blob: 783933b860da04716952818dfcfe2c4a1321f166 [file] [log] [blame]
/*
* 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_