220 Commits

Author SHA1 Message Date
Emil Lundmark
6c81a42eb1 Simulate generic dependency structure for VP8
This will be used as a fall-back when the encoder adapter doesn't
provide any dependency structure. This ensures we can always generate a
dependency descriptor RTP header extension for VP8.

Before, when switching between encoder adapters where the old one
generated a dependency structure but the new one didn't we had to make
sure the structure was cleared so that packets weren't sent with the
dependency structure from the previous adapter. This will not be a
problem anymore since the new adapter will use the simulated dependency
structure.

Bug: b/227749056
Change-Id: I8463c48a9dcde4b8d32c519819dd8a92acd8e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262765
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36930}
2022-05-19 11:53:08 +00:00
Ali Tofigh
641a1b11b6 Adopt absl::string_view in call/
Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36910}
2022-05-17 12:00:45 +00:00
Jonas Oreland
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
Jonas Oreland
c7f691a71a WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
convert call/ (and the collaterals)

Bug: webrtc:10335
Change-Id: I8f6bc13c032713aa2a947724b464f6f35454d39a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36165}
2022-03-09 22:17:52 +00:00
Tommi
6fba6b795a Reland "(Un/)Subscribe RtpVideoSender for feedback on the transport queue."
This is a reland of 9d230d54c7eef31ac1100f0aeef1374dd1ac62fa

Original change's description:
> (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
>
> * RtpVideoSender now registers/unregisters for feedback callback
>   inside of SetActive(), which runs on the transport queue.
> * Transport feedback is given on the transport queue
> * Registration/unregistration for feedback is done on the same
> * Removed the last mutex from TransportFeedbackDemuxer.
>
> Ultimately, this work is related to moving state from the Call
> class, that's related to network configuration, but due to the code
> is currently written is attached to the worker thread, over to the
> Transport, where it's used (e.g. suspended_video_send_ssrcs_).
>
> Bug: webrtc:13517, webrtc:11993
> Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35777}

Bug: webrtc:13517, webrtc:11993
Change-Id: I766e569abea8bae96d32267a951fcdc195ced8a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249782
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35863}
2022-02-01 06:45:17 +00:00
Tomas Gunnarsson
7734fc64b9 Revert "(Un/)Subscribe RtpVideoSender for feedback on the transport queue."
This reverts commit 9d230d54c7eef31ac1100f0aeef1374dd1ac62fa.

Reason for revert: Speculative revert to see if it's the cause of a few perf changes (some bad, some not so bad).

Bug: webrtc:13613

Original change's description:
> (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
>
> * RtpVideoSender now registers/unregisters for feedback callback
>   inside of SetActive(), which runs on the transport queue.
> * Transport feedback is given on the transport queue
> * Registration/unregistration for feedback is done on the same
> * Removed the last mutex from TransportFeedbackDemuxer.
>
> Ultimately, this work is related to moving state from the Call
> class, that's related to network configuration, but due to the code
> is currently written is attached to the worker thread, over to the
> Transport, where it's used (e.g. suspended_video_send_ssrcs_).
>
> Bug: webrtc:13517, webrtc:11993
> Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35777}

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

Bug: webrtc:13517, webrtc:11993
Change-Id: I824623b3b1c14f0ca7049a2a0890c6d97b7fb608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249600
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35815}
2022-01-27 13:43:44 +00:00
Tommi
9d230d54c7 (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
* RtpVideoSender now registers/unregisters for feedback callback
  inside of SetActive(), which runs on the transport queue.
* Transport feedback is given on the transport queue
* Registration/unregistration for feedback is done on the same
* Removed the last mutex from TransportFeedbackDemuxer.

Ultimately, this work is related to moving state from the Call
class, that's related to network configuration, but due to the code
is currently written is attached to the worker thread, over to the
Transport, where it's used (e.g. suspended_video_send_ssrcs_).

Bug: webrtc:13517, webrtc:11993
Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35777}
2022-01-24 19:50:03 +00:00
Niels Möller
7336422fe3 Delete some unneeded references to ProcessThread.
Bug: None
Change-Id: I77528df2a8bd2d461440cf59ada8229e732a1e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242370
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35613}
2022-01-03 15:36:02 +00:00
Tommi
2d319df955 Add a sequence checker and a few checks to RtpVideoSender.
Moving the following TODO into a bug for tracking.
  // TODO(holmer): Remove mutex_ once RtpVideoSender runs on the
  // transport task queue.

Bug: webrtc:13517
Change-Id: Ie3deb1276c2edaf9894001501ce79409f5437dd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242368
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35612}
2022-01-03 15:32:33 +00:00
Tommi
8695282243 Remove unnecessary copy of suspended_ssrcs.
Also removing pass-by-value in ctor.

Bug: none
Change-Id: I09e36fd955c8f306c4a347d8befc6eea38384cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239183
Auto-Submit: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35427}
2021-11-29 10:54:30 +00:00
Emil Lundmark
823ba0b038 Cleanup WebRTC-Vp9DependencyDescriptor field trial
Bug: chromium:1178444
Change-Id: Ie2ec796e207fa427fdbe00c8ea41a6b4fefea155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235374
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35241}
2021-10-19 14:29:29 +00:00
Emil Lundmark
b01e6457fe Reland "Reland "Enable WebRTC-Vp9DependencyDescriptor by default""
This is a reland of b062829311bf1962a7f264cecf36d17ef41951df

> Reland "Enable WebRTC-Vp9DependencyDescriptor by default"
>
> This is a reland of 472707150662bc4e174072e445938e5c405aa884
>
> Original change's description:
> > Enable WebRTC-Vp9DependencyDescriptor by default
> >
> > Bug: chromium:1178444
> > Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34584}
>
> Bug: chromium:1178444
> Change-Id: I874412b41e657179be6ffbe399617e18a29ec804
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230121
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34890}

Bug: chromium:1178444
Change-Id: I5bb3e3bd2da26f9a24d5e8161bd66b447543fc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231843
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35004}
2021-09-15 13:48:58 +00:00
Emil Lundmark
5498676edd Revert "Reland "Enable WebRTC-Vp9DependencyDescriptor by default""
This reverts commit b062829311bf1962a7f264cecf36d17ef41951df.

Reason for revert: Still causes crashes in perf tests.

Original change's description:
> Reland "Enable WebRTC-Vp9DependencyDescriptor by default"
>
> This is a reland of 472707150662bc4e174072e445938e5c405aa884
>
> Original change's description:
> > Enable WebRTC-Vp9DependencyDescriptor by default
> >
> > Bug: chromium:1178444
> > Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34584}
>
> Bug: chromium:1178444
> Change-Id: I874412b41e657179be6ffbe399617e18a29ec804
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230121
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34890}

Bug: chromium:1178444
Change-Id: I8a789ee60d0cca6db72612ef3660fe595255c537
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231221
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34928}
2021-09-06 13:11:51 +00:00
philipel
5b231de486 Make RtpPayloadParams::MinimalisticVp9Structure codec agnostic.
Bug: none
Change-Id: I97f603aad53933b09c761da954130b06ea5a5501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230760
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34894}
2021-09-01 14:15:59 +00:00
Emil Lundmark
b062829311 Reland "Enable WebRTC-Vp9DependencyDescriptor by default"
This is a reland of 472707150662bc4e174072e445938e5c405aa884

Original change's description:
> Enable WebRTC-Vp9DependencyDescriptor by default
>
> Bug: chromium:1178444
> Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34584}

Bug: chromium:1178444
Change-Id: I874412b41e657179be6ffbe399617e18a29ec804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34890}
2021-09-01 09:27:39 +00:00
Erik Språng
ac09f0dc92 Remove last traces of deferred sequencing.
Bug: webrtc:11340
Change-Id: I761be67d42959192355f9f6f54ed1f735da1fe96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228646
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34770}
2021-08-16 12:44:37 +00:00
Erik Språng
b6bbdeb24d Allow RTP module thread checking to know PacketRouter status.
Since https://webrtc-review.googlesource.com/c/src/+/228433 both audio
and video now only call Get/SetRtpState while not registered to the
packet router.

We can thus remove the lock around packet sequencer and just use a
thread checker.

Bug: webrtc:11340
Change-Id: Ie6865cc96c36208700c31a75747ff4dd992ce68d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228435
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34755}
2021-08-13 15:04:49 +00:00
Erik Språng
51ff855f05 Default-enable deferred sequence numbering for video.
Bug: webrtc:11340, webrtc:12470
Change-Id: I1a2a2a33be8720ecb7dbaf4ce8466c5257e8e0ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227774
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34663}
2021-08-06 15:01:38 +00:00
Björn Terelius
53adc7b1c8 Revert "Enable WebRTC-Vp9DependencyDescriptor by default"
This reverts commit 472707150662bc4e174072e445938e5c405aa884.

Reason for revert: Suspected cause for crashes in perf tests.

Original change's description:
> Enable WebRTC-Vp9DependencyDescriptor by default
>
> Bug: chromium:1178444
> Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34584}

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

Bug: chromium:1178444
Change-Id: I582d6d1c9d2091ca37b0943235b5cea8d4e2790d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227282
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34619}
2021-08-02 09:52:24 +00:00
Emil Lundmark
4727071506 Enable WebRTC-Vp9DependencyDescriptor by default
Bug: chromium:1178444
Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34584}
2021-07-28 12:08:36 +00:00
Artem Titov
ea24027e83 Use backticks not vertical bars to denote variables in comments for /call
Bug: webrtc:12338
Change-Id: I8f92127b61352bd4b98a0690e9a0435bb6c6f870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226943
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34569}
2021-07-27 18:29:33 +00:00
Markus Handell
eb61b7f620 ModuleRtcRtcpImpl2: remove Module inheritance.
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}
2021-06-22 14:51:04 +00:00
Erik Språng
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
Erik Språng
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
Erik Språng
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
Markus Handell
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
Danil Chapovalov
f01c2c96f2 Delete RtcpStatisticsCallback in favor of ReportBlockDataObserver
Bug: webrtc:10678
Change-Id: Ie016cbc47dbba15176fc5e7ad7d01a438db7dfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218842
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34013}
2021-05-16 15:09:29 +00:00
Danil Chapovalov
748550d6ce Provide FrameDependecyStructure for VP9 when encoder doesn't fill it
To produce suboptimal but valid DependencyDescriptor rtp header extension.

Bug: webrtc:11999
Change-Id: I78fbca4d750cc1316b1e4086fc7ae1ad90ce25f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216328
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33873}
2021-04-29 12:28:46 +00:00
Erik Språng
c96aa30c13 Minor refactoring of RtpVideoSender
Bug: None
Change-Id: I7cc16d4e435974090310bd3a7cea9eaf0586be8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208761
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33351}
2021-02-26 13:20:52 +00:00
Erik Språng
61d1773c03 Remove deactivated RTP modules from PacketRouter map.
Apart from making the map smaller, a purpose of this is guaranteeing
that if a module has been deactived it will not receive new packets
from the pacer, which will be needed for deferred sequencing.

Bug: webrtc:11340
Change-Id: I171a13413c5b8d3fa569c2d56bd9a54bff7c7976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208542
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33335}
2021-02-24 20:09:13 +00:00
Danil Chapovalov
37dfddd5e8 Avoid treating VP8 key frame in simulcast as delta frame
Bug: None
Change-Id: Ief3c759571f02af6cd9517acb2851861e190ceaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203891
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33088}
2021-01-28 12:11:50 +00:00
Danil Chapovalov
0be1846477 Fix enabling DependencyDescriptor for VP9 with spatial layers
DependencyDescriptor and vp9 wrapper understand key frame differently
when it comes to the first layer frame with spatial_id>0
This CL adds and use DD's interpretation of the key frame when deciding
if DD should be supported going forward.

Bug: webrtc:11999
Change-Id: I11a809a315e18bd856bb391576c6ea1f427e33be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202760
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33046}
2021-01-20 15:03:37 +00:00
Niels Moller
2accc7d6e0 Revert "Add task queue to RtpRtcpInterface::Configuration."
This reverts commit f23e2144e86400e2d68097345d4b3dc7a4b7f8a4.

Reason for revert: Need further discussion on appropriate thread/tq requirements.

Original change's description:
> Add task queue to RtpRtcpInterface::Configuration.
>
> Let ModuleRtpRtcpImpl2 use the configured value instead of
> TaskQueueBase::Current().
>
> Intention is to allow construction of RtpRtcpImpl2 on any thread.
> If a task queue is provided (required for periodic rtt updates), the
> destruction of the object must be done on that same task queue.
>
> Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
>
> Bug: None
> Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32949}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,saza@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: I7e5007f524a39a6552973ec9744cd04c13162432
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201420
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32953}
2021-01-12 17:47:32 +00:00
Niels Möller
f23e2144e8 Add task queue to RtpRtcpInterface::Configuration.
Let ModuleRtpRtcpImpl2 use the configured value instead of
TaskQueueBase::Current().

Intention is to allow construction of RtpRtcpImpl2 on any thread.
If a task queue is provided (required for periodic rtt updates), the
destruction of the object must be done on that same task queue.

Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.

Bug: None
Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32949}
2021-01-12 12:42:58 +00:00
Jakob Ivarsson
9a12ee5d37 Use frame rate in video overhead calculation.
Assume bitrate is evenly distributed between frames. This is wrong for
uneven frame sizes and will underestimate the overhead for simulcast.
However, it will be more correct than the current calculation,
especially for low bitrates when each frame is smaller than one packet.
This is also when overhead matters more since it is a larger fraction
of the total bitrate.

It is also unclear what will happen when using FEC.

Bug: b/166341943
Change-Id: I247b9d0fc7a8ad5daa9b577f55ec16c56efa34c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195221
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32725}
2020-12-01 09:42:40 +00:00
Jakob Ivarsson
36274f9158 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This is a reland of 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

Bug: webrtc:6762
Change-Id: I6d79894a213fc42d2338409e7513247725881b1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32534}
2020-11-02 11:05:56 +00:00
Björn Terelius
d546186b89 Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This reverts commit 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6.

Reason for revert: Speculative revert due to failing tests.

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
2020-10-27 10:51:46 +00:00
Jakob Ivarsson
1dbe30c7e8 Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

Bug: webrtc:6762
Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32492}
2020-10-26 12:35:47 +00:00
Jakob Ivarsson
27af3c4c24 Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
This reverts commit 87c1950841c3f5e465e1663cc922717ce191e192.

Reason for revert: breaks downstream tests

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
Jakob Ivarsson
87c1950841 Default enable WebRTC-SendSideBwe-WithOverhead.
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
Per Kjellander
af70418357 Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
Bug: webrtc:12000
Change-Id: Ieddbad8e6f4e7456441153d432f4dfb32e16d255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188627
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32439}
2020-10-19 11:37:23 +00:00
Erik Språng
b6477858ac Cleans up code related to legacy pre-pacing fec generation.
Bug: webrtc:11340
Change-Id: If3493db9fafdd3ad041f78999e304c8714be517f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186562
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32349}
2020-10-08 09:05:29 +00:00
Erik Språng
3c65a2b5d4 Cleans up code related to video packetization overhead.
Bug: webrtc:10155
Change-Id: Idc3aaa916e38a18e5ca9dc861587e96465915c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186660
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32312}
2020-10-05 13:22:33 +00:00
Erik Språng
ab89d02b05 Cleans up code related to early loss detection flag.
Bug: webrtc:10676
Change-Id: I32dab4d4cfc4e8e09c3611b1484fb4394b7efbad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186563
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32295}
2020-10-02 15:37:39 +00:00
Erik Språng
10ea118410 Default-enables WebRTC-DeferredFecGeneration.
Bug: webrtc:11340
Change-Id: I9575fcf2ac12ce9b71b27f32deeb7870a1dff64b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185814
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32232}
2020-09-29 13:50:54 +00:00
Danil Chapovalov
383f2cfca4 Stop mentioning RTPFragmentationHeader in call/
Bug: webrtc:6471
Change-Id: I07ab95524369fa996b8dde68f421281989d04e0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181461
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31921}
2020-08-12 13:59:38 +00:00
Danil Chapovalov
31cb3abd36 Do not propage RTPFragmentationHeader into rtp_rtcp
It is not longer needed by the rtp_rtcp module.

Bug: webrtc:6471
Change-Id: I89a4374a50c54a02e9f20a5ce789eac308aaffeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179523
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31773}
2020-07-21 14:37:08 +00:00
Markus Handell
8fe932a5a3 Migrate call/ to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: Iab7142c77bc0c1a026cf5121b756094e05bccefe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176742
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31636}
2020-07-06 15:48:30 +00:00
Erik Språng
1d50cb61d8 Reland "Reland "Allows FEC generation after pacer step.""
This is a reland of 19df870d924662e3b6efb86078d31a8e086b38b5
Patchset 1 is the original.
Subsequent patchset changes threadchecker that crashed with downstream
code.

Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}

Bug: webrtc:11340
Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31619}
2020-07-03 07:20:06 +00:00
Erik Språng
a1888ae791 Revert "Reland "Allows FEC generation after pacer step.""
This reverts commit 19df870d924662e3b6efb86078d31a8e086b38b5.

Reason for revert: Downstream project failure

Original change's description:
> Reland "Allows FEC generation after pacer step."
> 
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
> 
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
> 
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
> 
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: I3b2b25898ce88b64c2322f68ef83f9f86ac2edb0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178563
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31614}
2020-07-02 12:03:07 +00:00