blob: e9ca1518e89cf21868494406780ecceca50a8ab6 [file] [log] [blame]
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/test/audio_processing_simulator.h"
#include <algorithm>
#include <fstream>
#include <iostream>
#include <sstream>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/audio/echo_canceller3_factory.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/test/fake_recording_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/json.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringutils.h"
namespace webrtc {
namespace test {
namespace {
void ReadParam(const Json::Value& root, std::string param_name, bool* param) {
RTC_CHECK(param);
bool v;
if (rtc::GetBoolFromJsonObject(root, param_name, &v)) {
*param = v;
std::cout << param_name << ":" << (*param ? "true" : "false") << std::endl;
}
}
void ReadParam(const Json::Value& root, std::string param_name, size_t* param) {
RTC_CHECK(param);
int v;
if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
*param = v;
std::cout << param_name << ":" << *param << std::endl;
}
}
void ReadParam(const Json::Value& root, std::string param_name, int* param) {
RTC_CHECK(param);
int v;
if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
*param = v;
std::cout << param_name << ":" << *param << std::endl;
}
}
void ReadParam(const Json::Value& root, std::string param_name, float* param) {
RTC_CHECK(param);
double v;
if (rtc::GetDoubleFromJsonObject(root, param_name, &v)) {
*param = static_cast<float>(v);
std::cout << param_name << ":" << *param << std::endl;
}
}
void ReadParam(const Json::Value& root,
std::string param_name,
EchoCanceller3Config::Filter::MainConfiguration* param) {
RTC_CHECK(param);
Json::Value json_array;
if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
std::vector<double> v;
rtc::JsonArrayToDoubleVector(json_array, &v);
if (v.size() != 5) {
std::cout << "Incorrect array size for " << param_name << std::endl;
RTC_CHECK(false);
}
param->length_blocks = static_cast<size_t>(v[0]);
param->leakage_converged = static_cast<float>(v[1]);
param->leakage_diverged = static_cast<float>(v[2]);
param->error_floor = static_cast<float>(v[3]);
param->noise_gate = static_cast<float>(v[4]);
std::cout << param_name << ":"
<< "[" << param->length_blocks << "," << param->leakage_converged
<< "," << param->leakage_diverged << "," << param->error_floor
<< "," << param->noise_gate << "]" << std::endl;
}
}
void ReadParam(const Json::Value& root,
std::string param_name,
EchoCanceller3Config::Filter::ShadowConfiguration* param) {
RTC_CHECK(param);
Json::Value json_array;
if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
std::vector<double> v;
rtc::JsonArrayToDoubleVector(json_array, &v);
if (v.size() != 3) {
std::cout << "Incorrect array size for " << param_name << std::endl;
RTC_CHECK(false);
}
param->length_blocks = static_cast<size_t>(v[0]);
param->rate = static_cast<float>(v[1]);
param->noise_gate = static_cast<float>(v[2]);
std::cout << param_name << ":"
<< "[" << param->length_blocks << "," << param->rate << ","
<< param->noise_gate << "]" << std::endl;
}
}
void ReadParam(const Json::Value& root,
std::string param_name,
EchoCanceller3Config::GainUpdates::GainChanges* param) {
RTC_CHECK(param);
Json::Value json_array;
if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
std::vector<double> v;
rtc::JsonArrayToDoubleVector(json_array, &v);
if (v.size() != 6) {
std::cout << "Incorrect array size for " << param_name << std::endl;
RTC_CHECK(false);
}
param->max_inc = static_cast<float>(v[0]);
param->max_dec = static_cast<float>(v[1]);
param->rate_inc = static_cast<float>(v[2]);
param->rate_dec = static_cast<float>(v[3]);
param->min_inc = static_cast<float>(v[4]);
param->min_dec = static_cast<float>(v[5]);
std::cout << param_name << ":"
<< "[" << param->max_inc << "," << param->max_dec << ","
<< param->rate_inc << "," << param->rate_dec << ","
<< param->min_inc << "," << param->min_dec << "]" << std::endl;
}
}
EchoCanceller3Config ParseAec3Parameters(const std::string& filename) {
EchoCanceller3Config cfg;
Json::Value root;
std::string s;
std::string json_string;
std::ifstream f(filename.c_str());
if (f.fail()) {
std::cout << "Failed to open the file " << filename << std::endl;
RTC_CHECK(false);
}
while (std::getline(f, s)) {
json_string += s;
}
bool success = Json::Reader().parse(json_string, root);
if (!success) {
std::cout << "Incorrect JSON format:" << std::endl;
std::cout << json_string << std::endl;
RTC_CHECK(false);
}
std::cout << "AEC3 Parameters from JSON input:" << std::endl;
Json::Value section;
if (rtc::GetValueFromJsonObject(root, "delay", &section)) {
ReadParam(section, "default_delay", &cfg.delay.default_delay);
ReadParam(section, "down_sampling_factor", &cfg.delay.down_sampling_factor);
ReadParam(section, "num_filters", &cfg.delay.num_filters);
ReadParam(section, "api_call_jitter_blocks",
&cfg.delay.api_call_jitter_blocks);
ReadParam(section, "min_echo_path_delay_blocks",
&cfg.delay.min_echo_path_delay_blocks);
ReadParam(section, "delay_headroom_blocks",
&cfg.delay.delay_headroom_blocks);
ReadParam(section, "hysteresis_limit_1_blocks",
&cfg.delay.hysteresis_limit_1_blocks);
ReadParam(section, "hysteresis_limit_2_blocks",
&cfg.delay.hysteresis_limit_2_blocks);
ReadParam(section, "skew_hysteresis_blocks",
&cfg.delay.skew_hysteresis_blocks);
}
if (rtc::GetValueFromJsonObject(root, "filter", &section)) {
ReadParam(section, "main", &cfg.filter.main);
ReadParam(section, "shadow", &cfg.filter.shadow);
ReadParam(section, "main_initial", &cfg.filter.main_initial);
ReadParam(section, "shadow_initial", &cfg.filter.shadow_initial);
}
if (rtc::GetValueFromJsonObject(root, "erle", &section)) {
ReadParam(section, "min", &cfg.erle.min);
ReadParam(section, "max_l", &cfg.erle.max_l);
ReadParam(section, "max_h", &cfg.erle.max_h);
}
if (rtc::GetValueFromJsonObject(root, "ep_strength", &section)) {
ReadParam(section, "lf", &cfg.ep_strength.lf);
ReadParam(section, "mf", &cfg.ep_strength.mf);
ReadParam(section, "hf", &cfg.ep_strength.hf);
ReadParam(section, "default_len", &cfg.ep_strength.default_len);
ReadParam(section, "reverb_based_on_render",
&cfg.ep_strength.reverb_based_on_render);
ReadParam(section, "echo_can_saturate", &cfg.ep_strength.echo_can_saturate);
ReadParam(section, "bounded_erl", &cfg.ep_strength.bounded_erl);
}
if (rtc::GetValueFromJsonObject(root, "gain_mask", &section)) {
ReadParam(section, "m1", &cfg.gain_mask.m1);
ReadParam(section, "m2", &cfg.gain_mask.m2);
ReadParam(section, "m3", &cfg.gain_mask.m3);
ReadParam(section, "m5", &cfg.gain_mask.m5);
ReadParam(section, "m6", &cfg.gain_mask.m6);
ReadParam(section, "m7", &cfg.gain_mask.m7);
ReadParam(section, "m8", &cfg.gain_mask.m8);
ReadParam(section, "m9", &cfg.gain_mask.m9);
ReadParam(section, "gain_curve_offset", &cfg.gain_mask.gain_curve_offset);
ReadParam(section, "gain_curve_slope", &cfg.gain_mask.gain_curve_slope);
ReadParam(section, "temporal_masking_lf",
&cfg.gain_mask.temporal_masking_lf);
ReadParam(section, "temporal_masking_hf",
&cfg.gain_mask.temporal_masking_hf);
ReadParam(section, "temporal_masking_lf_bands",
&cfg.gain_mask.temporal_masking_lf_bands);
}
if (rtc::GetValueFromJsonObject(root, "echo_audibility", &section)) {
ReadParam(section, "low_render_limit",
&cfg.echo_audibility.low_render_limit);
ReadParam(section, "normal_render_limit",
&cfg.echo_audibility.normal_render_limit);
ReadParam(section, "floor_power", &cfg.echo_audibility.floor_power);
ReadParam(section, "audibility_threshold_lf",
&cfg.echo_audibility.audibility_threshold_lf);
ReadParam(section, "audibility_threshold_mf",
&cfg.echo_audibility.audibility_threshold_mf);
ReadParam(section, "audibility_threshold_hf",
&cfg.echo_audibility.audibility_threshold_hf);
ReadParam(section, "use_stationary_properties",
&cfg.echo_audibility.use_stationary_properties);
}
if (rtc::GetValueFromJsonObject(root, "gain_updates", &section)) {
ReadParam(section, "low_noise", &cfg.gain_updates.low_noise);
ReadParam(section, "initial", &cfg.gain_updates.initial);
ReadParam(section, "normal", &cfg.gain_updates.normal);
ReadParam(section, "saturation", &cfg.gain_updates.saturation);
ReadParam(section, "nonlinear", &cfg.gain_updates.nonlinear);
ReadParam(section, "floor_first_increase",
&cfg.gain_updates.floor_first_increase);
}
if (rtc::GetValueFromJsonObject(root, "echo_removal_control", &section)) {
Json::Value subsection;
if (rtc::GetValueFromJsonObject(section, "gain_rampup", &subsection)) {
ReadParam(subsection, "initial_gain",
&cfg.echo_removal_control.gain_rampup.initial_gain);
ReadParam(subsection, "first_non_zero_gain",
&cfg.echo_removal_control.gain_rampup.first_non_zero_gain);
ReadParam(subsection, "non_zero_gain_blocks",
&cfg.echo_removal_control.gain_rampup.non_zero_gain_blocks);
ReadParam(subsection, "full_gain_blocks",
&cfg.echo_removal_control.gain_rampup.full_gain_blocks);
}
ReadParam(section, "has_clock_drift",
&cfg.echo_removal_control.has_clock_drift);
ReadParam(section, "linear_and_stable_echo_path",
&cfg.echo_removal_control.linear_and_stable_echo_path);
}
if (rtc::GetValueFromJsonObject(root, "echo_model", &section)) {
Json::Value subsection;
ReadParam(section, "noise_floor_hold", &cfg.echo_model.noise_floor_hold);
ReadParam(section, "min_noise_floor_power",
&cfg.echo_model.min_noise_floor_power);
ReadParam(section, "stationary_gate_slope",
&cfg.echo_model.stationary_gate_slope);
ReadParam(section, "noise_gate_power", &cfg.echo_model.noise_gate_power);
ReadParam(section, "noise_gate_slope", &cfg.echo_model.noise_gate_slope);
ReadParam(section, "render_pre_window_size",
&cfg.echo_model.render_pre_window_size);
ReadParam(section, "render_post_window_size",
&cfg.echo_model.render_post_window_size);
ReadParam(section, "render_pre_window_size_init",
&cfg.echo_model.render_pre_window_size_init);
ReadParam(section, "render_post_window_size_init",
&cfg.echo_model.render_post_window_size_init);
ReadParam(section, "nonlinear_hold", &cfg.echo_model.nonlinear_hold);
ReadParam(section, "nonlinear_release", &cfg.echo_model.nonlinear_release);
}
std::cout << std::endl;
return cfg;
}
void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
// Copy the data from the input buffer.
std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
S16ToFloat(src.data(), tmp.size(), tmp.data());
Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_,
dest->channels());
}
std::string GetIndexedOutputWavFilename(const std::string& wav_name,
int counter) {
std::stringstream ss;
ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
<< wav_name.substr(wav_name.size() - 4);
return ss.str();
}
void WriteEchoLikelihoodGraphFileHeader(std::ofstream* output_file) {
(*output_file) << "import numpy as np" << std::endl
<< "import matplotlib.pyplot as plt" << std::endl
<< "y = np.array([";
}
void WriteEchoLikelihoodGraphFileFooter(std::ofstream* output_file) {
(*output_file) << "])" << std::endl
<< "if __name__ == '__main__':" << std::endl
<< " x = np.arange(len(y))*.01" << std::endl
<< " plt.plot(x, y)" << std::endl
<< " plt.ylabel('Echo likelihood')" << std::endl
<< " plt.xlabel('Time (s)')" << std::endl
<< " plt.ylim([0,1])" << std::endl
<< " plt.show()" << std::endl;
}
} // namespace
SimulationSettings::SimulationSettings() = default;
SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
SimulationSettings::~SimulationSettings() = default;
void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
int16_t* dest_data = dest->mutable_data();
for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
dest_data[sample * dest->num_channels_ + ch] =
src.channels()[ch][sample] * 32767;
}
}
}
AudioProcessingSimulator::AudioProcessingSimulator(
const SimulationSettings& settings,
std::unique_ptr<AudioProcessingBuilder> ap_builder)
: settings_(settings),
ap_builder_(ap_builder ? std::move(ap_builder)
: absl::make_unique<AudioProcessingBuilder>()),
analog_mic_level_(settings.initial_mic_level),
fake_recording_device_(
settings.initial_mic_level,
settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
worker_queue_("file_writer_task_queue") {
if (settings_.ed_graph_output_filename &&
!settings_.ed_graph_output_filename->empty()) {
residual_echo_likelihood_graph_writer_.open(
*settings_.ed_graph_output_filename);
RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open());
WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_);
}
if (settings_.simulate_mic_gain)
RTC_LOG(LS_VERBOSE) << "Simulating analog mic gain";
}
AudioProcessingSimulator::~AudioProcessingSimulator() {
if (residual_echo_likelihood_graph_writer_.is_open()) {
WriteEchoLikelihoodGraphFileFooter(&residual_echo_likelihood_graph_writer_);
residual_echo_likelihood_graph_writer_.close();
}
}
AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
int64_t interval = rtc::TimeNanos() - start_time_;
proc_time_->sum += interval;
proc_time_->max = std::max(proc_time_->max, interval);
proc_time_->min = std::min(proc_time_->min, interval);
}
void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
// Optionally use the fake recording device to simulate analog gain.
if (settings_.simulate_mic_gain) {
if (settings_.aec_dump_input_filename) {
// When the analog gain is simulated and an AEC dump is used as input, set
// the undo level to |aec_dump_mic_level_| to virtually restore the
// unmodified microphone signal level.
fake_recording_device_.SetUndoMicLevel(aec_dump_mic_level_);
}
if (fixed_interface) {
fake_recording_device_.SimulateAnalogGain(&fwd_frame_);
} else {
fake_recording_device_.SimulateAnalogGain(in_buf_.get());
}
// Notify the current mic level to AGC.
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_stream_analog_level(
fake_recording_device_.MicLevel()));
} else {
// Notify the current mic level to AGC.
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_stream_analog_level(
settings_.aec_dump_input_filename ? aec_dump_mic_level_
: analog_mic_level_));
}
// Process the current audio frame.
if (fixed_interface) {
{
const auto st = ScopedTimer(mutable_proc_time());
RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
}
CopyFromAudioFrame(fwd_frame_, out_buf_.get());
} else {
const auto st = ScopedTimer(mutable_proc_time());
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->ProcessStream(in_buf_->channels(), in_config_,
out_config_, out_buf_->channels()));
}
// Store the mic level suggested by AGC.
// Note that when the analog gain is simulated and an AEC dump is used as
// input, |analog_mic_level_| will not be used with set_stream_analog_level().
analog_mic_level_ = ap_->gain_control()->stream_analog_level();
if (settings_.simulate_mic_gain) {
fake_recording_device_.SetMicLevel(analog_mic_level_);
}
if (buffer_writer_) {
buffer_writer_->Write(*out_buf_);
}
if (residual_echo_likelihood_graph_writer_.is_open()) {
auto stats = ap_->GetStatistics();
residual_echo_likelihood_graph_writer_ << stats.residual_echo_likelihood
<< ", ";
}
++num_process_stream_calls_;
}
void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
if (fixed_interface) {
const auto st = ScopedTimer(mutable_proc_time());
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->ProcessReverseStream(&rev_frame_));
CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get());
} else {
const auto st = ScopedTimer(mutable_proc_time());
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->ProcessReverseStream(
reverse_in_buf_->channels(), reverse_in_config_,
reverse_out_config_, reverse_out_buf_->channels()));
}
if (reverse_buffer_writer_) {
reverse_buffer_writer_->Write(*reverse_out_buf_);
}
++num_reverse_process_stream_calls_;
}
void AudioProcessingSimulator::SetupBuffersConfigsOutputs(
int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_input_sample_rate_hz,
int reverse_output_sample_rate_hz,
int input_num_channels,
int output_num_channels,
int reverse_input_num_channels,
int reverse_output_num_channels) {
in_config_ = StreamConfig(input_sample_rate_hz, input_num_channels);
in_buf_.reset(new ChannelBuffer<float>(
rtc::CheckedDivExact(input_sample_rate_hz, kChunksPerSecond),
input_num_channels));
reverse_in_config_ =
StreamConfig(reverse_input_sample_rate_hz, reverse_input_num_channels);
reverse_in_buf_.reset(new ChannelBuffer<float>(
rtc::CheckedDivExact(reverse_input_sample_rate_hz, kChunksPerSecond),
reverse_input_num_channels));
out_config_ = StreamConfig(output_sample_rate_hz, output_num_channels);
out_buf_.reset(new ChannelBuffer<float>(
rtc::CheckedDivExact(output_sample_rate_hz, kChunksPerSecond),
output_num_channels));
reverse_out_config_ =
StreamConfig(reverse_output_sample_rate_hz, reverse_output_num_channels);
reverse_out_buf_.reset(new ChannelBuffer<float>(
rtc::CheckedDivExact(reverse_output_sample_rate_hz, kChunksPerSecond),
reverse_output_num_channels));
fwd_frame_.sample_rate_hz_ = input_sample_rate_hz;
fwd_frame_.samples_per_channel_ =
rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond);
fwd_frame_.num_channels_ = input_num_channels;
rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz;
rev_frame_.samples_per_channel_ =
rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond);
rev_frame_.num_channels_ = reverse_input_num_channels;
if (settings_.use_verbose_logging) {
rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
std::cout << "Sample rates:" << std::endl;
std::cout << " Forward input: " << input_sample_rate_hz << std::endl;
std::cout << " Forward output: " << output_sample_rate_hz << std::endl;
std::cout << " Reverse input: " << reverse_input_sample_rate_hz
<< std::endl;
std::cout << " Reverse output: " << reverse_output_sample_rate_hz
<< std::endl;
std::cout << "Number of channels: " << std::endl;
std::cout << " Forward input: " << input_num_channels << std::endl;
std::cout << " Forward output: " << output_num_channels << std::endl;
std::cout << " Reverse input: " << reverse_input_num_channels << std::endl;
std::cout << " Reverse output: " << reverse_output_num_channels
<< std::endl;
}
SetupOutput();
}
void AudioProcessingSimulator::SetupOutput() {
if (settings_.output_filename) {
std::string filename;
if (settings_.store_intermediate_output) {
filename = GetIndexedOutputWavFilename(*settings_.output_filename,
output_reset_counter_);
} else {
filename = *settings_.output_filename;
}
std::unique_ptr<WavWriter> out_file(
new WavWriter(filename, out_config_.sample_rate_hz(),
static_cast<size_t>(out_config_.num_channels())));
buffer_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
}
if (settings_.reverse_output_filename) {
std::string filename;
if (settings_.store_intermediate_output) {
filename = GetIndexedOutputWavFilename(*settings_.reverse_output_filename,
output_reset_counter_);
} else {
filename = *settings_.reverse_output_filename;
}
std::unique_ptr<WavWriter> reverse_out_file(
new WavWriter(filename, reverse_out_config_.sample_rate_hz(),
static_cast<size_t>(reverse_out_config_.num_channels())));
reverse_buffer_writer_.reset(
new ChannelBufferWavWriter(std::move(reverse_out_file)));
}
++output_reset_counter_;
}
void AudioProcessingSimulator::DestroyAudioProcessor() {
if (settings_.aec_dump_output_filename) {
ap_->DetachAecDump();
}
}
void AudioProcessingSimulator::CreateAudioProcessor() {
Config config;
AudioProcessing::Config apm_config;
std::unique_ptr<EchoControlFactory> echo_control_factory;
if (settings_.use_ts) {
config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
}
if (settings_.use_ie) {
config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie));
}
if (settings_.use_agc2) {
apm_config.gain_controller2.enabled = *settings_.use_agc2;
apm_config.gain_controller2.fixed_gain_db = settings_.agc2_fixed_gain_db;
}
if (settings_.use_pre_amplifier) {
apm_config.pre_amplifier.enabled = *settings_.use_pre_amplifier;
apm_config.pre_amplifier.fixed_gain_factor =
settings_.pre_amplifier_gain_factor;
}
if (settings_.use_aec3 && *settings_.use_aec3) {
EchoCanceller3Config cfg;
if (settings_.aec3_settings_filename) {
cfg = ParseAec3Parameters(*settings_.aec3_settings_filename);
}
echo_control_factory.reset(new EchoCanceller3Factory(cfg));
}
if (settings_.use_hpf) {
apm_config.high_pass_filter.enabled = *settings_.use_hpf;
}
if (settings_.use_refined_adaptive_filter) {
config.Set<RefinedAdaptiveFilter>(
new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
}
config.Set<ExtendedFilter>(new ExtendedFilter(
!settings_.use_extended_filter || *settings_.use_extended_filter));
config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
*settings_.use_delay_agnostic));
config.Set<ExperimentalAgc>(new ExperimentalAgc(
!settings_.use_experimental_agc || *settings_.use_experimental_agc,
!!settings_.use_experimental_agc_agc2_level_estimator &&
*settings_.use_experimental_agc_agc2_level_estimator,
!!settings_.use_experimental_agc_agc2_digital_adaptive &&
*settings_.use_experimental_agc_agc2_digital_adaptive));
if (settings_.use_ed) {
apm_config.residual_echo_detector.enabled = *settings_.use_ed;
}
RTC_CHECK(ap_builder_);
ap_.reset((*ap_builder_)
.SetEchoControlFactory(std::move(echo_control_factory))
.Create(config));
RTC_CHECK(ap_);
ap_->ApplyConfig(apm_config);
if (settings_.use_aec) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_cancellation()->Enable(*settings_.use_aec));
}
if (settings_.use_aecm) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_control_mobile()->Enable(*settings_.use_aecm));
}
if (settings_.use_agc) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->Enable(*settings_.use_agc));
}
if (settings_.use_ns) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->noise_suppression()->Enable(*settings_.use_ns));
}
if (settings_.use_le) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->level_estimator()->Enable(*settings_.use_le));
}
if (settings_.use_vad) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->voice_detection()->Enable(*settings_.use_vad));
}
if (settings_.use_agc_limiter) {
RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->gain_control()->enable_limiter(
*settings_.use_agc_limiter));
}
if (settings_.agc_target_level) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_target_level_dbfs(
*settings_.agc_target_level));
}
if (settings_.agc_compression_gain) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_compression_gain_db(
*settings_.agc_compression_gain));
}
if (settings_.agc_mode) {
RTC_CHECK_EQ(
AudioProcessing::kNoError,
ap_->gain_control()->set_mode(
static_cast<webrtc::GainControl::Mode>(*settings_.agc_mode)));
}
if (settings_.use_drift_compensation) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_cancellation()->enable_drift_compensation(
*settings_.use_drift_compensation));
}
if (settings_.aec_suppression_level) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_cancellation()->set_suppression_level(
static_cast<webrtc::EchoCancellation::SuppressionLevel>(
*settings_.aec_suppression_level)));
}
if (settings_.aecm_routing_mode) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_control_mobile()->set_routing_mode(
static_cast<webrtc::EchoControlMobile::RoutingMode>(
*settings_.aecm_routing_mode)));
}
if (settings_.use_aecm_comfort_noise) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_control_mobile()->enable_comfort_noise(
*settings_.use_aecm_comfort_noise));
}
if (settings_.vad_likelihood) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->voice_detection()->set_likelihood(
static_cast<webrtc::VoiceDetection::Likelihood>(
*settings_.vad_likelihood)));
}
if (settings_.ns_level) {
RTC_CHECK_EQ(
AudioProcessing::kNoError,
ap_->noise_suppression()->set_level(
static_cast<NoiseSuppression::Level>(*settings_.ns_level)));
}
if (settings_.use_ts) {
ap_->set_stream_key_pressed(*settings_.use_ts);
}
if (settings_.aec_dump_output_filename) {
ap_->AttachAecDump(AecDumpFactory::Create(
*settings_.aec_dump_output_filename, -1, &worker_queue_));
}
}
} // namespace test
} // namespace webrtc