| /* |
| * 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 <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "api/audio/echo_canceller3_config_json.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/echo_cancellation_impl.h" |
| #include "modules/audio_processing/echo_control_mobile_impl.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/logging.h" |
| #include "rtc_base/strings/json.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/stringutils.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| // Helper for reading JSON from a file and parsing it to an AEC3 configuration. |
| EchoCanceller3Config ReadAec3ConfigFromJsonFile(const std::string& filename) { |
| std::string json_string; |
| std::string s; |
| 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; |
| } |
| return Aec3ConfigFromJsonString(json_string); |
| } |
| |
| 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) { |
| rtc::StringBuilder ss; |
| ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter |
| << wav_name.substr(wav_name.size() - 4); |
| return ss.Release(); |
| } |
| |
| 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_agc2) { |
| apm_config.gain_controller2.enabled = *settings_.use_agc2; |
| apm_config.gain_controller2.fixed_gain_db = settings_.agc2_fixed_gain_db; |
| if (settings_.agc2_use_adaptive_gain) { |
| apm_config.gain_controller2.adaptive_digital_mode = |
| *settings_.agc2_use_adaptive_gain; |
| } |
| } |
| 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; |
| } |
| |
| bool use_aec2 = settings_.use_aec && *settings_.use_aec; |
| bool use_aec3 = settings_.use_aec3 && *settings_.use_aec3; |
| bool use_aecm = settings_.use_aecm && *settings_.use_aecm; |
| if (use_aec2 || use_aec3 || use_aecm) { |
| apm_config.echo_canceller.enabled = true; |
| apm_config.echo_canceller.mobile_mode = use_aecm; |
| } |
| |
| if (settings_.use_aec3 && *settings_.use_aec3) { |
| EchoCanceller3Config cfg; |
| if (settings_.aec3_settings_filename) { |
| if (settings_.use_verbose_logging) { |
| std::cout << "Reading AEC3 Parameters from JSON input." << std::endl; |
| } |
| cfg = ReadAec3ConfigFromJsonFile(*settings_.aec3_settings_filename); |
| } |
| echo_control_factory.reset(new EchoCanceller3Factory(cfg)); |
| |
| if (settings_.print_aec3_parameter_values) { |
| if (!settings_.use_quiet_output) { |
| std::cout << "AEC3 settings:" << std::endl; |
| } |
| std::cout << Aec3ConfigToJsonString(cfg) << std::endl; |
| } |
| } |
| |
| if (settings_.use_drift_compensation && *settings_.use_drift_compensation) { |
| RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AEC2 drift compensation"; |
| } |
| if (settings_.aec_suppression_level) { |
| auto level = static_cast<webrtc::EchoCancellationImpl::SuppressionLevel>( |
| *settings_.aec_suppression_level); |
| if (level == |
| webrtc::EchoCancellationImpl::SuppressionLevel::kLowSuppression) { |
| RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AEC2 low suppression"; |
| } else { |
| apm_config.echo_canceller.legacy_moderate_suppression_level = |
| (level == webrtc::EchoCancellationImpl::SuppressionLevel:: |
| kModerateSuppression); |
| } |
| } |
| |
| 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, |
| !!settings_.experimental_agc_analyze_before_aec && |
| *settings_.experimental_agc_analyze_before_aec)); |
| 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_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_.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 |
