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Side by Side Diff: webrtc/voice_engine/transmit_mixer.cc

Issue 2720253002: Remove saturation warning support from TransmitMixer. (Closed)
Patch Set: Remove comment Created 3 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 #include "webrtc/voice_engine/transmit_mixer.h" 11 #include "webrtc/voice_engine/transmit_mixer.h"
12 12
13 #include <memory> 13 #include <memory>
14 14
15 #include "webrtc/audio/utility/audio_frame_operations.h" 15 #include "webrtc/audio/utility/audio_frame_operations.h"
16 #include "webrtc/base/format_macros.h" 16 #include "webrtc/base/format_macros.h"
17 #include "webrtc/base/logging.h" 17 #include "webrtc/base/logging.h"
18 #include "webrtc/system_wrappers/include/event_wrapper.h" 18 #include "webrtc/system_wrappers/include/event_wrapper.h"
19 #include "webrtc/system_wrappers/include/trace.h" 19 #include "webrtc/system_wrappers/include/trace.h"
20 #include "webrtc/voice_engine/channel.h" 20 #include "webrtc/voice_engine/channel.h"
21 #include "webrtc/voice_engine/channel_manager.h" 21 #include "webrtc/voice_engine/channel_manager.h"
22 #include "webrtc/voice_engine/statistics.h" 22 #include "webrtc/voice_engine/statistics.h"
23 #include "webrtc/voice_engine/utility.h" 23 #include "webrtc/voice_engine/utility.h"
24 #include "webrtc/voice_engine/voe_base_impl.h" 24 #include "webrtc/voice_engine/voe_base_impl.h"
25 25
26 namespace webrtc { 26 namespace webrtc {
27 namespace voe { 27 namespace voe {
28 28
29 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
29 // TODO(ajm): The thread safety of this is dubious... 30 // TODO(ajm): The thread safety of this is dubious...
30 void 31 void TransmitMixer::OnPeriodicProcess()
31 TransmitMixer::OnPeriodicProcess()
32 { 32 {
33 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), 33 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
34 "TransmitMixer::OnPeriodicProcess()"); 34 "TransmitMixer::OnPeriodicProcess()");
35 35
36 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
37 bool send_typing_noise_warning = false; 36 bool send_typing_noise_warning = false;
38 bool typing_noise_detected = false; 37 bool typing_noise_detected = false;
39 { 38 {
40 rtc::CritScope cs(&_critSect); 39 rtc::CritScope cs(&_critSect);
41 if (_typingNoiseWarningPending) { 40 if (_typingNoiseWarningPending) {
42 send_typing_noise_warning = true; 41 send_typing_noise_warning = true;
43 typing_noise_detected = _typingNoiseDetected; 42 typing_noise_detected = _typingNoiseDetected;
44 _typingNoiseWarningPending = false; 43 _typingNoiseWarningPending = false;
45 } 44 }
46 } 45 }
(...skipping 10 matching lines...) Expand all
57 } else { 56 } else {
58 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), 57 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
59 "TransmitMixer::OnPeriodicProcess() => " 58 "TransmitMixer::OnPeriodicProcess() => "
60 "CallbackOnError(VE_TYPING_NOISE_OFF_WARNING)"); 59 "CallbackOnError(VE_TYPING_NOISE_OFF_WARNING)");
61 _voiceEngineObserverPtr->CallbackOnError( 60 _voiceEngineObserverPtr->CallbackOnError(
62 -1, 61 -1,
63 VE_TYPING_NOISE_OFF_WARNING); 62 VE_TYPING_NOISE_OFF_WARNING);
64 } 63 }
65 } 64 }
66 } 65 }
66 }
67 #endif // WEBRTC_VOICE_ENGINE_TYPING_DETECTION 67 #endif // WEBRTC_VOICE_ENGINE_TYPING_DETECTION
68 68
69 bool saturationWarning = false;
70 {
71 // Modify |_saturationWarning| under lock to avoid conflict with write op
72 // in ProcessAudio and also ensure that we don't hold the lock during the
73 // callback.
74 rtc::CritScope cs(&_critSect);
75 saturationWarning = _saturationWarning;
76 if (_saturationWarning)
77 _saturationWarning = false;
78 }
79
80 if (saturationWarning)
81 {
82 rtc::CritScope cs(&_callbackCritSect);
83 if (_voiceEngineObserverPtr)
84 {
85 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
86 "TransmitMixer::OnPeriodicProcess() =>"
87 " CallbackOnError(VE_SATURATION_WARNING)");
88 _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
89 }
90 }
91 }
92
93
94 void TransmitMixer::PlayNotification(int32_t id, 69 void TransmitMixer::PlayNotification(int32_t id,
95 uint32_t durationMs) 70 uint32_t durationMs)
96 { 71 {
97 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), 72 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
98 "TransmitMixer::PlayNotification(id=%d, durationMs=%d)", 73 "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
99 id, durationMs); 74 id, durationMs);
100 75
101 // Not implement yet 76 // Not implement yet
102 } 77 }
103 78
(...skipping 65 matching lines...) Expand 10 before | Expand all | Expand 10 after
169 TransmitMixer::Destroy(TransmitMixer*& mixer) 144 TransmitMixer::Destroy(TransmitMixer*& mixer)
170 { 145 {
171 if (mixer) 146 if (mixer)
172 { 147 {
173 delete mixer; 148 delete mixer;
174 mixer = NULL; 149 mixer = NULL;
175 } 150 }
176 } 151 }
177 152
178 TransmitMixer::TransmitMixer(uint32_t instanceId) : 153 TransmitMixer::TransmitMixer(uint32_t instanceId) :
179 _monitorModule(this),
180 // Avoid conflict with other channels by adding 1024 - 1026, 154 // Avoid conflict with other channels by adding 1024 - 1026,
181 // won't use as much as 1024 channels. 155 // won't use as much as 1024 channels.
182 _filePlayerId(instanceId + 1024), 156 _filePlayerId(instanceId + 1024),
183 _fileRecorderId(instanceId + 1025), 157 _fileRecorderId(instanceId + 1025),
184 _fileCallRecorderId(instanceId + 1026), 158 _fileCallRecorderId(instanceId + 1026),
159 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
160 _monitorModule(this),
161 #endif
185 _instanceId(instanceId) 162 _instanceId(instanceId)
186 { 163 {
187 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), 164 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
188 "TransmitMixer::TransmitMixer() - ctor"); 165 "TransmitMixer::TransmitMixer() - ctor");
189 } 166 }
190 167
191 TransmitMixer::~TransmitMixer() 168 TransmitMixer::~TransmitMixer()
192 { 169 {
193 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), 170 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
194 "TransmitMixer::~TransmitMixer() - dtor"); 171 "TransmitMixer::~TransmitMixer() - dtor");
172 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
195 if (_processThreadPtr) 173 if (_processThreadPtr)
196 {
197 _processThreadPtr->DeRegisterModule(&_monitorModule); 174 _processThreadPtr->DeRegisterModule(&_monitorModule);
198 } 175 #endif
199 { 176 {
200 rtc::CritScope cs(&_critSect); 177 rtc::CritScope cs(&_critSect);
201 if (file_recorder_) { 178 if (file_recorder_) {
202 file_recorder_->RegisterModuleFileCallback(NULL); 179 file_recorder_->RegisterModuleFileCallback(NULL);
203 file_recorder_->StopRecording(); 180 file_recorder_->StopRecording();
204 } 181 }
205 if (file_call_recorder_) { 182 if (file_call_recorder_) {
206 file_call_recorder_->RegisterModuleFileCallback(NULL); 183 file_call_recorder_->RegisterModuleFileCallback(NULL);
207 file_call_recorder_->StopRecording(); 184 file_call_recorder_->StopRecording();
208 } 185 }
209 if (file_player_) { 186 if (file_player_) {
210 file_player_->RegisterModuleFileCallback(NULL); 187 file_player_->RegisterModuleFileCallback(NULL);
211 file_player_->StopPlayingFile(); 188 file_player_->StopPlayingFile();
212 } 189 }
213 } 190 }
214 } 191 }
215 192
216 int32_t 193 int32_t
217 TransmitMixer::SetEngineInformation(ProcessThread& processThread, 194 TransmitMixer::SetEngineInformation(ProcessThread& processThread,
218 Statistics& engineStatistics, 195 Statistics& engineStatistics,
219 ChannelManager& channelManager) 196 ChannelManager& channelManager)
220 { 197 {
221 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), 198 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
222 "TransmitMixer::SetEngineInformation()"); 199 "TransmitMixer::SetEngineInformation()");
223 200
224 _processThreadPtr = &processThread; 201 _processThreadPtr = &processThread;
225 _engineStatisticsPtr = &engineStatistics; 202 _engineStatisticsPtr = &engineStatistics;
226 _channelManagerPtr = &channelManager; 203 _channelManagerPtr = &channelManager;
227 204
205 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
228 _processThreadPtr->RegisterModule(&_monitorModule); 206 _processThreadPtr->RegisterModule(&_monitorModule);
229 207 #endif
230 return 0; 208 return 0;
231 } 209 }
232 210
233 int32_t 211 int32_t
234 TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) 212 TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
235 { 213 {
236 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), 214 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
237 "TransmitMixer::RegisterVoiceEngineObserver()"); 215 "TransmitMixer::RegisterVoiceEngineObserver()");
238 rtc::CritScope cs(&_callbackCritSect); 216 rtc::CritScope cs(&_callbackCritSect);
239 217
(...skipping 830 matching lines...) Expand 10 before | Expand all | Expand 10 after
1070 audioproc_->set_stream_key_pressed(key_pressed); 1048 audioproc_->set_stream_key_pressed(key_pressed);
1071 1049
1072 int err = audioproc_->ProcessStream(&_audioFrame); 1050 int err = audioproc_->ProcessStream(&_audioFrame);
1073 if (err != 0) { 1051 if (err != 0) {
1074 LOG(LS_ERROR) << "ProcessStream() error: " << err; 1052 LOG(LS_ERROR) << "ProcessStream() error: " << err;
1075 assert(false); 1053 assert(false);
1076 } 1054 }
1077 1055
1078 // Store new capture level. Only updated when analog AGC is enabled. 1056 // Store new capture level. Only updated when analog AGC is enabled.
1079 _captureLevel = agc->stream_analog_level(); 1057 _captureLevel = agc->stream_analog_level();
1080
1081 rtc::CritScope cs(&_critSect);
1082 // Triggers a callback in OnPeriodicProcess().
1083 _saturationWarning |= agc->stream_is_saturated();
1084 } 1058 }
1085 1059
1086 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1060 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
1087 void TransmitMixer::TypingDetection(bool keyPressed) 1061 void TransmitMixer::TypingDetection(bool keyPressed)
1088 { 1062 {
1089 // We let the VAD determine if we're using this feature or not. 1063 // We let the VAD determine if we're using this feature or not.
1090 if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) { 1064 if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) {
1091 return; 1065 return;
1092 } 1066 }
1093 1067
(...skipping 44 matching lines...) Expand 10 before | Expand all | Expand 10 after
1138 void TransmitMixer::EnableStereoChannelSwapping(bool enable) { 1112 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
1139 swap_stereo_channels_ = enable; 1113 swap_stereo_channels_ = enable;
1140 } 1114 }
1141 1115
1142 bool TransmitMixer::IsStereoChannelSwappingEnabled() { 1116 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
1143 return swap_stereo_channels_; 1117 return swap_stereo_channels_;
1144 } 1118 }
1145 1119
1146 } // namespace voe 1120 } // namespace voe
1147 } // namespace webrtc 1121 } // namespace webrtc
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