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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/string_number_conversions.h" | |
10 #include "base/strings/utf_string_conversions.h" | 11 #include "base/strings/utf_string_conversions.h" |
11 #include "content/common/media/media_stream_options.h" | 12 #include "content/common/media/media_stream_options.h" |
13 #include "content/renderer/media/media_stream_constraints_util.h" | |
12 #include "content/renderer/media/rtc_media_constraints.h" | 14 #include "content/renderer/media/rtc_media_constraints.h" |
13 #include "media/audio/audio_parameters.h" | 15 #include "media/audio/audio_parameters.h" |
14 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | |
15 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" | |
16 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " | 16 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " |
17 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" | 17 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" |
18 | 18 |
19 namespace content { | 19 namespace content { |
20 | 20 |
21 const char MediaAudioConstraints::kEchoCancellation[] = "echoCancellation"; | |
22 const char MediaAudioConstraints::kGoogEchoCancellation[] = | |
23 "googEchoCancellation"; | |
24 const char MediaAudioConstraints::kGoogExperimentalEchoCancellation[] = | |
25 "googEchoCancellation2"; | |
26 const char MediaAudioConstraints::kGoogAutoGainControl[] = | |
27 "googAutoGainControl"; | |
28 const char MediaAudioConstraints::kGoogExperimentalAutoGainControl[] = | |
29 "googAutoGainControl2"; | |
30 const char MediaAudioConstraints::kGoogNoiseSuppression[] = | |
31 "googNoiseSuppression"; | |
32 const char MediaAudioConstraints::kGoogExperimentalNoiseSuppression[] = | |
33 "googNoiseSuppression2"; | |
34 const char MediaAudioConstraints::kGoogHighpassFilter[] = "googHighpassFilter"; | |
35 const char MediaAudioConstraints::kGoogTypingNoiseDetection[] = | |
36 "googTypingNoiseDetection"; | |
37 const char MediaAudioConstraints::kGoogAudioMirroring[] = "googAudioMirroring"; | |
38 | |
21 namespace { | 39 namespace { |
22 | 40 |
23 // Constant constraint keys which enables default audio constraints on | 41 // Constant constraint keys which enables default audio constraints on |
24 // mediastreams with audio. | 42 // mediastreams with audio. |
25 struct { | 43 struct { |
26 const char* key; | 44 const char* key; |
27 const char* value; | 45 bool value; |
28 } const kDefaultAudioConstraints[] = { | 46 } const kDefaultAudioConstraints[] = { |
29 { webrtc::MediaConstraintsInterface::kEchoCancellation, | 47 { MediaAudioConstraints::kEchoCancellation, true }, |
30 webrtc::MediaConstraintsInterface::kValueTrue }, | 48 { MediaAudioConstraints::kGoogEchoCancellation, true }, |
31 #if defined(OS_CHROMEOS) || defined(OS_MACOSX) | 49 #if defined(OS_CHROMEOS) || defined(OS_MACOSX) |
32 // Enable the extended filter mode AEC on platforms with known echo issues. | 50 // Enable the extended filter mode AEC on platforms with known echo issues. |
33 { webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation, | 51 { MediaAudioConstraints::kGoogExperimentalEchoCancellation, true }, |
34 webrtc::MediaConstraintsInterface::kValueTrue }, | 52 #else |
53 { MediaAudioConstraints::kGoogExperimentalEchoCancellation, false }, | |
35 #endif | 54 #endif |
36 { webrtc::MediaConstraintsInterface::kAutoGainControl, | 55 { MediaAudioConstraints::kGoogAutoGainControl, true }, |
37 webrtc::MediaConstraintsInterface::kValueTrue }, | 56 { MediaAudioConstraints::kGoogExperimentalAutoGainControl, true }, |
38 { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl, | 57 { MediaAudioConstraints::kGoogNoiseSuppression, true }, |
39 webrtc::MediaConstraintsInterface::kValueTrue }, | 58 { MediaAudioConstraints::kGoogHighpassFilter, true }, |
40 { webrtc::MediaConstraintsInterface::kNoiseSuppression, | 59 { MediaAudioConstraints::kGoogTypingNoiseDetection, true }, |
41 webrtc::MediaConstraintsInterface::kValueTrue }, | 60 { MediaAudioConstraints::kGoogExperimentalNoiseSuppression, false }, |
42 { webrtc::MediaConstraintsInterface::kHighpassFilter, | |
43 webrtc::MediaConstraintsInterface::kValueTrue }, | |
44 { webrtc::MediaConstraintsInterface::kTypingNoiseDetection, | |
45 webrtc::MediaConstraintsInterface::kValueTrue }, | |
46 #if defined(OS_WIN) | 61 #if defined(OS_WIN) |
47 { content::kMediaStreamAudioDucking, | 62 { kMediaStreamAudioDucking, true }, |
48 webrtc::MediaConstraintsInterface::kValueTrue }, | 63 #else |
64 { kMediaStreamAudioDucking, false }, | |
49 #endif | 65 #endif |
50 }; | 66 }; |
51 | 67 |
68 bool GetDefaultValueForConstraint(const blink::WebMediaConstraints& constraints, | |
69 const std::string& key) { | |
70 // The default audio processing is false for gUM with a specific | |
71 // kMediaStreamSource. | |
perkj_chrome
2014/04/14 12:15:19
Please explain what kMediaSource is.
no longer working on chromium
2014/04/14 14:40:50
Done with adding "which is used by tab capture and
| |
72 std::string value; | |
73 if (GetConstraintValue(constraints, kMediaStreamSource, &value)) | |
74 return false; | |
75 | |
76 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { | |
perkj_chrome
2014/04/14 12:15:19
arraysize ?
no longer working on chromium
2014/04/14 14:40:50
It is not possible before of the struct type used
| |
77 if (kDefaultAudioConstraints[i].key == key) | |
78 return kDefaultAudioConstraints[i].value; | |
79 } | |
80 | |
81 return false; | |
82 } | |
83 | |
52 } // namespace | 84 } // namespace |
53 | 85 |
54 void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) { | 86 // TODO(xians): Remove this method after the APM in WebRtc is deprecated. |
87 void MediaAudioConstraints::ApplyFixedAudioConstraints( | |
88 RTCMediaConstraints* constraints) { | |
55 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { | 89 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { |
perkj_chrome
2014/04/14 12:15:19
arraysize ?
no longer working on chromium
2014/04/14 14:40:50
ditto.
| |
56 bool already_set_value; | 90 bool already_set_value; |
57 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key, | 91 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key, |
58 &already_set_value, NULL)) { | 92 &already_set_value, NULL)) { |
59 constraints->AddOptional(kDefaultAudioConstraints[i].key, | 93 const std::string value = kDefaultAudioConstraints[i].value ? |
60 kDefaultAudioConstraints[i].value, false); | 94 webrtc::MediaConstraintsInterface::kValueTrue : |
95 webrtc::MediaConstraintsInterface::kValueFalse; | |
96 constraints->AddOptional(kDefaultAudioConstraints[i].key, value, false); | |
61 } else { | 97 } else { |
62 DVLOG(1) << "Constraint " << kDefaultAudioConstraints[i].key | 98 DVLOG(1) << "Constraint " << kDefaultAudioConstraints[i].key |
63 << " already set to " << already_set_value; | 99 << " already set to " << already_set_value; |
64 } | 100 } |
65 } | 101 } |
66 } | 102 } |
67 | 103 |
68 bool NeedsAudioProcessing(const blink::WebMediaConstraints& constraints, | 104 MediaAudioConstraints::MediaAudioConstraints( |
69 int effects) { | 105 const blink::WebMediaConstraints& constraints, int effects) |
70 RTCMediaConstraints native_constraints(constraints); | 106 : constraints_(constraints), effects_(effects) {} |
71 ApplyFixedAudioConstraints(&native_constraints); | 107 |
72 if (effects & media::AudioParameters::ECHO_CANCELLER) { | 108 MediaAudioConstraints::~MediaAudioConstraints() {} |
73 // If platform echo canceller is enabled, disable the software AEC. | 109 |
74 native_constraints.AddOptional( | 110 // TODO(xians): Remove this method after the APM in WebRtc is deprecated. |
75 MediaConstraintsInterface::kEchoCancellation, | 111 bool MediaAudioConstraints::NeedsAudioProcessing() { |
76 MediaConstraintsInterface::kValueFalse, true); | |
77 } | |
78 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { | 112 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { |
79 bool value = false; | 113 if (GetProperty(kDefaultAudioConstraints[i].key)) |
80 if (webrtc::FindConstraint(&native_constraints, | |
81 kDefaultAudioConstraints[i].key, &value, NULL) && | |
82 value) { | |
83 return true; | 114 return true; |
84 } | |
85 } | 115 } |
86 | 116 |
87 return false; | 117 return false; |
88 } | 118 } |
89 | 119 |
90 bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints, | 120 bool MediaAudioConstraints::GetProperty(const std::string& key) { |
91 const std::string& key) { | 121 if (effects_ & media::AudioParameters::ECHO_CANCELLER && |
no longer working on chromium
2014/04/11 16:47:05
Andrew, could you please take a look at this?
Pre
| |
122 key == kGoogEchoCancellation) { | |
123 // If platform echo canceller is enabled, disable the software AEC. | |
124 return false; | |
125 } | |
126 | |
127 // Return the value if the constraint is specified in |constraints|, | |
128 // otherwise return the default value. | |
92 bool value = false; | 129 bool value = false; |
93 return webrtc::FindConstraint(constraints, key, &value, NULL) && value; | 130 if (!GetConstraintValue(constraints_, key, &value)) |
131 value = GetDefaultValueForConstraint(constraints_, key); | |
132 | |
133 return value; | |
134 } | |
135 | |
136 bool MediaAudioConstraints::IsValid() { | |
137 blink::WebVector<blink::WebMediaConstraint> mandatory; | |
138 constraints_.getMandatoryConstraints(mandatory); | |
139 for (size_t i = 0; i < mandatory.size(); ++i) { | |
140 if (mandatory[i].m_name.utf8() == kMediaStreamSource) | |
141 continue; | |
142 | |
143 bool valid = false; | |
144 for (size_t j = 0; j < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++j) { | |
145 if (mandatory[i].m_name.utf8() == kDefaultAudioConstraints[j].key) { | |
146 valid = true; | |
147 break; | |
148 } | |
149 } | |
150 | |
151 if (!valid) { | |
152 DLOG(ERROR) << "Invalid MediaStream constraint. Name:" | |
153 << mandatory[i].m_name.utf8(); | |
154 return false; | |
155 } | |
156 } | |
157 | |
158 return true; | |
94 } | 159 } |
95 | 160 |
96 void EnableEchoCancellation(AudioProcessing* audio_processing) { | 161 void EnableEchoCancellation(AudioProcessing* audio_processing) { |
97 #if defined(OS_ANDROID) | 162 #if defined(OS_ANDROID) |
98 // Mobile devices are using AECM. | 163 // Mobile devices are using AECM. |
99 int err = audio_processing->echo_control_mobile()->set_routing_mode( | 164 int err = audio_processing->echo_control_mobile()->set_routing_mode( |
100 webrtc::EchoControlMobile::kSpeakerphone); | 165 webrtc::EchoControlMobile::kSpeakerphone); |
101 err |= audio_processing->echo_control_mobile()->Enable(true); | 166 err |= audio_processing->echo_control_mobile()->Enable(true); |
102 CHECK_EQ(err, 0); | 167 CHECK_EQ(err, 0); |
103 #else | 168 #else |
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120 } | 185 } |
121 | 186 |
122 void EnableExperimentalNoiseSuppression(AudioProcessing* audio_processing) { | 187 void EnableExperimentalNoiseSuppression(AudioProcessing* audio_processing) { |
123 CHECK_EQ(audio_processing->EnableExperimentalNs(true), 0); | 188 CHECK_EQ(audio_processing->EnableExperimentalNs(true), 0); |
124 } | 189 } |
125 | 190 |
126 void EnableHighPassFilter(AudioProcessing* audio_processing) { | 191 void EnableHighPassFilter(AudioProcessing* audio_processing) { |
127 CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0); | 192 CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0); |
128 } | 193 } |
129 | 194 |
130 void EnableTypingDetection(AudioProcessing* audio_processing, | 195 void EnableTypingDetection(AudioProcessing* audio_processing, |
perkj_chrome
2014/04/14 12:15:19
nit: allign paramers or move audio_processing to n
no longer working on chromium
2014/04/14 14:40:50
Done.
| |
131 webrtc::TypingDetection* typing_detector) { | 196 webrtc::TypingDetection* typing_detector) { |
132 int err = audio_processing->voice_detection()->Enable(true); | 197 int err = audio_processing->voice_detection()->Enable(true); |
133 err |= audio_processing->voice_detection()->set_likelihood( | 198 err |= audio_processing->voice_detection()->set_likelihood( |
134 webrtc::VoiceDetection::kVeryLowLikelihood); | 199 webrtc::VoiceDetection::kVeryLowLikelihood); |
135 CHECK_EQ(err, 0); | 200 CHECK_EQ(err, 0); |
136 | 201 |
137 // Configure the update period to 1s (100 * 10ms) in the typing detector. | 202 // Configure the update period to 1s (100 * 10ms) in the typing detector. |
138 typing_detector->SetParameters(0, 0, 0, 0, 0, 100); | 203 typing_detector->SetParameters(0, 0, 0, 0, 0, 100); |
139 } | 204 } |
140 | 205 |
141 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) { | 206 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) { |
142 webrtc::Config config; | 207 webrtc::Config config; |
143 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); | 208 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); |
144 audio_processing->SetExtraOptions(config); | 209 audio_processing->SetExtraOptions(config); |
145 } | 210 } |
146 | 211 |
147 void StartEchoCancellationDump(AudioProcessing* audio_processing, | 212 void StartEchoCancellationDump(AudioProcessing* audio_processing, |
148 const base::PlatformFile& aec_dump_file) { | 213 const base::PlatformFile& aec_dump_file) { |
perkj_chrome
2014/04/14 12:15:19
dio
no longer working on chromium
2014/04/14 14:40:50
Done.
| |
149 DCHECK_NE(aec_dump_file, base::kInvalidPlatformFileValue); | 214 DCHECK_NE(aec_dump_file, base::kInvalidPlatformFileValue); |
150 | 215 |
151 FILE* stream = base::FdopenPlatformFile(aec_dump_file, "w"); | 216 FILE* stream = base::FdopenPlatformFile(aec_dump_file, "w"); |
152 if (!stream) { | 217 if (!stream) { |
153 LOG(ERROR) << "Failed to open AEC dump file"; | 218 LOG(ERROR) << "Failed to open AEC dump file"; |
154 return; | 219 return; |
155 } | 220 } |
156 | 221 |
157 if (audio_processing->StartDebugRecording(stream)) | 222 if (audio_processing->StartDebugRecording(stream)) |
158 DLOG(ERROR) << "Fail to start AEC debug recording"; | 223 DLOG(ERROR) << "Fail to start AEC debug recording"; |
(...skipping 48 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
207 } | 272 } |
208 | 273 |
209 int median = 0, std = 0; | 274 int median = 0, std = 0; |
210 if (!audio_processing->echo_cancellation()->GetDelayMetrics(&median, &std)) { | 275 if (!audio_processing->echo_cancellation()->GetDelayMetrics(&median, &std)) { |
211 stats->echo_delay_median_ms = median; | 276 stats->echo_delay_median_ms = median; |
212 stats->echo_delay_std_ms = std; | 277 stats->echo_delay_std_ms = std; |
213 } | 278 } |
214 } | 279 } |
215 | 280 |
216 } // namespace content | 281 } // namespace content |
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