org.webrtc.voiceengine does not exist
> import org.webrtc.voiceengine.WebRtcAudioRecord;
> ^
> /builds/worker/checkouts/gecko/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java:14: error: package org.webrtc.voiceengine does not exist
> import org.webrtc.voiceengine.WebRtcAudioTrack;
> ^
> /builds/worker/checkouts/gecko/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java:39: error: non-static method setSpeakerMute(boolean) cannot be referenced from a static context
> WebRtcAudioTrack.setSpeakerMute(mute);
> ^
> /builds/worker/checkouts/gecko/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java:44: error: non-static method setMicrophoneMute(boolean) cannot be referenced from a static context
> WebRtcAudioRecord.setMicrophoneMute(mute);
Bug: webrtc:7452
Change-Id: Icaa4447ec6dc274d89f827ce4d1cc13c3e9f55ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323880
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40953}
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}
https://webrtc-review.googlesource.com/c/src/+/321081 made PEP-8 mandatory for WebRTC python file.
This CL allows to keep the old formatting style for existing python files because switching all methods and functions name from PascalCase to snake_case is non trivial.
Change-Id: Id094bbf72ee1c3c32027a49bc9763bc65dfb9ad2
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#40951}
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}
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}
Because of our asymmetrical codec situation, it's possible to have
send only codecs that we cannot negotiate even with ourselves.
This means that we should not have a DCHECK, but just a plain error.
Bug: webrtc:15064
Change-Id: I0c170e5c7f356197bcb04bcecb8259c344423ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40939}
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}
