69 Commits

Author SHA1 Message Date
Philipp Hancke
902bc24b6f iwyu test/fuzzers
Note that this needs to be done with a work directory that supports
fuzzer builds, otherwise IWYU will bail out with complaints about
find-bad-constructs and raw-ptr-plugin

Some manual work was required to resolve the TaskQueueFactory which
is forward-declared by environment which required a manual include
of the header file.

The DcSctp packet fuzzer was also updated use the
disable_checksum_verification option which was moved to the
DcSctpOptions struct.

vp9_encoder_references_fuzzer was trying to include libvpx includes
which had to be reverted.

BUG=webrtc:42226242

Change-Id: I9fdcf979e73fdee77106c4583faff21ca7abf19f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43873}
2025-02-11 12:04:42 -08:00
Danil Chapovalov
dc03d8731f Rename AudioProcessingFactory to Builder
To stress there is no intention to use each instance more than once.

Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
2024-10-29 16:34:01 +00:00
Danil Chapovalov
9c21f6386f Replace AudioProcessingBuilderForTesting with the BuiltinAudioProcessingFactory
Bug: webrtc:369904700
Change-Id: Ie96dc1a9c052cb5340b10bf834d95f88f0a96a14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43247}
2024-10-16 10:55:38 +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
Danil Chapovalov
b64eef1234 In AecDump take raw pointer to TaskQueueBase instead of legacy rtc::TaskQueue
Bug: webrtc:14169
Change-Id: I1e50a945a7637da07bec00ccd7b6b1847a7481cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333480
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41477}
2024-01-08 12:17:06 +00:00
Emil Lundmark
4e86aa0870 Remove mentions of already deleted field trials
- WebRTC-Audio-Agc2ForceExtraSaturationMargin
- WebRTC-Audio-Agc2ForceInitialSaturationMargin
- WebRTC-Audio-BitrateAdaptation
- WebRTC-Audio-TransientSuppressorVadMode
- WebRTC-FrameBuffer3
- WebRTC-IntelVP8
- WebRTC-UseActiveIceController

Bug: None
Change-Id: I3545727c09f761867f2f4c2bb5c400012ce146d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39444}
2023-03-01 15:53:37 +00:00
Sam Zackrisson
306eee3d17 APM fuzzer: fuzz more sample rates, clean up input generation
APM has historically allowed sample rates not divisible by 100, but there is also code that explicitly states that such rates are not supported.
It is unclear how well rates like 22050 are handled in practice.
This CL adds support for fuzzing more sample rates, to help find issues.

We usually preserve fuzzer data reads to avoid invalidating unresolved fuzzer-found issues, but to make the code a little more readable this CL removes the discarded reads. This renders the only currently open bug non-reproducible, crbug.com/1299393.

Bug: webrtc:9413, chromium:1299393
Change-Id: I98ac1c653627c20adc73b8edede02f1526d80d9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264504
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37114}
2022-06-03 13:00:43 +00:00
Alessio Bazzica
1db0a261ca Reland "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1.

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

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

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

Bug: webrtc:11226,webrtc:11292
Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36012}
2022-02-16 08:41:30 +00:00
Henrik Boström
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

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

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

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

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

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
2022-02-14 12:25:51 +00:00
Alessio Bazzica
54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.

Reason for revert: dependency in a downstream project removed

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

# Not skipping CQ checks because this is a reland.

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

Reason for revert: breaking downstream projects

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

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

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
2022-02-11 12:15:44 +00:00
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
Sam Zackrisson
03cb7e5a61 APM: Make echo detector an optionally compilable and injectable component
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.

Important: The echo detector is no longer enabled by default.

API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ

This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.

The echo detector implementation is marked poisonous, to avoid accidental dependencies.

Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.

Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps

Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
2021-12-16 17:39:11 +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
0b45462389 APM fuzzer: add SetConfig() to test builder
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.

[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc

Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
2021-10-18 14:53:37 +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
Alessio Bazzica
be1b8989d1 ExperimentalNs removed + APM not depending anymore on webrtc::Config
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.

Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
2021-09-17 10:53:43 +00:00
Alessio Bazzica
ff7e1bad1f APM config: remove ExperimentalAgc
Bug: webrtc:5298,webrtc:7494
Change-Id: Ic9bcb702603ec7900fbe9ae38ab49dff8fe99318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219463
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35015}
2021-09-16 13:28:51 +00:00
Niels Möller
4f776ac7de Use make_ref_counted in AudioProcessingBuilder
Bug: webrtc:12701
Change-Id: I51ca5a54f812a1620ee2e6605c9ff67b92e2a5f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34725}
2021-08-11 15:40:28 +00:00
Per Åhgren
cc73ed3e70 APM: Add build flag to allow building WebRTC without APM
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.

Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
2020-04-26 23:06:44 +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
Sam Zackrisson
31b01c0534 Fuzz APM float interface with up to 8 channels
Bug: webrtc:10859
Change-Id: Ie50b5fc102296bd71917852674cd2289e690ad78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160305
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29891}
2019-11-25 08:50:44 +00:00
saza
6787f232ae Remove AudioProcessing::level_estimator() getter
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.

Bug: webrtc:9878
Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29454}
2019-10-11 18:08:17 +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
Danil Chapovalov
32b183432d Use default task queue factory in fuzzers
Bug: webrtc:10284
Change-Id: I31a7fe08f1ff3c4842ba657586a158bf632501d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145217
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28549}
2019-07-12 07:56:10 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Danil Chapovalov
cb2a4ffb2b Reland "Remove TaskQueue constructor that uses GlobalTaskQueueFactory"
This reverts commit 57f2a5485a4c83c8045d0dbe9db6281d6bcda847.

Reason for revert: the breakage addressed with a separate change
https://webrtc-review.googlesource.com/c/src/+/131398

Original change's description:
> Revert "Remove TaskQueue constructor that uses GlobalTaskQueueFactory"
> 
> This reverts commit 7b7485b796ad77809e3343f3256013488b418235.
> 
> Reason for revert: Breaks Chrome autoroll 
> 
> video/video_stream_decoder_impl.cc:28:7: error: no matching constructor for initialization of 'rtc::TaskQueue'
>       bookkeeping_queue_("video_stream_decoder_bookkeeping_queue"),
> 
> Original change's description:
> > Remove TaskQueue constructor that uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: I9547fb7110222ce3a3c2323ae2a004024eab911e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130471
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27464}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10284
> Change-Id: I7684f7c7d5501cc910ac9f9daa8ccf6bdb10f8e1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131338
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27491}

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

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

Bug: webrtc:10284
Change-Id: I0a0544d4b82adaec468d3445b6554a7b94d52db5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132225
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27537}
2019-04-10 10:27:07 +00:00
Florent Castelli
57f2a5485a Revert "Remove TaskQueue constructor that uses GlobalTaskQueueFactory"
This reverts commit 7b7485b796ad77809e3343f3256013488b418235.

Reason for revert: Breaks Chrome autoroll 

video/video_stream_decoder_impl.cc:28:7: error: no matching constructor for initialization of 'rtc::TaskQueue'
      bookkeeping_queue_("video_stream_decoder_bookkeeping_queue"),

Original change's description:
> Remove TaskQueue constructor that uses GlobalTaskQueueFactory
> 
> Bug: webrtc:10284
> Change-Id: I9547fb7110222ce3a3c2323ae2a004024eab911e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130471
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27464}

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

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

Bug: webrtc:10284
Change-Id: I7684f7c7d5501cc910ac9f9daa8ccf6bdb10f8e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131338
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27491}
2019-04-08 14:44:48 +00:00
Danil Chapovalov
7b7485b796 Remove TaskQueue constructor that uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: I9547fb7110222ce3a3c2323ae2a004024eab911e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130471
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27464}
2019-04-05 13:34:26 +00:00
Sam Zackrisson
f0d1c03c31 Add replacement interface for webrtc::GainConrol
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
   to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
   GainControlImpl into the GainControlConfigProxy, as it becomes the
   sole AGC object with functionality exposed to the client.

Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
2019-03-27 15:19:41 +00:00
Gustaf Ullberg
9249fbf3a6 AEC3: Redesign delay headroom
This change reduces the risk of echo due to noise in the headroom
of the linear filter.

Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced

Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
2019-03-14 11:04:47 +00:00
Sam Zackrisson
d71edac904 Add an input size limit to APM fuzzer
The fuzzer times out on too long inputs.
This CL limits tests to 400 000 bytes, ~ 12 seconds of 8 kHz float audio.

Bug: chromium:940209
Change-Id: I86b772f9d8989a8b129d933d25ece3631a6a365f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126780
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27059}
2019-03-11 14:01:09 +00:00
Gustaf Ullberg
aa1a43e31f AEC3: Use minimum ERLE during onsets
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.

Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
2019-02-21 14:18:44 +00:00
Gustaf Ullberg
68d6d44197 AEC3: Remove remaining kill-switches
This CL concludes the post-launch removal of kill-switches is AEC3.

Kill-switches removed:
WebRTC-Aec3AdaptErleOnLowRenderKillSwitch
WebRTC-Aec3AgcGainChangeResponseKillSwitch
WebRTC-Aec3BoundedNearendKillSwitch
WebRTC-Aec3EarlyShadowFilterJumpstartKillSwitch
WebRTC-Aec3EnableAdaptiveEchoReverbEstimation
WebRTC-Aec3EnforceSkewHysteresis1
WebRTC-Aec3EnforceSkewHysteresis2
WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch
WebRTC-Aec3MisadjustmentEstimatorKillSwitch
WebRTC-Aec3OverrideEchoPathGainKillSwitch
WebRTC-Aec3RapidAgcGainRecoveryKillSwitch
WebRTC-Aec3ResetErleAtGainChangesKillSwitch
WebRTC-Aec3ShadowFilterBoostedJumpstartKillSwitch
WebRTC-Aec3ShadowFilterJumpstartKillSwitch
WebRTC-Aec3SmoothSignalTransitionsKillSwitch
WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch
WebRTC-Aec3SoftTransparentModeKillSwitch
WebRTC-Aec3StandardNonlinearReverbModelKillSwitch
WebRTC-Aec3StrictDivergenceCheckKillSwitch
WebRTC-Aec3UseOffsetBlocks
WebRTC-Aec3UseStationarityPropertiesKillSwitch
WebRTC-Aec3UtilizeShadowFilterOutputKillSwitch
WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch
WebRTC-Aec3FilterQualityStateKillSwitch
WebRTC-Aec3NewSaturationBehaviorKillSwitch
WebRTC-Aec3GainLimiterDeactivationKillSwitch
WebRTC-Aec3EnableErleUpdatesDuringReverbKillSwitch

The change has been tested for bit-exactness.

Bug: webrtc:8671
Change-Id: I42816b9d1c875cec0347034c6e2ed4ff5db6ec0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119942
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26443}
2019-01-29 10:31:45 +00:00
Gustaf Ullberg
99ec6f39b9 AEC3: Remove unused kill-switches from AdjustConfig
Kill-switches removed:
WebRTC-Aec3UseShortDelayEstimatorWindow
WebRTC-Aec3ReverbBasedOnRenderKillSwitch
WebRTC-Aec3ReverbModellingKillSwitch
WebRTC-Aec3EnableUnityInitialRampupGain
WebRTC-Aec3EnableUnityNonZeroRampupGain
WebRTC-Aec3ShortReverbKillSwitch
WebRTC-Aec3NewFilterParamsKillSwitch
WebRTC-Aec3EnableLegacyDominantNearend
WebRTC-Aec3UseLegacyNormalSuppressorTuning
WebRTC-Aec3UseStationarityProperties
WebRTC-Aec3UseStationarityPropertiesAtInit
WebRTC-Aec3EarlyDelayDetectionKillSwitch

The change is tested for bit-exactness.

Bug: webrtc:8671
Change-Id: Ic7638002c0ca1bc5fc911e048285134c4df5d134
Reviewed-on: https://webrtc-review.googlesource.com/c/119921
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26403}
2019-01-25 13:37:13 +00:00
Gustaf Ullberg
e47433f017 AEC3: Remove legacy render buffering
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.

Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
2019-01-25 08:31:12 +00:00
Sam Zackrisson
235131303b Add noise suppression settings to AudioProcessing::Config
This Config configuration will eventually replace the AudioProcessing::noise_suppression() interface.

This also introduces a proxy NoiseSuppression, returned by AudioProcessing::noise_suppression.
Without this proxy, ApplyConfig could overwrite NS settings for clients who currently use noise_suppression(). For example, the following code will not preserve the noise suppression level:

apm->noise_suppression()->set_level(NoiseSuppression::kHigh);
auto cfg = apm->GetConfig();
apm->ApplyConfig(cfg);

The NoiseSuppression instance returned by noise_suppression() has no way to update the config inside APM, so GetConfig() will return an out-of-date config which is then re-applied. This CL adds a proxy that makes this update, by forwarding Enable() and set_level() calls to ApplyConfig().

Drive-by change: AudioProcessing::Config substructs are reordered to mirror the capture processing pipeline.

Tested: Ran ToT and this CL builds of audioproc_f and verified identical settings/aecdumps.
Bug: webrtc:9947
Change-Id: I823eade894be115c254d656562564108b2b63b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/116521
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26248}
2019-01-14 16:17:19 +00:00
Sam Zackrisson
6c330ab63f Update some audio processing tests to new VAD API
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.

Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.

Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
2019-01-04 10:31:42 +00:00
Sam Zackrisson
6c95e2d56a Fuzz unfuzzed AEC3 killswitch field trials
Bug: webrtc:9413
Change-Id: Iccf861453c1c49c306ad18542074a792592491a9
Reviewed-on: https://webrtc-review.googlesource.com/c/113501
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25928}
2018-12-07 09:38:49 +00:00
Alessio Bazzica
dc107965bd Fix AGC2 fuzzer coverage.
Bug: webrtc:10084
Change-Id: Icc51994fe5ab16188c41452e887cbe7a6b8b9aff
Reviewed-on: https://webrtc-review.googlesource.com/c/112941
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25906}
2018-12-05 15:55:42 +00:00
Alessio Bazzica
1e2542f593 AGC2: adding level estimation option (RMS or peak-based).
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).

Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.

Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:32:13 +00:00
Alex Loiko
20f60f0dc6 Fuzzer crash in AGC2.
Gain specified by fuzzer in APM config was too high.

Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25594}
2018-11-12 12:16:47 +00:00
Per Åhgren
8b7d206d37 AEC3: Decrease latency until the delay has been detected
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.

On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.

Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
2018-10-31 07:29:48 +00:00
Yves Gerey
9516c38538 [Fuzzer] Check FieldTrial bitmask size at compile time.
Rather fail at compile time than at run-time.

Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
2018-10-25 13:21:31 +00:00
Jonas Olsson
e068ad6262 Use a sufficiently large bitmask.
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.

Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
2018-10-24 09:27:18 +00:00
Alex Loiko
4842c78e93 Increasing APM fuzzer coverage.
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.

Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
2018-10-23 15:11:02 +00:00
Gustaf Ullberg
6ed37ba1b5 AEC3: Enable fuzzer testing of old render buffering code.
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.

Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
2018-10-23 08:43:00 +00:00
Gustaf Ullberg
53e22113fd AEC3: Kill kill-switches
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."

This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch

It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.

The change has been tested for bit-exactness.

Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
2018-10-11 16:11:07 +00:00
Mirko Bonadei
17f4878419 Remove deprecated field_trial_default and metrics_default.
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default

It also refreshes all the dependencies on field_trial.h and metrics.h.

A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm

Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 07:21:07 +00:00