This new name emphasizes that the field trial keys are only allowed
within the current scope. We already have test::ScopedFieldTrials that
can be used to ensure that the global field trials string itself is
isolated.
Bug: webrtc:14705
Change-Id: I8b66bbd9c11d97985292c334d2d3496a047074a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284862
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38796}
M108 Stable has been released, which does not contain googIPv6 anymore,
and today the last downstream dependency on this flag was removed.
Let's delete!
Bug: webrtc:14608
Change-Id: Ia2d201f0da04b14961f891687b6135fc69b7767e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285720
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38786}
Extends the mocks for rtpreceiver rtpsender and videotrack. This change
allows the external HangoutsKit client to remove its own mocks of rtc
types.
Bug: none
Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jack Smith <jackdsmith@google.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38782}
Limit the number of TURN servers to 32 in order to allow the
prioritization to assume a fixed offset for (de)prioritizing
candidates. See
https://github.com/w3c/webrtc-pc/pull/2679
for discussion including some data on current usage.
Guarded by WebRTC-LimitTurnServers which is used as a killswitch.
BUG=webrtc:13195
Change-Id: Ib12726af426ae4238aa7eb6aa062c71af52d495f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38767}
This is a pure move/rename. The reason for wanting the tests in
RTPVideoHeader is that it is the GetAsMetadata() function that we are
testing and in a future CL we'll also want to test SetFromMetadata().
// Bots green, no need to wait for the remaining ones, just a move
NOTRY=True
Bug: webrtc:14709
Change-Id: Iecb938e79e7e8d55e208baea190eef4c6730158e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285460
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38764}
Update GetAsMetadata() to include more of the RTPVideoHeader metadata.
The intent is to be able to both get and set all of these from
JavaScript behind a flag.
Planned follow-up CLs:
1. Also get codecs-specifics, starting with VP8.
2. Test refactoring/rename: Move tests to RTPVideoHeaderTest.
3. Add RTPVideoHeader::SetFromMetadata() covering everything gettable.
4. Chrome plumbing.
Bug: webrtc:14709
Change-Id: I78679b9ff4ca749d50f309a1713e71ceabb826dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285084
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38756}
In preparation of adding RTPVideoHeader::SetFromMetadata() method, the
VideoFrameMetadata construct-from-RTPVideoHeader is replaced by
RTPVideoHeader::GetAsMetadata(). This serves two purposes:
1. Having "GetAs" and "SetFrom" in the same file reduces the risk of
these two methods getting out of sync as we expand its usage.
2. This is necessary to avoid a circular dependency that would
otherwise be introduced by RTPVideoHeader::SetFromMetadata().
Bug: webrtc:14709
Change-Id: I127b3d15f9a8c6af210449a5a50d414c9ba79930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285080
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38735}
This will clone an encoded video frame into a sender frame,
preserving metadata as much as possible.
Bug: webrtc:14708
Change-Id: I6f68d2ee65ef85c32cc3c142a41346b81ba73533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284701
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38733}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.
When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.
NOTRY=True
Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
As to not break downstream projects, the old name api/stats_types.h is
kept around to help include api/legacy_stats_types.h. We can delete this
in a follow-up.
NOTRY=True
Bug: webrtc:14180
Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38610}
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.
Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb.
Reason for revert: Breaks upstream.
Original change's description:
> Add checks for api/test mocks to make sure they're complete
>
> Also unifies the mock inheritance if they inherited from a ref counted
> interface:
> - it should only inherit from the interface
> - it should use make_ref_counted
>
> Bug: webrtc:14594
> Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38602}
Bug: webrtc:14594
Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38604}
Also unifies the mock inheritance if they inherited from a ref counted
interface:
- it should only inherit from the interface
- it should use make_ref_counted
Bug: webrtc:14594
Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38602}
The Chromium implementation unfortunately has a rare deadlock.
Rather than patching that up, we're changing the metronome
implementation to be able to use a single-threaded environment
instead.
The metronome functionality is disabled in VideoReceiveStream2
construction inside call.cc.
The new design does not have listener registration or
deresigstration and instead accepts and invokes callbacks, on
the same sequence that requested the callback. This allows
the clients to use features such as WeakPtrFactories or
ScopedThreadSafety for cancellation.
The CL will be followed up with cleanup CLs that removes
registration APIs once downstream consumers have adapted.
Bug: chromium:1381982
Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38582}
As the exposure of power efficient stats to JavaScript are limited as
to reduce the fingerprinting surface to getStats, a new RTCStatsMember
derivation, RTCLimitedStatsMember, was added in this change. This sets
the exposure criteria of the stat on the type, which keeps the size of
the RTCStatsMember class the same and allows for extension in the future
for new types of stat restrictions.
Bug: webrtc:14483
Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38576}
This is a reland of commit 38fcd58429b29c9474f1647efed7ebeb543c0637
Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}
Bug: none
Change-Id: Id634799286f6d1f1eaf315ebe8e70de669d589db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38572}
Classes defined inside the class PeerConnectionE2EQualityTestFixture are replaced by the ones define in media_configuration.h.
Change-Id: I1c025ff10aacf8cbc3df9bfa742a40622fe0807a
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281860
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38568}
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.
BUG=chromium:1354101
Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
The goal is to remove the dependency between PeerConfigurerImpl and PeerConnectionE2EQualityTestFixture so that PeerConfigurerImpl can be used in PeerConnectionE2EQualityTestFixture API.
Change-Id: I29ae44b9d0e39075d0c395ff9d9f8d313be12176
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38560}
This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138, b/256595485
Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38557}
This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
Bug: none
Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38536}
These metrics were not only non-standard, but residing in the
non-standard "track" stats object that we want to delete. As per
https://github.com/w3c/webrtc-stats/issues/695#issuecomment-1259611462
these metrics are no longer needed because we already have
inbound-rtp.totalInterFrameDelay/totalSquaredInterFrameDelay which is
basically the same thing.
// mac_rel infra failures are unrelated
NOTRY=True
Bug: webrtc:14522
Change-Id: I565da42514a93f15532ba8357dd006547a5296ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38509}
