122 lines
4.1 KiB
C
122 lines
4.1 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_BUFFER_H
|
||
|
|
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
||
|
|
|
||
|
|
#include "typedefs.h"
|
||
|
|
#include "resampler.h"
|
||
|
|
#include "file_wrapper.h"
|
||
|
|
#include "audio_device.h"
|
||
|
|
#include "list_wrapper.h"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
class CriticalSectionWrapper;
|
||
|
|
|
||
|
|
const WebRtc_UWord32 kPulsePeriodMs = 1000;
|
||
|
|
|
||
|
|
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 @ 48kHz
|
||
|
|
WebRtc_Word8 _recBuffer[1920];
|
||
|
|
|
||
|
|
// one sample <=> 2 or 4 bytes
|
||
|
|
WebRtc_UWord32 _recSamples;
|
||
|
|
WebRtc_UWord32 _recSize; // in bytes
|
||
|
|
|
||
|
|
// 10ms in stereo @ 48kHz
|
||
|
|
WebRtc_Word8 _playBuffer[1920];
|
||
|
|
|
||
|
|
// 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
|