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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/base/timeutils.h" | 20 #include "webrtc/base/timeutils.h" |
21 #include "webrtc/base/constructormagic.h" | 21 #include "webrtc/base/constructormagic.h" |
22 #include "webrtc/base/optional.h" | 22 #include "webrtc/base/optional.h" |
23 #include "webrtc/common_audio/channel_buffer.h" | 23 #include "webrtc/common_audio/channel_buffer.h" |
24 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 24 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
25 #include "webrtc/modules/audio_processing/test/test_utils.h" | 25 #include "webrtc/modules/audio_processing/test/test_utils.h" |
26 | 26 |
27 namespace webrtc { | 27 namespace webrtc { |
28 namespace test { | 28 namespace test { |
29 | 29 |
| 30 // TODO(alessiob): Check what initial value makes sense, 100 was used in |
| 31 // WavBasedSimulator::last_specified_microphone_level_. |
| 32 constexpr int kInitialMicrophoneGainLevel = 100; |
| 33 |
30 // Holds all the parameters available for controlling the simulation. | 34 // Holds all the parameters available for controlling the simulation. |
31 struct SimulationSettings { | 35 struct SimulationSettings { |
32 SimulationSettings(); | 36 SimulationSettings(); |
33 SimulationSettings(const SimulationSettings&); | 37 SimulationSettings(const SimulationSettings&); |
34 ~SimulationSettings(); | 38 ~SimulationSettings(); |
35 rtc::Optional<int> stream_delay; | 39 rtc::Optional<int> stream_delay; |
36 rtc::Optional<int> stream_drift_samples; | 40 rtc::Optional<int> stream_drift_samples; |
37 rtc::Optional<int> output_sample_rate_hz; | 41 rtc::Optional<int> output_sample_rate_hz; |
38 rtc::Optional<int> output_num_channels; | 42 rtc::Optional<int> output_num_channels; |
39 rtc::Optional<int> reverse_output_sample_rate_hz; | 43 rtc::Optional<int> reverse_output_sample_rate_hz; |
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67 rtc::Optional<bool> use_experimental_agc; | 71 rtc::Optional<bool> use_experimental_agc; |
68 rtc::Optional<int> aecm_routing_mode; | 72 rtc::Optional<int> aecm_routing_mode; |
69 rtc::Optional<bool> use_aecm_comfort_noise; | 73 rtc::Optional<bool> use_aecm_comfort_noise; |
70 rtc::Optional<int> agc_mode; | 74 rtc::Optional<int> agc_mode; |
71 rtc::Optional<int> agc_target_level; | 75 rtc::Optional<int> agc_target_level; |
72 rtc::Optional<bool> use_agc_limiter; | 76 rtc::Optional<bool> use_agc_limiter; |
73 rtc::Optional<int> agc_compression_gain; | 77 rtc::Optional<int> agc_compression_gain; |
74 rtc::Optional<int> vad_likelihood; | 78 rtc::Optional<int> vad_likelihood; |
75 rtc::Optional<int> ns_level; | 79 rtc::Optional<int> ns_level; |
76 rtc::Optional<bool> use_refined_adaptive_filter; | 80 rtc::Optional<bool> use_refined_adaptive_filter; |
| 81 bool simulate_mic_gain = false; |
77 bool report_performance = false; | 82 bool report_performance = false; |
78 bool report_bitexactness = false; | 83 bool report_bitexactness = false; |
79 bool use_verbose_logging = false; | 84 bool use_verbose_logging = false; |
80 bool discard_all_settings_in_aecdump = true; | 85 bool discard_all_settings_in_aecdump = true; |
81 rtc::Optional<std::string> aec_dump_input_filename; | 86 rtc::Optional<std::string> aec_dump_input_filename; |
82 rtc::Optional<std::string> aec_dump_output_filename; | 87 rtc::Optional<std::string> aec_dump_output_filename; |
83 bool fixed_interface = false; | 88 bool fixed_interface = false; |
84 bool store_intermediate_output = false; | 89 bool store_intermediate_output = false; |
85 rtc::Optional<std::string> custom_call_order_filename; | 90 rtc::Optional<std::string> custom_call_order_filename; |
86 }; | 91 }; |
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176 TickIntervalStats proc_time_; | 181 TickIntervalStats proc_time_; |
177 std::ofstream residual_echo_likelihood_graph_writer_; | 182 std::ofstream residual_echo_likelihood_graph_writer_; |
178 | 183 |
179 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator); | 184 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator); |
180 }; | 185 }; |
181 | 186 |
182 } // namespace test | 187 } // namespace test |
183 } // namespace webrtc | 188 } // namespace webrtc |
184 | 189 |
185 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ | 190 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_ |
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