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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h')
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1 files changed, 247 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h new file mode 100644 index 0000000000..e40d818bd8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h @@ -0,0 +1,247 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ +#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ + +#include <algorithm> +#include <fstream> +#include <limits> +#include <memory> +#include <string> + +#include "absl/types/optional.h" +#include "common_audio/channel_buffer.h" +#include "common_audio/include/audio_util.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "modules/audio_processing/test/api_call_statistics.h" +#include "modules/audio_processing/test/fake_recording_device.h" +#include "modules/audio_processing/test/test_utils.h" +#include "rtc_base/task_queue_for_test.h" +#include "rtc_base/time_utils.h" + +namespace webrtc { +namespace test { + +static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs; + +struct Int16Frame { + void SetFormat(int sample_rate_hz, int num_channels) { + this->sample_rate_hz = sample_rate_hz; + samples_per_channel = + rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond); + this->num_channels = num_channels; + config = StreamConfig(sample_rate_hz, num_channels); + data.resize(num_channels * samples_per_channel); + } + + void CopyTo(ChannelBuffer<float>* dest) { + RTC_DCHECK(dest); + RTC_CHECK_EQ(num_channels, dest->num_channels()); + RTC_CHECK_EQ(samples_per_channel, dest->num_frames()); + // Copy the data from the input buffer. + std::vector<float> tmp(samples_per_channel * num_channels); + S16ToFloat(data.data(), tmp.size(), tmp.data()); + Deinterleave(tmp.data(), samples_per_channel, num_channels, + dest->channels()); + } + + void CopyFrom(const ChannelBuffer<float>& src) { + RTC_CHECK_EQ(src.num_channels(), num_channels); + RTC_CHECK_EQ(src.num_frames(), samples_per_channel); + data.resize(num_channels * samples_per_channel); + int16_t* dest_data = data.data(); + for (int ch = 0; ch < num_channels; ++ch) { + for (int sample = 0; sample < samples_per_channel; ++sample) { + dest_data[sample * num_channels + ch] = + src.channels()[ch][sample] * 32767; + } + } + } + + int sample_rate_hz; + int samples_per_channel; + int num_channels; + + StreamConfig config; + + std::vector<int16_t> data; +}; + +// Holds all the parameters available for controlling the simulation. +struct SimulationSettings { + SimulationSettings(); + SimulationSettings(const SimulationSettings&); + ~SimulationSettings(); + absl::optional<int> stream_delay; + absl::optional<bool> use_stream_delay; + absl::optional<int> output_sample_rate_hz; + absl::optional<int> output_num_channels; + absl::optional<int> reverse_output_sample_rate_hz; + absl::optional<int> reverse_output_num_channels; + absl::optional<std::string> output_filename; + absl::optional<std::string> reverse_output_filename; + absl::optional<std::string> input_filename; + absl::optional<std::string> reverse_input_filename; + absl::optional<std::string> artificial_nearend_filename; + absl::optional<std::string> linear_aec_output_filename; + absl::optional<bool> use_aec; + absl::optional<bool> use_aecm; + absl::optional<bool> use_ed; // Residual Echo Detector. + absl::optional<std::string> ed_graph_output_filename; + absl::optional<bool> use_agc; + absl::optional<bool> use_agc2; + absl::optional<bool> use_pre_amplifier; + absl::optional<bool> use_capture_level_adjustment; + absl::optional<bool> use_analog_mic_gain_emulation; + absl::optional<bool> use_hpf; + absl::optional<bool> use_ns; + absl::optional<int> use_ts; + absl::optional<bool> use_analog_agc; + absl::optional<bool> use_all; + absl::optional<bool> analog_agc_use_digital_adaptive_controller; + absl::optional<int> agc_mode; + absl::optional<int> agc_target_level; + absl::optional<bool> use_agc_limiter; + absl::optional<int> agc_compression_gain; + absl::optional<bool> agc2_use_adaptive_gain; + absl::optional<float> agc2_fixed_gain_db; + absl::optional<float> pre_amplifier_gain_factor; + absl::optional<float> pre_gain_factor; + absl::optional<float> post_gain_factor; + absl::optional<float> analog_mic_gain_emulation_initial_level; + absl::optional<int> ns_level; + absl::optional<bool> ns_analysis_on_linear_aec_output; + absl::optional<bool> override_key_pressed; + absl::optional<int> maximum_internal_processing_rate; + int initial_mic_level; + bool simulate_mic_gain = false; + absl::optional<bool> multi_channel_render; + absl::optional<bool> multi_channel_capture; + absl::optional<int> simulated_mic_kind; + absl::optional<int> frame_for_sending_capture_output_used_false; + absl::optional<int> frame_for_sending_capture_output_used_true; + bool report_performance = false; + absl::optional<std::string> performance_report_output_filename; + bool report_bitexactness = false; + bool use_verbose_logging = false; + bool use_quiet_output = false; + bool discard_all_settings_in_aecdump = true; + absl::optional<std::string> aec_dump_input_filename; + absl::optional<std::string> aec_dump_output_filename; + bool fixed_interface = false; + bool store_intermediate_output = false; + bool print_aec_parameter_values = false; + bool dump_internal_data = false; + WavFile::SampleFormat wav_output_format = WavFile::SampleFormat::kInt16; + absl::optional<std::string> dump_internal_data_output_dir; + absl::optional<int> dump_set_to_use; + absl::optional<std::string> call_order_input_filename; + absl::optional<std::string> call_order_output_filename; + absl::optional<std::string> aec_settings_filename; + absl::optional<absl::string_view> aec_dump_input_string; + std::vector<float>* processed_capture_samples = nullptr; + bool analysis_only = false; + absl::optional<int> dump_start_frame; + absl::optional<int> dump_end_frame; + absl::optional<int> init_to_process; +}; + +// Provides common functionality for performing audioprocessing simulations. +class AudioProcessingSimulator { + public: + AudioProcessingSimulator(const SimulationSettings& settings, + rtc::scoped_refptr<AudioProcessing> audio_processing, + std::unique_ptr<AudioProcessingBuilder> ap_builder); + + AudioProcessingSimulator() = delete; + AudioProcessingSimulator(const AudioProcessingSimulator&) = delete; + AudioProcessingSimulator& operator=(const AudioProcessingSimulator&) = delete; + + virtual ~AudioProcessingSimulator(); + + // Processes the data in the input. + virtual void Process() = 0; + + // Returns the execution times of all AudioProcessing calls. + const ApiCallStatistics& GetApiCallStatistics() const { + return api_call_statistics_; + } + + // Analyzes the data in the input and reports the resulting statistics. + virtual void Analyze() = 0; + + // Reports whether the processed recording was bitexact. + bool OutputWasBitexact() { return bitexact_output_; } + + size_t get_num_process_stream_calls() { return num_process_stream_calls_; } + size_t get_num_reverse_process_stream_calls() { + return num_reverse_process_stream_calls_; + } + + protected: + void ProcessStream(bool fixed_interface); + void ProcessReverseStream(bool fixed_interface); + void ConfigureAudioProcessor(); + void DetachAecDump(); + void 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); + void SelectivelyToggleDataDumping(int init_index, + int capture_frames_since_init) const; + + const SimulationSettings settings_; + rtc::scoped_refptr<AudioProcessing> ap_; + + std::unique_ptr<ChannelBuffer<float>> in_buf_; + std::unique_ptr<ChannelBuffer<float>> out_buf_; + std::unique_ptr<ChannelBuffer<float>> reverse_in_buf_; + std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_; + std::vector<std::array<float, 160>> linear_aec_output_buf_; + StreamConfig in_config_; + StreamConfig out_config_; + StreamConfig reverse_in_config_; + StreamConfig reverse_out_config_; + std::unique_ptr<ChannelBufferWavReader> buffer_reader_; + std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_; + Int16Frame rev_frame_; + Int16Frame fwd_frame_; + bool bitexact_output_ = true; + absl::optional<int> aec_dump_applied_input_level_ = 0; + + protected: + size_t output_reset_counter_ = 0; + + private: + void SetupOutput(); + + size_t num_process_stream_calls_ = 0; + size_t num_reverse_process_stream_calls_ = 0; + std::unique_ptr<ChannelBufferWavWriter> buffer_file_writer_; + std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_file_writer_; + std::unique_ptr<ChannelBufferVectorWriter> buffer_memory_writer_; + std::unique_ptr<WavWriter> linear_aec_output_file_writer_; + ApiCallStatistics api_call_statistics_; + std::ofstream residual_echo_likelihood_graph_writer_; + int applied_input_volume_; + FakeRecordingDevice fake_recording_device_; + + TaskQueueForTest worker_queue_; +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ |