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/*
* 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_BUFFER_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
#include "typedefs.h"
#include "../../../../common_audio/resampler/include/resampler.h"
#include "file_wrapper.h"
#include "audio_device.h"
#include "list_wrapper.h"
namespace webrtc {
class CriticalSectionWrapper;
const WebRtc_UWord32 kPulsePeriodMs = 1000;
const WebRtc_UWord32 kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
class AudioDeviceObserver;
class MediaFile;
class AudioDeviceBuffer
{
public:
void SetId(WebRtc_UWord32 id);
WebRtc_Word32 RegisterAudioCallback(AudioTransport* audioCallback);
WebRtc_Word32 InitPlayout();
WebRtc_Word32 InitRecording();
WebRtc_Word32 SetRecordingSampleRate(WebRtc_UWord32 fsHz);
WebRtc_Word32 SetPlayoutSampleRate(WebRtc_UWord32 fsHz);
WebRtc_Word32 RecordingSampleRate() const;
WebRtc_Word32 PlayoutSampleRate() const;
WebRtc_Word32 SetRecordingChannels(WebRtc_UWord8 channels);
WebRtc_Word32 SetPlayoutChannels(WebRtc_UWord8 channels);
WebRtc_UWord8 RecordingChannels() const;
WebRtc_UWord8 PlayoutChannels() const;
WebRtc_Word32 SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
WebRtc_Word32 RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
WebRtc_Word32 SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples);
WebRtc_Word32 SetCurrentMicLevel(WebRtc_UWord32 level);
WebRtc_Word32 SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift);
WebRtc_Word32 DeliverRecordedData();
WebRtc_UWord32 NewMicLevel() const;
WebRtc_Word32 RequestPlayoutData(WebRtc_UWord32 nSamples);
WebRtc_Word32 GetPlayoutData(WebRtc_Word8* audioBuffer);
WebRtc_Word32 StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
WebRtc_Word32 StopInputFileRecording();
WebRtc_Word32 StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
WebRtc_Word32 StopOutputFileRecording();
AudioDeviceBuffer();
~AudioDeviceBuffer();
private:
void _EmptyList();
private:
WebRtc_Word32 _id;
CriticalSectionWrapper& _critSect;
CriticalSectionWrapper& _critSectCb;
AudioTransport* _ptrCbAudioTransport;
WebRtc_UWord32 _recSampleRate;
WebRtc_UWord32 _playSampleRate;
WebRtc_UWord8 _recChannels;
WebRtc_UWord8 _playChannels;
// selected recording channel (left/right/both)
AudioDeviceModule::ChannelType _recChannel;
// 2 or 4 depending on mono or stereo
WebRtc_UWord8 _recBytesPerSample;
WebRtc_UWord8 _playBytesPerSample;
// 10ms in stereo @ 96kHz
WebRtc_Word8 _recBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
WebRtc_UWord32 _recSamples;
WebRtc_UWord32 _recSize; // in bytes
// 10ms in stereo @ 96kHz
WebRtc_Word8 _playBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
WebRtc_UWord32 _playSamples;
WebRtc_UWord32 _playSize; // in bytes
FileWrapper& _recFile;
FileWrapper& _playFile;
WebRtc_UWord32 _currentMicLevel;
WebRtc_UWord32 _newMicLevel;
WebRtc_UWord32 _playDelayMS;
WebRtc_UWord32 _recDelayMS;
WebRtc_Word32 _clockDrift;
bool _measureDelay;
ListWrapper _pulseList;
WebRtc_UWord32 _lastPulseTime;
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H