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webrtc_m130/modules/audio_processing/include/audio_processing.cc

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Removed inline definitions and added destructors to fix chromium-style. BUG=webrtc:163 Review-Url: https://codereview.webrtc.org/2272563004 Cr-Commit-Position: refs/heads/master@{#13911}
2016-08-25 00:45:31 -07:00
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
#include "modules/audio_processing/include/audio_processing.h"
Removed inline definitions and added destructors to fix chromium-style. BUG=webrtc:163 Review-Url: https://codereview.webrtc.org/2272563004 Cr-Commit-Position: refs/heads/master@{#13911}
2016-08-25 00:45:31 -07:00
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
#include "rtc_base/strings/string_builder.h"
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}
2019-09-15 00:27:58 +02:00
#include "rtc_base/system/arch.h"
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
Removed inline definitions and added destructors to fix chromium-style. BUG=webrtc:163 Review-Url: https://codereview.webrtc.org/2272563004 Cr-Commit-Position: refs/heads/master@{#13911}
2016-08-25 00:45:31 -07:00
namespace webrtc {
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
namespace {
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
using Agc1Config = AudioProcessing::Config::GainController1;
using Agc2Config = AudioProcessing::Config::GainController2;
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
std::string NoiseSuppressionLevelToString(
const AudioProcessing::Config::NoiseSuppression::Level& level) {
switch (level) {
case AudioProcessing::Config::NoiseSuppression::Level::kLow:
return "Low";
case AudioProcessing::Config::NoiseSuppression::Level::kModerate:
return "Moderate";
case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
return "High";
case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
return "VeryHigh";
}
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() And use it in a few places that were using RTC_CHECK(false) or FATAL() to do the exact same job. There should be no change in behavior. Bug: none Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-08 00:49:37 +01:00
RTC_CHECK_NOTREACHED();
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
}
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
switch (mode) {
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
case Agc1Config::Mode::kAdaptiveAnalog:
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
return "AdaptiveAnalog";
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
case Agc1Config::Mode::kAdaptiveDigital:
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
return "AdaptiveDigital";
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
case Agc1Config::Mode::kFixedDigital:
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
return "FixedDigital";
}
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() And use it in a few places that were using RTC_CHECK(false) or FATAL() to do the exact same job. There should be no change in behavior. Bug: none Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-08 00:49:37 +01:00
RTC_CHECK_NOTREACHED();
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
}
} // namespace
Removed inline definitions and added destructors to fix chromium-style. BUG=webrtc:163 Review-Url: https://codereview.webrtc.org/2272563004 Cr-Commit-Position: refs/heads/master@{#13911}
2016-08-25 00:45:31 -07:00
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
constexpr int AudioProcessing::kNativeSampleRatesHz[];
Add RuntimeSettings to CustomProcessing. CustomProcessing is the interface to injectable audio processing submodules to AudioProcessing. This CL makes it possible to set runtime settings on the injected render processing component. Note that the current runtime setting handling happens on the capture thread. Therefore, we add another SwapQueue to communicate with the render thread. Bug: webrtc:9138, webrtc:9262 Change-Id: I665ce2d83a2b35ca8b25cca813d2cef7bd0ba911 Reviewed-on: https://webrtc-review.googlesource.com/76123 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23236}
2018-05-15 10:52:28 +02:00
void CustomProcessing::SetRuntimeSetting(
AudioProcessing::RuntimeSetting setting) {}
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
bool Agc1Config::operator==(const Agc1Config& rhs) const {
const auto& analog_lhs = analog_gain_controller;
const auto& analog_rhs = rhs.analog_gain_controller;
return enabled == rhs.enabled && mode == rhs.mode &&
target_level_dbfs == rhs.target_level_dbfs &&
compression_gain_db == rhs.compression_gain_db &&
enable_limiter == rhs.enable_limiter &&
analog_lhs.enabled == analog_rhs.enabled &&
analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
analog_lhs.enable_digital_adaptive ==
Make AgcManagerDirect clipping parameters configurable Bug: webrtc:12774 Change-Id: I99824b5aabe6f921a5db425dd1c1c1d4c606186c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219681 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34069}
2021-05-20 17:37:56 +02:00
analog_rhs.enable_digital_adaptive &&
analog_lhs.clipped_level_step == analog_rhs.clipped_level_step &&
analog_lhs.clipped_ratio_threshold ==
analog_rhs.clipped_ratio_threshold &&
Add ClippingPredictor config in AudioProcessing config Bug: webrtc:12774 Change-Id: Id8cdb6b5499a22cbca40d424cf936f81c1e7d8d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221104 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34204}
2021-06-02 17:13:24 +02:00
analog_lhs.clipped_wait_frames == analog_rhs.clipped_wait_frames &&
analog_lhs.clipping_predictor.mode ==
analog_rhs.clipping_predictor.mode &&
analog_lhs.clipping_predictor.window_length ==
analog_rhs.clipping_predictor.window_length &&
analog_lhs.clipping_predictor.reference_window_length ==
analog_rhs.clipping_predictor.reference_window_length &&
analog_lhs.clipping_predictor.reference_window_delay ==
analog_rhs.clipping_predictor.reference_window_delay &&
analog_lhs.clipping_predictor.clipping_threshold ==
analog_rhs.clipping_predictor.clipping_threshold &&
analog_lhs.clipping_predictor.crest_factor_margin ==
AudioProcessing: Add use_predicted_step in GainController1 comparison and string conversion Bug: webrtc:12774 Change-Id: I8d268b343420108a715698ad45fec3c54f0962fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231644 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35084}
2021-09-09 12:14:39 +02:00
analog_rhs.clipping_predictor.crest_factor_margin &&
analog_lhs.clipping_predictor.use_predicted_step ==
analog_rhs.clipping_predictor.use_predicted_step;
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
}
AGC2 config operator= and unit tests update Bug: webrtc:7494 Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@google.com> Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:17:49 +02:00
bool Agc2Config::AdaptiveDigital::operator==(
const Agc2Config::AdaptiveDigital& rhs) const {
return enabled == rhs.enabled && dry_run == rhs.dry_run &&
AGC2 config: allow tuning of headroom, max gain and initial gain This CL does *not* change the behavior of the AGC2 adaptive digital controller - bitexactness verified with audioproc_f on a collection of AEC dumps and Wav files (42 recordings in total). Tested: compiled Chrome with this patch and made an appr.tc test call Bug: webrtc:7494 Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35140}
2021-10-04 13:35:55 +02:00
headroom_db == rhs.headroom_db && max_gain_db == rhs.max_gain_db &&
initial_gain_db == rhs.initial_gain_db &&
AGC2 config operator= and unit tests update Bug: webrtc:7494 Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@google.com> Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:17:49 +02:00
vad_reset_period_ms == rhs.vad_reset_period_ms &&
adjacent_speech_frames_threshold ==
rhs.adjacent_speech_frames_threshold &&
max_gain_change_db_per_second == rhs.max_gain_change_db_per_second &&
AGC2: SIMD allowed config flags to field trials Bug: webrtc:7494 Change-Id: I41fa05d2ef6d969750f3d4c1e40ecbcd30293b5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233741 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35144}
2021-10-04 18:33:37 +02:00
max_output_noise_level_dbfs == rhs.max_output_noise_level_dbfs;
AGC2 config operator= and unit tests update Bug: webrtc:7494 Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@google.com> Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:17:49 +02:00
}
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
AGC2 config operator= and unit tests update Bug: webrtc:7494 Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@google.com> Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:17:49 +02:00
bool Agc2Config::operator==(const Agc2Config& rhs) const {
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
return enabled == rhs.enabled &&
fixed_digital.gain_db == rhs.fixed_digital.gain_db &&
AGC2 config operator= and unit tests update Bug: webrtc:7494 Change-Id: I7957602d60658b84175a617f9c283ea44eafe57d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216260 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@google.com> Cr-Commit-Position: refs/heads/master@{#33876}
2021-04-29 16:17:49 +02:00
adaptive_digital == rhs.adaptive_digital;
AGC2 config change detection fixed AGC2 is correctly (re)initialized when its config changes. This CL also improves the `AudioProcessingImpl::ApplyConfig` readability by defining operator!= also for the AGC1 config. Bug: webrtc:7494 Change-Id: I62068de32c941e6b18d4618c656f569647042345 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32402}
2020-10-14 12:47:50 +02:00
}
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 16:31:04 +00:00
bool AudioProcessing::Config::CaptureLevelAdjustment::operator==(
const AudioProcessing::Config::CaptureLevelAdjustment& rhs) const {
return enabled == rhs.enabled && pre_gain_factor == rhs.pre_gain_factor &&
APM: fix CaptureLevelAdjustment::operator== Bug: webrtc:7494 Change-Id: I0ea13af2e23ed1490fa22d75d104bdd45b0452bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233460 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35142}
2021-10-04 11:48:29 +02:00
post_gain_factor == rhs.post_gain_factor &&
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 16:31:04 +00:00
analog_mic_gain_emulation == rhs.analog_mic_gain_emulation;
}
bool AudioProcessing::Config::CaptureLevelAdjustment::AnalogMicGainEmulation::
operator==(const AudioProcessing::Config::CaptureLevelAdjustment::
AnalogMicGainEmulation& rhs) const {
return enabled == rhs.enabled && initial_level == rhs.initial_level;
}
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
std::string AudioProcessing::Config::ToString() const {
Move AGC2 config ToString to the right place and update Validate() The APM config to string mapping must be in one place (namely, in `audio_processing.cc`). This CL moves the AGC2 config to string impl to the right place. This CL also updates `GainController2::Validate()` and adds the missing unit tests for the parameters that have recently been added. Stack buffer size in `AudioProcessing::Config::ToString()` increased because of the extra params. Syntax near `multi_channel_capture` fixed. Output string format verified with a JS linter. Bug: webrtc:7494 Change-Id: I692e1549b7d40c970d88a14c8e83da16325fb54c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187080 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32400}
2020-10-14 12:49:54 +02:00
char buf[2048];
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
rtc::SimpleStringBuilder builder(buf);
AGC2: SIMD allowed config flags to field trials Bug: webrtc:7494 Change-Id: I41fa05d2ef6d969750f3d4c1e40ecbcd30293b5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233741 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35144}
2021-10-04 18:33:37 +02:00
builder << "AudioProcessing::Config{ "
"pipeline: { "
"maximum_internal_processing_rate: "
<< pipeline.maximum_internal_processing_rate
<< ", multi_channel_render: " << pipeline.multi_channel_render
<< ", multi_channel_capture: " << pipeline.multi_channel_capture
<< " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
<< ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
<< " },capture_level_adjustment: { enabled: "
<< capture_level_adjustment.enabled
<< ", pre_gain_factor: " << capture_level_adjustment.pre_gain_factor
<< ", post_gain_factor: " << capture_level_adjustment.post_gain_factor
<< ", analog_mic_gain_emulation: { enabled: "
<< capture_level_adjustment.analog_mic_gain_emulation.enabled
<< ", initial_level: "
<< capture_level_adjustment.analog_mic_gain_emulation.initial_level
<< " }}, high_pass_filter: { enabled: " << high_pass_filter.enabled
<< " }, echo_canceller: { enabled: " << echo_canceller.enabled
<< ", mobile_mode: " << echo_canceller.mobile_mode
<< ", enforce_high_pass_filtering: "
<< echo_canceller.enforce_high_pass_filtering
<< " }, noise_suppression: { enabled: " << noise_suppression.enabled
<< ", level: "
<< NoiseSuppressionLevelToString(noise_suppression.level)
<< " }, transient_suppression: { enabled: "
<< transient_suppression.enabled
<< " }, gain_controller1: { enabled: " << gain_controller1.enabled
<< ", mode: " << GainController1ModeToString(gain_controller1.mode)
<< ", target_level_dbfs: " << gain_controller1.target_level_dbfs
<< ", compression_gain_db: " << gain_controller1.compression_gain_db
<< ", enable_limiter: " << gain_controller1.enable_limiter
<< ", analog_gain_controller { enabled: "
<< gain_controller1.analog_gain_controller.enabled
<< ", startup_min_volume: "
<< gain_controller1.analog_gain_controller.startup_min_volume
<< ", clipped_level_min: "
<< gain_controller1.analog_gain_controller.clipped_level_min
<< ", enable_digital_adaptive: "
<< gain_controller1.analog_gain_controller.enable_digital_adaptive
<< ", clipped_level_step: "
<< gain_controller1.analog_gain_controller.clipped_level_step
<< ", clipped_ratio_threshold: "
<< gain_controller1.analog_gain_controller.clipped_ratio_threshold
<< ", clipped_wait_frames: "
<< gain_controller1.analog_gain_controller.clipped_wait_frames
<< ", clipping_predictor: { enabled: "
<< gain_controller1.analog_gain_controller.clipping_predictor.enabled
<< ", mode: "
<< gain_controller1.analog_gain_controller.clipping_predictor.mode
<< ", window_length: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.window_length
<< ", reference_window_length: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.reference_window_length
<< ", reference_window_delay: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.reference_window_delay
<< ", clipping_threshold: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.clipping_threshold
<< ", crest_factor_margin: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.crest_factor_margin
<< ", use_predicted_step: "
<< gain_controller1.analog_gain_controller.clipping_predictor
.use_predicted_step
<< " }}}, gain_controller2: { enabled: " << gain_controller2.enabled
<< ", fixed_digital: { gain_db: "
<< gain_controller2.fixed_digital.gain_db
<< " }, adaptive_digital: { enabled: "
<< gain_controller2.adaptive_digital.enabled
<< ", dry_run: " << gain_controller2.adaptive_digital.dry_run
<< ", headroom_db: " << gain_controller2.adaptive_digital.headroom_db
<< ", max_gain_db: " << gain_controller2.adaptive_digital.max_gain_db
<< ", initial_gain_db: "
<< gain_controller2.adaptive_digital.initial_gain_db
<< ", vad_reset_period_ms: "
<< gain_controller2.adaptive_digital.vad_reset_period_ms
<< ", adjacent_speech_frames_threshold: "
<< gain_controller2.adaptive_digital.adjacent_speech_frames_threshold
<< ", max_gain_change_db_per_second: "
<< gain_controller2.adaptive_digital.max_gain_change_db_per_second
<< ", max_output_noise_level_dbfs: "
<< gain_controller2.adaptive_digital.max_output_noise_level_dbfs
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}
2021-12-06 15:40:04 +01:00
<< "}}";
Add ToString method for AudioProcessing::Config Bug: webrtc:10138 Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 11:31:37 +02:00
return builder.str();
}
Removed inline definitions and added destructors to fix chromium-style. BUG=webrtc:163 Review-Url: https://codereview.webrtc.org/2272563004 Cr-Commit-Position: refs/heads/master@{#13911}
2016-08-25 00:45:31 -07:00
} // namespace webrtc
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