Commit Graph

4868 Commits

Author SHA1 Message Date
f4e3e2b83f Delete rtc::Callback0 and friends.
Replaced with std::function.

Bug: webrtc:6424
Change-Id: Iacc43822cb854ddde3cb1e5ddd863676cb07510a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205005
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33281}
2021-02-16 12:41:35 +00:00
067b050213 Delete deprecated unused functions from RtpRtcp interface
Bug: None
Change-Id: Iceb59d726c328974c3ccbf52a782ac9e25bd57c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205581
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33278}
2021-02-16 10:23:41 +00:00
9aa9b8dbbe Prepare to replace VideoLayerFrameId with int64_t.
Bug: webrtc:12206
Change-Id: I10bfdefbc95a79e0595956c1a0e688051da6d2b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33265}
2021-02-15 14:42:02 +00:00
563fbc1dc5 Replace RecursiveCriticalSection with Mutex in DxgiDuplicatorController
Bug: webrtc:11567
Change-Id: I6d59de7ca60b69765118787fff023c485b1f405e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207160
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33264}
2021-02-15 14:40:57 +00:00
f91f8b517a Consolidate full svc structures in one source file
Keeping structures in the same file makes it clearer which are missing
and makes it easier to see if structures are consistent with one another.

No-Try: True
Bug: None
Change-Id: I4e5e6971054dd28dd326c68369ee57b6df62725e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206987
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33256}
2021-02-13 16:17:54 +00:00
8eea117dea Make PostRuntimeSetting pure virtual
Bug: b/177830919
Change-Id: I92e30e9b65c8f851444268f0824a676044504814
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206640
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33252}
2021-02-12 22:49:43 +00:00
bc1cdef4e8 EncoderInfoSettings: Add common string which applies to all encoders.
Change "-LibvpxVp9Encoder-" to "-VP9-" for consistency.

Bug: none
Change-Id: I7a73759db00e92286fe9a4bbed8512baf91decdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206982
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33249}
2021-02-12 14:49:28 +00:00
f109193fba Remove VideoLayerFrameId::spatial_layer, use EncodedImage::SpatialIndex instead.
Next step is to replace VideoLayerFrameId with int64_t.

Bug: webrtc:12206
Change-Id: I414f491e383acf7f8efd97f7bf93dc55a5194fbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206804
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33245}
2021-02-12 11:16:23 +00:00
590b1bad08 Add lock annotations to DxgiDuplicatorController
Bug: webrtc:11567
Change-Id: I34b9138cc15cd534059dd64bb990d41174eeef21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206471
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33242}
2021-02-12 07:51:19 +00:00
5a585952da Update AGC2 tuning
Bug: webrtc:7494
Change-Id: Ifcc5b6c846476ce7d6862fba2cb53e426b5855dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206800
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33238}
2021-02-11 18:14:34 +00:00
45d2234a5c Test and fix unscalable video structure.
Bug: webrtc:11999
Change-Id: I94e3a97ebadbf92ca741d750f67fbea5cbd2b66f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206984
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33237}
2021-02-11 17:23:56 +00:00
ff0e01f689 Implement audio_interruption metric for kCodecPlc
Audio interruption metric is not implemented for codecs doing their own PLC.

R=ivoc@webrtc.org, jakobi@webrtc.org

Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
2021-02-11 09:37:24 +00:00
de7ee3a53d Reland "AV1: change update freq and disable denoiser explicitly."
This is a reland of abf5701c378329115838f3405ff48d43d2502559

Original change's description:
> AV1: change update freq and disable denoiser explicitly.
>
> Change speed/thread settings for faster encoding.
>
> Change-Id: I74d93eac26ae8700a48c437fe235643810de1ca0
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206480
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Reviewed-by: Marco Paniconi <marpan@google.com>
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Cr-Commit-Position: refs/heads/master@{#33208}

Bug: None
Change-Id: Icc8e064b4af175214a7fdec16f3c8078c0220e50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206900
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33226}
2021-02-11 08:25:27 +00:00
4b1c72c2f9 Reland "Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation"
This reverts commit 7bad75b3906ae78b67b2a8cec095d877deb58215.

Reason for revert: Downstream projects fixed.

Original change's description:
> Revert "Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation"
>
> This reverts commit 51f8e09540b1816236ceb1eaa540a7adb019b393.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation
> >
> >
> > Bug: webrtc:10335
> > Change-Id: I85d62b9b63e0b6ec5dd4b957738a67a9a11e3a1f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205627
> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33210}
>
> TBR=perkj@webrtc.org,crodbro@webrtc.org
>
> Change-Id: I220a0e5316c54c435d04bc2bbd714b9d9b92be26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10335
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206645
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33214}

TBR=mbonadei@webrtc.org,perkj@webrtc.org,crodbro@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:10335
Change-Id: I894be638d987e1ac39d7e8a9e642622f14e1acd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206806
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33221}
2021-02-10 16:56:59 +00:00
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
7bad75b390 Revert "Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation"
This reverts commit 51f8e09540b1816236ceb1eaa540a7adb019b393.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation
>
>
> Bug: webrtc:10335
> Change-Id: I85d62b9b63e0b6ec5dd4b957738a67a9a11e3a1f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205627
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33210}

TBR=perkj@webrtc.org,crodbro@webrtc.org

Change-Id: I220a0e5316c54c435d04bc2bbd714b9d9b92be26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206645
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33214}
2021-02-10 09:34:48 +00:00
fa5ad8c0b5 Revert "AV1: change update freq and disable denoiser explicitly."
This reverts commit abf5701c378329115838f3405ff48d43d2502559.

Reason for revert: Breaks downstream tests.

Original change's description:
> AV1: change update freq and disable denoiser explicitly.
>
> Change speed/thread settings for faster encoding.
>
> Change-Id: I74d93eac26ae8700a48c437fe235643810de1ca0
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206480
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Reviewed-by: Marco Paniconi <marpan@google.com>
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Cr-Commit-Position: refs/heads/master@{#33208}

TBR=jianj@google.com,marpan@google.com,marpan@webrtc.org

Change-Id: I47b65e1c78ccb055238a44886dac87f8fc2f5330
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206644
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33213}
2021-02-10 09:30:10 +00:00
51f8e09540 Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation
Bug: webrtc:10335
Change-Id: I85d62b9b63e0b6ec5dd4b957738a67a9a11e3a1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205627
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33210}
2021-02-09 21:50:41 +00:00
abf5701c37 AV1: change update freq and disable denoiser explicitly.
Change speed/thread settings for faster encoding.

Change-Id: I74d93eac26ae8700a48c437fe235643810de1ca0
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206480
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#33208}
2021-02-09 20:36:37 +00:00
82ce7e5515 Fix PacedSender class to use plain mutex, rather than RecursiveCriticalSection
Bug: webrtc:11567
Change-Id: I51f17ddebdda2fafeb9b721d038b16e784e7bd8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206464
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33202}
2021-02-09 12:11:25 +00:00
0a144a705a Adding initial support for lock-less informing of muting
This CL adds the initial support for letting APM know when its output
will be used or not.
It also adds a new method for passing RuntimeInformation to APM that
returns a bool indicating the success of the passing of information.

Bug: b/177830919
Change-Id: Ic2e1b92c37241d74ca6394b785b91736ca7532aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206061
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33201}
2021-02-09 12:08:54 +00:00
1142e0bfb2 Avoid crashing on error code 6450 in isac.
Isac is not able to effectively compress all types of signals. This
should be extremely rare with real audio signals, but mostly happens
with artificially created test signals. When this happens, we should
avoid crashing and just carry on.

Bug: chromium:1170167
Change-Id: I97b551fbbdcccb0186f3e6497991ac52d2301f68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205626
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33193}
2021-02-08 16:16:55 +00:00
68c225d76d Make 48 kHz maximum rate default for all devices
Bug: b/169918549
Change-Id: I2f4b7ced5ae6efcae3cd59c0a42610a54f5e2dc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203260
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33192}
2021-02-08 15:06:05 +00:00
a0848ddeff Correct SpatialLayer in VP9 unittest.
Bug: None
Change-Id: If8b26c8e7afa380f109d71a93b78bad784da34ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205961
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33187}
2021-02-08 10:13:18 +00:00
9554a7b641 Account for extra capacity rtx packet might need
When calculating maximum allowed size for a media packet.
In particular take in account that rtx packet might need to
include mid and repaired-rsid extensions when media packet can omit them.

Bug: webrtc:11031
Change-Id: I3e7bc36437c23e0330316588d2a46978407c8c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206060
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33184}
2021-02-06 21:34:08 +00:00
879d33b9f8 Add more refined control over dumping of data and the aecdump content
This CL adds the ability in audioproc_f and unpack_aecdump to:
-Clearly identify the Init events and when those occur.
-Optionally only process a specific Init section of an aecdump.
-Optionally selectively turn on dumping of internal data for a
 specific init section, and a specific time interval.
-Optionally let unpack_aecdump produce file names based on inits.

Bug: webrtc:5298
Change-Id: Id654b7175407a23ef634fca832994d87d1073239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33181}
2021-02-06 00:36:10 +00:00
3b9abd8dee Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder.
Bug: webrtc:12380
Change-Id: I60e4684150cb4880224f402a9bf42a72811863b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202920
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33174}
2021-02-05 10:05:25 +00:00
483b31c231 Reland "Enable Video-QualityScaling experiment by default"
This time exclude iOS from the default behaviour.

Bug: webrtc:12401
Change-Id: Ib1f77123b72c3365591b56455332b3d97b307b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205006
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33173}
2021-02-05 09:49:13 +00:00
65b901bbb1 Clean up previously deleted RTCP VOIP metrics block.
Bug: None
Change-Id: I6f9ddb09927200444dbccd24ed522c9b8f936b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205623
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33169}
2021-02-04 18:34:28 +00:00
426b6e49de changed src\modules\audio_device\win\audio_device_core_win.cc , and it is working
Bug: webrtc:12384
Change-Id: Ie9fddc3fa8016eb6a0bcc4c6757f30c4b087c10a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203821
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33158}
2021-02-04 09:31:33 +00:00
b5823055be In VP9 encoder avoid crashing when encoder produce an unexpected frame
Since for such frame SvcController haven't setup how buffer should be
referenced and updated, the frame would likely have unexpected configuration.
Log an error to note resource have been wasted produce it and drop such frame.

Bug: webrtc:11999
Change-Id: I1784403e67b7207092d46016510460738994404e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205140
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33148}
2021-02-03 14:56:09 +00:00
53610223a8 Delete unused function webrtc::AudioProcessing::MutateConfig
Bug: None
Change-Id: Ibc70e5246a3f7b89775c65a19c808c1f030b8ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205522
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33147}
2021-02-03 12:55:23 +00:00
6f75f6b3bd APM: add AGC2 SIMD kill switches in AudioProcessing::Config::ToString()
Bug: webrtc:7494
Change-Id: Icba5f6be689a57ef4748ae816565349fd1ad2108
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205322
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33146}
2021-02-03 12:31:56 +00:00
9111bd18b1 LibvpxVp8Encoder: add option to configure resolution_bitrate_limits.
Bug: none
Change-Id: Ia01d630fc95e19a4a08cd7a004238c22d823b4dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205521
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33144}
2021-02-03 11:25:32 +00:00
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
eee0e9e9d4 Remove passing rtp packet metadata through webrtc as array of bytes
Instead metadata is now passed via refcounted class.

Bug: b/178094662
Change-Id: I9591fb12990282b60310ca01aea2a7b73d92487a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204060
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33134}
2021-02-02 12:22:57 +00:00
e7ded686d5 Fix integer overflow.
Bug: chromium:1172583
Change-Id: I72c6c07f6f5702311c1a73eb4551e92a34c87e47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205007
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33127}
2021-02-01 17:37:19 +00:00
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.

`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.

Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 13:46:19 +00:00
d6604df27f Revert "Enable Video-QualityScaling experiment by default"
This reverts commit 066b5b6ed7069d78e17b8ad6fb8c82546b31acea.

Reason for revert: Regressions on iOS testbots.

Original change's description:
> Enable Video-QualityScaling experiment by default
>
> Bug: webrtc:12401
> Change-Id: Iebf3130e785892bb9fddf1012bc46027a21085a4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204000
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33091}

TBR=ilnik@webrtc.org,asapersson@webrtc.org

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

Bug: webrtc:12401
Change-Id: I489b805c7741b63c22c16cfce03347179a3e2602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205001
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33123}
2021-02-01 13:20:49 +00:00
3faea70d1a allow empty scalability mode in AV1 encoder
Bug: chromium:1170699
Change-Id: I74c633e74c85c3b940d6302cdc8fa319e187b1e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204221
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33122}
2021-02-01 13:00:09 +00:00
c91c4233e3 LibvpxVp9Encoder: add option to configure resolution_bitrate_limits.
Bug: none
Change-Id: Icdd7333296d652b1e0c159226df702084303475c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204701
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33121}
2021-02-01 11:48:00 +00:00
7864600a6e Add absl_deps field for rtc_test and rtc_executable
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".

Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 16:40:49 +00:00
b79acd8ce9 Format webrtc/modules/audio_processing/transient/BUILD.gn file
Bug: webrtc:12404
Change-Id: I3cb6b96255409709fa3144d2f83d13d12e39ab2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204064
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33107}
2021-01-29 16:33:19 +00:00
1a29a5da84 Delete rtc::Bind
Bug: webrtc:11339
Change-Id: Id53d17bbf37a15f482e9eb9f8762d2000c772dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202250
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33099}
2021-01-29 08:24:43 +00:00
8bd0f97fe6 Address CL comments from 200161.
This is a follow up to
https://webrtc-review.googlesource.com/c/src/+/200161

I forgot to actually upload the code that addressed reviewer's comments,
and since they were minor comments I went ahead and completed the CL
before I realized.

Fortunately no harm done because all of this code is behind a disabled
build flag. This is a great example of move fast, make mistakes. Lesson
learned.

Bug: webrtc:9273
Change-Id: I5e35b31719b264855568de4b5e595fe0f192654e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204420
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33095}
2021-01-28 22:59:30 +00:00
066b5b6ed7 Enable Video-QualityScaling experiment by default
Bug: webrtc:12401
Change-Id: Iebf3130e785892bb9fddf1012bc46027a21085a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204000
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33091}
2021-01-28 18:17:03 +00:00
3b68aa346a Move some RTC_LOG to RTC_DLOG.
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.

Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
2021-01-28 10:05:00 +00:00
2aad81259e Refactor and implement WgcCapturerWin, a source agnostic capturer.
This change refactors WgcWindowCapturer into WgcCapturerWin, a source
agnostic capturer, and finishes the implementation to enable both window
and screen capture.

This CL depends on another which must complete first:
196622: Add ability to load CreateDirect3DDeviceFromDXGIDevice from
d3d11.dll | https://webrtc-review.googlesource.com/c/src/+/196622

This feature remains off by default behind a build flag, due to it
adding a depency on the Win10 SDK vv10.0.19041 which not all consumers
of WebRTC have upgraded to. A follow up change later will enable the
rtc_enable_win_wgc build flag, but for now it should remain off.

The basic operation of this class is as follows:
Consumers call either WgcCapturerWin::CreateRawWindowCapturer or
CreateRawScreenCapturer to receive a correctly initialized
WgcCapturerWin object suitable for the desired source type.

Callers then indicate via SelectSource and a SourceId the desired
capture target, and the capturer creates an appropriate WgcCaptureSource
for the correct type (window or screen) using the
WgcCaptureSourceFactory supplied at construction.

Next, callers request frames for the currently selected source and the
capturer then creates a WgcCaptureSession and stores it in a map for
more efficient capture of multiple sources.

The WgcCaptureSession is supplied with a GraphicsCaptureItem created by
the WgcCaptureSource. It uses this item to create a
Direct3D11CaptureFramePool and create and start a
GraphicsCaptureSession.

Once started, captured frames will begin to be deposited into the
FramePool. Typically, one would listen for the FrameArrived event and
process the frame then, but due to the synchronous nature of the
DesktopCapturer interface, and to avoid a more complicated multi-
threaded architecture we ignore the FrameArrived event. Instead, we
wait for a request for a frame from the caller, then we check the
FramePool for a frame, and process it on demand.

Processing a frame involves moving the image data from an
ID3D11Texture2D stored in the GPU into a texture that is accessible
from the CPU, and then copying the data into the new WgcDesktopFrame
class. This copy is necessary as otherwise we would need to manage the
lifetimes of the CaptureFrame and ID3D11Texture2D objects, lest the
buffer be invalidated.

Once we've copied the data and returned it to the caller, we can unmap
the texture and exit the scope of the GetFrame method, which will
destruct the CaptureFrame object. At this point, the CaptureSession
will begin capturing a new frame, and will soon deposit it into the
FramePool and we can repeat.

Bug: webrtc:9273
Change-Id: I02263c4fd587df652b04d5267fad8965330d0f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200161
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33083}
2021-01-27 21:17:19 +00:00
103876f379 av1: turn off a few tools that are not used for rtc
Explicitly turn off obmc, warped motion, global motion and ref frame mv.

Bug: chromium:1095763, chromium:1170346
Change-Id: I19bc4fceef4cd5e35ea6699e6af883af244f8954
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203900
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#33075}
2021-01-26 17:49:34 +00:00
5312a8f532 Add option to attach custom object to an rtp packet
As an alternative to attaching custom array of bytes.

Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
2021-01-25 18:31:34 +00:00