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Author SHA1 Message Date
Andrey Logvin
fef0026f2f Revert "Reland "remove NV12 to I420 conversion in webrtc AV1 Encoder."" 
This reverts commit d7031692e3ba9eed78ead07f8bf34a847ca1fce6.

Reason for revert: Breaks downstream project

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Reason for revert: speculative revert: breaks downstream project

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

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

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

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

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

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

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

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

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

Reason for revert: downstream problem addressed

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

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

Reason for revert: Causes problems downstream:

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

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

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
 2022-02-26 10:35:13 +00:00
Danil Chapovalov
385eb9714d Represent RtpPacketToSend::capture_time with Timestamp 
Bug: webrtc:13757
Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36083}
 2022-02-25 16:44:07 +00:00
Jan Grulich
0825daf2ed PipeWire capturer: search for epoxy headers 
We actually use headers from libepoxy (it's part of the sysroot) so this
was removed accicentally in one of previous changes and it just
magically worked as we include those headers with their full path

Bug: webrtc:13429
Change-Id: I4f5684521a76287a725272ce3833daae673d9332
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252002
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36073}
 2022-02-25 01:20:32 +00:00
Niels Möller
ecd5ba15cb Fix missing include of rtc_base/system/no_unique_address.h 
Bug: None
Change-Id: I047c456cde647282824e8c51122ae53bef7cb7b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252440
Auto-Submit: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36070}
 2022-02-24 19:25:12 +00:00
Danil Chapovalov
3ed1dbb56e In RtpPacketReceived delete deprecated accessors for handling time in ms 
Bug: None
Change-Id: I02a43a16e8d9bf3a1e2c9f6442a1c119620e1288
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252286
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36067}
 2022-02-24 15:37:51 +00:00
“Michael
3147e29c4e Refactor encoder-complexity param in VideoCodec w/backward compatibility 
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.

Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
 2022-02-21 19:40:44 +00:00
Danil Chapovalov
808531653e In RtcpTransceiver implement handling incoming RRTR 
Bug: webrtc:8239
Change-Id: I4a469b6a0c2e387e35262798f4686fbf310d00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251902
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36037}
 2022-02-21 14:07:34 +00:00
Henrik Lundin
c4ed5f0b1a Adding fuzzer for G711/PCM u/A decoders and fixing a fuzzer problem 
Bug: chromium:1279775
Change-Id: I8cc3f5fe25b9e707e9d171251026bd5a8bad5da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251844
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36036}
 2022-02-21 13:35:24 +00:00
Henrik Lundin
ba2677061a Add fuzzer test for G722 and fix a fuzzer problem 
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().


Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
 2022-02-21 10:16:47 +00:00
Alessio Bazzica
feac97bb25 AgcManagerDirect: improve AgcMinMicLevelExperimentEnabled50 test 
Also test the field trial with valid parameter and non-empty suffix.

Bug: webrtc:7494
Change-Id: I3d871b41dd71c951ac56e180b3c09cda4c3627d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251441
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36031}
 2022-02-18 17:52:01 +00:00
Danil Chapovalov
8968bcae8d In RtcpTransceiver avoid generating rtcp sender reports for inactive senders 
Bug: webrtc:8239
Change-Id: I97d50c628db04c56669179ab7039a3fe3bd61d34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251901
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36030}
 2022-02-18 16:44:52 +00:00
Danil Chapovalov
27d5f14cf2 in RTPSender disallow enabling misconfigured rtx 
Bug: None
Change-Id: Id94771626ef723212e4d92d9093af3ec9e647990
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251780
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36020}
 2022-02-16 16:08:40 +00:00
Sergey Silkin
454d2309de Add bitrate adaptation tests 
Bug: none
Change-Id: I3e2c503efc7a85a3daaa40cd8118c1b02d3b81cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251680
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36013}
 2022-02-16 10:25:13 +00:00
Alessio Bazzica
1db0a261ca Reland "Reland "Remove unused APM voice activity detection sub-module"" 
This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1.

Reason for revert: downstream fixed (see https://chromium-review.googlesource.com/c/chromium/src/+/3461371)

Original change's description:
> Revert "Reland "Remove unused APM voice activity detection sub-module""
>
> This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.
>
> Reason for revert: Breaks chromium roll, see 
> https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview
>
> https://chromium-review.googlesource.com/c/chromium/src/+/3461512
>
> Original change's description:
> > Reland "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
> >
> > Reason for revert: dependency in a downstream project removed
> >
> > Original change's description:
> > > Revert "Remove unused APM voice activity detection sub-module"
> > >
> > > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> > >
> > > Reason for revert: breaking downstream projects
> > >
> > > Original change's description:
> > > > Remove unused APM voice activity detection sub-module
> > > >
> > > > API changes:
> > > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > > - cricket::AudioOptions::typing_detection deprecated
> > > > - webrtc::StatsReport::StatsValueName::
> > > >   kStatsValueNameTypingNoiseState deprecated
> > > >
> > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > > >
> > > > Bug: webrtc:11226,webrtc:11292
> > > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#35975}
> > >
> > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> > >
> > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:11226,webrtc:11292
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35977}
> >
> > # Not skipping CQ checks because this is a reland.
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35984}
>
> TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35990}

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

Bug: webrtc:11226,webrtc:11292
Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36012}
 2022-02-16 08:41:30 +00:00
Jan Grulich
5ae9b260ff Implement MouseCursorMonitorPipeWire to track cursor changes separately 
Current implementation has mouse cursor as part of the screen itself
which means that everytime a cursor changes location, we have to update
whole screen content, which brings unnecessary load overhead. Using our
own mouse cursor monitor implementation allows us to track only mouse
cursor changes and update them separately for much better performance.

Bug: webrtc:13429
Change-Id: I224e9145f0bc7e45eafe4490de160f2ad4c8b545
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244507
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36011}
 2022-02-15 23:03:41 +00:00
Alex Cooper
0c6e34ce5c Ensure PipeWire doesn't use a Null SourceId 
This has mostly seemed to work fine until now; but there's a collision
happening in chromium where if the source is being shown in the Window
Picker it collides with the (also null) Dialog ID and is ignored. While
we could patch that code to not count Null as a collision, there's the
potential for other (future) code to simply ignore a capture source
that it thinks is Null.

Fixed: chromium:1295375
Change-Id: I4356084f0af97f4d56632938b0d9a24d327f7107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251500
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36008}
 2022-02-15 20:03:33 +00:00
Sergey Silkin
0b02d637c0 Calculate max/avg encode/decode latency in codec tests 
Bug: none
Change-Id: Ie42461dd06b1764c99308393477921ea25319ab4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251687
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36007}
 2022-02-15 18:14:41 +00:00
Henrik Lundin
ac341df436 Adding fuzzer for PCM16b decoder and fixing a fuzzer problem 
Bug: chromium:1280852
Change-Id: I7f6c5de86ceee01156743c0389c59f875e53bb5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251580
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36005}
 2022-02-15 15:59:01 +00:00
Evan Shrubsole
f7a1937e70 Add FrameBufferProxy test for low-latency renderer 
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.

* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.

R=kron@webrtc.org

Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
 2022-02-15 08:30:51 +00:00
Mirko Bonadei
405ac4e840 Add objc_class_prefix to the Audio Network Adaptor proto. 
WANA: WebRTC Audio Network Adaptor

No-Try: True
Bug: None
Change-Id: I291e02ab70323ecc45d87cea0ea8d7e8cb62db9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249784
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35998}
 2022-02-14 21:04:20 +00:00
Jan Grulich
2f194e0325 PipeWire capturer: Import DMA-BUFs with correct render node 
With more GPUs it might happen that server used different render
node from the one we pick from the list. This would cause DMA-BUF to
fail to import so we use Wayland client library to obtain wl_display in
order to initialize EGLDisplay using same render node and have previous
approach as a fallback. Also everyone else uses EGL_LINUX_DMA_BUF_EXT
target for importing EGLImages from DMA-BUF file descriptors so use it
as well to be sure we import buffers same way as they are produced.

Bug: chromium:1290566
Bug: webrtc:13429
Change-Id: I32bbb0bdb28c08b6e7fcb3f94009f82a2041b6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250661
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35997}
 2022-02-14 20:08:50 +00:00
Danil Chapovalov
f2b987377b in RtcpTransceiver implement sending rtcp sender reports 
Bug: webrtc:8239
Change-Id: Id3298bf4e0eb18a3fc8072fb19416e67a126705f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249788
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35995}
 2022-02-14 15:58:40 +00:00
Henrik Boström
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module"" 
This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

Reason for revert: Breaks chromium roll, see 
https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview

https://chromium-review.googlesource.com/c/chromium/src/+/3461512

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
 2022-02-14 12:25:51 +00:00
Sergey Silkin
eb6c6fcf27 Fix delta frame delay calculation 
The issue was introduced in https://webrtc-review.googlesource.com/c/src/+/132460.

Bug: webrtc:10412
Change-Id: I92d1bd2be63ea34d150145cec63c282f7aa49ce8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35989}
 2022-02-14 11:15:50 +00:00
Alessio Bazzica
54d1344d98 Reland "Remove unused APM voice activity detection sub-module" 
This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
>
> Reason for revert: breaking downstream projects
>
> Original change's description:
> > Remove unused APM voice activity detection sub-module
> >
> > API changes:
> > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > - webrtc::AudioProcessingStats::voice_detected deprecated
> > - cricket::AudioOptions::typing_detection deprecated
> > - webrtc::StatsReport::StatsValueName::
> >   kStatsValueNameTypingNoiseState deprecated
> >
> > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35975}
>
> TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35977}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
 2022-02-13 14:02:08 +00:00
Alessio Bazzica
a751f167c6 Revert "Remove unused APM voice activity detection sub-module" 
This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.

Reason for revert: breaking downstream projects

Original change's description:
> Remove unused APM voice activity detection sub-module
>
> API changes:
> - webrtc::AudioProcessing::Config::VoiceDetection removed
> - webrtc::AudioProcessingStats::voice_detected deprecated
> - cricket::AudioOptions::typing_detection deprecated
> - webrtc::StatsReport::StatsValueName::
>   kStatsValueNameTypingNoiseState deprecated
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35975}

TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
 2022-02-11 12:15:44 +00:00