Commit Graph

5728 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
11fdb08282 Implement RTCInboundRTPStreamStats.JitterBufferTargetDelay
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}
2022-07-05 11:34:53 +00:00
63299a3124 Add absl::string_view overload for RtcEventLogOutput::Write
Bug: webrtc:13579
Change-Id: I13f63fb6be6aa62c2e011c18327499fa16b5824e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267641
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37440}
2022-07-05 10:47:47 +00:00
2ad75b3956 Remove testonly from unpack_aecdump.
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}
2022-07-05 10:23:53 +00:00
6939f63ca1 Update old TODO comments
Bug: None
Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37436}
2022-07-05 09:59:33 +00:00
c8152fe4a8 Update/delete old TODOs
Bug: webrtc:10198
Change-Id: I226768c2a6bd97ffcd0638e5bc6a1c286b71815f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267704
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37435}
2022-07-05 09:44:53 +00:00
fb9fbdf395 Delete unused UlpfecReceiver::ProcessReceivedFec return value
Bug: webrtc:10198
Change-Id: Ibb85f1b9094d09dabe677ccbc11e00f3a3590c50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267705
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37434}
2022-07-05 09:40:53 +00:00
56257afe10 Cleanup ReceiveSideCongestionController: remove inner wrapper helper
Bug: None
Change-Id: Iff388a56176d90e300e0c12b34414ee21fa26bc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267406
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37429}
2022-07-05 08:28:55 +00:00
f82e8fa911 Remove WebRTC-Bwe-AlrLimitedBackoff field trial.
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}
2022-07-04 16:29:42 +00:00
0fd2ed516b Delete ProcessThread and related Module interface
Bug: webrtc:7219
Change-Id: Id71430a24b21e591494557cf54419d2bc8b3f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267400
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37416}
2022-07-04 10:20:35 +00:00
67d23043f3 Fix config of number of temporal layers
Needed to produce correct VideoLayersAllocation extension for
scalability mode L1T2. The value in the `spatialLayers` array
is used on this line:
https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;drc=c374d11fac252535ccba15975568b1f6552c117e;l=320

Bug: webrtc:11607
Change-Id: I3bcfe738627e0af6f203a9b0f6e5323492e68987
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267621
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37414}
2022-07-04 08:59:02 +00:00
dd32562f24 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}
2022-07-02 07:41:21 +00:00
48cc54e4ce Remove code for unused field trial WebRTC-BweCappedProbing
Bug: None
Change-Id: I6799794659dce52f0d9f98dc1b5c63e0806d152d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267403
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37406}
2022-07-01 19:11:15 +00:00
93bb305149 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}
2022-07-01 17:42:20 +00:00
c50e423d3b Fix possible integer overflow.
Bug: chromium:1340143
Change-Id: Ia874c90b53e5c527d163a0fe566743713a55ca6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206986
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37402}
2022-07-01 16:26:10 +00:00
609aef3149 Remove WebRTC-ExponentialNackBackoff field trial from NackRequester.
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}
2022-07-01 15:32:44 +00:00
74680c0234 Cleanup RemoteBitrateEstimate::LatestEstimate function
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}
2022-07-01 13:05:05 +00:00
3e4e05d28b Use generate_stubs without //base dependency
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}
2022-07-01 10:50:54 +00:00
2bc41bc980 Detach RemoteBitrateEstimator interface from Module
Bug: webrtc:7219
Change-Id: I8302c5044582d73b0918013a0df89b9390788728
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267140
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37393}
2022-07-01 10:17:40 +00:00
3c24c096ef Add support for scalability modes L2T3 and S2T3
Bug: webrtc:11607
Change-Id: I1d0bd171564d2852f2f6ee2bbee26c7a1c0e1c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267103
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37389}
2022-07-01 08:17:04 +00:00
ed665521e4 in RtpRtcp configuration delete unused remote bitrate estimator
No code sets that configuration field.

Bug: None
Change-Id: Idd611d15ec54b3bd9115eac77d2222b97620d675
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267180
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37382}
2022-06-30 14:07:49 +00:00
fb60796b64 fec: simplify fec generation
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}
2022-06-30 11:53:58 +00:00
3a08d2a42d Reland "Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface"
This reverts commit 6769e95bbc0e4f29c482ee1356b3e250df1fde3b.

Reason for revert: downstream code adjusted

Original change's description:
> Revert "Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface"
>
> This reverts commit 08c7e7589218da1f26e9621e703dd790b4e7e7d7.
>
> Reason for revert: breaks downstream tests
>
> Original change's description:
> > Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface
> >
> > Bug: None
> > Change-Id: I47b7c83320b0c7327c0d2ee59f7a0a30704cd331
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266540
> > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37354}
>
> Bug: None
> Change-Id: Ia355be085890856141fc943432f6e2edef1c0900
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267065
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37361}

Bug: None
Change-Id: Ifaf8ff84a37a768b388b1f79c8c7829390d1905e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267104
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37367}
2022-06-29 09:38:16 +00:00
cc7bd85748 Don't add libopus to public_deps, its headers are only used directly
Bug: webrtc:8603
Change-Id: I2ce1f96a80dd23e420b3693b899d2b14382fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266765
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Oleh Prypin <oprypin@google.com>
Cr-Commit-Position: refs/heads/main@{#37363}
2022-06-28 19:13:14 +00:00
6769e95bbc Revert "Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface"
This reverts commit 08c7e7589218da1f26e9621e703dd790b4e7e7d7.

Reason for revert: breaks downstream tests

Original change's description:
> Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface
>
> Bug: None
> Change-Id: I47b7c83320b0c7327c0d2ee59f7a0a30704cd331
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266540
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37354}

Bug: None
Change-Id: Ia355be085890856141fc943432f6e2edef1c0900
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267065
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37361}
2022-06-28 17:07:44 +00:00
8783c678a5 delay estimator: Look for early reverberation
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}
2022-06-28 15:16:03 +00:00
7534ebd2bf Revert "Reland "Reland "Delete old Android ADM."""
This reverts commit db30009304ab97a5fde02977ed1239aa249e2656.

Reason for revert: ... and it's out again :(
 
Original change's description:
> Reland "Reland "Delete old Android ADM.""
>
> This reverts commit 38a28603fd7b2eec46a362105b225dd6f08b4137.
>
> Reason for revert: Attempt to reland, now that WebRTC dependency cycle has been broken.
>
> Original change's description:
> > Revert "Reland "Delete old Android ADM.""
> >
> > This reverts commit 6e4d7e606c4327eaa9298193e22794fcb9b30218.
> >
> > Reason for revert: Still breaks downstream build (though in a different way this time)
> >
> > Original change's description:
> > > Reland "Delete old Android ADM."
> > >
> > > This is a reland of commit 4ec3e9c98873520b3171d40ab0426b2f05edbbd2
> > >
> > > Original change's description:
> > > > Delete old Android ADM.
> > > >
> > > > The schedule move Android ADM code to sdk directory have been around
> > > > for several years, but the old code still not delete.
> > > >
> > > > Bug: webrtc:7452
> > > > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#37174}
> > >
> > > Bug: webrtc:7452
> > > Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#37236}
> >
> > Bug: webrtc:7452
> > Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Owners-Override: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37242}
>
> Bug: webrtc:7452
> Change-Id: I6946d0fc28cf4c08387e451e6a07765f7410ce7c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266980
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37356}

Bug: webrtc:7452
Change-Id: I1ef4004e89c8bea322bda0dc697a7ba45abeffcc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267067
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37359}
2022-06-28 14:37:43 +00:00
3f659b1b3c Continue looking for frames after padding packets
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}
2022-06-28 13:03:03 +00:00
db30009304 Reland "Reland "Delete old Android ADM.""
This reverts commit 38a28603fd7b2eec46a362105b225dd6f08b4137.

Reason for revert: Attempt to reland, now that WebRTC dependency cycle has been broken.

Original change's description:
> Revert "Reland "Delete old Android ADM.""
>
> This reverts commit 6e4d7e606c4327eaa9298193e22794fcb9b30218.
>
> Reason for revert: Still breaks downstream build (though in a different way this time)
>
> Original change's description:
> > Reland "Delete old Android ADM."
> >
> > This is a reland of commit 4ec3e9c98873520b3171d40ab0426b2f05edbbd2
> >
> > Original change's description:
> > > Delete old Android ADM.
> > >
> > > The schedule move Android ADM code to sdk directory have been around
> > > for several years, but the old code still not delete.
> > >
> > > Bug: webrtc:7452
> > > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#37174}
> >
> > Bug: webrtc:7452
> > Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37236}
>
> Bug: webrtc:7452
> Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37242}

Bug: webrtc:7452
Change-Id: I6946d0fc28cf4c08387e451e6a07765f7410ce7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37356}
2022-06-28 12:58:23 +00:00
08c7e75892 Detach RemoteEstimatorProxy from RemoteBitrateEstimator interface
Bug: None
Change-Id: I47b7c83320b0c7327c0d2ee59f7a0a30704cd331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266540
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37354}
2022-06-28 11:31:23 +00:00
7769dc87d7 Detach legacy RtpRtcp from Module interface
Bug: webrtc:7219
Change-Id: I5faf8f68b043994a86d227926c13b07d0141f382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267063
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37353}
2022-06-28 11:17:43 +00:00
ce80886bf2 Remove video_coding dependency on ProcessThread and Module
Bug: webrtc:7219
Change-Id: I360f7df5554389274fcaef64070b9441ce0ef984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266486
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@{#37351}
2022-06-28 10:04:23 +00:00
e58f1991dc Add Timestamp -> AbsoluteSendTime conversion function
instead of ms -> AbsoluteSendTime helper

Bug: webrtc:13757
Change-Id: I57389a66a43b4f4838023f9c224a985f2cd57107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266024
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37350}
2022-06-28 09:26:42 +00:00
7a66900683 Delete rtc_base/atomic_ops.h
Bug: webrtc:9305
Change-Id: I3e8b0db03b84b5361d63db31ee23e6db3deabfe4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266497
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37348}
2022-06-28 08:32:13 +00:00
450da27933 Wayland screencast: use stream size to adjust source stride of DMA-BUFs
In commit a6ed749b12c63d252c6d893d5b5b62fcf35773d9 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}
2022-06-27 20:51:22 +00:00
6e03c98873 Make "failed to query DMA-BUF modifiers" just warning message
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}
2022-06-27 19:26:42 +00:00
fe053426e2 Add missing lib dependency on X11.
Bug: b/36882554
Change-Id: I723d8c2876b963b43429d4fa322d6e09380d8f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267001
Reviewed-by: Oleh Prypin <oprypin@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37339}
2022-06-27 12:27:27 +00:00
e6ac4b263e Remove desktop_capture_generic target
This was a backwards compatible target.

Bug: b/36882554
Change-Id: I1faaf89656a540311af8c68ddd43df6d54ae87b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267000
Reviewed-by: Oleh Prypin <oprypin@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37338}
2022-06-27 12:03:46 +00:00
bf0da440ea Refactor desktop_capture build.
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}
2022-06-27 10:57:03 +00:00
fb698490bf Add missing absl dep.
Bug: b/36882554
Change-Id: I37e13338af8a2c75f56df283d20b1be4579074b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266763
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@google.com>
Cr-Commit-Position: refs/heads/main@{#37334}
2022-06-27 09:33:01 +00:00