| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| |
| namespace webrtc { |
| |
| enum { |
| kDefaultSampleRate = 44100, |
| kBitsPerSample = 16, |
| kNumChannels = 1, |
| kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000, |
| // Number of bytes per audio frame. |
| // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame] |
| kBytesPerFrame = kNumChannels * (kBitsPerSample / 8), |
| }; |
| |
| class PlayoutDelayProvider { |
| public: |
| virtual int PlayoutDelayMs() = 0; |
| |
| protected: |
| PlayoutDelayProvider() {} |
| virtual ~PlayoutDelayProvider() {} |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |