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 have all been moved to "inbound-rtp" and now that upstream
projects have migrated we can delete the old location.
Unblocks https://crbug.com/webrtc/14175
Bug: webrtc:14521, webrtc:14524
Change-Id: Ia2bfa399d62304cc0ead0e65c340dfad20acc530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38532}
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}
It's deprecated and has been removed from Chrome. Let's follow suite.
// Passing all but unrelated bots
NOTRY=True
Bug: webrtc:14608
Change-Id: I6f2601af5b1dc08164230ebf15db2d2f1754f9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38488}
This information is now readily available. Let's expose it.
In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.
Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
defined in
https://w3c.github.io/webrtc-encoded-transform/#rtcrtpsender-extension
Note: this does not implement the "rid(s)" parameter which will be done in a future CL.
VP8 still synchronizes keyframes on all layers even when asked for ones on individual layers while H264 (when implemented as three different encoders in SimulcastEncoderAdapter) can actually utilize this.
This does not change the behavior when receiving a RTCP PLI for a particular layer.
BUG=chromium:1354101
Change-Id: Ic8b14d155242e32c9aeafa55fe6652f346ac76b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274169
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38472}
This is needed to get rid of a mock in Chrome.
Bug: webrtc:14594
Change-Id: I27df2a1466e6a2dea87a211f803b3f2c7aa57478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280041
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38458}
This new class implements the existing FieldTrialsView interface,
extending it with the verification functionality. For now, the
verification will only be performed if the rtc_strict_field_trials GN
arg is set.
Most classes extending FieldTrialsView today have been converted to
extend from FieldTrialsRegistry instead to automatically perform
verification.
Bug: webrtc:14154
Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38453}
Enable by default the look for the first echo.
Bug: webrtc:14205
Change-Id: Iae904679c1432f3a0766263907cf376903685b97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278043
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38452}
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.
In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.
[1] https://crbug.com/webrtc/10335
Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.
Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}
which is a more modern way to represent something that either has a value or is not set
BUG=webrtc:14544
Change-Id: I0a06b30b1c7f802247eb1f60e69271594b94a6f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278421
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38443}
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats
There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.
In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.
# Unrelated infra failures
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
This patch adds variant of PeerConnectionInterface::AddTrack
that takes an initial_send_encodings.
This allows for setting/modifying encoding parameters before sdp
negotiation is performed/complete (e.g requested_resolution).
This is already available if using RtpTransciverInit and AddTransceiver,
but was not added to AddTrack because of concerns that it complicated matching with existing transceivers. This CL sidesteps that by never matching to a preexisting transceiver if initial_send_encodings are specified.
Note:
1) The patch adds a new method rather than an extra (e.g optional)
argument to existing AddTrack. This is to avoid problems with downstream mocks.
2) chromium "problems" was fixed in https://chromium-review.googlesource.com/c/chromium/src/+/3952684 and https://chromium-review.googlesource.com/c/chromium/src/+/3956060
Bug: webrtc:14451
Change-Id: I19b5a03872730280fbf868ca5d3a2f46443359f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38437}
Typically, remote candidates come from signalling and are deserialized
into C++ objects. The network_type field of these candidates is
always ADAPTER_TYPE_UNKNOWN.
However, in tests it is common to pass SDP and remote candidates as C++
objects. In this case, the network_type property of remote candidates
is preserved, so DCHECK might be triggered when GetStats is called.
Clearing fields that are not suitable as remote candidates fixes
this issue.
Bug: None
Change-Id: Ida01b0224bce5cf3e87bcad1ddaca35c9f4fffe7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38436}
The tool will generate a C++ header with all field trials in
REGISTERED_FIELD_TRIALS. This registry will later be used while looking
up field trials from native code to ensure they have been properly
registered in accordance with the policy.
Bug: webrtc:14154
Change-Id: I29bf880735121034585c541c46ef19f617d0afb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276268
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38426}
This reverts commit c1d5fda22c.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
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
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
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}
Also call out the places where it happens explicitly - these are places
that need to be redesigned.
Bug: chromium:1177125
Change-Id: I3237d028dbb22380e8fbf7cedb03e965d1fcf2aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279022
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38384}
Recent WebRTC stats spec changes have added restrictions on what stats
are available to JavaScript. This is done to reduce that fingerprinting
surface of WebRTC getStats. For example, stats exposing hardware
capabilities have requirements that must be met by the browser. See [1]
for more details.
This CL adds the types and the enumerations. Stats with these
restrictions should not be added until Chromium has implemented
filtering based on the stat type.
[1] https://w3c.github.io/webrtc-stats/#limiting-exposure-of-hardware-capabilities
Bug: webrtc:14546
Change-Id: I6dae5d4921c7a2bc828a4fc8f7d68e0c59f3be82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279043
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38381}
Only use the network thread for sending and receiving packets.
The one and only network thread is used as a worker thread in all
PeerConnections. Pacing when sending packets is done on the worker thread.
Bug: webrtc:14502
Change-Id: Ib373315688ae4d810ae1e4421101a859fca93b31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278621
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38354}
RTPHeader is not exported, so the TransformableIncomingAudioFrame can't be mocked in chrome tests, using a getter instead.
Bug: chromium:1247260
Change-Id: I2af4e6a88b3f4772b3bb50ee0ae9d5c80fed3ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278785
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@{#38352}
This also tests the UMA stats newly added to it.
Bug: chromium:1177125
Change-Id: I89bb17c1897565cd91ea5bbd92062018317738ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278600
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38345}
The one to use is StunMessage::ValidateMessageIntegrity(password).
Bug: chromium:1177125
Change-Id: I345f4d6b60090651bc23c3aa6358d4fb24723f9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278601
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38338}
This will allow us to see if bad integrity ever occurs, and where integrity
is not applied.
Bug: chromium:1177125
Change-Id: I7abdaba93088e4eef8121205e7dd76b21204cae8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38330}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
This patch
1) modifies VideoAdapter to use requested_resolution
instead on OnOutputFormatRequest, iff there are no active encoders
that is not using requested_resolution (i.e all "old" encoder(s) are
not active).
2) modifies VideoBroadcaster to not broadcast wants from
encoders that are not active (iff there is an active encoder
using requested_resolution).
3) fixes a bug in encoder_stream_factor in that the
requested_resolution was not propagated to return value
(must have been lost in merge?).
Bug: webrtc:14451
Change-Id: I00e0907f0fe9329141ed169576fa46cdc5384886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38323}
They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.
To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.
Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
tkchin and deadbeef are not working on webrtc on a daily basis at the
moment, so non-urgent approvals should not go to them.
Not mentioning this has led to misunderstandings.
Bug: chromium:1371843
Change-Id: I91e99249d32e52d6083de9c2b1bfebfc4693acac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278201
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38314}
These metrics were recently standardized. Part of the standardization
effort was to move them from obsolete "track" stats (on track for
deprecation and removal: https://crbug.com/webrtc/14175) into the
"inbound-rtp" stats which are not deprecated.
To ease transition for downstream projects, the metrics are temporarily
duplicated in both the old and new locations. In a follow-up CL, they
will be deleted from "track".
Bug: webrtc:14521
Change-Id: I0d9036472607a8c717ec823a458a79a49ddb80c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38308}
Now that SDES is (largely) removed, this is no longer useful.
Bug: chromium:1365484
Change-Id: I3e626a7d5d83130a70958851de3df0aa27616bd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38278}
