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platform-external-webrtc/webrtc/modules/audio_processing/audio_processing_impl.cc
Alejandro Luebs ef00925cd0 Compensate for the IntelligibilityEnhancer processing delay in high bands 
Before this CL, the IntelligibilityEnhancer introduced a processing delay to the lower band, without compensating for it in the higher bands. This CL corrects this.

BUG=b/30780909
R=henrik.lundin@webrtc.org, peah@webrtc.org

Review URL: https://codereview.webrtc.org/2320833002 .

Cr-Commit-Position: refs/heads/master@{#14311}
2016-09-20 21:52:08 +00:00

1592 lines
60 KiB
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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 "webrtc/modules/audio_processing/audio_processing_impl.h"
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_audio/audio_converter.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
#include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
#include "webrtc/modules/audio_processing/gain_control_for_experimental_agc.h"
#include "webrtc/modules/audio_processing/gain_control_impl.h"
#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
#if WEBRTC_INTELLIGIBILITY_ENHANCER
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#endif
#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
#include "webrtc/modules/audio_processing/level_estimator_impl.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/voice_detection_impl.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
#else
#include "webrtc/modules/audio_processing/debug.pb.h"
#endif
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
// Check to verify that the define for the intelligibility enhancer is properly
// set.
#if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \
(WEBRTC_INTELLIGIBILITY_ENHANCER != 0 && \
WEBRTC_INTELLIGIBILITY_ENHANCER != 1)
#error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1"
#endif
#define RETURN_ON_ERR(expr) \
do { \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
} while (0)
namespace webrtc {
constexpr int AudioProcessing::kNativeSampleRatesHz[];
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;
}
int FindNativeProcessRateToUse(int minimum_rate, bool band_splitting_required) {
#ifdef WEBRTC_ARCH_ARM_FAMILY
constexpr int kMaxSplittingNativeProcessRate =
AudioProcessing::kSampleRate32kHz;
#else
constexpr int kMaxSplittingNativeProcessRate =
AudioProcessing::kSampleRate48kHz;
#endif
static_assert(
kMaxSplittingNativeProcessRate <= AudioProcessing::kMaxNativeSampleRateHz,
"");
const int uppermost_native_rate = band_splitting_required
? kMaxSplittingNativeProcessRate
: AudioProcessing::kSampleRate48kHz;
for (auto rate : AudioProcessing::kNativeSampleRatesHz) {
if (rate >= uppermost_native_rate) {
return uppermost_native_rate;
}
if (rate >= minimum_rate) {
return rate;
}
}
RTC_NOTREACHED();
return uppermost_native_rate;
}
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
AudioProcessingImpl::ApmSubmoduleStates::ApmSubmoduleStates() {}
bool AudioProcessingImpl::ApmSubmoduleStates::Update(
bool high_pass_filter_enabled,
bool echo_canceller_enabled,
bool mobile_echo_controller_enabled,
bool noise_suppressor_enabled,
bool intelligibility_enhancer_enabled,
bool beamformer_enabled,
bool adaptive_gain_controller_enabled,
bool level_controller_enabled,
bool voice_activity_detector_enabled,
bool level_estimator_enabled,
bool transient_suppressor_enabled) {
bool changed = false;
changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
changed |= (echo_canceller_enabled != echo_canceller_enabled_);
changed |=
(mobile_echo_controller_enabled != mobile_echo_controller_enabled_);
changed |= (noise_suppressor_enabled != noise_suppressor_enabled_);
changed |=
(intelligibility_enhancer_enabled != intelligibility_enhancer_enabled_);
changed |= (beamformer_enabled != beamformer_enabled_);
changed |=
(adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
changed |= (level_controller_enabled != level_controller_enabled_);
changed |= (level_estimator_enabled != level_estimator_enabled_);
changed |=
(voice_activity_detector_enabled != voice_activity_detector_enabled_);
changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
if (changed) {
high_pass_filter_enabled_ = high_pass_filter_enabled;
echo_canceller_enabled_ = echo_canceller_enabled;
mobile_echo_controller_enabled_ = mobile_echo_controller_enabled;
noise_suppressor_enabled_ = noise_suppressor_enabled;
intelligibility_enhancer_enabled_ = intelligibility_enhancer_enabled;
beamformer_enabled_ = beamformer_enabled;
adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
level_controller_enabled_ = level_controller_enabled;
level_estimator_enabled_ = level_estimator_enabled;
voice_activity_detector_enabled_ = voice_activity_detector_enabled;
transient_suppressor_enabled_ = transient_suppressor_enabled;
}
changed |= first_update_;
first_update_ = false;
return changed;
}
bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandSubModulesActive()
const {
#if WEBRTC_INTELLIGIBILITY_ENHANCER
return CaptureMultiBandProcessingActive() ||
intelligibility_enhancer_enabled_ || voice_activity_detector_enabled_;
#else
return CaptureMultiBandProcessingActive() || voice_activity_detector_enabled_;
#endif
}
bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive()
const {
return high_pass_filter_enabled_ || echo_canceller_enabled_ ||
mobile_echo_controller_enabled_ || noise_suppressor_enabled_ ||
beamformer_enabled_ || adaptive_gain_controller_enabled_;
}
bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()
const {
return RenderMultiBandProcessingActive() || echo_canceller_enabled_ ||
mobile_echo_controller_enabled_ || adaptive_gain_controller_enabled_;
}
bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandProcessingActive()
const {
#if WEBRTC_INTELLIGIBILITY_ENHANCER
return intelligibility_enhancer_enabled_;
#else
return false;
#endif
}
struct AudioProcessingImpl::ApmPublicSubmodules {
ApmPublicSubmodules() {}
// Accessed externally of APM without any lock acquired.
std::unique_ptr<EchoCancellationImpl> echo_cancellation;
std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
std::unique_ptr<GainControlImpl> gain_control;
std::unique_ptr<HighPassFilterImpl> high_pass_filter;
std::unique_ptr<LevelEstimatorImpl> level_estimator;
std::unique_ptr<NoiseSuppressionImpl> noise_suppression;
std::unique_ptr<VoiceDetectionImpl> voice_detection;
std::unique_ptr<GainControlForExperimentalAgc>
gain_control_for_experimental_agc;
// Accessed internally from both render and capture.
std::unique_ptr<TransientSuppressor> transient_suppressor;
#if WEBRTC_INTELLIGIBILITY_ENHANCER
std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
#endif
};
struct AudioProcessingImpl::ApmPrivateSubmodules {
explicit ApmPrivateSubmodules(NonlinearBeamformer* beamformer)
: beamformer(beamformer) {}
// Accessed internally from capture or during initialization
std::unique_ptr<NonlinearBeamformer> beamformer;
std::unique_ptr<AgcManagerDirect> agc_manager;
std::unique_ptr<LevelController> level_controller;
};
AudioProcessing* AudioProcessing::Create() {
webrtc::Config config;
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const webrtc::Config& config) {
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
apm = nullptr;
}
return apm;
}
AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
: AudioProcessingImpl(config, nullptr) {}
AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
NonlinearBeamformer* beamformer)
: public_submodules_(new ApmPublicSubmodules()),
private_submodules_(new ApmPrivateSubmodules(beamformer)),
constants_(config.Get<ExperimentalAgc>().startup_min_volume,
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
false),
#else
config.Get<ExperimentalAgc>().enabled),
#endif
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
capture_(false,
#else
capture_(config.Get<ExperimentalNs>().enabled,
#endif
config.Get<Beamforming>().array_geometry,
config.Get<Beamforming>().target_direction),
capture_nonlocked_(config.Get<Beamforming>().enabled,
config.Get<Intelligibility>().enabled) {
{
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
public_submodules_->echo_cancellation.reset(
new EchoCancellationImpl(&crit_render_, &crit_capture_));
public_submodules_->echo_control_mobile.reset(
new EchoControlMobileImpl(&crit_render_, &crit_capture_));
public_submodules_->gain_control.reset(
new GainControlImpl(&crit_capture_, &crit_capture_));
public_submodules_->high_pass_filter.reset(
new HighPassFilterImpl(&crit_capture_));
public_submodules_->level_estimator.reset(
new LevelEstimatorImpl(&crit_capture_));
public_submodules_->noise_suppression.reset(
new NoiseSuppressionImpl(&crit_capture_));
public_submodules_->voice_detection.reset(
new VoiceDetectionImpl(&crit_capture_));
public_submodules_->gain_control_for_experimental_agc.reset(
new GainControlForExperimentalAgc(
public_submodules_->gain_control.get(), &crit_capture_));
private_submodules_->level_controller.reset(new LevelController());
}
SetExtraOptions(config);
}
AudioProcessingImpl::~AudioProcessingImpl() {
// Depends on gain_control_ and
// public_submodules_->gain_control_for_experimental_agc.
private_submodules_->agc_manager.reset();
// Depends on gain_control_.
public_submodules_->gain_control_for_experimental_agc.reset();
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
debug_dump_.debug_file->CloseFile();
#endif
}
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) {
return MaybeInitialize(processing_config, false);
}
int AudioProcessingImpl::MaybeInitializeCapture(
const ProcessingConfig& processing_config,
bool force_initialization) {
return MaybeInitialize(processing_config, force_initialization);
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
AudioProcessingImpl::ApmDebugDumpThreadState::ApmDebugDumpThreadState()
: event_msg(new audioproc::Event()) {}
AudioProcessingImpl::ApmDebugDumpThreadState::~ApmDebugDumpThreadState() {}
AudioProcessingImpl::ApmDebugDumpState::ApmDebugDumpState()
: debug_file(FileWrapper::Create()) {}
AudioProcessingImpl::ApmDebugDumpState::~ApmDebugDumpState() {}
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
// Calls InitializeLocked() if any of the audio parameters have changed from
// their current values (needs to be called while holding the crit_render_lock).
int AudioProcessingImpl::MaybeInitialize(
const ProcessingConfig& processing_config,
bool force_initialization) {
// Called from both threads. Thread check is therefore not possible.
if (processing_config == formats_.api_format && !force_initialization) {
return kNoError;
}
rtc::CritScope cs_capture(&crit_capture_);
return InitializeLocked(processing_config);
}
int AudioProcessingImpl::InitializeLocked() {
const int capture_audiobuffer_num_channels =
capture_nonlocked_.beamformer_enabled
? formats_.api_format.input_stream().num_channels()
: formats_.api_format.output_stream().num_channels();
const int render_audiobuffer_num_output_frames =
formats_.api_format.reverse_output_stream().num_frames() == 0
? formats_.render_processing_format.num_frames()
: formats_.api_format.reverse_output_stream().num_frames();
if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
render_.render_audio.reset(new AudioBuffer(
formats_.api_format.reverse_input_stream().num_frames(),
formats_.api_format.reverse_input_stream().num_channels(),
formats_.render_processing_format.num_frames(),
formats_.render_processing_format.num_channels(),
render_audiobuffer_num_output_frames));
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().num_frames(),
formats_.api_format.input_stream().num_channels(),
capture_nonlocked_.capture_processing_format.num_frames(),
capture_audiobuffer_num_channels,
formats_.api_format.output_stream().num_frames()));
public_submodules_->gain_control->Initialize(num_proc_channels(),
proc_sample_rate_hz());
public_submodules_->echo_cancellation->Initialize(
proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
num_proc_channels());
public_submodules_->echo_control_mobile->Initialize(
proc_split_sample_rate_hz(), num_reverse_channels(),
num_output_channels());
if (constants_.use_experimental_agc) {
if (!private_submodules_->agc_manager.get()) {
private_submodules_->agc_manager.reset(new AgcManagerDirect(
public_submodules_->gain_control.get(),
public_submodules_->gain_control_for_experimental_agc.get(),
constants_.agc_startup_min_volume));
}
private_submodules_->agc_manager->Initialize();
private_submodules_->agc_manager->SetCaptureMuted(
capture_.output_will_be_muted);
}
InitializeTransient();
InitializeBeamformer();
#if WEBRTC_INTELLIGIBILITY_ENHANCER
InitializeIntelligibility();
#endif
public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
proc_sample_rate_hz());
public_submodules_->noise_suppression->Initialize(num_proc_channels(),
proc_sample_rate_hz());
public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
public_submodules_->level_estimator->Initialize();
InitializeLevelController();
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
int err = WriteInitMessage();
if (err != kNoError) {
return err;
}
}
#endif
return kNoError;
}
int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
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;
}
if (capture_nonlocked_.beamformer_enabled &&
num_in_channels != capture_.array_geometry.size()) {
return kBadNumberChannelsError;
}
formats_.api_format = config;
int capture_processing_rate = FindNativeProcessRateToUse(
std::min(formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.output_stream().sample_rate_hz()),
submodule_states_.CaptureMultiBandSubModulesActive() ||
submodule_states_.RenderMultiBandSubModulesActive());
capture_nonlocked_.capture_processing_format =
StreamConfig(capture_processing_rate);
int render_processing_rate = FindNativeProcessRateToUse(
std::min(formats_.api_format.reverse_input_stream().sample_rate_hz(),
formats_.api_format.reverse_output_stream().sample_rate_hz()),
submodule_states_.CaptureMultiBandSubModulesActive() ||
submodule_states_.RenderMultiBandSubModulesActive());
// TODO(aluebs): Remove this restriction once we figure out why the 3-band
// splitting filter degrades the AEC performance.
if (render_processing_rate > kSampleRate32kHz) {
render_processing_rate = submodule_states_.RenderMultiBandProcessingActive()
? kSampleRate32kHz
: kSampleRate16kHz;
}
// 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));
}
// Always downmix the render stream to mono for analysis. This has been
// demonstrated to work well for AEC in most practical scenarios.
formats_.render_processing_format = StreamConfig(render_processing_rate, 1);
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) {
AudioProcessing::Config config_to_use = config;
bool config_ok = LevelController::Validate(config_to_use.level_controller);
if (!config_ok) {
LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
<< "level_controller: "
<< LevelController::ToString(config_to_use.level_controller)
<< std::endl
<< "Reverting to default parameter set";
config_to_use.level_controller = AudioProcessing::Config::LevelController();
}
// Run in a single-threaded manner when applying the settings.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
if (config.level_controller.enabled !=
capture_nonlocked_.level_controller_enabled) {
InitializeLevelController();
LOG(LS_INFO) << "Level controller activated: "
<< capture_nonlocked_.level_controller_enabled;
capture_nonlocked_.level_controller_enabled =
config.level_controller.enabled;
}
}
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
// Run in a single-threaded manner when setting the extra options.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
public_submodules_->echo_cancellation->SetExtraOptions(config);
if (capture_.transient_suppressor_enabled !=
config.Get<ExperimentalNs>().enabled) {
capture_.transient_suppressor_enabled =
config.Get<ExperimentalNs>().enabled;
InitializeTransient();
}
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if(capture_nonlocked_.intelligibility_enabled !=
config.Get<Intelligibility>().enabled) {
capture_nonlocked_.intelligibility_enabled =
config.Get<Intelligibility>().enabled;
InitializeIntelligibility();
}
#endif
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
if (capture_nonlocked_.beamformer_enabled !=
config.Get<Beamforming>().enabled) {
capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
if (config.Get<Beamforming>().array_geometry.size() > 1) {
capture_.array_geometry = config.Get<Beamforming>().array_geometry;
}
capture_.target_direction = config.Get<Beamforming>().target_direction;
InitializeBeamformer();
}
#endif // WEBRTC_ANDROID_PLATFORM_BUILD
}
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_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.
return capture_nonlocked_.beamformer_enabled ? 1 : 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 (private_submodules_->agc_manager.get()) {
private_submodules_->agc_manager->SetCaptureMuted(
capture_.output_will_be_muted);
}
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
size_t samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_ChannelLayout");
StreamConfig input_stream;
StreamConfig output_stream;
{
// Access the formats_.api_format.input_stream beneath the capture lock.
// The lock must be released as it is later required in the call
// to ProcessStream(,,,);
rtc::CritScope cs(&crit_capture_);
input_stream = formats_.api_format.input_stream();
output_stream = formats_.api_format.output_stream();
}
input_stream.set_sample_rate_hz(input_sample_rate_hz);
input_stream.set_num_channels(ChannelsFromLayout(input_layout));
input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
output_stream.set_sample_rate_hz(output_sample_rate_hz);
output_stream.set_num_channels(ChannelsFromLayout(output_layout));
output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
if (samples_per_channel != input_stream.num_frames()) {
return kBadDataLengthError;
}
return ProcessStream(src, input_stream, output_stream, dest);
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Acquire the capture lock in order to safely call the function
// that retrieves the render side data. This function accesses apm
// getters that need the capture lock held when being called.
rtc::CritScope cs_capture(&crit_capture_);
public_submodules_->echo_cancellation->ReadQueuedRenderData();
public_submodules_->echo_control_mobile->ReadQueuedRenderData();
public_submodules_->gain_control->ReadQueuedRenderData();
if (!src || !dest) {
return kNullPointerError;
}
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
processing_config.input_stream() = input_config;
processing_config.output_stream() = output_config;
{
// Do conditional reinitialization.
rtc::CritScope cs_render(&crit_render_);
RETURN_ON_ERR(
MaybeInitializeCapture(processing_config, reinitialization_required));
}
rtc::CritScope cs_capture(&crit_capture_);
RTC_DCHECK_EQ(processing_config.input_stream().num_frames(),
formats_.api_format.input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
RETURN_ON_ERR(WriteConfigMessage(false));
debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.input_stream().num_frames();
for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
++i)
msg->add_input_channel(src[i], channel_size);
}
#endif
capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
RETURN_ON_ERR(ProcessCaptureStreamLocked());
capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.output_stream().num_frames();
for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
}
#endif
return kNoError;
}
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
{
// Acquire the capture lock in order to safely call the function
// that retrieves the render side data. This function accesses apm
// getters that need the capture lock held when being called.
// The lock needs to be released as
// public_submodules_->echo_control_mobile->is_enabled() aquires this lock
// as well.
rtc::CritScope cs_capture(&crit_capture_);
public_submodules_->echo_cancellation->ReadQueuedRenderData();
public_submodules_->echo_control_mobile->ReadQueuedRenderData();
public_submodules_->gain_control->ReadQueuedRenderData();
}
if (!frame) {
return kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != kSampleRate8kHz &&
frame->sample_rate_hz_ != kSampleRate16kHz &&
frame->sample_rate_hz_ != kSampleRate32kHz &&
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Aquire lock for the access of api_format.
// The lock is released immediately due to the conditional
// reinitialization.
rtc::CritScope cs_capture(&crit_capture_);
// TODO(ajm): The input and output rates and channels are currently
// constrained to be identical in the int16 interface.
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.input_stream().set_num_channels(frame->num_channels_);
processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.output_stream().set_num_channels(frame->num_channels_);
{
// Do conditional reinitialization.
rtc::CritScope cs_render(&crit_render_);
RETURN_ON_ERR(
MaybeInitializeCapture(processing_config, reinitialization_required));
}
rtc::CritScope cs_capture(&crit_capture_);
if (frame->samples_per_channel_ !=
formats_.api_format.input_stream().num_frames()) {
return kBadDataLengthError;
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
RETURN_ON_ERR(WriteConfigMessage(false));
debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_input_data(frame->data_, data_size);
}
#endif
capture_.capture_audio->DeinterleaveFrom(frame);
RETURN_ON_ERR(ProcessCaptureStreamLocked());
capture_.capture_audio->InterleaveTo(
frame, submodule_states_.CaptureMultiBandProcessingActive());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_output_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
}
#endif
return kNoError;
}
int AudioProcessingImpl::ProcessCaptureStreamLocked() {
// 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(!(public_submodules_->echo_cancellation->is_enabled() &&
public_submodules_->echo_control_mobile->is_enabled()));
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
msg->set_delay(capture_nonlocked_.stream_delay_ms);
msg->set_drift(
public_submodules_->echo_cancellation->stream_drift_samples());
msg->set_level(gain_control()->stream_analog_level());
msg->set_keypress(capture_.key_pressed);
}
#endif
MaybeUpdateHistograms();
AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity.
if (constants_.use_experimental_agc &&
public_submodules_->gain_control->is_enabled()) {
private_submodules_->agc_manager->AnalyzePreProcess(
capture_buffer->channels()[0], capture_buffer->num_channels(),
capture_nonlocked_.capture_processing_format.num_frames());
}
if (submodule_states_.CaptureMultiBandSubModulesActive() &&
SampleRateSupportsMultiBand(
capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
capture_buffer->SplitIntoFrequencyBands();
}
if (capture_nonlocked_.beamformer_enabled) {
private_submodules_->beamformer->AnalyzeChunk(
*capture_buffer->split_data_f());
// Discards all channels by the leftmost one.
capture_buffer->set_num_channels(1);
}
public_submodules_->high_pass_filter->ProcessCaptureAudio(capture_buffer);
RETURN_ON_ERR(
public_submodules_->gain_control->AnalyzeCaptureAudio(capture_buffer));
public_submodules_->noise_suppression->AnalyzeCaptureAudio(capture_buffer);
// Ensure that the stream delay was set before the call to the
// AEC ProcessCaptureAudio function.
if (public_submodules_->echo_cancellation->is_enabled() &&
!was_stream_delay_set()) {
return AudioProcessing::kStreamParameterNotSetError;
}
RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(
capture_buffer, stream_delay_ms()));
if (public_submodules_->echo_control_mobile->is_enabled() &&
public_submodules_->noise_suppression->is_enabled()) {
capture_buffer->CopyLowPassToReference();
}
public_submodules_->noise_suppression->ProcessCaptureAudio(capture_buffer);
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if (capture_nonlocked_.intelligibility_enabled) {
RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
int gain_db = public_submodules_->gain_control->is_enabled() ?
public_submodules_->gain_control->compression_gain_db() :
0;
float gain = std::pow(10.f, gain_db / 20.f);
gain *= capture_nonlocked_.level_controller_enabled ?
private_submodules_->level_controller->GetLastGain() :
1.f;
public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
public_submodules_->noise_suppression->NoiseEstimate(), gain);
}
#endif
// Ensure that the stream delay was set before the call to the
// AECM ProcessCaptureAudio function.
if (public_submodules_->echo_control_mobile->is_enabled() &&
!was_stream_delay_set()) {
return AudioProcessing::kStreamParameterNotSetError;
}
RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessCaptureAudio(
capture_buffer, stream_delay_ms()));
if (capture_nonlocked_.beamformer_enabled) {
private_submodules_->beamformer->PostFilter(capture_buffer->split_data_f());
}
public_submodules_->voice_detection->ProcessCaptureAudio(capture_buffer);
if (constants_.use_experimental_agc &&
public_submodules_->gain_control->is_enabled() &&
(!capture_nonlocked_.beamformer_enabled ||
private_submodules_->beamformer->is_target_present())) {
private_submodules_->agc_manager->Process(
capture_buffer->split_bands_const(0)[kBand0To8kHz],
capture_buffer->num_frames_per_band(), capture_nonlocked_.split_rate);
}
RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(
capture_buffer, echo_cancellation()->stream_has_echo()));
if (submodule_states_.CaptureMultiBandProcessingActive() &&
SampleRateSupportsMultiBand(
capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
capture_buffer->MergeFrequencyBands();
}
// TODO(aluebs): Investigate if the transient suppression placement should be
// before or after the AGC.
if (capture_.transient_suppressor_enabled) {
float voice_probability =
private_submodules_->agc_manager.get()
? private_submodules_->agc_manager->voice_probability()
: 1.f;
public_submodules_->transient_suppressor->Suppress(
capture_buffer->channels_f()[0], capture_buffer->num_frames(),
capture_buffer->num_channels(),
capture_buffer->split_bands_const_f(0)[kBand0To8kHz],
capture_buffer->num_frames_per_band(), capture_buffer->keyboard_data(),
capture_buffer->num_keyboard_frames(), voice_probability,
capture_.key_pressed);
}
if (capture_nonlocked_.level_controller_enabled) {
private_submodules_->level_controller->Process(capture_buffer);
}
// The level estimator operates on the recombined data.
public_submodules_->level_estimator->ProcessStream(capture_buffer);
capture_.was_stream_delay_set = false;
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
rtc::CritScope cs(&crit_render_);
const StreamConfig reverse_config = {
sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
};
if (samples_per_channel != reverse_config.num_frames()) {
return kBadDataLengthError;
}
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()) {
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));
assert(input_config.num_frames() ==
formats_.api_format.reverse_input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg =
debug_dump_.render.event_msg->mutable_reverse_stream();
const size_t channel_size =
sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
for (size_t i = 0;
i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
msg->add_channel(src[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.render));
}
#endif
render_.render_audio->CopyFrom(src,
formats_.api_format.reverse_input_stream());
return ProcessRenderStreamLocked();
}
int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
rtc::CritScope cs(&crit_render_);
if (frame == nullptr) {
return kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != kSampleRate8kHz &&
frame->sample_rate_hz_ != kSampleRate16kHz &&
frame->sample_rate_hz_ != kSampleRate32kHz &&
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
if (frame->num_channels_ <= 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.reverse_input_stream().set_num_channels(
frame->num_channels_);
processing_config.reverse_output_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.reverse_output_stream().set_num_channels(
frame->num_channels_);
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
if (frame->samples_per_channel_ !=
formats_.api_format.reverse_input_stream().num_frames()) {
return kBadDataLengthError;
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->is_open()) {
debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg =
debug_dump_.render.event_msg->mutable_reverse_stream();
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.render));
}
#endif
render_.render_audio->DeinterleaveFrom(frame);
RETURN_ON_ERR(ProcessRenderStreamLocked());
render_.render_audio->InterleaveTo(
frame, submodule_states_.RenderMultiBandProcessingActive());
return kNoError;
}
int AudioProcessingImpl::ProcessRenderStreamLocked() {
AudioBuffer* render_buffer = render_.render_audio.get(); // For brevity.
if (submodule_states_.RenderMultiBandSubModulesActive() &&
SampleRateSupportsMultiBand(
formats_.render_processing_format.sample_rate_hz())) {
render_buffer->SplitIntoFrequencyBands();
}
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if (capture_nonlocked_.intelligibility_enabled) {
public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
render_buffer);
}
#endif
RETURN_ON_ERR(
public_submodules_->echo_cancellation->ProcessRenderAudio(render_buffer));
RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessRenderAudio(
render_buffer));
if (!constants_.use_experimental_agc) {
RETURN_ON_ERR(
public_submodules_->gain_control->ProcessRenderAudio(render_buffer));
}
if (submodule_states_.RenderMultiBandProcessingActive() &&
SampleRateSupportsMultiBand(
formats_.render_processing_format.sample_rate_hz())) {
render_buffer->MergeFrequencyBands();
}
return kNoError;
}
int AudioProcessingImpl::set_stream_delay_ms(int delay) {
rtc::CritScope cs(&crit_capture_);
Error retval = kNoError;
capture_.was_stream_delay_set = true;
delay += capture_.delay_offset_ms;
if (delay < 0) {
delay = 0;
retval = kBadStreamParameterWarning;
}
// TODO(ajm): the max is rather arbitrarily chosen; investigate.
if (delay > 500) {
delay = 500;
retval = kBadStreamParameterWarning;
}
capture_nonlocked_.stream_delay_ms = delay;
return retval;
}
int AudioProcessingImpl::stream_delay_ms() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.stream_delay_ms;
}
bool AudioProcessingImpl::was_stream_delay_set() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_.was_stream_delay_set;
}
void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
rtc::CritScope cs(&crit_capture_);
capture_.key_pressed = key_pressed;
}
void AudioProcessingImpl::set_delay_offset_ms(int offset) {
rtc::CritScope cs(&crit_capture_);
capture_.delay_offset_ms = offset;
}
int AudioProcessingImpl::delay_offset_ms() const {
rtc::CritScope cs(&crit_capture_);
return capture_.delay_offset_ms;
}
int AudioProcessingImpl::StartDebugRecording(
const char filename[AudioProcessing::kMaxFilenameSize],
int64_t max_log_size_bytes) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
if (filename == nullptr) {
return kNullPointerError;
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
// Stop any ongoing recording.
debug_dump_.debug_file->CloseFile();
if (!debug_dump_.debug_file->OpenFile(filename, false)) {
return kFileError;
}
RETURN_ON_ERR(WriteConfigMessage(true));
RETURN_ON_ERR(WriteInitMessage());
return kNoError;
#else
return kUnsupportedFunctionError;
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
int AudioProcessingImpl::StartDebugRecording(FILE* handle,
int64_t max_log_size_bytes) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
if (handle == nullptr) {
return kNullPointerError;
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
// Stop any ongoing recording.
debug_dump_.debug_file->CloseFile();
if (!debug_dump_.debug_file->OpenFromFileHandle(handle)) {
return kFileError;
}
RETURN_ON_ERR(WriteConfigMessage(true));
RETURN_ON_ERR(WriteInitMessage());
return kNoError;
#else
return kUnsupportedFunctionError;
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
rtc::PlatformFile handle) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
return StartDebugRecording(stream, -1);
}
int AudioProcessingImpl::StopDebugRecording() {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// We just return if recording hasn't started.
debug_dump_.debug_file->CloseFile();
return kNoError;
#else
return kUnsupportedFunctionError;
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
return public_submodules_->echo_cancellation.get();
}
EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
return public_submodules_->echo_control_mobile.get();
}
GainControl* AudioProcessingImpl::gain_control() const {
if (constants_.use_experimental_agc) {
return public_submodules_->gain_control_for_experimental_agc.get();
}
return public_submodules_->gain_control.get();
}
HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
return public_submodules_->high_pass_filter.get();
}
LevelEstimator* AudioProcessingImpl::level_estimator() const {
return public_submodules_->level_estimator.get();
}
NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
return public_submodules_->noise_suppression.get();
}
VoiceDetection* AudioProcessingImpl::voice_detection() const {
return public_submodules_->voice_detection.get();
}
bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
return submodule_states_.Update(
public_submodules_->high_pass_filter->is_enabled(),
public_submodules_->echo_cancellation->is_enabled(),
public_submodules_->echo_control_mobile->is_enabled(),
public_submodules_->noise_suppression->is_enabled(),
capture_nonlocked_.intelligibility_enabled,
capture_nonlocked_.beamformer_enabled,
public_submodules_->gain_control->is_enabled(),
capture_nonlocked_.level_controller_enabled,
public_submodules_->voice_detection->is_enabled(),
public_submodules_->level_estimator->is_enabled(),
capture_.transient_suppressor_enabled);
}
void AudioProcessingImpl::InitializeTransient() {
if (capture_.transient_suppressor_enabled) {
if (!public_submodules_->transient_suppressor.get()) {
public_submodules_->transient_suppressor.reset(new TransientSuppressor());
}
public_submodules_->transient_suppressor->Initialize(
capture_nonlocked_.capture_processing_format.sample_rate_hz(),
capture_nonlocked_.split_rate, num_proc_channels());
}
}
void AudioProcessingImpl::InitializeBeamformer() {
if (capture_nonlocked_.beamformer_enabled) {
if (!private_submodules_->beamformer) {
private_submodules_->beamformer.reset(new NonlinearBeamformer(
capture_.array_geometry, 1u, capture_.target_direction));
}
private_submodules_->beamformer->Initialize(kChunkSizeMs,
capture_nonlocked_.split_rate);
}
}
void AudioProcessingImpl::InitializeIntelligibility() {
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if (capture_nonlocked_.intelligibility_enabled) {
public_submodules_->intelligibility_enhancer.reset(
new IntelligibilityEnhancer(capture_nonlocked_.split_rate,
render_.render_audio->num_channels(),
render_.render_audio->num_bands(),
NoiseSuppressionImpl::num_noise_bins()));
}
#endif
}
void AudioProcessingImpl::InitializeLevelController() {
private_submodules_->level_controller->Initialize(proc_sample_rate_hz());
}
void AudioProcessingImpl::MaybeUpdateHistograms() {
static const int kMinDiffDelayMs = 60;
if (echo_cancellation()->is_enabled()) {
// Activate delay_jumps_ counters if we know echo_cancellation is runnning.
// If a stream has echo we know that the echo_cancellation is in process.
if (capture_.stream_delay_jumps == -1 &&
echo_cancellation()->stream_has_echo()) {
capture_.stream_delay_jumps = 0;
}
if (capture_.aec_system_delay_jumps == -1 &&
echo_cancellation()->stream_has_echo()) {
capture_.aec_system_delay_jumps = 0;
}
// Detect a jump in platform reported system delay and log the difference.
const int diff_stream_delay_ms =
capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
if (diff_stream_delay_ms > kMinDiffDelayMs &&
capture_.last_stream_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
if (capture_.stream_delay_jumps == -1) {
capture_.stream_delay_jumps = 0; // Activate counter if needed.
}
capture_.stream_delay_jumps++;
}
capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
// Detect a jump in AEC system delay and log the difference.
const int samples_per_ms =
rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
RTC_DCHECK_LT(0, samples_per_ms);
const int aec_system_delay_ms =
public_submodules_->echo_cancellation->GetSystemDelayInSamples() /
samples_per_ms;
const int diff_aec_system_delay_ms =
aec_system_delay_ms - capture_.last_aec_system_delay_ms;
if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
capture_.last_aec_system_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
100);
if (capture_.aec_system_delay_jumps == -1) {
capture_.aec_system_delay_jumps = 0; // Activate counter if needed.
}
capture_.aec_system_delay_jumps++;
}
capture_.last_aec_system_delay_ms = aec_system_delay_ms;
}
}
void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
if (capture_.stream_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
capture_.stream_delay_jumps, 51);
}
capture_.stream_delay_jumps = -1;
capture_.last_stream_delay_ms = 0;
if (capture_.aec_system_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
capture_.aec_system_delay_jumps, 51);
}
capture_.aec_system_delay_jumps = -1;
capture_.last_aec_system_delay_ms = 0;
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
int AudioProcessingImpl::WriteMessageToDebugFile(
FileWrapper* debug_file,
int64_t* filesize_limit_bytes,
rtc::CriticalSection* crit_debug,
ApmDebugDumpThreadState* debug_state) {
int32_t size = debug_state->event_msg->ByteSize();
if (size <= 0) {
return kUnspecifiedError;
}
#if defined(WEBRTC_ARCH_BIG_ENDIAN)
// TODO(ajm): Use little-endian "on the wire". For the moment, we can be
// pretty safe in assuming little-endian.
#endif
if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
return kUnspecifiedError;
}
{
// Ensure atomic writes of the message.
rtc::CritScope cs_debug(crit_debug);
RTC_DCHECK(debug_file->is_open());
// Update the byte counter.
if (*filesize_limit_bytes >= 0) {
*filesize_limit_bytes -=
(sizeof(int32_t) + debug_state->event_str.length());
if (*filesize_limit_bytes < 0) {
// Not enough bytes are left to write this message, so stop logging.
debug_file->CloseFile();
return kNoError;
}
}
// Write message preceded by its size.
if (!debug_file->Write(&size, sizeof(int32_t))) {
return kFileError;
}
if (!debug_file->Write(debug_state->event_str.data(),
debug_state->event_str.length())) {
return kFileError;
}
}
debug_state->event_msg->Clear();
return kNoError;
}
int AudioProcessingImpl::WriteInitMessage() {
debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
msg->set_num_input_channels(static_cast<google::protobuf::int32>(
formats_.api_format.input_stream().num_channels()));
msg->set_num_output_channels(static_cast<google::protobuf::int32>(
formats_.api_format.output_stream().num_channels()));
msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
formats_.api_format.reverse_input_stream().num_channels()));
msg->set_reverse_sample_rate(
formats_.api_format.reverse_input_stream().sample_rate_hz());
msg->set_output_sample_rate(
formats_.api_format.output_stream().sample_rate_hz());
msg->set_reverse_output_sample_rate(
formats_.api_format.reverse_output_stream().sample_rate_hz());
msg->set_num_reverse_output_channels(
formats_.api_format.reverse_output_stream().num_channels());
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
return kNoError;
}
int AudioProcessingImpl::WriteConfigMessage(bool forced) {
audioproc::Config config;
config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
config.set_aec_delay_agnostic_enabled(
public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
config.set_aec_drift_compensation_enabled(
public_submodules_->echo_cancellation->is_drift_compensation_enabled());
config.set_aec_extended_filter_enabled(
public_submodules_->echo_cancellation->is_extended_filter_enabled());
config.set_aec_suppression_level(static_cast<int>(
public_submodules_->echo_cancellation->suppression_level()));
config.set_aecm_enabled(
public_submodules_->echo_control_mobile->is_enabled());
config.set_aecm_comfort_noise_enabled(
public_submodules_->echo_control_mobile->is_comfort_noise_enabled());
config.set_aecm_routing_mode(static_cast<int>(
public_submodules_->echo_control_mobile->routing_mode()));
config.set_agc_enabled(public_submodules_->gain_control->is_enabled());
config.set_agc_mode(
static_cast<int>(public_submodules_->gain_control->mode()));
config.set_agc_limiter_enabled(
public_submodules_->gain_control->is_limiter_enabled());
config.set_noise_robust_agc_enabled(constants_.use_experimental_agc);
config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
config.set_ns_enabled(public_submodules_->noise_suppression->is_enabled());
config.set_ns_level(
static_cast<int>(public_submodules_->noise_suppression->level()));
config.set_transient_suppression_enabled(
capture_.transient_suppressor_enabled);
config.set_intelligibility_enhancer_enabled(
capture_nonlocked_.intelligibility_enabled);
std::string experiments_description =
public_submodules_->echo_cancellation->GetExperimentsDescription();
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
if (capture_nonlocked_.level_controller_enabled) {
experiments_description += "LevelController;";
}
config.set_experiments_description(experiments_description);
std::string serialized_config = config.SerializeAsString();
if (!forced &&
debug_dump_.capture.last_serialized_config == serialized_config) {
return kNoError;
}
debug_dump_.capture.last_serialized_config = serialized_config;
debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
&debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
return kNoError;
}
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
bool transient_suppressor_enabled,
const std::vector<Point>& array_geometry,
SphericalPointf target_direction)
: aec_system_delay_jumps(-1),
delay_offset_ms(0),
was_stream_delay_set(false),
last_stream_delay_ms(0),
last_aec_system_delay_ms(0),
stream_delay_jumps(-1),
output_will_be_muted(false),
key_pressed(false),
transient_suppressor_enabled(transient_suppressor_enabled),
array_geometry(array_geometry),
target_direction(target_direction),
capture_processing_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz) {}
AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
} // namespace webrtc
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