Commit graph

944 commits

Author SHA1 Message Date
Alessio Bazzica
8845f7e32b AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (3/3)
This is the last CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.

This CL adds a second state property to hold temporary updates and a
counter for consecutive speech frames. When enough speech frames are
observed, the reliable state is updated; otherwise, the temporary state
is discarded.

The default for `AdaptiveModeLevelEstimator::min_consecutive_speech_frames_`
is 1, which means that the new feature is disabled.

Tested:
- Bit-exactness verified with audioproc_f
- Not bit-exact if `min_consecutive_speech_frames_` set to 10

Bug: webrtc:7494
No-Try: True
Change-Id: I0daa00e90c27c418c00baec39fb8eacd26eed858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185125
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32250}
2020-09-30 13:08:08 +00:00
Alessio Bazzica
9cf3ea0ab2 AGC2 AdaptiveModeLevelEstimator: minor code improvements
- State -> LevelEstimatorState
- Mark two methods as const
- Call DumpDebugData() in one place
- DumpDebugData: don't check if data dumper is provided
- Add LevelEstimatorState::operator==

The changes will reduce clutter in follow up CL.

Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".

Bug: webrtc:7494
Change-Id: If39ce4b787069bef4af910d718cdfae3af1784a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185811
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32247}
2020-09-30 10:31:30 +00:00
Alessio Bazzica
307fab9e02 AGC2 AdaptiveModeLevelEstimator: cache last level estimate
`AdaptiveModeLevelEstimator::last_level_dbfs_` doesn't need to be optional.

Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".

Bug: webrtc:7494
Change-Id: Id5b409ca5cb5f11ed132c861b7995b9721e167bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185809
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32237}
2020-09-29 15:45:22 +00:00
Alessio Bazzica
435f279433 AGC2 remove incorrect field trial parsing functions
The AGC2 params must be exposed via
`AudioProcessing::Config::GainController2` and the Finch params must
be parsed in blink (see [1]).

Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".

[1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/blink/renderer/modules/mediastream/media_stream_audio_processor.cc;l=593-596?q=HybridAgc&start=11

Bug: webrtc:7494
Change-Id: Ie7bd1bef1d6caf7d2b20600a1626c12171b67c82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185044
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32230}
2020-09-29 13:33:15 +00:00
Alessio Bazzica
56f63c3e7e AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (2/3)
This is the second CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.

In this CL, the `SaturationProtector` class has been replaced by a
struct that define the state and two functions to change it.
This is done in order to use the saturation protector state in
`AdaptiveModeLevelEstimator::State` and will allow to add a
temporary state in `AdaptiveModeLevelEstimator` (see the child CL).

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: Ic5ecd1e174010656ed20664ef7b7e5798ebb7978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185041
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32226}
2020-09-29 11:02:10 +00:00
Niels Möller
de95329daa Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.

Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
2020-09-29 10:19:20 +00:00
Gustaf Ullberg
46be80d349 Remove echo suppression in transparent mode
This change completely disables the use of suppression
when operating in transparent mode.

It also removes the following field trials:
* WebRTC-Aec3UseLowEarlyReflectionsTransparentModeGain
* WebRTC-Aec3UseLowLateReflectionsTransparentModeGain

Bug: webrtc:11985
Change-Id: I1c75efdad2d9c9d0a1aced86bf0278fc96616ea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185402
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32223}
2020-09-29 09:35:09 +00:00
Per Åhgren
e5d669ed28 Reland "Activating AVX2 support by default"
This is a reland of ad148272b8

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

Bug: webrtc:11663
Change-Id: I669435c2f4e451ee0766d809443484f2dde09d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32200}
2020-09-25 14:46:20 +00:00
Alessio Bazzica
cd92b0be9a AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (1/3)
This is the first CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.

In this CL, the state of the estimator is defined in a separate struct
so that in a follow-up CL a new member of that type can be added to
hold a temporary state (that can be either confirmed or discarded).

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: Ic2ea5ed63c493b9f3a79f19e7f5eaecaa6808ace
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184931
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32199}
2020-09-25 14:04:57 +00:00
Alessio Bazzica
c1ece012cb AGC2 VAD probability: instant decay / slow attack
Feature added to gain robustness to occasional VAD speech probability
spikes. In such a case, the attack process reduces the chance that the
smoothed values are greater than the speech threshold.

Bug: webrtc:7494
Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32198}
2020-09-25 14:03:17 +00:00
Per Åhgren
9ccbe17abb Revert "Activating AVX2 support by default"
This reverts commit ad148272b8.

Reason for revert: Causing test failures downstream.

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org

Change-Id: If2287a0a4b37931ce5f85baae093a66b19d0a78b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32196}
2020-09-25 12:13:46 +00:00
Alessio Bazzica
530781d03f AGC2 VadWithLevel(::LevelAndProbability) renamed + injectable VAD
Refactoring CL to improve names and allow to inject a VAD into
`VadLevelAnalyzer` (new name for `VadWithLevel`).

The injectable VAD is needed to inject a mock VAD and write better
unit tests as new features are going to be added to the class.

Bug: webrtc:7494
Change-Id: Ic0cea1e86a19a82533bd40fa04c061be3c44f068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185180
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32195}
2020-09-25 12:11:12 +00:00
Per Åhgren
ad148272b8 Activating AVX2 support by default
This CL activates the newly added AVX2 support by default.
The activation is done beneath a kill-switch.

Beyond the above, the CL also changes an incorrect DCHECK_GT
to a DCHECK_GE.

Bug: webrtc:11663
Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32193}
2020-09-25 11:30:22 +00:00
Gustaf Ullberg
39a3f081c1 Disable transparent mode when ERL is bounded
A bounded ERL disables all TransparentMode implementations.

Bug: webrtc:10232
Change-Id: I5a3641b356c2abf42cd6d055cee4007327528a41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185301
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32187}
2020-09-24 20:53:09 +00:00
Alessio Bazzica
9e81182f8e RNN VAD build file refactoring: split into separate (private) targets
First step to only expose the relevant RNN VAD API to AGC2.

Bug: webrtc:7494
Change-Id: I7f11f6eebded124c30cabd64963c8e3ccc35e58f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185124
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32185}
2020-09-24 14:33:14 +00:00
Gustaf Ullberg
afef7a74a7 HMM based transparent mode classifier
This change introduces a new Hidden Markov Model based classifier for
AEC3's 'transparent mode'. Transparent mode is used with
headsets/headphones where the speaker signal does not leak into the
microphone signal.

The current classifier suffers from two problems:
1. It sometimes takes a long time to enter transparent mode.
2. Sometimes transparent mode is left (and it once again takes a long
time to re-enter).

Both problems have a severe effect on AEC transparency.

The new classifier enters transparent mode quicker and is less likely
to exit transparent mode when there is no echo. This improves the
audio experience when using headset/headphones.

Another (minor) benefit of this change is that when transparent mode
is disabled no classifier is run (or even created) saving some memory
and CPU cycles.

Bug: webrtc:10232
Change-Id: I509af0e22b59463aeaead53c78c35be1e97fe8c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32182}
2020-09-24 08:25:50 +00:00
Alessio Bazzica
1922fb0ec3 AGC2 saturation protector: extra margin added by level estimator
In preparation for a coming refactoring CL, the (fixed) extra saturation
margin is now applied into `AdaptiveModeLevelEstimator`.

This CL also improves the unit tests by hard-coding its saturation
params instead of reading them from a field trial.
This reduces the chances of making the test flaky if a default value
changes.

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: I6765def9887a2f4e55b04d929af754cfecbb1626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184927
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32172}
2020-09-23 11:33:37 +00:00
Alessio Bazzica
736ff83e69 AGC2 saturation protector: simplify interface and impl
- Passing the speech peak power instead of VAD data
- The private class SaturationProtector::PeakEnveloper has been removed
- Added `initial_saturation_margin_db_` parameter to correctly
  initialize `last_margin_` (renamed to `margin_db_`)
- Member names have been fixed and/or shortened for better readability

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: I6cad2974397319737c8ac201d44311bf16275f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184925
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32168}
2020-09-23 07:56:44 +00:00
Alessio Bazzica
10f6eadd48 AGC2 Saturation Protector: switch to ring buffer
Even if small, the peak delay buffer copies N-1 elements for each frame
whereas a ring buffer is copy-free and scales better if the buffer size
increases.

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32165}
2020-09-22 19:34:04 +00:00
Per Åhgren
2f0f93a0c9 Add explicit initialization for the FilterAnalyzer in AEC3
This CL adds explicit initialization of the FilterAnalyzer in AEC3.
While the current code never uses any fields before they are initialized,
it makes sense to be on the safe side and add initialization during
construction.

Bug: webrtc:11918
Change-Id: I467c4c8b8d6dd859a1b216baef28ac1e9d3f76c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183764
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32069}
2020-09-10 10:50:43 +00:00
Mirko Bonadei
bef022bde0 Refactor cpu_features_wrapper.h functions from C to C++.
As mentioned on https://webrtc-review.googlesource.com/c/src/+/183380,
then relanded as https://webrtc-review.googlesource.com/c/src/+/183444,
functions in cpu_features_wrapper.h should be refactored to use
C++ features like namespaces and drop the WebRtc_ prefix.

Bug: None
Change-Id: I3e83e1668f9bf48a5d8e85d809f006666b7fa45e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183445
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32045}
2020-09-07 08:44:24 +00:00
Mirko Bonadei
66e7338617 Reland "Merge cpu_features build targets into //system_wrappers."
This is a reland of 2b242d8fba

Original change's description:
> Merge cpu_features build targets into //system_wrappers.
>
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
>
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
>
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
>
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}

TBR=kwiberg@webrtc.org

Bug: None
Change-Id: I1695b9a34d3ec20c50c1202a745f64fac58edef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183444
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32042}
2020-09-06 10:38:23 +00:00
Mirko Bonadei
d156a0d061 Revert "Merge cpu_features build targets into //system_wrappers."
This reverts commit 2b242d8fba.

Reason for revert: Breaks downstream project.

Original change's description:
> Merge cpu_features build targets into //system_wrappers.
> 
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
> 
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
> 
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
> 
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I4daa7582e55a0343eef72f08ed023c73e0b6456b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32040}
2020-09-05 21:18:40 +00:00
Mirko Bonadei
2b242d8fba Merge cpu_features build targets into //system_wrappers.
Before this CL, functions declared in cpu_features_wrapper.h where
not defined in the same build target, causing brittle builds that
might fail at link time if the binary was not depending on
//system_wrappers (the target with the definitions), violating [1].

This CL moves everything into //system_wrappers and also moves
cpu_features_wrapper.h definitions from C to C++ (in order to be able
to add the definitions to a C++ build target like //system_wrappers).

[1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs

Bug: None
Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32039}
2020-09-05 20:43:27 +00:00
Mirko Bonadei
c94650d88f Remove AudioProcessing::SetExtraOptions.
Bug: webrtc:5298
Change-Id: I28be75df69b66aa59ae91b05cb7f9afad4f55aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32033}
2020-09-03 12:43:14 +00:00
Mirko Bonadei
ea8f739bce Make AudioProcessing::SetExtraOptions non pure-virtual.
Bug: webrtc:5298
Change-Id: Ib5ba81c8e333b1e7189fbb7e9e9f143579c84b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182181
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32032}
2020-09-03 06:17:44 +00:00
Per Åhgren
0ade98316c Cleanup of the APM creation
This CL removes the possibility that APM cannot be created, i.e., that
the create method can return nullptr. That was already the case
implicitly but this CL makes that behavior explicit.


Bug: webrtc:5298
Change-Id: I2706ea538c9d1b4bcd65faecab637640a209a4dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183101
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32029}
2020-09-02 20:28:49 +00:00
Zhaoliang Ma
e537e9ca13 aec3: Support AVX2/FMA intrinsics in AEC3
Bug: webrtc:11663
Change-Id: Ib75eb616ef0cb62698b0d96af7ebe42e93825222
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179006
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32023}
2020-09-01 11:58:16 +00:00
Per Åhgren
0796b58a7e Removing call to deprecated SetExtraOptions method
Bug: webrtc:5298
Change-Id: If81d74727bb231f6e61b1647cc7b80ef13107b62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182121
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31972}
2020-08-20 16:13:12 +00:00
Per Åhgren
552d3e3d5e Reland "Add ability to state whether the APM output will be used"
This is a reland of 8be2f201ba

Original change's description:
> Add ability to state whether the APM output will be used
> 
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
> 
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}

Bug: b/154437967
Bug: b/163802450
Change-Id: Ia77a9e43f913929d1afa72212f1ea6c192d0e519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181887
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31957}
2020-08-18 12:29:32 +00:00
Erik Språng
1804b339a9 Revert "Add ability to state whether the APM output will be used"
This reverts commit 8be2f201ba.

Reason for revert: Breaks downstream

Original change's description:
> Add ability to state whether the APM output will be used
> 
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
> 
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}

TBR=alessiob@webrtc.org,peah@webrtc.org

Change-Id: I1e56dafbbfa6ea69cccbbb5cdc2b1e2a6c122c11
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/154437967
Bug: b/163802450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181884
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31953}
2020-08-17 14:10:08 +00:00
Per Åhgren
8be2f201ba Add ability to state whether the APM output will be used
This CL adds the ability for the surrounding code to state that the
APM output will not be used. The intended usecase for this is to allow
APM to run at a lower complexity when the endpoint is muted.
When APM has been informed that the output will not be used, it can
turn off code that is needed only for ensuring that the output audio
will sound good.

Bug: b/154437967,b/163802450
Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31949}
2020-08-17 12:56:24 +00:00
Sam Zackrisson
ff571c60a9 AEC3: Fix render delay buffer alignment issue at call start
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment.
This CL gives the RenderDelayBuffer an opportunity to update its counters.

Tested:
Passes: modules_unittests --gtest_filter=BlockProcessor.*
Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/"

audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps.

Bug: webrtc:11803
Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31795}
2020-07-27 15:19:58 +00:00
Markus Handell
3d2210876e Remove unused critical section includes.
Bug: webrtc:11567
Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31745}
2020-07-16 13:52:28 +00:00
Markus Handell
0df0faefd5 Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31649}
2020-07-07 14:35:58 +00:00
Brian Sheedy
8e144aa33f Remove WebRTC usage of //third_party/pymock
Removes usage of Chromium's //third_party/pymock in favor of the version
provided by vpython. This is so that the third_party version can
eventually be removed.

TBR=aleloi@webrtc.org

Bug: chromium:1094489
Change-Id: I68511e11ed1e517c2b6d3bb832090a3c27e480e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#31568}
2020-06-25 18:01:30 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
Per Åhgren
80b301255e Reland "Change to using the new Ooura constructor"
This is a reland of a6c70741e7

Original change's description:
> Change to using the new Ooura constructor
> 
> 
> Bug: b/155316201
> Change-Id: I40000e30df7a495a0937885abea19caeb599e00a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176378
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31411}

Bug: b/155316201
Change-Id: Iba9748509d66b9ad4b210be83d515db684c5d5e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176410
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31422}
2020-06-03 12:27:14 +00:00
Stephan Hartmann
3ca2836f70 IWYU: size_t is defined in stddef.h
This required for gcc-10.

Bug: None
Change-Id: Iba3ff0881005cb2ae4574e47284a1b881594de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176361
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31417}
2020-06-03 08:43:50 +00:00
Per Åhgren
43ccfecdb7 Remove unused instance of Ooura FFT
Bug: b/155316201
Change-Id: I29a2dcf620091665b6d48820022fac3a3b672c04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176405
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31414}
2020-06-03 00:33:54 +00:00
Alex Glaznev
955e9039ea Revert "Change to using the new Ooura constructor"
This reverts commit a6c70741e7.

Reason for revert: breaks TAP build

Original change's description:
> Change to using the new Ooura constructor
> 
> 
> Bug: b/155316201
> Change-Id: I40000e30df7a495a0937885abea19caeb599e00a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176378
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31411}

TBR=mbonadei@webrtc.org,peah@webrtc.org

Change-Id: Id8149889fddf96276cc15f37686ef9d1190c5c94
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/155316201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176420
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31413}
2020-06-02 22:28:07 +00:00
Per Åhgren
a6c70741e7 Change to using the new Ooura constructor
Bug: b/155316201
Change-Id: I40000e30df7a495a0937885abea19caeb599e00a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176378
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31411}
2020-06-02 14:38:58 +00:00
Tommi
a5e07cc3db Rename more death test to *DeathTest
Bug: webrtc:11577
Change-Id: If45e322fed3f2935e64c9e4d7e8c096eccc53ac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31362}
2020-05-26 20:27:34 +00:00
Danil Chapovalov
704fb55255 In common_audio/ and modules/audio_* replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: Ib0ffce4de50a13b018926f6ea2865a2ec2fb2ec7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175621
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31333}
2020-05-20 13:17:31 +00:00
Per Åhgren
e9cd6177eb Add ability for audioproc_f to operate on any AudioProcessing object.
This CL extends the WebRTC testing API to allow audioproc_f -based
testing using a pre-created AudioProcessing object. This is an
important feature to allow testing any AudioProcessing objects
that are injected into WebRTC.

Beyond adding this, the CL also changes the simulation code to
operate on a scoped_refptr<AudioProcessing> object instead of a
std::unique<AudioProcessing> object

Bug: webrtc:5298
Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31319}
2020-05-19 11:37:18 +00:00
Tommi
909f3a5339 Rename several more tests that use EXPECT_DEATH to *DeathTest.
Bug: webrtc:11577
Change-Id: I0397ee933464496e4885bb0f8030f3d669e5e612
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175641
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31309}
2020-05-18 16:10:04 +00:00
Tommi
da357a9495 Rename EchoPathDelayEstimator to EchoPathDelayEstimatorDeathTest.
...for the NullDataDumper, WrongCaptureBlockSize and
DISABLED_WrongRenderBlockSize tests. This is to avoid creation
of additional threads on Mac, which can cause issues on asan bots.

Bug: webrtc:11577
Change-Id: I4e6a64d47ec3b0a0e0018b19a0486208ba7e6ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31307}
2020-05-18 14:27:14 +00:00
Markus Handell
02ba1d252e AudioProcessingImpl: remove lock recursions.
This change removes lock recursions and adds thread annotations.

Bug: webrtc:11567
Change-Id: Ibefb49bb5b865cb0bb33e4580d34d9837fb41bff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31260}
2020-05-14 14:53:58 +00:00
Per Åhgren
09e9a83d91 Change the way that AecDumps are created in APM
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.

This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.

The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.

The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
   surface of APM.
2) Those files anyway needed to be moved to a separate build-
   target to avoid a circular build-file dependency caused by
   the other changes in this CL

Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
2020-05-11 10:33:00 +00:00
Sam Zackrisson
ab866a2ccb Remove audio playout generator from APM API
This API is and has always been unused.

Bug: webrtc:5298
Change-Id: If1201d37a00e387567d44a9ed8be99a157915b47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174661
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31180}
2020-05-07 12:14:50 +00:00