Commit graph

91 commits

Author SHA1 Message Date
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Tommi
06af5b5c64 More use of DeinterleavedView and MonoView in audio classes
Adopt DeinterleavedView and MonoView in the following classes
and deprecate existing versions where external dependencies exist:

* GainApplier
* AdaptiveDigitalGainController
* NoiseLevelEstimator
* VoiceActivityDetectorWrapper (including MonoVad)

Bug: chromium:335805780
Change-Id: I15dad833a87d31476d147dd2456bd1cc39f901ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355861
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42611}
2024-07-09 13:29:37 +00:00
Tommi
6056976709 Updates to AudioFrameView and VectorFloatFrame
Using DeinterleavedView<> simplifies these two classes, so now the
classes are arguably thin wrappers on top of DeinterleavedView<> and
AudioFrameView<> can be replaced with DeinterleavedView<>.

The changes are:
* Make VectorFloatFrame not use a vector of vectors but rather
  just hold a one dimensional vector of samples and leaves the mapping
  into the buffer up to DeinterleavedView<>.
* Remove the `channel_ptrs_` vector which was required due to an
  issue with AudioFrameView.
* AudioFrameView is now a wrapper over DeinterleavedView<>. The most
  important change is to remove the `audio_samples_` pointer, which
  pointed into an externally owned pointer array (in addition to
  the array that holds the samples themselves). Now AudioFrameView
  can be initialized without requiring such a long-lived array.

Bug: chromium:335805780
Change-Id: I8f3c23c0ac4b5a337f68e9161fc3a97271f4e87d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352504
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42498}
2024-06-17 12:13:40 +00:00
Tommi
093824c4d2 Switch away from hz to samples per channel for FrameCombiner et al
This simplifies the following steps:
* FrameCombiner infers the sample rate from channel size
* Sends the inferred sample rate to FixedDigitalLevelEstimator
  and Limiter.
* Those classes then convert the sample rate to channel size.
  Along the way perform checks that the derived channel size value
  is a legal value (which has already been done by FrameCombiner).

To:
* FrameCombiner sends channel size to FixedDigitalLevelEstimator and
  Limiter.

Bug: chromium:335805780
Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42480}
2024-06-13 19:00:39 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Florent Castelli
0afde7614d Move webrtc::AudioProcessing include to api/ folder
Bug: webrtc:15874
Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-20 07:02:50 +00:00
Alessio Bazzica
a106095333 Fix WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget tests
- Reset the tested metrics to avoid interactions between tests that
  depend on the execution order
- Address the comment in [1] by adding a function to log the same
  histogram in two different places

[1] https://chromium-review.googlesource.com/c/chromium/src/+/4087426/4/tools/metrics/histograms/metadata/web_rtc/histograms.xml#179

Bug: webrtc:7494
Change-Id: Ia4d339b03c8078eb63626c91579f8a9547f087f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287681
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38873}
2022-12-12 19:26:01 +00:00
Alessio Bazzica
f72bc5f1e2 AGC2: rename AdaptiveDigitalGainApplier -> AdaptiveDigitalGainController
Bug: webrtc:7494
Change-Id: Id45495d1742f7d2027429c97a3b286468da99b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287220
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38857}
2022-12-09 10:31:34 +00:00
Alessio Bazzica
17e14fdf34 APM AGC2: consolidate GainController2
Now that `InputVolumeController` is finalized, it's time to
consolidate AGC2.

Main changes:
- Remove `AdaptiveDigitalGainController`: it's too simple to justify
  a dedicated class and some components of it are also used by
  `InputVolumeController`
- Remove unwanted temporal dependency: make `InputVolumeController`
  adapt the volume based on the current speech level estimation and
  not on the estimation from the previous frame

Tested: AGC2 adaptive digital bit-exactness verified

Bug: webrtc:7494
Change-Id: I175c2741cafc52be81794219c996a3824c3bbf5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280560
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38841}
2022-12-07 21:47:45 +00:00
Alessio Bazzica
8b4a81fb55 APM: Prepare to remove AdaptiveDigitalGainController wrapper
Isolates the build targets for the `AdaptiveDigitalGainController`
dependencies that will be moved into `GainController2`.

`AdaptiveDigitalGainController` will be removed because the wrapper
itself adds little - that's the reason why it has no unit tests.

Bug: webrtc:7494
Change-Id: I2ca41f9255c8faefe4b2cb4ec1f8db536e582f39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280482
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38799}
2022-12-02 14:47:33 +00:00
Hanna Silen
9f06ef1cc3 Implement InputVolumeController
Implement InputVolumeController and RecommendedInputVolumeEstimator based on the copy of agc classes AgcManagerDirect and MonoAgc.
Copies of the original files created in https://webrtc-review.googlesource.com/c/src/+/278624.

Bug: webrtc:7494
Change-Id: I74acee57b0db5cc8a6b666be9ba619c6c98a1773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278625
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38533}
2022-11-02 11:31:59 +00:00
Hanna Silen
7587755d29 Copy AgcManagerDirect files to agc2 and rename the classes
Copy AgcManagerDirect files from agc to agc2. Rename the newly
created files and classes ahead of refactoring. Add a build
target.

This change is done to enable creating a class
InputVolumeController based on AgcManagerDirect. The added
temporary dependency on files in agc will be removed
in https://webrtc-review.googlesource.com/c/src/+/278625.

The exact copy of the files happened in the 1st patchset and it
has been verified as follows:

Checksum check:
```
$ git checkout main && git pull
# Go back to the tree state before [1] landed
$ git new-branch tmp
$ git reset --hard 2235776597
$ cd modules/audio_processing/agc/
$ md5 agc_manager_direct*
MD5 (agc_manager_direct.cc) = e661481a85f72596cae4599b62907f5b
MD5 (agc_manager_direct.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (agc_manager_direct_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```

Patchset 1 (see [2])
```
$ cd modules/audio_processing/agc2/
$ md5 input_volume_controlle*
MD5 (input_volume_controller.cc) = e661481a85f72596cae4599b62907f5b
MD5 (input_volume_controller.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (input_volume_controller_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```

[1] https://webrtc-review.googlesource.com/c/src/+/278781
[2] https://webrtc-review.googlesource.com/c/src/+/278624/1

Bug: webrtc:7494
Change-Id: I7804da899d18adf556b089c76a567ce27c299a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278624
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38512}
2022-10-31 15:58:11 +00:00
Alessio Bazzica
fbe5d7c3d4 Reland "APM: log both applied and recommended input volume stats"
This is a reland of commit 8d7273357d

Root cause:
audioproc_f doesn't call `metrics::Enable()` and therefore the stats
reporter crashed when `metrics::HistogramFactoryGetCountsLinear()`
returned a nullptr.

Bug fix:
Added `InputVolumeStatsReporter::cannot_log_stats_`, a const flag
that is set to true if any histogram factory returns a nullptr.
When true, the class does nothing.

This CL also includes other code readability improvements that were
not part of the original CL.

Original change's description:
> APM: log both applied and recommended input volume stats
>
> This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
> with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
> `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
>
> Bug: webrtc:7494
> Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38468}

Bug: webrtc:7494
Change-Id: I8373d16beb06b84f439d2c2274ededea7c5e95b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38484}
2022-10-27 14:40:40 +00:00
Alessio Bazzica
35b3c63ba4 Revert "APM: log both applied and recommended input volume stats"
This reverts commit 8d7273357d.

Reason for revert: revert needed to land https://webrtc-review.googlesource.com/c/src/+/280600

Original change's description:
> APM: log both applied and recommended input volume stats
>
> This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
> with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
> `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
>
> Bug: webrtc:7494
> Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38468}

Bug: webrtc:7494
Change-Id: I4a2acfd5a983d9397932b2879cfa057deaf0eb2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280581
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38476}
2022-10-26 13:27:01 +00:00
Alessio Bazzica
d89dff767c AGC2: prepare to move speech level estimator into GainController2
- build target isolated
- `AdaptiveModeLevelEstimator` renamed to `SpeechLevelEstimator`

Bug: webrtc:7494
Change-Id: If16caec2269b2ed1b2ee27c3687a8f8875f55c8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280441
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38469}
2022-10-25 16:15:07 +00:00
Alessio Bazzica
8d7273357d APM: log both applied and recommended input volume stats
This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
`WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.

Bug: webrtc:7494
Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38468}
2022-10-25 14:02:22 +00:00
Alessio Bazzica
d226c5731d APM: move AnalogGainStatsReporter to AGC2
Bug: webrtc:7494
Change-Id: Ifb924e6eda47dd96a591a0b55b1e7fcfdbbbbe18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280222
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38464}
2022-10-25 08:35:02 +00:00
Hanna Silen
b37a9c5f88 Remove ClippingPredictorEvaluator
Bug: webrtc:7494
Change-Id: Idba27a5dbe72726f9e1469e955c5958558d93a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278403
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38321}
2022-10-07 13:50:04 +00:00
Hanna Silen
767898c048 Add SpeechProbabilityBuffer
Add a buffer class to store speech probabilities and to estimate speech
activity. Follows the implementation of speech activity computation in
LoudnessHistogram but uses floats for computations.

Bug: webrtc:7494
Change-Id: I6ee72ec52919904ea4e1fbe51d61993aa7813c9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277801
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38309}
2022-10-06 11:23:03 +00:00
Hanna Silen
cfbda697ec ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2
Bug: webrtc:7494
Change-Id: If88795fe34a73faa267a9c0bd5250e36455d4d81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277741
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38296}
2022-10-05 08:35:42 +00:00
Ali Tofigh
f3592cb2a2 Adopt absl::string_view in modules/audio_processing/
Bug: webrtc:13579
Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37800}
2022-08-16 13:49:14 +00:00
Florent Castelli
c3e6e3a3e8 Remove dependency on rtc_base_approved from most targets
Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
Florent Castelli
71337f387e Move random out of rtc_base_approved
Bug: webrtc:9838
Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-19 14:00:47 +00:00
Florent Castelli
4467ad7835 Remove //rtc_base:macromagic from public deps
Bug: webrtc:8603
Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-05 12:36:12 +00:00
Alessio Bazzica
2fa4618a3b AGC2: AdaptiveAgc ctor with sample rate and # of channels
The class has also been renamed to better reflect its purpose.

Bug: webrtc:7494
Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35277}
2021-10-28 15:28:12 +00:00
Alessio Bazzica
b4d4ae2c23 AGC2: VAD moved into GainController2
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).

Bug: webrtc:7494
Change-Id: Id9849c4463791f5a203afe31efc163efb4d4458e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234583
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35248}
2021-10-20 15:50:33 +00:00
Alessio Bazzica
8dbdf5e3bf AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 2/2
Internal refactoring of AGC2 to decouple the VAD, its wrapper and the
peak and RMS level measurements.

Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).

Bug: webrtc:7494
Change-Id: Ib560f1fcaa601557f4f30e47025c69e91b1b62e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234524
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35208}
2021-10-14 13:32:25 +00:00
Alessio Bazzica
f77f35b764 AGC2: gain_controller2 target isolated
Needed to restrict visibility.

Bug: webrtc:7494
Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 11:24:55 +00:00
Alessio Bazzica
585aad7323 AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2
Internal refactoring of AGC2. This CL is needed in preparation for its
child CL to correctly show the upcoming changes in the diff.

Bug: webrtc:7494
Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 16:15:24 +00:00
Alessio Bazzica
5c3ae49b44 AudioFrameView: size_t -> int
Bug: webrtc:7494
Change-Id: I46b1328f3d7da721e144cc3752ed4f458084cf62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234522
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35163}
2021-10-07 14:41:03 +00:00
Alessio Bazzica
7b80d4480e AGC2: SIMD allowed config flags to field trials
Bug: webrtc:7494
Change-Id: I41fa05d2ef6d969750f3d4c1e40ecbcd30293b5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233741
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35144}
2021-10-05 12:01:38 +00:00
Alessio Bazzica
a850e6c8b6 AGC2 config: allow tuning of headroom, max gain and initial gain
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).

Tested: compiled Chrome with this patch and made an appr.tc test call

Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
2021-10-04 16:11:00 +00:00
Alessio Bazzica
b8a19df71c AGC2: removed unused noise estimator implementation
This CL also includes the following changes:
- `AudioProcessing::Config::GainController2::noise_estimator`
  deprecated
- `EnergyToDbfs()` optimized by removing unnecessary `sqrt`
- Unit test minor fix, incorrect type was used

Bug: webrtc:7494
Change-Id: I88a6672d6f7cd03fcf6a3031883522d256880140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230940
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34893}
2021-09-01 12:45:20 +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
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
42eef86c4f Remove unused code in APM
- The injection of the AGC2 level estimator into `AgcManagerDirect`
  is not used anymore
- `ExperimentalAgc::enabled_agc2_level_estimator` can also be removed
- 3 ctors of `ExperimentalAgc` are unused
- `AgcManagerDirectStandaloneTest::AgcMinMicLevelExperiment` can be
  split into separate unit tests (better code clarity)

Bug: webrtc:7494
Change-Id: I5843147c38cf7cb5ee484b0a72fe13dcf363efaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202025
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33027}
2021-01-18 13:40:27 +00:00
Alessio Bazzica
ed9f5f85fd RNN VAD optimizations: VectorMath::DotProduct() NEON arm64
Results: RNN VAD realtime factor improved from 140x to 195x (+55x)
Test device: Pixel 2 XL
Benchmark setup: max clock speed forced on all the cores by
setting "performance" as scaling governor

Bug: webrtc:10480
Change-Id: I3e92f643853ad1fe990db909c578ce78ee826c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198842
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32888}
2020-12-30 10:47:01 +00:00
Alessio Bazzica
253f8369bb AGC2 RNN VAD: safe SIMD optimizations scheme + AVX2 kill switch
In preparation for adding AVX2 code, a safe scheme to support
different SIMD optimizations is added.

Safety features:
- AVX2 kill switch to stop using it even if supported by the
  architecture
- struct indicating the available CPU features propagated from
  AGC2 to each component; in this way
  - better control over the unit tests
  - no need to propagate individual kill switches but just
    set to false features that are turned off

Note that (i) this CL does not change the performance of the RNN VAD
and (ii) no AVX2 optimization is added yet.

Bug: webrtc:10480
Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32739}
2020-12-02 10:09:24 +00:00
Alessio Bazzica
59f1d1e36d AGC2 adaptive digital controller config: new params
This CL adds and wires up the following parameters:
- VAD probability attack used in `VadLevelAnalyzer`
- Adjacent spech frames threshold used in `AdaptiveModeLevelEstimator`
- Initial saturation margin used in `AdaptiveModeLevelEstimator`

The deprecated ctor in `AdaptiveModeLevelEstimator` is removed.

Tested: bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: Idf94aaadba1476757f845e696bfb47ff6252d5f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186048
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32265}
2020-09-30 22:29:14 +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
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
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
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
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
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
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
Mirko Bonadei
146137adee Merge OOURA BUILD.gn files.
No-Tree-Checks: True
No-Try: True
Bug: webrtc:11509
Change-Id: If66483a02085396e231986e39ef0dafcdd3f4bed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173740
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31114}
2020-04-21 09:06:37 +00:00