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webrtc / src / bcce4530d43b046d6e9ddf8e4e859dfba89137a2 / . / modules / audio_processing / audio_processing_impl.cc
blob: 3e49c0cb76ed16b1bad5a2ed6a3c9e4cae235c80 [file] [log] [blame]
/*
* 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 "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 "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 AudioProcessing::kNativeSampleRatesHz[];
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 legacy ns functionality should be enforced.
bool DetectLegacyNsEnforcement() {
return field_trial::IsEnabled("WebRTC-NewNoiseSuppressionKillSwitch");
}
// 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;
}
NoiseSuppression::Level NsConfigLevelToInterfaceLevel(
AudioProcessing::Config::NoiseSuppression::Level level) {
using NsConfig = AudioProcessing::Config::NoiseSuppression;
switch (level) {
case NsConfig::kLow:
return NoiseSuppression::Level::kLow;
case NsConfig::kModerate:
return NoiseSuppression::Level::kModerate;
case NsConfig::kHigh:
return NoiseSuppression::Level::kHigh;
case NsConfig::kVeryHigh:
return NoiseSuppression::Level::kVeryHigh;
default:
RTC_NOTREACHED();
}
}
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 echo_canceller_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 |= (echo_canceller_enabled != echo_canceller_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;
echo_canceller_enabled_ = echo_canceller_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_ || echo_canceller_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() || echo_canceller_enabled_ ||
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_ || echo_canceller_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_))),
enforced_usage_of_legacy_ns_(DetectLegacyNsEnforcement()),
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_(config.Get<ExperimentalAgc>().startup_min_volume,
config.Get<ExperimentalAgc>().clipped_level_min,
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
/* enabled= */ false,
/* enabled_agc2_level_estimator= */ false,
/* digital_adaptive_disabled= */ false,
#else
config.Get<ExperimentalAgc>().enabled,
config.Get<ExperimentalAgc>().enabled_agc2_level_estimator,
config.Get<ExperimentalAgc>().digital_adaptive_disabled,
#endif
!field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
!field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch")),
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
capture_(false),
#else
capture_(config.Get<ExperimentalNs>().enabled),
#endif
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_);
submodules_.gain_control.reset(new GainControlImpl());
// If no echo detector is injected, use the ResidualEchoDetector.
if (!submodules_.echo_detector) {
submodules_.echo_detector =
new rtc::RefCountedObject<ResidualEchoDetector>();
}
// TODO(alessiob): Move the injected gain controller once injection is
// implemented.
submodules_.gain_controller2.reset(new GainController2());
SetExtraOptions(config);
}
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();
submodules_.gain_control->Initialize(num_proc_channels(),
proc_sample_rate_hz());
if (constants_.use_experimental_agc) {
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(), constants_.agc_startup_min_volume,
constants_.agc_clipped_level_min,
constants_.use_experimental_agc_agc2_level_estimation,
constants_.use_experimental_agc_agc2_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);
}
InitializeTransient();
InitializeHighPassFilter();
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;
const bool aec_config_changed =
config_.echo_canceller.enabled != config.echo_canceller.enabled ||
config_.echo_canceller.use_legacy_aec !=
config.echo_canceller.use_legacy_aec ||
config_.echo_canceller.mobile_mode != config.echo_canceller.mobile_mode ||
(config_.echo_canceller.enabled && config.echo_canceller.use_legacy_aec &&
config_.echo_canceller.legacy_moderate_suppression_level !=
config.echo_canceller.legacy_moderate_suppression_level);
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;
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;
config_ = config;
if (aec_config_changed) {
InitializeEchoController();
}
if (ns_config_changed) {
InitializeNoiseSuppressor();
}
InitializeHighPassFilter();
if (agc1_config_changed) {
ApplyAgc1Config(config_.gain_controller1);
}
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();
}
InitializeGainController2();
InitializePreAmplifier();
submodules_.gain_controller2->ApplyConfig(config_.gain_controller2);
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);
}
}
void AudioProcessingImpl::ApplyAgc1Config(
const Config::GainController1& config) {
int error = submodules_.gain_control->Enable(config.enabled);
RTC_DCHECK_EQ(kNoError, error);
if (!submodules_.agc_manager) {
error = submodules_.gain_control->set_mode(
Agc1ConfigModeToInterfaceMode(config.mode));
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_target_level_dbfs(
config.target_level_dbfs);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_compression_gain_db(
config.compression_gain_db);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->enable_limiter(config.enable_limiter);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_analog_level_limits(
config.analog_level_minimum, config.analog_level_maximum);
RTC_DCHECK_EQ(kNoError, error);
}
}
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
// Run in a single-threaded manner when setting the extra options.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
capture_nonlocked_.use_aec2_extended_filter =
config.Get<ExtendedFilter>().enabled;
capture_nonlocked_.use_aec2_delay_agnostic =
config.Get<DelayAgnostic>().enabled;
capture_nonlocked_.use_aec2_refined_adaptive_filter =
config.Get<RefinedAdaptiveFilter>().enabled;
if (capture_.transient_suppressor_enabled !=
config.Get<ExperimentalNs>().enabled) {
capture_.transient_suppressor_enabled =
config.Get<ExperimentalNs>().enabled;
InitializeTransient();
}
}
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::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Acquire the capture lock in order to safely call the function
// that retrieves the render side data. This function accesses apm
// getters that need the capture lock held when being called.
rtc::CritScope cs_capture(&crit_capture_);
EmptyQueuedRenderAudio();
if (!src || !dest) {
return kNullPointerError;
}
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) {
// Reinitialize.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
rtc::CritScope cs_capture(&crit_capture_);
RTC_DCHECK_EQ(processing_config.input_stream().num_frames(),
formats_.api_format.input_stream().num_frames());
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;
int error =
submodules_.gain_control->set_compression_gain_db(int_value);
RTC_DCHECK_EQ(kNoError, error);
}
break;
}
case RuntimeSetting::Type::kCaptureFixedPostGain: {
if (config_.gain_controller2.enabled) {
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());
// Insert the samples into the queue.
if (submodules_.echo_cancellation) {
RTC_DCHECK(aec_render_signal_queue_);
EchoCancellationImpl::PackRenderAudioBuffer(audio, num_output_channels(),
num_reverse_channels(),
&aec_render_queue_buffer_);
if (!aec_render_signal_queue_->Insert(&aec_render_queue_buffer_)) {
// The data queue is full and needs to be emptied.
EmptyQueuedRenderAudio();
// Retry the insert (should always work).
bool result = aec_render_signal_queue_->Insert(&aec_render_queue_buffer_);
RTC_DCHECK(result);
}
}
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) {
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_cancellation) {
RTC_DCHECK(aec_render_signal_queue_);
while (aec_render_signal_queue_->Remove(&aec_capture_queue_buffer_)) {
submodules_.echo_cancellation->ProcessRenderAudio(
aec_capture_queue_buffer_);
}
}
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_);
}
}
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(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
{
// Acquire the capture lock in order to safely call the function
// that retrieves the render side data. This function accesses APM
// getters that need the capture lock held when being called.
rtc::CritScope cs_capture(&crit_capture_);
EmptyQueuedRenderAudio();
}
if (!frame) {
return kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != kSampleRate8kHz &&
frame->sample_rate_hz_ != kSampleRate16kHz &&
frame->sample_rate_hz_ != kSampleRate32kHz &&
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Aquire lock for the access of api_format.
// The lock is released immediately due to the conditional
// reinitialization.
rtc::CritScope cs_capture(&crit_capture_);
// TODO(ajm): The input and output rates and channels are currently
// constrained to be identical in the int16 interface.
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
reinitialization_required =
reinitialization_required ||
processing_config.input_stream().sample_rate_hz() !=
frame->sample_rate_hz_ ||
processing_config.input_stream().num_channels() != frame->num_channels_ ||
processing_config.output_stream().sample_rate_hz() !=
frame->sample_rate_hz_ ||
processing_config.output_stream().num_channels() != frame->num_channels_;
if (reinitialization_required) {
processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.input_stream().set_num_channels(frame->num_channels_);
processing_config.output_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.output_stream().set_num_channels(frame->num_channels_);
// Reinitialize.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
rtc::CritScope cs_capture(&crit_capture_);
if (frame->samples_per_channel_ !=
formats_.api_format.input_stream().num_frames()) {
return kBadDataLengthError;
}
if (aec_dump_) {
RecordUnprocessedCaptureStream(*frame);
}
capture_.capture_audio->CopyFrom(frame);
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyFrom(frame);
}
RETURN_ON_ERR(ProcessCaptureStreamLocked());
if (submodule_states_.CaptureMultiBandProcessingPresent() ||
submodule_states_.CaptureFullBandProcessingActive()) {
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyTo(frame);
} else {
capture_.capture_audio->CopyTo(frame);
}
}
if (capture_.stats.voice_detected) {
frame->vad_activity_ = *capture_.stats.voice_detected
? AudioFrame::kVadActive
: AudioFrame::kVadPassive;
}
if (aec_dump_) {
RecordProcessedCaptureStream(*frame);
}
return kNoError;
}
int AudioProcessingImpl::ProcessCaptureStreamLocked() {
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_cancellation +
!!submodules_.echo_control_mobile,
1);
AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity.
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 (constants_.use_experimental_agc &&
submodules_.gain_control->is_enabled()) {
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) {
submodules_.high_pass_filter->Process(capture_buffer);
}
RETURN_ON_ERR(submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
RTC_DCHECK(
!(submodules_.legacy_noise_suppressor && submodules_.noise_suppressor));
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->Analyze(*capture_buffer);
} else if (submodules_.legacy_noise_suppressor) {
submodules_.legacy_noise_suppressor->AnalyzeCaptureAudio(capture_buffer);
}
if (submodules_.echo_control_mobile) {
// Ensure that the stream delay was set before the call to the
// AECM ProcessCaptureAudio function.
if (!was_stream_delay_set()) {
return AudioProcessing::kStreamParameterNotSetError;
}
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->Process(capture_buffer);
} else if (submodules_.legacy_noise_suppressor) {
submodules_.echo_control_mobile->CopyLowPassReference(capture_buffer);
submodules_.legacy_noise_suppressor->ProcessCaptureAudio(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 (was_stream_delay_set()) {
submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms());
}
AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
submodules_.echo_controller->ProcessCapture(
capture_buffer, linear_aec_buffer, capture_.echo_path_gain_change);
} else if (submodules_.echo_cancellation) {
// Ensure that the stream delay was set before the call to the
// AEC ProcessCaptureAudio function.
if (!was_stream_delay_set()) {
return AudioProcessing::kStreamParameterNotSetError;
}
RETURN_ON_ERR(submodules_.echo_cancellation->ProcessCaptureAudio(
capture_buffer, stream_delay_ms()));
}
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->Process(capture_buffer);
} else if (submodules_.legacy_noise_suppressor) {
submodules_.legacy_noise_suppressor->ProcessCaptureAudio(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 (constants_.use_experimental_agc &&
submodules_.gain_control->is_enabled()) {
submodules_.agc_manager->Process(capture_buffer);
absl::optional<int> new_digital_gain =
submodules_.agc_manager->GetDigitalComressionGain();
if (new_digital_gain) {
submodules_.gain_control->set_compression_gain_db(*new_digital_gain);
}
}
// TODO(peah): Add reporting from AEC3 whether there is echo.
RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio(
capture_buffer, submodules_.echo_cancellation &&
submodules_.echo_cancellation->stream_has_echo()));
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 (capture_.transient_suppressor_enabled) {
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 (config_.gain_controller2.enabled) {
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);
}
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(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
rtc::CritScope cs(&crit_render_);
if (frame == nullptr) {
return kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != kSampleRate8kHz &&
frame->sample_rate_hz_ != kSampleRate16kHz &&
frame->sample_rate_hz_ != kSampleRate32kHz &&
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
if (frame->num_channels_ <= 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.reverse_input_stream().set_num_channels(
frame->num_channels_);
processing_config.reverse_output_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.reverse_output_stream().set_num_channels(
frame->num_channels_);
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
if (frame->samples_per_channel_ !=
formats_.api_format.reverse_input_stream().num_frames()) {
return kBadDataLengthError;
}
if (aec_dump_) {
aec_dump_->WriteRenderStreamMessage(*frame);
}
render_.render_audio->CopyFrom(frame);
RETURN_ON_ERR(ProcessRenderStreamLocked());
if (submodule_states_.RenderMultiBandProcessingActive() ||
submodule_states_.RenderFullBandProcessingActive()) {
render_.render_audio->CopyTo(frame);
}
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;
delay += capture_.delay_offset_ms;
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;
}
bool AudioProcessingImpl::was_stream_delay_set() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_.was_stream_delay_set;
}
void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
rtc::CritScope cs(&crit_capture_);
capture_.key_pressed = key_pressed;
}
void AudioProcessingImpl::set_delay_offset_ms(int offset) {
rtc::CritScope cs(&crit_capture_);
capture_.delay_offset_ms = offset;
}
int AudioProcessingImpl::delay_offset_ms() const {
rtc::CritScope cs(&crit_capture_);
return capture_.delay_offset_ms;
}
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 {
int error = submodules_.gain_control->set_stream_analog_level(level);
RTC_DCHECK_EQ(kNoError, error);
}
}
int AudioProcessingImpl::recommended_stream_analog_level() const {
rtc::CritScope cs_capture(&crit_capture_);
if (submodules_.agc_manager) {
return submodules_.agc_manager->stream_analog_level();
}
return submodules_.gain_control->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.
}
AudioProcessingStats AudioProcessingImpl::GetStatistics(
bool has_remote_tracks) const {
rtc::CritScope cs_capture(&crit_capture_);
if (!has_remote_tracks) {
return capture_.stats;
}
AudioProcessingStats stats = capture_.stats;
EchoCancellationImpl::Metrics metrics;
if (submodules_.echo_controller) {
auto ec_metrics = submodules_.echo_controller->GetMetrics();
stats.echo_return_loss = ec_metrics.echo_return_loss;
stats.echo_return_loss_enhancement =
ec_metrics.echo_return_loss_enhancement;
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();
stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
stats.residual_echo_likelihood_recent_max =
ed_metrics.echo_likelihood_recent_max;
}
return stats;
}
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_cancellation,
!!submodules_.echo_control_mobile, config_.residual_echo_detector.enabled,
!!submodules_.legacy_noise_suppressor || !!submodules_.noise_suppressor,
submodules_.gain_control->is_enabled(), config_.gain_controller2.enabled,
config_.pre_amplifier.enabled, capture_nonlocked_.echo_controller_enabled,
config_.voice_detection.enabled, capture_.transient_suppressor_enabled);
}
void AudioProcessingImpl::InitializeTransient() {
if (capture_.transient_suppressor_enabled) {
if (!submodules_.transient_suppressor.get()) {
submodules_.transient_suppressor.reset(new TransientSuppressor());
}
submodules_.transient_suppressor->Initialize(proc_fullband_sample_rate_hz(),
capture_nonlocked_.split_rate,
num_proc_channels());
}
}
void AudioProcessingImpl::InitializeHighPassFilter() {
if (submodule_states_.HighPassFilteringRequired()) {
submodules_.high_pass_filter.reset(new HighPassFilter(num_proc_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 &&
!config_.echo_canceller.use_legacy_aec);
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_cancellation.reset();
aec_render_signal_queue_.reset();
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_cancellation.reset();
aec_render_signal_queue_.reset();
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());
submodules_.echo_cancellation.reset();
aec_render_signal_queue_.reset();
return;
}
submodules_.echo_control_mobile.reset();
aecm_render_signal_queue_.reset();
// Create and activate AEC2.
submodules_.echo_cancellation.reset(new EchoCancellationImpl());
submodules_.echo_cancellation->SetExtraOptions(
capture_nonlocked_.use_aec2_extended_filter,
capture_nonlocked_.use_aec2_delay_agnostic,
capture_nonlocked_.use_aec2_refined_adaptive_filter);
size_t element_max_size =
std::max(static_cast<size_t>(1),
kMaxAllowedValuesOfSamplesPerBand *
EchoCancellationImpl::NumCancellersRequired(
num_output_channels(), num_reverse_channels()));
std::vector<float> template_queue_element(element_max_size);
aec_render_signal_queue_.reset(
new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
kMaxNumFramesToBuffer, template_queue_element,
RenderQueueItemVerifier<float>(element_max_size)));
aec_render_queue_buffer_.resize(element_max_size);
aec_capture_queue_buffer_.resize(element_max_size);
submodules_.echo_cancellation->Initialize(
proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
num_proc_channels());
submodules_.echo_cancellation->set_suppression_level(
config_.echo_canceller.legacy_moderate_suppression_level
? EchoCancellationImpl::SuppressionLevel::kModerateSuppression
: EchoCancellationImpl::SuppressionLevel::kHighSuppression);
}
void AudioProcessingImpl::InitializeGainController2() {
if (config_.gain_controller2.enabled) {
submodules_.gain_controller2->Initialize(proc_fullband_sample_rate_hz());
}
}
void AudioProcessingImpl::InitializeNoiseSuppressor() {
submodules_.legacy_noise_suppressor.reset();
submodules_.noise_suppressor.reset();
if (config_.noise_suppression.enabled) {
const bool use_legacy_ns =
config_.noise_suppression.use_legacy_ns || enforced_usage_of_legacy_ns_;
if (!use_legacy_ns) {
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());
} else {
auto ns_level =
NsConfigLevelToInterfaceLevel(config_.noise_suppression.level);
submodules_.legacy_noise_suppressor = std::make_unique<NoiseSuppression>(
num_proc_channels(), proc_sample_rate_hz(), ns_level);
}
}
}
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 = "";
if (submodules_.echo_cancellation) {
experiments_description +=
submodules_.echo_cancellation->GetExperimentsDescription();
}
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
if (constants_.agc_clipped_level_min != kClippedLevelMin) {
experiments_description += "AgcClippingLevelExperiment;";
}
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 =
submodules_.echo_cancellation &&
submodules_.echo_cancellation->is_delay_agnostic_enabled();
apm_config.aec_drift_compensation_enabled =
submodules_.echo_cancellation &&
submodules_.echo_cancellation->is_drift_compensation_enabled();
apm_config.aec_extended_filter_enabled =
submodules_.echo_cancellation &&
submodules_.echo_cancellation->is_extended_filter_enabled();
apm_config.aec_suppression_level =
submodules_.echo_cancellation
? static_cast<int>(submodules_.echo_cancellation->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->is_enabled();
apm_config.agc_mode = static_cast<int>(submodules_.gain_control->mode());
apm_config.agc_limiter_enabled =
submodules_.gain_control->is_limiter_enabled();
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 =
capture_.transient_suppressor_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 AudioFrame& capture_frame) {
RTC_DCHECK(aec_dump_);
WriteAecDumpConfigMessage(false);
aec_dump_->AddCaptureStreamInput(capture_frame);
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 AudioFrame& processed_capture_frame) {
RTC_DCHECK(aec_dump_);
aec_dump_->AddCaptureStreamOutput(processed_capture_frame);
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 =
submodules_.echo_cancellation
? submodules_.echo_cancellation->stream_drift_samples()
: 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(
bool transient_suppressor_enabled)
: delay_offset_ms(0),
was_stream_delay_set(false),
output_will_be_muted(false),
key_pressed(false),
transient_suppressor_enabled(transient_suppressor_enabled),
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;
} // namespace webrtc