| /* |
| * Copyright (c) 2012 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. |
| */ |
| |
| #include "modules/audio_processing/audio_processing_impl.h" |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <memory> |
| #include <string> |
| #include <type_traits> |
| #include <utility> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/audio/audio_frame.h" |
| #include "common_audio/audio_converter.h" |
| #include "common_audio/include/audio_util.h" |
| #include "modules/audio_processing/agc2/gain_applier.h" |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/common.h" |
| #include "modules/audio_processing/include/audio_frame_view.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "modules/audio_processing/transient/transient_suppressor_creator.h" |
| #include "rtc_base/atomic_ops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| #define RETURN_ON_ERR(expr) \ |
| do { \ |
| int err = (expr); \ |
| if (err != kNoError) { \ |
| return err; \ |
| } \ |
| } while (0) |
| |
| namespace webrtc { |
| |
| constexpr int kRuntimeSettingQueueSize = 100; |
| |
| namespace { |
| |
| static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) { |
| switch (layout) { |
| case AudioProcessing::kMono: |
| case AudioProcessing::kStereo: |
| return false; |
| case AudioProcessing::kMonoAndKeyboard: |
| case AudioProcessing::kStereoAndKeyboard: |
| return true; |
| } |
| |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| bool SampleRateSupportsMultiBand(int sample_rate_hz) { |
| return sample_rate_hz == AudioProcessing::kSampleRate32kHz || |
| sample_rate_hz == AudioProcessing::kSampleRate48kHz; |
| } |
| |
| // Checks whether the high-pass filter should be done in the full-band. |
| bool EnforceSplitBandHpf() { |
| return field_trial::IsEnabled("WebRTC-FullBandHpfKillSwitch"); |
| } |
| |
| // Checks whether AEC3 should be allowed to decide what the default |
| // configuration should be based on the render and capture channel configuration |
| // at hand. |
| bool UseSetupSpecificDefaultAec3Congfig() { |
| return !field_trial::IsEnabled( |
| "WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch"); |
| } |
| |
| // Identify the native processing rate that best handles a sample rate. |
| int SuitableProcessRate(int minimum_rate, |
| int max_splitting_rate, |
| bool band_splitting_required) { |
| const int uppermost_native_rate = |
| band_splitting_required ? max_splitting_rate : 48000; |
| for (auto rate : {16000, 32000, 48000}) { |
| if (rate >= uppermost_native_rate) { |
| return uppermost_native_rate; |
| } |
| if (rate >= minimum_rate) { |
| return rate; |
| } |
| } |
| RTC_NOTREACHED(); |
| return uppermost_native_rate; |
| } |
| |
| GainControl::Mode Agc1ConfigModeToInterfaceMode( |
| AudioProcessing::Config::GainController1::Mode mode) { |
| using Agc1Config = AudioProcessing::Config::GainController1; |
| switch (mode) { |
| case Agc1Config::kAdaptiveAnalog: |
| return GainControl::kAdaptiveAnalog; |
| case Agc1Config::kAdaptiveDigital: |
| return GainControl::kAdaptiveDigital; |
| case Agc1Config::kFixedDigital: |
| return GainControl::kFixedDigital; |
| } |
| } |
| |
| // Maximum lengths that frame of samples being passed from the render side to |
| // the capture side can have (does not apply to AEC3). |
| static const size_t kMaxAllowedValuesOfSamplesPerBand = 160; |
| static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480; |
| |
| // Maximum number of frames to buffer in the render queue. |
| // TODO(peah): Decrease this once we properly handle hugely unbalanced |
| // reverse and forward call numbers. |
| static const size_t kMaxNumFramesToBuffer = 100; |
| } // namespace |
| |
| // Throughout webrtc, it's assumed that success is represented by zero. |
| static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero"); |
| |
| AudioProcessingImpl::SubmoduleStates::SubmoduleStates( |
| bool capture_post_processor_enabled, |
| bool render_pre_processor_enabled, |
| bool capture_analyzer_enabled) |
| : capture_post_processor_enabled_(capture_post_processor_enabled), |
| render_pre_processor_enabled_(render_pre_processor_enabled), |
| capture_analyzer_enabled_(capture_analyzer_enabled) {} |
| |
| bool AudioProcessingImpl::SubmoduleStates::Update( |
| bool high_pass_filter_enabled, |
| bool mobile_echo_controller_enabled, |
| bool residual_echo_detector_enabled, |
| bool noise_suppressor_enabled, |
| bool adaptive_gain_controller_enabled, |
| bool gain_controller2_enabled, |
| bool pre_amplifier_enabled, |
| bool echo_controller_enabled, |
| bool voice_detector_enabled, |
| bool transient_suppressor_enabled) { |
| bool changed = false; |
| changed |= (high_pass_filter_enabled != high_pass_filter_enabled_); |
| changed |= |
| (mobile_echo_controller_enabled != mobile_echo_controller_enabled_); |
| changed |= |
| (residual_echo_detector_enabled != residual_echo_detector_enabled_); |
| changed |= (noise_suppressor_enabled != noise_suppressor_enabled_); |
| changed |= |
| (adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_); |
| changed |= (gain_controller2_enabled != gain_controller2_enabled_); |
| changed |= (pre_amplifier_enabled_ != pre_amplifier_enabled); |
| changed |= (echo_controller_enabled != echo_controller_enabled_); |
| changed |= (voice_detector_enabled != voice_detector_enabled_); |
| changed |= (transient_suppressor_enabled != transient_suppressor_enabled_); |
| if (changed) { |
| high_pass_filter_enabled_ = high_pass_filter_enabled; |
| mobile_echo_controller_enabled_ = mobile_echo_controller_enabled; |
| residual_echo_detector_enabled_ = residual_echo_detector_enabled; |
| noise_suppressor_enabled_ = noise_suppressor_enabled; |
| adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled; |
| gain_controller2_enabled_ = gain_controller2_enabled; |
| pre_amplifier_enabled_ = pre_amplifier_enabled; |
| echo_controller_enabled_ = echo_controller_enabled; |
| voice_detector_enabled_ = voice_detector_enabled; |
| transient_suppressor_enabled_ = transient_suppressor_enabled; |
| } |
| |
| changed |= first_update_; |
| first_update_ = false; |
| return changed; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandSubModulesActive() |
| const { |
| return CaptureMultiBandProcessingPresent() || voice_detector_enabled_; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingPresent() |
| const { |
| // If echo controller is present, assume it performs active processing. |
| return CaptureMultiBandProcessingActive(/*ec_processing_active=*/true); |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingActive( |
| bool ec_processing_active) const { |
| return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ || |
| noise_suppressor_enabled_ || adaptive_gain_controller_enabled_ || |
| (echo_controller_enabled_ && ec_processing_active); |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::CaptureFullBandProcessingActive() |
| const { |
| return gain_controller2_enabled_ || capture_post_processor_enabled_ || |
| pre_amplifier_enabled_; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::CaptureAnalyzerActive() const { |
| return capture_analyzer_enabled_; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandSubModulesActive() |
| const { |
| return RenderMultiBandProcessingActive() || mobile_echo_controller_enabled_ || |
| adaptive_gain_controller_enabled_ || echo_controller_enabled_; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::RenderFullBandProcessingActive() |
| const { |
| return render_pre_processor_enabled_; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandProcessingActive() |
| const { |
| return false; |
| } |
| |
| bool AudioProcessingImpl::SubmoduleStates::HighPassFilteringRequired() const { |
| return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ || |
| noise_suppressor_enabled_; |
| } |
| |
| AudioProcessingBuilder::AudioProcessingBuilder() = default; |
| AudioProcessingBuilder::~AudioProcessingBuilder() = default; |
| |
| AudioProcessingBuilder& AudioProcessingBuilder::SetCapturePostProcessing( |
| std::unique_ptr<CustomProcessing> capture_post_processing) { |
| capture_post_processing_ = std::move(capture_post_processing); |
| return *this; |
| } |
| |
| AudioProcessingBuilder& AudioProcessingBuilder::SetRenderPreProcessing( |
| std::unique_ptr<CustomProcessing> render_pre_processing) { |
| render_pre_processing_ = std::move(render_pre_processing); |
| return *this; |
| } |
| |
| AudioProcessingBuilder& AudioProcessingBuilder::SetCaptureAnalyzer( |
| std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) { |
| capture_analyzer_ = std::move(capture_analyzer); |
| return *this; |
| } |
| |
| AudioProcessingBuilder& AudioProcessingBuilder::SetEchoControlFactory( |
| std::unique_ptr<EchoControlFactory> echo_control_factory) { |
| echo_control_factory_ = std::move(echo_control_factory); |
| return *this; |
| } |
| |
| AudioProcessingBuilder& AudioProcessingBuilder::SetEchoDetector( |
| rtc::scoped_refptr<EchoDetector> echo_detector) { |
| echo_detector_ = std::move(echo_detector); |
| return *this; |
| } |
| |
| AudioProcessing* AudioProcessingBuilder::Create() { |
| webrtc::Config config; |
| return Create(config); |
| } |
| |
| AudioProcessing* AudioProcessingBuilder::Create(const webrtc::Config& config) { |
| AudioProcessingImpl* apm = new rtc::RefCountedObject<AudioProcessingImpl>( |
| config, std::move(capture_post_processing_), |
| std::move(render_pre_processing_), std::move(echo_control_factory_), |
| std::move(echo_detector_), std::move(capture_analyzer_)); |
| if (apm->Initialize() != AudioProcessing::kNoError) { |
| delete apm; |
| apm = nullptr; |
| } |
| return apm; |
| } |
| |
| AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config) |
| : AudioProcessingImpl(config, |
| /*capture_post_processor=*/nullptr, |
| /*render_pre_processor=*/nullptr, |
| /*echo_control_factory=*/nullptr, |
| /*echo_detector=*/nullptr, |
| /*capture_analyzer=*/nullptr) {} |
| |
| int AudioProcessingImpl::instance_count_ = 0; |
| |
| AudioProcessingImpl::AudioProcessingImpl( |
| const webrtc::Config& config, |
| std::unique_ptr<CustomProcessing> capture_post_processor, |
| std::unique_ptr<CustomProcessing> render_pre_processor, |
| std::unique_ptr<EchoControlFactory> echo_control_factory, |
| rtc::scoped_refptr<EchoDetector> echo_detector, |
| std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) |
| : data_dumper_( |
| new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))), |
| use_setup_specific_default_aec3_config_( |
| UseSetupSpecificDefaultAec3Congfig()), |
| capture_runtime_settings_(kRuntimeSettingQueueSize), |
| render_runtime_settings_(kRuntimeSettingQueueSize), |
| capture_runtime_settings_enqueuer_(&capture_runtime_settings_), |
| render_runtime_settings_enqueuer_(&render_runtime_settings_), |
| echo_control_factory_(std::move(echo_control_factory)), |
| submodule_states_(!!capture_post_processor, |
| !!render_pre_processor, |
| !!capture_analyzer), |
| submodules_(std::move(capture_post_processor), |
| std::move(render_pre_processor), |
| std::move(echo_detector), |
| std::move(capture_analyzer)), |
| constants_(!field_trial::IsEnabled( |
| "WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"), |
| !field_trial::IsEnabled( |
| "WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"), |
| EnforceSplitBandHpf()), |
| capture_nonlocked_() { |
| RTC_LOG(LS_INFO) << "Injected APM submodules:" |
| "\nEcho control factory: " |
| << !!echo_control_factory_ |
| << "\nEcho detector: " << !!submodules_.echo_detector |
| << "\nCapture analyzer: " << !!submodules_.capture_analyzer |
| << "\nCapture post processor: " |
| << !!submodules_.capture_post_processor |
| << "\nRender pre processor: " |
| << !!submodules_.render_pre_processor; |
| |
| // Mark Echo Controller enabled if a factory is injected. |
| capture_nonlocked_.echo_controller_enabled = |
| static_cast<bool>(echo_control_factory_); |
| |
| // If no echo detector is injected, use the ResidualEchoDetector. |
| if (!submodules_.echo_detector) { |
| submodules_.echo_detector = |
| new rtc::RefCountedObject<ResidualEchoDetector>(); |
| } |
| |
| #if !(defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)) |
| // TODO(webrtc:5298): Remove once the use of ExperimentalNs has been |
| // deprecated. |
| config_.transient_suppression.enabled = config.Get<ExperimentalNs>().enabled; |
| |
| // TODO(webrtc:5298): Remove once the use of ExperimentalAgc has been |
| // deprecated. |
| config_.gain_controller1.analog_gain_controller.enabled = |
| config.Get<ExperimentalAgc>().enabled; |
| config_.gain_controller1.analog_gain_controller.startup_min_volume = |
| config.Get<ExperimentalAgc>().startup_min_volume; |
| config_.gain_controller1.analog_gain_controller.clipped_level_min = |
| config.Get<ExperimentalAgc>().clipped_level_min; |
| config_.gain_controller1.analog_gain_controller.enable_agc2_level_estimator = |
| config.Get<ExperimentalAgc>().enabled_agc2_level_estimator; |
| config_.gain_controller1.analog_gain_controller.enable_digital_adaptive = |
| !config.Get<ExperimentalAgc>().digital_adaptive_disabled; |
| #endif |
| } |
| |
| AudioProcessingImpl::~AudioProcessingImpl() = default; |
| |
| int AudioProcessingImpl::Initialize() { |
| // Run in a single-threaded manner during initialization. |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| return InitializeLocked(); |
| } |
| |
| int AudioProcessingImpl::Initialize(int capture_input_sample_rate_hz, |
| int capture_output_sample_rate_hz, |
| int render_input_sample_rate_hz, |
| ChannelLayout capture_input_layout, |
| ChannelLayout capture_output_layout, |
| ChannelLayout render_input_layout) { |
| const ProcessingConfig processing_config = { |
| {{capture_input_sample_rate_hz, ChannelsFromLayout(capture_input_layout), |
| LayoutHasKeyboard(capture_input_layout)}, |
| {capture_output_sample_rate_hz, |
| ChannelsFromLayout(capture_output_layout), |
| LayoutHasKeyboard(capture_output_layout)}, |
| {render_input_sample_rate_hz, ChannelsFromLayout(render_input_layout), |
| LayoutHasKeyboard(render_input_layout)}, |
| {render_input_sample_rate_hz, ChannelsFromLayout(render_input_layout), |
| LayoutHasKeyboard(render_input_layout)}}}; |
| |
| return Initialize(processing_config); |
| } |
| |
| int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) { |
| // Run in a single-threaded manner during initialization. |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| return InitializeLocked(processing_config); |
| } |
| |
| int AudioProcessingImpl::MaybeInitializeRender( |
| const ProcessingConfig& processing_config) { |
| // Called from both threads. Thread check is therefore not possible. |
| if (processing_config == formats_.api_format) { |
| return kNoError; |
| } |
| |
| rtc::CritScope cs_capture(&crit_capture_); |
| return InitializeLocked(processing_config); |
| } |
| |
| int AudioProcessingImpl::InitializeLocked() { |
| UpdateActiveSubmoduleStates(); |
| |
| const int render_audiobuffer_sample_rate_hz = |
| formats_.api_format.reverse_output_stream().num_frames() == 0 |
| ? formats_.render_processing_format.sample_rate_hz() |
| : formats_.api_format.reverse_output_stream().sample_rate_hz(); |
| if (formats_.api_format.reverse_input_stream().num_channels() > 0) { |
| render_.render_audio.reset(new AudioBuffer( |
| formats_.api_format.reverse_input_stream().sample_rate_hz(), |
| formats_.api_format.reverse_input_stream().num_channels(), |
| formats_.render_processing_format.sample_rate_hz(), |
| formats_.render_processing_format.num_channels(), |
| render_audiobuffer_sample_rate_hz, |
| formats_.render_processing_format.num_channels())); |
| if (formats_.api_format.reverse_input_stream() != |
| formats_.api_format.reverse_output_stream()) { |
| render_.render_converter = AudioConverter::Create( |
| formats_.api_format.reverse_input_stream().num_channels(), |
| formats_.api_format.reverse_input_stream().num_frames(), |
| formats_.api_format.reverse_output_stream().num_channels(), |
| formats_.api_format.reverse_output_stream().num_frames()); |
| } else { |
| render_.render_converter.reset(nullptr); |
| } |
| } else { |
| render_.render_audio.reset(nullptr); |
| render_.render_converter.reset(nullptr); |
| } |
| |
| capture_.capture_audio.reset(new AudioBuffer( |
| formats_.api_format.input_stream().sample_rate_hz(), |
| formats_.api_format.input_stream().num_channels(), |
| capture_nonlocked_.capture_processing_format.sample_rate_hz(), |
| formats_.api_format.output_stream().num_channels(), |
| formats_.api_format.output_stream().sample_rate_hz(), |
| formats_.api_format.output_stream().num_channels())); |
| |
| if (capture_nonlocked_.capture_processing_format.sample_rate_hz() < |
| formats_.api_format.output_stream().sample_rate_hz() && |
| formats_.api_format.output_stream().sample_rate_hz() == 48000) { |
| capture_.capture_fullband_audio.reset( |
| new AudioBuffer(formats_.api_format.input_stream().sample_rate_hz(), |
| formats_.api_format.input_stream().num_channels(), |
| formats_.api_format.output_stream().sample_rate_hz(), |
| formats_.api_format.output_stream().num_channels(), |
| formats_.api_format.output_stream().sample_rate_hz(), |
| formats_.api_format.output_stream().num_channels())); |
| } else { |
| capture_.capture_fullband_audio.reset(); |
| } |
| |
| AllocateRenderQueue(); |
| |
| InitializeGainController1(); |
| InitializeTransientSuppressor(); |
| InitializeHighPassFilter(true); |
| InitializeVoiceDetector(); |
| InitializeResidualEchoDetector(); |
| InitializeEchoController(); |
| InitializeGainController2(); |
| InitializeNoiseSuppressor(); |
| InitializeAnalyzer(); |
| InitializePostProcessor(); |
| InitializePreProcessor(); |
| |
| if (aec_dump_) { |
| aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis()); |
| } |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) { |
| UpdateActiveSubmoduleStates(); |
| |
| for (const auto& stream : config.streams) { |
| if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) { |
| return kBadSampleRateError; |
| } |
| } |
| |
| const size_t num_in_channels = config.input_stream().num_channels(); |
| const size_t num_out_channels = config.output_stream().num_channels(); |
| |
| // Need at least one input channel. |
| // Need either one output channel or as many outputs as there are inputs. |
| if (num_in_channels == 0 || |
| !(num_out_channels == 1 || num_out_channels == num_in_channels)) { |
| return kBadNumberChannelsError; |
| } |
| |
| formats_.api_format = config; |
| |
| // Choose maximum rate to use for the split filtering. |
| RTC_DCHECK(config_.pipeline.maximum_internal_processing_rate == 48000 || |
| config_.pipeline.maximum_internal_processing_rate == 32000); |
| int max_splitting_rate = 48000; |
| if (config_.pipeline.maximum_internal_processing_rate == 32000) { |
| max_splitting_rate = config_.pipeline.maximum_internal_processing_rate; |
| } |
| |
| int capture_processing_rate = SuitableProcessRate( |
| std::min(formats_.api_format.input_stream().sample_rate_hz(), |
| formats_.api_format.output_stream().sample_rate_hz()), |
| max_splitting_rate, |
| submodule_states_.CaptureMultiBandSubModulesActive() || |
| submodule_states_.RenderMultiBandSubModulesActive()); |
| RTC_DCHECK_NE(8000, capture_processing_rate); |
| |
| capture_nonlocked_.capture_processing_format = |
| StreamConfig(capture_processing_rate); |
| |
| int render_processing_rate; |
| if (!capture_nonlocked_.echo_controller_enabled) { |
| render_processing_rate = SuitableProcessRate( |
| std::min(formats_.api_format.reverse_input_stream().sample_rate_hz(), |
| formats_.api_format.reverse_output_stream().sample_rate_hz()), |
| max_splitting_rate, |
| submodule_states_.CaptureMultiBandSubModulesActive() || |
| submodule_states_.RenderMultiBandSubModulesActive()); |
| } else { |
| render_processing_rate = capture_processing_rate; |
| } |
| |
| // If the forward sample rate is 8 kHz, the render stream is also processed |
| // at this rate. |
| if (capture_nonlocked_.capture_processing_format.sample_rate_hz() == |
| kSampleRate8kHz) { |
| render_processing_rate = kSampleRate8kHz; |
| } else { |
| render_processing_rate = |
| std::max(render_processing_rate, static_cast<int>(kSampleRate16kHz)); |
| } |
| |
| RTC_DCHECK_NE(8000, render_processing_rate); |
| |
| if (submodule_states_.RenderMultiBandSubModulesActive()) { |
| // By default, downmix the render stream to mono for analysis. This has been |
| // demonstrated to work well for AEC in most practical scenarios. |
| const bool multi_channel_render = config_.pipeline.multi_channel_render && |
| constants_.multi_channel_render_support; |
| int render_processing_num_channels = |
| multi_channel_render |
| ? formats_.api_format.reverse_input_stream().num_channels() |
| : 1; |
| formats_.render_processing_format = |
| StreamConfig(render_processing_rate, render_processing_num_channels); |
| } else { |
| formats_.render_processing_format = StreamConfig( |
| formats_.api_format.reverse_input_stream().sample_rate_hz(), |
| formats_.api_format.reverse_input_stream().num_channels()); |
| } |
| |
| if (capture_nonlocked_.capture_processing_format.sample_rate_hz() == |
| kSampleRate32kHz || |
| capture_nonlocked_.capture_processing_format.sample_rate_hz() == |
| kSampleRate48kHz) { |
| capture_nonlocked_.split_rate = kSampleRate16kHz; |
| } else { |
| capture_nonlocked_.split_rate = |
| capture_nonlocked_.capture_processing_format.sample_rate_hz(); |
| } |
| |
| return InitializeLocked(); |
| } |
| |
| void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) { |
| RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString(); |
| |
| // Run in a single-threaded manner when applying the settings. |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| |
| const bool pipeline_config_changed = |
| config_.pipeline.multi_channel_render != |
| config.pipeline.multi_channel_render || |
| config_.pipeline.multi_channel_capture != |
| config.pipeline.multi_channel_capture || |
| config_.pipeline.maximum_internal_processing_rate != |
| config.pipeline.maximum_internal_processing_rate; |
| |
| const bool aec_config_changed = |
| config_.echo_canceller.enabled != config.echo_canceller.enabled || |
| config_.echo_canceller.mobile_mode != config.echo_canceller.mobile_mode; |
| |
| const bool agc1_config_changed = |
| config_.gain_controller1.enabled != config.gain_controller1.enabled || |
| config_.gain_controller1.mode != config.gain_controller1.mode || |
| config_.gain_controller1.target_level_dbfs != |
| config.gain_controller1.target_level_dbfs || |
| config_.gain_controller1.compression_gain_db != |
| config.gain_controller1.compression_gain_db || |
| config_.gain_controller1.enable_limiter != |
| config.gain_controller1.enable_limiter || |
| config_.gain_controller1.analog_level_minimum != |
| config.gain_controller1.analog_level_minimum || |
| config_.gain_controller1.analog_level_maximum != |
| config.gain_controller1.analog_level_maximum || |
| config_.gain_controller1.analog_gain_controller.enabled != |
| config.gain_controller1.analog_gain_controller.enabled || |
| config_.gain_controller1.analog_gain_controller.startup_min_volume != |
| config.gain_controller1.analog_gain_controller.startup_min_volume || |
| config_.gain_controller1.analog_gain_controller.clipped_level_min != |
| config.gain_controller1.analog_gain_controller.clipped_level_min || |
| config_.gain_controller1.analog_gain_controller |
| .enable_agc2_level_estimator != |
| config.gain_controller1.analog_gain_controller |
| .enable_agc2_level_estimator || |
| config_.gain_controller1.analog_gain_controller.enable_digital_adaptive != |
| config.gain_controller1.analog_gain_controller |
| .enable_digital_adaptive; |
| |
| const bool agc2_config_changed = |
| config_.gain_controller2.enabled != config.gain_controller2.enabled; |
| |
| const bool voice_detection_config_changed = |
| config_.voice_detection.enabled != config.voice_detection.enabled; |
| |
| const bool ns_config_changed = |
| config_.noise_suppression.enabled != config.noise_suppression.enabled || |
| config_.noise_suppression.level != config.noise_suppression.level; |
| |
| const bool ts_config_changed = config_.transient_suppression.enabled != |
| config.transient_suppression.enabled; |
| |
| const bool pre_amplifier_config_changed = |
| config_.pre_amplifier.enabled != config.pre_amplifier.enabled || |
| config_.pre_amplifier.fixed_gain_factor != |
| config.pre_amplifier.fixed_gain_factor; |
| |
| config_ = config; |
| |
| if (aec_config_changed) { |
| InitializeEchoController(); |
| } |
| |
| if (ns_config_changed) { |
| InitializeNoiseSuppressor(); |
| } |
| |
| if (ts_config_changed) { |
| InitializeTransientSuppressor(); |
| } |
| |
| InitializeHighPassFilter(false); |
| |
| if (agc1_config_changed) { |
| InitializeGainController1(); |
| } |
| |
| const bool config_ok = GainController2::Validate(config_.gain_controller2); |
| if (!config_ok) { |
| RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n" |
| "Gain Controller 2: " |
| << GainController2::ToString(config_.gain_controller2) |
| << "\nReverting to default parameter set"; |
| config_.gain_controller2 = AudioProcessing::Config::GainController2(); |
| } |
| |
| if (agc2_config_changed) { |
| InitializeGainController2(); |
| } |
| |
| if (pre_amplifier_config_changed) { |
| InitializePreAmplifier(); |
| } |
| |
| if (config_.level_estimation.enabled && !submodules_.output_level_estimator) { |
| submodules_.output_level_estimator = std::make_unique<LevelEstimator>(); |
| } |
| |
| if (voice_detection_config_changed) { |
| InitializeVoiceDetector(); |
| } |
| |
| // Reinitialization must happen after all submodule configuration to avoid |
| // additional reinitializations on the next capture / render processing call. |
| if (pipeline_config_changed) { |
| InitializeLocked(formats_.api_format); |
| } |
| } |
| |
| // TODO(webrtc:5298): Remove. |
| void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {} |
| |
| int AudioProcessingImpl::proc_sample_rate_hz() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return capture_nonlocked_.capture_processing_format.sample_rate_hz(); |
| } |
| |
| int AudioProcessingImpl::proc_fullband_sample_rate_hz() const { |
| return capture_.capture_fullband_audio |
| ? capture_.capture_fullband_audio->num_frames() * 100 |
| : capture_nonlocked_.capture_processing_format.sample_rate_hz(); |
| } |
| |
| int AudioProcessingImpl::proc_split_sample_rate_hz() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return capture_nonlocked_.split_rate; |
| } |
| |
| size_t AudioProcessingImpl::num_reverse_channels() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return formats_.render_processing_format.num_channels(); |
| } |
| |
| size_t AudioProcessingImpl::num_input_channels() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return formats_.api_format.input_stream().num_channels(); |
| } |
| |
| size_t AudioProcessingImpl::num_proc_channels() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| const bool multi_channel_capture = config_.pipeline.multi_channel_capture && |
| constants_.multi_channel_capture_support; |
| if (capture_nonlocked_.echo_controller_enabled && !multi_channel_capture) { |
| return 1; |
| } |
| return num_output_channels(); |
| } |
| |
| size_t AudioProcessingImpl::num_output_channels() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return formats_.api_format.output_stream().num_channels(); |
| } |
| |
| void AudioProcessingImpl::set_output_will_be_muted(bool muted) { |
| rtc::CritScope cs(&crit_capture_); |
| capture_.output_will_be_muted = muted; |
| if (submodules_.agc_manager.get()) { |
| submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted); |
| } |
| } |
| |
| void AudioProcessingImpl::SetRuntimeSetting(RuntimeSetting setting) { |
| switch (setting.type()) { |
| case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting: |
| case RuntimeSetting::Type::kPlayoutAudioDeviceChange: |
| render_runtime_settings_enqueuer_.Enqueue(setting); |
| return; |
| case RuntimeSetting::Type::kCapturePreGain: |
| case RuntimeSetting::Type::kCaptureCompressionGain: |
| case RuntimeSetting::Type::kCaptureFixedPostGain: |
| capture_runtime_settings_enqueuer_.Enqueue(setting); |
| return; |
| case RuntimeSetting::Type::kPlayoutVolumeChange: |
| capture_runtime_settings_enqueuer_.Enqueue(setting); |
| render_runtime_settings_enqueuer_.Enqueue(setting); |
| return; |
| case RuntimeSetting::Type::kNotSpecified: |
| RTC_NOTREACHED(); |
| return; |
| } |
| // The language allows the enum to have a non-enumerator |
| // value. Check that this doesn't happen. |
| RTC_NOTREACHED(); |
| } |
| |
| AudioProcessingImpl::RuntimeSettingEnqueuer::RuntimeSettingEnqueuer( |
| SwapQueue<RuntimeSetting>* runtime_settings) |
| : runtime_settings_(*runtime_settings) { |
| RTC_DCHECK(runtime_settings); |
| } |
| |
| AudioProcessingImpl::RuntimeSettingEnqueuer::~RuntimeSettingEnqueuer() = |
| default; |
| |
| void AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue( |
| RuntimeSetting setting) { |
| size_t remaining_attempts = 10; |
| while (!runtime_settings_.Insert(&setting) && remaining_attempts-- > 0) { |
| RuntimeSetting setting_to_discard; |
| if (runtime_settings_.Remove(&setting_to_discard)) |
| RTC_LOG(LS_ERROR) |
| << "The runtime settings queue is full. Oldest setting discarded."; |
| } |
| if (remaining_attempts == 0) |
| RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting."; |
| } |
| |
| int AudioProcessingImpl::MaybeInitializeCapture( |
| const StreamConfig& input_config, |
| const StreamConfig& output_config) { |
| ProcessingConfig processing_config; |
| bool reinitialization_required = false; |
| { |
| // Acquire the capture lock in order to access api_format. The lock is |
| // released immediately, as we may need to acquire the render lock as part |
| // of the conditional reinitialization. |
| rtc::CritScope cs_capture(&crit_capture_); |
| processing_config = formats_.api_format; |
| reinitialization_required = UpdateActiveSubmoduleStates(); |
| } |
| |
| if (processing_config.input_stream() != input_config) { |
| processing_config.input_stream() = input_config; |
| reinitialization_required = true; |
| } |
| |
| if (processing_config.output_stream() != output_config) { |
| processing_config.output_stream() = output_config; |
| reinitialization_required = true; |
| } |
| |
| if (reinitialization_required) { |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| RETURN_ON_ERR(InitializeLocked(processing_config)); |
| } |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::ProcessStream(const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| float* const* dest) { |
| TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig"); |
| if (!src || !dest) { |
| return kNullPointerError; |
| } |
| |
| RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config)); |
| |
| rtc::CritScope cs_capture(&crit_capture_); |
| |
| if (aec_dump_) { |
| RecordUnprocessedCaptureStream(src); |
| } |
| |
| capture_.keyboard_info.Extract(src, formats_.api_format.input_stream()); |
| capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream()); |
| if (capture_.capture_fullband_audio) { |
| capture_.capture_fullband_audio->CopyFrom( |
| src, formats_.api_format.input_stream()); |
| } |
| RETURN_ON_ERR(ProcessCaptureStreamLocked()); |
| if (capture_.capture_fullband_audio) { |
| capture_.capture_fullband_audio->CopyTo(formats_.api_format.output_stream(), |
| dest); |
| } else { |
| capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest); |
| } |
| |
| if (aec_dump_) { |
| RecordProcessedCaptureStream(dest); |
| } |
| return kNoError; |
| } |
| |
| void AudioProcessingImpl::HandleCaptureRuntimeSettings() { |
| RuntimeSetting setting; |
| while (capture_runtime_settings_.Remove(&setting)) { |
| if (aec_dump_) { |
| aec_dump_->WriteRuntimeSetting(setting); |
| } |
| switch (setting.type()) { |
| case RuntimeSetting::Type::kCapturePreGain: |
| if (config_.pre_amplifier.enabled) { |
| float value; |
| setting.GetFloat(&value); |
| config_.pre_amplifier.fixed_gain_factor = value; |
| submodules_.pre_amplifier->SetGainFactor(value); |
| } |
| // TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump. |
| break; |
| case RuntimeSetting::Type::kCaptureCompressionGain: { |
| if (!submodules_.agc_manager) { |
| float value; |
| setting.GetFloat(&value); |
| int int_value = static_cast<int>(value + .5f); |
| config_.gain_controller1.compression_gain_db = int_value; |
| if (submodules_.gain_control) { |
| int error = |
| submodules_.gain_control->set_compression_gain_db(int_value); |
| RTC_DCHECK_EQ(kNoError, error); |
| } |
| } |
| break; |
| } |
| case RuntimeSetting::Type::kCaptureFixedPostGain: { |
| if (submodules_.gain_controller2) { |
| float value; |
| setting.GetFloat(&value); |
| config_.gain_controller2.fixed_digital.gain_db = value; |
| submodules_.gain_controller2->ApplyConfig(config_.gain_controller2); |
| } |
| break; |
| } |
| case RuntimeSetting::Type::kPlayoutVolumeChange: { |
| int value; |
| setting.GetInt(&value); |
| capture_.playout_volume = value; |
| break; |
| } |
| case RuntimeSetting::Type::kPlayoutAudioDeviceChange: |
| RTC_NOTREACHED(); |
| break; |
| case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting: |
| RTC_NOTREACHED(); |
| break; |
| case RuntimeSetting::Type::kNotSpecified: |
| RTC_NOTREACHED(); |
| break; |
| } |
| } |
| } |
| |
| void AudioProcessingImpl::HandleRenderRuntimeSettings() { |
| RuntimeSetting setting; |
| while (render_runtime_settings_.Remove(&setting)) { |
| if (aec_dump_) { |
| aec_dump_->WriteRuntimeSetting(setting); |
| } |
| switch (setting.type()) { |
| case RuntimeSetting::Type::kPlayoutAudioDeviceChange: // fall-through |
| case RuntimeSetting::Type::kPlayoutVolumeChange: // fall-through |
| case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting: |
| if (submodules_.render_pre_processor) { |
| submodules_.render_pre_processor->SetRuntimeSetting(setting); |
| } |
| break; |
| case RuntimeSetting::Type::kCapturePreGain: // fall-through |
| case RuntimeSetting::Type::kCaptureCompressionGain: // fall-through |
| case RuntimeSetting::Type::kCaptureFixedPostGain: // fall-through |
| case RuntimeSetting::Type::kNotSpecified: |
| RTC_NOTREACHED(); |
| break; |
| } |
| } |
| } |
| |
| void AudioProcessingImpl::QueueBandedRenderAudio(AudioBuffer* audio) { |
| RTC_DCHECK_GE(160, audio->num_frames_per_band()); |
| |
| if (submodules_.echo_control_mobile) { |
| EchoControlMobileImpl::PackRenderAudioBuffer(audio, num_output_channels(), |
| num_reverse_channels(), |
| &aecm_render_queue_buffer_); |
| RTC_DCHECK(aecm_render_signal_queue_); |
| // Insert the samples into the queue. |
| if (!aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_)) { |
| // The data queue is full and needs to be emptied. |
| EmptyQueuedRenderAudio(); |
| |
| // Retry the insert (should always work). |
| bool result = |
| aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_); |
| RTC_DCHECK(result); |
| } |
| } |
| |
| if (!submodules_.agc_manager && submodules_.gain_control) { |
| GainControlImpl::PackRenderAudioBuffer(*audio, &agc_render_queue_buffer_); |
| // Insert the samples into the queue. |
| if (!agc_render_signal_queue_->Insert(&agc_render_queue_buffer_)) { |
| // The data queue is full and needs to be emptied. |
| EmptyQueuedRenderAudio(); |
| |
| // Retry the insert (should always work). |
| bool result = agc_render_signal_queue_->Insert(&agc_render_queue_buffer_); |
| RTC_DCHECK(result); |
| } |
| } |
| } |
| |
| void AudioProcessingImpl::QueueNonbandedRenderAudio(AudioBuffer* audio) { |
| ResidualEchoDetector::PackRenderAudioBuffer(audio, &red_render_queue_buffer_); |
| |
| // Insert the samples into the queue. |
| if (!red_render_signal_queue_->Insert(&red_render_queue_buffer_)) { |
| // The data queue is full and needs to be emptied. |
| EmptyQueuedRenderAudio(); |
| |
| // Retry the insert (should always work). |
| bool result = red_render_signal_queue_->Insert(&red_render_queue_buffer_); |
| RTC_DCHECK(result); |
| } |
| } |
| |
| void AudioProcessingImpl::AllocateRenderQueue() { |
| const size_t new_agc_render_queue_element_max_size = |
| std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerBand); |
| |
| const size_t new_red_render_queue_element_max_size = |
| std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerFrame); |
| |
| // Reallocate the queues if the queue item sizes are too small to fit the |
| // data to put in the queues. |
| |
| if (agc_render_queue_element_max_size_ < |
| new_agc_render_queue_element_max_size) { |
| agc_render_queue_element_max_size_ = new_agc_render_queue_element_max_size; |
| |
| std::vector<int16_t> template_queue_element( |
| agc_render_queue_element_max_size_); |
| |
| agc_render_signal_queue_.reset( |
| new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>( |
| kMaxNumFramesToBuffer, template_queue_element, |
| RenderQueueItemVerifier<int16_t>( |
| agc_render_queue_element_max_size_))); |
| |
| agc_render_queue_buffer_.resize(agc_render_queue_element_max_size_); |
| agc_capture_queue_buffer_.resize(agc_render_queue_element_max_size_); |
| } else { |
| agc_render_signal_queue_->Clear(); |
| } |
| |
| if (red_render_queue_element_max_size_ < |
| new_red_render_queue_element_max_size) { |
| red_render_queue_element_max_size_ = new_red_render_queue_element_max_size; |
| |
| std::vector<float> template_queue_element( |
| red_render_queue_element_max_size_); |
| |
| red_render_signal_queue_.reset( |
| new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>( |
| kMaxNumFramesToBuffer, template_queue_element, |
| RenderQueueItemVerifier<float>( |
| red_render_queue_element_max_size_))); |
| |
| red_render_queue_buffer_.resize(red_render_queue_element_max_size_); |
| red_capture_queue_buffer_.resize(red_render_queue_element_max_size_); |
| } else { |
| red_render_signal_queue_->Clear(); |
| } |
| } |
| |
| void AudioProcessingImpl::EmptyQueuedRenderAudio() { |
| rtc::CritScope cs_capture(&crit_capture_); |
| if (submodules_.echo_control_mobile) { |
| RTC_DCHECK(aecm_render_signal_queue_); |
| while (aecm_render_signal_queue_->Remove(&aecm_capture_queue_buffer_)) { |
| submodules_.echo_control_mobile->ProcessRenderAudio( |
| aecm_capture_queue_buffer_); |
| } |
| } |
| |
| if (submodules_.gain_control) { |
| while (agc_render_signal_queue_->Remove(&agc_capture_queue_buffer_)) { |
| submodules_.gain_control->ProcessRenderAudio(agc_capture_queue_buffer_); |
| } |
| } |
| |
| while (red_render_signal_queue_->Remove(&red_capture_queue_buffer_)) { |
| RTC_DCHECK(submodules_.echo_detector); |
| submodules_.echo_detector->AnalyzeRenderAudio(red_capture_queue_buffer_); |
| } |
| } |
| |
| int AudioProcessingImpl::ProcessStream(const int16_t* const src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| int16_t* const dest) { |
| TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame"); |
| RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config)); |
| |
| rtc::CritScope cs_capture(&crit_capture_); |
| |
| if (aec_dump_) { |
| RecordUnprocessedCaptureStream(src, input_config); |
| } |
| |
| capture_.capture_audio->CopyFrom(src, input_config); |
| if (capture_.capture_fullband_audio) { |
| capture_.capture_fullband_audio->CopyFrom(src, input_config); |
| } |
| RETURN_ON_ERR(ProcessCaptureStreamLocked()); |
| if (submodule_states_.CaptureMultiBandProcessingPresent() || |
| submodule_states_.CaptureFullBandProcessingActive()) { |
| if (capture_.capture_fullband_audio) { |
| capture_.capture_fullband_audio->CopyTo(output_config, dest); |
| } else { |
| capture_.capture_audio->CopyTo(output_config, dest); |
| } |
| } |
| |
| if (aec_dump_) { |
| RecordProcessedCaptureStream(dest, output_config); |
| } |
| |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::ProcessCaptureStreamLocked() { |
| EmptyQueuedRenderAudio(); |
| HandleCaptureRuntimeSettings(); |
| |
| // Ensure that not both the AEC and AECM are active at the same time. |
| // TODO(peah): Simplify once the public API Enable functions for these |
| // are moved to APM. |
| RTC_DCHECK_LE( |
| !!submodules_.echo_controller + !!submodules_.echo_control_mobile, 1); |
| |
| AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity. |
| AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get(); |
| |
| if (submodules_.high_pass_filter && |
| config_.high_pass_filter.apply_in_full_band && |
| !constants_.enforce_split_band_hpf) { |
| submodules_.high_pass_filter->Process(capture_buffer, |
| /*use_split_band_data=*/false); |
| } |
| |
| if (submodules_.pre_amplifier) { |
| submodules_.pre_amplifier->ApplyGain(AudioFrameView<float>( |
| capture_buffer->channels(), capture_buffer->num_channels(), |
| capture_buffer->num_frames())); |
| } |
| |
| capture_input_rms_.Analyze(rtc::ArrayView<const float>( |
| capture_buffer->channels_const()[0], |
| capture_nonlocked_.capture_processing_format.num_frames())); |
| const bool log_rms = ++capture_rms_interval_counter_ >= 1000; |
| if (log_rms) { |
| capture_rms_interval_counter_ = 0; |
| RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak(); |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms", |
| levels.average, 1, RmsLevel::kMinLevelDb, 64); |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms", |
| levels.peak, 1, RmsLevel::kMinLevelDb, 64); |
| } |
| |
| if (submodules_.echo_controller) { |
| // Detect and flag any change in the analog gain. |
| int analog_mic_level = recommended_stream_analog_level(); |
| capture_.echo_path_gain_change = |
| capture_.prev_analog_mic_level != analog_mic_level && |
| capture_.prev_analog_mic_level != -1; |
| capture_.prev_analog_mic_level = analog_mic_level; |
| |
| // Detect and flag any change in the pre-amplifier gain. |
| if (submodules_.pre_amplifier) { |
| float pre_amp_gain = submodules_.pre_amplifier->GetGainFactor(); |
| capture_.echo_path_gain_change = |
| capture_.echo_path_gain_change || |
| (capture_.prev_pre_amp_gain != pre_amp_gain && |
| capture_.prev_pre_amp_gain >= 0.f); |
| capture_.prev_pre_amp_gain = pre_amp_gain; |
| } |
| |
| // Detect volume change. |
| capture_.echo_path_gain_change = |
| capture_.echo_path_gain_change || |
| (capture_.prev_playout_volume != capture_.playout_volume && |
| capture_.prev_playout_volume >= 0); |
| capture_.prev_playout_volume = capture_.playout_volume; |
| |
| submodules_.echo_controller->AnalyzeCapture(capture_buffer); |
| } |
| |
| if (submodules_.agc_manager) { |
| submodules_.agc_manager->AnalyzePreProcess(capture_buffer); |
| } |
| |
| if (submodule_states_.CaptureMultiBandSubModulesActive() && |
| SampleRateSupportsMultiBand( |
| capture_nonlocked_.capture_processing_format.sample_rate_hz())) { |
| capture_buffer->SplitIntoFrequencyBands(); |
| } |
| |
| const bool multi_channel_capture = config_.pipeline.multi_channel_capture && |
| constants_.multi_channel_capture_support; |
| if (submodules_.echo_controller && !multi_channel_capture) { |
| // Force down-mixing of the number of channels after the detection of |
| // capture signal saturation. |
| // TODO(peah): Look into ensuring that this kind of tampering with the |
| // AudioBuffer functionality should not be needed. |
| capture_buffer->set_num_channels(1); |
| } |
| |
| if (submodules_.high_pass_filter && |
| (!config_.high_pass_filter.apply_in_full_band || |
| constants_.enforce_split_band_hpf)) { |
| submodules_.high_pass_filter->Process(capture_buffer, |
| /*use_split_band_data=*/true); |
| } |
| |
| if (submodules_.gain_control) { |
| RETURN_ON_ERR( |
| submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer)); |
| } |
| |
| if ((!config_.noise_suppression.analyze_linear_aec_output_when_available || |
| !linear_aec_buffer || submodules_.echo_control_mobile) && |
| submodules_.noise_suppressor) { |
| submodules_.noise_suppressor->Analyze(*capture_buffer); |
| } |
| |
| if (submodules_.echo_control_mobile) { |
| // Ensure that the stream delay was set before the call to the |
| // AECM ProcessCaptureAudio function. |
| if (!capture_.was_stream_delay_set) { |
| return AudioProcessing::kStreamParameterNotSetError; |
| } |
| |
| if (submodules_.noise_suppressor) { |
| submodules_.noise_suppressor->Process(capture_buffer); |
| } |
| |
| RETURN_ON_ERR(submodules_.echo_control_mobile->ProcessCaptureAudio( |
| capture_buffer, stream_delay_ms())); |
| } else { |
| if (submodules_.echo_controller) { |
| data_dumper_->DumpRaw("stream_delay", stream_delay_ms()); |
| |
| if (capture_.was_stream_delay_set) { |
| submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms()); |
| } |
| |
| submodules_.echo_controller->ProcessCapture( |
| capture_buffer, linear_aec_buffer, capture_.echo_path_gain_change); |
| } |
| |
| if (config_.noise_suppression.analyze_linear_aec_output_when_available && |
| linear_aec_buffer && submodules_.noise_suppressor) { |
| submodules_.noise_suppressor->Analyze(*linear_aec_buffer); |
| } |
| |
| if (submodules_.noise_suppressor) { |
| submodules_.noise_suppressor->Process(capture_buffer); |
| } |
| } |
| |
| if (config_.voice_detection.enabled) { |
| capture_.stats.voice_detected = |
| submodules_.voice_detector->ProcessCaptureAudio(capture_buffer); |
| } else { |
| capture_.stats.voice_detected = absl::nullopt; |
| } |
| |
| if (submodules_.agc_manager) { |
| submodules_.agc_manager->Process(capture_buffer); |
| |
| absl::optional<int> new_digital_gain = |
| submodules_.agc_manager->GetDigitalComressionGain(); |
| if (new_digital_gain && submodules_.gain_control) { |
| submodules_.gain_control->set_compression_gain_db(*new_digital_gain); |
| } |
| } |
| |
| if (submodules_.gain_control) { |
| // TODO(peah): Add reporting from AEC3 whether there is echo. |
| RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio( |
| capture_buffer, /*stream_has_echo*/ false)); |
| } |
| |
| if (submodule_states_.CaptureMultiBandProcessingPresent() && |
| SampleRateSupportsMultiBand( |
| capture_nonlocked_.capture_processing_format.sample_rate_hz())) { |
| capture_buffer->MergeFrequencyBands(); |
| } |
| |
| if (capture_.capture_fullband_audio) { |
| const auto& ec = submodules_.echo_controller; |
| bool ec_active = ec ? ec->ActiveProcessing() : false; |
| // Only update the fullband buffer if the multiband processing has changed |
| // the signal. Keep the original signal otherwise. |
| if (submodule_states_.CaptureMultiBandProcessingActive(ec_active)) { |
| capture_buffer->CopyTo(capture_.capture_fullband_audio.get()); |
| } |
| capture_buffer = capture_.capture_fullband_audio.get(); |
| } |
| |
| if (config_.residual_echo_detector.enabled) { |
| RTC_DCHECK(submodules_.echo_detector); |
| submodules_.echo_detector->AnalyzeCaptureAudio(rtc::ArrayView<const float>( |
| capture_buffer->channels()[0], capture_buffer->num_frames())); |
| } |
| |
| // TODO(aluebs): Investigate if the transient suppression placement should be |
| // before or after the AGC. |
| if (submodules_.transient_suppressor) { |
| float voice_probability = submodules_.agc_manager.get() |
| ? submodules_.agc_manager->voice_probability() |
| : 1.f; |
| |
| submodules_.transient_suppressor->Suppress( |
| capture_buffer->channels()[0], capture_buffer->num_frames(), |
| capture_buffer->num_channels(), |
| capture_buffer->split_bands_const(0)[kBand0To8kHz], |
| capture_buffer->num_frames_per_band(), |
| capture_.keyboard_info.keyboard_data, |
| capture_.keyboard_info.num_keyboard_frames, voice_probability, |
| capture_.key_pressed); |
| } |
| |
| // Experimental APM sub-module that analyzes |capture_buffer|. |
| if (submodules_.capture_analyzer) { |
| submodules_.capture_analyzer->Analyze(capture_buffer); |
| } |
| |
| if (submodules_.gain_controller2) { |
| submodules_.gain_controller2->NotifyAnalogLevel( |
| recommended_stream_analog_level()); |
| submodules_.gain_controller2->Process(capture_buffer); |
| } |
| |
| if (submodules_.capture_post_processor) { |
| submodules_.capture_post_processor->Process(capture_buffer); |
| } |
| |
| // The level estimator operates on the recombined data. |
| if (config_.level_estimation.enabled) { |
| submodules_.output_level_estimator->ProcessStream(*capture_buffer); |
| capture_.stats.output_rms_dbfs = submodules_.output_level_estimator->RMS(); |
| } else { |
| capture_.stats.output_rms_dbfs = absl::nullopt; |
| } |
| |
| capture_output_rms_.Analyze(rtc::ArrayView<const float>( |
| capture_buffer->channels_const()[0], |
| capture_nonlocked_.capture_processing_format.num_frames())); |
| if (log_rms) { |
| RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak(); |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelAverageRms", |
| levels.average, 1, RmsLevel::kMinLevelDb, 64); |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelPeakRms", |
| levels.peak, 1, RmsLevel::kMinLevelDb, 64); |
| } |
| |
| if (submodules_.agc_manager) { |
| int level = recommended_stream_analog_level(); |
| data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1, |
| &level); |
| } |
| |
| // Compute echo-related stats. |
| if (submodules_.echo_controller) { |
| auto ec_metrics = submodules_.echo_controller->GetMetrics(); |
| capture_.stats.echo_return_loss = ec_metrics.echo_return_loss; |
| capture_.stats.echo_return_loss_enhancement = |
| ec_metrics.echo_return_loss_enhancement; |
| capture_.stats.delay_ms = ec_metrics.delay_ms; |
| } |
| if (config_.residual_echo_detector.enabled) { |
| RTC_DCHECK(submodules_.echo_detector); |
| auto ed_metrics = submodules_.echo_detector->GetMetrics(); |
| capture_.stats.residual_echo_likelihood = ed_metrics.echo_likelihood; |
| capture_.stats.residual_echo_likelihood_recent_max = |
| ed_metrics.echo_likelihood_recent_max; |
| } |
| |
| // Pass stats for reporting. |
| stats_reporter_.UpdateStatistics(capture_.stats); |
| |
| capture_.was_stream_delay_set = false; |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::AnalyzeReverseStream( |
| const float* const* data, |
| const StreamConfig& reverse_config) { |
| TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_StreamConfig"); |
| rtc::CritScope cs(&crit_render_); |
| return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config); |
| } |
| |
| int AudioProcessingImpl::ProcessReverseStream(const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| float* const* dest) { |
| TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig"); |
| rtc::CritScope cs(&crit_render_); |
| RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, input_config, output_config)); |
| if (submodule_states_.RenderMultiBandProcessingActive() || |
| submodule_states_.RenderFullBandProcessingActive()) { |
| render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(), |
| dest); |
| } else if (formats_.api_format.reverse_input_stream() != |
| formats_.api_format.reverse_output_stream()) { |
| render_.render_converter->Convert(src, input_config.num_samples(), dest, |
| output_config.num_samples()); |
| } else { |
| CopyAudioIfNeeded(src, input_config.num_frames(), |
| input_config.num_channels(), dest); |
| } |
| |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::AnalyzeReverseStreamLocked( |
| const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config) { |
| if (src == nullptr) { |
| return kNullPointerError; |
| } |
| |
| if (input_config.num_channels() == 0) { |
| return kBadNumberChannelsError; |
| } |
| |
| ProcessingConfig processing_config = formats_.api_format; |
| processing_config.reverse_input_stream() = input_config; |
| processing_config.reverse_output_stream() = output_config; |
| |
| RETURN_ON_ERR(MaybeInitializeRender(processing_config)); |
| RTC_DCHECK_EQ(input_config.num_frames(), |
| formats_.api_format.reverse_input_stream().num_frames()); |
| |
| if (aec_dump_) { |
| const size_t channel_size = |
| formats_.api_format.reverse_input_stream().num_frames(); |
| const size_t num_channels = |
| formats_.api_format.reverse_input_stream().num_channels(); |
| aec_dump_->WriteRenderStreamMessage( |
| AudioFrameView<const float>(src, num_channels, channel_size)); |
| } |
| render_.render_audio->CopyFrom(src, |
| formats_.api_format.reverse_input_stream()); |
| return ProcessRenderStreamLocked(); |
| } |
| |
| int AudioProcessingImpl::ProcessReverseStream(const int16_t* const src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| int16_t* const dest) { |
| TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame"); |
| |
| if (input_config.num_channels() <= 0) { |
| return AudioProcessing::Error::kBadNumberChannelsError; |
| } |
| |
| rtc::CritScope cs(&crit_render_); |
| ProcessingConfig processing_config = formats_.api_format; |
| processing_config.reverse_input_stream().set_sample_rate_hz( |
| input_config.sample_rate_hz()); |
| processing_config.reverse_input_stream().set_num_channels( |
| input_config.num_channels()); |
| processing_config.reverse_output_stream().set_sample_rate_hz( |
| output_config.sample_rate_hz()); |
| processing_config.reverse_output_stream().set_num_channels( |
| output_config.num_channels()); |
| |
| RETURN_ON_ERR(MaybeInitializeRender(processing_config)); |
| if (input_config.num_frames() != |
| formats_.api_format.reverse_input_stream().num_frames()) { |
| return kBadDataLengthError; |
| } |
| |
| if (aec_dump_) { |
| aec_dump_->WriteRenderStreamMessage(src, input_config.num_frames(), |
| input_config.num_channels()); |
| } |
| |
| render_.render_audio->CopyFrom(src, input_config); |
| RETURN_ON_ERR(ProcessRenderStreamLocked()); |
| if (submodule_states_.RenderMultiBandProcessingActive() || |
| submodule_states_.RenderFullBandProcessingActive()) { |
| render_.render_audio->CopyTo(output_config, dest); |
| } |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::ProcessRenderStreamLocked() { |
| AudioBuffer* render_buffer = render_.render_audio.get(); // For brevity. |
| |
| HandleRenderRuntimeSettings(); |
| |
| if (submodules_.render_pre_processor) { |
| submodules_.render_pre_processor->Process(render_buffer); |
| } |
| |
| QueueNonbandedRenderAudio(render_buffer); |
| |
| if (submodule_states_.RenderMultiBandSubModulesActive() && |
| SampleRateSupportsMultiBand( |
| formats_.render_processing_format.sample_rate_hz())) { |
| render_buffer->SplitIntoFrequencyBands(); |
| } |
| |
| if (submodule_states_.RenderMultiBandSubModulesActive()) { |
| QueueBandedRenderAudio(render_buffer); |
| } |
| |
| // TODO(peah): Perform the queuing inside QueueRenderAudiuo(). |
| if (submodules_.echo_controller) { |
| submodules_.echo_controller->AnalyzeRender(render_buffer); |
| } |
| |
| if (submodule_states_.RenderMultiBandProcessingActive() && |
| SampleRateSupportsMultiBand( |
| formats_.render_processing_format.sample_rate_hz())) { |
| render_buffer->MergeFrequencyBands(); |
| } |
| |
| return kNoError; |
| } |
| |
| int AudioProcessingImpl::set_stream_delay_ms(int delay) { |
| rtc::CritScope cs(&crit_capture_); |
| Error retval = kNoError; |
| capture_.was_stream_delay_set = true; |
| |
| if (delay < 0) { |
| delay = 0; |
| retval = kBadStreamParameterWarning; |
| } |
| |
| // TODO(ajm): the max is rather arbitrarily chosen; investigate. |
| if (delay > 500) { |
| delay = 500; |
| retval = kBadStreamParameterWarning; |
| } |
| |
| capture_nonlocked_.stream_delay_ms = delay; |
| return retval; |
| } |
| |
| bool AudioProcessingImpl::GetLinearAecOutput( |
| rtc::ArrayView<std::array<float, 160>> linear_output) const { |
| rtc::CritScope cs(&crit_capture_); |
| AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get(); |
| |
| RTC_DCHECK(linear_aec_buffer); |
| if (linear_aec_buffer) { |
| RTC_DCHECK_EQ(1, linear_aec_buffer->num_bands()); |
| RTC_DCHECK_EQ(linear_output.size(), linear_aec_buffer->num_channels()); |
| |
| for (size_t ch = 0; ch < linear_aec_buffer->num_channels(); ++ch) { |
| RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames()); |
| rtc::ArrayView<const float> channel_view = |
| rtc::ArrayView<const float>(linear_aec_buffer->channels_const()[ch], |
| linear_aec_buffer->num_frames()); |
| std::copy(channel_view.begin(), channel_view.end(), |
| linear_output[ch].begin()); |
| } |
| return true; |
| } |
| RTC_LOG(LS_ERROR) << "No linear AEC output available"; |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| int AudioProcessingImpl::stream_delay_ms() const { |
| // Used as callback from submodules, hence locking is not allowed. |
| return capture_nonlocked_.stream_delay_ms; |
| } |
| |
| void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) { |
| rtc::CritScope cs(&crit_capture_); |
| capture_.key_pressed = key_pressed; |
| } |
| |
| void AudioProcessingImpl::set_stream_analog_level(int level) { |
| rtc::CritScope cs_capture(&crit_capture_); |
| |
| if (submodules_.agc_manager) { |
| submodules_.agc_manager->set_stream_analog_level(level); |
| data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level", |
| 1, &level); |
| } else if (submodules_.gain_control) { |
| int error = submodules_.gain_control->set_stream_analog_level(level); |
| RTC_DCHECK_EQ(kNoError, error); |
| } else { |
| capture_.cached_stream_analog_level_ = level; |
| } |
| } |
| |
| int AudioProcessingImpl::recommended_stream_analog_level() const { |
| rtc::CritScope cs_capture(&crit_capture_); |
| if (submodules_.agc_manager) { |
| return submodules_.agc_manager->stream_analog_level(); |
| } else if (submodules_.gain_control) { |
| return submodules_.gain_control->stream_analog_level(); |
| } else { |
| return capture_.cached_stream_analog_level_; |
| } |
| } |
| |
| void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) { |
| RTC_DCHECK(aec_dump); |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| |
| // The previously attached AecDump will be destroyed with the |
| // 'aec_dump' parameter, which is after locks are released. |
| aec_dump_.swap(aec_dump); |
| WriteAecDumpConfigMessage(true); |
| aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis()); |
| } |
| |
| void AudioProcessingImpl::DetachAecDump() { |
| // The d-tor of a task-queue based AecDump blocks until all pending |
| // tasks are done. This construction avoids blocking while holding |
| // the render and capture locks. |
| std::unique_ptr<AecDump> aec_dump = nullptr; |
| { |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| aec_dump = std::move(aec_dump_); |
| } |
| } |
| |
| void AudioProcessingImpl::AttachPlayoutAudioGenerator( |
| std::unique_ptr<AudioGenerator> audio_generator) { |
| // TODO(bugs.webrtc.org/8882) Stub. |
| // Reset internal audio generator with audio_generator. |
| } |
| |
| void AudioProcessingImpl::DetachPlayoutAudioGenerator() { |
| // TODO(bugs.webrtc.org/8882) Stub. |
| // Delete audio generator, if one is attached. |
| } |
| |
| void AudioProcessingImpl::MutateConfig( |
| rtc::FunctionView<void(AudioProcessing::Config*)> mutator) { |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| mutator(&config_); |
| ApplyConfig(config_); |
| } |
| |
| AudioProcessing::Config AudioProcessingImpl::GetConfig() const { |
| rtc::CritScope cs_render(&crit_render_); |
| rtc::CritScope cs_capture(&crit_capture_); |
| return config_; |
| } |
| |
| bool AudioProcessingImpl::UpdateActiveSubmoduleStates() { |
| return submodule_states_.Update( |
| config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile, |
| config_.residual_echo_detector.enabled, !!submodules_.noise_suppressor, |
| !!submodules_.gain_control, !!submodules_.gain_controller2, |
| config_.pre_amplifier.enabled, capture_nonlocked_.echo_controller_enabled, |
| config_.voice_detection.enabled, !!submodules_.transient_suppressor); |
| } |
| |
| void AudioProcessingImpl::InitializeTransientSuppressor() { |
| if (config_.transient_suppression.enabled) { |
| // Attempt to create a transient suppressor, if one is not already created. |
| if (!submodules_.transient_suppressor) { |
| submodules_.transient_suppressor = CreateTransientSuppressor(); |
| } |
| if (submodules_.transient_suppressor) { |
| submodules_.transient_suppressor->Initialize( |
| proc_fullband_sample_rate_hz(), capture_nonlocked_.split_rate, |
| num_proc_channels()); |
| } else { |
| RTC_LOG(LS_WARNING) |
| << "No transient suppressor created (probably disabled)"; |
| } |
| } else { |
| submodules_.transient_suppressor.reset(); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializeHighPassFilter(bool forced_reset) { |
| bool high_pass_filter_needed_by_aec = |
| config_.echo_canceller.enabled && |
| config_.echo_canceller.enforce_high_pass_filtering && |
| !config_.echo_canceller.mobile_mode; |
| if (submodule_states_.HighPassFilteringRequired() || |
| high_pass_filter_needed_by_aec) { |
| bool use_full_band = config_.high_pass_filter.apply_in_full_band && |
| !constants_.enforce_split_band_hpf; |
| int rate = use_full_band ? proc_fullband_sample_rate_hz() |
| : proc_split_sample_rate_hz(); |
| size_t num_channels = |
| use_full_band ? num_output_channels() : num_proc_channels(); |
| |
| if (!submodules_.high_pass_filter || |
| rate != submodules_.high_pass_filter->sample_rate_hz() || |
| forced_reset || |
| num_channels != submodules_.high_pass_filter->num_channels()) { |
| submodules_.high_pass_filter.reset( |
| new HighPassFilter(rate, num_channels)); |
| } |
| } else { |
| submodules_.high_pass_filter.reset(); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializeVoiceDetector() { |
| if (config_.voice_detection.enabled) { |
| submodules_.voice_detector = std::make_unique<VoiceDetection>( |
| proc_split_sample_rate_hz(), VoiceDetection::kVeryLowLikelihood); |
| } else { |
| submodules_.voice_detector.reset(); |
| } |
| } |
| void AudioProcessingImpl::InitializeEchoController() { |
| bool use_echo_controller = |
| echo_control_factory_ || |
| (config_.echo_canceller.enabled && !config_.echo_canceller.mobile_mode); |
| |
| if (use_echo_controller) { |
| // Create and activate the echo controller. |
| if (echo_control_factory_) { |
| submodules_.echo_controller = echo_control_factory_->Create( |
| proc_sample_rate_hz(), num_reverse_channels(), num_proc_channels()); |
| RTC_DCHECK(submodules_.echo_controller); |
| } else { |
| EchoCanceller3Config config = |
| use_setup_specific_default_aec3_config_ |
| ? EchoCanceller3::CreateDefaultConfig(num_reverse_channels(), |
| num_proc_channels()) |
| : EchoCanceller3Config(); |
| submodules_.echo_controller = std::make_unique<EchoCanceller3>( |
| config, proc_sample_rate_hz(), num_reverse_channels(), |
| num_proc_channels()); |
| } |
| |
| // Setup the storage for returning the linear AEC output. |
| if (config_.echo_canceller.export_linear_aec_output) { |
| constexpr int kLinearOutputRateHz = 16000; |
| capture_.linear_aec_output = std::make_unique<AudioBuffer>( |
| kLinearOutputRateHz, num_proc_channels(), kLinearOutputRateHz, |
| num_proc_channels(), kLinearOutputRateHz, num_proc_channels()); |
| } else { |
| capture_.linear_aec_output.reset(); |
| } |
| |
| capture_nonlocked_.echo_controller_enabled = true; |
| |
| submodules_.echo_control_mobile.reset(); |
| aecm_render_signal_queue_.reset(); |
| return; |
| } |
| |
| submodules_.echo_controller.reset(); |
| capture_nonlocked_.echo_controller_enabled = false; |
| capture_.linear_aec_output.reset(); |
| |
| if (!config_.echo_canceller.enabled) { |
| submodules_.echo_control_mobile.reset(); |
| aecm_render_signal_queue_.reset(); |
| return; |
| } |
| |
| if (config_.echo_canceller.mobile_mode) { |
| // Create and activate AECM. |
| size_t max_element_size = |
| std::max(static_cast<size_t>(1), |
| kMaxAllowedValuesOfSamplesPerBand * |
| EchoControlMobileImpl::NumCancellersRequired( |
| num_output_channels(), num_reverse_channels())); |
| |
| std::vector<int16_t> template_queue_element(max_element_size); |
| |
| aecm_render_signal_queue_.reset( |
| new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>( |
| kMaxNumFramesToBuffer, template_queue_element, |
| RenderQueueItemVerifier<int16_t>(max_element_size))); |
| |
| aecm_render_queue_buffer_.resize(max_element_size); |
| aecm_capture_queue_buffer_.resize(max_element_size); |
| |
| submodules_.echo_control_mobile.reset(new EchoControlMobileImpl()); |
| |
| submodules_.echo_control_mobile->Initialize(proc_split_sample_rate_hz(), |
| num_reverse_channels(), |
| num_output_channels()); |
| return; |
| } |
| |
| submodules_.echo_control_mobile.reset(); |
| aecm_render_signal_queue_.reset(); |
| } |
| |
| void AudioProcessingImpl::InitializeGainController1() { |
| if (!config_.gain_controller1.enabled) { |
| submodules_.agc_manager.reset(); |
| submodules_.gain_control.reset(); |
| return; |
| } |
| |
| if (!submodules_.gain_control) { |
| submodules_.gain_control.reset(new GainControlImpl()); |
| } |
| |
| submodules_.gain_control->Initialize(num_proc_channels(), |
| proc_sample_rate_hz()); |
| |
| if (!config_.gain_controller1.analog_gain_controller.enabled) { |
| int error = submodules_.gain_control->set_mode( |
| Agc1ConfigModeToInterfaceMode(config_.gain_controller1.mode)); |
| RTC_DCHECK_EQ(kNoError, error); |
| error = submodules_.gain_control->set_target_level_dbfs( |
| config_.gain_controller1.target_level_dbfs); |
| RTC_DCHECK_EQ(kNoError, error); |
| error = submodules_.gain_control->set_compression_gain_db( |
| config_.gain_controller1.compression_gain_db); |
| RTC_DCHECK_EQ(kNoError, error); |
| error = submodules_.gain_control->enable_limiter( |
| config_.gain_controller1.enable_limiter); |
| RTC_DCHECK_EQ(kNoError, error); |
| error = submodules_.gain_control->set_analog_level_limits( |
| config_.gain_controller1.analog_level_minimum, |
| config_.gain_controller1.analog_level_maximum); |
| RTC_DCHECK_EQ(kNoError, error); |
| |
| submodules_.agc_manager.reset(); |
| return; |
| } |
| |
| if (!submodules_.agc_manager.get() || |
| submodules_.agc_manager->num_channels() != |
| static_cast<int>(num_proc_channels()) || |
| submodules_.agc_manager->sample_rate_hz() != |
| capture_nonlocked_.split_rate) { |
| int stream_analog_level = -1; |
| const bool re_creation = !!submodules_.agc_manager; |
| if (re_creation) { |
| stream_analog_level = submodules_.agc_manager->stream_analog_level(); |
| } |
| submodules_.agc_manager.reset(new AgcManagerDirect( |
| num_proc_channels(), |
| config_.gain_controller1.analog_gain_controller.startup_min_volume, |
| config_.gain_controller1.analog_gain_controller.clipped_level_min, |
| config_.gain_controller1.analog_gain_controller |
| .enable_agc2_level_estimator, |
| !config_.gain_controller1.analog_gain_controller |
| .enable_digital_adaptive, |
| capture_nonlocked_.split_rate)); |
| if (re_creation) { |
| submodules_.agc_manager->set_stream_analog_level(stream_analog_level); |
| } |
| } |
| submodules_.agc_manager->Initialize(); |
| submodules_.agc_manager->SetupDigitalGainControl( |
| submodules_.gain_control.get()); |
| submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted); |
| } |
| |
| void AudioProcessingImpl::InitializeGainController2() { |
| if (config_.gain_controller2.enabled) { |
| if (!submodules_.gain_controller2) { |
| // TODO(alessiob): Move the injected gain controller once injection is |
| // implemented. |
| submodules_.gain_controller2.reset(new GainController2()); |
| } |
| |
| submodules_.gain_controller2->Initialize(proc_fullband_sample_rate_hz()); |
| submodules_.gain_controller2->ApplyConfig(config_.gain_controller2); |
| } else { |
| submodules_.gain_controller2.reset(); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializeNoiseSuppressor() { |
| submodules_.noise_suppressor.reset(); |
| |
| if (config_.noise_suppression.enabled) { |
| auto map_level = |
| [](AudioProcessing::Config::NoiseSuppression::Level level) { |
| using NoiseSuppresionConfig = |
| AudioProcessing::Config::NoiseSuppression; |
| switch (level) { |
| case NoiseSuppresionConfig::kLow: |
| return NsConfig::SuppressionLevel::k6dB; |
| case NoiseSuppresionConfig::kModerate: |
| return NsConfig::SuppressionLevel::k12dB; |
| case NoiseSuppresionConfig::kHigh: |
| return NsConfig::SuppressionLevel::k18dB; |
| case NoiseSuppresionConfig::kVeryHigh: |
| return NsConfig::SuppressionLevel::k21dB; |
| default: |
| RTC_NOTREACHED(); |
| } |
| }; |
| |
| NsConfig cfg; |
| cfg.target_level = map_level(config_.noise_suppression.level); |
| submodules_.noise_suppressor = std::make_unique<NoiseSuppressor>( |
| cfg, proc_sample_rate_hz(), num_proc_channels()); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializePreAmplifier() { |
| if (config_.pre_amplifier.enabled) { |
| submodules_.pre_amplifier.reset( |
| new GainApplier(true, config_.pre_amplifier.fixed_gain_factor)); |
| } else { |
| submodules_.pre_amplifier.reset(); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializeResidualEchoDetector() { |
| RTC_DCHECK(submodules_.echo_detector); |
| submodules_.echo_detector->Initialize( |
| proc_fullband_sample_rate_hz(), 1, |
| formats_.render_processing_format.sample_rate_hz(), 1); |
| } |
| |
| void AudioProcessingImpl::InitializeAnalyzer() { |
| if (submodules_.capture_analyzer) { |
| submodules_.capture_analyzer->Initialize(proc_fullband_sample_rate_hz(), |
| num_proc_channels()); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializePostProcessor() { |
| if (submodules_.capture_post_processor) { |
| submodules_.capture_post_processor->Initialize( |
| proc_fullband_sample_rate_hz(), num_proc_channels()); |
| } |
| } |
| |
| void AudioProcessingImpl::InitializePreProcessor() { |
| if (submodules_.render_pre_processor) { |
| submodules_.render_pre_processor->Initialize( |
| formats_.render_processing_format.sample_rate_hz(), |
| formats_.render_processing_format.num_channels()); |
| } |
| } |
| |
| void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {} |
| |
| void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) { |
| if (!aec_dump_) { |
| return; |
| } |
| |
| std::string experiments_description = ""; |
| // TODO(peah): Add semicolon-separated concatenations of experiment |
| // descriptions for other submodules. |
| if (config_.gain_controller1.analog_gain_controller.clipped_level_min != |
| kClippedLevelMin) { |
| experiments_description += "AgcClippingLevelExperiment;"; |
| } |
| if (!!submodules_.capture_post_processor) { |
| experiments_description += "CapturePostProcessor;"; |
| } |
| if (!!submodules_.render_pre_processor) { |
| experiments_description += "RenderPreProcessor;"; |
| } |
| if (capture_nonlocked_.echo_controller_enabled) { |
| experiments_description += "EchoController;"; |
| } |
| if (config_.gain_controller2.enabled) { |
| experiments_description += "GainController2;"; |
| } |
| |
| InternalAPMConfig apm_config; |
| |
| apm_config.aec_enabled = config_.echo_canceller.enabled; |
| apm_config.aec_delay_agnostic_enabled = false; |
| apm_config.aec_extended_filter_enabled = false; |
| apm_config.aec_suppression_level = 0; |
| |
| apm_config.aecm_enabled = !!submodules_.echo_control_mobile; |
| apm_config.aecm_comfort_noise_enabled = |
| submodules_.echo_control_mobile && |
| submodules_.echo_control_mobile->is_comfort_noise_enabled(); |
| apm_config.aecm_routing_mode = |
| submodules_.echo_control_mobile |
| ? static_cast<int>(submodules_.echo_control_mobile->routing_mode()) |
| : 0; |
| |
| apm_config.agc_enabled = !!submodules_.gain_control; |
| |
| apm_config.agc_mode = submodules_.gain_control |
| ? static_cast<int>(submodules_.gain_control->mode()) |
| : GainControl::kAdaptiveAnalog; |
| apm_config.agc_limiter_enabled = |
| submodules_.gain_control ? submodules_.gain_control->is_limiter_enabled() |
| : false; |
| apm_config.noise_robust_agc_enabled = !!submodules_.agc_manager; |
| |
| apm_config.hpf_enabled = config_.high_pass_filter.enabled; |
| |
| apm_config.ns_enabled = config_.noise_suppression.enabled; |
| apm_config.ns_level = static_cast<int>(config_.noise_suppression.level); |
| |
| apm_config.transient_suppression_enabled = |
| config_.transient_suppression.enabled; |
| apm_config.experiments_description = experiments_description; |
| apm_config.pre_amplifier_enabled = config_.pre_amplifier.enabled; |
| apm_config.pre_amplifier_fixed_gain_factor = |
| config_.pre_amplifier.fixed_gain_factor; |
| |
| if (!forced && apm_config == apm_config_for_aec_dump_) { |
| return; |
| } |
| aec_dump_->WriteConfig(apm_config); |
| apm_config_for_aec_dump_ = apm_config; |
| } |
| |
| void AudioProcessingImpl::RecordUnprocessedCaptureStream( |
| const float* const* src) { |
| RTC_DCHECK(aec_dump_); |
| WriteAecDumpConfigMessage(false); |
| |
| const size_t channel_size = formats_.api_format.input_stream().num_frames(); |
| const size_t num_channels = formats_.api_format.input_stream().num_channels(); |
| aec_dump_->AddCaptureStreamInput( |
| AudioFrameView<const float>(src, num_channels, channel_size)); |
| RecordAudioProcessingState(); |
| } |
| |
| void AudioProcessingImpl::RecordUnprocessedCaptureStream( |
| const int16_t* const data, |
| const StreamConfig& config) { |
| RTC_DCHECK(aec_dump_); |
| WriteAecDumpConfigMessage(false); |
| |
| aec_dump_->AddCaptureStreamInput(data, config.num_channels(), |
| config.num_frames()); |
| RecordAudioProcessingState(); |
| } |
| |
| void AudioProcessingImpl::RecordProcessedCaptureStream( |
| const float* const* processed_capture_stream) { |
| RTC_DCHECK(aec_dump_); |
| |
| const size_t channel_size = formats_.api_format.output_stream().num_frames(); |
| const size_t num_channels = |
| formats_.api_format.output_stream().num_channels(); |
| aec_dump_->AddCaptureStreamOutput(AudioFrameView<const float>( |
| processed_capture_stream, num_channels, channel_size)); |
| aec_dump_->WriteCaptureStreamMessage(); |
| } |
| |
| void AudioProcessingImpl::RecordProcessedCaptureStream( |
| const int16_t* const data, |
| const StreamConfig& config) { |
| RTC_DCHECK(aec_dump_); |
| |
| aec_dump_->AddCaptureStreamOutput(data, config.num_channels(), |
| config.num_frames()); |
| aec_dump_->WriteCaptureStreamMessage(); |
| } |
| |
| void AudioProcessingImpl::RecordAudioProcessingState() { |
| RTC_DCHECK(aec_dump_); |
| AecDump::AudioProcessingState audio_proc_state; |
| audio_proc_state.delay = capture_nonlocked_.stream_delay_ms; |
| audio_proc_state.drift = 0; |
| audio_proc_state.level = recommended_stream_analog_level(); |
| audio_proc_state.keypress = capture_.key_pressed; |
| aec_dump_->AddAudioProcessingState(audio_proc_state); |
| } |
| |
| AudioProcessingImpl::ApmCaptureState::ApmCaptureState() |
| : was_stream_delay_set(false), |
| output_will_be_muted(false), |
| key_pressed(false), |
| capture_processing_format(kSampleRate16kHz), |
| split_rate(kSampleRate16kHz), |
| echo_path_gain_change(false), |
| prev_analog_mic_level(-1), |
| prev_pre_amp_gain(-1.f), |
| playout_volume(-1), |
| prev_playout_volume(-1) {} |
| |
| AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; |
| |
| void AudioProcessingImpl::ApmCaptureState::KeyboardInfo::Extract( |
| const float* const* data, |
| const StreamConfig& stream_config) { |
| if (stream_config.has_keyboard()) { |
| keyboard_data = data[stream_config.num_channels()]; |
| } else { |
| keyboard_data = NULL; |
| } |
| num_keyboard_frames = stream_config.num_frames(); |
| } |
| |
| AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; |
| |
| AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; |
| |
| AudioProcessingImpl::ApmStatsReporter::ApmStatsReporter() |
| : stats_message_queue_(1) {} |
| |
| AudioProcessingImpl::ApmStatsReporter::~ApmStatsReporter() = default; |
| |
| AudioProcessingStats AudioProcessingImpl::ApmStatsReporter::GetStatistics() { |
| rtc::CritScope cs_stats(&crit_stats_); |
| bool new_stats_available = stats_message_queue_.Remove(&cached_stats_); |
| // If the message queue is full, return the cached stats. |
| static_cast<void>(new_stats_available); |
| |
| return cached_stats_; |
| } |
| |
| void AudioProcessingImpl::ApmStatsReporter::UpdateStatistics( |
| const AudioProcessingStats& new_stats) { |
| AudioProcessingStats stats_to_queue = new_stats; |
| bool stats_message_passed = stats_message_queue_.Insert(&stats_to_queue); |
| // If the message queue is full, discard the new stats. |
| static_cast<void>(stats_message_passed); |
| } |
| |
| } // namespace webrtc |
