webrtc/modules/audio_device/audio_device_buffer.h

/*
 *  Copyright (c) 2012 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 uint32_t kPulsePeriodMs = 1000;
const uint32_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz

class AudioDeviceObserver;
class MediaFile;

class AudioDeviceBuffer
{
public:
    void SetId(uint32_t id);
    int32_t RegisterAudioCallback(AudioTransport* audioCallback);

    int32_t InitPlayout();
    int32_t InitRecording();

    int32_t SetRecordingSampleRate(uint32_t fsHz);
    int32_t SetPlayoutSampleRate(uint32_t fsHz);
    int32_t RecordingSampleRate() const;
    int32_t PlayoutSampleRate() const;

    int32_t SetRecordingChannels(uint8_t channels);
    int32_t SetPlayoutChannels(uint8_t channels);
    uint8_t RecordingChannels() const;
    uint8_t PlayoutChannels() const;
    int32_t SetRecordingChannel(
        const AudioDeviceModule::ChannelType channel);
    int32_t RecordingChannel(
        AudioDeviceModule::ChannelType& channel) const;

    int32_t SetRecordedBuffer(const void* audioBuffer, uint32_t nSamples);
    int32_t SetCurrentMicLevel(uint32_t level);
    int32_t SetVQEData(uint32_t playDelayMS,
                       uint32_t recDelayMS,
                       int32_t clockDrift);
    int32_t DeliverRecordedData();
    uint32_t NewMicLevel() const;

    int32_t RequestPlayoutData(uint32_t nSamples);
    int32_t GetPlayoutData(void* audioBuffer);

    int32_t StartInputFileRecording(
        const char fileName[kAdmMaxFileNameSize]);
    int32_t StopInputFileRecording();
    int32_t StartOutputFileRecording(
        const char fileName[kAdmMaxFileNameSize]);
    int32_t StopOutputFileRecording();

    AudioDeviceBuffer();
    ~AudioDeviceBuffer();

private:
    void _EmptyList();

private:
    int32_t                   _id;
    CriticalSectionWrapper&         _critSect;
    CriticalSectionWrapper&         _critSectCb;

    AudioTransport*                 _ptrCbAudioTransport;

    uint32_t                  _recSampleRate;
    uint32_t                  _playSampleRate;

    uint8_t                   _recChannels;
    uint8_t                   _playChannels;

    // selected recording channel (left/right/both)
    AudioDeviceModule::ChannelType _recChannel;

    // 2 or 4 depending on mono or stereo
    uint8_t                   _recBytesPerSample;
    uint8_t                   _playBytesPerSample;

    // 10ms in stereo @ 96kHz
    int8_t                          _recBuffer[kMaxBufferSizeBytes];

    // one sample <=> 2 or 4 bytes
    uint32_t                  _recSamples;
    uint32_t                  _recSize;           // in bytes

    // 10ms in stereo @ 96kHz
    int8_t                          _playBuffer[kMaxBufferSizeBytes];

    // one sample <=> 2 or 4 bytes
    uint32_t                  _playSamples;
    uint32_t                  _playSize;          // in bytes

    FileWrapper&                    _recFile;
    FileWrapper&                    _playFile;

    uint32_t                  _currentMicLevel;
    uint32_t                  _newMicLevel;

    uint32_t                  _playDelayMS;
    uint32_t                  _recDelayMS;

    int32_t                   _clockDrift;

    bool                            _measureDelay;
    ListWrapper                     _pulseList;
    uint32_t                  _lastPulseTime;
};

}  // namespace webrtc

#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H