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03cb7e5a61
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}
212 lines
9.9 KiB
C++
212 lines
9.9 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/system/arch.h"
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namespace webrtc {
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namespace {
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using Agc1Config = AudioProcessing::Config::GainController1;
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using Agc2Config = AudioProcessing::Config::GainController2;
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std::string NoiseSuppressionLevelToString(
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const AudioProcessing::Config::NoiseSuppression::Level& level) {
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switch (level) {
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case AudioProcessing::Config::NoiseSuppression::Level::kLow:
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return "Low";
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case AudioProcessing::Config::NoiseSuppression::Level::kModerate:
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return "Moderate";
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case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
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return "High";
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case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
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return "VeryHigh";
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}
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RTC_CHECK_NOTREACHED();
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}
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std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
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switch (mode) {
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case Agc1Config::Mode::kAdaptiveAnalog:
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return "AdaptiveAnalog";
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case Agc1Config::Mode::kAdaptiveDigital:
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return "AdaptiveDigital";
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case Agc1Config::Mode::kFixedDigital:
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return "FixedDigital";
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}
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RTC_CHECK_NOTREACHED();
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}
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} // namespace
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constexpr int AudioProcessing::kNativeSampleRatesHz[];
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void CustomProcessing::SetRuntimeSetting(
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AudioProcessing::RuntimeSetting setting) {}
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bool Agc1Config::operator==(const Agc1Config& rhs) const {
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const auto& analog_lhs = analog_gain_controller;
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const auto& analog_rhs = rhs.analog_gain_controller;
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return enabled == rhs.enabled && mode == rhs.mode &&
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target_level_dbfs == rhs.target_level_dbfs &&
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compression_gain_db == rhs.compression_gain_db &&
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enable_limiter == rhs.enable_limiter &&
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analog_lhs.enabled == analog_rhs.enabled &&
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analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
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analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
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analog_lhs.enable_digital_adaptive ==
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analog_rhs.enable_digital_adaptive &&
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analog_lhs.clipped_level_step == analog_rhs.clipped_level_step &&
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analog_lhs.clipped_ratio_threshold ==
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analog_rhs.clipped_ratio_threshold &&
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analog_lhs.clipped_wait_frames == analog_rhs.clipped_wait_frames &&
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analog_lhs.clipping_predictor.mode ==
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analog_rhs.clipping_predictor.mode &&
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analog_lhs.clipping_predictor.window_length ==
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analog_rhs.clipping_predictor.window_length &&
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analog_lhs.clipping_predictor.reference_window_length ==
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analog_rhs.clipping_predictor.reference_window_length &&
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analog_lhs.clipping_predictor.reference_window_delay ==
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analog_rhs.clipping_predictor.reference_window_delay &&
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analog_lhs.clipping_predictor.clipping_threshold ==
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analog_rhs.clipping_predictor.clipping_threshold &&
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analog_lhs.clipping_predictor.crest_factor_margin ==
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analog_rhs.clipping_predictor.crest_factor_margin &&
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analog_lhs.clipping_predictor.use_predicted_step ==
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analog_rhs.clipping_predictor.use_predicted_step;
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}
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bool Agc2Config::AdaptiveDigital::operator==(
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const Agc2Config::AdaptiveDigital& rhs) const {
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return enabled == rhs.enabled && dry_run == rhs.dry_run &&
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headroom_db == rhs.headroom_db && max_gain_db == rhs.max_gain_db &&
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initial_gain_db == rhs.initial_gain_db &&
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vad_reset_period_ms == rhs.vad_reset_period_ms &&
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adjacent_speech_frames_threshold ==
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rhs.adjacent_speech_frames_threshold &&
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max_gain_change_db_per_second == rhs.max_gain_change_db_per_second &&
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max_output_noise_level_dbfs == rhs.max_output_noise_level_dbfs;
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}
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bool Agc2Config::operator==(const Agc2Config& rhs) const {
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return enabled == rhs.enabled &&
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fixed_digital.gain_db == rhs.fixed_digital.gain_db &&
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adaptive_digital == rhs.adaptive_digital;
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}
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bool AudioProcessing::Config::CaptureLevelAdjustment::operator==(
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const AudioProcessing::Config::CaptureLevelAdjustment& rhs) const {
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return enabled == rhs.enabled && pre_gain_factor == rhs.pre_gain_factor &&
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post_gain_factor == rhs.post_gain_factor &&
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analog_mic_gain_emulation == rhs.analog_mic_gain_emulation;
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}
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bool AudioProcessing::Config::CaptureLevelAdjustment::AnalogMicGainEmulation::
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operator==(const AudioProcessing::Config::CaptureLevelAdjustment::
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AnalogMicGainEmulation& rhs) const {
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return enabled == rhs.enabled && initial_level == rhs.initial_level;
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}
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std::string AudioProcessing::Config::ToString() const {
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char buf[2048];
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rtc::SimpleStringBuilder builder(buf);
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builder << "AudioProcessing::Config{ "
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"pipeline: { "
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"maximum_internal_processing_rate: "
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<< pipeline.maximum_internal_processing_rate
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<< ", multi_channel_render: " << pipeline.multi_channel_render
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<< ", multi_channel_capture: " << pipeline.multi_channel_capture
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<< " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
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<< ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
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<< " },capture_level_adjustment: { enabled: "
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<< capture_level_adjustment.enabled
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<< ", pre_gain_factor: " << capture_level_adjustment.pre_gain_factor
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<< ", post_gain_factor: " << capture_level_adjustment.post_gain_factor
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<< ", analog_mic_gain_emulation: { enabled: "
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<< capture_level_adjustment.analog_mic_gain_emulation.enabled
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<< ", initial_level: "
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<< capture_level_adjustment.analog_mic_gain_emulation.initial_level
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<< " }}, high_pass_filter: { enabled: " << high_pass_filter.enabled
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<< " }, echo_canceller: { enabled: " << echo_canceller.enabled
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<< ", mobile_mode: " << echo_canceller.mobile_mode
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<< ", enforce_high_pass_filtering: "
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<< echo_canceller.enforce_high_pass_filtering
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<< " }, noise_suppression: { enabled: " << noise_suppression.enabled
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<< ", level: "
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<< NoiseSuppressionLevelToString(noise_suppression.level)
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<< " }, transient_suppression: { enabled: "
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<< transient_suppression.enabled
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<< " }, voice_detection: { enabled: " << voice_detection.enabled
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<< " }, gain_controller1: { enabled: " << gain_controller1.enabled
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<< ", mode: " << GainController1ModeToString(gain_controller1.mode)
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<< ", target_level_dbfs: " << gain_controller1.target_level_dbfs
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<< ", compression_gain_db: " << gain_controller1.compression_gain_db
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<< ", enable_limiter: " << gain_controller1.enable_limiter
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<< ", analog_gain_controller { enabled: "
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<< gain_controller1.analog_gain_controller.enabled
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<< ", startup_min_volume: "
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<< gain_controller1.analog_gain_controller.startup_min_volume
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<< ", clipped_level_min: "
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<< gain_controller1.analog_gain_controller.clipped_level_min
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<< ", enable_digital_adaptive: "
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<< gain_controller1.analog_gain_controller.enable_digital_adaptive
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<< ", clipped_level_step: "
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<< gain_controller1.analog_gain_controller.clipped_level_step
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<< ", clipped_ratio_threshold: "
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<< gain_controller1.analog_gain_controller.clipped_ratio_threshold
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<< ", clipped_wait_frames: "
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<< gain_controller1.analog_gain_controller.clipped_wait_frames
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<< ", clipping_predictor: { enabled: "
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<< gain_controller1.analog_gain_controller.clipping_predictor.enabled
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<< ", mode: "
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<< gain_controller1.analog_gain_controller.clipping_predictor.mode
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<< ", window_length: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.window_length
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<< ", reference_window_length: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.reference_window_length
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<< ", reference_window_delay: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.reference_window_delay
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<< ", clipping_threshold: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.clipping_threshold
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<< ", crest_factor_margin: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.crest_factor_margin
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<< ", use_predicted_step: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.use_predicted_step
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<< " }}}, gain_controller2: { enabled: " << gain_controller2.enabled
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<< ", fixed_digital: { gain_db: "
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<< gain_controller2.fixed_digital.gain_db
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<< " }, adaptive_digital: { enabled: "
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<< gain_controller2.adaptive_digital.enabled
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<< ", dry_run: " << gain_controller2.adaptive_digital.dry_run
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<< ", headroom_db: " << gain_controller2.adaptive_digital.headroom_db
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<< ", max_gain_db: " << gain_controller2.adaptive_digital.max_gain_db
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<< ", initial_gain_db: "
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<< gain_controller2.adaptive_digital.initial_gain_db
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<< ", vad_reset_period_ms: "
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<< gain_controller2.adaptive_digital.vad_reset_period_ms
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<< ", adjacent_speech_frames_threshold: "
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<< gain_controller2.adaptive_digital.adjacent_speech_frames_threshold
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<< ", max_gain_change_db_per_second: "
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<< gain_controller2.adaptive_digital.max_gain_change_db_per_second
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<< ", max_output_noise_level_dbfs: "
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<< gain_controller2.adaptive_digital.max_output_noise_level_dbfs
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<< "}}";
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return builder.str();
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}
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} // namespace webrtc
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