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| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 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 | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_ | |
| 12 #define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_ | |
| 13 | |
| 14 #include "webrtc/base/refcountedobject.h" | |
| 15 #include "webrtc/modules/audio_device/include/audio_device.h" | |
| 16 #include "webrtc/modules/include/module.h" | |
| 17 | |
| 18 namespace webrtc { | |
| 19 | |
| 20 class AudioTransportCapture { | |
| 21 public: | |
| 22 virtual void CaptureRecordedData( | |
| 23 const void* audioSamples, const size_t nSamples, | |
|
henrika_webrtc
2017/03/23 09:02:42
Can't we use new notation in this interface?
audio
lliuu
2017/03/24 00:12:58
Acknowledged.
| |
| 24 const size_t nBytesPerSample, const size_t nChannels, | |
| 25 const uint32_t samplesPerSec) = 0; | |
| 26 | |
| 27 virtual void CapturePlayoutData( | |
| 28 const void* audioSamples, const size_t nSamples, | |
| 29 const size_t nBytesPerSample, const size_t nChannels, | |
| 30 const uint32_t samplesPerSec) = 0; | |
| 31 | |
| 32 AudioTransportCapture() = default; | |
| 33 virtual ~AudioTransportCapture() = default; | |
| 34 }; | |
| 35 | |
| 36 // Creates an ADM instance with AudioTransportCapture callback. | |
| 37 rtc::scoped_refptr<AudioDeviceModule> | |
| 38 CreateAudioDeviceWithTransportCapture( | |
| 39 const int32_t id, | |
| 40 const AudioDeviceModule::AudioLayer audio_layer, | |
| 41 AudioTransportCapture* transport_cb); | |
| 42 | |
| 43 } // namespace webrtc | |
| 44 | |
| 45 #endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_ | |
| 46 | |
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