Commit Graph

4983 Commits

Author SHA1 Message Date
3f7b7170cc RTCPSender: remove compatibility ctor & method.
This change removes compatibility APIs in RTCPSender now
that downstream consumers updated.

Bug: webrtc:11581, webrtc:6458
Change-Id: I82d70f1ab6b522b3884480b0b16cbdff9a1490c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222323
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34356}
2021-06-22 13:30:20 +00:00
885d538cdd ModuleRtpRtcpImpl2: remove RTCP send polling.
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}
2021-06-22 07:49:05 +00:00
049ed447b0 ModuleRtpRtcpImpl2: update test code.
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}
2021-06-21 23:36:49 +00:00
c6b9ac782a RTCPSender: migrate to Timestamp.
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}
2021-06-21 22:26:34 +00:00
2e3edc1da9 RTCPSender: migrate to own configuration struct.
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}
2021-06-21 20:23:01 +00:00
f906ec40d4 Handle null return from ToI420 in encoders
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}
2021-06-21 12:45:11 +00:00
7c719b0db1 Fixes off-by-one error in video capture module
Fixed: webrtc:11290
Change-Id: I471b409c27d6ee577a0ed84e3a09d31fbbc16fcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222609
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34333}
2021-06-18 14:07:28 +00:00
c6d76489e3 Add jakobi to modules/audio_coding OWNERS
Bug: None
Change-Id: I299f38126dc1bb419448dcf6f61d3d0323e33885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223040
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34331}
2021-06-18 11:52:58 +00:00
42dacda82c AGC analog clipping predictor: integrate evaluator
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}
2021-06-17 16:16:53 +00:00
7d5418233d Avoid assembling complicated but unused video rtp header extensions
Bug: chromium:1219407
Change-Id: I017de10813a1e80f4af0ba55d8d1aa73077dd131
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222615
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34326}
2021-06-17 16:09:13 +00:00
ac82bd386a Add timestamp to log message in generic_decoder.cc
Bug: None
Change-Id: Ib558247d887aff880853ef824f8d80d8e7e4feee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222610
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34319}
2021-06-17 10:14:14 +00:00
b4100ad06a Avoid using legacy rtp parser in neteq test::Packet
Bug: None
Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-17 08:38:14 +00:00
35b21ba8d4 In RtcpTransceiver avoid extra PostTask during construction
it is not required because during construction members can be set on
wrong thread, and in some corner cases it may even cause a crash.

Bug: none
Change-Id: I37d7f2a7772b6ab5e574077d3f53bca2529f9ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222651
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34315}
2021-06-17 08:36:34 +00:00
a3796c8090 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap.
The behavior was changed on https://webrtc-review.googlesource.com/c/src/+/219696. The revert is due to unknown implications for a downstream project. As REMB caps are not used with send-side bandwidth estimation it should be a noop.

Bug: webrtc:12306
Change-Id: Idecc49fda007f72512a8fc1e35d62e673b00df3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222607
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34313}
2021-06-17 07:44:02 +00:00
355c47309d Fix VideoRtpDepacketizerVp{8,9} copy assignment signature.
Bug: none
Change-Id: I4adca8b4cbf4ffa15172fabc1eaba8c2b65c6fb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222650
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34306}
2021-06-16 17:09:05 +00:00
5b9d0c70c2 AGC1 add clipping predictor evaluator
Observes clipping predictions and detections and computes evaluation
metrics for the predictor.

Bug: webrtc:12774
Change-Id: I83f5942a3b6491de288510f2200f2f5c0e099bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221619
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34305}
2021-06-16 15:33:51 +00:00
98ff0280ce AGC analog ClippingPredictor refactoring 2/2
Uunit test code readability improvements.

Bug: webrtc:12774
Change-Id: I66f552d23680ddb03824618dab869946e0940334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34300}
2021-06-16 10:28:57 +00:00
08be9baaa3 Don't recreate the audio receive stream when updating the local_ssrc.
Bug: webrtc:11993
Change-Id: Ic5d8a8a8b7c12fb1d906e0b3cbdf657fd9e8eafc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34299}
2021-06-16 10:03:31 +00:00
6a0a55907b Reland "Correctly handle retransmissions/padding in early loss detection."
This is a reland of e9ae4729e03f60dbe3b1828dd9009b401097cd3f

TBR=philipel@webrtc.org,terelius@webrtc.org

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

Bug: webrtc:12713
Change-Id: Iec123d71edafea98fe289acde007b57e212681f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34297}
2021-06-16 08:14:27 +00:00
d6957c2eed Revert "Correctly handle retransmissions/padding in early loss detection."
This reverts commit e9ae4729e03f60dbe3b1828dd9009b401097cd3f.

Reason for revert: Internal test failure

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

TBR=danilchap@webrtc.org,terelius@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34294}
2021-06-15 15:59:10 +00:00
e9ae4729e0 Correctly handle retransmissions/padding in early loss detection.
This CL makes sure we don't cull packets from the history based on
incorrect ack mapping, just like it's predecessor:
https://webrtc-review.googlesource.com/c/src/+/218000

It also changes the logic to make sure retransmits counts towards
history pruning - and properly ignores padding/fec.

Bug: webrtc:12713
Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34293}
2021-06-15 15:39:19 +00:00
be53049555 Reland "Avoid sending empty receiver reports with RtcpTransceiver"
This reverts commit 48420fa947cea4c618d51dc5f87908765a3a69db.

Reason for revert: downstream unittests adjusted

Original change's description:
> Revert "Avoid sending empty receiver reports with RtcpTransceiver"
>
> This reverts commit e5f1a3992e3bbfa0445b90f317576c8229524d74.
>
> Reason for revert: Speculative revert due to failing downstream unittest.
>
> Original change's description:
> > Avoid sending empty receiver reports with RtcpTransceiver
> >
> > Bug: None
> > Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
> > Reviewed-by: Per Kjellander <perkj@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34281}
>
> TBR=danilchap@webrtc.org,perkj@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I895317ad0381756e97e501a36d6440f83a68b6f8
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222440
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34284}

# Not skipping CQ checks because this is a reland.

Bug: None
Change-Id: I3481b9b12ddabaef7303ba80e9cd885930988caa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34291}
2021-06-15 12:57:56 +00:00
d350006b70 Add rtp_config() accessor to ReceiveStream.
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.

Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34285}
2021-06-14 17:57:57 +00:00
48420fa947 Revert "Avoid sending empty receiver reports with RtcpTransceiver"
This reverts commit e5f1a3992e3bbfa0445b90f317576c8229524d74.

Reason for revert: Speculative revert due to failing downstream unittest.

Original change's description:
> Avoid sending empty receiver reports with RtcpTransceiver
>
> Bug: None
> Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34281}

TBR=danilchap@webrtc.org,perkj@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I895317ad0381756e97e501a36d6440f83a68b6f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222440
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34284}
2021-06-14 17:29:09 +00:00
e5f1a3992e Avoid sending empty receiver reports with RtcpTransceiver
Bug: None
Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34281}
2021-06-14 16:19:47 +00:00
b237a87a25 AGC analog ClippingPredictor refactoring 1/2
- ClippingPredictor API and docstring changes
- Unified ClippingPredictor factory function

Bug: webrtc:12774
Change-Id: Iafaddae52addc00eb790ac165bf407a4bdd1cb52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221540
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34279}
2021-06-14 12:21:31 +00:00
a63d152423 AEC3: Unbounded echo spectrum for dominant nearend detection.
The dominant nearend detector uses the residual echo spectrum for
determining whether in nearend state. The residual echo spectrum in
computed using the ERLE. To reduce the risk of echo leaks in the
suppressor, the ERLE is capped. While minimizing echo leaks, the
capping of the ERLE can affect the dominant nearend classification
negatively as the residual echo spectrum is often over estimated.

This change enables the dominant nearend detector to use a residual
echo spectrum computed with a virtually non-capped ERLE. This ERLE
is only used for dominant nearend detection and leads to increased
transparency.

The feature is currently disabled by default and can be enabled
with the field trial "WebRTC-Aec3UseUnboundedEchoSpectrum".

Bug: webrtc:12870
Change-Id: Icb675c6f5d42ab9286e623b5fb38424d5c9cbee4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221920
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34270}
2021-06-11 13:30:00 +00:00
3cc68ec32e Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
This is a change from the previous 100Hz frequency.
Also changing the  locks slightly in AcmReceiver so that grabbing the
neteq lock right after we've let it go, isn't necessary inside of
AcmReceiver::GetAudio and also to avoid grabbing the neteq lock while
holding the AcmReceiver lock.

Bug: webrtc:12868
Change-Id: If6ee35f3dca20eb5bdbc615123aa099ccecf57c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221371
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34258}
2021-06-09 18:41:47 +00:00
58126f92bf Update the only 3 remaining kFilterBilinear to kFilterBox.
Bilinear is faster but lesser quality, box is best quality. Our code
base has disagreed about which filter to use for quite some time,
causing aliasing bug reports. In an effort to avoid aliasing artifacts
and make our scaling filters more predictable, we're updating all uses
to kFilterBox.

WebRTC already uses kFilterBox everywhere except for these three
places. The main discrepency was between Chromium and WebRTC but that
has already been fixed. This CL fixes the last remaining bilinears.

This brings the WebRTC kFilterBox use count up from 11 to 14 and the
kFilterBilinear use count down from 3 to 0.

Bug: chromium:1212630
Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
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/master@{#34248}
2021-06-08 13:19:23 +00:00
1a778a24ba Avoid using legacy rtp header parser in the rtp_to_text tool
Bug: None
Change-Id: I4c0ab1ba7730bdcdd826aa41b67b80a96d92c8f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221204
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34231}
2021-06-04 16:41:23 +00:00
1050fbca91 Remove synchronization from VideoSendStream construction.
* Make VideoSendStream and VideoSendStreamImpl construction non-blocking.
* Move ownership of the rtp video sender to VideoSendStream.
* Most state is constructed in initializer lists.
* More state is now const (including VideoSendStreamImpl ptr)
* Adding thread checks to classes that appear to have had a race before
  E.g. RtpTransportControllerSend. The change in threading now actually
  fixes an issue we weren't aware of.
* Moved from using weak_ptr to safety flag and made some PostTask calls
  cancellable that could potentially have been problematic. Initalizing
  the flag without thread synchronization is also simpler.

This should speed up renegotiation significantly when there are
multiple channels. A follow-up change will improve SetSend as well
which is another costly step during renegotiation.

Bug: webrtc:12840
Change-Id: If4b28da5a085643ce132c7cfcf80a62cd1a625c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221105
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34224}
2021-06-03 19:13:45 +00:00
52c7fd6be5 Modernize style in RemoteBitrateEstimatorAbsSendTime implementation
Use dedicated DataSize/DataRate/Time classes instead plain integers
this avoid subtle overflows and makes code easier to follow.

Hide helper structs Probe and Cluster as private structs.
User foreach loops where possible.
Make private constants constexpr instead of using enum hack

Bug: None
Change-Id: I3e71dc1254d7ff8ce71e051de53f0459bfa5264d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219795
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34222}
2021-06-03 14:37:33 +00:00
096014345f Add a function to check if the packet in a PacketResult has been received.
Bug: webrtc:12839
Change-Id: I0ee2b8fa0dfffd2bda2cba0e360b5f5815bbca9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221102
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34220}
2021-06-03 12:42:49 +00:00
47f5f8c160 Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket
As a step to delete the legacy rtp packet parser.

Bug: None
Change-Id: I2aae86bc8847acd76cdd89007273a99f0298fdb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221109
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34219}
2021-06-03 12:29:09 +00:00
943e2e6a57 Revert "Fix incorrect SSRC in RtpPacketSendInfo for RTX packets."
This reverts commit 82aa094a970a2c37634378910116bbe1d5abc633.

Reason for revert: Causes regression for an upstream project

Original change's description:
> Fix incorrect SSRC in RtpPacketSendInfo for RTX packets.
>
> Bug: webrtc:12713
> Change-Id: I1b5fb947ffe4ac80e23a6b891ea1a2c2156ba81f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218000
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34177}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12713
Change-Id: I20facf724bdb0136e7eb079c4834575184764174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221202
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34218}
2021-06-03 11:59:39 +00:00
f3ff3c5b77 Reinstate killswitch for WebRTC-Bwe-ReceiverLimitCapsOnly.
Bug: webrtc:12306
Change-Id: Idd643c3152252732562553f207d0a6335773e98a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221043
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34211}
2021-06-03 09:15:58 +00:00
a004715d13 Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect
Integrate ClippingPredictor in AgcManagerDirect and
AudioProcessingImpl. Disable functionality by default.

Bug: webrtc:12774
Change-Id: Ic67a47f439c89b75066506fca8acaf636d8812f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221100
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34207}
2021-06-03 02:35:05 +00:00
4b3a06139b Add ClippingPredictor implementation
Add implementation for clipping prediction and clipped level step estimation.

Bug: webrtc:12774
Change-Id: I855d22980302aac7d49078ca29755f9422af9cb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220935
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34206}
2021-06-02 22:45:46 +00:00
a43953a518 Add ClippingPredictor config in AudioProcessing config
Bug: webrtc:12774
Change-Id: Id8cdb6b5499a22cbca40d424cf936f81c1e7d8d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221104
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34204}
2021-06-02 16:16:25 +00:00
cbdbb8c166 Add ability to adjust the suppressor smoothing in AEC3
Bug: b/177359044
Change-Id: I5eddb6fa6f01aa14426161204e37a9097b182234
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217889
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34203}
2021-06-02 15:21:35 +00:00
fccb052ee3 Add event traces to interesting places in WebRTC.
Bug: webrtc:12840
Change-Id: I2fe749039059c9f3d6da064dce10d9c24a27d02e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34199}
2021-06-02 13:06:04 +00:00
486b0401c5 Make VP8 DefaultTemporalLayers always report TL count even with no rate.
If at creation of a VP8 encoder there is not enough bitrate to enable a
given spatial layer - the configuration won't be updated to indicate
the correct temporal layer count. This means GetEncoderInfo() will
indicate lack of temporal layer support, which triggers issues with
rate allocation.

This CL fixes that by always setting an initial bitrate of 0bps.

Bug: webrtc:12788
Change-Id: I10974e85446b58e597d2ca415eaf2550306ce986
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220929
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34198}
2021-06-02 10:35:07 +00:00
78c73477c7 Add DesktopCaptureOption enumerate_current_process_windows to avoid hang
Enumerating windows owned by the current process on Windows has some
complications due to the GetWindowText*() APIs potentially causing a
deadlock. The APIs will send messages to the window's message loop, and
if the message loop is waiting on this operation we will enter a
deadlock.

I previously put in a mitigation for this [1] which brought the
incidence rate down by an order of magnitude, but we are still seeing
this issue fairly frequently.

So, I've added  DesktopCaptureOption enumerate_current_process_windows
which allows consumers to avoid this issue completely by ignoring
these potentially problematic windows.

By default the flag is set to true which equates with the current
behavior, consumers can set the flag to false to get the new behavior.

I've also updated all the capturers that enumerate windows on Windows
to respect the option.

[1] https://webrtc-review.googlesource.com/c/src/+/195365

Bug: chromium:1152841
Change-Id: I0e0d868957d6fbe1e607a440b3a909d005c93ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219380
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34191}
2021-06-01 18:20:50 +00:00
f865444877 Make AV1 respect spatial layer active flag.
Bug: webrtc:12788
Change-Id: Ied629e1635b6ff9bf92fab2d1af708163f9dd28c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220928
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34189}
2021-06-01 16:07:25 +00:00
82aa094a97 Fix incorrect SSRC in RtpPacketSendInfo for RTX packets.
Bug: webrtc:12713
Change-Id: I1b5fb947ffe4ac80e23a6b891ea1a2c2156ba81f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218000
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34177}
2021-05-31 20:51:07 +00:00
ea72ee6350 Add ClippingPredictorLevelBuffer circular buffer.
Bug: webrtc:12774
Change-Id: I2b26660e3fe051ab358dd5298ba5098f275943da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219631
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34167}
2021-05-31 14:04:54 +00:00
e52cfab633 PipeWire capturer: request mouse cursor to be part of the stream
We need to specify that the cursor should be included in the stream as
by default xdg-desktop-portal defaults to hidden cursor.

Bug: chromium:1202526
Change-Id: Ic4742da2e51f7ed28cb9d7b6b0c069c1fa7d0cee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214782
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34137}
2021-05-26 19:08:17 +00:00
2182096e66 RtpFrameReferenceFinder return frames directly instead of via callback.
Bug: webrtc:12579
Change-Id: I41263f70a6f3dc60167e41f8b015a7d3b0dc3dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219633
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34136}
2021-05-26 15:47:03 +00:00
7f11067110 Clean up RtpSenderTest and remove RtpSenderEgress dependencies.
Since all test cases that used RtpSenderEgress have been refactored or
moved, we can now get rid of lot of test fixture crud:
* Remove RtpSenderContext helper, make sender normal member.
* Remove test transport helper
* Remove task queue helper (needed for thread checks in egress)
* Remove various mocks no longer used
* Remove RtpSenderWithoutPacer subclass
* Remove WithWithoutOverhead parametrization (only affect egress)

..plus some cleanup of how configs are created.

Bug: webrtc:11340
Change-Id: I5c581d60862fc6dc2b99f76058782309dc7aef4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220280
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34135}
2021-05-26 15:25:58 +00:00
8f8bf252e6 Remove usage of InjectPacket and transport_ in rtp_sender_unittest
Thus removing dependency on RtpSenderEgress, allowing simplification of
the test fixture in a follow-up.

Bug: webrtc:11340
Change-Id: I9772bab18d1f4a04e0deccc9125d4b1c16c30d7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219627
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34132}
2021-05-26 10:44:29 +00:00