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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 #ifndef VOICE_ENGINE_TRANSMIT_MIXER_H_ | 11 #ifndef VOICE_ENGINE_TRANSMIT_MIXER_H_ |
12 #define VOICE_ENGINE_TRANSMIT_MIXER_H_ | 12 #define VOICE_ENGINE_TRANSMIT_MIXER_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "common_audio/resampler/include/push_resampler.h" | 16 #include "common_audio/resampler/include/push_resampler.h" |
17 #include "common_types.h" // NOLINT(build/include) | 17 #include "common_types.h" // NOLINT(build/include) |
18 #include "modules/audio_processing/typing_detection.h" | 18 #include "modules/audio_processing/typing_detection.h" |
19 #include "modules/include/module_common_types.h" | 19 #include "modules/include/module_common_types.h" |
20 #include "rtc_base/criticalsection.h" | 20 #include "rtc_base/criticalsection.h" |
21 #include "voice_engine/audio_level.h" | 21 #include "voice_engine/audio_level.h" |
22 #include "voice_engine/file_player.h" | |
23 #include "voice_engine/file_recorder.h" | |
24 #include "voice_engine/include/voe_base.h" | 22 #include "voice_engine/include/voe_base.h" |
25 #include "voice_engine/monitor_module.h" | 23 #include "voice_engine/monitor_module.h" |
26 #include "voice_engine/voice_engine_defines.h" | 24 #include "voice_engine/voice_engine_defines.h" |
27 | 25 |
28 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) | 26 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) |
29 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1 | 27 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1 |
30 #else | 28 #else |
31 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0 | 29 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0 |
32 #endif | 30 #endif |
33 | 31 |
34 namespace webrtc { | 32 namespace webrtc { |
35 class AudioProcessing; | 33 class AudioProcessing; |
36 class ProcessThread; | 34 class ProcessThread; |
37 | 35 |
38 namespace voe { | 36 namespace voe { |
39 | 37 |
40 class ChannelManager; | 38 class ChannelManager; |
41 class MixedAudio; | 39 class MixedAudio; |
42 class Statistics; | 40 class Statistics; |
43 | 41 |
44 class TransmitMixer : public FileCallback { | 42 class TransmitMixer { |
45 public: | 43 public: |
46 static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId); | 44 static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId); |
47 | 45 |
48 static void Destroy(TransmitMixer*& mixer); | 46 static void Destroy(TransmitMixer*& mixer); |
49 | 47 |
50 int32_t SetEngineInformation(ProcessThread& processThread, | 48 int32_t SetEngineInformation(ProcessThread& processThread, |
51 Statistics& engineStatistics, | 49 Statistics& engineStatistics, |
52 ChannelManager& channelManager); | 50 ChannelManager& channelManager); |
53 | 51 |
54 int32_t SetAudioProcessingModule( | 52 int32_t SetAudioProcessingModule( |
(...skipping 22 matching lines...) Expand all Loading... |
77 virtual int16_t AudioLevelFullRange() const; | 75 virtual int16_t AudioLevelFullRange() const; |
78 | 76 |
79 // See description of "totalAudioEnergy" in the WebRTC stats spec: | 77 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
80 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaud
ioenergy | 78 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaud
ioenergy |
81 // 'virtual' to allow mocking. | 79 // 'virtual' to allow mocking. |
82 virtual double GetTotalInputEnergy() const; | 80 virtual double GetTotalInputEnergy() const; |
83 | 81 |
84 // 'virtual' to allow mocking. | 82 // 'virtual' to allow mocking. |
85 virtual double GetTotalInputDuration() const; | 83 virtual double GetTotalInputDuration() const; |
86 | 84 |
87 bool IsRecordingCall(); | |
88 | |
89 bool IsRecordingMic(); | |
90 | |
91 int StartPlayingFileAsMicrophone(const char* fileName, | |
92 bool loop, | |
93 FileFormats format, | |
94 int startPosition, | |
95 float volumeScaling, | |
96 int stopPosition, | |
97 const CodecInst* codecInst); | |
98 | |
99 int StartPlayingFileAsMicrophone(InStream* stream, | |
100 FileFormats format, | |
101 int startPosition, | |
102 float volumeScaling, | |
103 int stopPosition, | |
104 const CodecInst* codecInst); | |
105 | |
106 int StopPlayingFileAsMicrophone(); | |
107 | |
108 int IsPlayingFileAsMicrophone() const; | |
109 | |
110 int StartRecordingMicrophone(const char* fileName, | |
111 const CodecInst* codecInst); | |
112 | |
113 int StartRecordingMicrophone(OutStream* stream, | |
114 const CodecInst* codecInst); | |
115 | |
116 int StopRecordingMicrophone(); | |
117 | |
118 int StartRecordingCall(const char* fileName, const CodecInst* codecInst); | |
119 | |
120 int StartRecordingCall(OutStream* stream, const CodecInst* codecInst); | |
121 | |
122 int StopRecordingCall(); | |
123 | |
124 void SetMixWithMicStatus(bool mix); | |
125 | |
126 int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); | 85 int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); |
127 | 86 |
128 virtual ~TransmitMixer(); | 87 virtual ~TransmitMixer(); |
129 | 88 |
130 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 89 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
131 // Periodic callback from the MonitorModule. | 90 // Periodic callback from the MonitorModule. |
132 void OnPeriodicProcess(); | 91 void OnPeriodicProcess(); |
133 #endif | 92 #endif |
134 | 93 |
135 // FileCallback | |
136 void PlayNotification(const int32_t id, | |
137 const uint32_t durationMs); | |
138 | |
139 void RecordNotification(const int32_t id, | |
140 const uint32_t durationMs); | |
141 | |
142 void PlayFileEnded(const int32_t id); | |
143 | |
144 void RecordFileEnded(const int32_t id); | |
145 | |
146 // Virtual to allow mocking. | 94 // Virtual to allow mocking. |
147 virtual void EnableStereoChannelSwapping(bool enable); | 95 virtual void EnableStereoChannelSwapping(bool enable); |
148 bool IsStereoChannelSwappingEnabled(); | 96 bool IsStereoChannelSwappingEnabled(); |
149 | 97 |
150 protected: | 98 protected: |
151 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 99 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
152 TransmitMixer() : _monitorModule(this) {} | 100 TransmitMixer() : _monitorModule(this) {} |
153 #else | 101 #else |
154 TransmitMixer() = default; | 102 TransmitMixer() = default; |
155 #endif | 103 #endif |
156 | 104 |
157 private: | 105 private: |
158 TransmitMixer(uint32_t instanceId); | 106 TransmitMixer(uint32_t instanceId); |
159 | 107 |
160 // Gets the maximum sample rate and number of channels over all currently | 108 // Gets the maximum sample rate and number of channels over all currently |
161 // sending codecs. | 109 // sending codecs. |
162 void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels); | 110 void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels); |
163 | 111 |
164 void GenerateAudioFrame(const int16_t audioSamples[], | 112 void GenerateAudioFrame(const int16_t audioSamples[], |
165 size_t nSamples, | 113 size_t nSamples, |
166 size_t nChannels, | 114 size_t nChannels, |
167 int samplesPerSec); | 115 int samplesPerSec); |
168 int32_t RecordAudioToFile(uint32_t mixingFrequency); | |
169 | |
170 int32_t MixOrReplaceAudioWithFile( | |
171 int mixingFrequency); | |
172 | 116 |
173 void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level, | 117 void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level, |
174 bool key_pressed); | 118 bool key_pressed); |
175 | 119 |
176 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 120 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
177 void TypingDetection(bool keyPressed); | 121 void TypingDetection(bool keyPressed); |
178 #endif | 122 #endif |
179 | 123 |
180 // uses | 124 // uses |
181 Statistics* _engineStatisticsPtr = nullptr; | 125 Statistics* _engineStatisticsPtr = nullptr; |
182 ChannelManager* _channelManagerPtr = nullptr; | 126 ChannelManager* _channelManagerPtr = nullptr; |
183 AudioProcessing* audioproc_ = nullptr; | 127 AudioProcessing* audioproc_ = nullptr; |
184 VoiceEngineObserver* _voiceEngineObserverPtr = nullptr; | 128 VoiceEngineObserver* _voiceEngineObserverPtr = nullptr; |
185 ProcessThread* _processThreadPtr = nullptr; | 129 ProcessThread* _processThreadPtr = nullptr; |
186 | 130 |
187 // owns | 131 // owns |
188 AudioFrame _audioFrame; | 132 AudioFrame _audioFrame; |
189 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate | 133 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate |
190 std::unique_ptr<FilePlayer> file_player_; | |
191 std::unique_ptr<FileRecorder> file_recorder_; | |
192 std::unique_ptr<FileRecorder> file_call_recorder_; | |
193 int _filePlayerId = 0; | |
194 int _fileRecorderId = 0; | |
195 int _fileCallRecorderId = 0; | |
196 bool _filePlaying = false; | |
197 bool _fileRecording = false; | |
198 bool _fileCallRecording = false; | |
199 voe::AudioLevel _audioLevel; | 134 voe::AudioLevel _audioLevel; |
200 // protect file instances and their variables in MixedParticipants() | 135 // protect file instances and their variables in MixedParticipants() |
201 rtc::CriticalSection _critSect; | 136 rtc::CriticalSection _critSect; |
202 rtc::CriticalSection _callbackCritSect; | 137 rtc::CriticalSection _callbackCritSect; |
203 | 138 |
204 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 139 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
205 MonitorModule<TransmitMixer> _monitorModule; | 140 MonitorModule<TransmitMixer> _monitorModule; |
206 webrtc::TypingDetection _typingDetection; | 141 webrtc::TypingDetection _typingDetection; |
207 bool _typingNoiseWarningPending = false; | 142 bool _typingNoiseWarningPending = false; |
208 bool _typingNoiseDetected = false; | 143 bool _typingNoiseDetected = false; |
209 #endif | 144 #endif |
210 | 145 |
211 int _instanceId = 0; | 146 int _instanceId = 0; |
212 bool _mixFileWithMicrophone = false; | |
213 uint32_t _captureLevel = 0; | 147 uint32_t _captureLevel = 0; |
214 bool stereo_codec_ = false; | 148 bool stereo_codec_ = false; |
215 bool swap_stereo_channels_ = false; | 149 bool swap_stereo_channels_ = false; |
216 }; | 150 }; |
217 } // namespace voe | 151 } // namespace voe |
218 } // namespace webrtc | 152 } // namespace webrtc |
219 | 153 |
220 #endif // VOICE_ENGINE_TRANSMIT_MIXER_H_ | 154 #endif // VOICE_ENGINE_TRANSMIT_MIXER_H_ |
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