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Side by Side Diff: webrtc/modules/audio_processing/test/audio_processing_simulator.h

Issue 2834643002: audioproc_f with simulated mic analog gain (Closed)
Patch Set: AEC dump + fake rec device bugfix Created 3 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
13 13
14 #include <algorithm> 14 #include <algorithm>
15 #include <fstream> 15 #include <fstream>
16 #include <limits> 16 #include <limits>
17 #include <memory> 17 #include <memory>
18 #include <string> 18 #include <string>
19 19
20 #include "webrtc/common_audio/channel_buffer.h" 20 #include "webrtc/common_audio/channel_buffer.h"
21 #include "webrtc/modules/audio_processing/include/audio_processing.h" 21 #include "webrtc/modules/audio_processing/include/audio_processing.h"
22 #include "webrtc/modules/audio_processing/test/fake_recording_device.h"
22 #include "webrtc/modules/audio_processing/test/test_utils.h" 23 #include "webrtc/modules/audio_processing/test/test_utils.h"
23 #include "webrtc/rtc_base/constructormagic.h" 24 #include "webrtc/rtc_base/constructormagic.h"
24 #include "webrtc/rtc_base/optional.h" 25 #include "webrtc/rtc_base/optional.h"
25 #include "webrtc/rtc_base/task_queue.h" 26 #include "webrtc/rtc_base/task_queue.h"
26 #include "webrtc/rtc_base/timeutils.h" 27 #include "webrtc/rtc_base/timeutils.h"
27 28
28 namespace webrtc { 29 namespace webrtc {
29 namespace test { 30 namespace test {
30 31
31 // Holds all the parameters available for controlling the simulation. 32 // Holds all the parameters available for controlling the simulation.
(...skipping 37 matching lines...) Expand 10 before | Expand all | Expand 10 after
69 rtc::Optional<bool> use_experimental_agc; 70 rtc::Optional<bool> use_experimental_agc;
70 rtc::Optional<int> aecm_routing_mode; 71 rtc::Optional<int> aecm_routing_mode;
71 rtc::Optional<bool> use_aecm_comfort_noise; 72 rtc::Optional<bool> use_aecm_comfort_noise;
72 rtc::Optional<int> agc_mode; 73 rtc::Optional<int> agc_mode;
73 rtc::Optional<int> agc_target_level; 74 rtc::Optional<int> agc_target_level;
74 rtc::Optional<bool> use_agc_limiter; 75 rtc::Optional<bool> use_agc_limiter;
75 rtc::Optional<int> agc_compression_gain; 76 rtc::Optional<int> agc_compression_gain;
76 rtc::Optional<int> vad_likelihood; 77 rtc::Optional<int> vad_likelihood;
77 rtc::Optional<int> ns_level; 78 rtc::Optional<int> ns_level;
78 rtc::Optional<bool> use_refined_adaptive_filter; 79 rtc::Optional<bool> use_refined_adaptive_filter;
80 int initial_mic_level;
81 bool simulate_mic_gain = false;
82 rtc::Optional<int> simulated_mic_kind;
79 bool report_performance = false; 83 bool report_performance = false;
80 bool report_bitexactness = false; 84 bool report_bitexactness = false;
81 bool use_verbose_logging = false; 85 bool use_verbose_logging = false;
82 bool discard_all_settings_in_aecdump = true; 86 bool discard_all_settings_in_aecdump = true;
83 rtc::Optional<std::string> aec_dump_input_filename; 87 rtc::Optional<std::string> aec_dump_input_filename;
84 rtc::Optional<std::string> aec_dump_output_filename; 88 rtc::Optional<std::string> aec_dump_output_filename;
85 bool fixed_interface = false; 89 bool fixed_interface = false;
86 bool store_intermediate_output = false; 90 bool store_intermediate_output = false;
87 rtc::Optional<std::string> custom_call_order_filename; 91 rtc::Optional<std::string> custom_call_order_filename;
88 }; 92 };
(...skipping 70 matching lines...) Expand 10 before | Expand all | Expand 10 after
159 std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_; 163 std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
160 StreamConfig in_config_; 164 StreamConfig in_config_;
161 StreamConfig out_config_; 165 StreamConfig out_config_;
162 StreamConfig reverse_in_config_; 166 StreamConfig reverse_in_config_;
163 StreamConfig reverse_out_config_; 167 StreamConfig reverse_out_config_;
164 std::unique_ptr<ChannelBufferWavReader> buffer_reader_; 168 std::unique_ptr<ChannelBufferWavReader> buffer_reader_;
165 std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_; 169 std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_;
166 AudioFrame rev_frame_; 170 AudioFrame rev_frame_;
167 AudioFrame fwd_frame_; 171 AudioFrame fwd_frame_;
168 bool bitexact_output_ = true; 172 bool bitexact_output_ = true;
173 rtc::Optional<int> aec_dump_mic_level_;
169 174
170 private: 175 private:
171 void SetupOutput(); 176 void SetupOutput();
172 177
173 size_t num_process_stream_calls_ = 0; 178 size_t num_process_stream_calls_ = 0;
174 size_t num_reverse_process_stream_calls_ = 0; 179 size_t num_reverse_process_stream_calls_ = 0;
175 size_t output_reset_counter_ = 0; 180 size_t output_reset_counter_ = 0;
176 std::unique_ptr<ChannelBufferWavWriter> buffer_writer_; 181 std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
177 std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_; 182 std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
178 TickIntervalStats proc_time_; 183 TickIntervalStats proc_time_;
179 std::ofstream residual_echo_likelihood_graph_writer_; 184 std::ofstream residual_echo_likelihood_graph_writer_;
180 185
peah-webrtc 2017/09/15 09:36:20 Please remove the empty line on 185
AleBzk 2017/09/22 12:33:55 Done.
186 int analog_mic_level_;
187 FakeRecordingDevice fake_recording_device_;
188
181 rtc::TaskQueue worker_queue_; 189 rtc::TaskQueue worker_queue_;
182 190
183 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator); 191 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
184 }; 192 };
185 193
186 } // namespace test 194 } // namespace test
187 } // namespace webrtc 195 } // namespace webrtc
188 196
189 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ 197 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
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