In order to experiment with AGC2 and TS at the same time, 3 field
trials are removed and merged into `WebRTC-Audio-GainController2`,
which is existing.
New parameters for the `WebRTC-Audio-GainController2` field trial:
- `switch_to_agc2`: true by default; when true, the gain control
switches to AGC2 (both for the input volume and for the adaptive
digital gain);
- `min_input_volume`: minimum input volume enforced by the input
volume controller when the applied input volume is not zero;
- `disallow_transient_suppressor_usage`: when true, TS is never
created.
Removed field trials:
- `WebRTC-Audio-Agc2-MinInputVolume`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-ApmTransientSuppressorKillSwitch`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-Audio-TransientSuppressorVadMode`: automatically inferred
from `WebRTC-Audio-GainController2`.
Bug: webrtc:7494
Change-Id: I452798c0862d71f9adae6d163fe841df05ca44d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287861
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38890}
Add AGC2 digital adaptive config parameters in the field trial
"WebRTC-Audio-InputVolumeControllerExperiment". Rename it as
"WebRTC-Audio-GainController2" to reflect that the override now adjusts
the parameters for both input volume controller and adaptive digital
controller.
Bug: webrtc:7494
Change-Id: Ifbc1b8be76cf23b0b6b74b22b5167a45972cab38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286880
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38855}
Regardless of the APM config, the transient suppressor (TS) submodule
won't be created if the `WebRTC-ApmTransientSuppressorKillSwitch`
field trial, disabled by default, is enabled.
Bug: webrtc:13663
Change-Id: Ic1ef9aa57c728296d671d4ef253630c581a86610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286382
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38839}
Add a field trial WebRTC-Audio-InputVolumeControllerExperiment and
a mechanism to adjust the config accordingly. Pass the additional
input volume controller config to GainController2.
Bug: webrtc:7494
Change-Id: I3dd624df1f4774cb533417747627995e1f60aa68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284101
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38780}
This CL adds more explicit tests for unsupported sample rates in the WebRTC audio processing module (APM). Rates are restricted to the range [8000, 384000] Hz. Rates outside this range are handled as best as possible, depending on the format.
Tested: bitexact on a large number of aecdumps
Bug: chromium:1332484, chromium:1334991
Change-Id: I9639d03dc837e1fdff64d1f9d1fff0edc0fb299f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276920
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38663}
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}
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}
Adopt the new naming convention, which replaces "analog gain" and
"mic level" with "input volume", in the input volume stats reporter.
Bug: webrtc:7494
Change-Id: Ia24876151f51dd1dcc4e4f9db56c64d11ae3b442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279840
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38467}
The current design of the modified getter is error-prone since the
returned value changes meaning based on when (which point in the code)
the getter is called - namely, before `ProcessStream()` is called the
getter returns the stream analog level, after it returns the
recommended level.
Plus, the new implementation, which essentially returns a local
member, removes the risks that the non-trivial implementation
is computationally expensive.
Bug: webrtc:7494, b/241923537
Change-Id: I6714444df27bcc055ae693974ecd1f77f79e6ec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271580
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38055}
The `recommended_stream_analog_level()` getter is used to retrieve
both the applied and the recommended input volume. This behavior is
error-prone since the caller must know what is returned based on
the point in the code (namely, before/after the AGC has changed
the last applied input volume into a recommended level).
This CL is a first step to make clarity on which input volume is
handled in different parts of APM. Next in the pipeline: make
`recommended_stream_analog_level()` a trivial getter that always
returns the recommended level.
Main changes:
- When `recommended_stream_analog_level()` is called but
`set_stream_analog_level()` is not called, APM logs an error
and returns a fall-back volume (which should not be applied
since, when `set_stream_analog_level()` is not called, no
external input volume is expected to be present
- When APM is used without calling the `*_stream_analog_level()`
methods (e.g., when the caller does not provide any input volume),
the recorded AEC dumps won't store `Stream::applied_input_level`
Other changes:
- Removed `AudioProcessingImpl::capture_::prev_analog_mic_level`
- Removed redundant code in `GainController2` around detecting
input volume changes (already done by APM)
- Adapted the `audioproc_f` and `unpack_aecdump` tools
- Data dumps clean-up: the applied and the recommended input
volumes are now recorded in an AGC implementation agnostic way
Bug: webrtc:7494, b/241923537
Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38054}
Switching to an AGC implementation agnostic solution for the input
volume emulation functionality offered by the
`capture_levels_adjuster` sub-module.
This CL also fixes a (silent) bug due to which, when the input
volume is emulated via the capture adjuster sub-module, AGC2
reads an incorrect value for the applied input volume.
Tested: audioproc_f with `--analog_mic_gain_emulation 1` used
to verify bit-exactness for one Wav file and one AEC dump for
which the input volume varies.
Bug: webrtc:7494, b/241923537
Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38053}
This is the first step of migrating
AudioProcessing::CreateAndAttachAecDump() from using std::string to
absl::string_view.
Bug: webrtc:13579
Change-Id: I8fc373e7ac55fd8e96bb0b01d1a30e28883ac9a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269400
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37631}
Add a VoiceActivityDetectorWrapper submodule in AudioProcessingImpl
and enable injecting speech probability into GainController2.
Bug: webrtc:13663
Change-Id: I05e13b737d085b45ac8ce76660191867c56834c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265166
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37275}
It is now required to specify which VAD is used to compute the speech
probability passed when `TransientSuppressor::Suppress()` is called.
In this way, it is possible to adapt parameters and/or logic of a
`TransientSuppressor` implementation to the behavior of the used
VAD. This CL also adds a "no VAD" mode option, which ignores the speech
probability argument passed when `Suppress()` and always applies mild
suppression to preserve transparency.
Finally, this CL adds a field trial to choose which VAD is used by
APM for transient suppression. Wiring the RNN VAD to TS will be done
in a follow-up CL.
Bug: webrtc:13663
Change-Id: I21ed49f91875a4ee0f04db97ea87c0dbc3db7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250962
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36485}
This enum is no longer needed. Also moving the last piece of code from
common.h to audio_processing_impl.h, allowing to delete common.h.
Bug: chromium:1271981, b/217349489
Change-Id: If115336c36d6d7b5845a903e421c18aebfe434ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251242
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35946}
- Switch from ptr+size to rtc::ArrayView
- Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz
- Stop passing nullptr in agc_manager_direct_unittest.cc when
`AgcManagerDirect::Process()` is called
- Allow to correctly run the tests added in the child CL (see [1])
[1] https://webrtc-review.googlesource.com/c/src/+/250141
Bug: webrtc:7494
Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35910}
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}
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}
Implement AnalogGainStatsReporter and add it in AudioProcessingImpl.
This class computes statistics for analog gain updates and
periodically reports them into a histogram.
The added histograms for analog gain update statistics:
- WebRTC.Audio.ApmAnalogGainDecreaseRate
- WebRTC.Audio.ApmAnalogGainIncreaseRate
- WebRTC.Audio.ApmAnalogGainUpdateRate
- WebRTC.Audio.ApmAnalogGainDecreaseAverage
- WebRTC.Audio.ApmAnalogGainIncreaseAverage
- WebRTC.Audio.ApmAnalogGainUpdateAverage
The histograms are defined in
https://chromium-review.googlesource.com/c/chromium/src/+/3207987
Bug: webrtc:12774
Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35301}
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}
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized
and then the new config is applied. That is error prone and for example
breaks bit exactness in [1].
Changes:
- `GainController2` must be created by passing configuration,
sample rate and number of channels
- `GainController2::ApplyConfig()` removed
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234587.
Bug: webrtc:7494
Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35206}
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}
Denormal numbers (see [1]) may origin in APM when the input is zeroed
after a non-zero signal. In extreme cases, instructions involving
denormal operands may run as much as 100 times slower, which seems to
be the case (to some extent) of crbug.com/1227566.
This CL adds a class that disables denormals only via hardware on x86
and on ARM. The class is used in APM and it is an adaption of [2].
Tested: appr.tc call on Chromium (Win, Mac)
[1] https://en.wikipedia.org/wiki/Denormal_number
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/platform/audio/denormal_disabler.h
Fixed: chromium:1227566
Change-Id: I0ed2eab55dc597529f09f93c26c7a01de051fdbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227768
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34701}
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.
Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
This change adds the field trial "WebRTC-TransientSuppressorForcedOff"
that can be used to disable the transient suppressor (removal of
keyboard typing sounds). The field trial can be enabled by users via
command-line or via experimentation.
Bug: chromium:1186705
Change-Id: I7272df6a20fbbee24a7ba0904502c76bd775d275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219282
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34038}
This is a reland of aa6adffba3
What was changed in the reland is that the merging of the bands is
excluded from the code that is not run when the output is not used.
I.e., the merging is always done.
This is important to have since some clients may apply muting before APM,
and still flag to APM that the signal is muted. If the merging is not
always done, those clients will get nonzero output from APM during muting.
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: Ib74dd1cefa173d45101e26c4f2b931860abc6d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33478}
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}
This CL adds one frame (10 ms) of silence in APM output after unmuting to mask
audio resulting from the turning on the processing that was deactivated
during the muting.
Bug: b/177830919
Change-Id: If44cfb0ef270dde839dcd3f0b98d1c91e81668dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33454}
This reverts commit aa6adffba3.
Reason for revert: breaks webrtc-importer
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: I937cd61dedcd43150933eb1b9d65aebe68401e91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211348
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33433}
This CL selectively turns off parts of the audio processing when
the output of APM is not used. The parts turned off are such that
don't need to continuously need to be trained, but rather can be
temporarily deactivated.
The purpose of this CL is to allow CPU to be reduced when the
client is muted.
The CL will be follow by additional CLs, adding similar functionality
in the echo canceller and the noiser suppressor
Bug: b/177830919
Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33431}
This CL adds the initial support for letting APM know when its output
will be used or not.
It also adds a new method for passing RuntimeInformation to APM that
returns a bool indicating the success of the passing of information.
Bug: b/177830919
Change-Id: Ic2e1b92c37241d74ca6394b785b91736ca7532aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206061
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33201}
