This reverts commit 8c9f8a0fc0.
Reason for revert: The issues in code that this CL relies on have now been addressed and the CL should now work as intended.
Original change's description:
> Revert "Activate adaptive channel mixing when the echo canceller runs in stereo"
>
> This reverts commit 82a3659e11.
>
> Reason for revert: Seems to be breaking in chromium:
> https://chromium-review.googlesource.com/c/chromium/src/+/7490942?tab=checks
>
> #
> # Fatal error in: third_party/webrtc/modules/audio_processing/audio_buffer.cc, line 152
> # last system error: 2
> # Check failed: !use_adaptive_downmixing || buffer_internal_num_channels_ == 2
> #
>
> Original change's description:
> > Activate adaptive channel mixing when the echo canceller runs in stereo
> >
> > This CL activates adaptive channel mixing when the echo canceller runs in stereo mode.
> > The mixing is provided by the CaptureRemixer class.
> > The adaptive mixing is activated behind a finch-based kill-switch that allows it to be
> > activated.
> >
> > Bug: webrtc:42221468
> > Change-Id: I5c4e79edb69752bbbb4c1cfe9271d1224fb695aa
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433640
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Tomas Lundqvist <tomasl@google.com>
> > Cr-Commit-Position: refs/heads/main@{#46678}
>
> Bug: webrtc:42221468
> Change-Id: Ifc3f489e0fe8b4da1b891fd0dd425280a8698fdf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442441
> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46685}
Bug: webrtc:42221468
Change-Id: I720df6d70de705a1027907d3cd2758d6b569cd75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/444140
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46800}
This change introduces specific priority levels for audio and
video tasks within the TaskQueueFactory to allow for better
platform-specific thread scheduling.
Changes:
* New field trial WebRTC-MediaTaskQueuePriorities that activates the
priority change.
* New Priority Enums: Added kVideo and kAudio to
TaskQueueFactory::Priority and corresponding kVideo and
kAudio to ThreadPriority.
* WebRTC platform thread implementation:
- Video priority initially maps to high priority.
- On Mac/iOS (GCD), both audio and video map to
DISPATCH_QUEUE_PRIORITY_HIGH because of API limitation.
* Task queue updates:
- Updated the audio encoder queue to use kAudio priority.
- Updated the video encoder & decoder, incoming video
stream, and video frame transformer queues to use kVideo priority.
* Queue renaming: Standardized several task queue names by
appending "Queue" (e.g., "AudioEncoder" became "AudioEncoderQueue").
Bug: chromium:470337728
Change-Id: I4790990340a72a54945750c2c4e1f97314edf375
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/436580
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46744}
This change makes network slices on cellular networks more preferable
than the regular network by decreasing the network cost slightly when a
slice is in use.
Network slicing is typically offered as a premium feature on 5G networks
at an additional price.
The network cost is only reduced when the underlying adapter supports
slicing, currently 5G or undifferentiated cellular. It is gated behind
the WebRTC-UnifiedCommunications field trial.
BUG=webrtc:466507512
Change-Id: I230264c95d2d9f0386a096e7ca51cf44b5684922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442681
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#46710}
This reverts commit 82a3659e11.
Reason for revert: Seems to be breaking in chromium:
https://chromium-review.googlesource.com/c/chromium/src/+/7490942?tab=checks
#
# Fatal error in: third_party/webrtc/modules/audio_processing/audio_buffer.cc, line 152
# last system error: 2
# Check failed: !use_adaptive_downmixing || buffer_internal_num_channels_ == 2
#
Original change's description:
> Activate adaptive channel mixing when the echo canceller runs in stereo
>
> This CL activates adaptive channel mixing when the echo canceller runs in stereo mode.
> The mixing is provided by the CaptureRemixer class.
> The adaptive mixing is activated behind a finch-based kill-switch that allows it to be
> activated.
>
> Bug: webrtc:42221468
> Change-Id: I5c4e79edb69752bbbb4c1cfe9271d1224fb695aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433640
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Tomas Lundqvist <tomasl@google.com>
> Cr-Commit-Position: refs/heads/main@{#46678}
Bug: webrtc:42221468
Change-Id: Ifc3f489e0fe8b4da1b891fd0dd425280a8698fdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442441
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46685}
This CL activates adaptive channel mixing when the echo canceller runs in stereo mode.
The mixing is provided by the CaptureRemixer class.
The adaptive mixing is activated behind a finch-based kill-switch that allows it to be
activated.
Bug: webrtc:42221468
Change-Id: I5c4e79edb69752bbbb4c1cfe9271d1224fb695aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433640
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#46678}
The WebRTC-UseNtpTimeAbsoluteSendTime field trial has been enabled by default for a while now and can be cleaned up. This commit removes the field trial from experiments/field_trials.py and hardcodes the behavior in RtpSenderEgress.
Bug: webrtc:42226305
Change-Id: I15103cec73e7c7e3399d2fad3e79291cf1f888af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440101
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46630}
This cl adds a field trial `WebRTC-CorruptionDetectionFrameSelector`
and a utility class for parsing the configured time spans.
Bug: webrtc:358039777
Change-Id: I6b997a0ed63168d803375d931a8ebe601e763b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441022
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46603}
Appears unused as no one is reading the weak_transport_interval_ var in
P2PTransportChannel. This looks like a bug, so we should either clean it
up, or pass it into the IceConfig ctor where kWeakPingInterval is used
as the default.
For now proposing to clean it up unless there is downstream usage.
Bug: webrtc:42221607
Change-Id: Ic2c559e68f9fc799f75da6599856528c9dface8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/427500
Commit-Queue: Joachim Reiersen <joachimr@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46313}
This CL adds a new (experimental) speech level estimator to AGC2. The
goal is make the AGC less likely to adapt to (and boost) background
noise.
Bug: webrtc:457791164
Change-Id: I425d72790b2e1dd993ffb7530c02b0485a80f4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/422966
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46143}
This reverts commit ddb627e555.
Reason for revert: The downstream tests causing the breaking of the Chromium roll are now addressed.
Original change's description:
> Revert "Switch to use 32 kHz processing inside APM"
>
> This reverts commit 179be29133.
>
> Reason for revert: Broke Chromium roll.
>
> Error: https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/2423056/overview
>
> Original change's description:
> > Switch to use 32 kHz processing inside APM
> >
> > This CL switches to using 32 kHz processing internally inside WebRTC APM
> > to avoid using the current 3-band split filter which has been shown to
> > have issues with aliasing impacting the speech quality.
> >
> > The intention is to revert this change once the issues in the 3-band
> > split filter have been addressed.
> >
> > Bug: webrtc:454695115
> > Change-Id: Id87e7f8d2ba37a915b3640f7eeb5c996037c59aa
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/419860
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#46026}
>
> Bug: webrtc:454695115
> Change-Id: I599adf846217f0cb81588e84f541277465ef856a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/420580
> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46037}
Bug: webrtc:454695115
Change-Id: Ic0bb60b8c7103c7eac9563bae2d06935c152764e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/420463
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46046}
This reverts commit 179be29133.
Reason for revert: Broke Chromium roll.
Error: https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/2423056/overview
Original change's description:
> Switch to use 32 kHz processing inside APM
>
> This CL switches to using 32 kHz processing internally inside WebRTC APM
> to avoid using the current 3-band split filter which has been shown to
> have issues with aliasing impacting the speech quality.
>
> The intention is to revert this change once the issues in the 3-band
> split filter have been addressed.
>
> Bug: webrtc:454695115
> Change-Id: Id87e7f8d2ba37a915b3640f7eeb5c996037c59aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/419860
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46026}
Bug: webrtc:454695115
Change-Id: I599adf846217f0cb81588e84f541277465ef856a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/420580
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46037}
This CL switches to using 32 kHz processing internally inside WebRTC APM
to avoid using the current 3-band split filter which has been shown to
have issues with aliasing impacting the speech quality.
The intention is to revert this change once the issues in the 3-band
split filter have been addressed.
Bug: webrtc:454695115
Change-Id: Id87e7f8d2ba37a915b3640f7eeb5c996037c59aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/419860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46026}
This feature has been Stable for 2 weeks meaning we did not have to kill
switch it. Delete the flag such that the only valid code path is that
this feature is enabled.
Bug: chromium:440975167
Change-Id: I8f83ee3962dc5dc8fae565c20cce92be7bb3266e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/416540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45933}
This CL adds a helper class to simplify parsing of the
WebRTC-EncoderSpeed field trial string, which specifies the
encoder complexity to use per codec and camera/screenshare permutation.
Bug: webrtc:443906251
Change-Id: I2a89e9149b8583b117e448e2dd3e9cf309729bb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/414401
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45854}
enabled by default and guarded by the killswitch
WebRTC-EnforceTransceiverDirection
Bug: chromium:448408148
Change-Id: I15e98d371d1b398aff4b9cb88bf34a483b36fe1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/413340
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45847}
This is the only keep sorted usage in WebRTC, let's make the analyzer happy with it.
Bug: b/446606314
Change-Id: I0b086eb4cc9581e6b6f63d88466ae27e209a9851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/413000
Auto-Submit: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45786}
The field trial name is
WebRTC-Sctp-Snap
This prevents it from being enabled by default until it is ready
and has at least a submitted IETF draft
Bug: webrtc:426480601
Change-Id: I3de79ce60c70d74774fa30f9c174987889a66f25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397381
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#45735}
This CL introduces a field trial which allows the libaom AV1 encoder
wrapper to drop frames if the input frame is repat frame and the
scalabilit controller assigns it a non-base layer.
Bug: webrtc:445115234
Change-Id: I56692808c07251c01136b2246b5ee3bf7c7bfeda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/409564
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45712}
And implemented it for destruction of MediaEngine in ConnectionContext.
Having it as a PostTask means that it can happen arbitrarely
later...if worker thread is loaded...which it can be, in some
apps. The asynchronous destruction of MediaEngine means
that audio "data" can flow through the system even
after the destructor of PeerConnectionFactory has been
run.
Work in progress doc: https://docs.google.com/document/d/1tZHCWfCjIFtPHtm_tg9hAksHJ-JOurASckv4k-aJInw/edit?usp=sharing
And related bug: webrtc:14449
Bug: webrtc:443588673
Change-Id: I2d0f117ca13fe166d913d71af1ba1e6ff8598b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408745
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45633}
This changes HeaderExtensionsToNegotiate to remember the result
of the last negotiation.
Corresponding spec change: https://github.com/w3c/webrtc-extensions/pull/238
Bug: webrtc:439514253
Change-Id: I420b65f252398b1bb72d1938c48dc548ec18fd60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404946
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45493}
No issues with the fix has been reported, thus cleanup the killswitch.
Bug: webrtc:361124449
Change-Id: Id7ed12e1bfa03e835b47cc86316553d8b2f9b153
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406540
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45449}
When field trial "WebRTC-RTP-Lifetime" is enabled, the creation of the
inbound-rtp stats object is delayed until the first packet is received.
- This aligns with spec and Firefox behavior.
To aid testing, PeerConnectionTestWrapper has new negotiation methods
added to give the test more control without having to write a lot of
boilerplate code.
(A separate CL will deal with outbound-rtp which, while the plan is
they continue to be created before first packet is sent, should be
delayed until after O/A has completed.)
Bug: chromium:406585888
Change-Id: Ibac2128e80e0153659b68cc0f00869e5d1f27a69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/405740
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45428}
This reverts commit c432ba14db.
Reason for revert: Investigate downstream test breakage - b/429391976
Bug: webrtc:423841921, webrtc:42222445
Original change's description:
> Refactor class SendSideBandwidthEstimation
>
> The goal is to make states more clear and be able to log where a certain decision come from.
> In this cl:
> - loss bases BWE handling moved to separate files
> - remove field trial WebRTC-Bwe-ReceiverLimitCapsOnly
>
> Bug: webrtc:423841921, webrtc:42222445
> Change-Id: I502bee094e18606f8a188214fafa421a868023ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396342
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45056}
Bug: webrtc:423841921, webrtc:42222445
Change-Id: I8dcda24877dd8b1fbab4e1bb5235e2e7b903dabf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/399080
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45088}
The goal is to make states more clear and be able to log where a certain decision come from.
In this cl:
- loss bases BWE handling moved to separate files
- remove field trial WebRTC-Bwe-ReceiverLimitCapsOnly
Bug: webrtc:423841921, webrtc:42222445
Change-Id: I502bee094e18606f8a188214fafa421a868023ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396342
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45056}
WebRTC-NoSdpMangle/Enabled,1,2,83/
allows for finch rollout of restrictions without requiring a code change to turn off low-usage munging.
WebRTC-NoSdpMangleForTesting/Enabled,1,2,83/
will reject any modification by default unless it is part of the
comma-separated list of exceptions. This allows for proactive deactivation and adding E2E tests.
Bug: chromium:40567530
Change-Id: If57602089dcc3e9372044b37996ec4c468fdd0cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#44981}
Removed from
- AudioEncoderOpus
- BitrateAllocator
stable_target rate in GoogCC is not used for anything except as a stable rate for opus. In order to simplify GoogCC, it will be be deprecated since it does not work that well in loss limited network scenarios.
Bug: webrtc:423841921
Change-Id: I6356ab4f11542816b6200cbb055d1a98ecebad74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396740
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44961}
The feature is mature enough that we don't need these kill switches
and field trials anymore.
Bug: chromium:40200151
Change-Id: I0344a42c8637034226925bc95f1eda10c69df977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396801
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44959}
TargetTransferRate.stable_target_bitrate is planned to be deprecated since it is mostly unused and does not currently work well in networks limited by packet loss.
Bug: webrtc:42220156, webrtc:423841921
Change-Id: Ib2a0321a6771974e679ec9a0da16e175a554ccc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396720
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44953}
Field trial itself was already cleaned up from the code
Bug: webrtc:42233254
Change-Id: I5a8c842eea84f1e210c38aa9bc7f5b7deccab310
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393200
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44739}
This is now disabled by default, guarded by
WebRTC-NoSdpMangleNumberOfContents
which can be disabled if necessary.
BUG=webrtc:40567530
Change-Id: I02a3d4d21678f41e0910144be1aa159828c40757
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387560
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#44517}
Gated behind the default-empty list of restricted addresses tracked by
the field trial WebRTC-NoSdpMangleUfragRestrictedAddresses.
Bug: b/409713509
Change-Id: Iabb5c8b5307c66215635336bee55f7bfbd040f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388340
Auto-Submit: Tom Van Goethem <tov@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44477}
G2G (glass to glass) latency metric uses RTP timestamps to estimate the metric. The RTP timestamps are converted to senders NTP time using data from Sender Reports. This work fine in most cases, but not when Virtual Video SSRCs are used.
This field trail changes the G2G metric to use the absolute capture time header extension to get senders NTP timestamp. This is designed to work with Virtual SSRCs, and is currently used for audio end to end latency metric.
Enables this should improve G2G latency metric calculations with VVSSRCs , and create the same results in other cases. It should not create any other side effects. The feature is hidden behind the flag so we can make sure that's the case as we roll this out.
Bug: webrtc:401512883
Change-Id: I7a75d0a430f1fd1bcd79d9a228a7429300d5fafe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/380421
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#44126}
This unlaunches the experiment for H264 due to issues reported for H264
where the old packet buffer behaves better than the new one. A recently
introduced issue will be fixed separately from this CL and rollout
restarted.
This CL is not a clean revert though (see delta with PS#1):
we will continue to use the H26xPacketBuffer for the new codec H265
because it does not work with the old packet buffer anyway.
Bug: webrtc:41480904
Change-Id: Icac49fa70f1c78d1ed596a7838d1417f6588a8b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/380861
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44104}
The old experiment WebRTC-Video-QualityScaling only applies for VP8,
VP9, H264 and Generic, which does not cover H265.
In order to fine tune QP thresholds for H265, a new field trial is
added, this allows us to experiment with those without affecting the
old field trial which has a bunch of existing dependencies.
Some drive-by comments are added to the existing code.
Bug: webrtc:402154973, chromium:391907171
Change-Id: I13c8b496a1c708528d7538c8d1d2ff66de4c2ae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/380420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44088}
