Commit graph

1092 commits

Author SHA1 Message Date
Mirko Bonadei
f135800b5c Roll chromium_revision 47dc8e2f50..680b7dae9d (912091:912910)
Change log: 47dc8e2f50..680b7dae9d
Full diff: 47dc8e2f50..680b7dae9d

Changed dependencies
* src/base: 959457e3f3..4bcc0feab1
* src/build: a0d51919fe..02ca29f24d
* src/buildtools: 6810b870e0..6f9b470988
* src/buildtools/third_party/libc++abi/trunk: 671803fd96..8452f0657d
* src/ios: 6a9bd7348f..4bdd6cc72d
* src/testing: c0ea7c3386..e3201c323d
* src/third_party: 56c558ed2e..0f2f057998
* src/third_party/androidx: v5A41FDtUTUgWmjkgJS42X4yMcKx2zbPp8fWod32rhsC..8ehN1uRQQBM3VrBh28TpSvhV4AmGQRMCfN6Fm1L5y9QC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/126f6a8996..77a7089299
* src/third_party/depot_tools: 0c42eff6d1..9a0189cd7a
* src/third_party/perfetto: 303b88cfe5..95e9c5e207
* src/tools: b54abb9ed0..bb864a1e83
DEPS diff: 47dc8e2f50..680b7dae9d/DEPS

Clang version changed llvmorg-14-init-1002-gb5e470aa:llvmorg-14-init-1380-gee659383
Details: 47dc8e2f50..680b7dae9d/tools/clang/scripts/update.py

TBR=mbonadei@webrtc.org,
BUG=None


Fix roll

Change-Id: Ie0b20fe417ce893b6905f0b3c02053e09b83de8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229102
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34792}
2021-08-18 10:22:27 +00:00
Mirko Bonadei
24e79f6962 Add missing header (for unique_ptr).
Bug: None
Change-Id: I2ee004ac4feca9a0c25551fc1709069e8df836b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229026
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34783}
2021-08-17 12:19:01 +00:00
Nico Weber
19214818d7 Fix some -Wunreachable-code-aggressive warnings
Bug: chromium:1066980
Change-Id: I24fea094f28577799c5fcbcf2e9657ffa9bfd076
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228760
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34766}
2021-08-16 11:09:16 +00:00
Niels Möller
4f776ac7de Use make_ref_counted in AudioProcessingBuilder
Bug: webrtc:12701
Change-Id: I51ca5a54f812a1620ee2e6605c9ff67b92e2a5f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34725}
2021-08-11 15:40:28 +00:00
Alessio Bazzica
0441bb625f APM: add HW-only denormal disabler
Denormal numbers (see [1]) may origin in APM when the input is zeroed
after a non-zero signal. In extreme cases, instructions involving
denormal operands may run as much as 100 times slower, which seems to
be the case (to some extent) of crbug.com/1227566.

This CL adds a class that disables denormals only via hardware on x86
and on ARM. The class is used in APM and it is an adaption of [2].

Tested: appr.tc call on Chromium (Win, Mac)

[1] https://en.wikipedia.org/wiki/Denormal_number
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/platform/audio/denormal_disabler.h

Fixed: chromium:1227566
Change-Id: I0ed2eab55dc597529f09f93c26c7a01de051fdbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227768
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34701}
2021-08-10 15:48:22 +00:00
Artem Titov
cfea2182f8 Use backticks not vertical bars to denote variables in comments
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
Artem Titov
0b489303d2 Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
Bug: webrtc:12338
Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34690}
2021-08-09 21:49:02 +00:00
Peter Kasting
55ec1a43bb Fix some instances of -Wunused-but-set-variable.
Bug: chromium:1203071
Change-Id: I1ef3c8fd1f8e2bbf980d5d5217257e919f4564c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226961
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34579}
2021-07-28 02:08:30 +00:00
Niels Möller
5b747233a3 Add method Mutex::AssertHeld
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.

Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
2021-07-27 07:46:32 +00:00
Tony Herre
b0ed12099f Update links to point at main branch
As part of go/coil update code search links to not point to the
"master" branch.

Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
2021-07-22 16:41:26 +00:00
Hanna Silen
e7e9292fe8 Analog AGC: Add clipping rate metrics
Add a histogram WebRTC.Audio.Agc.InputClippingRate and logging of
max clipping rate in AgcManagerDirect.

Bug: webrtc:12774
Change-Id: I4a72119b65ad032fc50672e2a8fb4a4d55e1ff24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225264
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34450}
2021-07-10 13:13:46 +00:00
Peter Kasting
286b1db1b2 Fix -Wunreachable-code-aggressive.
Bug: chromium:1066980
Change-Id: I6888ea1fbc458c9b3063b3f60a7732af16ab5fc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224266
Reviewed-by: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34393}
2021-06-30 11:14:37 +00:00
Alessio Bazzica
42dacda82c AGC analog clipping predictor: integrate evaluator
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the
possibility to run the predictor without affecting the analog gain
adjustment process.

The evaluator is used to compute precision, recall and F1 score.
F1 score and the measured clipping prediction intervals are logged as
`WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval`
histograms respectively.

Bug: webrtc:12774
Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34327}
2021-06-17 16:16:53 +00:00
Alessio Bazzica
5b9d0c70c2 AGC1 add clipping predictor evaluator
Observes clipping predictions and detections and computes evaluation
metrics for the predictor.

Bug: webrtc:12774
Change-Id: I83f5942a3b6491de288510f2200f2f5c0e099bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221619
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34305}
2021-06-16 15:33:51 +00:00
Alessio Bazzica
98ff0280ce AGC analog ClippingPredictor refactoring 2/2
Uunit test code readability improvements.

Bug: webrtc:12774
Change-Id: I66f552d23680ddb03824618dab869946e0940334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34300}
2021-06-16 10:28:57 +00:00
Alessio Bazzica
b237a87a25 AGC analog ClippingPredictor refactoring 1/2
- ClippingPredictor API and docstring changes
- Unified ClippingPredictor factory function

Bug: webrtc:12774
Change-Id: Iafaddae52addc00eb790ac165bf407a4bdd1cb52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221540
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34279}
2021-06-14 12:21:31 +00:00
Gustaf Ullberg
a63d152423 AEC3: Unbounded echo spectrum for dominant nearend detection.
The dominant nearend detector uses the residual echo spectrum for
determining whether in nearend state. The residual echo spectrum in
computed using the ERLE. To reduce the risk of echo leaks in the
suppressor, the ERLE is capped. While minimizing echo leaks, the
capping of the ERLE can affect the dominant nearend classification
negatively as the residual echo spectrum is often over estimated.

This change enables the dominant nearend detector to use a residual
echo spectrum computed with a virtually non-capped ERLE. This ERLE
is only used for dominant nearend detection and leads to increased
transparency.

The feature is currently disabled by default and can be enabled
with the field trial "WebRTC-Aec3UseUnboundedEchoSpectrum".

Bug: webrtc:12870
Change-Id: Icb675c6f5d42ab9286e623b5fb38424d5c9cbee4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221920
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34270}
2021-06-11 13:30:00 +00:00
Hanna Silen
a004715d13 Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect
Integrate ClippingPredictor in AgcManagerDirect and
AudioProcessingImpl. Disable functionality by default.

Bug: webrtc:12774
Change-Id: Ic67a47f439c89b75066506fca8acaf636d8812f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221100
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34207}
2021-06-03 02:35:05 +00:00
Hanna Silen
4b3a06139b Add ClippingPredictor implementation
Add implementation for clipping prediction and clipped level step estimation.

Bug: webrtc:12774
Change-Id: I855d22980302aac7d49078ca29755f9422af9cb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220935
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34206}
2021-06-02 22:45:46 +00:00
Hanna Silen
a43953a518 Add ClippingPredictor config in AudioProcessing config
Bug: webrtc:12774
Change-Id: Id8cdb6b5499a22cbca40d424cf936f81c1e7d8d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221104
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34204}
2021-06-02 16:16:25 +00:00
Per Åhgren
cbdbb8c166 Add ability to adjust the suppressor smoothing in AEC3
Bug: b/177359044
Change-Id: I5eddb6fa6f01aa14426161204e37a9097b182234
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217889
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34203}
2021-06-02 15:21:35 +00:00
Hanna Silen
ea72ee6350 Add ClippingPredictorLevelBuffer circular buffer.
Bug: webrtc:12774
Change-Id: I2b26660e3fe051ab358dd5298ba5098f275943da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219631
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34167}
2021-05-31 14:04:54 +00:00
Hanna Silen
b8dc7fa5a6 Make AgcManagerDirect clipping parameters configurable
Bug: webrtc:12774
Change-Id: I99824b5aabe6f921a5db425dd1c1c1d4c606186c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219681
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34069}
2021-05-20 16:53:59 +00:00
Gustaf Ullberg
398def6828 Improvements to AEC3 logging to simplify debugging
Adds log messages for
- AEC3 creation
- Transparent mode implementation selection
- Config parameter changes via RetrieveFieldTrialValue

Bug: webrtc:8671
Change-Id: Ibb1e76d66975a3a3c1227e31b9916a17b76e6c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219468
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34056}
2021-05-19 14:28:44 +00:00
Gustaf Ullberg
aeb8ce882f AEC3: Change adaptation speed of the matched filter after a delay is found
This change enables the use of two different adaptation speeds of the
matched filter of the delay estimator of AEC3.

One speed is used when no delay has been found, and one is used after a
reliable delay has been found. The purpose is to use a slower adaptation
speed to reduce the risk of divergence during double-talk without
slowing down the search for the initial delay.

The CL prepares for experimentation by adding field trials for
controlling the two adaptation speeds.

Bug: webrtc:12775
Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34055}
2021-05-19 13:37:13 +00:00
Per Åhgren
91a892f8ed Add ability to dump the coarse filter in the echo subtractor
Bug: b/155316201
Change-Id: I008cdf1531af3c3c0fff4ce19ad5dd74f8e73f65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217891
Reviewed-by: Sam Zackrisson <saza@google.com>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34047}
2021-05-19 08:24:16 +00:00
Gustaf Ullberg
a399c823bb Field trial to disable the transient suppressor
This change adds the field trial "WebRTC-TransientSuppressorForcedOff"
that can be used to disable the transient suppressor (removal of
keyboard typing sounds). The field trial can be enabled by users via
command-line or via experimentation.

Bug: chromium:1186705
Change-Id: I7272df6a20fbbee24a7ba0904502c76bd775d275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219282
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34038}
2021-05-18 12:34:37 +00:00
Alessio Bazzica
7ddadbc108 APM: dump GainController1::AnalogGainController in Config::ToString
Update `AudioProcessing::Config::ToString()` to also dump the config
from `AnalogGainController` which is missing.

Bug: webrtc:7494
Change-Id: Iea5dab1f6abb9ec8581ce690a2a119f202b4d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219082
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34025}
2021-05-17 16:37:37 +00:00
Markus Handell
ad5037b4a8 Reland "Refactor the PlatformThread API."
This reverts commit 793bac569f.

Reason for revert: rare compilation error fixed

Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> >   a) Handles for detached threads are simply closed where applicable.
> >   b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
2021-05-07 14:14:43 +00:00
Alessio Bazzica
dfc11d55af Add RTC_EXPORT to APM config structs with overloaded operators
Bug: webrtc:7494
Change-Id: I9b627709d8c5bb47d73fd7981259a95e6b51e16c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217766
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33946}
2021-05-07 11:31:18 +00:00
Guido Urdaneta
793bac569f Revert "Refactor the PlatformThread API."
This reverts commit c89fdd716c.

Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?

Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
>   a) Handles for detached threads are simply closed where applicable.
>   b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}

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

TBR=handellm@webrtc.org

Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
2021-05-06 14:22:57 +00:00
Markus Handell
c89fdd716c Refactor the PlatformThread API.
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.

Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
  a) Handles for detached threads are simply closed where applicable.
  b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.

Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
2021-05-05 09:59:07 +00:00
Alessio Bazzica
a2efd15627 AGC2 config operator= and unit tests update
Bug: webrtc:7494
Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@google.com>
Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:37:17 +00:00
Alessio Bazzica
d66a60597d AGC2 adaptive digital dry run mode
Add the option to run the adaptive digital controller of AGC2 without
side-effects - i.e., no gain applied.

Tested: adapation verified during a video call in chromium

Bug: webrtc:7494
Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33875}
2021-04-29 16:05:57 +00:00
Tommi
87f7090fd9 Replace more instances of rtc::RefCountedObject with make_ref_counted.
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.

Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
2021-04-27 17:01:59 +00:00
Markus Handell
97c4458c8f PlatformThread: add support for detached threads.
The change introduces support for detachable PlatformThreads, for which
the Stop() call doesn't wait until the thread has finished executing.

The change also introduces rtc::ThreadAttributes that carries priority
and detachability thread attributes. It additionally refactors all
known use to use the new semantics.

Bug: b:181572711, webrtc:12659
Change-Id: Id96e87c2a0dafabc8047767d241fd5da4505d14c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214704
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33796}
2021-04-21 07:44:31 +00:00
Gustaf Ullberg
437d129ef5 AEC3: Avoid overcompensating for render onsets during dominant nearend
The ERLE is used to estimate residual echo for echo suppression. The
ERLE is reduced during far-end offset to avoid echo leakage. When there
is a strong near-end present this can cause unnecessary transparency loss.

This change adds an ERLE estimation that does not compensate for onsets and
uses it for residual echo estimation when the suppressor considers the near-end to be dominant.

Bug: webrtc:12686
Change-Id: Ida78eeacf1f95c6e62403f86ba3f2ff055898a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215323
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33786}
2021-04-20 12:33:02 +00:00
Alessio Bazzica
980c4601e1 AGC2: retuning and large refactoring
- Bug fix: the desired initial gain quickly dropped to 0 dB hence
  starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
  of adjacent speech frames, the gain applier temporarily allows a
  faster gain increase to deal with a longer time spent waiting for
  enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
  simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming

Bug: webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33736}
2021-04-14 19:01:01 +00:00
Alessio Bazzica
61982a7f2d AGC2 lightweight noise floor estimator
The current noise level estimator has a bug due to which the estimated
level decays to the lower bound in a few seconds when speech is observed.
Instead of fixing the current implementation, which is based on a
stationarity classifier, an alternative, lightweight, noise floor
estimator has been added and tuned for AGC2.

Tested on several AEC dumps including HW mute, music and fast talking.

Bug: webrtc:7494
Change-Id: Iae4cff9fc955a716878f830957e893cd5bc59446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214133
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33733}
2021-04-14 15:56:41 +00:00
Per Åhgren
dea5721efb Adding g3doc for AudioProcessingModule (APM)
Bug: webrtc:12569
Change-Id: I8fa896a5afa9791ad6d8c2b5011d1e75ca068df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215141
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33726}
2021-04-14 09:40:25 +00:00
Gustaf Ullberg
1fad94f502 Remove ErleUncertainty
Erle Uncertainty changes the residual echo computation during saturated
echo. However, the case of saturated echo is already handled by the
residual echo estimator causing the ErleUncertainty to be a no-op.

The change has been tested for bit-exactness.

Bug: webrtc:8671
Change-Id: I779ba67f99f29d4475a0465d05da03d42d50e075
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215072
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33719}
2021-04-14 07:01:14 +00:00
Alessio Bazzica
11bd143974 AGC2 add an interface for the noise level estimator
Done in preparation for the child CL which adds an alternative
implementation.

Bug: webrtc:7494
Change-Id: I4963376afc917eae434a0d0ccee18f21880eefe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214125
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33646}
2021-04-08 07:34:22 +00:00
Alessio Bazzica
70b775d77f AGC2 noise estimator code style improvements
Code style improvements done in preparation for a bug fix (TODO added)
which requires changes in the unit tests.

Note that one expected value in the unit tests has been adjusted since
the white noise generator is now instanced in each separate test and
therefore, even if the seed remained the same, the generated sequences
differ.

Bug: webrtc:7494
Change-Id: I497513b84f50b5c66cf6241a09946ce853eb1cd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214122
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33636}
2021-04-07 11:57:55 +00:00
Alessio Bazzica
8aaa604375 AGC2 new data dumps
Bug: webrtc:7494
Change-Id: Id288dd426e1c2754805bc548fbffe0eaeaacf3da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213420
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33605}
2021-03-31 14:55:42 +00:00
Alessio Bazzica
841d74ea80 AGC2 periodically reset VAD state
Bug: webrtc:7494
Change-Id: I880ef3991ade4e429ccde843571f069ede149c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213342
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33604}
2021-03-31 14:15:10 +00:00
Alessio Bazzica
b995bb86df AGC2 size_t -> int
Bug: webrtc:7494
Change-Id: I5ecf242e83b509931c1764a37339d11506c5afc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213341
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33600}
2021-03-31 11:18:30 +00:00
Per Åhgren
19775cbd29 Reland "Reduce complexity in the APM pipeline when the output is not used"
This is a reland of aa6adffba3

What was changed in the reland is that the merging of the bands is
excluded from the code that is not run when the output is not used.
I.e., the merging is always done.

This is important to have since some clients may apply muting before APM,
and still flag to APM that the signal is muted. If the merging is not
always done, those clients will get nonzero output from APM during muting.


Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}

Bug: b/177830919
Change-Id: Ib74dd1cefa173d45101e26c4f2b931860abc6d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33478}
2021-03-16 09:33:02 +00:00
Per Åhgren
15179a9986 Allowing reduced computations in the noise suppressor when the output is not used
This CL adds functionality in the noise suppressor that allows the
computational complexity to be reduced when the output of APM is not used.

Bug: b/177830919
Change-Id: I849351ba9559fae770e4667d78e38abde5230eed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211342
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33477}
2021-03-16 09:28:42 +00:00
Per Åhgren
8ee1ec82e4 Allowing reduced computations in the AEC3 when the output is not used
This CL adds functionality in AEC3 that allows the computational
complexity to be reduced when the output of APM is not used.

Bug: b/177830919
Change-Id: I08121364bf966f34311f54ffa5affbfd8b4db1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211341
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33476}
2021-03-16 09:16:32 +00:00
Per Åhgren
db5d728878 Add refined handling of the internal scaling of the audio in APM
This CL adds functionality that allows adjusting the audio levels
internally in APM. The main purpose of the functionality is to allow
APM to optionally be moved to an integration that does not provide an
analog gain to control, and the implementation of this has been
tailored specifically to meet the requirements for that.

More specifically, this CL does
-Add a new variant of the pre-amplifier gain that is intended to replace
 the pre-amplifier gain (but at the moment can coexist with that). The
 main differences with the pre-amplifier gain is that an attenuating
 gain is allowed, the gain is applied jointly with any emulated analog
 gain, and that its packaging fits better with the post gain.
-Add an emulation of an analog microphone gain. The emulation is
 designed to match the analog mic gain functionality in Chrome OS (which
 is digital) but should be usable also on other platforms.
-Add a post-gain which is applied after all processing has been applied.
 The purpose of this gain is for it to work well with the integration
 in ChromeOS, and be used to compensate for the offset that there is
 applied on some USB audio devices.


Bug: b/177830918
Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33466}
2021-03-15 19:12:02 +00:00