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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.
*/
#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 "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/constructor_magic.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
namespace test {
// 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> stream_drift_samples;
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<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_hpf;
absl::optional<bool> use_ns;
absl::optional<bool> use_ts;
absl::optional<bool> use_ie;
absl::optional<bool> use_vad;
absl::optional<bool> use_le;
absl::optional<bool> use_all;
absl::optional<int> aec_suppression_level;
absl::optional<bool> use_delay_agnostic;
absl::optional<bool> use_extended_filter;
absl::optional<bool> use_drift_compensation;
absl::optional<bool> use_legacy_aec;
absl::optional<bool> use_experimental_agc;
absl::optional<bool> use_experimental_agc_agc2_level_estimator;
absl::optional<bool> experimental_agc_disable_digital_adaptive;
absl::optional<bool> experimental_agc_analyze_before_aec;
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;
float agc2_fixed_gain_db;
AudioProcessing::Config::GainController2::LevelEstimator
agc2_adaptive_level_estimator;
float pre_amplifier_gain_factor;
absl::optional<int> vad_likelihood;
absl::optional<int> ns_level;
absl::optional<bool> use_refined_adaptive_filter;
int initial_mic_level;
bool simulate_mic_gain = false;
absl::optional<int> simulated_mic_kind;
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;
absl::optional<std::string> dump_internal_data_output_dir;
absl::optional<std::string> call_order_input_filename;
absl::optional<std::string> call_order_output_filename;
absl::optional<std::string> aec_settings_filename;
};
// Copies samples present in a ChannelBuffer into an AudioFrame.
void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest);
// Provides common functionality for performing audioprocessing simulations.
class AudioProcessingSimulator {
public:
static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
AudioProcessingSimulator(const SimulationSettings& settings,
std::unique_ptr<AudioProcessingBuilder> ap_builder);
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_;
}
// 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 CreateAudioProcessor();
void DestroyAudioProcessor();
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);
const SimulationSettings settings_;
std::unique_ptr<AudioProcessing> ap_;
std::unique_ptr<AudioProcessingBuilder> ap_builder_;
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_;
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_;
AudioFrame rev_frame_;
AudioFrame fwd_frame_;
bool bitexact_output_ = true;
int aec_dump_mic_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_writer_;
std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
ApiCallStatistics api_call_statistics_;
std::ofstream residual_echo_likelihood_graph_writer_;
int analog_mic_level_;
FakeRecordingDevice fake_recording_device_;
TaskQueueForTest worker_queue_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_