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webrtc/modules/audio_processing/test/audio_processing_simulator.cc
Alex Loiko 9489c3a2ea Optionally disable digital gain control in ExperimentalAgc. 
This CL adds a flag to optionally disable the digital gain control in
ExperimentalAgc. With the flag, Experimental Agc (henceforth AGC1)
only controls the adaptive analog gain. This flag can be combined to
that which activates AGC2. That way, one can enable the hybrid AGC
configuration AGC1 analog only + AGC2 fixed+adaptive digital.

Previously, there was a flag "use_agc2_digital_adaptive" in
AgcManagerDirect. Our ambition was that to activate the hybrid mode
described above with this flag. The behavior of the flag was not
implemented.

To activate the hybrid mode after this CL, set
ExperimentalAgc::digital_adaptive_disabled=true and
AudioProcessing::Config::GainController2::enabled=true.

We also add flags for these settings in audioproc_f.
Then the required settings are currently

  audioproc_f --agc2 1 --agc 1 --experimental_agc 1 \
      --experimental_agc_disable_digital_adaptive 1 \
      -i [INPUT]

Bug: webrtc:7494
Change-Id: Iea798dc3899cec83d30ba71caba787262fcaef41
Reviewed-on: https://webrtc-review.googlesource.com/89740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24249}
2018-08-09 13:37:30 +00:00

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/*
* 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_.experimental_agc_disable_digital_adaptive &&
*settings_.experimental_agc_disable_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
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