mozilla-central/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc (file symbol)

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/*
* 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 <cstring>
#include <memory>
#include <string>
#include <type_traits>
#include <utility>
#include "absl/base/nullability.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "api/task_queue/task_queue_base.h"
#include "common_audio/audio_converter.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/denormal_disabler.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 {
namespace {
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_DCHECK_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;
}
RTC_CHECK_NOTREACHED();
}
bool MinimizeProcessingForUnusedOutput() {
return !field_trial::IsEnabled("WebRTC-MutedStateKillSwitch");
}
// 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;
void PackRenderAudioBufferForEchoDetector(const AudioBuffer& audio,
std::vector<float>& packed_buffer) {
packed_buffer.clear();
packed_buffer.insert(packed_buffer.end(), audio.channels_const()[0],
audio.channels_const()[0] + audio.num_frames());
}
// Options for gracefully handling processing errors.
enum class FormatErrorOutputOption {
kOutputExactCopyOfInput,
kOutputBroadcastCopyOfFirstInputChannel,
kOutputSilence,
kDoNothing
};
enum class AudioFormatValidity {
// Format is supported by APM.
kValidAndSupported,
// Format has a reasonable interpretation but is not supported.
kValidButUnsupportedSampleRate,
// The remaining enums values signal that the audio does not have a reasonable
// interpretation and cannot be used.
kInvalidSampleRate,
kInvalidChannelCount
};
AudioFormatValidity ValidateAudioFormat(const StreamConfig& config) {
if (config.sample_rate_hz() < 0)
return AudioFormatValidity::kInvalidSampleRate;
if (config.num_channels() == 0)
return AudioFormatValidity::kInvalidChannelCount;
// Format has a reasonable interpretation, but may still be unsupported.
if (config.sample_rate_hz() < 8000 ||
config.sample_rate_hz() > AudioBuffer::kMaxSampleRate)
return AudioFormatValidity::kValidButUnsupportedSampleRate;
// Format is fully supported.
return AudioFormatValidity::kValidAndSupported;
}
int AudioFormatValidityToErrorCode(AudioFormatValidity validity) {
switch (validity) {
case AudioFormatValidity::kValidAndSupported:
return AudioProcessing::kNoError;
case AudioFormatValidity::kValidButUnsupportedSampleRate: // fall-through
case AudioFormatValidity::kInvalidSampleRate:
return AudioProcessing::kBadSampleRateError;
case AudioFormatValidity::kInvalidChannelCount:
return AudioProcessing::kBadNumberChannelsError;
}
RTC_DCHECK(false);
}
// Returns an AudioProcessing::Error together with the best possible option for
// output audio content.
std::pair<int, FormatErrorOutputOption> ChooseErrorOutputOption(
const StreamConfig& input_config,
const StreamConfig& output_config) {
AudioFormatValidity input_validity = ValidateAudioFormat(input_config);
AudioFormatValidity output_validity = ValidateAudioFormat(output_config);
if (input_validity == AudioFormatValidity::kValidAndSupported &&
output_validity == AudioFormatValidity::kValidAndSupported &&
(output_config.num_channels() == 1 ||
output_config.num_channels() == input_config.num_channels())) {
return {AudioProcessing::kNoError, FormatErrorOutputOption::kDoNothing};
}
int error_code = AudioFormatValidityToErrorCode(input_validity);
if (error_code == AudioProcessing::kNoError) {
error_code = AudioFormatValidityToErrorCode(output_validity);
}
if (error_code == AudioProcessing::kNoError) {
// The individual formats are valid but there is some error - must be
// channel mismatch.
error_code = AudioProcessing::kBadNumberChannelsError;
}
FormatErrorOutputOption output_option;
if (output_validity != AudioFormatValidity::kValidAndSupported &&
output_validity != AudioFormatValidity::kValidButUnsupportedSampleRate) {
// The output format is uninterpretable: cannot do anything.
output_option = FormatErrorOutputOption::kDoNothing;
} else if (input_validity != AudioFormatValidity::kValidAndSupported &&
input_validity !=
AudioFormatValidity::kValidButUnsupportedSampleRate) {
// The input format is uninterpretable: cannot use it, must output silence.
output_option = FormatErrorOutputOption::kOutputSilence;
} else if (input_config.sample_rate_hz() != output_config.sample_rate_hz()) {
// Sample rates do not match: Cannot copy input into output, output silence.
// Note: If the sample rates are in a supported range, we could resample.
// However, that would significantly increase complexity of this error
// handling code.
output_option = FormatErrorOutputOption::kOutputSilence;
} else if (input_config.num_channels() != output_config.num_channels()) {
// Channel counts do not match: We cannot easily map input channels to
// output channels.
output_option =
FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel;
} else {
// The formats match exactly.
RTC_DCHECK(input_config == output_config);
output_option = FormatErrorOutputOption::kOutputExactCopyOfInput;
}
return std::make_pair(error_code, output_option);
}
// Checks if the audio format is supported. If not, the output is populated in a
// best-effort manner and an APM error code is returned.
int HandleUnsupportedAudioFormats(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) {
RTC_DCHECK(src);
RTC_DCHECK(dest);
auto [error_code, output_option] =
ChooseErrorOutputOption(input_config, output_config);
if (error_code == AudioProcessing::kNoError)
return AudioProcessing::kNoError;
const size_t num_output_channels = output_config.num_channels();
switch (output_option) {
case FormatErrorOutputOption::kOutputSilence:
memset(dest, 0, output_config.num_samples() * sizeof(int16_t));
break;
case FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel:
for (size_t i = 0; i < output_config.num_frames(); ++i) {
int16_t sample = src[input_config.num_channels() * i];
for (size_t ch = 0; ch < num_output_channels; ++ch) {
dest[ch + num_output_channels * i] = sample;
}
}
break;
case FormatErrorOutputOption::kOutputExactCopyOfInput:
memcpy(dest, src, output_config.num_samples() * sizeof(int16_t));
break;
case FormatErrorOutputOption::kDoNothing:
break;
}
return error_code;
}
// Checks if the audio format is supported. If not, the output is populated in a
// best-effort manner and an APM error code is returned.
int HandleUnsupportedAudioFormats(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
RTC_DCHECK(src);
RTC_DCHECK(dest);
for (size_t i = 0; i < input_config.num_channels(); ++i) {
RTC_DCHECK(src[i]);
}
for (size_t i = 0; i < output_config.num_channels(); ++i) {
RTC_DCHECK(dest[i]);
}
auto [error_code, output_option] =
ChooseErrorOutputOption(input_config, output_config);
if (error_code == AudioProcessing::kNoError)
return AudioProcessing::kNoError;
const size_t num_output_channels = output_config.num_channels();
switch (output_option) {
case FormatErrorOutputOption::kOutputSilence:
for (size_t ch = 0; ch < num_output_channels; ++ch) {
memset(dest[ch], 0, output_config.num_frames() * sizeof(float));
}
break;
case FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel:
for (size_t ch = 0; ch < num_output_channels; ++ch) {
memcpy(dest[ch], src[0], output_config.num_frames() * sizeof(float));
}
break;
case FormatErrorOutputOption::kOutputExactCopyOfInput:
for (size_t ch = 0; ch < num_output_channels; ++ch) {
memcpy(dest[ch], src[ch], output_config.num_frames() * sizeof(float));
}
break;
case FormatErrorOutputOption::kDoNothing:
break;
}
return error_code;
}
using DownmixMethod = AudioProcessing::Config::Pipeline::DownmixMethod;
void SetDownmixMethod(AudioBuffer& buffer, DownmixMethod method) {
switch (method) {
case DownmixMethod::kAverageChannels:
buffer.set_downmixing_by_averaging();
break;
case DownmixMethod::kUseFirstChannel:
buffer.set_downmixing_to_specific_channel(/*channel=*/0);
break;
}
}
constexpr int kUnspecifiedDataDumpInputVolume = -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 noise_suppressor_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
bool gain_adjustment_enabled,
bool echo_controller_enabled) {
bool changed = false;
changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
changed |=
(mobile_echo_controller_enabled != mobile_echo_controller_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 |= (gain_adjustment_enabled != gain_adjustment_enabled_);
changed |= (echo_controller_enabled != echo_controller_enabled_);
if (changed) {
high_pass_filter_enabled_ = high_pass_filter_enabled;
mobile_echo_controller_enabled_ = mobile_echo_controller_enabled;
noise_suppressor_enabled_ = noise_suppressor_enabled;
adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
gain_controller2_enabled_ = gain_controller2_enabled;
gain_adjustment_enabled_ = gain_adjustment_enabled;
echo_controller_enabled_ = echo_controller_enabled;
}
changed |= first_update_;
first_update_ = false;
return changed;
}
bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandSubModulesActive()
const {
return CaptureMultiBandProcessingPresent();
}
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_ ||
gain_adjustment_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_;
}
AudioProcessingImpl::AudioProcessingImpl()
: AudioProcessingImpl(/*config=*/{},
/*capture_post_processor=*/nullptr,
/*render_pre_processor=*/nullptr,
/*echo_control_factory=*/nullptr,
/*echo_detector=*/nullptr,
/*capture_analyzer=*/nullptr) {}
std::atomic<int> AudioProcessingImpl::instance_count_(0);
AudioProcessingImpl::AudioProcessingImpl(
const AudioProcessing::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(instance_count_.fetch_add(1) + 1)),
use_setup_specific_default_aec3_config_(
UseSetupSpecificDefaultAec3Congfig()),
capture_runtime_settings_(RuntimeSettingQueueSize()),
render_runtime_settings_(RuntimeSettingQueueSize()),
capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
render_runtime_settings_enqueuer_(&render_runtime_settings_),
echo_control_factory_(std::move(echo_control_factory)),
config_(config),
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(),
MinimizeProcessingForUnusedOutput()),
capture_(),
capture_nonlocked_(),
applied_input_volume_stats_reporter_(
InputVolumeStatsReporter::InputVolumeType::kApplied),
recommended_input_volume_stats_reporter_(
InputVolumeStatsReporter::InputVolumeType::kRecommended) {
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;
if (!DenormalDisabler::IsSupported()) {
RTC_LOG(LS_INFO) << "Denormal disabler unsupported";
}
RTC_LOG(LS_INFO) << "AudioProcessing: " << config_.ToString();
// Mark Echo Controller enabled if a factory is injected.
capture_nonlocked_.echo_controller_enabled =
static_cast<bool>(echo_control_factory_);
Initialize();
}
AudioProcessingImpl::~AudioProcessingImpl() = default;
int AudioProcessingImpl::Initialize() {
// Run in a single-threaded manner during initialization.
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
InitializeLocked();
return kNoError;
}
int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
// Run in a single-threaded manner during initialization.
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
InitializeLocked(processing_config);
return kNoError;
}
void AudioProcessingImpl::MaybeInitializeRender(
const StreamConfig& input_config,
const StreamConfig& output_config) {
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream() = input_config;
processing_config.reverse_output_stream() = output_config;
if (processing_config == formats_.api_format) {
return;
}
MutexLock lock_capture(&mutex_capture_);
InitializeLocked(processing_config);
}
void 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()));
SetDownmixMethod(*capture_.capture_audio,
config_.pipeline.capture_downmix_method);
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()));
SetDownmixMethod(*capture_.capture_fullband_audio,
config_.pipeline.capture_downmix_method);
} else {
capture_.capture_fullband_audio.reset();
}
AllocateRenderQueue();
InitializeGainController1();
InitializeHighPassFilter(true);
InitializeResidualEchoDetector();
InitializeEchoController();
InitializeGainController2();
InitializeNoiseSuppressor();
InitializeAnalyzer();
InitializePostProcessor();
InitializePreProcessor();
InitializeCaptureLevelsAdjuster();
if (aec_dump_) {
aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
}
}
void AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
UpdateActiveSubmoduleStates();
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();
}
InitializeLocked();
}
void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
// Run in a single-threaded manner when applying the settings.
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
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 ||
config_.pipeline.capture_downmix_method !=
config.pipeline.capture_downmix_method;
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 != config.gain_controller1;
const bool agc2_config_changed =
config_.gain_controller2 != config.gain_controller2;
const bool ns_config_changed =
config_.noise_suppression.enabled != config.noise_suppression.enabled ||
config_.noise_suppression.level != config.noise_suppression.level;
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;
const bool gain_adjustment_config_changed =
config_.capture_level_adjustment != config.capture_level_adjustment;
config_ = config;
if (aec_config_changed) {
InitializeEchoController();
}
if (ns_config_changed) {
InitializeNoiseSuppressor();
}
InitializeHighPassFilter(false);
if (agc1_config_changed) {
InitializeGainController1();
}
const bool config_ok = GainController2::Validate(config_.gain_controller2);
if (!config_ok) {
RTC_LOG(LS_ERROR)
<< "Invalid Gain Controller 2 config; using the default config.";
config_.gain_controller2 = AudioProcessing::Config::GainController2();
}
if (agc2_config_changed) {
InitializeGainController2();
}
if (pre_amplifier_config_changed || gain_adjustment_config_changed) {
InitializeCaptureLevelsAdjuster();
}
// 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);
}
}
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) {
MutexLock lock(&mutex_capture_);
HandleCaptureOutputUsedSetting(!muted);
}
void AudioProcessingImpl::HandleCaptureOutputUsedSetting(
bool capture_output_used) {
capture_.capture_output_used =
capture_output_used || !constants_.minimize_processing_for_unused_output;
if (submodules_.agc_manager.get()) {
submodules_.agc_manager->HandleCaptureOutputUsedChange(
capture_.capture_output_used);
}
if (submodules_.echo_controller) {
submodules_.echo_controller->SetCaptureOutputUsage(
capture_.capture_output_used);
}
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->SetCaptureOutputUsage(
capture_.capture_output_used);
}
if (submodules_.gain_controller2) {
submodules_.gain_controller2->SetCaptureOutputUsed(
capture_.capture_output_used);
}
}
void AudioProcessingImpl::SetRuntimeSetting(RuntimeSetting setting) {
PostRuntimeSetting(setting);
}
bool AudioProcessingImpl::PostRuntimeSetting(RuntimeSetting setting) {
switch (setting.type()) {
case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
return render_runtime_settings_enqueuer_.Enqueue(setting);
case RuntimeSetting::Type::kCapturePreGain:
case RuntimeSetting::Type::kCapturePostGain:
case RuntimeSetting::Type::kCaptureCompressionGain:
case RuntimeSetting::Type::kCaptureFixedPostGain:
case RuntimeSetting::Type::kCaptureOutputUsed:
return capture_runtime_settings_enqueuer_.Enqueue(setting);
case RuntimeSetting::Type::kPlayoutVolumeChange: {
bool enqueueing_successful;
enqueueing_successful =
capture_runtime_settings_enqueuer_.Enqueue(setting);
enqueueing_successful =
render_runtime_settings_enqueuer_.Enqueue(setting) &&
enqueueing_successful;
return enqueueing_successful;
}
case RuntimeSetting::Type::kNotSpecified:
RTC_DCHECK_NOTREACHED();
return true;
}
// The language allows the enum to have a non-enumerator
// value. Check that this doesn't happen.
RTC_DCHECK_NOTREACHED();
return true;
}
AudioProcessingImpl::RuntimeSettingEnqueuer::RuntimeSettingEnqueuer(
SwapQueue<RuntimeSetting>* runtime_settings)
: runtime_settings_(*runtime_settings) {
RTC_DCHECK(runtime_settings);
}
AudioProcessingImpl::RuntimeSettingEnqueuer::~RuntimeSettingEnqueuer() =
default;
bool AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
RuntimeSetting setting) {
const bool successful_insert = runtime_settings_.Insert(&setting);
if (!successful_insert) {
RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
}
return successful_insert;
}
void 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.
MutexLock lock_capture(&mutex_capture_);
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
if (processing_config.input_stream() != input_config) {
reinitialization_required = true;
}
if (processing_config.output_stream() != output_config) {
reinitialization_required = true;
}
if (reinitialization_required) {
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
// Reread the API format since the render format may have changed.
processing_config = formats_.api_format;
processing_config.input_stream() = input_config;
processing_config.output_stream() = output_config;
InitializeLocked(processing_config);
}
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
DenormalDisabler denormal_disabler;
RETURN_ON_ERR(
HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
MaybeInitializeCapture(input_config, output_config);
MutexLock lock_capture(&mutex_capture_);
if (aec_dump_) {
RecordUnprocessedCaptureStream(src);
}
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;
int num_settings_processed = 0;
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 ||
config_.capture_level_adjustment.enabled) {
float value;
setting.GetFloat(&value);
// If the pre-amplifier is used, apply the new gain to the
// pre-amplifier regardless if the capture level adjustment is
// activated. This approach allows both functionalities to coexist
// until they have been properly merged.
if (config_.pre_amplifier.enabled) {
config_.pre_amplifier.fixed_gain_factor = value;
} else {
config_.capture_level_adjustment.pre_gain_factor = value;
}
// Use both the pre-amplifier and the capture level adjustment gains
// as pre-gains.
float gain = 1.f;
if (config_.pre_amplifier.enabled) {
gain *= config_.pre_amplifier.fixed_gain_factor;
}
if (config_.capture_level_adjustment.enabled) {
gain *= config_.capture_level_adjustment.pre_gain_factor;
}
submodules_.capture_levels_adjuster->SetPreGain(gain);
}
// TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
break;
case RuntimeSetting::Type::kCapturePostGain:
if (config_.capture_level_adjustment.enabled) {
float value;
setting.GetFloat(&value);
config_.capture_level_adjustment.post_gain_factor = value;
submodules_.capture_levels_adjuster->SetPostGain(
config_.capture_level_adjustment.post_gain_factor);
}
// TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
break;
case RuntimeSetting::Type::kCaptureCompressionGain: {
if (!submodules_.agc_manager &&
!(submodules_.gain_controller2 &&
config_.gain_controller2.input_volume_controller.enabled)) {
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->SetFixedGainDb(value);
}
break;
}
case RuntimeSetting::Type::kPlayoutVolumeChange: {
int value;
setting.GetInt(&value);
capture_.playout_volume = value;
break;
}
case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
RTC_DCHECK_NOTREACHED();
break;
case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
RTC_DCHECK_NOTREACHED();
break;
case RuntimeSetting::Type::kNotSpecified:
RTC_DCHECK_NOTREACHED();
break;
case RuntimeSetting::Type::kCaptureOutputUsed:
bool value;
setting.GetBool(&value);
HandleCaptureOutputUsedSetting(value);
break;
}
++num_settings_processed;
}
if (num_settings_processed >= RuntimeSettingQueueSize()) {
// Handle overrun of the runtime settings queue, which likely will has
// caused settings to be discarded.
HandleOverrunInCaptureRuntimeSettingsQueue();
}
}
void AudioProcessingImpl::HandleOverrunInCaptureRuntimeSettingsQueue() {
// Fall back to a safe state for the case when a setting for capture output
// usage setting has been missed.
HandleCaptureOutputUsedSetting(/*capture_output_used=*/true);
}
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::kCapturePostGain: // fall-through
case RuntimeSetting::Type::kCaptureCompressionGain: // fall-through
case RuntimeSetting::Type::kCaptureFixedPostGain: // fall-through
case RuntimeSetting::Type::kCaptureOutputUsed: // fall-through
case RuntimeSetting::Type::kNotSpecified:
RTC_DCHECK_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) {
if (submodules_.echo_detector) {
PackRenderAudioBufferForEchoDetector(*audio, red_render_queue_buffer_);
RTC_DCHECK(red_render_signal_queue_);
// 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 (submodules_.echo_detector) {
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() {
MutexLock lock_capture(&mutex_capture_);
EmptyQueuedRenderAudioLocked();
}
void AudioProcessingImpl::EmptyQueuedRenderAudioLocked() {
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_);
}
}
if (submodules_.echo_detector) {
while (red_render_signal_queue_->Remove(&red_capture_queue_buffer_)) {
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(
HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
MaybeInitializeCapture(input_config, output_config);
MutexLock lock_capture(&mutex_capture_);
DenormalDisabler denormal_disabler;
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() {
EmptyQueuedRenderAudioLocked();
HandleCaptureRuntimeSettings();
DenormalDisabler denormal_disabler;
// 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);
data_dumper_->DumpRaw(
"applied_input_volume",
capture_.applied_input_volume.value_or(kUnspecifiedDataDumpInputVolume));
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_.capture_levels_adjuster) {
if (config_.capture_level_adjustment.analog_mic_gain_emulation.enabled) {
// When the input volume is emulated, retrieve the volume applied to the
// input audio and notify that to APM so that the volume is passed to the
// active AGC.
set_stream_analog_level_locked(
submodules_.capture_levels_adjuster->GetAnalogMicGainLevel());
}
submodules_.capture_levels_adjuster->ApplyPreLevelAdjustment(
*capture_buffer);
}
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 (capture_.applied_input_volume.has_value()) {
applied_input_volume_stats_reporter_.UpdateStatistics(
*capture_.applied_input_volume);
}
if (submodules_.echo_controller) {
// Determine if the echo path gain has changed by checking all the gains
// applied before AEC.
capture_.echo_path_gain_change = capture_.applied_input_volume_changed;
// Detect and flag any change in the capture level adjustment pre-gain.
if (submodules_.capture_levels_adjuster) {
float pre_adjustment_gain =
submodules_.capture_levels_adjuster->GetPreAdjustmentGain();
capture_.echo_path_gain_change =
capture_.echo_path_gain_change ||
(capture_.prev_pre_adjustment_gain != pre_adjustment_gain &&
capture_.prev_pre_adjustment_gain >= 0.0f);
capture_.prev_pre_adjustment_gain = pre_adjustment_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 (submodules_.gain_controller2 &&
config_.gain_controller2.input_volume_controller.enabled) {
// Expect the volume to be available if the input controller is enabled.
RTC_DCHECK(capture_.applied_input_volume.has_value());
if (capture_.applied_input_volume.has_value()) {
submodules_.gain_controller2->Analyze(*capture_.applied_input_volume,
*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 (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_output_used) {
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 (submodules_.echo_detector) {
submodules_.echo_detector->AnalyzeCaptureAudio(
rtc::ArrayView<const float>(capture_buffer->channels()[0],
capture_buffer->num_frames()));
}
// Experimental APM sub-module that analyzes `capture_buffer`.
if (submodules_.capture_analyzer) {
submodules_.capture_analyzer->Analyze(capture_buffer);
}
if (submodules_.gain_controller2) {
// TODO(bugs.webrtc.org/7494): Let AGC2 detect applied input volume
// changes.
submodules_.gain_controller2->Process(
/*speech_probability=*/absl::nullopt,
capture_.applied_input_volume_changed, capture_buffer);
}
if (submodules_.capture_post_processor) {
submodules_.capture_post_processor->Process(capture_buffer);
}
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);
}
// Compute echo-detector stats.
if (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;
}
}
// Compute echo-controller 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;
}
// Pass stats for reporting.
stats_reporter_.UpdateStatistics(capture_.stats);
UpdateRecommendedInputVolumeLocked();
if (capture_.recommended_input_volume.has_value()) {
recommended_input_volume_stats_reporter_.UpdateStatistics(
*capture_.recommended_input_volume);
}
if (submodules_.capture_levels_adjuster) {
submodules_.capture_levels_adjuster->ApplyPostLevelAdjustment(
*capture_buffer);
if (config_.capture_level_adjustment.analog_mic_gain_emulation.enabled) {
// If the input volume emulation is used, retrieve the recommended input
// volume and set that to emulate the input volume on the next processed
// audio frame.
RTC_DCHECK(capture_.recommended_input_volume.has_value());
submodules_.capture_levels_adjuster->SetAnalogMicGainLevel(
*capture_.recommended_input_volume);
}
}
// Temporarily set the output to zero after the stream has been unmuted
// (capture output is again used). The purpose of this is to avoid clicks and
// artefacts in the audio that results when the processing again is
// reactivated after unmuting.
if (!capture_.capture_output_used_last_frame &&
capture_.capture_output_used) {
for (size_t ch = 0; ch < capture_buffer->num_channels(); ++ch) {
rtc::ArrayView<float> channel_view(capture_buffer->channels()[ch],
capture_buffer->num_frames());
std::fill(channel_view.begin(), channel_view.end(), 0.f);
}
}
capture_.capture_output_used_last_frame = capture_.capture_output_used;
capture_.was_stream_delay_set = false;
data_dumper_->DumpRaw("recommended_input_volume",
capture_.recommended_input_volume.value_or(
kUnspecifiedDataDumpInputVolume));
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(
const float* const* data,
const StreamConfig& reverse_config) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_StreamConfig");
MutexLock lock(&mutex_render_);
DenormalDisabler denormal_disabler;
RTC_DCHECK(data);
for (size_t i = 0; i < reverse_config.num_channels(); ++i) {
RTC_DCHECK(data[i]);
}
RETURN_ON_ERR(
AudioFormatValidityToErrorCode(ValidateAudioFormat(reverse_config)));
MaybeInitializeRender(reverse_config, reverse_config);
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");
MutexLock lock(&mutex_render_);
DenormalDisabler denormal_disabler;
RETURN_ON_ERR(
HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
MaybeInitializeRender(input_config, output_config);
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 (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");
MutexLock lock(&mutex_render_);
DenormalDisabler denormal_disabler;
RETURN_ON_ERR(
HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
MaybeInitializeRender(input_config, output_config);
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();
DenormalDisabler denormal_disabler;
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) {
MutexLock lock(&mutex_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 {
MutexLock lock(&mutex_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());
FloatS16ToFloat(channel_view.data(), channel_view.size(),
linear_output[ch].data());
}
return true;
}
RTC_LOG(LS_ERROR) << "No linear AEC output available";
RTC_DCHECK_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) {
MutexLock lock(&mutex_capture_);
capture_.key_pressed = key_pressed;
}
void AudioProcessingImpl::set_stream_analog_level(int level) {
MutexLock lock_capture(&mutex_capture_);
set_stream_analog_level_locked(level);
}
void AudioProcessingImpl::set_stream_analog_level_locked(int level) {
capture_.applied_input_volume_changed =
capture_.applied_input_volume.has_value() &&
*capture_.applied_input_volume != level;
capture_.applied_input_volume = level;
// Invalidate any previously recommended input volume which will be updated by
// `ProcessStream()`.
capture_.recommended_input_volume = absl::nullopt;
if (submodules_.agc_manager) {
submodules_.agc_manager->set_stream_analog_level(level);
return;
}
if (submodules_.gain_control) {
int error = submodules_.gain_control->set_stream_analog_level(level);
RTC_DCHECK_EQ(kNoError, error);
return;
}
}
int AudioProcessingImpl::recommended_stream_analog_level() const {
MutexLock lock_capture(&mutex_capture_);
if (!capture_.applied_input_volume.has_value()) {
RTC_LOG(LS_ERROR) << "set_stream_analog_level has not been called";
}
// Input volume to recommend when `set_stream_analog_level()` is not called.
constexpr int kFallBackInputVolume = 255;
// When APM has no input volume to recommend, return the latest applied input
// volume that has been observed in order to possibly produce no input volume
// change. If no applied input volume has been observed, return a fall-back
// value.
return capture_.recommended_input_volume.value_or(
capture_.applied_input_volume.value_or(kFallBackInputVolume));
}
void AudioProcessingImpl::UpdateRecommendedInputVolumeLocked() {
if (!capture_.applied_input_volume.has_value()) {
// When `set_stream_analog_level()` is not called, no input level can be
// recommended.
capture_.recommended_input_volume = absl::nullopt;
return;
}
if (submodules_.agc_manager) {
capture_.recommended_input_volume =
submodules_.agc_manager->recommended_analog_level();
return;
}
if (submodules_.gain_control) {
capture_.recommended_input_volume =
submodules_.gain_control->stream_analog_level();
return;
}
if (submodules_.gain_controller2 &&
config_.gain_controller2.input_volume_controller.enabled) {
capture_.recommended_input_volume =
submodules_.gain_controller2->recommended_input_volume();
return;
}
capture_.recommended_input_volume = capture_.applied_input_volume;
}
bool AudioProcessingImpl::CreateAndAttachAecDump(
absl::string_view file_name,
int64_t max_log_size_bytes,
absl::Nonnull<TaskQueueBase*> worker_queue) {
std::unique_ptr<AecDump> aec_dump =
AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
if (!aec_dump) {
return false;
}
AttachAecDump(std::move(aec_dump));
return true;
}
bool AudioProcessingImpl::CreateAndAttachAecDump(
FILE* handle,
int64_t max_log_size_bytes,
absl::Nonnull<TaskQueueBase*> worker_queue) {
std::unique_ptr<AecDump> aec_dump =
AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
if (!aec_dump) {
return false;
}
AttachAecDump(std::move(aec_dump));
return true;
}
void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
RTC_DCHECK(aec_dump);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_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;
{
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
aec_dump = std::move(aec_dump_);
}
}
AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
return config_;
}
bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
return submodule_states_.Update(
config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile,
!!submodules_.noise_suppressor, !!submodules_.gain_control,
!!submodules_.gain_controller2,
config_.pre_amplifier.enabled || config_.capture_level_adjustment.enabled,
capture_nonlocked_.echo_controller_enabled);
}
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::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;
absl::optional<EchoCanceller3Config> multichannel_config;
if (use_setup_specific_default_aec3_config_) {
multichannel_config = EchoCanceller3::CreateDefaultMultichannelConfig();
}
submodules_.echo_controller = std::make_unique<EchoCanceller3>(
config, multichannel_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_controller2.enabled &&
config_.gain_controller2.input_volume_controller.enabled &&
config_.gain_controller1.enabled &&
(config_.gain_controller1.mode ==
AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
config_.gain_controller1.analog_gain_controller.enabled)) {
RTC_LOG(LS_ERROR) << "APM configuration not valid: "
<< "Multiple input volume controllers enabled.";
}
if (!config_.gain_controller1.enabled) {
submodules_.agc_manager.reset();
submodules_.gain_control.reset();
return;
}
RTC_HISTOGRAM_BOOLEAN(
"WebRTC.Audio.GainController.Analog.Enabled",
config_.gain_controller1.analog_gain_controller.enabled);
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);
constexpr int kAnalogLevelMinimum = 0;
constexpr int kAnalogLevelMaximum = 255;
error = submodules_.gain_control->set_analog_level_limits(
kAnalogLevelMinimum, kAnalogLevelMaximum);
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())) {
int stream_analog_level = -1;
const bool re_creation = !!submodules_.agc_manager;
if (re_creation) {
stream_analog_level = submodules_.agc_manager->recommended_analog_level();
}
submodules_.agc_manager.reset(new AgcManagerDirect(
num_proc_channels(), config_.gain_controller1.analog_gain_controller));
if (re_creation) {
submodules_.agc_manager->set_stream_analog_level(stream_analog_level);
}
}
submodules_.agc_manager->Initialize();
submodules_.agc_manager->SetupDigitalGainControl(*submodules_.gain_control);
submodules_.agc_manager->HandleCaptureOutputUsedChange(
capture_.capture_output_used);
}
void AudioProcessingImpl::InitializeGainController2() {
if (!config_.gain_controller2.enabled) {
submodules_.gain_controller2.reset();
return;
}
// Input volume controller configuration if the AGC2 is running
// and its parameters require to fully switch the gain control to
// AGC2.
const InputVolumeController::Config input_volume_controller_config =
InputVolumeController::Config{};
submodules_.gain_controller2 = std::make_unique<GainController2>(
config_.gain_controller2, input_volume_controller_config,
proc_fullband_sample_rate_hz(), num_output_channels(),
/*use_internal_vad=*/true);
submodules_.gain_controller2->SetCaptureOutputUsed(
capture_.capture_output_used);
}
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;
}
RTC_CHECK_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::InitializeCaptureLevelsAdjuster() {
if (config_.pre_amplifier.enabled ||
config_.capture_level_adjustment.enabled) {
// Use both the pre-amplifier and the capture level adjustment gains as
// pre-gains.
float pre_gain = 1.f;
if (config_.pre_amplifier.enabled) {
pre_gain *= config_.pre_amplifier.fixed_gain_factor;
}
if (config_.capture_level_adjustment.enabled) {
pre_gain *= config_.capture_level_adjustment.pre_gain_factor;
}
submodules_.capture_levels_adjuster =
std::make_unique<CaptureLevelsAdjuster>(
config_.capture_level_adjustment.analog_mic_gain_emulation.enabled,
config_.capture_level_adjustment.analog_mic_gain_emulation
.initial_level,
pre_gain, config_.capture_level_adjustment.post_gain_factor);
} else {
submodules_.capture_levels_adjuster.reset();
}
}
void AudioProcessingImpl::InitializeResidualEchoDetector() {
if (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::WriteAecDumpConfigMessage(bool forced) {
if (!aec_dump_) {
return;
}
std::string experiments_description = "";
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
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.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.applied_input_volume = capture_.applied_input_volume;
audio_proc_state.keypress = capture_.key_pressed;
aec_dump_->AddAudioProcessingState(audio_proc_state);
}
AudioProcessingImpl::ApmCaptureState::ApmCaptureState()
: was_stream_delay_set(false),
capture_output_used(true),
capture_output_used_last_frame(true),
key_pressed(false),
capture_processing_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz),
echo_path_gain_change(false),
prev_pre_adjustment_gain(-1.0f),
playout_volume(-1),
prev_playout_volume(-1),
applied_input_volume_changed(false) {}
AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
AudioProcessingImpl::ApmStatsReporter::ApmStatsReporter()
: stats_message_queue_(1) {}
AudioProcessingImpl::ApmStatsReporter::~ApmStatsReporter() = default;
AudioProcessingStats AudioProcessingImpl::ApmStatsReporter::GetStatistics() {
MutexLock lock_stats(&mutex_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