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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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98 | 98 |
99 int32_t NeedMorePlayData(const size_t nSamples, | 99 int32_t NeedMorePlayData(const size_t nSamples, |
100 const size_t nBytesPerSample, | 100 const size_t nBytesPerSample, |
101 const uint8_t nChannels, | 101 const uint8_t nChannels, |
102 const uint32_t samplesPerSec, | 102 const uint32_t samplesPerSec, |
103 void* audioSamples, | 103 void* audioSamples, |
104 size_t& nSamplesOut, | 104 size_t& nSamplesOut, |
105 int64_t* elapsed_time_ms, | 105 int64_t* elapsed_time_ms, |
106 int64_t* ntp_time_ms) override; | 106 int64_t* ntp_time_ms) override; |
107 | 107 |
108 int OnDataAvailable(const int voe_channels[], | |
109 int number_of_voe_channels, | |
110 const int16_t* audio_data, | |
111 int sample_rate, | |
112 int number_of_channels, | |
113 size_t number_of_frames, | |
114 int audio_delay_milliseconds, | |
115 int current_volume, | |
116 bool key_pressed, | |
117 bool need_audio_processing) override; | |
118 | |
119 void PushCaptureData(int voe_channel, const void* audio_data, | |
120 int bits_per_sample, int sample_rate, | |
121 int number_of_channels, | |
122 size_t number_of_frames) override; | |
123 | |
124 void PullRenderData(int bits_per_sample, int sample_rate, | |
125 int number_of_channels, size_t number_of_frames, | |
126 void* audio_data, | |
127 int64_t* elapsed_time_ms, | |
128 int64_t* ntp_time_ms) override; | |
129 | |
130 AudioTransportImpl(AudioDeviceModule* audioDevice); | 108 AudioTransportImpl(AudioDeviceModule* audioDevice); |
131 ~AudioTransportImpl(); | 109 ~AudioTransportImpl(); |
132 | 110 |
133 public: | 111 public: |
134 int32_t SetFilePlayout(bool enable, const char* fileName = NULL); | 112 int32_t SetFilePlayout(bool enable, const char* fileName = NULL); |
135 void SetFullDuplex(bool enable); | 113 void SetFullDuplex(bool enable); |
136 void SetSpeakerVolume(bool enable) | 114 void SetSpeakerVolume(bool enable) |
137 { | 115 { |
138 _speakerVolume = enable; | 116 _speakerVolume = enable; |
139 } | 117 } |
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230 | 208 |
231 rtc::scoped_ptr<ProcessThread> _processThread; | 209 rtc::scoped_ptr<ProcessThread> _processThread; |
232 AudioDeviceModule* _audioDevice; | 210 AudioDeviceModule* _audioDevice; |
233 AudioEventObserver* _audioEventObserver; | 211 AudioEventObserver* _audioEventObserver; |
234 AudioTransportImpl* _audioTransport; | 212 AudioTransportImpl* _audioTransport; |
235 }; | 213 }; |
236 | 214 |
237 } // namespace webrtc | 215 } // namespace webrtc |
238 | 216 |
239 #endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H | 217 #endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H |
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