| /* |
| * 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 <algorithm> |
| #include <cstddef> |
| #include <cstdint> |
| #include <cstdio> |
| #include <cstring> |
| #include <fstream> |
| #include <iostream> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/base/nullability.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/scoped_refptr.h" |
| #include "common_audio/channel_buffer.h" |
| #include "common_audio/wav_file.h" |
| #include "modules/audio_processing/debug.pb.h" |
| #include "modules/audio_processing/echo_control_mobile_impl.h" |
| #include "modules/audio_processing/test/aec_dump_based_simulator.h" |
| #include "modules/audio_processing/test/audio_processing_simulator.h" |
| #include "modules/audio_processing/test/protobuf_utils.h" |
| #include "modules/audio_processing/test/test_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, |
| absl::Nonnull<scoped_refptr<AudioProcessing>> audio_processing) |
| : AudioProcessingSimulator(settings, std::move(audio_processing)) { |
| 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() |
| ? std::optional<int>(msg.applied_input_volume()) |
| : std::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 |
