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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 |
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55 | 55 |
56 int GetVersion(char version[1024]) override; | 56 int GetVersion(char version[1024]) override; |
57 | 57 |
58 int LastError() override; | 58 int LastError() override; |
59 | 59 |
60 AudioTransport* audio_transport() override { return this; } | 60 AudioTransport* audio_transport() override { return this; } |
61 | 61 |
62 int AssociateSendChannel(int channel, int accociate_send_channel) override; | 62 int AssociateSendChannel(int channel, int accociate_send_channel) override; |
63 | 63 |
64 // AudioTransport | 64 // AudioTransport |
65 int32_t RecordedDataIsAvailable(const void* audioSamples, | 65 int32_t RecordedDataIsAvailable(const void* audio_data, |
66 const size_t nSamples, | 66 const size_t number_of_frames, |
67 const size_t nBytesPerSample, | 67 const size_t bytes_per_sample, |
68 const size_t nChannels, | 68 const size_t number_of_channels, |
69 const uint32_t samplesPerSec, | 69 const uint32_t sample_rate, |
70 const uint32_t totalDelayMS, | 70 const uint32_t audio_delay_milliseconds, |
71 const int32_t clockDrift, | 71 const int32_t clock_drift, |
72 const uint32_t currentMicLevel, | 72 const uint32_t volume, |
73 const bool keyPressed, | 73 const bool key_pressed, |
74 uint32_t& newMicLevel) override; | 74 uint32_t& new_mic_volume) override; |
75 int32_t NeedMorePlayData(const size_t nSamples, | 75 int32_t NeedMorePlayData(const size_t nSamples, |
76 const size_t nBytesPerSample, | 76 const size_t nBytesPerSample, |
77 const size_t nChannels, | 77 const size_t nChannels, |
78 const uint32_t samplesPerSec, | 78 const uint32_t samplesPerSec, |
79 void* audioSamples, | 79 void* audioSamples, |
80 size_t& nSamplesOut, | 80 size_t& nSamplesOut, |
81 int64_t* elapsed_time_ms, | 81 int64_t* elapsed_time_ms, |
82 int64_t* ntp_time_ms) override; | 82 int64_t* ntp_time_ms) override; |
83 void PushCaptureData(int voe_channel, | 83 void PushCaptureData(int voe_channel, |
84 const void* audio_data, | 84 const void* audio_data, |
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102 VoEBaseImpl(voe::SharedData* shared); | 102 VoEBaseImpl(voe::SharedData* shared); |
103 ~VoEBaseImpl() override; | 103 ~VoEBaseImpl() override; |
104 | 104 |
105 private: | 105 private: |
106 int32_t StartPlayout(); | 106 int32_t StartPlayout(); |
107 int32_t StopPlayout(); | 107 int32_t StopPlayout(); |
108 int32_t StartSend(); | 108 int32_t StartSend(); |
109 int32_t StopSend(); | 109 int32_t StopSend(); |
110 int32_t TerminateInternal(); | 110 int32_t TerminateInternal(); |
111 | 111 |
112 // Helper function to process the recorded data with AudioProcessing Module, | |
113 // demultiplex the data to specific voe channels, encode and send to the | |
114 // network. When |number_of_VoE_channels| is 0, it will demultiplex the | |
115 // data to all the existing VoE channels. | |
116 // It returns new AGC microphone volume or 0 if no volume changes | |
117 // should be done. | |
118 int ProcessRecordedDataWithAPM( | |
119 const int voe_channels[], size_t number_of_voe_channels, | |
120 const void* audio_data, uint32_t sample_rate, size_t number_of_channels, | |
121 size_t number_of_frames, uint32_t audio_delay_milliseconds, | |
122 int32_t clock_drift, uint32_t volume, bool key_pressed); | |
123 | |
124 void GetPlayoutData(int sample_rate, size_t number_of_channels, | 112 void GetPlayoutData(int sample_rate, size_t number_of_channels, |
125 size_t number_of_frames, bool feed_data_to_apm, | 113 size_t number_of_frames, bool feed_data_to_apm, |
126 void* audio_data, int64_t* elapsed_time_ms, | 114 void* audio_data, int64_t* elapsed_time_ms, |
127 int64_t* ntp_time_ms); | 115 int64_t* ntp_time_ms); |
128 | 116 |
129 // Initialize channel by setting Engine Information then initializing | 117 // Initialize channel by setting Engine Information then initializing |
130 // channel. | 118 // channel. |
131 int InitializeChannel(voe::ChannelOwner* channel_owner); | 119 int InitializeChannel(voe::ChannelOwner* channel_owner); |
132 VoiceEngineObserver* voiceEngineObserverPtr_; | 120 VoiceEngineObserver* voiceEngineObserverPtr_; |
133 rtc::CriticalSection callbackCritSect_; | 121 rtc::CriticalSection callbackCritSect_; |
134 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 122 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
135 | 123 |
136 AudioFrame audioFrame_; | 124 AudioFrame audioFrame_; |
137 voe::SharedData* shared_; | 125 voe::SharedData* shared_; |
138 }; | 126 }; |
139 | 127 |
140 } // namespace webrtc | 128 } // namespace webrtc |
141 | 129 |
142 #endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H | 130 #endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H |
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