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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 Loading... | |
| 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 |
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| 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"); |
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| 1070 audioproc_->set_stream_key_pressed(key_pressed); | 1047 audioproc_->set_stream_key_pressed(key_pressed); |
| 1071 | 1048 |
| 1072 int err = audioproc_->ProcessStream(&_audioFrame); | 1049 int err = audioproc_->ProcessStream(&_audioFrame); |
| 1073 if (err != 0) { | 1050 if (err != 0) { |
| 1074 LOG(LS_ERROR) << "ProcessStream() error: " << err; | 1051 LOG(LS_ERROR) << "ProcessStream() error: " << err; |
| 1075 assert(false); | 1052 assert(false); |
| 1076 } | 1053 } |
| 1077 | 1054 |
| 1078 // Store new capture level. Only updated when analog AGC is enabled. | 1055 // Store new capture level. Only updated when analog AGC is enabled. |
| 1079 _captureLevel = agc->stream_analog_level(); | 1056 _captureLevel = agc->stream_analog_level(); |
| 1080 | |
| 1081 rtc::CritScope cs(&_critSect); | |
|
tommi
2017/02/28 11:46:43
look, we can haz one less locks!
| |
| 1082 // Triggers a callback in OnPeriodicProcess(). | |
| 1083 _saturationWarning |= agc->stream_is_saturated(); | |
| 1084 } | 1057 } |
| 1085 | 1058 |
| 1086 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 1059 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 1087 void TransmitMixer::TypingDetection(bool keyPressed) | 1060 void TransmitMixer::TypingDetection(bool keyPressed) |
| 1088 { | 1061 { |
| 1089 // We let the VAD determine if we're using this feature or not. | 1062 // We let the VAD determine if we're using this feature or not. |
| 1090 if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) { | 1063 if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) { |
| 1091 return; | 1064 return; |
| 1092 } | 1065 } |
| 1093 | 1066 |
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| 1138 void TransmitMixer::EnableStereoChannelSwapping(bool enable) { | 1111 void TransmitMixer::EnableStereoChannelSwapping(bool enable) { |
| 1139 swap_stereo_channels_ = enable; | 1112 swap_stereo_channels_ = enable; |
| 1140 } | 1113 } |
| 1141 | 1114 |
| 1142 bool TransmitMixer::IsStereoChannelSwappingEnabled() { | 1115 bool TransmitMixer::IsStereoChannelSwappingEnabled() { |
| 1143 return swap_stereo_channels_; | 1116 return swap_stereo_channels_; |
| 1144 } | 1117 } |
| 1145 | 1118 |
| 1146 } // namespace voe | 1119 } // namespace voe |
| 1147 } // namespace webrtc | 1120 } // namespace webrtc |
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