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webrtc/modules/audio_processing/audio_processing_impl.cc
Per Åhgren 09e9a83d91 Change the way that AecDumps are created in APM 
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.

This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.

The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.

The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
   surface of APM.
2) Those files anyway needed to be moved to a separate build-
   target to avoid a circular build-file dependency caused by
   the other changes in this CL

Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
2020-05-11 10:33:00 +00:00

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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 <memory>
#include <string>
#include <type_traits>
#include <utility>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "common_audio/audio_converter.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#define RETURN_ON_ERR(expr) \
do { \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
} while (0)
namespace webrtc {
constexpr int kRuntimeSettingQueueSize = 100;
namespace {
static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
case AudioProcessing::kStereo:
return false;
case AudioProcessing::kMonoAndKeyboard:
case AudioProcessing::kStereoAndKeyboard:
return true;
}
RTC_NOTREACHED();
return false;
}
bool SampleRateSupportsMultiBand(int sample_rate_hz) {
return sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
sample_rate_hz == AudioProcessing::kSampleRate48kHz;
}
// Checks whether the high-pass filter should be done in the full-band.
bool EnforceSplitBandHpf() {
return field_trial::IsEnabled("WebRTC-FullBandHpfKillSwitch");
}
// Checks whether AEC3 should be allowed to decide what the default
// configuration should be based on the render and capture channel configuration
// at hand.
bool UseSetupSpecificDefaultAec3Congfig() {
return !field_trial::IsEnabled(
"WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch");
}
// Identify the native processing rate that best handles a sample rate.
int SuitableProcessRate(int minimum_rate,
int max_splitting_rate,
bool band_splitting_required) {
const int uppermost_native_rate =
band_splitting_required ? max_splitting_rate : 48000;
for (auto rate : {16000, 32000, 48000}) {
if (rate >= uppermost_native_rate) {
return uppermost_native_rate;
}
if (rate >= minimum_rate) {
return rate;
}
}
RTC_NOTREACHED();
return uppermost_native_rate;
}
GainControl::Mode Agc1ConfigModeToInterfaceMode(
AudioProcessing::Config::GainController1::Mode mode) {
using Agc1Config = AudioProcessing::Config::GainController1;
switch (mode) {
case Agc1Config::kAdaptiveAnalog:
return GainControl::kAdaptiveAnalog;
case Agc1Config::kAdaptiveDigital:
return GainControl::kAdaptiveDigital;
case Agc1Config::kFixedDigital:
return GainControl::kFixedDigital;
}
}
// Maximum lengths that frame of samples being passed from the render side to
// the capture side can have (does not apply to AEC3).
static const size_t kMaxAllowedValuesOfSamplesPerBand = 160;
static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480;
// Maximum number of frames to buffer in the render queue.
// TODO(peah): Decrease this once we properly handle hugely unbalanced
// reverse and forward call numbers.
static const size_t kMaxNumFramesToBuffer = 100;
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
AudioProcessingImpl::SubmoduleStates::SubmoduleStates(
bool capture_post_processor_enabled,
bool render_pre_processor_enabled,
bool capture_analyzer_enabled)
: capture_post_processor_enabled_(capture_post_processor_enabled),
render_pre_processor_enabled_(render_pre_processor_enabled),
capture_analyzer_enabled_(capture_analyzer_enabled) {}
bool AudioProcessingImpl::SubmoduleStates::Update(
bool high_pass_filter_enabled,
bool mobile_echo_controller_enabled,
bool residual_echo_detector_enabled,
bool noise_suppressor_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
bool pre_amplifier_enabled,
bool echo_controller_enabled,
bool voice_detector_enabled,
bool transient_suppressor_enabled) {
bool changed = false;
changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
changed |=
(mobile_echo_controller_enabled != mobile_echo_controller_enabled_);
changed |=
(residual_echo_detector_enabled != residual_echo_detector_enabled_);
changed |= (noise_suppressor_enabled != noise_suppressor_enabled_);
changed |=
(adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
changed |= (gain_controller2_enabled != gain_controller2_enabled_);
changed |= (pre_amplifier_enabled_ != pre_amplifier_enabled);
changed |= (echo_controller_enabled != echo_controller_enabled_);
changed |= (voice_detector_enabled != voice_detector_enabled_);
changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
if (changed) {
high_pass_filter_enabled_ = high_pass_filter_enabled;
mobile_echo_controller_enabled_ = mobile_echo_controller_enabled;
residual_echo_detector_enabled_ = residual_echo_detector_enabled;
noise_suppressor_enabled_ = noise_suppressor_enabled;
adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
gain_controller2_enabled_ = gain_controller2_enabled;
pre_amplifier_enabled_ = pre_amplifier_enabled;
echo_controller_enabled_ = echo_controller_enabled;
voice_detector_enabled_ = voice_detector_enabled;
transient_suppressor_enabled_ = transient_suppressor_enabled;
}
changed |= first_update_;
first_update_ = false;
return changed;
}
bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandSubModulesActive()
const {
return CaptureMultiBandProcessingPresent() || voice_detector_enabled_;
}
bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingPresent()
const {
// If echo controller is present, assume it performs active processing.
return CaptureMultiBandProcessingActive(/*ec_processing_active=*/true);
}
bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingActive(
bool ec_processing_active) const {
return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ ||
noise_suppressor_enabled_ || adaptive_gain_controller_enabled_ ||
(echo_controller_enabled_ && ec_processing_active);
}
bool AudioProcessingImpl::SubmoduleStates::CaptureFullBandProcessingActive()
const {
return gain_controller2_enabled_ || capture_post_processor_enabled_ ||
pre_amplifier_enabled_;
}
bool AudioProcessingImpl::SubmoduleStates::CaptureAnalyzerActive() const {
return capture_analyzer_enabled_;
}
bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandSubModulesActive()
const {
return RenderMultiBandProcessingActive() || mobile_echo_controller_enabled_ ||
adaptive_gain_controller_enabled_ || echo_controller_enabled_;
}
bool AudioProcessingImpl::SubmoduleStates::RenderFullBandProcessingActive()
const {
return render_pre_processor_enabled_;
}
bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandProcessingActive()
const {
return false;
}
bool AudioProcessingImpl::SubmoduleStates::HighPassFilteringRequired() const {
return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ ||
noise_suppressor_enabled_;
}
AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
: AudioProcessingImpl(config,
/*capture_post_processor=*/nullptr,
/*render_pre_processor=*/nullptr,
/*echo_control_factory=*/nullptr,
/*echo_detector=*/nullptr,
/*capture_analyzer=*/nullptr) {}
int AudioProcessingImpl::instance_count_ = 0;
AudioProcessingImpl::AudioProcessingImpl(
const webrtc::Config& config,
std::unique_ptr<CustomProcessing> capture_post_processor,
std::unique_ptr<CustomProcessing> render_pre_processor,
std::unique_ptr<EchoControlFactory> echo_control_factory,
rtc::scoped_refptr<EchoDetector> echo_detector,
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer)
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
use_setup_specific_default_aec3_config_(
UseSetupSpecificDefaultAec3Congfig()),
capture_runtime_settings_(kRuntimeSettingQueueSize),
render_runtime_settings_(kRuntimeSettingQueueSize),
capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
render_runtime_settings_enqueuer_(&render_runtime_settings_),
echo_control_factory_(std::move(echo_control_factory)),
submodule_states_(!!capture_post_processor,
!!render_pre_processor,
!!capture_analyzer),
submodules_(std::move(capture_post_processor),
std::move(render_pre_processor),
std::move(echo_detector),
std::move(capture_analyzer)),
constants_(!field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
!field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"),
EnforceSplitBandHpf()),
capture_nonlocked_() {
RTC_LOG(LS_INFO) << "Injected APM submodules:"
"\nEcho control factory: "
<< !!echo_control_factory_
<< "\nEcho detector: " << !!submodules_.echo_detector
<< "\nCapture analyzer: " << !!submodules_.capture_analyzer
<< "\nCapture post processor: "
<< !!submodules_.capture_post_processor
<< "\nRender pre processor: "
<< !!submodules_.render_pre_processor;
// Mark Echo Controller enabled if a factory is injected.
capture_nonlocked_.echo_controller_enabled =
static_cast<bool>(echo_control_factory_);
// If no echo detector is injected, use the ResidualEchoDetector.
if (!submodules_.echo_detector) {
submodules_.echo_detector =
new rtc::RefCountedObject<ResidualEchoDetector>();
}
#if !(defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS))
// TODO(webrtc:5298): Remove once the use of ExperimentalNs has been
// deprecated.
config_.transient_suppression.enabled = config.Get<ExperimentalNs>().enabled;
// TODO(webrtc:5298): Remove once the use of ExperimentalAgc has been
// deprecated.
config_.gain_controller1.analog_gain_controller.enabled =
config.Get<ExperimentalAgc>().enabled;
config_.gain_controller1.analog_gain_controller.startup_min_volume =
config.Get<ExperimentalAgc>().startup_min_volume;
config_.gain_controller1.analog_gain_controller.clipped_level_min =
config.Get<ExperimentalAgc>().clipped_level_min;
config_.gain_controller1.analog_gain_controller.enable_agc2_level_estimator =
config.Get<ExperimentalAgc>().enabled_agc2_level_estimator;
config_.gain_controller1.analog_gain_controller.enable_digital_adaptive =
!config.Get<ExperimentalAgc>().digital_adaptive_disabled;
#endif
}
AudioProcessingImpl::~AudioProcessingImpl() = default;
int AudioProcessingImpl::Initialize() {
// Run in a single-threaded manner during initialization.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
return InitializeLocked();
}
int AudioProcessingImpl::Initialize(int capture_input_sample_rate_hz,
int capture_output_sample_rate_hz,
int render_input_sample_rate_hz,
ChannelLayout capture_input_layout,
ChannelLayout capture_output_layout,
ChannelLayout render_input_layout) {
const ProcessingConfig processing_config = {
{{capture_input_sample_rate_hz, ChannelsFromLayout(capture_input_layout),
LayoutHasKeyboard(capture_input_layout)},
{capture_output_sample_rate_hz,
ChannelsFromLayout(capture_output_layout),
LayoutHasKeyboard(capture_output_layout)},
{render_input_sample_rate_hz, ChannelsFromLayout(render_input_layout),
LayoutHasKeyboard(render_input_layout)},
{render_input_sample_rate_hz, ChannelsFromLayout(render_input_layout),
LayoutHasKeyboard(render_input_layout)}}};
return Initialize(processing_config);
}
int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
// Run in a single-threaded manner during initialization.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
return InitializeLocked(processing_config);
}
int AudioProcessingImpl::MaybeInitializeRender(
const ProcessingConfig& processing_config) {
// Called from both threads. Thread check is therefore not possible.
if (processing_config == formats_.api_format) {
return kNoError;
}
rtc::CritScope cs_capture(&crit_capture_);
return InitializeLocked(processing_config);
}
int AudioProcessingImpl::InitializeLocked() {
UpdateActiveSubmoduleStates();
const int render_audiobuffer_sample_rate_hz =
formats_.api_format.reverse_output_stream().num_frames() == 0
? formats_.render_processing_format.sample_rate_hz()
: formats_.api_format.reverse_output_stream().sample_rate_hz();
if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
render_.render_audio.reset(new AudioBuffer(
formats_.api_format.reverse_input_stream().sample_rate_hz(),
formats_.api_format.reverse_input_stream().num_channels(),
formats_.render_processing_format.sample_rate_hz(),
formats_.render_processing_format.num_channels(),
render_audiobuffer_sample_rate_hz,
formats_.render_processing_format.num_channels()));
if (formats_.api_format.reverse_input_stream() !=
formats_.api_format.reverse_output_stream()) {
render_.render_converter = AudioConverter::Create(
formats_.api_format.reverse_input_stream().num_channels(),
formats_.api_format.reverse_input_stream().num_frames(),
formats_.api_format.reverse_output_stream().num_channels(),
formats_.api_format.reverse_output_stream().num_frames());
} else {
render_.render_converter.reset(nullptr);
}
} else {
render_.render_audio.reset(nullptr);
render_.render_converter.reset(nullptr);
}
capture_.capture_audio.reset(new AudioBuffer(
formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.input_stream().num_channels(),
capture_nonlocked_.capture_processing_format.sample_rate_hz(),
formats_.api_format.output_stream().num_channels(),
formats_.api_format.output_stream().sample_rate_hz(),
formats_.api_format.output_stream().num_channels()));
if (capture_nonlocked_.capture_processing_format.sample_rate_hz() <
formats_.api_format.output_stream().sample_rate_hz() &&
formats_.api_format.output_stream().sample_rate_hz() == 48000) {
capture_.capture_fullband_audio.reset(
new AudioBuffer(formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.input_stream().num_channels(),
formats_.api_format.output_stream().sample_rate_hz(),
formats_.api_format.output_stream().num_channels(),
formats_.api_format.output_stream().sample_rate_hz(),
formats_.api_format.output_stream().num_channels()));
} else {
capture_.capture_fullband_audio.reset();
}
AllocateRenderQueue();
InitializeGainController1();
InitializeTransientSuppressor();
InitializeHighPassFilter(true);
InitializeVoiceDetector();
InitializeResidualEchoDetector();
InitializeEchoController();
InitializeGainController2();
InitializeNoiseSuppressor();
InitializeAnalyzer();
InitializePostProcessor();
InitializePreProcessor();
if (aec_dump_) {
aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
}
return kNoError;
}
int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
UpdateActiveSubmoduleStates();
for (const auto& stream : config.streams) {
if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
return kBadSampleRateError;
}
}
const size_t num_in_channels = config.input_stream().num_channels();
const size_t num_out_channels = config.output_stream().num_channels();
// Need at least one input channel.
// Need either one output channel or as many outputs as there are inputs.
if (num_in_channels == 0 ||
!(num_out_channels == 1 || num_out_channels == num_in_channels)) {
return kBadNumberChannelsError;
}
formats_.api_format = config;
// Choose maximum rate to use for the split filtering.
RTC_DCHECK(config_.pipeline.maximum_internal_processing_rate == 48000 ||
config_.pipeline.maximum_internal_processing_rate == 32000);
int max_splitting_rate = 48000;
if (config_.pipeline.maximum_internal_processing_rate == 32000) {
max_splitting_rate = config_.pipeline.maximum_internal_processing_rate;
}
int capture_processing_rate = SuitableProcessRate(
std::min(formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.output_stream().sample_rate_hz()),
max_splitting_rate,
submodule_states_.CaptureMultiBandSubModulesActive() ||
submodule_states_.RenderMultiBandSubModulesActive());
RTC_DCHECK_NE(8000, capture_processing_rate);
capture_nonlocked_.capture_processing_format =
StreamConfig(capture_processing_rate);
int render_processing_rate;
if (!capture_nonlocked_.echo_controller_enabled) {
render_processing_rate = SuitableProcessRate(
std::min(formats_.api_format.reverse_input_stream().sample_rate_hz(),
formats_.api_format.reverse_output_stream().sample_rate_hz()),
max_splitting_rate,
submodule_states_.CaptureMultiBandSubModulesActive() ||
submodule_states_.RenderMultiBandSubModulesActive());
} else {
render_processing_rate = capture_processing_rate;
}
// If the forward sample rate is 8 kHz, the render stream is also processed
// at this rate.
if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
kSampleRate8kHz) {
render_processing_rate = kSampleRate8kHz;
} else {
render_processing_rate =
std::max(render_processing_rate, static_cast<int>(kSampleRate16kHz));
}
RTC_DCHECK_NE(8000, render_processing_rate);
if (submodule_states_.RenderMultiBandSubModulesActive()) {
// By default, downmix the render stream to mono for analysis. This has been
// demonstrated to work well for AEC in most practical scenarios.
const bool multi_channel_render = config_.pipeline.multi_channel_render &&
constants_.multi_channel_render_support;
int render_processing_num_channels =
multi_channel_render
? formats_.api_format.reverse_input_stream().num_channels()
: 1;
formats_.render_processing_format =
StreamConfig(render_processing_rate, render_processing_num_channels);
} else {
formats_.render_processing_format = StreamConfig(
formats_.api_format.reverse_input_stream().sample_rate_hz(),
formats_.api_format.reverse_input_stream().num_channels());
}
if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
kSampleRate32kHz ||
capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
kSampleRate48kHz) {
capture_nonlocked_.split_rate = kSampleRate16kHz;
} else {
capture_nonlocked_.split_rate =
capture_nonlocked_.capture_processing_format.sample_rate_hz();
}
return InitializeLocked();
}
void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
// Run in a single-threaded manner when applying the settings.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
const bool pipeline_config_changed =
config_.pipeline.multi_channel_render !=
config.pipeline.multi_channel_render ||
config_.pipeline.multi_channel_capture !=
config.pipeline.multi_channel_capture ||
config_.pipeline.maximum_internal_processing_rate !=
config.pipeline.maximum_internal_processing_rate;
const bool aec_config_changed =
config_.echo_canceller.enabled != config.echo_canceller.enabled ||
config_.echo_canceller.mobile_mode != config.echo_canceller.mobile_mode;
const bool agc1_config_changed =
config_.gain_controller1.enabled != config.gain_controller1.enabled ||
config_.gain_controller1.mode != config.gain_controller1.mode ||
config_.gain_controller1.target_level_dbfs !=
config.gain_controller1.target_level_dbfs ||
config_.gain_controller1.compression_gain_db !=
config.gain_controller1.compression_gain_db ||
config_.gain_controller1.enable_limiter !=
config.gain_controller1.enable_limiter ||
config_.gain_controller1.analog_level_minimum !=
config.gain_controller1.analog_level_minimum ||
config_.gain_controller1.analog_level_maximum !=
config.gain_controller1.analog_level_maximum ||
config_.gain_controller1.analog_gain_controller.enabled !=
config.gain_controller1.analog_gain_controller.enabled ||
config_.gain_controller1.analog_gain_controller.startup_min_volume !=
config.gain_controller1.analog_gain_controller.startup_min_volume ||
config_.gain_controller1.analog_gain_controller.clipped_level_min !=
config.gain_controller1.analog_gain_controller.clipped_level_min ||
config_.gain_controller1.analog_gain_controller
.enable_agc2_level_estimator !=
config.gain_controller1.analog_gain_controller
.enable_agc2_level_estimator ||
config_.gain_controller1.analog_gain_controller.enable_digital_adaptive !=
config.gain_controller1.analog_gain_controller
.enable_digital_adaptive;
const bool agc2_config_changed =
config_.gain_controller2.enabled != config.gain_controller2.enabled;
const bool voice_detection_config_changed =
config_.voice_detection.enabled != config.voice_detection.enabled;
const bool ns_config_changed =
config_.noise_suppression.enabled != config.noise_suppression.enabled ||
config_.noise_suppression.level != config.noise_suppression.level;
const bool ts_config_changed = config_.transient_suppression.enabled !=
config.transient_suppression.enabled;
const bool pre_amplifier_config_changed =
config_.pre_amplifier.enabled != config.pre_amplifier.enabled ||
config_.pre_amplifier.fixed_gain_factor !=
config.pre_amplifier.fixed_gain_factor;
config_ = config;
if (aec_config_changed) {
InitializeEchoController();
}
if (ns_config_changed) {
InitializeNoiseSuppressor();
}
if (ts_config_changed) {
InitializeTransientSuppressor();
}
InitializeHighPassFilter(false);
if (agc1_config_changed) {
InitializeGainController1();
}
const bool config_ok = GainController2::Validate(config_.gain_controller2);
if (!config_ok) {
RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
"Gain Controller 2: "
<< GainController2::ToString(config_.gain_controller2)
<< "\nReverting to default parameter set";
config_.gain_controller2 = AudioProcessing::Config::GainController2();
}
if (agc2_config_changed) {
InitializeGainController2();
}
if (pre_amplifier_config_changed) {
InitializePreAmplifier();
}
if (config_.level_estimation.enabled && !submodules_.output_level_estimator) {
submodules_.output_level_estimator = std::make_unique<LevelEstimator>();
}
if (voice_detection_config_changed) {
InitializeVoiceDetector();
}
// Reinitialization must happen after all submodule configuration to avoid
// additional reinitializations on the next capture / render processing call.
if (pipeline_config_changed) {
InitializeLocked(formats_.api_format);
}
}
// TODO(webrtc:5298): Remove.
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {}
void AudioProcessingImpl::OverrideSubmoduleCreationForTesting(
const ApmSubmoduleCreationOverrides& overrides) {
rtc::CritScope cs(&crit_capture_);
submodule_creation_overrides_ = overrides;
}
int AudioProcessingImpl::proc_sample_rate_hz() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.capture_processing_format.sample_rate_hz();
}
int AudioProcessingImpl::proc_fullband_sample_rate_hz() const {
return capture_.capture_fullband_audio
? capture_.capture_fullband_audio->num_frames() * 100
: capture_nonlocked_.capture_processing_format.sample_rate_hz();
}
int AudioProcessingImpl::proc_split_sample_rate_hz() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.split_rate;
}
size_t AudioProcessingImpl::num_reverse_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.render_processing_format.num_channels();
}
size_t AudioProcessingImpl::num_input_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.api_format.input_stream().num_channels();
}
size_t AudioProcessingImpl::num_proc_channels() const {
// Used as callback from submodules, hence locking is not allowed.
const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
constants_.multi_channel_capture_support;
if (capture_nonlocked_.echo_controller_enabled && !multi_channel_capture) {
return 1;
}
return num_output_channels();
}
size_t AudioProcessingImpl::num_output_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.api_format.output_stream().num_channels();
}
void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
rtc::CritScope cs(&crit_capture_);
capture_.output_will_be_muted = muted;
if (submodules_.agc_manager.get()) {
submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted);
}
}
void AudioProcessingImpl::SetRuntimeSetting(RuntimeSetting setting) {
switch (setting.type()) {
case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
render_runtime_settings_enqueuer_.Enqueue(setting);
return;
case RuntimeSetting::Type::kCapturePreGain:
case RuntimeSetting::Type::kCaptureCompressionGain:
case RuntimeSetting::Type::kCaptureFixedPostGain:
capture_runtime_settings_enqueuer_.Enqueue(setting);
return;
case RuntimeSetting::Type::kPlayoutVolumeChange:
capture_runtime_settings_enqueuer_.Enqueue(setting);
render_runtime_settings_enqueuer_.Enqueue(setting);
return;
case RuntimeSetting::Type::kNotSpecified:
RTC_NOTREACHED();
return;
}
// The language allows the enum to have a non-enumerator
// value. Check that this doesn't happen.
RTC_NOTREACHED();
}
AudioProcessingImpl::RuntimeSettingEnqueuer::RuntimeSettingEnqueuer(
SwapQueue<RuntimeSetting>* runtime_settings)
: runtime_settings_(*runtime_settings) {
RTC_DCHECK(runtime_settings);
}
AudioProcessingImpl::RuntimeSettingEnqueuer::~RuntimeSettingEnqueuer() =
default;
void AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
RuntimeSetting setting) {
size_t remaining_attempts = 10;
while (!runtime_settings_.Insert(&setting) && remaining_attempts-- > 0) {
RuntimeSetting setting_to_discard;
if (runtime_settings_.Remove(&setting_to_discard))
RTC_LOG(LS_ERROR)
<< "The runtime settings queue is full. Oldest setting discarded.";
}
if (remaining_attempts == 0)
RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
}
int AudioProcessingImpl::MaybeInitializeCapture(
const StreamConfig& input_config,
const StreamConfig& output_config) {
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Acquire the capture lock in order to access api_format. The lock is
// released immediately, as we may need to acquire the render lock as part
// of the conditional reinitialization.
rtc::CritScope cs_capture(&crit_capture_);
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
if (processing_config.input_stream() != input_config) {
processing_config.input_stream() = input_config;
reinitialization_required = true;
}
if (processing_config.output_stream() != output_config) {
processing_config.output_stream() = output_config;
reinitialization_required = true;
}
if (reinitialization_required) {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
return kNoError;
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
if (!src || !dest) {
return kNullPointerError;
}
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
if (aec_dump_) {
RecordUnprocessedCaptureStream(src);
}
capture_.keyboard_info.Extract(src, formats_.api_format.input_stream());
capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyFrom(
src, formats_.api_format.input_stream());
}
RETURN_ON_ERR(ProcessCaptureStreamLocked());
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyTo(formats_.api_format.output_stream(),
dest);
} else {
capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
}
if (aec_dump_) {
RecordProcessedCaptureStream(dest);
}
return kNoError;
}
void AudioProcessingImpl::HandleCaptureRuntimeSettings() {
RuntimeSetting setting;
while (capture_runtime_settings_.Remove(&setting)) {
if (aec_dump_) {
aec_dump_->WriteRuntimeSetting(setting);
}
switch (setting.type()) {
case RuntimeSetting::Type::kCapturePreGain:
if (config_.pre_amplifier.enabled) {
float value;
setting.GetFloat(&value);
config_.pre_amplifier.fixed_gain_factor = value;
submodules_.pre_amplifier->SetGainFactor(value);
}
// TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
break;
case RuntimeSetting::Type::kCaptureCompressionGain: {
if (!submodules_.agc_manager) {
float value;
setting.GetFloat(&value);
int int_value = static_cast<int>(value + .5f);
config_.gain_controller1.compression_gain_db = int_value;
if (submodules_.gain_control) {
int error =
submodules_.gain_control->set_compression_gain_db(int_value);
RTC_DCHECK_EQ(kNoError, error);
}
}
break;
}
case RuntimeSetting::Type::kCaptureFixedPostGain: {
if (submodules_.gain_controller2) {
float value;
setting.GetFloat(&value);
config_.gain_controller2.fixed_digital.gain_db = value;
submodules_.gain_controller2->ApplyConfig(config_.gain_controller2);
}
break;
}
case RuntimeSetting::Type::kPlayoutVolumeChange: {
int value;
setting.GetInt(&value);
capture_.playout_volume = value;
break;
}
case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
RTC_NOTREACHED();
break;
case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
RTC_NOTREACHED();
break;
case RuntimeSetting::Type::kNotSpecified:
RTC_NOTREACHED();
break;
}
}
}
void AudioProcessingImpl::HandleRenderRuntimeSettings() {
RuntimeSetting setting;
while (render_runtime_settings_.Remove(&setting)) {
if (aec_dump_) {
aec_dump_->WriteRuntimeSetting(setting);
}
switch (setting.type()) {
case RuntimeSetting::Type::kPlayoutAudioDeviceChange: // fall-through
case RuntimeSetting::Type::kPlayoutVolumeChange: // fall-through
case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
if (submodules_.render_pre_processor) {
submodules_.render_pre_processor->SetRuntimeSetting(setting);
}
break;
case RuntimeSetting::Type::kCapturePreGain: // fall-through
case RuntimeSetting::Type::kCaptureCompressionGain: // fall-through
case RuntimeSetting::Type::kCaptureFixedPostGain: // fall-through
case RuntimeSetting::Type::kNotSpecified:
RTC_NOTREACHED();
break;
}
}
}
void AudioProcessingImpl::QueueBandedRenderAudio(AudioBuffer* audio) {
RTC_DCHECK_GE(160, audio->num_frames_per_band());
if (submodules_.echo_control_mobile) {
EchoControlMobileImpl::PackRenderAudioBuffer(audio, num_output_channels(),
num_reverse_channels(),
&aecm_render_queue_buffer_);
RTC_DCHECK(aecm_render_signal_queue_);
// Insert the samples into the queue.
if (!aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_)) {
// The data queue is full and needs to be emptied.
EmptyQueuedRenderAudio();
// Retry the insert (should always work).
bool result =
aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_);
RTC_DCHECK(result);
}
}
if (!submodules_.agc_manager && submodules_.gain_control) {
GainControlImpl::PackRenderAudioBuffer(*audio, &agc_render_queue_buffer_);
// Insert the samples into the queue.
if (!agc_render_signal_queue_->Insert(&agc_render_queue_buffer_)) {
// The data queue is full and needs to be emptied.
EmptyQueuedRenderAudio();
// Retry the insert (should always work).
bool result = agc_render_signal_queue_->Insert(&agc_render_queue_buffer_);
RTC_DCHECK(result);
}
}
}
void AudioProcessingImpl::QueueNonbandedRenderAudio(AudioBuffer* audio) {
ResidualEchoDetector::PackRenderAudioBuffer(audio, &red_render_queue_buffer_);
// Insert the samples into the queue.
if (!red_render_signal_queue_->Insert(&red_render_queue_buffer_)) {
// The data queue is full and needs to be emptied.
EmptyQueuedRenderAudio();
// Retry the insert (should always work).
bool result = red_render_signal_queue_->Insert(&red_render_queue_buffer_);
RTC_DCHECK(result);
}
}
void AudioProcessingImpl::AllocateRenderQueue() {
const size_t new_agc_render_queue_element_max_size =
std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerBand);
const size_t new_red_render_queue_element_max_size =
std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerFrame);
// Reallocate the queues if the queue item sizes are too small to fit the
// data to put in the queues.
if (agc_render_queue_element_max_size_ <
new_agc_render_queue_element_max_size) {
agc_render_queue_element_max_size_ = new_agc_render_queue_element_max_size;
std::vector<int16_t> template_queue_element(
agc_render_queue_element_max_size_);
agc_render_signal_queue_.reset(
new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
kMaxNumFramesToBuffer, template_queue_element,
RenderQueueItemVerifier<int16_t>(
agc_render_queue_element_max_size_)));
agc_render_queue_buffer_.resize(agc_render_queue_element_max_size_);
agc_capture_queue_buffer_.resize(agc_render_queue_element_max_size_);
} else {
agc_render_signal_queue_->Clear();
}
if (red_render_queue_element_max_size_ <
new_red_render_queue_element_max_size) {
red_render_queue_element_max_size_ = new_red_render_queue_element_max_size;
std::vector<float> template_queue_element(
red_render_queue_element_max_size_);
red_render_signal_queue_.reset(
new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
kMaxNumFramesToBuffer, template_queue_element,
RenderQueueItemVerifier<float>(
red_render_queue_element_max_size_)));
red_render_queue_buffer_.resize(red_render_queue_element_max_size_);
red_capture_queue_buffer_.resize(red_render_queue_element_max_size_);
} else {
red_render_signal_queue_->Clear();
}
}
void AudioProcessingImpl::EmptyQueuedRenderAudio() {
rtc::CritScope cs_capture(&crit_capture_);
if (submodules_.echo_control_mobile) {
RTC_DCHECK(aecm_render_signal_queue_);
while (aecm_render_signal_queue_->Remove(&aecm_capture_queue_buffer_)) {
submodules_.echo_control_mobile->ProcessRenderAudio(
aecm_capture_queue_buffer_);
}
}
if (submodules_.gain_control) {
while (agc_render_signal_queue_->Remove(&agc_capture_queue_buffer_)) {
submodules_.gain_control->ProcessRenderAudio(agc_capture_queue_buffer_);
}
}
while (red_render_signal_queue_->Remove(&red_capture_queue_buffer_)) {
RTC_DCHECK(submodules_.echo_detector);
submodules_.echo_detector->AnalyzeRenderAudio(red_capture_queue_buffer_);
}
}
int AudioProcessingImpl::ProcessStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
if (aec_dump_) {
RecordUnprocessedCaptureStream(src, input_config);
}
capture_.capture_audio->CopyFrom(src, input_config);
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyFrom(src, input_config);
}
RETURN_ON_ERR(ProcessCaptureStreamLocked());
if (submodule_states_.CaptureMultiBandProcessingPresent() ||
submodule_states_.CaptureFullBandProcessingActive()) {
if (capture_.capture_fullband_audio) {
capture_.capture_fullband_audio->CopyTo(output_config, dest);
} else {
capture_.capture_audio->CopyTo(output_config, dest);
}
}
if (aec_dump_) {
RecordProcessedCaptureStream(dest, output_config);
}
return kNoError;
}
int AudioProcessingImpl::ProcessCaptureStreamLocked() {
EmptyQueuedRenderAudio();
HandleCaptureRuntimeSettings();
// Ensure that not both the AEC and AECM are active at the same time.
// TODO(peah): Simplify once the public API Enable functions for these
// are moved to APM.
RTC_DCHECK_LE(
!!submodules_.echo_controller + !!submodules_.echo_control_mobile, 1);
AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity.
AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
if (submodules_.high_pass_filter &&
config_.high_pass_filter.apply_in_full_band &&
!constants_.enforce_split_band_hpf) {
submodules_.high_pass_filter->Process(capture_buffer,
/*use_split_band_data=*/false);
}
if (submodules_.pre_amplifier) {
submodules_.pre_amplifier->ApplyGain(AudioFrameView<float>(
capture_buffer->channels(), capture_buffer->num_channels(),
capture_buffer->num_frames()));
}
capture_input_rms_.Analyze(rtc::ArrayView<const float>(
capture_buffer->channels_const()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
if (log_rms) {
capture_rms_interval_counter_ = 0;
RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
levels.average, 1, RmsLevel::kMinLevelDb, 64);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
levels.peak, 1, RmsLevel::kMinLevelDb, 64);
}
if (submodules_.echo_controller) {
// Detect and flag any change in the analog gain.
int analog_mic_level = recommended_stream_analog_level();
capture_.echo_path_gain_change =
capture_.prev_analog_mic_level != analog_mic_level &&
capture_.prev_analog_mic_level != -1;
capture_.prev_analog_mic_level = analog_mic_level;
// Detect and flag any change in the pre-amplifier gain.
if (submodules_.pre_amplifier) {
float pre_amp_gain = submodules_.pre_amplifier->GetGainFactor();
capture_.echo_path_gain_change =
capture_.echo_path_gain_change ||
(capture_.prev_pre_amp_gain != pre_amp_gain &&
capture_.prev_pre_amp_gain >= 0.f);
capture_.prev_pre_amp_gain = pre_amp_gain;
}
// Detect volume change.
capture_.echo_path_gain_change =
capture_.echo_path_gain_change ||
(capture_.prev_playout_volume != capture_.playout_volume &&
capture_.prev_playout_volume >= 0);
capture_.prev_playout_volume = capture_.playout_volume;
submodules_.echo_controller->AnalyzeCapture(capture_buffer);
}
if (submodules_.agc_manager) {
submodules_.agc_manager->AnalyzePreProcess(capture_buffer);
}
if (submodule_states_.CaptureMultiBandSubModulesActive() &&
SampleRateSupportsMultiBand(
capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
capture_buffer->SplitIntoFrequencyBands();
}
const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
constants_.multi_channel_capture_support;
if (submodules_.echo_controller && !multi_channel_capture) {
// Force down-mixing of the number of channels after the detection of
// capture signal saturation.
// TODO(peah): Look into ensuring that this kind of tampering with the
// AudioBuffer functionality should not be needed.
capture_buffer->set_num_channels(1);
}
if (submodules_.high_pass_filter &&
(!config_.high_pass_filter.apply_in_full_band ||
constants_.enforce_split_band_hpf)) {
submodules_.high_pass_filter->Process(capture_buffer,
/*use_split_band_data=*/true);
}
if (submodules_.gain_control) {
RETURN_ON_ERR(
submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
}
if ((!config_.noise_suppression.analyze_linear_aec_output_when_available ||
!linear_aec_buffer || submodules_.echo_control_mobile) &&
submodules_.noise_suppressor) {
submodules_.noise_suppressor->Analyze(*capture_buffer);
}
if (submodules_.echo_control_mobile) {
// Ensure that the stream delay was set before the call to the
// AECM ProcessCaptureAudio function.
if (!capture_.was_stream_delay_set) {
return AudioProcessing::kStreamParameterNotSetError;
}
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->Process(capture_buffer);
}
RETURN_ON_ERR(submodules_.echo_control_mobile->ProcessCaptureAudio(
capture_buffer, stream_delay_ms()));
} else {
if (submodules_.echo_controller) {
data_dumper_->DumpRaw("stream_delay", stream_delay_ms());
if (capture_.was_stream_delay_set) {
submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms());
}
submodules_.echo_controller->ProcessCapture(
capture_buffer, linear_aec_buffer, capture_.echo_path_gain_change);
}
if (config_.noise_suppression.analyze_linear_aec_output_when_available &&
linear_aec_buffer && submodules_.noise_suppressor) {
submodules_.noise_suppressor->Analyze(*linear_aec_buffer);
}
if (submodules_.noise_suppressor) {
submodules_.noise_suppressor->Process(capture_buffer);
}
}
if (config_.voice_detection.enabled) {
capture_.stats.voice_detected =
submodules_.voice_detector->ProcessCaptureAudio(capture_buffer);
} else {
capture_.stats.voice_detected = absl::nullopt;
}
if (submodules_.agc_manager) {
submodules_.agc_manager->Process(capture_buffer);
absl::optional<int> new_digital_gain =
submodules_.agc_manager->GetDigitalComressionGain();
if (new_digital_gain && submodules_.gain_control) {
submodules_.gain_control->set_compression_gain_db(*new_digital_gain);
}
}
if (submodules_.gain_control) {
// TODO(peah): Add reporting from AEC3 whether there is echo.
RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio(
capture_buffer, /*stream_has_echo*/ false));
}
if (submodule_states_.CaptureMultiBandProcessingPresent() &&
SampleRateSupportsMultiBand(
capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
capture_buffer->MergeFrequencyBands();
}
if (capture_.capture_fullband_audio) {
const auto& ec = submodules_.echo_controller;
bool ec_active = ec ? ec->ActiveProcessing() : false;
// Only update the fullband buffer if the multiband processing has changed
// the signal. Keep the original signal otherwise.
if (submodule_states_.CaptureMultiBandProcessingActive(ec_active)) {
capture_buffer->CopyTo(capture_.capture_fullband_audio.get());
}
capture_buffer = capture_.capture_fullband_audio.get();
}
if (config_.residual_echo_detector.enabled) {
RTC_DCHECK(submodules_.echo_detector);
submodules_.echo_detector->AnalyzeCaptureAudio(rtc::ArrayView<const float>(
capture_buffer->channels()[0], capture_buffer->num_frames()));
}
// TODO(aluebs): Investigate if the transient suppression placement should be
// before or after the AGC.
if (submodules_.transient_suppressor) {
float voice_probability = submodules_.agc_manager.get()
? submodules_.agc_manager->voice_probability()
: 1.f;
submodules_.transient_suppressor->Suppress(
capture_buffer->channels()[0], capture_buffer->num_frames(),
capture_buffer->num_channels(),
capture_buffer->split_bands_const(0)[kBand0To8kHz],
capture_buffer->num_frames_per_band(),
capture_.keyboard_info.keyboard_data,
capture_.keyboard_info.num_keyboard_frames, voice_probability,
capture_.key_pressed);
}
// Experimental APM sub-module that analyzes |capture_buffer|.
if (submodules_.capture_analyzer) {
submodules_.capture_analyzer->Analyze(capture_buffer);
}
if (submodules_.gain_controller2) {
submodules_.gain_controller2->NotifyAnalogLevel(
recommended_stream_analog_level());
submodules_.gain_controller2->Process(capture_buffer);
}
if (submodules_.capture_post_processor) {
submodules_.capture_post_processor->Process(capture_buffer);
}
// The level estimator operates on the recombined data.
if (config_.level_estimation.enabled) {
submodules_.output_level_estimator->ProcessStream(*capture_buffer);
capture_.stats.output_rms_dbfs = submodules_.output_level_estimator->RMS();
} else {
capture_.stats.output_rms_dbfs = absl::nullopt;
}
capture_output_rms_.Analyze(rtc::ArrayView<const float>(
capture_buffer->channels_const()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
if (log_rms) {
RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelAverageRms",
levels.average, 1, RmsLevel::kMinLevelDb, 64);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelPeakRms",
levels.peak, 1, RmsLevel::kMinLevelDb, 64);
}
if (submodules_.agc_manager) {
int level = recommended_stream_analog_level();
data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1,
&level);
}
// Compute echo-related stats.
if (submodules_.echo_controller) {
auto ec_metrics = submodules_.echo_controller->GetMetrics();
capture_.stats.echo_return_loss = ec_metrics.echo_return_loss;
capture_.stats.echo_return_loss_enhancement =
ec_metrics.echo_return_loss_enhancement;
capture_.stats.delay_ms = ec_metrics.delay_ms;
}
if (config_.residual_echo_detector.enabled) {
RTC_DCHECK(submodules_.echo_detector);
auto ed_metrics = submodules_.echo_detector->GetMetrics();
capture_.stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
capture_.stats.residual_echo_likelihood_recent_max =
ed_metrics.echo_likelihood_recent_max;
}
// Pass stats for reporting.
stats_reporter_.UpdateStatistics(capture_.stats);
capture_.was_stream_delay_set = false;
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(
const float* const* data,
const StreamConfig& reverse_config) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_StreamConfig");
rtc::CritScope cs(&crit_render_);
return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
}
int AudioProcessingImpl::ProcessReverseStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
rtc::CritScope cs(&crit_render_);
RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, input_config, output_config));
if (submodule_states_.RenderMultiBandProcessingActive() ||
submodule_states_.RenderFullBandProcessingActive()) {
render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
dest);
} else if (formats_.api_format.reverse_input_stream() !=
formats_.api_format.reverse_output_stream()) {
render_.render_converter->Convert(src, input_config.num_samples(), dest,
output_config.num_samples());
} else {
CopyAudioIfNeeded(src, input_config.num_frames(),
input_config.num_channels(), dest);
}
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStreamLocked(
const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config) {
if (src == nullptr) {
return kNullPointerError;
}
if (input_config.num_channels() == 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream() = input_config;
processing_config.reverse_output_stream() = output_config;
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
RTC_DCHECK_EQ(input_config.num_frames(),
formats_.api_format.reverse_input_stream().num_frames());
if (aec_dump_) {
const size_t channel_size =
formats_.api_format.reverse_input_stream().num_frames();
const size_t num_channels =
formats_.api_format.reverse_input_stream().num_channels();
aec_dump_->WriteRenderStreamMessage(
AudioFrameView<const float>(src, num_channels, channel_size));
}
render_.render_audio->CopyFrom(src,
formats_.api_format.reverse_input_stream());
return ProcessRenderStreamLocked();
}
int AudioProcessingImpl::ProcessReverseStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
if (input_config.num_channels() <= 0) {
return AudioProcessing::Error::kBadNumberChannelsError;
}
rtc::CritScope cs(&crit_render_);
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream().set_sample_rate_hz(
input_config.sample_rate_hz());
processing_config.reverse_input_stream().set_num_channels(
input_config.num_channels());
processing_config.reverse_output_stream().set_sample_rate_hz(
output_config.sample_rate_hz());
processing_config.reverse_output_stream().set_num_channels(
output_config.num_channels());
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
if (input_config.num_frames() !=
formats_.api_format.reverse_input_stream().num_frames()) {
return kBadDataLengthError;
}
if (aec_dump_) {
aec_dump_->WriteRenderStreamMessage(src, input_config.num_frames(),
input_config.num_channels());
}
render_.render_audio->CopyFrom(src, input_config);
RETURN_ON_ERR(ProcessRenderStreamLocked());
if (submodule_states_.RenderMultiBandProcessingActive() ||
submodule_states_.RenderFullBandProcessingActive()) {
render_.render_audio->CopyTo(output_config, dest);
}
return kNoError;
}
int AudioProcessingImpl::ProcessRenderStreamLocked() {
AudioBuffer* render_buffer = render_.render_audio.get(); // For brevity.
HandleRenderRuntimeSettings();
if (submodules_.render_pre_processor) {
submodules_.render_pre_processor->Process(render_buffer);
}
QueueNonbandedRenderAudio(render_buffer);
if (submodule_states_.RenderMultiBandSubModulesActive() &&
SampleRateSupportsMultiBand(
formats_.render_processing_format.sample_rate_hz())) {
render_buffer->SplitIntoFrequencyBands();
}
if (submodule_states_.RenderMultiBandSubModulesActive()) {
QueueBandedRenderAudio(render_buffer);
}
// TODO(peah): Perform the queuing inside QueueRenderAudiuo().
if (submodules_.echo_controller) {
submodules_.echo_controller->AnalyzeRender(render_buffer);
}
if (submodule_states_.RenderMultiBandProcessingActive() &&
SampleRateSupportsMultiBand(
formats_.render_processing_format.sample_rate_hz())) {
render_buffer->MergeFrequencyBands();
}
return kNoError;
}
int AudioProcessingImpl::set_stream_delay_ms(int delay) {
rtc::CritScope cs(&crit_capture_);
Error retval = kNoError;
capture_.was_stream_delay_set = true;
if (delay < 0) {
delay = 0;
retval = kBadStreamParameterWarning;
}
// TODO(ajm): the max is rather arbitrarily chosen; investigate.
if (delay > 500) {
delay = 500;
retval = kBadStreamParameterWarning;
}
capture_nonlocked_.stream_delay_ms = delay;
return retval;
}
bool AudioProcessingImpl::GetLinearAecOutput(
rtc::ArrayView<std::array<float, 160>> linear_output) const {
rtc::CritScope cs(&crit_capture_);
AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
RTC_DCHECK(linear_aec_buffer);
if (linear_aec_buffer) {
RTC_DCHECK_EQ(1, linear_aec_buffer->num_bands());
RTC_DCHECK_EQ(linear_output.size(), linear_aec_buffer->num_channels());
for (size_t ch = 0; ch < linear_aec_buffer->num_channels(); ++ch) {
RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames());
rtc::ArrayView<const float> channel_view =
rtc::ArrayView<const float>(linear_aec_buffer->channels_const()[ch],
linear_aec_buffer->num_frames());
std::copy(channel_view.begin(), channel_view.end(),
linear_output[ch].begin());
}
return true;
}
RTC_LOG(LS_ERROR) << "No linear AEC output available";
RTC_NOTREACHED();
return false;
}
int AudioProcessingImpl::stream_delay_ms() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.stream_delay_ms;
}
void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
rtc::CritScope cs(&crit_capture_);
capture_.key_pressed = key_pressed;
}
void AudioProcessingImpl::set_stream_analog_level(int level) {
rtc::CritScope cs_capture(&crit_capture_);
if (submodules_.agc_manager) {
submodules_.agc_manager->set_stream_analog_level(level);
data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level",
1, &level);
} else if (submodules_.gain_control) {
int error = submodules_.gain_control->set_stream_analog_level(level);
RTC_DCHECK_EQ(kNoError, error);
} else {
capture_.cached_stream_analog_level_ = level;
}
}
int AudioProcessingImpl::recommended_stream_analog_level() const {
rtc::CritScope cs_capture(&crit_capture_);
if (submodules_.agc_manager) {
return submodules_.agc_manager->stream_analog_level();
} else if (submodules_.gain_control) {
return submodules_.gain_control->stream_analog_level();
} else {
return capture_.cached_stream_analog_level_;
}
}
bool AudioProcessingImpl::CreateAndAttachAecDump(const std::string& file_name,
int64_t max_log_size_bytes,
rtc::TaskQueue* 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,
rtc::TaskQueue* 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);
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::MutateConfig(
rtc::FunctionView<void(AudioProcessing::Config*)> mutator) {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
mutator(&config_);
ApplyConfig(config_);
}
AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
return config_;
}
bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
return submodule_states_.Update(
config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile,
config_.residual_echo_detector.enabled, !!submodules_.noise_suppressor,
!!submodules_.gain_control, !!submodules_.gain_controller2,
config_.pre_amplifier.enabled, capture_nonlocked_.echo_controller_enabled,
config_.voice_detection.enabled, !!submodules_.transient_suppressor);
}
void AudioProcessingImpl::InitializeTransientSuppressor() {
if (config_.transient_suppression.enabled) {
// Attempt to create a transient suppressor, if one is not already created.
if (!submodules_.transient_suppressor) {
submodules_.transient_suppressor =
CreateTransientSuppressor(submodule_creation_overrides_);
}
if (submodules_.transient_suppressor) {
submodules_.transient_suppressor->Initialize(
proc_fullband_sample_rate_hz(), capture_nonlocked_.split_rate,
num_proc_channels());
} else {
RTC_LOG(LS_WARNING)
<< "No transient suppressor created (probably disabled)";
}
} else {
submodules_.transient_suppressor.reset();
}
}
void AudioProcessingImpl::InitializeHighPassFilter(bool forced_reset) {
bool high_pass_filter_needed_by_aec =
config_.echo_canceller.enabled &&
config_.echo_canceller.enforce_high_pass_filtering &&
!config_.echo_canceller.mobile_mode;
if (submodule_states_.HighPassFilteringRequired() ||
high_pass_filter_needed_by_aec) {
bool use_full_band = config_.high_pass_filter.apply_in_full_band &&
!constants_.enforce_split_band_hpf;
int rate = use_full_band ? proc_fullband_sample_rate_hz()
: proc_split_sample_rate_hz();
size_t num_channels =
use_full_band ? num_output_channels() : num_proc_channels();
if (!submodules_.high_pass_filter ||
rate != submodules_.high_pass_filter->sample_rate_hz() ||
forced_reset ||
num_channels != submodules_.high_pass_filter->num_channels()) {
submodules_.high_pass_filter.reset(
new HighPassFilter(rate, num_channels));
}
} else {
submodules_.high_pass_filter.reset();
}
}
void AudioProcessingImpl::InitializeVoiceDetector() {
if (config_.voice_detection.enabled) {
submodules_.voice_detector = std::make_unique<VoiceDetection>(
proc_split_sample_rate_hz(), VoiceDetection::kVeryLowLikelihood);
} else {
submodules_.voice_detector.reset();
}
}
void AudioProcessingImpl::InitializeEchoController() {
bool use_echo_controller =
echo_control_factory_ ||
(config_.echo_canceller.enabled && !config_.echo_canceller.mobile_mode);
if (use_echo_controller) {
// Create and activate the echo controller.
if (echo_control_factory_) {
submodules_.echo_controller = echo_control_factory_->Create(
proc_sample_rate_hz(), num_reverse_channels(), num_proc_channels());
RTC_DCHECK(submodules_.echo_controller);
} else {
EchoCanceller3Config config =
use_setup_specific_default_aec3_config_
? EchoCanceller3::CreateDefaultConfig(num_reverse_channels(),
num_proc_channels())
: EchoCanceller3Config();
submodules_.echo_controller = std::make_unique<EchoCanceller3>(
config, proc_sample_rate_hz(), num_reverse_channels(),
num_proc_channels());
}
// Setup the storage for returning the linear AEC output.
if (config_.echo_canceller.export_linear_aec_output) {
constexpr int kLinearOutputRateHz = 16000;
capture_.linear_aec_output = std::make_unique<AudioBuffer>(
kLinearOutputRateHz, num_proc_channels(), kLinearOutputRateHz,
num_proc_channels(), kLinearOutputRateHz, num_proc_channels());
} else {
capture_.linear_aec_output.reset();
}
capture_nonlocked_.echo_controller_enabled = true;
submodules_.echo_control_mobile.reset();
aecm_render_signal_queue_.reset();
return;
}
submodules_.echo_controller.reset();
capture_nonlocked_.echo_controller_enabled = false;
capture_.linear_aec_output.reset();
if (!config_.echo_canceller.enabled) {
submodules_.echo_control_mobile.reset();
aecm_render_signal_queue_.reset();
return;
}
if (config_.echo_canceller.mobile_mode) {
// Create and activate AECM.
size_t max_element_size =
std::max(static_cast<size_t>(1),
kMaxAllowedValuesOfSamplesPerBand *
EchoControlMobileImpl::NumCancellersRequired(
num_output_channels(), num_reverse_channels()));
std::vector<int16_t> template_queue_element(max_element_size);
aecm_render_signal_queue_.reset(
new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
kMaxNumFramesToBuffer, template_queue_element,
RenderQueueItemVerifier<int16_t>(max_element_size)));
aecm_render_queue_buffer_.resize(max_element_size);
aecm_capture_queue_buffer_.resize(max_element_size);
submodules_.echo_control_mobile.reset(new EchoControlMobileImpl());
submodules_.echo_control_mobile->Initialize(proc_split_sample_rate_hz(),
num_reverse_channels(),
num_output_channels());
return;
}
submodules_.echo_control_mobile.reset();
aecm_render_signal_queue_.reset();
}
void AudioProcessingImpl::InitializeGainController1() {
if (!config_.gain_controller1.enabled) {
submodules_.agc_manager.reset();
submodules_.gain_control.reset();
return;
}
if (!submodules_.gain_control) {
submodules_.gain_control.reset(new GainControlImpl());
}
submodules_.gain_control->Initialize(num_proc_channels(),
proc_sample_rate_hz());
if (!config_.gain_controller1.analog_gain_controller.enabled) {
int error = submodules_.gain_control->set_mode(
Agc1ConfigModeToInterfaceMode(config_.gain_controller1.mode));
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_target_level_dbfs(
config_.gain_controller1.target_level_dbfs);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_compression_gain_db(
config_.gain_controller1.compression_gain_db);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->enable_limiter(
config_.gain_controller1.enable_limiter);
RTC_DCHECK_EQ(kNoError, error);
error = submodules_.gain_control->set_analog_level_limits(
config_.gain_controller1.analog_level_minimum,
config_.gain_controller1.analog_level_maximum);
RTC_DCHECK_EQ(kNoError, error);
submodules_.agc_manager.reset();
return;
}
if (!submodules_.agc_manager.get() ||
submodules_.agc_manager->num_channels() !=
static_cast<int>(num_proc_channels()) ||
submodules_.agc_manager->sample_rate_hz() !=
capture_nonlocked_.split_rate) {
int stream_analog_level = -1;
const bool re_creation = !!submodules_.agc_manager;
if (re_creation) {
stream_analog_level = submodules_.agc_manager->stream_analog_level();
}
submodules_.agc_manager.reset(new AgcManagerDirect(
num_proc_channels(),
config_.gain_controller1.analog_gain_controller.startup_min_volume,
config_.gain_controller1.analog_gain_controller.clipped_level_min,
config_.gain_controller1.analog_gain_controller
.enable_agc2_level_estimator,
!config_.gain_controller1.analog_gain_controller
.enable_digital_adaptive,
capture_nonlocked_.split_rate));
if (re_creation) {
submodules_.agc_manager->set_stream_analog_level(stream_analog_level);
}
}
submodules_.agc_manager->Initialize();
submodules_.agc_manager->SetupDigitalGainControl(
submodules_.gain_control.get());
submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted);
}
void AudioProcessingImpl::InitializeGainController2() {
if (config_.gain_controller2.enabled) {
if (!submodules_.gain_controller2) {
// TODO(alessiob): Move the injected gain controller once injection is
// implemented.
submodules_.gain_controller2.reset(new GainController2());
}
submodules_.gain_controller2->Initialize(proc_fullband_sample_rate_hz());
submodules_.gain_controller2->ApplyConfig(config_.gain_controller2);
} else {
submodules_.gain_controller2.reset();
}
}
void AudioProcessingImpl::InitializeNoiseSuppressor() {
submodules_.noise_suppressor.reset();
if (config_.noise_suppression.enabled) {
auto map_level =
[](AudioProcessing::Config::NoiseSuppression::Level level) {
using NoiseSuppresionConfig =
AudioProcessing::Config::NoiseSuppression;
switch (level) {
case NoiseSuppresionConfig::kLow:
return NsConfig::SuppressionLevel::k6dB;
case NoiseSuppresionConfig::kModerate:
return NsConfig::SuppressionLevel::k12dB;
case NoiseSuppresionConfig::kHigh:
return NsConfig::SuppressionLevel::k18dB;
case NoiseSuppresionConfig::kVeryHigh:
return NsConfig::SuppressionLevel::k21dB;
default:
RTC_NOTREACHED();
}
};
NsConfig cfg;
cfg.target_level = map_level(config_.noise_suppression.level);
submodules_.noise_suppressor = std::make_unique<NoiseSuppressor>(
cfg, proc_sample_rate_hz(), num_proc_channels());
}
}
void AudioProcessingImpl::InitializePreAmplifier() {
if (config_.pre_amplifier.enabled) {
submodules_.pre_amplifier.reset(
new GainApplier(true, config_.pre_amplifier.fixed_gain_factor));
} else {
submodules_.pre_amplifier.reset();
}
}
void AudioProcessingImpl::InitializeResidualEchoDetector() {
RTC_DCHECK(submodules_.echo_detector);
submodules_.echo_detector->Initialize(
proc_fullband_sample_rate_hz(), 1,
formats_.render_processing_format.sample_rate_hz(), 1);
}
void AudioProcessingImpl::InitializeAnalyzer() {
if (submodules_.capture_analyzer) {
submodules_.capture_analyzer->Initialize(proc_fullband_sample_rate_hz(),
num_proc_channels());
}
}
void AudioProcessingImpl::InitializePostProcessor() {
if (submodules_.capture_post_processor) {
submodules_.capture_post_processor->Initialize(
proc_fullband_sample_rate_hz(), num_proc_channels());
}
}
void AudioProcessingImpl::InitializePreProcessor() {
if (submodules_.render_pre_processor) {
submodules_.render_pre_processor->Initialize(
formats_.render_processing_format.sample_rate_hz(),
formats_.render_processing_format.num_channels());
}
}
void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
if (!aec_dump_) {
return;
}
std::string experiments_description = "";
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
if (config_.gain_controller1.analog_gain_controller.clipped_level_min !=
kClippedLevelMin) {
experiments_description += "AgcClippingLevelExperiment;";
}
if (!!submodules_.capture_post_processor) {
experiments_description += "CapturePostProcessor;";
}
if (!!submodules_.render_pre_processor) {
experiments_description += "RenderPreProcessor;";
}
if (capture_nonlocked_.echo_controller_enabled) {
experiments_description += "EchoController;";
}
if (config_.gain_controller2.enabled) {
experiments_description += "GainController2;";
}
InternalAPMConfig apm_config;
apm_config.aec_enabled = config_.echo_canceller.enabled;
apm_config.aec_delay_agnostic_enabled = false;
apm_config.aec_extended_filter_enabled = false;
apm_config.aec_suppression_level = 0;
apm_config.aecm_enabled = !!submodules_.echo_control_mobile;
apm_config.aecm_comfort_noise_enabled =
submodules_.echo_control_mobile &&
submodules_.echo_control_mobile->is_comfort_noise_enabled();
apm_config.aecm_routing_mode =
submodules_.echo_control_mobile
? static_cast<int>(submodules_.echo_control_mobile->routing_mode())
: 0;
apm_config.agc_enabled = !!submodules_.gain_control;
apm_config.agc_mode = submodules_.gain_control
? static_cast<int>(submodules_.gain_control->mode())
: GainControl::kAdaptiveAnalog;
apm_config.agc_limiter_enabled =
submodules_.gain_control ? submodules_.gain_control->is_limiter_enabled()
: false;
apm_config.noise_robust_agc_enabled = !!submodules_.agc_manager;
apm_config.hpf_enabled = config_.high_pass_filter.enabled;
apm_config.ns_enabled = config_.noise_suppression.enabled;
apm_config.ns_level = static_cast<int>(config_.noise_suppression.level);
apm_config.transient_suppression_enabled =
config_.transient_suppression.enabled;
apm_config.experiments_description = experiments_description;
apm_config.pre_amplifier_enabled = config_.pre_amplifier.enabled;
apm_config.pre_amplifier_fixed_gain_factor =
config_.pre_amplifier.fixed_gain_factor;
if (!forced && apm_config == apm_config_for_aec_dump_) {
return;
}
aec_dump_->WriteConfig(apm_config);
apm_config_for_aec_dump_ = apm_config;
}
void AudioProcessingImpl::RecordUnprocessedCaptureStream(
const float* const* src) {
RTC_DCHECK(aec_dump_);
WriteAecDumpConfigMessage(false);
const size_t channel_size = formats_.api_format.input_stream().num_frames();
const size_t num_channels = formats_.api_format.input_stream().num_channels();
aec_dump_->AddCaptureStreamInput(
AudioFrameView<const float>(src, num_channels, channel_size));
RecordAudioProcessingState();
}
void AudioProcessingImpl::RecordUnprocessedCaptureStream(
const int16_t* const data,
const StreamConfig& config) {
RTC_DCHECK(aec_dump_);
WriteAecDumpConfigMessage(false);
aec_dump_->AddCaptureStreamInput(data, config.num_channels(),
config.num_frames());
RecordAudioProcessingState();
}
void AudioProcessingImpl::RecordProcessedCaptureStream(
const float* const* processed_capture_stream) {
RTC_DCHECK(aec_dump_);
const size_t channel_size = formats_.api_format.output_stream().num_frames();
const size_t num_channels =
formats_.api_format.output_stream().num_channels();
aec_dump_->AddCaptureStreamOutput(AudioFrameView<const float>(
processed_capture_stream, num_channels, channel_size));
aec_dump_->WriteCaptureStreamMessage();
}
void AudioProcessingImpl::RecordProcessedCaptureStream(
const int16_t* const data,
const StreamConfig& config) {
RTC_DCHECK(aec_dump_);
aec_dump_->AddCaptureStreamOutput(data, config.num_channels(),
config.num_frames());
aec_dump_->WriteCaptureStreamMessage();
}
void AudioProcessingImpl::RecordAudioProcessingState() {
RTC_DCHECK(aec_dump_);
AecDump::AudioProcessingState audio_proc_state;
audio_proc_state.delay = capture_nonlocked_.stream_delay_ms;
audio_proc_state.drift = 0;
audio_proc_state.level = recommended_stream_analog_level();
audio_proc_state.keypress = capture_.key_pressed;
aec_dump_->AddAudioProcessingState(audio_proc_state);
}
AudioProcessingImpl::ApmCaptureState::ApmCaptureState()
: was_stream_delay_set(false),
output_will_be_muted(false),
key_pressed(false),
capture_processing_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz),
echo_path_gain_change(false),
prev_analog_mic_level(-1),
prev_pre_amp_gain(-1.f),
playout_volume(-1),
prev_playout_volume(-1) {}
AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
void AudioProcessingImpl::ApmCaptureState::KeyboardInfo::Extract(
const float* const* data,
const StreamConfig& stream_config) {
if (stream_config.has_keyboard()) {
keyboard_data = data[stream_config.num_channels()];
} else {
keyboard_data = NULL;
}
num_keyboard_frames = stream_config.num_frames();
}
AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
AudioProcessingImpl::ApmStatsReporter::ApmStatsReporter()
: stats_message_queue_(1) {}
AudioProcessingImpl::ApmStatsReporter::~ApmStatsReporter() = default;
AudioProcessingStats AudioProcessingImpl::ApmStatsReporter::GetStatistics() {
rtc::CritScope cs_stats(&crit_stats_);
bool new_stats_available = stats_message_queue_.Remove(&cached_stats_);
// If the message queue is full, return the cached stats.
static_cast<void>(new_stats_available);
return cached_stats_;
}
void AudioProcessingImpl::ApmStatsReporter::UpdateStatistics(
const AudioProcessingStats& new_stats) {
AudioProcessingStats stats_to_queue = new_stats;
bool stats_message_passed = stats_message_queue_.Insert(&stats_to_queue);
// If the message queue is full, discard the new stats.
static_cast<void>(stats_message_passed);
}
} // namespace webrtc
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