webrtc/modules/audio_device/main/interface/audio_device_defines.h

81 lines
2.7 KiB
C++

/*
* Copyright (c) 2011 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_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
#include "typedefs.h"
namespace webrtc {
static const int kAdmMaxDeviceNameSize = 128;
static const int kAdmMaxFileNameSize = 512;
static const int kAdmMaxGuidSize = 128;
static const int kAdmMinPlayoutBufferSizeMs = 10;
static const int kAdmMaxPlayoutBufferSizeMs = 250;
// ----------------------------------------------------------------------------
// AudioDeviceObserver
// ----------------------------------------------------------------------------
class AudioDeviceObserver
{
public:
enum ErrorCode
{
kRecordingError = 0,
kPlayoutError = 1
};
enum WarningCode
{
kRecordingWarning = 0,
kPlayoutWarning = 1
};
virtual void OnErrorIsReported(const ErrorCode error) = 0;
virtual void OnWarningIsReported(const WarningCode warning) = 0;
protected:
virtual ~AudioDeviceObserver() {}
};
// ----------------------------------------------------------------------------
// AudioTransport
// ----------------------------------------------------------------------------
class AudioTransport
{
public:
virtual int32_t RecordedDataIsAvailable(const char* audioSamples,
const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
uint32_t& newMicLevel) = 0;
virtual int32_t NeedMorePlayData(const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
char* audioSamples,
uint32_t& nSamplesOut) = 0;
protected:
virtual ~AudioTransport() {}
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H