Commit Graph

5424 Commits

Author SHA1 Message Date
865d94e452 Revert "Apply lower bound of delay based estimate in AimdRateControl::ClampBitrate"
This reverts commit e66e6a845b50f212ebb60234446cfc9db897879c.

Reason for revert: 
May cause BWE to increase on delay increase if link capacity estimate is too high.

Original change's description:
> Apply lower bound of delay based estimate in AimdRateControl::ClampBitrate
>
> This move the functionality of applying the lower bound of a network estimate to AimdRateControl::ClampBitrate instead of ChangeBitrate.
> The purpose is to be able to also clamp probe estimates set by AimdRateControl::SetEstimate as well.
>
> Bug: none
> Change-Id: I6a4d64d2e98bb99da06010e2edaf20dc42880e37
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255823
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36219}

Bug: none
Change-Id: I8c65b1461160dbf3d35e50ef2cc6f9bc305c2b15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256011
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36248}
2022-03-18 12:06:47 +00:00
bb986cbddb APM audioproc_f: adding --override_key_pressed
Add a flag to override the key pressed state when simulating APM.

The current behavior changes as follows:
- Wav files simulation: instead of simulating continuous key press
  events only if the transient suppressor (TS) sub-module is active,
  allow to simulate the events regardless of whether TS is used;
  the default key pressed state is used if the command line flag is
  unspecified, otherwise it is overridden (either always false or
  always true)
- AEC dump simulation: instead of simulating continuous key press
  events when `--ts 2` is specified, allow to simulate the events
  regardless of whether TS is used; the state recorded in the AEC
  dump is used if the command line flag is unspecified, otherwise
  it is overridden (either always false or always true)
- The `--ts 2` option (continuous key events) is now equivalent to
  `--ts 1`.

Bug: webrtc:13663
Change-Id: I5ebe96283db73ee235ec2b2795d91d4e241a3527
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256003
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36247}
2022-03-18 10:51:37 +00:00
017a606836 Avoid sending empty receiver reports with RTCPSender
in reduced size mode, i.e. when rtcp-rsize sdp attribute is negotiated

Bug: webrtc:13833
Change-Id: I55fa5248d3f66dc2240d7a6fbbb399319f1a2e03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256004
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36246}
2022-03-18 10:45:27 +00:00
c2cd814cdd PipeWire capturer: check existence of cursor metadata
Check whether there are any cursor metadata before we try to validate
and use them, otherwise we might crash on this.

Bug: webrtc:13429
Change-Id: I365da59a189b6b974cebafc94fec49d5b942efae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255601
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36240}
2022-03-17 18:14:26 +00:00
1a08096998 Clean up the TaskQueuePacedSender constructor.
Removes the unused event log pointer and the default arguments.

Bug: webrtc:13417
Change-Id: I90341528cdfd7a5c102addaa4e7c83e875525386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255562
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36238}
2022-03-17 15:23:46 +00:00
4ceea65848 Integrate trendline estimator into loss based bwe v2.
Bug: webrtc:12707
Change-Id: I510d3799c14599344d1714178e42b29e7c0c06d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254380
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36236}
2022-03-17 13:07:44 +00:00
658b88a48e Delete rtc::string_trim. Replaced with absl::StripAsciiWhitespace.
Bug: webrtc:6424, webrtc:13579
Change-Id: I222e1bfb62d5f1f1a2c74e5fce1038e04e7bebfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255824
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36234}
2022-03-17 12:53:14 +00:00
ae4fb618d7 Cleanup RtpToNtpEstimator
- Use NtpTime instead of pair of uint32_t to represent ntp time
- Increase precision estimate with NtpTime precision instead of ms precision
- Hide helper structs as private types
- Modernize interface to prefer return values over output parameters
- embed LinearRegression helper into the only user: UpdateParameters

Bug: webrtc:13757
Change-Id: I0a62a03e2869b2ae1eacaa15253accc43ba0a598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36232}
2022-03-17 10:26:57 +00:00
a943e730b2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf
Convert audio/ and collateral (audio encoder copy red).

Bug: webrtc:10335
Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36229}
2022-03-17 07:11:44 +00:00
347f9b07b9 getDisplayMedia shows black window on Youtube PiP in Windows.
getDisplayMedia capture the view of the screens and windows
in the capture dialog, but the issue is that captured view
of the Youtube somehow is blank. It repros only in certain
circumstances, for example, Canary channel.
If user reinstall the Canary as fresh new, we observed that
it doesn't repro.

Cause:
We aren't sure what's cause of this one yet.

Solution:
We decided to provide fallback WGC capturer when the main
capturer (GDI) shows blank. WGC could show yellow outline
in prior Win11 OS, but yellow outline looks better than
blank.

The blank detector and fallback capturer are what screen capturer
already supported. So, the solution will follow similar
pattern in the window capturer.

Bug: webrtc:13726
Change-Id: I620c817d259d7bb5c295adab11c4444349ab1c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252625
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36224}
2022-03-16 22:06:04 +00:00
554bb39998 Reland "Experimentally reduce TaskQueuePacedSender's delayed task precision."
This is a reland of commit 43a69b3f46059b103c35079744dc09a0e2bb2948

Original change's description:
> Experimentally reduce TaskQueuePacedSender's delayed task precision.
>
> This CL reduces the delayed task precision of non-probes in accordance
> with DD go/slacked-task-queue-paced-sender. The precision is only
> deduced if field trial "WebRTC-SlackedTaskQueuePacedSender" is enabled
> though.
>
> Bug: webrtc:13824
> Change-Id: I37e53b24e343f4f08059be08a3cda74f5484cc05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255341
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36214}

Bug: webrtc:13824
Change-Id: I86cace6f4f6bf23d51c75b3d18f8d24fff0f5b74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255826
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36221}
2022-03-16 16:14:04 +00:00
6673437775 Move ownership of congestion window state to rtp sender controller.
When congestion window is used, two different mechanisms can currently
update the outstanding data state in the pacer:
* OnPacketSent() withing the pacer itself, when a packet is sent
* UpdateOutstandingData(), when RtpTransportControllerSend either:
  a. Receives an OnPacketSent() callback (increase outstanding data)
  b. Receives transport feedback (decrease outstanding data)

This creates a lot of calls to UpdateOutstandingData(), more than one
per sent packet. Each requires locking and/or thread jumps. To avoid
that, this CL moves the congestion window state to
RtpTransportController send - and we only post a congested flag down
the the pacer when the state is changed.

The only benefit I can see is of the old way is we prevent sending
new packets immedately when the window is full, rather than in some
edge cases queue extra packets on the network task queue before the
congestion signal is received. That should be rare and benign.
I think this simplified logic, which is easier to read and more
performant, is a better tradeoff.

Bug: webrtc:13417
Change-Id: I326dd88db86dc0d6dc685c61920654ac024e57ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255600
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36220}
2022-03-16 15:30:03 +00:00
e66e6a845b Apply lower bound of delay based estimate in AimdRateControl::ClampBitrate
This move the functionality of applying the lower bound of a network estimate to AimdRateControl::ClampBitrate instead of ChangeBitrate.
The purpose is to be able to also clamp probe estimates set by AimdRateControl::SetEstimate as well.

Bug: none
Change-Id: I6a4d64d2e98bb99da06010e2edaf20dc42880e37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255823
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36219}
2022-03-16 14:52:43 +00:00
9524c0fa4f AgcManagerDirect: Modify clipping_predictor_evaluator_ configuration
Increase the history size of clipping_predictor_evaluator_. Use one-sample
accuracy in clipping detection for the evaluator.

Bug: webrtc:12774
Change-Id: I8c1bbfe69fe55af73ce14992e49ef7295b3ce926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241602
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36218}
2022-03-16 14:42:23 +00:00
4a08d5013b Revert "Experimentally reduce TaskQueuePacedSender's delayed task precision."
This reverts commit 43a69b3f46059b103c35079744dc09a0e2bb2948.

Reason for revert: Breaks downstream project.

Original change's description:
> Experimentally reduce TaskQueuePacedSender's delayed task precision.
>
> This CL reduces the delayed task precision of non-probes in accordance
> with DD go/slacked-task-queue-paced-sender. The precision is only
> deduced if field trial "WebRTC-SlackedTaskQueuePacedSender" is enabled
> though.
>
> Bug: webrtc:13824
> Change-Id: I37e53b24e343f4f08059be08a3cda74f5484cc05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255341
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36214}

Bug: webrtc:13824
Change-Id: Iccdadcfaa0c490a1b9e5636cd695c5673c3c09a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255825
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36217}
2022-03-16 14:16:33 +00:00
43a69b3f46 Experimentally reduce TaskQueuePacedSender's delayed task precision.
This CL reduces the delayed task precision of non-probes in accordance
with DD go/slacked-task-queue-paced-sender. The precision is only
deduced if field trial "WebRTC-SlackedTaskQueuePacedSender" is enabled
though.

Bug: webrtc:13824
Change-Id: I37e53b24e343f4f08059be08a3cda74f5484cc05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255341
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36214}
2022-03-16 11:04:43 +00:00
6e2b9e2210 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf
Add field trials to audio api.

It is added as a pointer with nullptr as default.
It is not (yet) used anywhere.
Usage of field trials comes in subsequent patches.

Bug: webrtc:10335
Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521
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@{#36213}
2022-03-16 09:11:43 +00:00
e486a7bdf7 Make KeyValueConfig mandatory in the pacer.
This CL also removes dependency on the legacy field trial methods.

Bug: webrtc:11926
Change-Id: I53feeee86b92878cf0f2b8ebdce3d101f9e04014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255381
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36205}
2022-03-15 15:07:46 +00:00
a2d0a2ad99 Ignore v4l2 select() return when stopping capture
With some slightly broken webcams, it's possible that the select()
returns with a timeout or no event. In that case, the v4l2 thread
never returns. To fix this, just check if quit_ is set and exit
unconditionally in that case.

https://bugzilla.mozilla.org/show_bug.cgi?id=1752326

Bug: None
Change-Id: Ic07ce15afd0016ff9f967c2cf64e646c20127457
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36200}
2022-03-15 08:52:26 +00:00
d8543dedf2 Revert "Pacer: Reduce TQ wake up and improve packet size estimation"
This reverts commit 37195cf2e577cc09ad1362d046b5c8a9b65d4f99.

Reason for revert: Breaks downstream tests (more investigations and testing is necessary).

Original change's description:
> Pacer: Reduce TQ wake up and improve packet size estimation
>
> The TQ Pacer schedules delayed task according to target time of
> PacingController. It drains all valid ProcessPackets() in single loop,
> denies retired scheduled tasks, and round up the timeout to 1ms.
>
> This CL also improves packet size estimation in TQ Pacer by removing
> zero initialization, and introduces `include_overhead_` configuration.
>
> Tests:
> 1. webrtc_perf_tests: MaybeProcessPackets() calls
>   2075147 -> 2007995
>
> 2. module_unittests: MaybeProcessPackets() calls
>   203393 -> 183563
>
> 3. peerconnection_unittests: MaybeProcessPackets() calls
>   66713-> 64333
>
> Bug: webrtc:13417, webrtc:13437
> Change-Id: I18eb0a36dbe063c606b1f27014df74a65ebfc486
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242962
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36179}

No-Try: True
Bug: webrtc:13417, webrtc:13437
Change-Id: I5418d26d3978f21765ef38acfb002398e671e036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255301
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36185}
2022-03-14 09:35:56 +00:00
aaf9d051c7 Lower hd av1 quality threshold
Bug: None
Change-Id: I2b7bfbd8f5a2be13ede11df30272e5b001471453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36180}
2022-03-11 20:54:03 +00:00
37195cf2e5 Pacer: Reduce TQ wake up and improve packet size estimation
The TQ Pacer schedules delayed task according to target time of
PacingController. It drains all valid ProcessPackets() in single loop,
denies retired scheduled tasks, and round up the timeout to 1ms.

This CL also improves packet size estimation in TQ Pacer by removing
zero initialization, and introduces `include_overhead_` configuration.

Tests:
1. webrtc_perf_tests: MaybeProcessPackets() calls
  2075147 -> 2007995

2. module_unittests: MaybeProcessPackets() calls
  203393 -> 183563

3. peerconnection_unittests: MaybeProcessPackets() calls
  66713-> 64333

Bug: webrtc:13417, webrtc:13437
Change-Id: I18eb0a36dbe063c606b1f27014df74a65ebfc486
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242962
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36179}
2022-03-11 14:17:33 +00:00
a5f3018c24 [DesktopCapture][WGC] Avoid artifacts when capture source is resized
This CL fixes the issue where artifacts appear during capture with WGC
when the capture source is resized. A video of the issue is available
here: https://bugs.chromium.org/p/webrtc/issues/detail?id=9273#c44

The solution is to use CopySubresourceRegion instead of CopyResource to
only copy valid data into our texture. Additionally, we moved the call
to CreateMappedTexture to before the call to CopySubresourceRegion, as
the latter requires both textures to be of the same size.

Bug: webrtc:9273
Change-Id: I114458d95cbf58550ff653a985dd84db4741e0f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254100
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36163}
2022-03-09 17:14:42 +00:00
878c0299b3 flexfec: increase verbosity of logging
- add recovered sequence number and length of the recovered packet
- increase level of periodic logging to LS_INFO
- log for every packet on LS_VERBOSE

This makes it easier to validate and debug flexfec implementations.

BUG=None

Change-Id: I6f9e73e72ec3dcc0531f7adc62ac7019c7899270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36162}
2022-03-09 14:39:42 +00:00
ed99dae422 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.

Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 13:23:21 +00:00
fef0026f2f Revert "Reland "remove NV12 to I420 conversion in webrtc AV1 Encoder.""
This reverts commit d7031692e3ba9eed78ead07f8bf34a847ca1fce6.

Reason for revert: Breaks downstream project

Original change's description:
> Reland "remove NV12 to I420 conversion in webrtc AV1 Encoder."
>
> This reverts commit 66557e1af3f95a70753e782224d13a6186ed0d2e.
>
> Reason for revert: Some downstream projects seem to have an old libaom version with no NV12 support yet. It will be updated soon.
>
> Original change's description:
> > Revert "remove NV12 to I420 conversion in webrtc AV1 Encoder."
> >
> > This reverts commit 9558ab41eb4de39c62cda2dd1e559f5814a3a0c7.
> >
> > Reason for revert: speculative revert: breaks downstream project
> >
> > Original change's description:
> > > remove NV12 to I420 conversion in webrtc AV1 Encoder.
> > >
> > > libaom supports for NV12 inputs for encoding av1 stream. It will reduce
> > > unnecessary conversion from NV12 to I420 format.
> > > (https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)
> > >
> > > Bug: webrtc:13746
> > > Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > > Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
> > > Cr-Commit-Position: refs/heads/main@{#36111}
> >
> > Bug: webrtc:13746
> > Change-Id: Ie928f7f5b5992337a9d186fa70b7fdec20a33f00
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253122
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Owners-Override: Artem Titov <titovartem@webrtc.org>
> > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36114}
>
> Bug: webrtc:13746
> Change-Id: Ib26ff6204abceb863b03d55e5953797c9ca27fc2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253215
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36156}

Bug: webrtc:13746
Change-Id: Ia9f8024bf70a82f8e26cd7a80d3020ed796c1b40
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254262
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36159}
2022-03-09 11:47:54 +00:00
d7031692e3 Reland "remove NV12 to I420 conversion in webrtc AV1 Encoder."
This reverts commit 66557e1af3f95a70753e782224d13a6186ed0d2e.

Reason for revert: Some downstream projects seem to have an old libaom version with no NV12 support yet. It will be updated soon.

Original change's description:
> Revert "remove NV12 to I420 conversion in webrtc AV1 Encoder."
>
> This reverts commit 9558ab41eb4de39c62cda2dd1e559f5814a3a0c7.
>
> Reason for revert: speculative revert: breaks downstream project
>
> Original change's description:
> > remove NV12 to I420 conversion in webrtc AV1 Encoder.
> >
> > libaom supports for NV12 inputs for encoding av1 stream. It will reduce
> > unnecessary conversion from NV12 to I420 format.
> > (https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)
> >
> > Bug: webrtc:13746
> > Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
> > Cr-Commit-Position: refs/heads/main@{#36111}
>
> Bug: webrtc:13746
> Change-Id: Ie928f7f5b5992337a9d186fa70b7fdec20a33f00
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253122
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36114}

Bug: webrtc:13746
Change-Id: Ib26ff6204abceb863b03d55e5953797c9ca27fc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253215
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36156}
2022-03-09 11:15:13 +00:00
4d54260ae2 Field trial to not clamp delay based estimate to a lowered link estimate
This adds a new paramater to WebRTC-Bwe-EstimateBoundedIncrease that ensure that even if the link capacity has decreased, the delay based estimate does not immediately decrease unless an overuse has been detected.
This is a follow up to https://webrtc-review.googlesource.com/c/src/+/252442/

Bug: none
Change-Id: I98d77ba1e3f7856b06f2691575f2d248a500e659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253901
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36154}
2022-03-09 08:33:03 +00:00
f73b524a5a Add perkj@ as owner of remote_bitrate_estimator
And remove srte since they are no longer active.

Bug: none
Change-Id: I259898db1223d43d13b918ece6555c5f687ce23f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254060
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36153}
2022-03-09 07:40:12 +00:00
ddcfe405d3 Change PSNR threshold for av1 test
Bug: None
Change-Id: I47101a6625c2f1704599ea60ad3f2c05370da66e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/main@{#36151}
2022-03-08 22:14:51 +00:00
e9126c18bf Migrate VCMInterFrameDelay to use Time units
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.

Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-08 09:05:12 +00:00
773205dfb2 Save unwrapped tl0_pic_idx for inserted VP9 frames.
As stashed frames are retried their `tl0_pic_idx` are again unwrapped which can lead to the `tl0_unwrapper_` to unwrap the `tl0_pic_idx` of newer frames backwards. Instead unwrap the `tl0_pid_idx` only once and save it with the frame if necessary.

In this CL
  - Only unwrap the TL0 once in ManageFrame.
  - Split ManageFrameInternal into ManageFrameFlexible and ManageFrameGof.
  - Save the unwrapped TL0 with the stashed frame.

Bug: none
Change-Id: I56e6b071c0082682e010c049c537d66060635567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253844
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36146}
2022-03-07 18:36:50 +00:00
10ab697dcb Cleanup legacy functions to handle time as raw int in RtpPacketToSend
Bug: webrtc:13757
Change-Id: I28964cb7dbd6bc6363401a9658208b8f96aceb8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253820
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36145}
2022-03-07 16:49:10 +00:00
11cc804d97 Remove unused variable from RtpVp9RefFinder
Bug: none
Change-Id: Iaa1f2f8272a7e47f50a3572efb2e0765286c8a0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253843
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36144}
2022-03-07 16:12:30 +00:00
13e42a88df Use TimeDelta and Timestamp in VCMJitterEstimator
* Uses DataSize to represent incoming and outgoing bytes.
* Puts units into doubles as they enter the Kalman filter
* Moved to its own GN target.

Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 14:58:22 +00:00
0f50cc2849 Remove checks for SDK <= 21
WebRTC’s minSdk is 21, so all those checks are dead code.

Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
2022-03-07 09:56:42 +00:00
80260c226d Switch VCMRttFilter to use TimeDelta
* Moved into its own GN target
* Switched the internal buffer types to absl::InlinedVector as arrays
  are tricky to use with types that do not have default constructors.
* Update fields arnd variables to use style guide.
* Use constexpr for formerly const fields.
* Adds unit tests.

Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 16:03:28 +00:00
a2ee9234b4 Migrate to Timestamp and TimeDelta types in RtpPacketHistory
Bug: webrtc:13757
Change-Id: Ie542fca50b97fe9dc450e45da40f05e2b66c7da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252981
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36132}
2022-03-04 15:02:58 +00:00
15ee87fe0e Use VideoCodec complexity to determine AV1 encoder cpu_speed.
Bug: webrtc:13744
Change-Id: Ib6d62dcdf7346d886c0aca09735c7d5c1f3e2455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Michael Horowitz <mhoro@google.com>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36125}
2022-03-03 19:06:17 +00:00
45623a3c0f Remove operator= from VCMJitterEstimator and VCMRttFilter
Change-Id: I70846d9cdc17d904585a18983acee7980292e62e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253301
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36122}
2022-03-03 15:26:27 +00:00
b663cfaae4 Cleanup RtpPacketHistory from unused features
history no longer used for storing unsent packets and for legacy pacer.

Bug: None
Change-Id: I639c37de66857a64c620e80df6288fa6ce8326d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36120}
2022-03-03 14:30:27 +00:00
66557e1af3 Revert "remove NV12 to I420 conversion in webrtc AV1 Encoder."
This reverts commit 9558ab41eb4de39c62cda2dd1e559f5814a3a0c7.

Reason for revert: speculative revert: breaks downstream project

Original change's description:
> remove NV12 to I420 conversion in webrtc AV1 Encoder.
>
> libaom supports for NV12 inputs for encoding av1 stream. It will reduce
> unnecessary conversion from NV12 to I420 format.
> (https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)
>
> Bug: webrtc:13746
> Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
> Cr-Commit-Position: refs/heads/main@{#36111}

Bug: webrtc:13746
Change-Id: Ie928f7f5b5992337a9d186fa70b7fdec20a33f00
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253122
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36114}
2022-03-02 16:01:28 +00:00
5cd7d2aa0f audioproc_f: fix AGC1 digital adaptive flag bug
- missing negation causes the opposite behavior when
  `analog_agc_disable_digital_adaptive` is used
- flag replaced with `analog_agc_use_digital_adaptive_controller`
  which is less error-prone

Bug: webrtc:7494
Change-Id: If9e0ba4fc9e539c73269faf9096ca782620dac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251322
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36113}
2022-03-02 15:50:57 +00:00
d6cdf80072 Use Timestamp and TimeDelta in VCMTiming
* Switches TimestampExtrapolator to use Timestamp as well.

Bug: webrtc:13589
Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:07:25 +00:00
9558ab41eb remove NV12 to I420 conversion in webrtc AV1 Encoder.
libaom supports for NV12 inputs for encoding av1 stream. It will reduce
unnecessary conversion from NV12 to I420 format.
(https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)

Bug: webrtc:13746
Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
Cr-Commit-Position: refs/heads/main@{#36111}
2022-03-02 14:18:36 +00:00
c6d3a7a691 Ensure returned delay based estimate from probe can be clamped by
AimdRateControl

AimdRateControl can potentially clamp bitrate in SetEstimate.
DelayBasedBwe::MaybeUpdateEstimate should therefore check the result before using the probe bitrate.
Otherwise, BWE may drop on next update.

Bug: none
Change-Id: I8b1b3549a2bcd981e941b1cc802c984828d68261
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252444
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36099}
2022-03-01 09:45:30 +00:00
141007668c Add field trial for limiting probes and delay based estimates to link
capacity.

Allow delay based estimate to increase up to 85% of the current NetworkStateEstimate
even if in ALR. The estimate may not increase higher than that.
WebRTC-Bwe-EstimateBoundedIncrease/ratio:0.85,ignore_acked:true

Bug: none
Change-Id: I6f34af7fab03082ca168e624ddea06f216790fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252442
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36096}
2022-02-28 17:29:37 +00:00
c2b1bad4c8 In RtcpTransceiver use TimeDelta instead of raw int to represent time
Bug: webrtc:8239, webrtc:13757
Change-Id: Idda3fe5761665b4b3fedaf2dd1a28bb0119ae1f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252287
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36094}
2022-02-28 11:21:17 +00:00
9af4aa7cf4 Reland "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 56db8d09529d5ba92d24954a1d78a90c8ea2cd85.

Reason for revert: downstream problem addressed

Original change's description:
> Revert "Represent RtpPacketToSend::capture_time with Timestamp"
>
> This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.
>
> Reason for revert: Causes problems downstream:
>
> #
> # Fatal error in: rtc_base/units/unit_base.h, line 122
> # last system error: 0
> # Check failed: value >= 0 (-234 vs. 0)
>
> Original change's description:
> > Represent RtpPacketToSend::capture_time with Timestamp
> >
> > Bug: webrtc:13757
> > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36083}
>
> Bug: webrtc:13757
> Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36087}

Bug: webrtc:13757
Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36093}
2022-02-28 10:04:37 +00:00
56db8d0952 Revert "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.

Reason for revert: Causes problems downstream:

#
# Fatal error in: rtc_base/units/unit_base.h, line 122
# last system error: 0
# Check failed: value >= 0 (-234 vs. 0)

Original change's description:
> Represent RtpPacketToSend::capture_time with Timestamp
>
> Bug: webrtc:13757
> Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36083}

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
2022-02-26 10:35:13 +00:00