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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc')
| -rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc | 656 |
1 files changed, 656 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc new file mode 100644 index 0000000000..416e287751 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc @@ -0,0 +1,656 @@ +/* + * 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/aec_dump_based_simulator.h" + +#include <iostream> +#include <memory> + +#include "modules/audio_processing/echo_control_mobile_impl.h" +#include "modules/audio_processing/logging/apm_data_dumper.h" +#include "modules/audio_processing/test/aec_dump_based_simulator.h" +#include "modules/audio_processing/test/protobuf_utils.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" + +namespace webrtc { +namespace test { +namespace { + +// Verify output bitexactness for the fixed interface. +// TODO(peah): Check whether it would make sense to add a threshold +// to use for checking the bitexactness in a soft manner. +bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg, + const Int16Frame& frame) { + if (sizeof(frame.data[0]) * frame.data.size() != msg.output_data().size()) { + return false; + } else { + const int16_t* frame_data = frame.data.data(); + for (int k = 0; k < frame.num_channels * frame.samples_per_channel; ++k) { + if (msg.output_data().data()[k] != frame_data[k]) { + return false; + } + } + } + return true; +} + +// Verify output bitexactness for the float interface. +bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg, + const StreamConfig& out_config, + const ChannelBuffer<float>& out_buf) { + if (static_cast<size_t>(msg.output_channel_size()) != + out_config.num_channels() || + msg.output_channel(0).size() != out_config.num_frames()) { + return false; + } else { + for (int ch = 0; ch < msg.output_channel_size(); ++ch) { + for (size_t sample = 0; sample < out_config.num_frames(); ++sample) { + if (msg.output_channel(ch).data()[sample] != + out_buf.channels()[ch][sample]) { + return false; + } + } + } + } + return true; +} + +// Selectively reads the next proto-buf message from dump-file or string input. +// Returns a bool indicating whether a new message was available. +bool ReadNextMessage(bool use_dump_file, + FILE* dump_input_file, + std::stringstream& input, + webrtc::audioproc::Event& event_msg) { + if (use_dump_file) { + return ReadMessageFromFile(dump_input_file, &event_msg); + } + return ReadMessageFromString(&input, &event_msg); +} + +} // namespace + +AecDumpBasedSimulator::AecDumpBasedSimulator( + const SimulationSettings& settings, + rtc::scoped_refptr<AudioProcessing> audio_processing, + std::unique_ptr<AudioProcessingBuilder> ap_builder) + : AudioProcessingSimulator(settings, + std::move(audio_processing), + std::move(ap_builder)) { + MaybeOpenCallOrderFile(); +} + +AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; + +void AecDumpBasedSimulator::PrepareProcessStreamCall( + const webrtc::audioproc::Stream& msg) { + if (msg.has_input_data()) { + // Fixed interface processing. + // Verify interface invariance. + RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || + interface_used_ == InterfaceType::kNotSpecified); + interface_used_ = InterfaceType::kFixedInterface; + + // Populate input buffer. + RTC_CHECK_EQ(sizeof(fwd_frame_.data[0]) * fwd_frame_.data.size(), + msg.input_data().size()); + memcpy(fwd_frame_.data.data(), msg.input_data().data(), + msg.input_data().size()); + } else { + // Float interface processing. + // Verify interface invariance. + RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface || + interface_used_ == InterfaceType::kNotSpecified); + interface_used_ = InterfaceType::kFloatInterface; + + RTC_CHECK_EQ(in_buf_->num_channels(), + static_cast<size_t>(msg.input_channel_size())); + + // Populate input buffer. + for (size_t i = 0; i < in_buf_->num_channels(); ++i) { + RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]), + msg.input_channel(i).size()); + std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(), + msg.input_channel(i).size()); + } + } + + if (artificial_nearend_buffer_reader_) { + if (artificial_nearend_buffer_reader_->Read( + artificial_nearend_buf_.get())) { + if (msg.has_input_data()) { + int16_t* fwd_frame_data = fwd_frame_.data.data(); + for (size_t k = 0; k < in_buf_->num_frames(); ++k) { + fwd_frame_data[k] = rtc::saturated_cast<int16_t>( + fwd_frame_data[k] + + static_cast<int16_t>(32767 * + artificial_nearend_buf_->channels()[0][k])); + } + } else { + for (int i = 0; i < msg.input_channel_size(); ++i) { + for (size_t k = 0; k < in_buf_->num_frames(); ++k) { + in_buf_->channels()[i][k] += + artificial_nearend_buf_->channels()[0][k]; + in_buf_->channels()[i][k] = std::min( + 32767.f, std::max(-32768.f, in_buf_->channels()[i][k])); + } + } + } + } else { + if (!artificial_nearend_eof_reported_) { + std::cout << "The artificial nearend file ended before the recording."; + artificial_nearend_eof_reported_ = true; + } + } + } + + if (!settings_.use_stream_delay || *settings_.use_stream_delay) { + if (!settings_.stream_delay) { + if (msg.has_delay()) { + RTC_CHECK_EQ(AudioProcessing::kNoError, + ap_->set_stream_delay_ms(msg.delay())); + } + } else { + RTC_CHECK_EQ(AudioProcessing::kNoError, + ap_->set_stream_delay_ms(*settings_.stream_delay)); + } + } + + if (settings_.override_key_pressed.has_value()) { + // Key pressed state overridden. + ap_->set_stream_key_pressed(*settings_.override_key_pressed); + } else { + // Set the recorded key pressed state. + if (msg.has_keypress()) { + ap_->set_stream_key_pressed(msg.keypress()); + } + } + + // Set the applied input level if available. + aec_dump_applied_input_level_ = + msg.has_applied_input_volume() + ? absl::optional<int>(msg.applied_input_volume()) + : absl::nullopt; +} + +void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( + const webrtc::audioproc::Stream& msg) { + if (bitexact_output_) { + if (interface_used_ == InterfaceType::kFixedInterface) { + bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); + } else { + bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); + } + } +} + +void AecDumpBasedSimulator::PrepareReverseProcessStreamCall( + const webrtc::audioproc::ReverseStream& msg) { + if (msg.has_data()) { + // Fixed interface processing. + // Verify interface invariance. + RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || + interface_used_ == InterfaceType::kNotSpecified); + interface_used_ = InterfaceType::kFixedInterface; + + // Populate input buffer. + RTC_CHECK_EQ(sizeof(rev_frame_.data[0]) * rev_frame_.data.size(), + msg.data().size()); + memcpy(rev_frame_.data.data(), msg.data().data(), msg.data().size()); + } else { + // Float interface processing. + // Verify interface invariance. + RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface || + interface_used_ == InterfaceType::kNotSpecified); + interface_used_ = InterfaceType::kFloatInterface; + + RTC_CHECK_EQ(reverse_in_buf_->num_channels(), + static_cast<size_t>(msg.channel_size())); + + // Populate input buffer. + for (int i = 0; i < msg.channel_size(); ++i) { + RTC_CHECK_EQ(reverse_in_buf_->num_frames() * + sizeof(*reverse_in_buf_->channels()[i]), + msg.channel(i).size()); + std::memcpy(reverse_in_buf_->channels()[i], msg.channel(i).data(), + msg.channel(i).size()); + } + } +} + +void AecDumpBasedSimulator::Process() { + ConfigureAudioProcessor(); + + if (settings_.artificial_nearend_filename) { + std::unique_ptr<WavReader> artificial_nearend_file( + new WavReader(settings_.artificial_nearend_filename->c_str())); + + RTC_CHECK_EQ(1, artificial_nearend_file->num_channels()) + << "Only mono files for the artificial nearend are supported, " + "reverted to not using the artificial nearend file"; + + const int sample_rate_hz = artificial_nearend_file->sample_rate(); + artificial_nearend_buffer_reader_.reset( + new ChannelBufferWavReader(std::move(artificial_nearend_file))); + artificial_nearend_buf_.reset(new ChannelBuffer<float>( + rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1)); + } + + const bool use_dump_file = !settings_.aec_dump_input_string.has_value(); + std::stringstream input; + if (use_dump_file) { + dump_input_file_ = + OpenFile(settings_.aec_dump_input_filename->c_str(), "rb"); + } else { + input << settings_.aec_dump_input_string.value(); + } + + webrtc::audioproc::Event event_msg; + int capture_frames_since_init = 0; + int init_index = 0; + while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) { + SelectivelyToggleDataDumping(init_index, capture_frames_since_init); + HandleEvent(event_msg, capture_frames_since_init, init_index); + + // Perfom an early exit if the init block to process has been fully + // processed + if (finished_processing_specified_init_block_) { + break; + } + RTC_CHECK(!settings_.init_to_process || + *settings_.init_to_process >= init_index); + } + + if (use_dump_file) { + fclose(dump_input_file_); + } + + DetachAecDump(); +} + +void AecDumpBasedSimulator::Analyze() { + const bool use_dump_file = !settings_.aec_dump_input_string.has_value(); + std::stringstream input; + if (use_dump_file) { + dump_input_file_ = + OpenFile(settings_.aec_dump_input_filename->c_str(), "rb"); + } else { + input << settings_.aec_dump_input_string.value(); + } + + webrtc::audioproc::Event event_msg; + int num_capture_frames = 0; + int num_render_frames = 0; + int init_index = 0; + while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) { + if (event_msg.type() == webrtc::audioproc::Event::INIT) { + ++init_index; + constexpr float kNumFramesPerSecond = 100.f; + float capture_time_seconds = num_capture_frames / kNumFramesPerSecond; + float render_time_seconds = num_render_frames / kNumFramesPerSecond; + + std::cout << "Inits:" << std::endl; + std::cout << init_index << ": -->" << std::endl; + std::cout << " Time:" << std::endl; + std::cout << " Capture: " << capture_time_seconds << " s (" + << num_capture_frames << " frames) " << std::endl; + std::cout << " Render: " << render_time_seconds << " s (" + << num_render_frames << " frames) " << std::endl; + } else if (event_msg.type() == webrtc::audioproc::Event::STREAM) { + ++num_capture_frames; + } else if (event_msg.type() == webrtc::audioproc::Event::REVERSE_STREAM) { + ++num_render_frames; + } + } + + if (use_dump_file) { + fclose(dump_input_file_); + } +} + +void AecDumpBasedSimulator::HandleEvent( + const webrtc::audioproc::Event& event_msg, + int& capture_frames_since_init, + int& init_index) { + switch (event_msg.type()) { + case webrtc::audioproc::Event::INIT: + RTC_CHECK(event_msg.has_init()); + ++init_index; + capture_frames_since_init = 0; + HandleMessage(event_msg.init(), init_index); + break; + case webrtc::audioproc::Event::STREAM: + RTC_CHECK(event_msg.has_stream()); + ++capture_frames_since_init; + HandleMessage(event_msg.stream()); + break; + case webrtc::audioproc::Event::REVERSE_STREAM: + RTC_CHECK(event_msg.has_reverse_stream()); + HandleMessage(event_msg.reverse_stream()); + break; + case webrtc::audioproc::Event::CONFIG: + RTC_CHECK(event_msg.has_config()); + HandleMessage(event_msg.config()); + break; + case webrtc::audioproc::Event::RUNTIME_SETTING: + HandleMessage(event_msg.runtime_setting()); + break; + case webrtc::audioproc::Event::UNKNOWN_EVENT: + RTC_CHECK_NOTREACHED(); + } +} + +void AecDumpBasedSimulator::HandleMessage( + const webrtc::audioproc::Config& msg) { + if (settings_.use_verbose_logging) { + std::cout << "Config at frame:" << std::endl; + std::cout << " Forward: " << get_num_process_stream_calls() << std::endl; + std::cout << " Reverse: " << get_num_reverse_process_stream_calls() + << std::endl; + } + + if (!settings_.discard_all_settings_in_aecdump) { + if (settings_.use_verbose_logging) { + std::cout << "Setting used in config:" << std::endl; + } + AudioProcessing::Config apm_config = ap_->GetConfig(); + + if (msg.has_aec_enabled() || settings_.use_aec) { + bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled(); + apm_config.echo_canceller.enabled = enable; + if (settings_.use_verbose_logging) { + std::cout << " aec_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_aecm_enabled() || settings_.use_aecm) { + bool enable = + settings_.use_aecm ? *settings_.use_aecm : msg.aecm_enabled(); + apm_config.echo_canceller.enabled |= enable; + apm_config.echo_canceller.mobile_mode = enable; + if (settings_.use_verbose_logging) { + std::cout << " aecm_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_aecm_comfort_noise_enabled() && + msg.aecm_comfort_noise_enabled()) { + RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM comfort noise"; + } + + if (msg.has_aecm_routing_mode() && + static_cast<webrtc::EchoControlMobileImpl::RoutingMode>( + msg.aecm_routing_mode()) != EchoControlMobileImpl::kSpeakerphone) { + RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM routing mode: " + << msg.aecm_routing_mode(); + } + + if (msg.has_agc_enabled() || settings_.use_agc) { + bool enable = settings_.use_agc ? *settings_.use_agc : msg.agc_enabled(); + apm_config.gain_controller1.enabled = enable; + if (settings_.use_verbose_logging) { + std::cout << " agc_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_agc_mode() || settings_.agc_mode) { + int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode(); + apm_config.gain_controller1.mode = + static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>( + mode); + if (settings_.use_verbose_logging) { + std::cout << " agc_mode: " << mode << std::endl; + } + } + + if (msg.has_agc_limiter_enabled() || settings_.use_agc_limiter) { + bool enable = settings_.use_agc_limiter ? *settings_.use_agc_limiter + : msg.agc_limiter_enabled(); + apm_config.gain_controller1.enable_limiter = enable; + if (settings_.use_verbose_logging) { + std::cout << " agc_limiter_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (settings_.use_agc2) { + bool enable = *settings_.use_agc2; + apm_config.gain_controller2.enabled = enable; + if (settings_.agc2_fixed_gain_db) { + apm_config.gain_controller2.fixed_digital.gain_db = + *settings_.agc2_fixed_gain_db; + } + if (settings_.use_verbose_logging) { + std::cout << " agc2_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_noise_robust_agc_enabled()) { + apm_config.gain_controller1.analog_gain_controller.enabled = + settings_.use_analog_agc ? *settings_.use_analog_agc + : msg.noise_robust_agc_enabled(); + if (settings_.use_verbose_logging) { + std::cout << " noise_robust_agc_enabled: " + << (msg.noise_robust_agc_enabled() ? "true" : "false") + << std::endl; + } + } + + if (msg.has_transient_suppression_enabled() || settings_.use_ts) { + bool enable = settings_.use_ts ? *settings_.use_ts + : msg.transient_suppression_enabled(); + apm_config.transient_suppression.enabled = enable; + if (settings_.use_verbose_logging) { + std::cout << " transient_suppression_enabled: " + << (enable ? "true" : "false") << std::endl; + } + } + + if (msg.has_hpf_enabled() || settings_.use_hpf) { + bool enable = settings_.use_hpf ? *settings_.use_hpf : msg.hpf_enabled(); + apm_config.high_pass_filter.enabled = enable; + if (settings_.use_verbose_logging) { + std::cout << " hpf_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_ns_enabled() || settings_.use_ns) { + bool enable = settings_.use_ns ? *settings_.use_ns : msg.ns_enabled(); + apm_config.noise_suppression.enabled = enable; + if (settings_.use_verbose_logging) { + std::cout << " ns_enabled: " << (enable ? "true" : "false") + << std::endl; + } + } + + if (msg.has_ns_level() || settings_.ns_level) { + int level = settings_.ns_level ? *settings_.ns_level : msg.ns_level(); + apm_config.noise_suppression.level = + static_cast<AudioProcessing::Config::NoiseSuppression::Level>(level); + if (settings_.use_verbose_logging) { + std::cout << " ns_level: " << level << std::endl; + } + } + + if (msg.has_pre_amplifier_enabled() || settings_.use_pre_amplifier) { + const bool enable = settings_.use_pre_amplifier + ? *settings_.use_pre_amplifier + : msg.pre_amplifier_enabled(); + apm_config.pre_amplifier.enabled = enable; + } + + if (msg.has_pre_amplifier_fixed_gain_factor() || + settings_.pre_amplifier_gain_factor) { + const float gain = settings_.pre_amplifier_gain_factor + ? *settings_.pre_amplifier_gain_factor + : msg.pre_amplifier_fixed_gain_factor(); + apm_config.pre_amplifier.fixed_gain_factor = gain; + } + + if (settings_.use_verbose_logging && msg.has_experiments_description() && + !msg.experiments_description().empty()) { + std::cout << " experiments not included by default in the simulation: " + << msg.experiments_description() << std::endl; + } + + ap_->ApplyConfig(apm_config); + } +} + +void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg, + int init_index) { + RTC_CHECK(msg.has_sample_rate()); + RTC_CHECK(msg.has_num_input_channels()); + RTC_CHECK(msg.has_num_reverse_channels()); + RTC_CHECK(msg.has_reverse_sample_rate()); + + // Do not perform the init if the init block to process is fully processed + if (settings_.init_to_process && *settings_.init_to_process < init_index) { + finished_processing_specified_init_block_ = true; + } + + MaybeOpenCallOrderFile(); + + if (settings_.use_verbose_logging) { + std::cout << "Init at frame:" << std::endl; + std::cout << " Forward: " << get_num_process_stream_calls() << std::endl; + std::cout << " Reverse: " << get_num_reverse_process_stream_calls() + << std::endl; + } + + int num_output_channels; + if (settings_.output_num_channels) { + num_output_channels = *settings_.output_num_channels; + } else { + num_output_channels = msg.has_num_output_channels() + ? msg.num_output_channels() + : msg.num_input_channels(); + } + + int output_sample_rate; + if (settings_.output_sample_rate_hz) { + output_sample_rate = *settings_.output_sample_rate_hz; + } else { + output_sample_rate = msg.has_output_sample_rate() ? msg.output_sample_rate() + : msg.sample_rate(); + } + + int num_reverse_output_channels; + if (settings_.reverse_output_num_channels) { + num_reverse_output_channels = *settings_.reverse_output_num_channels; + } else { + num_reverse_output_channels = msg.has_num_reverse_output_channels() + ? msg.num_reverse_output_channels() + : msg.num_reverse_channels(); + } + + int reverse_output_sample_rate; + if (settings_.reverse_output_sample_rate_hz) { + reverse_output_sample_rate = *settings_.reverse_output_sample_rate_hz; + } else { + reverse_output_sample_rate = msg.has_reverse_output_sample_rate() + ? msg.reverse_output_sample_rate() + : msg.reverse_sample_rate(); + } + + SetupBuffersConfigsOutputs( + msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), + reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, + msg.num_reverse_channels(), num_reverse_output_channels); +} + +void AecDumpBasedSimulator::HandleMessage( + const webrtc::audioproc::Stream& msg) { + if (call_order_output_file_) { + *call_order_output_file_ << "c"; + } + PrepareProcessStreamCall(msg); + ProcessStream(interface_used_ == InterfaceType::kFixedInterface); + VerifyProcessStreamBitExactness(msg); +} + +void AecDumpBasedSimulator::HandleMessage( + const webrtc::audioproc::ReverseStream& msg) { + if (call_order_output_file_) { + *call_order_output_file_ << "r"; + } + PrepareReverseProcessStreamCall(msg); + ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); +} + +void AecDumpBasedSimulator::HandleMessage( + const webrtc::audioproc::RuntimeSetting& msg) { + RTC_CHECK(ap_.get()); + if (msg.has_capture_pre_gain()) { + // Handle capture pre-gain runtime setting only if not overridden. + const bool pre_amplifier_overridden = + (!settings_.use_pre_amplifier || *settings_.use_pre_amplifier) && + !settings_.pre_amplifier_gain_factor; + const bool capture_level_adjustment_overridden = + (!settings_.use_capture_level_adjustment || + *settings_.use_capture_level_adjustment) && + !settings_.pre_gain_factor; + if (pre_amplifier_overridden || capture_level_adjustment_overridden) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreateCapturePreGain( + msg.capture_pre_gain())); + } + } else if (msg.has_capture_post_gain()) { + // Handle capture post-gain runtime setting only if not overridden. + if ((!settings_.use_capture_level_adjustment || + *settings_.use_capture_level_adjustment) && + !settings_.post_gain_factor) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreateCapturePreGain( + msg.capture_pre_gain())); + } + } else if (msg.has_capture_fixed_post_gain()) { + // Handle capture fixed-post-gain runtime setting only if not overridden. + if ((!settings_.use_agc2 || *settings_.use_agc2) && + !settings_.agc2_fixed_gain_db) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain( + msg.capture_fixed_post_gain())); + } + } else if (msg.has_playout_volume_change()) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange( + msg.playout_volume_change())); + } else if (msg.has_playout_audio_device_change()) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange( + {msg.playout_audio_device_change().id(), + msg.playout_audio_device_change().max_volume()})); + } else if (msg.has_capture_output_used()) { + ap_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( + msg.capture_output_used())); + } +} + +void AecDumpBasedSimulator::MaybeOpenCallOrderFile() { + if (settings_.call_order_output_filename.has_value()) { + const std::string filename = settings_.store_intermediate_output + ? *settings_.call_order_output_filename + + "_" + + std::to_string(output_reset_counter_) + : *settings_.call_order_output_filename; + call_order_output_file_ = std::make_unique<std::ofstream>(filename); + } +} + +} // namespace test +} // namespace webrtc |