5817 Commits

Author SHA1 Message Date
Danil Chapovalov
6e7c2685e3 Allow recursive check for RTC_DCHECK_RUN_ON macro
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.

Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue

Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
2022-07-26 09:27:23 +00:00
Danil Chapovalov
8743db9865 Cleanup congestion controller min bitrate
Replace helper functions with the constant

Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.

Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
2022-07-25 17:31:28 +00:00
Erik Språng
06a6f3f818 Make PrioritizedPacketQueue the default.
Keeping the field trial around as a kill-switch for another milestone.

Bug: webrtc:11340
Change-Id: I3285baefab176f541cbb5ed3bacbc669d3e8836f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269384
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37611}
2022-07-25 16:24:58 +00:00
Danil Chapovalov
fc22dfa727 In RemoteEstimatorProxy treat zero arrival time as valid
Inserting packet with zero arrival time may trigger inconsistent state in the internal map where packet sometimes treated as received, but sometimes treated as not received.

Bug: chromium:1346959
Change-Id: I0809e41a873103dcd62528358e64794c1d3cb28f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37609}
2022-07-25 13:08:19 +00:00
Diep Bui
5edefa87f7 Update loss based bwe 2 to handle low bandwidth networks and no delay signal.
1. Add loss threshold for high bandwidth preference. If the average loss ratio is less than the threshold, then the model prefers higher bandwidth candidates. Otherwise, it prefers lower bandwidth candidates. Before, it always prefers higher bandwidth candidate. The default value is 0.99, means it always prefers high bandwidth candidates.

2. Only increase the estimate if the inherent loss (random loss) is equal to/greater than the average loss. If the inherent loss is less than the average loss, then it is oversending, thus should not increase the estimate.

Bug: webrtc:12707
Change-Id: I37eb536679ca29e017a4a47703b417efd4d103ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269101
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37608}
2022-07-25 10:45:47 +00:00
Ali Tofigh
82c29716c0 Adopt absl::string_view in modules/audio_device/
Bug: webrtc:13579
Change-Id: I6e8a90281a9d70a40364b6df5fee4f0a55b4a797
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269060
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37607}
2022-07-25 10:35:17 +00:00
philipel
f012bfaf96 Use Video{Encoder, Decoder}FactoryTemplate instead of Internal{Encoder, Decoder}Factory.
Bug: webrtc:13573
Change-Id: Id0e46a9b6053dedae3cbf0e5581768868900630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269247
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37596}
2022-07-22 12:07:25 +00:00
Austin Orion
de4fd2f9ef WindowCapturerWinGdi shouldn't deliver SUCCESS and nullptr.
Consumers expect the frame to be valid if Result::SUCCESS is delivered.
If the frame is nullptr, we should deliver ERROR_TEMPORARY instead.

Bug: webrtc:14265
Change-Id: If94a3ead38d7657d7b90bbe046256be697312216
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269223
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37590}
2022-07-21 21:03:24 +00:00
Diep Bui
9804aa5f6a Avoid fraction_loss overflowing when packet loss is negative in send_side_bandwidth_estimation.cc.
Update an affected unit test by the change in goog_cc.

Bug: webrtc:14272
Change-Id: I83e97530c861b126bed876d57f6d4f91aa45de7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269002
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37587}
2022-07-21 14:31:04 +00:00
Byoungchan Lee
e044ec572a Don't print warning for tasks running 1ms earlier than planned.
Bug: webrtc:12889
Change-Id: I33faa986130f2d7ae049466c303ef29b643d97ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268920
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37585}
2022-07-21 11:44:35 +00:00
Danil Chapovalov
be5258e61d Optimize adding many consecutive missing packets to rtcp TransportFeedback
Bug: chromium:1342840
Change-Id: I894157af2ed4f8b9dc97ccb8613cbf18db09f95a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269100
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37583}
2022-07-21 10:15:25 +00:00
Alessio Bazzica
d9f1208de7 AgcManagerDirect parametrized unit tests
It is now easier to fully test `AgcManagerDirect` with different values
for the used field trials. In particular, this CL adds tests for the
field trial named `WebRTC-Audio-2ndAgcMinMicLevelExperiment`.

1. `UnmutingRaisesTooLowVolume` and `MicVolumeIsLimited`
The expectations for the lowest input volume are not hard-coded anymore
since the parametrized tests use different values for the enforced
minimum.

2. `RecoveryAfterManualLevelChangeBelowMin`
The recovery behavior after manual input volume change depends on
whether the minimum input volume is overridden. When that's the case,
the minimum volume is applied immediately after the manual adjustment.
Hence, the existing test is left and a parametrized version of it has been added to test the "instant recovery" behavior. The latter test is
skipped when the minimum input volume is not overridden since that case
is covered by the existing test.

Bug: chromium:1275566
Change-Id: Ib0d4427b32b88f33138d4062b365916a3c47a406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37577}
2022-07-20 15:44:13 +00:00
Niels Möller
daddfee8c5 Delete tests with 4 temporal layers
Using 4 temporal layers is not quite supported: Not advertised, no
integration tests. When transitioning to configuration via scalability
mode, there are no corresponding modes defined. So delete these two
tests; they can be added back if/when support for corresponding
scalability modes are added.

Bug: webrtc:11607
Change-Id: I97f55dc95d6513ccf65fa887757a62e9c8659be7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269003
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37575}
2022-07-20 14:05:13 +00:00
Ali Tofigh
714e3cbb48 Adopt absl::string_view in modules/audio_coding/
Bug: webrtc:13579
Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 13:34:23 +00:00
Niels Möller
07d80675e2 Move test utilities into more specific build targets
Move audio- and video-specific utilities to audio_test_common (newly
added target) and video_test_common.

Bug: webrtc:10198
Change-Id: Ia10fa5c0a51d9b1f37db4964984d22fc5b269bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268980
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37570}
2022-07-20 10:14:03 +00:00
Alessio Bazzica
866caeb62c AgcManagerDirect ctor API and doc string improved
Bug: chromium:1275566
Change-Id: Iedc8f5cbbf65fbf018da9df1aaa1f8ade1bbc063
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268840
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37569}
2022-07-20 09:39:24 +00:00
Ivo Creusen
1a84b565ac Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
This metric was recently added to the standard (see https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay). This CL implements it for audio streams.

Bug: webrtc:14141
Change-Id: I79d918639cd12361ebbc28c2be41549e33fa7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262770
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37567}
2022-07-20 09:14:03 +00:00
Zhaoliang Ma
e7696f771d Plumb color space from VP8/VP9/H264 encoder
Bug: None
Change-Id: If771d9486bde01d5a2775d904a01ecf3953e75df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268944
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37566}
2022-07-20 09:08:41 +00:00
Danil Chapovalov
c05a1be5b4 Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
Bug: webrtc:14245
Change-Id: I8de2c23da5fbdfc0b1efbbe07fb6e8de744424a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268191
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37565}
2022-07-20 08:15:08 +00:00
Alessio Bazzica
7d4116855a AgcManagerDirect unit tests refactoring
Stop using TEST_F; that will make it easier to switch to parametric
tests that are needed to correctly test `AgcManagerDirect`.

"Avoid fixtures where reasonable."
Source: https://abseil.io/tips/122

Bug: chromium:1275566
Change-Id: I2d73a0913eb2349144f63bd17ab4d6efa245e472
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268766
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37556}
2022-07-19 12:46:42 +00:00
Bruno Pitrus
99465b1395 Add missing header to fix build error when using linux system libraries
Change-Id: I4fc04563c2cfe36fa2352f72f2ae61d47972f025
Bug: webrtc:11226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268194
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Auto-Submit: Bruno Pitrus <brunopitrus@hotmail.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37555}
2022-07-19 12:25:42 +00:00
Niels Möller
253f36f88e Delete rtp_sender_ check in ModuleRtpRtcpImpl2::SetSendingMediaStatus
Analogous to https://webrtc-review.googlesource.com/c/src/+/267845/

Bug: webrtc:10198
Change-Id: Ib7d5e9b2a456486a419c61e7b2ce36df8960c67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268762
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37550}
2022-07-18 14:28:31 +00:00
Danil Chapovalov
03f8b8a241 In video replace non-owning pointer to rtc::TaskQueue with non-owning pointer to TaskQueueBase
rtc::TaskQueue is a simple wrapper over TaskQueueBase and adds no
extra features when task queue is used without passing ownership.

Reducing usage of the internal rtc::TaskQueue wrapper gives users more flexibility how TaskQueueBase* is stored.

Bug: webrtc:14169
Change-Id: If5c8827544c843502c7dfcef775ac558de79ec3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268189
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37549}
2022-07-18 13:59:32 +00:00
Niels Möller
d78789eee2 Delete old TODOs.
Bug: webrtc:10198
Change-Id: I7ea6ddedd97db17a9fc8caf6434cf72f6cd0d6ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268761
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37544}
2022-07-18 12:09:31 +00:00
Danil Chapovalov
3e378d7efa Refactor AecDump not to rely on QueuedTask
Bug: webrtc:14245
Change-Id: Ib41765652745a247da2ae6c2ca6be714de927ca7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268185
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37542}
2022-07-18 11:07:51 +00:00
Alessio Bazzica
08480a599d Reland "AgcManagerDirect: stop enforcing min mic level override with 0 level"
This reverts commit d0a6fd239cef0d9fd5fdd5a41df389a696bff017.

Reason for revert: reland the bug fix

Original change's description:
> Revert "`AgcManagerDirect`: stop enforcing min mic level override with 0 level"
>
> This reverts commit e76daab8b36f8c2a16d59a116425a3a2f98022f6.
>
> Reason for revert: revert required to revert the parent CL
>
> Original change's description:
> > `AgcManagerDirect`: stop enforcing min mic level override with 0 level
> >
> > https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> > due to which the min mic level override is always enforced, if specified
> > even if the user manually adjusts the mic level to zero.
> >
> > This CL fixes that bug, the changes run behind a kill switch.
> >
> > TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
> >
> > Bug: chromium:1275566
> > Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> > Reviewed-by: Hanna Silen <silen@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37460}
>
> Bug: chromium:1275566
> Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37515}

Bug: chromium:1275566
Change-Id: I7198587dec2a153270e8beb714e9dacccdaae806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268544
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37530}
2022-07-15 09:43:22 +00:00
Alessio Bazzica
f3c86154d4 Revert "Min mic analog level: override minimum and behavior on Mac"
This reverts commit c9cad23274a837b135db98c6ce96f85bbbc81604.

Reason for revert: add back field trial

Original change's description:
> Min mic analog level: override minimum and behavior on Mac
>
> This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
> and always enables the code path behind that flag on Mac. In summary,
> the analog AGC behaves as follows on Mac:
> 1. the minimum level is overridden to 20
> 2. the minimum is applied even when clipping is detected
> 3. when the level is manually adjusted to 0, the minimum level is
>   enforced - i.e., 20
>
> Note that the 3rd property had been unintentionally added when the
> changes were added behind the aforementioned field trial. This will
> be fixed in a follow-up CL.
>
> Bug: chromium:1275566
> Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37459}

Bug: chromium:1275566
Change-Id: I00a37ad9e16efc49f721558d25af16efd5f3db8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268540
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37521}
2022-07-14 11:44:47 +00:00
Mirko Bonadei
9dcbfd8614 Revert "In bitrate estimator Improve handing send time of out of order packets"
This reverts commit 2295ddbff978b9954688ee6163c9f3f554a7a85e.

Reason for revert: Investigation required because it breaks some downstream tests.

Original change's description:
> In bitrate estimator Improve handing send time of out of order packets
>
> Bug: None
> Change-Id: I74da3b616fb9419de8f7d9d28326354cee1c178d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268061
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37494}

Bug: None
Change-Id: Ib8ab916b9eedb93aac5fc35c5d291b1f4ed16de0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268541
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@{#37520}
2022-07-14 11:35:47 +00:00
Alessio Bazzica
d0a6fd239c Revert "AgcManagerDirect: stop enforcing min mic level override with 0 level"
This reverts commit e76daab8b36f8c2a16d59a116425a3a2f98022f6.

Reason for revert: revert required to revert the parent CL

Original change's description:
> `AgcManagerDirect`: stop enforcing min mic level override with 0 level
>
> https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> due to which the min mic level override is always enforced, if specified
> even if the user manually adjusts the mic level to zero.
>
> This CL fixes that bug, the changes run behind a kill switch.
>
> TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
>
> Bug: chromium:1275566
> Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37460}

Bug: chromium:1275566
Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37515}
2022-07-13 20:58:07 +00:00
Sunggook Chue
b5d77a0c84 webrtc: Blank desktop capturer regards empty frame as a blank frame
There is a AV where GDI capturer sends empty frame to the
blank detector. It is fine operation from the GDI capturer
to pass an empty to the next handler. So, blank capturer
filter it and send it as blank frame to next handler.

Bug: webrtc:14265
Change-Id: Ifc90a210703e14fa6d0dc7fb2ae2942ae4e8125f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268444
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37513}
2022-07-13 18:03:07 +00:00
Ali Tofigh
b7821cea6b Remove unnecessary overload in RtcEventLogOutput
Bug: webrtc:13579
Change-Id: I3ea4b8ce8d111ae6b9ce7e92f75bd4196bc9656b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268420
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37508}
2022-07-12 22:09:36 +00:00
philipel
9e9bc644f8 Update visibility for dav1d_decoder.
Bug: webrtc:13573
Change-Id: Icb43b11cf0a6ad2b90f6876875bcb545be01ec0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268303
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37505}
2022-07-12 10:07:46 +00:00
Stephan Hartmann
cb56277a17 libstdc++: add missing atomic include for std::atomic
Bug: chromium:957519
Change-Id: I93242198ef8277d5f4d6044fb565d3126768b514
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268187
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37504}
2022-07-12 09:23:26 +00:00
Danil Chapovalov
52747f1744 Reland "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
This reverts commit 791294a647bfba8ebd26821a78020a2bb2f82b9b.

Reason for revert: downstream test adjusted

Original change's description:
> Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
>
> This reverts commit a17651f7d8748905d902eedf34471a0c227ca789.
>
> Reason for revert: triggers failure in downstream test
>
> Original change's description:
> > Fix overflow due to rounding in AbsoluteSendTime::To24Bits
> >
> > Actual rounding is not important for this time as long it is consistent
> > during the call: only difference between two absolute send time matter
> > Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
> >
> > Bug: None
> > Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37468}
>
> Bug: None
> Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37473}

Bug: None
Change-Id: I99bcc6c6b7c08cd9621bdce336cd5793f78ee657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268190
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37498}
2022-07-08 16:09:02 +00:00
Danil Chapovalov
2295ddbff9 In bitrate estimator Improve handing send time of out of order packets
Bug: None
Change-Id: I74da3b616fb9419de8f7d9d28326354cee1c178d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268061
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37494}
2022-07-08 13:19:43 +00:00
Erik Språng
28bc2ca92c Remove unused WebRTC-LimitPaddingSize field trial
Bug: webrtc:11508
Change-Id: Ib7d48e23bd44e2f948d51800090fc14b873d11eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268122
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37485}
2022-07-07 14:28:06 +00:00
philipel
26910ffe22 Make dav1d the default AV1 decoder.
Bug: chromium:1330308, b/234414450
Change-Id: Idc35c9e3612843001f8d7d9361f3769b45350e63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268183
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37483}
2022-07-07 14:04:46 +00:00
Danil Chapovalov
677c1ddde5 Migrate rtp_rtcp to absl::AnyInvocable based TaskQueueBase interface
Bug: webrtc:14245
Change-Id: I037f964130648caf0bd1de86611f8681d475b078
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268146
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37481}
2022-07-07 12:39:25 +00:00
Erik Språng
c52e627c83 Remove WebRTC-Pacer-DynamicPaddingTarget field trial
Bug: webrtc:10809
Change-Id: I9f4adbea5721408b339bbf7497c20834537e50c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268145
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37480}
2022-07-07 12:24:04 +00:00
Danil Chapovalov
791294a647 Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
This reverts commit a17651f7d8748905d902eedf34471a0c227ca789.

Reason for revert: triggers failure in downstream test

Original change's description:
> Fix overflow due to rounding in AbsoluteSendTime::To24Bits
>
> Actual rounding is not important for this time as long it is consistent
> during the call: only difference between two absolute send time matter
> Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
>
> Bug: None
> Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37468}

Bug: None
Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37473}
2022-07-07 07:19:44 +00:00
Austin Orion
81797744fd Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing."
This reverts commit dd32562f242b247aed8add4efecaf3e20c623b9a.

Reason for revert: Updated the original change to dynamically load
the CoreMessaging.dll instead of statically linking with the .lib.

Original change's description:
> Revert "Wait for frames to arrive in WgcCapturer instead of returning nothing."
>
> This reverts commit 93bb3051490253d56dc1cdab4701b91138a151c3.
>
> Reason for revert: It breaks a test while rolling into Chromium,
> see https://webrtc-review.googlesource.com/c/src/+/261780/21#message-4a96e33bfb475f19a618be82bbe72951b23085ef for details.
>
> Original change's description:
> > Wait for frames to arrive in WgcCapturer instead of returning nothing.
> >
> > We're seeing a high instance of "first capture failed" in Chromium when
> > using WGC. We can reduce this by waiting for frames to arrive if there
> > are none in the frame pool instead of returning a temporary error.
> >
> > I've set the maximum time to wait for a frame to 50ms. If no frame
> > arrives before 50ms has elapsed, we will return a temporary error.
> > Added a new test, FirstCaptureSucceeds, to verify that this is working
> > as expected.
> >
> > As part of this I updated the name of the `kCreateFreeThreadedFailed`
> > enum value to `kCreateFramePoolFailed`. The value remains the same
> > since they both report failures in frame pool creation.
> >
> > I also increased `kNumBuffers` from 1 to 2, so that the frame pool can
> > store two frames. This should prevent us from having to wait on the
> > event as frequently. This will increase the latency between capture
> > and display, however. High frame rate applications should not be
> > noticeably affected.
> >
> > Additionally, we uncovered a bug in the OS that prevents window capture
> > when there are displays attached, but none of them are active. Added
> > a new check to `IsWgcSupported` to cover this scenario.
> >
> > Finally, some issues with other WGC tests blocked moving the TryBots
> > to a newer version of Windows. This CL fixes those issues and updates
> > the TryBot configuration.
> >
> > bug: chromium:1314868
> > Change-Id: Id9c4d5ee98621e682ef04864c3848d50e761cdb7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261780
> > Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> > Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> > Commit-Queue: Austin Orion <auorion@microsoft.com>
> > Reviewed-by: Jeremy Leconte <jleconte@google.com>
> > Cr-Commit-Position: refs/heads/main@{#37404}
>
> Change-Id: If237df4826fe20b6fe2ca4b57253623321bf33c5
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267460
> 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>
> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37408}

Change-Id: I6cc2becd9ed363782ab2f326f58d9401bc8fb820
Bug: chromium:1314868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267902
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37470}
2022-07-06 20:28:26 +00:00
Danil Chapovalov
a17651f7d8 Fix overflow due to rounding in AbsoluteSendTime::To24Bits
Actual rounding is not important for this time as long it is consistent
during the call: only difference between two absolute send time matter
Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.

Bug: None
Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37468}
2022-07-06 16:25:54 +00:00
Danil Chapovalov
0be8eba07e Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface
Bug: webrtc:14245
Change-Id: Icfab3e6548055ea72a199a226eca5233b1ead20d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267983
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37467}
2022-07-06 15:46:04 +00:00
Alessio Bazzica
e76daab8b3 AgcManagerDirect: stop enforcing min mic level override with 0 level
https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
due to which the min mic level override is always enforced, if specified
even if the user manually adjusts the mic level to zero.

This CL fixes that bug, the changes run behind a kill switch.

TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI

Bug: chromium:1275566
Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37460}
2022-07-06 09:50:43 +00:00
Alessio Bazzica
c9cad23274 Min mic analog level: override minimum and behavior on Mac
This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
and always enables the code path behind that flag on Mac. In summary,
the analog AGC behaves as follows on Mac:
1. the minimum level is overridden to 20
2. the minimum is applied even when clipping is detected
3. when the level is manually adjusted to 0, the minimum level is
  enforced - i.e., 20

Note that the 3rd property had been unintentionally added when the
changes were added behind the aforementioned field trial. This will
be fixed in a follow-up CL.

Bug: chromium:1275566
Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37459}
2022-07-06 09:46:24 +00:00
Niels Möller
cb99ccd244 Update/delete old TODOs
Bug: webrtc:10198
Change-Id: I0341e068d792bc0b143db86e675988f4cd07ff2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267822
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37454}
2022-07-06 07:49:04 +00:00
Niels Möller
ea8eff3737 Delete rtp_sender_ check in ModuleRtpRtcpImpl::SetSendingMediaStatus
Bug: webrtc:10198
Change-Id: Ic40cd702717665a70f5aac0833963d467ea71dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267845
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37452}
2022-07-06 06:07:43 +00:00
Byoungchan Lee
a1a7c638ec Let PCF.GetRtpSenderCapabilities return codecs' scalabilityModes.
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.

Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
2022-07-05 13:28:33 +00:00
philipel
e1c707c40f Remove unused incomplete_frame argument from JitterEstimator.
Bug: webrtc:14151
Change-Id: I6764315f0c10b304f50e4639a3e49e4ed013c41e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267842
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37443}
2022-07-05 12:53:13 +00:00
Niels Möller
39b1b42487 Use designated initializers for webrtc::SimulcastStream
Style change extracted from
https://webrtc-review.googlesource.com/c/src/+/264800

Bug: webrtc:11607
Change-Id: I3dd5ca1eef8d70a61023af37d90032225e40b55d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267841
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37442}
2022-07-05 12:23:44 +00:00