This change aims to fix a flake that could have been caused by
threads lingering past the main thread lifetime.
Bug: webrtc:12913
Change-Id: I144078b41d59f46398212f48cdd4b0cf3f204bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224080
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34371}
Relanding after adding to chromium stats whitelist:
https://chromium-review.googlesource.com/c/chromium/src/+/2983329
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I3633ee428d07b283b0cc503a84d8fa2e79415dfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34367}
The Cl landing the alternative was dated 2018, and an (informal)
deprecation notice was landed. The users are now gone.
Bug: None
Change-Id: I1f49143fe5f2c236bf9574469f2573593b7a8086
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223663
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34365}
If timestamp_scaler_ is used, then rtp_header.timestamp, passed to UpdateLastDecodedPacket, will advance at a different rate than the scaled timestamp packet->timestamp, passed to UpdateLastDecodedPacket.
NackTracker::EstimateTimestamp uses timestamp_last_received_rtp_, and NackTracker::TimeToPlay uses timestamp_last_decoded_rtp_.
This difference causes TimeToPlay to continuously increase to huge values, so that every missing packet will be returned from GetNackList, even if RTT > real time to play.
Change-Id: Ie6ca347972edea98a202c9cdd26c6ab3f45a73c4
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222841
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34361}
This change achieves an Idle Wakeup savings of 200 Hz.
ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.
Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
TaskQueueStdlib initialized it's thread too early which
permitted it to access uninitialized attributes.
Also remove the |stopped_| event which isn't needed because of
the platform thread being joinable.
Fixed: webrtc:12876
Change-Id: Ibd27ce915e0e3ac92ebafca535c5a3fd72f9165e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223340
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34355}
Also change listen_queue_ member to use std::unique_ptr to
manage ownership.
Bug: webrtc:11567
Change-Id: I85171c9cd0253fdbcbce38b1cfebb1adb5bddd9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223063
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34353}
This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.
Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.
Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
> need to be taken from the audio processor attached to the track
> rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
> RTCMediaStreamTrackStats. For now, will populate the stats in both
> locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34344}
TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}
* Remove unnecessary decoder factory pointer.
* Set video decoder factory in the ctor of the config class.
* Prepare SetRecvParameters for not needing RecreateWebRtcVideoStream.
Bug: none
Change-Id: I48fbf2920c9fe50f3995ceab5667eb2f70618f25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34351}
This change migrates RTCP send polling happening in
ModuleRtpRtcpImpl2::Process to task queues.
ModuleRtpRtcpImpl2 would previously only cause RTCP sends while being
registered with a ProcessThread. This is now relaxed so that RTCP will
be sent regardless of ProcessThread registration status, and it seems
no tests cared.
Now there's only one piece of polling left in Process.
Bug: webrtc:11581
Change-Id: Ibdcffefccef7363f2089c34a9c7d694d222445c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222603
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34350}
This change prepares for later CLs that partly replaces
logic in the module that depends on the Module system
for logic that depends on task queues.
The change also changes SendTransport::SendRTCP
to schedule packet reception with the simulated time
controller. This fixes the problem that SendRTCP itself
updates the simulated time which makes it hard to
understand the tests.
Finally, GlobalSimulatedTimeController was updated
to support addition of custom SimulatedSequenceRunners
like SendTransport.
Bug: webrtc:11581
Change-Id: I0aa310ad0a10526479ad8c28affc38a413363ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222602
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34348}
This change migrates RTCPSender to use webrtc::Timestamp, preparing
for later improvements regarding bugs.webrtc.org/11581.
Fixed: webrtc:12873
Change-Id: I1159701dc373883367d9b2c86823f8fb59904d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222324
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34346}
This reverts commit a4aabb921353125f6d3a2caa2ceb9cda7e971f22.
Reason for revert: downstream tests fixed.
TBR=hta@webrtc.org
Original change's description:
> Revert "Port: migrate to TaskQueue."
>
> This reverts commit 06540166ca97028454adea48cec9bf109b771ddc.
>
> Reason for revert: breaks downstream test.
>
> Original change's description:
> > Port: migrate to TaskQueue.
> >
> > Port uses legacy rtc::Thread message handling. In order
> > to cancel callbacks it uses rtc::Thread::Clear() which uses locks and
> > necessitates looping through all currently queued (unbounded) messages
> > in the thread. In particular, these Clear calls are common during
> > negotiation and the probability of having a lot of queued messages is
> > high due to a long-running network thread function invoked on the
> > network thread.
> >
> > Fix this by migrating Port to task queues.
> >
> >
> > Bug: webrtc:12840, webrtc:9702
> > Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34338}
>
> TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I014ef9267d224c10595cfa1c12899eabe0093306
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12840, webrtc:9702
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223062
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34339}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12840, webrtc:9702
Change-Id: I4d2e086b686da8d5272d67293406300a07edef81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223260
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34345}
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34344}
The class depends on RtcRtcpInterface::Configuration which adds an
unneeded dependency, and inhibits well-manored changes to the
constructor interface.
Fix this so that RTCPSender uses it's own configuration struct which
can be extended in future CLs.
Also add a legacy constructor while downstream dependencies are
updated.
Bug: webrtc:11581
Change-Id: I8d166ab8253b27c08fcbe6aa7c7adde92688b7dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222322
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34343}
In cases where ToI420 fails it should be able to return null.
Bug: webrtc:12877
Change-Id: Ia13859c104d978a29712ae10f8e15acada8406ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222613
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34342}
Add GUARDED_BY annotation on members claimed to be protected by the
lock, and add missing lock operations. Also mark a few members const.
Bug: webrtc:11567, webrtc:2079
Change-Id: I8f12ca7627df0c24e07fa2ae24a387c6a0ed76cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208224
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34340}
This reverts commit 06540166ca97028454adea48cec9bf109b771ddc.
Reason for revert: breaks downstream test.
Original change's description:
> Port: migrate to TaskQueue.
>
> Port uses legacy rtc::Thread message handling. In order
> to cancel callbacks it uses rtc::Thread::Clear() which uses locks and
> necessitates looping through all currently queued (unbounded) messages
> in the thread. In particular, these Clear calls are common during
> negotiation and the probability of having a lot of queued messages is
> high due to a long-running network thread function invoked on the
> network thread.
>
> Fix this by migrating Port to task queues.
>
>
> Bug: webrtc:12840, webrtc:9702
> Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34338}
TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I014ef9267d224c10595cfa1c12899eabe0093306
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12840, webrtc:9702
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223062
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34339}
Port uses legacy rtc::Thread message handling. In order
to cancel callbacks it uses rtc::Thread::Clear() which uses locks and
necessitates looping through all currently queued (unbounded) messages
in the thread. In particular, these Clear calls are common during
negotiation and the probability of having a lot of queued messages is
high due to a long-running network thread function invoked on the
network thread.
Fix this by migrating Port to task queues.
Bug: webrtc:12840, webrtc:9702
Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34338}
The factory allows us to isolate the implementation from users who only
need to depend directly on the public folder now.
Bug: webrtc:12614
Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34330}
While in the COOKIE ECHO state, there is a TCB and there might be data
in the send buffer, and RFC4960 allows the COOKIE ECHO chunk to bundle
additional DATA chunks in the same packet, but there mustn't be more
than one such packet sent, and that packet must have a COOKIE ECHO chunk
as the first chunk in it.
When the COOKIE ACK chunk has been received, the socket is allowed to
send multiple packets.
Previously, this was state managed by the socket and not the TCB, as
the socket is responsible for moving between the different states. And
when the COOKIE ECHO chunk was sent, the TCB was instructed to only send
a single packet by the socket.
However, if there were retransmissions or anything else that could
result in calling TransmissionControlBlock::SendBufferedChunks, it would
do as instructed and send those, even if the socket was in a state where
that wasn't allowed.
When the peer was dcSCTP, this didn't cause any issues as dcSCTP tries
to be tolerant in what it receives (but strict in what it sends, except
for when there are bugs). When the peer was usrsctp, it would send an
ABORT for each received packet that didn't have a COOKIE ECHO as the
first chunk, and then restart the handshake (sending an INIT). So this
resulted in a longer handshake, but the connection would eventually be
correctly established and any DATA chunks that resulted in the ABORTs
would've been retransmitted.
By making the TCB aware of that particular state, and to make it
responsible for creating the SCTP packet with the COOKIE ECHO chunk
first, and also to only send a single packet when it is in that state,
there will not be any way to bypass this limitation.
Also, while not explicitly mentioned in the RFC, the retransmission
timer will not affect resending any outstanding DATA chunks that were
bundled together with the COOKIE ECHO chunk, as then there would be two
timers that both would drive resending COOKIE ECHO and DATA chunks.
Bug: webrtc:12880
Change-Id: I76f215a03cceab5bafe9f16eb4775f3dc68a6f05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34329}
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the
possibility to run the predictor without affecting the analog gain
adjustment process.
The evaluator is used to compute precision, recall and F1 score.
F1 score and the measured clipping prediction intervals are logged as
`WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval`
histograms respectively.
Bug: webrtc:12774
Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34327}
