Commit graph

68 commits

Author SHA1 Message Date
Henrik Boström
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 54d1344d98.

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

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

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c6.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

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

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
2022-02-14 12:25:51 +00:00
Alessio Bazzica
54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c6.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> This reverts commit b4e06d032e.
>
> Reason for revert: breaking downstream projects
>
> Original change's description:
> > Remove unused APM voice activity detection sub-module
> >
> > API changes:
> > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > - webrtc::AudioProcessingStats::voice_detected deprecated
> > - cricket::AudioOptions::typing_detection deprecated
> > - webrtc::StatsReport::StatsValueName::
> >   kStatsValueNameTypingNoiseState deprecated
> >
> > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35975}
>
> TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35977}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
2022-02-13 14:02:08 +00:00
Alessio Bazzica
a751f167c6 Revert "Remove unused APM voice activity detection sub-module"
This reverts commit b4e06d032e.

Reason for revert: breaking downstream projects

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

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

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
2022-02-11 12:15:44 +00:00
Alessio Bazzica
b4e06d032e Remove unused APM voice activity detection sub-module
API changes:
- webrtc::AudioProcessing::Config::VoiceDetection removed
- webrtc::AudioProcessingStats::voice_detected deprecated
- cricket::AudioOptions::typing_detection deprecated
- webrtc::StatsReport::StatsValueName::
  kStatsValueNameTypingNoiseState deprecated

PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0

Bug: webrtc:11226,webrtc:11292
Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35975}
2022-02-11 10:47:39 +00:00
Henrik Lundin
64253a93dc Remove more traces of keyboard mic support from APM
The 6-parameter Initialize method is removed. The has_keyboard parameter
in the StreamConfig constructor is removed together with the underlying
member and helper functions.

Bug: chromium:1271981, b/217349489
Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35908}
2022-02-04 14:27:51 +00:00
Alessio Bazzica
183c64ce19 APM: remove LevelEstimator
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.

This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.

Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
2021-10-20 10:52:17 +00:00
Alessio Bazzica
1ac4f2a29e AGC2: Remove unused parameters
- `NoiseEstimator` and `LevelEstimator` enums
- `vad_probability_attack`
- `level_estimator_adjacent_speech_frames_threshold`
- `use_saturation_protector`
- `gain_applier_adjacent_speech_frames_threshold`
- `initial_saturation_margin_db`
- `extra_saturation_margin_db`

Bug: webrtc:7494
Change-Id: I12e40c8efe2d2126d7597ec18a78cf9d5d39baf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232903
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35096}
2021-09-27 11:14:35 +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
Gustaf Ullberg
fdd6099348 Rework transient suppressor configuration in audioproc_f
The transient suppressor can be configured as:
0 - Deactivated
1 - Activated with key events from aecdump
2 - Activated with continuous key events (for debugging purposes)

Bug: webrtc:5298
Change-Id: I116eb08ad50178dc5116d5d967084e6c9967f258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211869
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33464}
2021-03-15 15:19:09 +00:00
Per Åhgren
34fdc92119 Add audioproc_f support for testing the runtime settings of whether the output is used
Bug: b/177830919
Change-Id: Iddcb79000f471eac165e3f44f14fad41435e6ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211241
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33426}
2021-03-10 23:19:30 +00:00
Per Åhgren
879d33b9f8 Add more refined control over dumping of data and the aecdump content
This CL adds the ability in audioproc_f and unpack_aecdump to:
-Clearly identify the Init events and when those occur.
-Optionally only process a specific Init section of an aecdump.
-Optionally selectively turn on dumping of internal data for a
 specific init section, and a specific time interval.
-Optionally let unpack_aecdump produce file names based on inits.

Bug: webrtc:5298
Change-Id: Id654b7175407a23ef634fca832994d87d1073239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33181}
2021-02-06 00:36:10 +00:00
Per Åhgren
c2ae4c8a37 Allow separate dump sets for the data dumper in APM
This CL allows separate dump sets to be used when dumping internal
APM data using audioproc_f, opening up for reducing the amount of
data to be dumped.

Bug: webrtc:5298
Change-Id: I8286933ceed10db074f2064414cc08e8b12653fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196089
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33047}
2021-01-20 16:38:17 +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
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
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
Sam Zackrisson
01c107e37a Correct int16 audio frame setup in audioproc_f
Currently, audioproc_f crashes on a DCHECK as the data vector of Int16Frame is not resized.

Bug: webrtc:5298
Change-Id: I897cf0fce07e0ed2c0a365a965fa50fd3d8ddd18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172624
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30969}
2020-04-02 04:05:02 +00:00
Per Åhgren
2507f8cdc9 APM: Replace all remaining usage of AudioFrame outside interfaces
This CL replaces all remaining usage of AudioFrame within APM,
with the exception of the AudioProcessing interface.

The main changes are within the unittests.

Bug: webrtc:5298
Change-Id: I219cdd08f81a8679b28d9dd1359a56837945f3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170362
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30831}
2020-03-19 12:40:18 +00:00
Per Åhgren
8ad9e74d62 Removing deprecated legacy noise suppressor
This CL removes the code for the deprecated legacy noise.

Bug: webrtc:5298
Change-Id: If287d8967a3079ef96bff4790afa31f37d178823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167922
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30434}
2020-01-31 07:14:25 +00:00
Per Åhgren
5dca3f1336 Add floating point support for writing and reading wav files
This CL adds support for reading and writing floating point
wav files in WebRTC. It also updates the former wav handling
code as well as adds some simplifications.

Beyond this, the CL also adds support in the APM data_dumper
and in the audioproc_f tool for using the floating point wav
format.

Bug: webrtc:11307
Change-Id: I2ea33fd12f590b6031ac85f75708f6cc88a266b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162902
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30423}
2020-01-30 13:38:19 +00:00
Per Åhgren
0695df1a59 Reland "Replace the ExperimentalAgc config with the new config format"
This is a reland of f3aa6326b8

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
2020-01-17 10:09:09 +00:00
Yves Gerey
eb3beb8504 Revert "Replace the ExperimentalAgc config with the new config format"
This reverts commit f3aa6326b8.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

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

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

Bug: webrtc:5298
Change-Id: I794d2ab4b8caa5330c5ad490ba604646a249a1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164530
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30153}
2020-01-07 05:22:01 +00:00
Per Åhgren
f3aa6326b8 Replace the ExperimentalAgc config with the new config format
This CL replaces the use of the ExperimentalAgc config with
using the new config format.

Beyond that, some further changes were made to how the analog
and digital AGCs are initialized/called. While these can be
made in a separate CL, I believe the code changes becomes more
clear by bundling those with the replacement of the
ExperimentalAgc config.

TBR: saza@webrtc.org
Bug: webrtc:5298
Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30149}
2020-01-03 23:14:13 +00:00
Per Åhgren
2e8e1c699e Open up for do the noise suppressor analysis on the linear AEC output
This CL allows the noise suppressor to use the linear AEC output
for analysis whenever that is available. This will potentially
lower the risk that the noise suppressor estimates the noise
based on echo.
The feature is off by default.

Bug: webrtc:5298,b:132164318
Change-Id: Idc6c8e197d96209d213819d87a8fb2533b7303ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162900
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30116}
2019-12-20 09:28:01 +00:00
Per Åhgren
62ea0aaea0 Remove deprecated legacy AEC code
This CL removes the deprecated legacy AEC code.

Note that this CL should not be landed before the M80 release has been cut.

Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
2019-12-09 10:37:49 +00:00
Per Åhgren
e14cb99408 Correct/update the activation of the multi-channel processing in APM
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.

It also splits the activation of multi-channel processing to be separate
for render and capture.


Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
2019-11-27 10:15:25 +00:00
Per Åhgren
c20a19cc4b Allow extracting the linear AEC output
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.

Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
2019-11-13 11:33:53 +00:00
Per Åhgren
0cbb58e046 Reland "Refactoring of the noise suppressor and adding true multichannel support"
This is a reland of 87a7b82520

Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
> 
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
> 
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
> 
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
> 
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}

Bug: webrtc:10895, b/143344262
Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29663}
2019-10-31 11:56:01 +00:00
Artem Titov
4778f6ce7a Revert "Refactoring of the noise suppressor and adding true multichannel support"
This reverts commit 87a7b82520.

Reason for revert: Speculative revert. Breaks downstream projects.

Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
> 
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
> 
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
> 
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
> 
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}

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

Change-Id: I4d4025bda01f484979961fe57380a705e4d78397
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10895, b/143344262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158701
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29651}
2019-10-30 09:49:31 +00:00
Per Åhgren
87a7b82520 Refactoring of the noise suppressor and adding true multichannel support
This CL adds proper multichannel support to the noise suppressor.
To accomplish that in a safe way, a full refactoring of the noise
suppressor code has been done.

Due to floating point precision, the changes made are not entirely
bitexact. They are, however, very close to being bitexact.

As a safety measure, the former noise suppressor code is preserved
and a kill-switch is added to allow revering to the legacy noise
suppressor in case issues arise.

Bug: webrtc:10895, b/143344262
Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29646}
2019-10-29 23:23:38 +00:00
Sam Zackrisson
41478c7c1b Remove AudioProcessing::gain_control() getter
This change also resolves a bug in audioproc_f:
The implicit ApplyConfig calls to enable gain control settings in
aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig
call on line 500 using a config from line 292.

Compared to a ToT build including a fix for that bug, these changes
are bitexact on a large number of aecdumps.

[1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b

Bug: webrtc:9878
Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 09:23:16 +00:00
Sam Zackrisson
0824c6f61a Delete voice_detection() pointer to submodule
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.

ApmTest.Process passes with unchanged reference files if
audio_processing_impl would initialize the VAD with
VoiceDetection::kLowLikelihood instead of kVeryLowLikelihood.
This was verified by testing this CL with that modification.

Bug: webrtc:9878
Change-Id: I4d08df37a07e5c72feeec02a07d6b9435f917d72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155445
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29395}
2019-10-07 13:06:05 +00:00
Sam Zackrisson
feee1e4c36 Add flag to APM to force multichannel even with AEC3
Currently, APM fakes multichannel in two ways:
 - With injected AECs, capture processing is only performed on the left
channel. The result is copied into the other channels.
 - With multichannel render audio, all channels are mixed into one
before analysing.

This CL adds a flag to disable these behaviors, ensuring proper
multichannel processing happens throughout the APM pipeline.

Adds killswitches to separately disable render / capture multichannel.

Additionally - AEC3 currently crashes when running with multichannel.
This CL adds the missing pieces to at least have it run without
triggering any DCHECKS, including making the high pass filter properly
handle multichannel.

Bug: webrtc:10913, webrtc:10907
Change-Id: I38795bf8f312b959fcc816a056fba2c68d4e424d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29248}
2019-09-20 06:36:12 +00:00
Per Åhgren
fcbe4071ce Adding more refined control over choice of band-splitting
This CL allows the user to have more refined control over what band
splitting-scheme is used inside the audio processing module.


Bug: webrtc:6181
Change-Id: I236c3b1f96ab80cc4ffb8c39c045c034764567a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29189}
2019-09-14 23:14:17 +00:00
Sonia-Florina Horchidan
b75d14c802 audioproc_f: input AEC dump as string, output audio to vector
This CL adds the following options:

pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector

Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
2019-08-12 09:17:36 +00:00
Per Åhgren
6ee75fdfcb Allow setting the AGC2 fixed gain during runtime
This CL extends the supported runtime settings in
APM to also comprise the AGC2 fixed gain.
The CL was originally created by Adam Whiteside.

Bug: webrtc:10574
Change-Id: I79b3d6501f1e202b66a9b6018f8a493a56b01f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27782}
2019-04-26 10:05:45 +00:00
Per Åhgren
ef3496095d Allow audioproc_f to override the pre-amp gain in aecdumps
This CL allows audioproc_f to overrule any runtime settings for the
pre-amplifier gain that are present in the aecdump file.

Bug: webrtc:10546
Change-Id: I74dbf8d043f59b516bf0abc80f266e965af0754d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132558
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27598}
2019-04-12 15:05:15 +00:00
Per Åhgren
ada9b89b99 Added more refined benchmarking code for audioproc_f
This CL extends, and partly corrects, the benchmarking
code in audioproc_f to provide statistics for the API
call durations in audioproc_f

Bug: chromium:939791
Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27443}
2019-04-04 08:37:16 +00:00
Danil Chapovalov
07122bc87e Use TaskQueueForTest instead or TaskQueue in unittests
To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue

Bug: webrtc:10284
Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:42:49 +00:00
Ivo Creusen
9a66d5ed65 Add support to audioproc_f to generate a custom call order file.
This adds a flag to audioproc_f to generate a custom call order
file from an AEC dump. This file can be used to get more realism
when simulating with wav-files.

Bug: webrtc:10393
Change-Id: I245533d18affaab2f6cef53138332d7d83c71822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126782
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27104}
2019-03-13 15:08:18 +00:00
Per Åhgren
200feba1c0 Make AEC3 the default desktop AEC option in WebRTC
Bug: webrtc:10366
Change-Id: I854ed62df1da489fdab43e9157dff79b7287cacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125081
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26983}
2019-03-06 08:43:48 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Alessio Bazzica
4bc60452f7 Add output directory option for audioproc_f data dump files.
Bug: webrtc:10000
Change-Id: Iac21f826e78d6cb339c68fdeeedf9fe39920ac31
Reviewed-on: https://webrtc-review.googlesource.com/c/110904
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25713}
2018-11-20 13:30:24 +00:00
Alessio Bazzica
68170388f4 APM audioproc_f: flag for AGC2 adaptive level estimator.
Bug: webrtc:7494
Change-Id: I603211570a0a46d8884749dab887cd572827cca6
Reviewed-on: https://webrtc-review.googlesource.com/c/110250
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25708}
2018-11-20 12:50:23 +00:00
Per Åhgren
7a95e0fcf4 APM: Add ability to turn on/off dumping of internal data
This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.

Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}
2018-10-25 09:03:53 +00:00
Per Åhgren
e4d23b1adf Hooked up the control of the adaptive AGC2 mode in audioproc_f
This CL adds the ability to toggle the AGC2 adaptive digital mode in
audioproc_f

Bug: webrtc:5298
Change-Id: If1567d8c87f88992dff89253edb293a56cee0a73
Reviewed-on: https://webrtc-review.googlesource.com/c/103361
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24954}
2018-10-03 14:21:55 +00:00
Sam Zackrisson
8c147b68e6 Reland "Remove APM-internal usage of EchoControlMobile"
This is a reland of 2fbb83b16b

Original change's description:
> Remove APM-internal usage of EchoControlMobile
> 
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
> 
>  - EchoControlMobileImpl will no longer inherit EchoControlMobile.
>  - Removes usage of AudioProcessing::echo_control_mobile() inside most of
>    the audio processing module and unit tests.
> 
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
> 
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}

Bug: webrtc:9535
Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/102881
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24943}
2018-10-03 07:45:33 +00:00
Sam Zackrisson
05a7004442 Revert "Remove APM-internal usage of EchoControlMobile"
This reverts commit 2fbb83b16b.

Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117

Original change's description:
> Remove APM-internal usage of EchoControlMobile
> 
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
> 
>  - EchoControlMobileImpl will no longer inherit EchoControlMobile.
>  - Removes usage of AudioProcessing::echo_control_mobile() inside most of
>    the audio processing module and unit tests.
> 
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
> 
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}

TBR=saza@webrtc.org,aleloi@webrtc.org

Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
2018-09-28 13:39:19 +00:00
Sam Zackrisson
2fbb83b16b Remove APM-internal usage of EchoControlMobile
This is a sibling CL to a similar one for EchoCancellation:
https://webrtc-review.googlesource.com/c/src/+/97603

 - EchoControlMobileImpl will no longer inherit EchoControlMobile.
 - Removes usage of AudioProcessing::echo_control_mobile() inside most of
   the audio processing module and unit tests.

The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (comfort noise, routing mode), but prints an
error message when unsupported settings are encountered.

Tested: audioproc_f with .wav and aecdump inputs.
Bug: webrtc:9535
Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
Reviewed-on: https://webrtc-review.googlesource.com/101621
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24888}
2018-09-28 11:11:44 +00:00
Per Åhgren
56b5a6c4b2 audioproc_f: Modified and added further logging of used aec3 parameters
This CL:
-Adds the option to log the aec3 parameters used for a simulation.
-Cleans up the logging of the custom setting of aec3 parameters to
 instead rely on the newly added logging.

Bug: webrtc:8671
Change-Id: If73a73d08e5a5077416033ded598a83fb1ade3e0
Reviewed-on: https://webrtc-review.googlesource.com/100381
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24742}
2018-09-14 13:56:52 +00:00
Alex Loiko
d934244feb Added flags for the adaptive analog AGC in audioproc_f.
Added back the 'agc2 level estimation' flag. Also added a flag for
moving the level measurement before AEC and NS. This is to run offline
experiments with audioproc_f.


Bug: webrtc:7494
Change-Id: I3e3ffceede7166b754130be2b707b620ba527e9f
Reviewed-on: https://webrtc-review.googlesource.com/97442
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24657}
2018-09-10 14:16:46 +00:00