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android / platform / external / webrtc / ae2c5ad12afc8cc29fe9c59dea432b697b871a87 / . / webrtc / modules / audio_processing / audio_processing_impl.cc
blob: b79b4f0c7695d18e952ed7f06c7a324fc17ae90f [file] [log] [blame]
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include <assert.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"
extern "C" {
#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_impl.h"
#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/level_estimator_impl.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/modules/audio_processing/processing_component.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/audio_processing/debug.pb.h"
#endif
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
#define RETURN_ON_ERR(expr) \
do { \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
} while (0)
namespace webrtc {
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;
}
assert(false);
return false;
}
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
// This class has two main functionalities:
//
// 1) It is returned instead of the real GainControl after the new AGC has been
// enabled in order to prevent an outside user from overriding compression
// settings. It doesn't do anything in its implementation, except for
// delegating the const methods and Enable calls to the real GainControl, so
// AGC can still be disabled.
//
// 2) It is injected into AgcManagerDirect and implements volume callbacks for
// getting and setting the volume level. It just caches this value to be used
// in VoiceEngine later.
class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
public:
explicit GainControlForNewAgc(GainControlImpl* gain_control)
: real_gain_control_(gain_control), volume_(0) {}
// GainControl implementation.
int Enable(bool enable) override {
return real_gain_control_->Enable(enable);
}
bool is_enabled() const override { return real_gain_control_->is_enabled(); }
int set_stream_analog_level(int level) override {
volume_ = level;
return AudioProcessing::kNoError;
}
int stream_analog_level() override { return volume_; }
int set_mode(Mode mode) override { return AudioProcessing::kNoError; }
Mode mode() const override { return GainControl::kAdaptiveAnalog; }
int set_target_level_dbfs(int level) override {
return AudioProcessing::kNoError;
}
int target_level_dbfs() const override {
return real_gain_control_->target_level_dbfs();
}
int set_compression_gain_db(int gain) override {
return AudioProcessing::kNoError;
}
int compression_gain_db() const override {
return real_gain_control_->compression_gain_db();
}
int enable_limiter(bool enable) override { return AudioProcessing::kNoError; }
bool is_limiter_enabled() const override {
return real_gain_control_->is_limiter_enabled();
}
int set_analog_level_limits(int minimum, int maximum) override {
return AudioProcessing::kNoError;
}
int analog_level_minimum() const override {
return real_gain_control_->analog_level_minimum();
}
int analog_level_maximum() const override {
return real_gain_control_->analog_level_maximum();
}
bool stream_is_saturated() const override {
return real_gain_control_->stream_is_saturated();
}
// VolumeCallbacks implementation.
void SetMicVolume(int volume) override { volume_ = volume; }
int GetMicVolume() override { return volume_; }
private:
GainControl* real_gain_control_;
int volume_;
};
struct AudioProcessingImpl::ApmPublicSubmodules {
ApmPublicSubmodules()
: echo_cancellation(nullptr),
echo_control_mobile(nullptr),
gain_control(nullptr) {}
// Accessed externally of APM without any lock acquired.
EchoCancellationImpl* echo_cancellation;
EchoControlMobileImpl* echo_control_mobile;
GainControlImpl* gain_control;
rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
rtc::scoped_ptr<VoiceDetectionImpl> voice_detection;
rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
// Accessed internally from both render and capture.
rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
};
struct AudioProcessingImpl::ApmPrivateSubmodules {
explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
: beamformer(beamformer) {}
// Accessed internally from capture or during initialization
std::list<ProcessingComponent*> component_list;
rtc::scoped_ptr<Beamformer<float>> beamformer;
rtc::scoped_ptr<AgcManagerDirect> agc_manager;
};
const int AudioProcessing::kNativeSampleRatesHz[] = {
AudioProcessing::kSampleRate8kHz,
AudioProcessing::kSampleRate16kHz,
AudioProcessing::kSampleRate32kHz,
AudioProcessing::kSampleRate48kHz};
const size_t AudioProcessing::kNumNativeSampleRates =
arraysize(AudioProcessing::kNativeSampleRatesHz);
const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
kNativeSampleRatesHz[AudioProcessing::kNumNativeSampleRates - 1];
const int AudioProcessing::kMaxAECMSampleRateHz = kSampleRate16kHz;
AudioProcessing* AudioProcessing::Create() {
Config config;
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const Config& config) {
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const Config& config,
Beamformer<float>* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
apm = nullptr;
}
return apm;
}
AudioProcessingImpl::AudioProcessingImpl(const Config& config)
: AudioProcessingImpl(config, nullptr) {}
AudioProcessingImpl::AudioProcessingImpl(const Config& config,
Beamformer<float>* beamformer)
: public_submodules_(new ApmPublicSubmodules()),
private_submodules_(new ApmPrivateSubmodules(beamformer)),
constants_(config.Get<ExperimentalAgc>().startup_min_volume,
config.Get<Beamforming>().array_geometry,
config.Get<Beamforming>().target_direction,
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
false,
#else
config.Get<ExperimentalAgc>().enabled,
#endif
config.Get<Intelligibility>().enabled,
config.Get<Beamforming>().enabled),
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
capture_(false)
#else
capture_(config.Get<ExperimentalNs>().enabled)
#endif
{
{
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
public_submodules_->echo_cancellation =
new EchoCancellationImpl(this, &crit_render_, &crit_capture_);
public_submodules_->echo_control_mobile =
new EchoControlMobileImpl(this, &crit_render_, &crit_capture_);
public_submodules_->gain_control =
new GainControlImpl(this, &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_new_agc.reset(
new GainControlForNewAgc(public_submodules_->gain_control));
private_submodules_->component_list.push_back(
public_submodules_->echo_cancellation);
private_submodules_->component_list.push_back(
public_submodules_->echo_control_mobile);
private_submodules_->component_list.push_back(
public_submodules_->gain_control);
}
SetExtraOptions(config);
}
AudioProcessingImpl::~AudioProcessingImpl() {
// Depends on gain_control_ and
// public_submodules_->gain_control_for_new_agc.
private_submodules_->agc_manager.reset();
// Depends on gain_control_.
public_submodules_->gain_control_for_new_agc.reset();
while (!private_submodules_->component_list.empty()) {
ProcessingComponent* component =
private_submodules_->component_list.front();
component->Destroy();
delete component;
private_submodules_->component_list.pop_front();
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->Open()) {
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 input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) {
const ProcessingConfig processing_config = {
{{input_sample_rate_hz,
ChannelsFromLayout(input_layout),
LayoutHasKeyboard(input_layout)},
{output_sample_rate_hz,
ChannelsFromLayout(output_layout),
LayoutHasKeyboard(output_layout)},
{reverse_sample_rate_hz,
ChannelsFromLayout(reverse_layout),
LayoutHasKeyboard(reverse_layout)},
{reverse_sample_rate_hz,
ChannelsFromLayout(reverse_layout),
LayoutHasKeyboard(reverse_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);
}
int AudioProcessingImpl::MaybeInitializeCapture(
const ProcessingConfig& processing_config) {
return MaybeInitialize(processing_config);
}
// 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) {
// 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() {
const int fwd_audio_buffer_channels =
constants_.beamformer_enabled
? formats_.api_format.input_stream().num_channels()
: formats_.api_format.output_stream().num_channels();
const int rev_audio_buffer_out_num_frames =
formats_.api_format.reverse_output_stream().num_frames() == 0
? formats_.rev_proc_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_.rev_proc_format.num_frames(),
formats_.rev_proc_format.num_channels(),
rev_audio_buffer_out_num_frames));
if (rev_conversion_needed()) {
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_.fwd_proc_format.num_frames(),
fwd_audio_buffer_channels,
formats_.api_format.output_stream().num_frames()));
// Initialize all components.
for (auto item : private_submodules_->component_list) {
int err = item->Initialize();
if (err != kNoError) {
return err;
}
}
InitializeExperimentalAgc();
InitializeTransient();
InitializeBeamformer();
InitializeIntelligibility();
InitializeHighPassFilter();
InitializeNoiseSuppression();
InitializeLevelEstimator();
InitializeVoiceDetection();
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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) {
return kBadNumberChannelsError;
}
if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
return kBadSampleRateError;
}
}
const int num_in_channels = config.input_stream().num_channels();
const int 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 (constants_.beamformer_enabled && (static_cast<size_t>(num_in_channels) !=
constants_.array_geometry.size() ||
num_out_channels > 1)) {
return kBadNumberChannelsError;
}
formats_.api_format = config;
// We process at the closest native rate >= min(input rate, output rate)...
const int min_proc_rate =
std::min(formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.output_stream().sample_rate_hz());
int fwd_proc_rate;
for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
fwd_proc_rate = kNativeSampleRatesHz[i];
if (fwd_proc_rate >= min_proc_rate) {
break;
}
}
// ...with one exception.
if (public_submodules_->echo_control_mobile->is_enabled() &&
min_proc_rate > kMaxAECMSampleRateHz) {
fwd_proc_rate = kMaxAECMSampleRateHz;
}
capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
// We normally process the reverse stream at 16 kHz. Unless...
int rev_proc_rate = kSampleRate16kHz;
if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
// ...the forward stream is at 8 kHz.
rev_proc_rate = kSampleRate8kHz;
} else {
if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
kSampleRate32kHz) {
// ...or the input is at 32 kHz, in which case we use the splitting
// filter rather than the resampler.
rev_proc_rate = kSampleRate32kHz;
}
}
// Always downmix the reverse stream to mono for analysis. This has been
// demonstrated to work well for AEC in most practical scenarios.
formats_.rev_proc_format = StreamConfig(rev_proc_rate, 1);
if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz ||
capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
capture_nonlocked_.split_rate = kSampleRate16kHz;
} else {
capture_nonlocked_.split_rate =
capture_nonlocked_.fwd_proc_format.sample_rate_hz();
}
return InitializeLocked();
}
void AudioProcessingImpl::SetExtraOptions(const 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_);
for (auto item : private_submodules_->component_list) {
item->SetExtraOptions(config);
}
if (capture_.transient_suppressor_enabled !=
config.Get<ExperimentalNs>().enabled) {
capture_.transient_suppressor_enabled =
config.Get<ExperimentalNs>().enabled;
InitializeTransient();
}
}
int AudioProcessingImpl::input_sample_rate_hz() const {
// Accessed from outside APM, hence a lock is needed.
rtc::CritScope cs(&crit_capture_);
return formats_.api_format.input_stream().sample_rate_hz();
}
int AudioProcessingImpl::proc_sample_rate_hz() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.fwd_proc_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;
}
int AudioProcessingImpl::num_reverse_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.rev_proc_format.num_channels();
}
int AudioProcessingImpl::num_input_channels() const {
// Used as callback from submodules, hence locking is not allowed.
return formats_.api_format.input_stream().num_channels();
}
int 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;
{
// 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;
}
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));
}
rtc::CritScope cs_capture(&crit_capture_);
assert(processing_config.input_stream().num_frames() ==
formats_.api_format.input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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 (int 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(ProcessStreamLocked());
capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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 (int 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;
}
if (public_submodules_->echo_control_mobile->is_enabled() &&
frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
return kUnsupportedComponentError;
}
ProcessingConfig processing_config;
{
// 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;
}
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));
}
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->Open()) {
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(ProcessStreamLocked());
capture_.capture_audio->InterleaveTo(frame,
output_copy_needed(is_data_processed()));
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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::ProcessStreamLocked() {
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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* ca = capture_.capture_audio.get(); // For brevity.
if (constants_.use_new_agc &&
public_submodules_->gain_control->is_enabled()) {
private_submodules_->agc_manager->AnalyzePreProcess(
ca->channels()[0], ca->num_channels(),
capture_nonlocked_.fwd_proc_format.num_frames());
}
bool data_processed = is_data_processed();
if (analysis_needed(data_processed)) {
ca->SplitIntoFrequencyBands();
}
if (constants_.intelligibility_enabled) {
public_submodules_->intelligibility_enhancer->AnalyzeCaptureAudio(
ca->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
ca->num_channels());
}
if (constants_.beamformer_enabled) {
private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
ca->split_data_f());
ca->set_num_channels(1);
}
public_submodules_->high_pass_filter->ProcessCaptureAudio(ca);
RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca);
RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
if (public_submodules_->echo_control_mobile->is_enabled() &&
public_submodules_->noise_suppression->is_enabled()) {
ca->CopyLowPassToReference();
}
public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
RETURN_ON_ERR(
public_submodules_->echo_control_mobile->ProcessCaptureAudio(ca));
public_submodules_->voice_detection->ProcessCaptureAudio(ca);
if (constants_.use_new_agc &&
public_submodules_->gain_control->is_enabled() &&
(!constants_.beamformer_enabled ||
private_submodules_->beamformer->is_target_present())) {
private_submodules_->agc_manager->Process(
ca->split_bands_const(0)[kBand0To8kHz], ca->num_frames_per_band(),
capture_nonlocked_.split_rate);
}
RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(ca));
if (synthesis_needed(data_processed)) {
ca->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(
ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
capture_.key_pressed);
}
// The level estimator operates on the recombined data.
public_submodules_->level_estimator->ProcessStream(ca);
capture_.was_stream_delay_set = false;
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int rev_sample_rate_hz,
ChannelLayout layout) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
rtc::CritScope cs(&crit_render_);
const StreamConfig reverse_config = {
rev_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& reverse_input_config,
const StreamConfig& reverse_output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
rtc::CritScope cs(&crit_render_);
RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, reverse_input_config,
reverse_output_config));
if (is_rev_processed()) {
render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
dest);
} else if (render_check_rev_conversion_needed()) {
render_.render_converter->Convert(src, reverse_input_config.num_samples(),
dest,
reverse_output_config.num_samples());
} else {
CopyAudioIfNeeded(src, reverse_input_config.num_frames(),
reverse_input_config.num_channels(), dest);
}
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStreamLocked(
const float* const* src,
const StreamConfig& reverse_input_config,
const StreamConfig& reverse_output_config) {
if (src == nullptr) {
return kNullPointerError;
}
if (reverse_input_config.num_channels() <= 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream() = reverse_input_config;
processing_config.reverse_output_stream() = reverse_output_config;
RETURN_ON_ERR(MaybeInitializeRender(processing_config));
assert(reverse_input_config.num_frames() ==
formats_.api_format.reverse_input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->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 (int 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 ProcessReverseStreamLocked();
}
int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
RETURN_ON_ERR(AnalyzeReverseStream(frame));
rtc::CritScope cs(&crit_render_);
if (is_rev_processed()) {
render_.render_audio->InterleaveTo(frame, true);
}
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_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;
}
// This interface does not tolerate different forward and reverse rates.
if (frame->sample_rate_hz_ !=
formats_.api_format.input_stream().sample_rate_hz()) {
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->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 ProcessReverseStreamLocked();
}
int AudioProcessingImpl::ProcessReverseStreamLocked() {
AudioBuffer* ra = render_.render_audio.get(); // For brevity.
if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
ra->SplitIntoFrequencyBands();
}
if (constants_.intelligibility_enabled) {
// Currently run in single-threaded mode when the intelligibility
// enhancer is activated.
// TODO(peah): Fix to be properly multi-threaded.
rtc::CritScope cs(&crit_capture_);
public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
ra->num_channels());
}
RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
RETURN_ON_ERR(
public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
if (!constants_.use_new_agc) {
RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
}
if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
is_rev_processed()) {
ra->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.
if (debug_dump_.debug_file->Open()) {
if (debug_dump_.debug_file->CloseFile() == -1) {
return kFileError;
}
}
if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
debug_dump_.debug_file->CloseFile();
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.
if (debug_dump_.debug_file->Open()) {
if (debug_dump_.debug_file->CloseFile() == -1) {
return kFileError;
}
}
if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
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.
if (debug_dump_.debug_file->Open()) {
if (debug_dump_.debug_file->CloseFile() == -1) {
return kFileError;
}
}
return kNoError;
#else
return kUnsupportedFunctionError;
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->echo_cancellation;
}
EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->echo_control_mobile;
}
GainControl* AudioProcessingImpl::gain_control() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
if (constants_.use_new_agc) {
return public_submodules_->gain_control_for_new_agc.get();
}
return public_submodules_->gain_control;
}
HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->high_pass_filter.get();
}
LevelEstimator* AudioProcessingImpl::level_estimator() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->level_estimator.get();
}
NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->noise_suppression.get();
}
VoiceDetection* AudioProcessingImpl::voice_detection() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
return public_submodules_->voice_detection.get();
}
bool AudioProcessingImpl::is_data_processed() const {
if (constants_.beamformer_enabled) {
return true;
}
int enabled_count = 0;
for (auto item : private_submodules_->component_list) {
if (item->is_component_enabled()) {
enabled_count++;
}
}
if (public_submodules_->high_pass_filter->is_enabled()) {
enabled_count++;
}
if (public_submodules_->noise_suppression->is_enabled()) {
enabled_count++;
}
if (public_submodules_->level_estimator->is_enabled()) {
enabled_count++;
}
if (public_submodules_->voice_detection->is_enabled()) {
enabled_count++;
}
// Data is unchanged if no components are enabled, or if only
// public_submodules_->level_estimator
// or public_submodules_->voice_detection is enabled.
if (enabled_count == 0) {
return false;
} else if (enabled_count == 1) {
if (public_submodules_->level_estimator->is_enabled() ||
public_submodules_->voice_detection->is_enabled()) {
return false;
}
} else if (enabled_count == 2) {
if (public_submodules_->level_estimator->is_enabled() &&
public_submodules_->voice_detection->is_enabled()) {
return false;
}
}
return true;
}
bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
// Check if we've upmixed or downmixed the audio.
return ((formats_.api_format.output_stream().num_channels() !=
formats_.api_format.input_stream().num_channels()) ||
is_data_processed || capture_.transient_suppressor_enabled);
}
bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
return (is_data_processed &&
(capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
kSampleRate32kHz ||
capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
kSampleRate48kHz));
}
bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
if (!is_data_processed &&
!public_submodules_->voice_detection->is_enabled() &&
!capture_.transient_suppressor_enabled) {
// Only public_submodules_->level_estimator is enabled.
return false;
} else if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
kSampleRate32kHz ||
capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
kSampleRate48kHz) {
// Something besides public_submodules_->level_estimator is enabled, and we
// have super-wb.
return true;
}
return false;
}
bool AudioProcessingImpl::is_rev_processed() const {
return constants_.intelligibility_enabled &&
public_submodules_->intelligibility_enhancer->active();
}
bool AudioProcessingImpl::render_check_rev_conversion_needed() const {
return rev_conversion_needed();
}
bool AudioProcessingImpl::rev_conversion_needed() const {
return (formats_.api_format.reverse_input_stream() !=
formats_.api_format.reverse_output_stream());
}
void AudioProcessingImpl::InitializeExperimentalAgc() {
if (constants_.use_new_agc) {
if (!private_submodules_->agc_manager.get()) {
private_submodules_->agc_manager.reset(new AgcManagerDirect(
public_submodules_->gain_control,
public_submodules_->gain_control_for_new_agc.get(),
constants_.agc_startup_min_volume));
}
private_submodules_->agc_manager->Initialize();
private_submodules_->agc_manager->SetCaptureMuted(
capture_.output_will_be_muted);
}
}
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_.fwd_proc_format.sample_rate_hz(),
capture_nonlocked_.split_rate,
formats_.api_format.output_stream().num_channels());
}
}
void AudioProcessingImpl::InitializeBeamformer() {
if (constants_.beamformer_enabled) {
if (!private_submodules_->beamformer) {
private_submodules_->beamformer.reset(new NonlinearBeamformer(
constants_.array_geometry, constants_.target_direction));
}
private_submodules_->beamformer->Initialize(kChunkSizeMs,
capture_nonlocked_.split_rate);
}
}
void AudioProcessingImpl::InitializeIntelligibility() {
if (constants_.intelligibility_enabled) {
IntelligibilityEnhancer::Config config;
config.sample_rate_hz = capture_nonlocked_.split_rate;
config.num_capture_channels = capture_.capture_audio->num_channels();
config.num_render_channels = render_.render_audio->num_channels();
public_submodules_->intelligibility_enhancer.reset(
new IntelligibilityEnhancer(config));
}
}
void AudioProcessingImpl::InitializeHighPassFilter() {
public_submodules_->high_pass_filter->Initialize(num_output_channels(),
proc_sample_rate_hz());
}
void AudioProcessingImpl::InitializeNoiseSuppression() {
public_submodules_->noise_suppression->Initialize(num_output_channels(),
proc_sample_rate_hz());
}
void AudioProcessingImpl::InitializeLevelEstimator() {
public_submodules_->level_estimator->Initialize();
}
void AudioProcessingImpl::InitializeVoiceDetection() {
public_submodules_->voice_detection->Initialize(proc_split_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 frames_per_ms =
rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
const int aec_system_delay_ms =
WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_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->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(
formats_.api_format.input_stream().num_channels());
msg->set_num_output_channels(
formats_.api_format.output_stream().num_channels());
msg->set_num_reverse_channels(
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());
// TODO(ekmeyerson): Add reverse output fields to
// debug_dump_.capture.event_msg.
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_new_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);
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
} // namespace webrtc