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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/media_stream_audio_processor_options.h" | 5 #include "content/renderer/media/media_stream_audio_processor_options.h" |
6 | 6 |
7 #include "base/files/file_path.h" | 7 #include "base/files/file_path.h" |
8 #include "base/logging.h" | 8 #include "base/logging.h" |
9 #include "base/path_service.h" | 9 #include "base/path_service.h" |
10 #include "base/strings/utf_string_conversions.h" | 10 #include "base/strings/utf_string_conversions.h" |
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32 webrtc::MediaConstraintsInterface::kValueTrue }, | 32 webrtc::MediaConstraintsInterface::kValueTrue }, |
33 #endif | 33 #endif |
34 { webrtc::MediaConstraintsInterface::kAutoGainControl, | 34 { webrtc::MediaConstraintsInterface::kAutoGainControl, |
35 webrtc::MediaConstraintsInterface::kValueTrue }, | 35 webrtc::MediaConstraintsInterface::kValueTrue }, |
36 { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl, | 36 { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl, |
37 webrtc::MediaConstraintsInterface::kValueTrue }, | 37 webrtc::MediaConstraintsInterface::kValueTrue }, |
38 { webrtc::MediaConstraintsInterface::kNoiseSuppression, | 38 { webrtc::MediaConstraintsInterface::kNoiseSuppression, |
39 webrtc::MediaConstraintsInterface::kValueTrue }, | 39 webrtc::MediaConstraintsInterface::kValueTrue }, |
40 { webrtc::MediaConstraintsInterface::kHighpassFilter, | 40 { webrtc::MediaConstraintsInterface::kHighpassFilter, |
41 webrtc::MediaConstraintsInterface::kValueTrue }, | 41 webrtc::MediaConstraintsInterface::kValueTrue }, |
42 // TODO(xians): Verify if it is OK to set typing detection to kValueFalse as | |
43 // default. | |
44 { webrtc::MediaConstraintsInterface::kTypingNoiseDetection, | 42 { webrtc::MediaConstraintsInterface::kTypingNoiseDetection, |
45 webrtc::MediaConstraintsInterface::kValueFalse }, | 43 webrtc::MediaConstraintsInterface::kValueTrue }, |
46 }; | 44 }; |
47 | 45 |
48 } // namespace | 46 } // namespace |
49 | 47 |
50 void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) { | 48 void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) { |
51 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { | 49 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { |
52 bool already_set_value; | 50 bool already_set_value; |
53 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key, | 51 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key, |
54 &already_set_value, NULL)) { | 52 &already_set_value, NULL)) { |
55 constraints->AddMandatory(kDefaultAudioConstraints[i].key, | 53 constraints->AddMandatory(kDefaultAudioConstraints[i].key, |
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83 return false; | 81 return false; |
84 } | 82 } |
85 | 83 |
86 bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints, | 84 bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints, |
87 const std::string& key) { | 85 const std::string& key) { |
88 bool value = false; | 86 bool value = false; |
89 return webrtc::FindConstraint(constraints, key, &value, NULL) && value; | 87 return webrtc::FindConstraint(constraints, key, &value, NULL) && value; |
90 } | 88 } |
91 | 89 |
92 void EnableEchoCancellation(AudioProcessing* audio_processing) { | 90 void EnableEchoCancellation(AudioProcessing* audio_processing) { |
93 #if defined(OS_IOS) | 91 #if defined(OS_ANDROID) |
94 // On iOS, VPIO provides built-in EC and AGC. | |
95 return; | |
96 #elif defined(OS_ANDROID) | |
97 // Mobile devices are using AECM. | 92 // Mobile devices are using AECM. |
98 int err = audio_processing->echo_control_mobile()->Enable(true); | 93 int err = audio_processing->echo_control_mobile()->set_routing_mode( |
99 err |= audio_processing->echo_control_mobile()->set_routing_mode( | |
100 webrtc::EchoControlMobile::kSpeakerphone); | 94 webrtc::EchoControlMobile::kSpeakerphone); |
95 err |= audio_processing->echo_control_mobile()->Enable(true); | |
101 CHECK_EQ(err, 0); | 96 CHECK_EQ(err, 0); |
102 #else | 97 #else |
103 int err = audio_processing->echo_cancellation()->Enable(true); | 98 int err = audio_processing->echo_cancellation()->set_suppression_level( |
104 err |= audio_processing->echo_cancellation()->set_suppression_level( | |
105 webrtc::EchoCancellation::kHighSuppression); | 99 webrtc::EchoCancellation::kHighSuppression); |
106 | 100 |
107 // Enable the metrics for AEC. | 101 // Enable the metrics for AEC. |
108 err |= audio_processing->echo_cancellation()->enable_metrics(true); | 102 err |= audio_processing->echo_cancellation()->enable_metrics(true); |
109 err |= audio_processing->echo_cancellation()->enable_delay_logging(true); | 103 err |= audio_processing->echo_cancellation()->enable_delay_logging(true); |
104 err |= audio_processing->echo_cancellation()->Enable(true); | |
110 CHECK_EQ(err, 0); | 105 CHECK_EQ(err, 0); |
111 #endif | 106 #endif |
112 } | 107 } |
113 | 108 |
114 void EnableNoiseSuppression(AudioProcessing* audio_processing) { | 109 void EnableNoiseSuppression(AudioProcessing* audio_processing) { |
115 int err = audio_processing->noise_suppression()->set_level( | 110 int err = audio_processing->noise_suppression()->set_level( |
116 webrtc::NoiseSuppression::kHigh); | 111 webrtc::NoiseSuppression::kHigh); |
117 err |= audio_processing->noise_suppression()->Enable(true); | 112 err |= audio_processing->noise_suppression()->Enable(true); |
118 CHECK_EQ(err, 0); | 113 CHECK_EQ(err, 0); |
119 } | 114 } |
120 | 115 |
121 void EnableHighPassFilter(AudioProcessing* audio_processing) { | 116 void EnableHighPassFilter(AudioProcessing* audio_processing) { |
122 CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0); | 117 CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0); |
123 } | 118 } |
124 | 119 |
125 // TODO(xians): stereo swapping | |
126 void EnableTypingDetection(AudioProcessing* audio_processing) { | 120 void EnableTypingDetection(AudioProcessing* audio_processing) { |
127 int err = audio_processing->voice_detection()->Enable(true); | 121 int err = audio_processing->voice_detection()->Enable(true); |
128 err |= audio_processing->voice_detection()->set_likelihood( | 122 err |= audio_processing->voice_detection()->set_likelihood( |
129 webrtc::VoiceDetection::kVeryLowLikelihood); | 123 webrtc::VoiceDetection::kVeryLowLikelihood); |
130 CHECK_EQ(err, 0); | 124 CHECK_EQ(err, 0); |
131 } | 125 } |
132 | 126 |
133 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) { | 127 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) { |
134 webrtc::Config config; | 128 webrtc::Config config; |
135 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); | 129 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); |
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156 #endif | 150 #endif |
157 if (audio_processing->StartDebugRecording(file_name.c_str())) | 151 if (audio_processing->StartDebugRecording(file_name.c_str())) |
158 DLOG(ERROR) << "Fail to start AEC debug recording"; | 152 DLOG(ERROR) << "Fail to start AEC debug recording"; |
159 } | 153 } |
160 | 154 |
161 void StopAecDump(AudioProcessing* audio_processing) { | 155 void StopAecDump(AudioProcessing* audio_processing) { |
162 if (audio_processing->StopDebugRecording()) | 156 if (audio_processing->StopDebugRecording()) |
163 DLOG(ERROR) << "Fail to stop AEC debug recording"; | 157 DLOG(ERROR) << "Fail to stop AEC debug recording"; |
164 } | 158 } |
165 | 159 |
160 void EnableAutomaticGainControl(AudioProcessing* audio_processing) { | |
161 #if defined(OS_ANDROID) || defined(OS_IOS) | |
162 const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog; | |
ajm
2014/01/24 06:48:59
As in off review comments, change this to kFixedDi
no longer working on chromium
2014/01/24 09:10:50
Ah, thanks. I noticed this mistake while I was mak
| |
163 #else | |
164 const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog; | |
165 #endif | |
166 int err = audio_processing->gain_control()->set_mode(mode); | |
167 err |= audio_processing->gain_control()->Enable(true); | |
168 CHECK_EQ(err, 0); | |
169 } | |
170 | |
166 } // namespace content | 171 } // namespace content |
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