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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
This reverts commit d0a6fd239c.
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 e76daab8b3.
>
> 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}
This reverts commit c9cad23274.
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}
This reverts commit e76daab8b3.
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}
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}
This reverts commit 791294a647.
Reason for revert: downstream test adjusted
Original change's description:
> Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
>
> This reverts commit a17651f7d8.
>
> 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}
This reverts commit a17651f7d8.
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}
This reverts commit dd32562f24.
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 93bb305149.
>
> 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}
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}
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}
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}
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}
This CL also removes the existing non-standard implementation of the metric.
Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
This trial has been unused for some time, time to clean it up.
Bug: webrtc:10144
Change-Id: I2b1bd9ff0335efdc07f47a361878915f1be383a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267410
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37421}
This reverts commit 93bb305149.
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}
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}
This flag has gone unused for a long time, time to clean it up.
While we're here, convert NackRequester to use unit types.
Bug: webrtc:8624
Change-Id: I1f314f9b5b6771d4f9c351a7a9a887130b86907c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267408
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37400}
Return the bitrate estimate as DataRate type
Remove list of affected ssrcs as unused
Bug: None
Change-Id: Ie31dce591d861624736d834194f90eb6c93f70f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37397}
For this I added a header called no_cfi_icall.h and use it.
Also, some files use the gio header, but if the //base dependency is
not used, compilation errors occur. So I added an explicit dependency
on gio.
Bug: webrtc:13662
Change-Id: If732ede202dd413be6702bf06bf024cd203fdae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267340
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37395}
since the fec packets are initialized to 0 there is no need
to special-case the first packet since
A XOR 0
is the identity operator.
BUG=None
Change-Id: I0cb55283ecdca06f8e3a7b5856ec1f9fbbad1ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251522
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37378}
Look for first echo (and not only the strongest one) on the same matched
filter.
This change is bit exact with previous version when `pre_echo` is false.
Author: Jesús de Vicente Peña <devicentepena@webrtc.org>
Bug: webrtc:14205
Change-Id: I6782eaa1d690b0df78d00f6d425a85c951b2ca9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266321
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37360}
In H264, reordered packets can cause a frame following padding to become stuck in the packet buffer.
A minimal example:
_, P, 1 - padding packet p and frame 1. Frame 1 has not been returned because of missing packet 0
0, P, 1 - when packet 0 arrives, FindFrames will stop incrementing i when it sees padding packet P, and frame 1 will never be returned
Bug: webrtc:14216
Change-Id: I78b76df9709fa8593c5025d647e2b868af3749f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266465
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37357}
In commit a6ed749b12 we used width of the
frame we copy into to calculate the source stride. This is a wrong
assumption as there might be implementations (e.g. GNOME) where we might
have to import a DMA-BUF with size of the whole screen and just having
information in SPA_META_VideoCrop metadata to get the real size of the
frame we will end up using. Given this, we always have to calculate
source stride using the size of the stream to not end up copying pixels
from the empty area of the imported DMA-BUF.
Also improve naming of variables to have names better describing what
they really represent and add some comments explaining why some things
are written the way they are.
Bug: chromium:1333304
Change-Id: I755a5139336c1da5abf95591a2b70a68659a255f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267002
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37344}
It's not a problem if we fail to query DMA-BUF modifiers as we can still
continue with modifier-less buffers.
Bug: webrtc:13429
Change-Id: Ia718362bdc9eef1ebc54c06b24a2b65206aa873e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267003
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#37342}
This CL removes the split of "desktop_capture" in 2 build targets
(one for C++ and one for Obj-C++) by moving the C++ part to
"desktop_capture" itself and keeping the Obj-C++ variant but allowing
it to include .h files that are also part of "desktop_capture".
This removes the build cycle between the two targets (which conceptually
are the same target).
Clients should never depend on "desktop_capture_objc", which will
be linked by "desktop_capture" when needed.
Bug: b/36882554
Change-Id: Id219a15e549275870c54375c07f00cfe704ab7cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266743
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@google.com>
Cr-Commit-Position: refs/heads/main@{#37337}
The tests check that the various scalability mode are supported
and the frames are marked properly by the encoder with their
spatial and temporal index.
The same information is then checked on the receiving side.
A new member is added on EncodedImage to store the temporal index,
and is filled by the encoders and retreived by the ref finder
objects on the decoding side.
Bug: webrtc:11607
Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37303}
When the `WebRTC-Audio-TransientSuppressorVadMode-RnnVad` field trial
is set, APM now uses (i) its RNN VAD sub-module to compute the voice
probability, (ii) that probability for TS and (iii) a temporally
delayed version of it for AGC2 (the delay introduced by TS is taken
into account).
Bug: webrtc:13663
Change-Id: Ic0f245c3f00d318c19bb01d3dbc2d5176c90f851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266362
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37291}
This way call no longer needs dedicated process thread
Bug: webrtc:7219
Change-Id: I8ab677b1e6b909eeb726aefed5e6d10ce4bc43b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37279}
Add a VoiceActivityDetectorWrapper submodule in AudioProcessingImpl
and enable injecting speech probability into GainController2.
Bug: webrtc:13663
Change-Id: I05e13b737d085b45ac8ce76660191867c56834c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265166
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37275}
