AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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410
webrtc/modules/audio_processing/aec3/echo_canceller3.cc
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@ -12,10 +12,10 @@
#include <algorithm>
#include <utility>
#include "absl/strings/string_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/high_pass_filter.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
@ -27,8 +27,8 @@ namespace {
enum class EchoCanceller3ApiCall { kCapture, kRender };
bool DetectSaturation(rtc::ArrayView<const float> y) {
for (auto y_k : y) {
if (y_k >= 32700.0f || y_k <= -32700.0f) {
for (size_t k = 0; k < y.size(); ++k) {
if (y[k] >= 32700.0f || y[k] <= -32700.0f) {
return true;
}
}
@ -38,7 +38,7 @@ bool DetectSaturation(rtc::ArrayView<const float> y) {
// Retrieves a value from a field trial if it is available. If no value is
// present, the default value is returned. If the retrieved value is beyond the
// specified limits, the default value is returned instead.
void RetrieveFieldTrialValue(const char* trial_name,
void RetrieveFieldTrialValue(absl::string_view trial_name,
float min,
float max,
float* value_to_update) {
@ -49,12 +49,16 @@ void RetrieveFieldTrialValue(const char* trial_name,
ParseFieldTrial({&field_trial_param}, field_trial_str);
float field_trial_value = static_cast<float>(field_trial_param.Get());
if (field_trial_value >= min && field_trial_value <= max) {
if (field_trial_value >= min && field_trial_value <= max &&
field_trial_value != *value_to_update) {
RTC_LOG(LS_INFO) << "Key " << trial_name
<< " changing AEC3 parameter value from "
<< *value_to_update << " to " << field_trial_value;
*value_to_update = field_trial_value;
}
}
void RetrieveFieldTrialValue(const char* trial_name,
void RetrieveFieldTrialValue(absl::string_view trial_name,
int min,
int max,
int* value_to_update) {
@ -65,7 +69,11 @@ void RetrieveFieldTrialValue(const char* trial_name,
ParseFieldTrial({&field_trial_param}, field_trial_str);
float field_trial_value = field_trial_param.Get();
if (field_trial_value >= min && field_trial_value <= max) {
if (field_trial_value >= min && field_trial_value <= max &&
field_trial_value != *value_to_update) {
RTC_LOG(LS_INFO) << "Key " << trial_name
<< " changing AEC3 parameter value from "
<< *value_to_update << " to " << field_trial_value;
*value_to_update = field_trial_value;
}
}
@ -88,18 +96,50 @@ void FillSubFrameView(
}
void FillSubFrameView(
bool proper_downmix_needed,
std::vector<std::vector<std::vector<float>>>* frame,
size_t sub_frame_index,
std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
RTC_DCHECK_GE(1, sub_frame_index);
RTC_DCHECK_EQ(frame->size(), sub_frame_view->size());
RTC_DCHECK_EQ((*frame)[0].size(), (*sub_frame_view)[0].size());
for (size_t band = 0; band < frame->size(); ++band) {
for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) {
(*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
&(*frame)[band][channel][sub_frame_index * kSubFrameLength],
const size_t frame_num_channels = (*frame)[0].size();
const size_t sub_frame_num_channels = (*sub_frame_view)[0].size();
if (frame_num_channels > sub_frame_num_channels) {
RTC_DCHECK_EQ(sub_frame_num_channels, 1u);
if (proper_downmix_needed) {
// When a proper downmix is needed (which is the case when proper stereo
// is present in the echo reference signal but the echo canceller does the
// processing in mono) downmix the echo reference by averaging the channel
// content (otherwise downmixing is done by selecting channel 0).
for (size_t band = 0; band < frame->size(); ++band) {
for (size_t ch = 1; ch < frame_num_channels; ++ch) {
for (size_t k = 0; k < kSubFrameLength; ++k) {
(*frame)[band][/*channel=*/0]
[sub_frame_index * kSubFrameLength + k] +=
(*frame)[band][ch][sub_frame_index * kSubFrameLength + k];
}
}
const float one_by_num_channels = 1.0f / frame_num_channels;
for (size_t k = 0; k < kSubFrameLength; ++k) {
(*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength +
k] *= one_by_num_channels;
}
}
}
for (size_t band = 0; band < frame->size(); ++band) {
(*sub_frame_view)[band][/*channel=*/0] = rtc::ArrayView<float>(
&(*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength],
kSubFrameLength);
}
} else {
RTC_DCHECK_EQ(frame_num_channels, sub_frame_num_channels);
for (size_t band = 0; band < frame->size(); ++band) {
for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) {
(*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
&(*frame)[band][channel][sub_frame_index * kSubFrameLength],
kSubFrameLength);
}
}
}
}
@ -107,16 +147,17 @@ void ProcessCaptureFrameContent(
AudioBuffer* linear_output,
AudioBuffer* capture,
bool level_change,
bool aec_reference_is_downmixed_stereo,
bool saturated_microphone_signal,
size_t sub_frame_index,
FrameBlocker* capture_blocker,
BlockFramer* linear_output_framer,
BlockFramer* output_framer,
BlockProcessor* block_processor,
std::vector<std::vector<std::vector<float>>>* linear_output_block,
Block* linear_output_block,
std::vector<std::vector<rtc::ArrayView<float>>>*
linear_output_sub_frame_view,
std::vector<std::vector<std::vector<float>>>* capture_block,
Block* capture_block,
std::vector<std::vector<rtc::ArrayView<float>>>* capture_sub_frame_view) {
FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view);
@ -130,8 +171,10 @@ void ProcessCaptureFrameContent(
capture_blocker->InsertSubFrameAndExtractBlock(*capture_sub_frame_view,
capture_block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
linear_output_block, capture_block);
block_processor->ProcessCapture(
/*echo_path_gain_change=*/level_change ||
aec_reference_is_downmixed_stereo,
saturated_microphone_signal, linear_output_block, capture_block);
output_framer->InsertBlockAndExtractSubFrame(*capture_block,
capture_sub_frame_view);
@ -142,22 +185,24 @@ void ProcessCaptureFrameContent(
}
}
void ProcessRemainingCaptureFrameContent(
bool level_change,
bool saturated_microphone_signal,
FrameBlocker* capture_blocker,
BlockFramer* linear_output_framer,
BlockFramer* output_framer,
BlockProcessor* block_processor,
std::vector<std::vector<std::vector<float>>>* linear_output_block,
std::vector<std::vector<std::vector<float>>>* block) {
void ProcessRemainingCaptureFrameContent(bool level_change,
bool aec_reference_is_downmixed_stereo,
bool saturated_microphone_signal,
FrameBlocker* capture_blocker,
BlockFramer* linear_output_framer,
BlockFramer* output_framer,
BlockProcessor* block_processor,
Block* linear_output_block,
Block* block) {
if (!capture_blocker->IsBlockAvailable()) {
return;
}
capture_blocker->ExtractBlock(block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
linear_output_block, block);
block_processor->ProcessCapture(
/*echo_path_gain_change=*/level_change ||
aec_reference_is_downmixed_stereo,
saturated_microphone_signal, linear_output_block, block);
output_framer->InsertBlock(*block);
if (linear_output_framer) {
@ -167,21 +212,22 @@ void ProcessRemainingCaptureFrameContent(
}
void BufferRenderFrameContent(
bool proper_downmix_needed,
std::vector<std::vector<std::vector<float>>>* render_frame,
size_t sub_frame_index,
FrameBlocker* render_blocker,
BlockProcessor* block_processor,
std::vector<std::vector<std::vector<float>>>* block,
Block* block,
std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
FillSubFrameView(render_frame, sub_frame_index, sub_frame_view);
FillSubFrameView(proper_downmix_needed, render_frame, sub_frame_index,
sub_frame_view);
render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
block_processor->BufferRender(*block);
}
void BufferRemainingRenderFrameContent(
FrameBlocker* render_blocker,
BlockProcessor* block_processor,
std::vector<std::vector<std::vector<float>>>* block) {
void BufferRemainingRenderFrameContent(FrameBlocker* render_blocker,
BlockProcessor* block_processor,
Block* block) {
if (!render_blocker->IsBlockAvailable()) {
return;
}
@ -213,6 +259,10 @@ void CopyBufferIntoFrame(const AudioBuffer& buffer,
EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
EchoCanceller3Config adjusted_cfg = config;
if (field_trial::IsEnabled("WebRTC-Aec3StereoContentDetectionKillSwitch")) {
adjusted_cfg.multi_channel.detect_stereo_content = false;
}
if (field_trial::IsEnabled("WebRTC-Aec3AntiHowlingMinimizationKillSwitch")) {
adjusted_cfg.suppressor.high_bands_suppression
.anti_howling_activation_threshold = 25.f;
@ -251,24 +301,39 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
adjusted_cfg.filter.initial_state_seconds = 2.0f;
}
if (field_trial::IsEnabled("WebRTC-Aec3HighPassFilterEchoReference")) {
adjusted_cfg.filter.high_pass_filter_echo_reference = true;
}
if (field_trial::IsEnabled("WebRTC-Aec3EchoSaturationDetectionKillSwitch")) {
adjusted_cfg.ep_strength.echo_can_saturate = false;
}
if (field_trial::IsEnabled("WebRTC-Aec3UseDot2ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.2f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot3ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.3f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot4ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.4f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot5ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.5f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot6ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.6f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot7ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.7f;
} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot8ReverbDefaultLen")) {
adjusted_cfg.ep_strength.default_len = 0.8f;
const std::string use_nearend_reverb_len_tunings =
field_trial::FindFullName("WebRTC-Aec3UseNearendReverbLen");
FieldTrialParameter<double> nearend_reverb_default_len(
"default_len", adjusted_cfg.ep_strength.default_len);
FieldTrialParameter<double> nearend_reverb_nearend_len(
"nearend_len", adjusted_cfg.ep_strength.nearend_len);
ParseFieldTrial({&nearend_reverb_default_len, &nearend_reverb_nearend_len},
use_nearend_reverb_len_tunings);
float default_len = static_cast<float>(nearend_reverb_default_len.Get());
float nearend_len = static_cast<float>(nearend_reverb_nearend_len.Get());
if (default_len > -1 && default_len < 1 && nearend_len > -1 &&
nearend_len < 1) {
adjusted_cfg.ep_strength.default_len =
static_cast<float>(nearend_reverb_default_len.Get());
adjusted_cfg.ep_strength.nearend_len =
static_cast<float>(nearend_reverb_nearend_len.Get());
}
if (field_trial::IsEnabled("WebRTC-Aec3ConservativeTailFreqResponse")) {
adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = true;
}
if (field_trial::IsDisabled("WebRTC-Aec3ConservativeTailFreqResponse")) {
adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = false;
}
if (field_trial::IsEnabled("WebRTC-Aec3ShortHeadroomKillSwitch")) {
@ -313,6 +378,14 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
false;
}
if (field_trial::IsEnabled("WebRTC-Aec3DelayEstimatorDetectPreEcho")) {
adjusted_cfg.delay.detect_pre_echo = true;
}
if (field_trial::IsDisabled("WebRTC-Aec3DelayEstimatorDetectPreEcho")) {
adjusted_cfg.delay.detect_pre_echo = false;
}
if (field_trial::IsEnabled("WebRTC-Aec3SensitiveDominantNearendActivation")) {
adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.5f;
} else if (field_trial::IsEnabled(
@ -368,6 +441,10 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf = .2f;
}
if (field_trial::IsEnabled("WebRTC-Aec3EnforceConservativeHfSuppression")) {
adjusted_cfg.suppressor.conservative_hf_suppression = true;
}
if (field_trial::IsEnabled("WebRTC-Aec3EnforceStationarityProperties")) {
adjusted_cfg.echo_audibility.use_stationarity_properties = true;
}
@ -443,8 +520,6 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
FieldTrialParameter<int> dominant_nearend_detection_trigger_threshold(
"dominant_nearend_detection_trigger_threshold",
adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold);
FieldTrialParameter<double> ep_strength_default_len(
"ep_strength_default_len", adjusted_cfg.ep_strength.default_len);
ParseFieldTrial(
{&nearend_tuning_mask_lf_enr_transparent,
@ -461,7 +536,7 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
&dominant_nearend_detection_enr_exit_threshold,
&dominant_nearend_detection_snr_threshold,
&dominant_nearend_detection_hold_duration,
&dominant_nearend_detection_trigger_threshold, &ep_strength_default_len},
&dominant_nearend_detection_trigger_threshold},
suppressor_tuning_override_trial_name);
adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent =
@ -498,8 +573,6 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
dominant_nearend_detection_hold_duration.Get();
adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold =
dominant_nearend_detection_trigger_threshold.Get();
adjusted_cfg.ep_strength.default_len =
static_cast<float>(ep_strength_default_len.Get());
// Field trial-based overrides of individual suppressor parameters.
RetrieveFieldTrialValue(
@ -561,8 +634,12 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
"WebRTC-Aec3SuppressorAntiHowlingGainOverride", 0.f, 10.f,
&adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain);
RetrieveFieldTrialValue("WebRTC-Aec3SuppressorEpStrengthDefaultLenOverride",
-1.f, 1.f, &adjusted_cfg.ep_strength.default_len);
// Field trial-based overrides of individual delay estimator parameters.
RetrieveFieldTrialValue("WebRTC-Aec3DelayEstimateSmoothingOverride", 0.f, 1.f,
&adjusted_cfg.delay.delay_estimate_smoothing);
RetrieveFieldTrialValue(
"WebRTC-Aec3DelayEstimateSmoothingDelayFoundOverride", 0.f, 1.f,
&adjusted_cfg.delay.delay_estimate_smoothing_delay_found);
return adjusted_cfg;
}
@ -570,6 +647,7 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
class EchoCanceller3::RenderWriter {
public:
RenderWriter(ApmDataDumper* data_dumper,
const EchoCanceller3Config& config,
SwapQueue<std::vector<std::vector<std::vector<float>>>,
Aec3RenderQueueItemVerifier>* render_transfer_queue,
size_t num_bands,
@ -586,7 +664,7 @@ class EchoCanceller3::RenderWriter {
ApmDataDumper* data_dumper_;
const size_t num_bands_;
const size_t num_channels_;
HighPassFilter high_pass_filter_;
std::unique_ptr<HighPassFilter> high_pass_filter_;
std::vector<std::vector<std::vector<float>>> render_queue_input_frame_;
SwapQueue<std::vector<std::vector<std::vector<float>>>,
Aec3RenderQueueItemVerifier>* render_transfer_queue_;
@ -594,6 +672,7 @@ class EchoCanceller3::RenderWriter {
EchoCanceller3::RenderWriter::RenderWriter(
ApmDataDumper* data_dumper,
const EchoCanceller3Config& config,
SwapQueue<std::vector<std::vector<std::vector<float>>>,
Aec3RenderQueueItemVerifier>* render_transfer_queue,
size_t num_bands,
@ -601,7 +680,6 @@ EchoCanceller3::RenderWriter::RenderWriter(
: data_dumper_(data_dumper),
num_bands_(num_bands),
num_channels_(num_channels),
high_pass_filter_(16000, num_channels),
render_queue_input_frame_(
num_bands_,
std::vector<std::vector<float>>(
@ -609,6 +687,9 @@ EchoCanceller3::RenderWriter::RenderWriter(
std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
render_transfer_queue_(render_transfer_queue) {
RTC_DCHECK(data_dumper);
if (config.filter.high_pass_filter_echo_reference) {
high_pass_filter_ = std::make_unique<HighPassFilter>(16000, num_channels);
}
}
EchoCanceller3::RenderWriter::~RenderWriter() = default;
@ -627,104 +708,124 @@ void EchoCanceller3::RenderWriter::Insert(const AudioBuffer& input) {
CopyBufferIntoFrame(input, num_bands_, num_channels_,
&render_queue_input_frame_);
high_pass_filter_.Process(&render_queue_input_frame_[0]);
if (high_pass_filter_) {
high_pass_filter_->Process(&render_queue_input_frame_[0]);
}
static_cast<void>(render_transfer_queue_->Insert(&render_queue_input_frame_));
}
int EchoCanceller3::instance_count_ = 0;
std::atomic<int> EchoCanceller3::instance_count_(0);
EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels)
: EchoCanceller3(AdjustConfig(config),
sample_rate_hz,
num_render_channels,
num_capture_channels,
std::unique_ptr<BlockProcessor>(
BlockProcessor::Create(AdjustConfig(config),
sample_rate_hz,
num_render_channels,
num_capture_channels))) {}
EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels,
std::unique_ptr<BlockProcessor> block_processor)
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
config_(config),
EchoCanceller3::EchoCanceller3(
const EchoCanceller3Config& config,
const absl::optional<EchoCanceller3Config>& multichannel_config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels)
: data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
config_(AdjustConfig(config)),
sample_rate_hz_(sample_rate_hz),
num_bands_(NumBandsForRate(sample_rate_hz_)),
num_render_channels_(num_render_channels),
num_render_input_channels_(num_render_channels),
num_capture_channels_(num_capture_channels),
config_selector_(AdjustConfig(config),
multichannel_config,
num_render_input_channels_),
multichannel_content_detector_(
config_selector_.active_config().multi_channel.detect_stereo_content,
num_render_input_channels_,
config_selector_.active_config()
.multi_channel.stereo_detection_threshold,
config_selector_.active_config()
.multi_channel.stereo_detection_timeout_threshold_seconds,
config_selector_.active_config()
.multi_channel.stereo_detection_hysteresis_seconds),
output_framer_(num_bands_, num_capture_channels_),
capture_blocker_(num_bands_, num_capture_channels_),
render_blocker_(num_bands_, num_render_channels_),
render_transfer_queue_(
kRenderTransferQueueSizeFrames,
std::vector<std::vector<std::vector<float>>>(
num_bands_,
std::vector<std::vector<float>>(
num_render_channels_,
num_render_input_channels_,
std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
Aec3RenderQueueItemVerifier(num_bands_,
num_render_channels_,
num_render_input_channels_,
AudioBuffer::kSplitBandSize)),
block_processor_(std::move(block_processor)),
render_queue_output_frame_(
num_bands_,
std::vector<std::vector<float>>(
num_render_channels_,
num_render_input_channels_,
std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
render_block_(
num_bands_,
std::vector<std::vector<float>>(num_render_channels_,
std::vector<float>(kBlockSize, 0.f))),
capture_block_(
num_bands_,
std::vector<std::vector<float>>(num_capture_channels_,
std::vector<float>(kBlockSize, 0.f))),
render_sub_frame_view_(
num_bands_,
std::vector<rtc::ArrayView<float>>(num_render_channels_)),
render_block_(num_bands_, num_render_input_channels_),
capture_block_(num_bands_, num_capture_channels_),
capture_sub_frame_view_(
num_bands_,
std::vector<rtc::ArrayView<float>>(num_capture_channels_)) {
RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
if (config_.delay.fixed_capture_delay_samples > 0) {
if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) {
block_delay_buffer_.reset(new BlockDelayBuffer(
num_capture_channels_, num_bands_, AudioBuffer::kSplitBandSize,
config_.delay.fixed_capture_delay_samples));
}
render_writer_.reset(new RenderWriter(data_dumper_.get(),
&render_transfer_queue_, num_bands_,
num_render_channels_));
render_writer_.reset(new RenderWriter(
data_dumper_.get(), config_selector_.active_config(),
&render_transfer_queue_, num_bands_, num_render_input_channels_));
RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000);
RTC_DCHECK_GE(kMaxNumBands, num_bands_);
if (config_.filter.export_linear_aec_output) {
linear_output_framer_.reset(new BlockFramer(1, num_capture_channels_));
if (config_selector_.active_config().filter.export_linear_aec_output) {
linear_output_framer_.reset(
new BlockFramer(/*num_bands=*/1, num_capture_channels_));
linear_output_block_ =
std::make_unique<std::vector<std::vector<std::vector<float>>>>(
1, std::vector<std::vector<float>>(
num_capture_channels_, std::vector<float>(kBlockSize, 0.f)));
std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_),
linear_output_sub_frame_view_ =
std::vector<std::vector<rtc::ArrayView<float>>>(
1, std::vector<rtc::ArrayView<float>>(num_capture_channels_));
}
Initialize();
RTC_LOG(LS_INFO) << "AEC3 created with sample rate: " << sample_rate_hz_
<< " Hz, num render channels: " << num_render_input_channels_
<< ", num capture channels: " << num_capture_channels_;
}
EchoCanceller3::~EchoCanceller3() = default;
void EchoCanceller3::Initialize() {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
num_render_channels_to_aec_ =
multichannel_content_detector_.IsProperMultiChannelContentDetected()
? num_render_input_channels_
: 1;
config_selector_.Update(
multichannel_content_detector_.IsProperMultiChannelContentDetected());
render_block_.SetNumChannels(num_render_channels_to_aec_);
render_blocker_.reset(
new FrameBlocker(num_bands_, num_render_channels_to_aec_));
block_processor_.reset(BlockProcessor::Create(
config_selector_.active_config(), sample_rate_hz_,
num_render_channels_to_aec_, num_capture_channels_));
render_sub_frame_view_ = std::vector<std::vector<rtc::ArrayView<float>>>(
num_bands_,
std::vector<rtc::ArrayView<float>>(num_render_channels_to_aec_));
}
void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) {
RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
RTC_DCHECK_EQ(render.num_channels(), num_render_channels_);
RTC_DCHECK_EQ(render.num_channels(), num_render_input_channels_);
data_dumper_->DumpRaw("aec3_call_order",
static_cast<int>(EchoCanceller3ApiCall::kRender));
@ -764,7 +865,7 @@ void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
if (linear_output && !linear_output_framer_) {
RTC_LOG(LS_ERROR) << "Trying to retrieve the linear AEC output without "
"properly configuring AEC3.";
RTC_NOTREACHED();
RTC_DCHECK_NOTREACHED();
}
// Report capture call in the metrics and periodically update API call
@ -772,7 +873,7 @@ void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
api_call_metrics_.ReportCaptureCall();
// Optionally delay the capture signal.
if (config_.delay.fixed_capture_delay_samples > 0) {
if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) {
RTC_DCHECK(block_delay_buffer_);
block_delay_buffer_->DelaySignal(capture);
}
@ -784,22 +885,26 @@ void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
EmptyRenderQueue();
ProcessCaptureFrameContent(linear_output, capture, level_change,
saturated_microphone_signal_, 0, &capture_blocker_,
linear_output_framer_.get(), &output_framer_,
block_processor_.get(), linear_output_block_.get(),
&linear_output_sub_frame_view_, &capture_block_,
&capture_sub_frame_view_);
ProcessCaptureFrameContent(
linear_output, capture, level_change,
multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
saturated_microphone_signal_, 0, &capture_blocker_,
linear_output_framer_.get(), &output_framer_, block_processor_.get(),
linear_output_block_.get(), &linear_output_sub_frame_view_,
&capture_block_, &capture_sub_frame_view_);
ProcessCaptureFrameContent(linear_output, capture, level_change,
saturated_microphone_signal_, 1, &capture_blocker_,
linear_output_framer_.get(), &output_framer_,
block_processor_.get(), linear_output_block_.get(),
&linear_output_sub_frame_view_, &capture_block_,
&capture_sub_frame_view_);
ProcessCaptureFrameContent(
linear_output, capture, level_change,
multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
saturated_microphone_signal_, 1, &capture_blocker_,
linear_output_framer_.get(), &output_framer_, block_processor_.get(),
linear_output_block_.get(), &linear_output_sub_frame_view_,
&capture_block_, &capture_sub_frame_view_);
ProcessRemainingCaptureFrameContent(
level_change, saturated_microphone_signal_, &capture_blocker_,
level_change,
multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
saturated_microphone_signal_, &capture_blocker_,
linear_output_framer_.get(), &output_framer_, block_processor_.get(),
linear_output_block_.get(), &capture_block_);
@ -819,29 +924,37 @@ void EchoCanceller3::SetAudioBufferDelay(int delay_ms) {
block_processor_->SetAudioBufferDelay(delay_ms);
}
void EchoCanceller3::SetCaptureOutputUsage(bool capture_output_used) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
block_processor_->SetCaptureOutputUsage(capture_output_used);
}
bool EchoCanceller3::ActiveProcessing() const {
return true;
}
EchoCanceller3Config EchoCanceller3::CreateDefaultConfig(
size_t num_render_channels,
size_t num_capture_channels) {
EchoCanceller3Config EchoCanceller3::CreateDefaultMultichannelConfig() {
EchoCanceller3Config cfg;
if (num_render_channels > 1) {
// Use shorter and more rapidly adapting coarse filter to compensate for
// thge increased number of total filter parameters to adapt.
cfg.filter.coarse.length_blocks = 11;
cfg.filter.coarse.rate = 0.95f;
cfg.filter.coarse_initial.length_blocks = 11;
cfg.filter.coarse_initial.rate = 0.95f;
// Use shorter and more rapidly adapting coarse filter to compensate for
// thge increased number of total filter parameters to adapt.
cfg.filter.coarse.length_blocks = 11;
cfg.filter.coarse.rate = 0.95f;
cfg.filter.coarse_initial.length_blocks = 11;
cfg.filter.coarse_initial.rate = 0.95f;
// Use more concervative suppressor behavior for non-nearend speech.
cfg.suppressor.normal_tuning.max_dec_factor_lf = 0.35f;
cfg.suppressor.normal_tuning.max_inc_factor = 1.5f;
}
// Use more concervative suppressor behavior for non-nearend speech.
cfg.suppressor.normal_tuning.max_dec_factor_lf = 0.35f;
cfg.suppressor.normal_tuning.max_inc_factor = 1.5f;
return cfg;
}
void EchoCanceller3::SetBlockProcessorForTesting(
std::unique_ptr<BlockProcessor> block_processor) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
RTC_DCHECK(block_processor);
block_processor_ = std::move(block_processor);
}
void EchoCanceller3::EmptyRenderQueue() {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
bool frame_to_buffer =
@ -850,16 +963,27 @@ void EchoCanceller3::EmptyRenderQueue() {
// Report render call in the metrics.
api_call_metrics_.ReportRenderCall();
BufferRenderFrameContent(&render_queue_output_frame_, 0, &render_blocker_,
block_processor_.get(), &render_block_,
&render_sub_frame_view_);
if (multichannel_content_detector_.UpdateDetection(
render_queue_output_frame_)) {
// Reinitialize the AEC when proper stereo is detected.
Initialize();
}
BufferRenderFrameContent(&render_queue_output_frame_, 1, &render_blocker_,
block_processor_.get(), &render_block_,
&render_sub_frame_view_);
// Buffer frame content.
BufferRenderFrameContent(
/*proper_downmix_needed=*/multichannel_content_detector_
.IsTemporaryMultiChannelContentDetected(),
&render_queue_output_frame_, 0, render_blocker_.get(),
block_processor_.get(), &render_block_, &render_sub_frame_view_);
BufferRemainingRenderFrameContent(&render_blocker_, block_processor_.get(),
&render_block_);
BufferRenderFrameContent(
/*proper_downmix_needed=*/multichannel_content_detector_
.IsTemporaryMultiChannelContentDetected(),
&render_queue_output_frame_, 1, render_blocker_.get(),
block_processor_.get(), &render_block_, &render_sub_frame_view_);
BufferRemainingRenderFrameContent(render_blocker_.get(),
block_processor_.get(), &render_block_);
frame_to_buffer =
render_transfer_queue_.Remove(&render_queue_output_frame_);