RtAudio.h

/************************************************************************/
/*! \class RtAudio
    \brief Realtime audio i/o C++ classes.

    RtAudio provides a common API (Application Programming Interface)
    for realtime audio input/output across Linux (native ALSA, Jack,
    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
    (DirectSound, ASIO and WASAPI) operating systems.

    RtAudio GitHub site: https://github.com/thestk/rtaudio
    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/

    RtAudio: realtime audio i/o C++ classes
    Copyright (c) 2001-2023 Gary P. Scavone

    Permission is hereby granted, free of charge, to any person
    obtaining a copy of this software and associated documentation files
    (the "Software"), to deal in the Software without restriction,
    including without limitation the rights to use, copy, modify, merge,
    publish, distribute, sublicense, and/or sell copies of the Software,
    and to permit persons to whom the Software is furnished to do so,
    subject to the following conditions:

    The above copyright notice and this permission notice shall be
    included in all copies or substantial portions of the Software.

    Any person wishing to distribute modifications to the Software is
    asked to send the modifications to the original developer so that
    they can be incorporated into the canonical version.  This is,
    however, not a binding provision of this license.

    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/************************************************************************/

/*!
  \file RtAudio.h
 */

#ifndef __RTAUDIO_H
#define __RTAUDIO_H

#define RTAUDIO_VERSION_MAJOR 6
#define RTAUDIO_VERSION_MINOR 0
#define RTAUDIO_VERSION_PATCH 1
#define RTAUDIO_VERSION_BETA  0

#define RTAUDIO_TOSTRING2(n) #n
#define RTAUDIO_TOSTRING(n) RTAUDIO_TOSTRING2(n)

#if RTAUDIO_VERSION_BETA > 0
#define RTAUDIO_VERSION RTAUDIO_TOSTRING(RTAUDIO_VERSION_MAJOR) \
                        "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_MINOR) \
                        "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_PATCH) \
                     "beta" RTAUDIO_TOSTRING(RTAUDIO_VERSION_BETA)
#else
#define RTAUDIO_VERSION RTAUDIO_TOSTRING(RTAUDIO_VERSION_MAJOR) \
                        "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_MINOR) \
                        "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_PATCH)
#endif

#if defined _WIN32 || defined __CYGWIN__
#if defined(RTAUDIO_EXPORT)
#define RTAUDIO_DLL_PUBLIC __declspec(dllexport)
#else
#define RTAUDIO_DLL_PUBLIC
#endif
#else
#if __GNUC__ >= 4
#define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) )
#else
#define RTAUDIO_DLL_PUBLIC
#endif
#endif

#include <string>
#include <vector>
#include <iostream>
#include <functional>
#include <optional>
#include <memory>

 /*! \typedef typedef unsigned long RtAudioFormat;
     \brief RtAudio data format type.

     Support for signed integers and floats.  Audio data fed to/from an
     RtAudio stream is assumed to ALWAYS be in host byte order.  The
     internal routines will automatically take care of any necessary
     byte-swapping between the host format and the soundcard.  Thus,
     endian-ness is not a concern in the following format definitions.
     Note that there are no range checks for floating-point values that
     extend beyond plus/minus 1.0.

     - \e RTAUDIO_SINT8:   8-bit signed integer.
     - \e RTAUDIO_SINT16:  16-bit signed integer.
     - \e RTAUDIO_SINT24:  24-bit signed integer.
     - \e RTAUDIO_SINT32:  32-bit signed integer.
     - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
     - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
 */
typedef unsigned long RtAudioFormat;
static const RtAudioFormat RTAUDIO_SINT8 = 0x1;    // 8-bit signed integer.
static const RtAudioFormat RTAUDIO_SINT16 = 0x2;   // 16-bit signed integer.
static const RtAudioFormat RTAUDIO_SINT24 = 0x4;   // 24-bit signed integer.
static const RtAudioFormat RTAUDIO_SINT32 = 0x8;   // 32-bit signed integer.
static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.

/*! \typedef typedef unsigned long RtAudioStreamFlags;
    \brief RtAudio stream option flags.

    The following flags can be OR'ed together to allow a client to
    make changes to the default stream behavior:

    - \e RTAUDIO_NONINTERLEAVED:   Use non-interleaved buffers (default = interleaved).
    - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
    - \e RTAUDIO_HOG_DEVICE:       Attempt grab device for exclusive use.
    - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
    - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only).

    By default, RtAudio streams pass and receive audio data from the
    client in an interleaved format.  By passing the
    RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
    data will instead be presented in non-interleaved buffers.  In
    this case, each buffer argument in the RtAudioCallback function
    will point to a single array of data, with \c nFrames samples for
    each channel concatenated back-to-back.  For example, the first
    sample of data for the second channel would be located at index \c
    nFrames (assuming the \c buffer pointer was recast to the correct
    data type for the stream).

    Certain audio APIs offer a number of parameters that influence the
    I/O latency of a stream.  By default, RtAudio will attempt to set
    these parameters internally for robust (glitch-free) performance
    (though some APIs, like Windows DirectSound, make this difficult).
    By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
    function, internal stream settings will be influenced in an attempt
    to minimize stream latency, though possibly at the expense of stream
    performance.

    If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
    open the input and/or output stream device(s) for exclusive use.
    Note that this is not possible with all supported audio APIs.

    If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
    to select realtime scheduling (round-robin) for the callback thread.

    If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
    open the "default" PCM device when using the ALSA API. Note that this
    will override any specified input or output device id.

    If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt
    to automatically connect the ports of the client to the audio device.
*/
typedef unsigned int RtAudioStreamFlags;
static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1;    // Use non-interleaved buffers (default = interleaved).
static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2;  // Attempt to set stream parameters for lowest possible latency.
static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4;        // Attempt grab device and prevent use by others.
static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only).
static const RtAudioStreamFlags RTAUDIO_ALSA_NONBLOCK = 0x40; // Use non-block mode for alsa io.

/*! \typedef typedef unsigned long RtAudioStreamStatus;
    \brief RtAudio stream status (over- or underflow) flags.

    Notification of a stream over- or underflow is indicated by a
    non-zero stream \c status argument in the RtAudioCallback function.
    The stream status can be one of the following two options,
    depending on whether the stream is open for output and/or input:

    - \e RTAUDIO_INPUT_OVERFLOW:   Input data was discarded because of an overflow condition at the driver.
    - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
*/
typedef unsigned int RtAudioStreamStatus;
static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1;    // Input data was discarded because of an overflow condition at the driver.
static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2;  // The output buffer ran low, likely causing a gap in the output sound.

//! RtAudio callback function prototype.
/*!
   All RtAudio clients must create a function of type RtAudioCallback
   to read and/or write data from/to the audio stream.  When the
   underlying audio system is ready for new input or output data, this
   function will be invoked.

   \param outputBuffer For output (or duplex) streams, the client
          should write \c nFrames of audio sample frames into this
          buffer.  This argument should be recast to the datatype
          specified when the stream was opened.  For input-only
          streams, this argument will be NULL.

   \param inputBuffer For input (or duplex) streams, this buffer will
          hold \c nFrames of input audio sample frames.  This
          argument should be recast to the datatype specified when the
          stream was opened.  For output-only streams, this argument
          will be NULL.

   \param nFrames The number of sample frames of input or output
          data in the buffers.  The actual buffer size in bytes is
          dependent on the data type and number of channels in use.

   \param streamTime The number of seconds that have elapsed since the
          stream was started.

   \param status If non-zero, this argument indicates a data overflow
          or underflow condition for the stream.  The particular
          condition can be determined by comparison with the
          RtAudioStreamStatus flags.

   \param userData A pointer to optional data provided by the client
          when opening the stream (default = NULL).

   \return
   To continue normal stream operation, the RtAudioCallback function
   should return a value of zero.  To stop the stream and drain the
   output buffer, the function should return a value of one.  To abort
   the stream immediately, the client should return a value of two.
 */
typedef int (*RtAudioCallback)(void *outputBuffer,
                               const void *inputBuffer,
                               unsigned int nFrames,
                               double streamTime,
                               RtAudioStreamStatus status,
                               void *userData);

enum RtAudioDeviceParam {
    DEFAULT_CHANGED,
    DEVICE_ADDED,
    DEVICE_REMOVED,
    DEVICE_STATE_CHANGED,
    DEVICE_PROPERTY_CHANGED
};

typedef void (*RtAudioDeviceCallback)(unsigned int deviceId, RtAudioDeviceParam param, void* userData);
typedef std::function<void(const std::string&, RtAudioDeviceParam)> RtAudioDeviceCallbackLambda;

enum RtAudioErrorType {
    RTAUDIO_NO_ERROR = 0,      /*!< No error. */
    RTAUDIO_WARNING,           /*!< A non-critical error. */
    RTAUDIO_UNKNOWN_ERROR,     /*!< An unspecified error type. */
    RTAUDIO_NO_DEVICES_FOUND,  /*!< No devices found on system. */
    RTAUDIO_INVALID_DEVICE,    /*!< An invalid device ID was specified. */
    RTAUDIO_DEVICE_DISCONNECT, /*!< A device in use was disconnected. */
    RTAUDIO_MEMORY_ERROR,      /*!< An error occurred during memory allocation. */
    RTAUDIO_INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
    RTAUDIO_INVALID_USE,       /*!< The function was called incorrectly. */
    RTAUDIO_DRIVER_ERROR,      /*!< A system driver error occurred. */
    RTAUDIO_SYSTEM_ERROR,      /*!< A system error occurred. */
    RTAUDIO_THREAD_ERROR       /*!< A thread error occurred. */
};

//! RtAudio error callback function prototype.
/*!
    \param type Type of error.
    \param errorText Error description.
 */
typedef std::function<void(RtAudioErrorType type,
    const std::string& errorText)>
    RtAudioErrorCallback;

// **************************************************************** //
//
// RtAudio class declaration.
//
// RtAudio is a "controller" used to select an available audio i/o
// interface.  It presents a common API for the user to call but all
// functionality is implemented by the class RtApi and its
// subclasses.  RtAudio creates an instance of an RtApi subclass
// based on the user's API choice.  If no choice is made, RtAudio
// attempts to make a "logical" API selection.
//
// **************************************************************** //

class RtApi;

class RtApiStreamClassFactory;
class RtApiProber;
class RtApiEnumerator;
class RtApiSystemCallback;


class RTAUDIO_DLL_PUBLIC RtAudio
{
public:
    static const unsigned int MAX_SAMPLE_RATES;
    static const unsigned int SAMPLE_RATES[];

    //! Audio API specifier arguments.
    enum Api {
        UNSPECIFIED,    /*!< Search for a working compiled API. */
        MACOSX_CORE,    /*!< Macintosh OS-X Core Audio API. */
        LINUX_ALSA,     /*!< The Advanced Linux Sound Architecture API. */
        UNIX_JACK,      /*!< The Jack Low-Latency Audio Server API. */
        LINUX_PULSE,    /*!< The Linux PulseAudio API. */
        WINDOWS_ASIO,   /*!< The Steinberg Audio Stream I/O API. */
        WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
        RTAUDIO_DUMMY,  /*!< A compilable but non-functional API. */
        NUM_APIS        /*!< Number of values in this enum. */
    };

    static std::shared_ptr<RtApiEnumerator> GetRtAudioEnumerator(RtAudio::Api api);

    static std::shared_ptr<RtApiProber> GetRtAudioProber(RtAudio::Api api);

    static std::shared_ptr<RtApiStreamClassFactory> GetRtAudioStreamFactory(RtAudio::Api api);

    static std::shared_ptr<RtApiSystemCallback> GetRtAudioSystemCallback(RtAudio::Api api, RtAudioDeviceCallbackLambda callback);

    struct DeviceInfoPartial {
        std::string name;               /*!< Character string device name. */
        std::string busID;              /*!< Unique ID of device bus. */
        bool supportsOutput = false;
        bool supportsInput = false;
    };
    //! The public device information structure for returning queried values.
    struct DeviceInfo {
        DeviceInfoPartial partial{};
        unsigned int outputChannels{};  /*!< Maximum output channels supported by device. */
        unsigned int inputChannels{};   /*!< Maximum input channels supported by device. */
        unsigned int duplexChannels{};  /*!< Maximum simultaneous input/output channels supported by device. */
        bool isDefaultOutput{ false };         /*!< true if this is the default output device. */
        bool isDefaultInput{ false };          /*!< true if this is the default input device. */
        std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
        unsigned int currentSampleRate{};   /*!< Current sample rate, system sample rate as currently configured. */
        unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */
        RtAudioFormat nativeFormats{};  /*!< Bit mask of supported data formats. */
    };

    //! The structure for specifying stream options.
    /*!
      The following flags can be OR'ed together to allow a client to
      make changes to the default stream behavior:

      - \e RTAUDIO_NONINTERLEAVED:    Use non-interleaved buffers (default = interleaved).
      - \e RTAUDIO_MINIMIZE_LATENCY:  Attempt to set stream parameters for lowest possible latency.
      - \e RTAUDIO_HOG_DEVICE:        Attempt grab device for exclusive use.
      - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
      - \e RTAUDIO_ALSA_USE_DEFAULT:  Use the "default" PCM device (ALSA only).

      By default, RtAudio streams pass and receive audio data from the
      client in an interleaved format.  By passing the
      RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
      data will instead be presented in non-interleaved buffers.  In
      this case, each buffer argument in the RtAudioCallback function
      will point to a single array of data, with \c nFrames samples for
      each channel concatenated back-to-back.  For example, the first
      sample of data for the second channel would be located at index \c
      nFrames (assuming the \c buffer pointer was recast to the correct
      data type for the stream).

      Certain audio APIs offer a number of parameters that influence the
      I/O latency of a stream.  By default, RtAudio will attempt to set
      these parameters internally for robust (glitch-free) performance
      (though some APIs, like Windows DirectSound, make this difficult).
      By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
      function, internal stream settings will be influenced in an attempt
      to minimize stream latency, though possibly at the expense of stream
      performance.

      If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
      open the input and/or output stream device(s) for exclusive use.
      Note that this is not possible with all supported audio APIs.

      If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
      to select realtime scheduling (round-robin) for the callback thread.
      The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
      flag is set. It defines the thread's realtime priority.

      If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
      open the "default" PCM device when using the ALSA API. Note that this
      will override any specified input or output device id.

      The \c numberOfBuffers parameter can be used to control stream
      latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
      only.  A value of two is usually the smallest allowed.  Larger
      numbers can potentially result in more robust stream performance,
      though likely at the cost of stream latency.  The value set by the
      user is replaced during execution of the RtAudio::openStream()
      function by the value actually used by the system.

      The \c streamName parameter can be used to set the client name
      when using the Jack API or the application name when using the
      Pulse API.  By default, the Jack client name is set to RtApiJack.
      However, if you wish to create multiple instances of RtAudio with
      Jack, each instance must have a unique client name. The default
      Pulse application name is set to "RtAudio."
    */
    struct StreamOptions {
        RtAudioStreamFlags flags{};      /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
        unsigned int numberOfBuffers{};  /*!< Number of stream buffers. */
        std::string streamName;        /*!< A stream name (currently used only in Jack and Pulse). */
        int priority{};                  /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
    };

    //! A static function to determine the current RtAudio version.
    static std::string getVersion(void);

    //! A static function to determine the available compiled audio APIs.
    /*!
      The values returned in the std::vector can be compared against
      the enumerated list values.  Note that there can be more than one
      API compiled for certain operating systems.
    */
    static void getCompiledApi(std::vector<RtAudio::Api>& apis);

    //! Return the name of a specified compiled audio API.
    /*!
      This obtains a short lower-case name used for identification purposes.
      This value is guaranteed to remain identical across library versions.
      If the API is unknown, this function will return the empty string.
    */
    static std::string getApiName(RtAudio::Api api);

    //! Return the display name of a specified compiled audio API.
    /*!
      This obtains a long name used for display purposes.
      If the API is unknown, this function will return the empty string.
    */
    static std::string getApiDisplayName(RtAudio::Api api);

    //! Return the compiled audio API having the given name.
    /*!
      A case insensitive comparison will check the specified name
      against the list of compiled APIs, and return the one that
      matches. On failure, the function returns UNSPECIFIED.
    */
    static RtAudio::Api getCompiledApiByName(const std::string& name);

    //! Return the compiled audio API having the given display name.
    /*!
    A case sensitive comparison will check the specified display name
    against the list of compiled APIs, and return the one that
    matches. On failure, the function returns UNSPECIFIED.
    */
    static RtAudio::Api getCompiledApiByDisplayName(const std::string& name);
};

// Operating system dependent thread functionality.
#if defined(_MSC_VER)

#ifndef NOMINMAX
#define NOMINMAX
#endif
#include <windows.h>
#include <process.h>
#include <stdint.h>

typedef uintptr_t ThreadHandle;
typedef CRITICAL_SECTION StreamMutex;

#else

  // Using pthread library for various flavors of unix.
#include <pthread.h>

typedef pthread_t ThreadHandle;
typedef pthread_mutex_t StreamMutex;

#endif

// Setup for "dummy" behavior if no apis specified.
#if !(defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) \
      || defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) \
      || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__))

#define __RTAUDIO_DUMMY__

#endif

// This global structure type is used to pass callback information
// between the private RtAudio stream structure and global callback
// handling functions.
struct CallbackInfo {
    void* object{};    // Used as a "this" pointer.
    ThreadHandle thread{};
    void* callback{};
    void* userData{};
    void* apiInfo{};   // void pointer for API specific callback information
    bool isRunning{ false };
    bool doRealtime{ false };
    int priority{};
    bool deviceDisconnected{ false };
};

#pragma pack(push, 1)
class S24 {

protected:
    unsigned char c3[3]{};

public:
    S24() {}

    S24& operator = (const int& i) {
        c3[0] = (unsigned char)(i & 0x000000ff);
        c3[1] = (unsigned char)((i & 0x0000ff00) >> 8);
        c3[2] = (unsigned char)((i & 0x00ff0000) >> 16);
        return *this;
    }

    S24(const double& d) { *this = (int)d; }
    S24(const float& f) { *this = (int)f; }
    S24(const signed short& s) { *this = (int)s; }
    S24(const char& c) { *this = (int)c; }

    int asInt() {
        int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
        if (i & 0x800000) i |= ~0xffffff;
        return i;
    }
};
#pragma pack(pop)

class RTAUDIO_DLL_PUBLIC RtApi
{
public:
    enum StreamMode {
        OUTPUT,
        INPUT,
        DUPLEX,
        UNINITIALIZED = -75
    };
    enum { FAILURE, SUCCESS };

    enum StreamState {
        STREAM_STOPPED,
        STREAM_STOPPING,
        STREAM_RUNNING,
        STREAM_ERROR,
        STREAM_CLOSED = -50
    };
    // A protected structure used for buffer conversion.
    struct ConvertInfo {
        int channels;
        int inJump, outJump;
        RtAudioFormat inFormat, outFormat;
        std::vector<int> inOffset;
        std::vector<int> outOffset;
    };

    struct RtApiStream {
        std::string deviceId;
        void* apiHandle;           // void pointer for API specific stream handle information
        StreamMode mode;           // OUTPUT, INPUT, or DUPLEX.
        StreamState state;         // STOPPED, RUNNING, or CLOSED
        bool errorState = false;   //TODO: add mutex or make atomic
        std::shared_ptr<char[]> userBuffer[2];       // Playback and record, respectively.
        std::shared_ptr<char[]> deviceBuffer;
        bool doConvertBuffer[2];   // Playback and record, respectively.
        bool userInterleaved;
        bool deviceInterleaved[2]; // Playback and record, respectively.
        bool doByteSwap[2];        // Playback and record, respectively.
        unsigned int sampleRate;
        unsigned int bufferSize;
        unsigned int nBuffers;
        unsigned int nUserChannels[2];    // Playback and record, respectively.
        unsigned int nDeviceChannels[2];  // Playback and record channels, respectively.
        unsigned int channelOffset[2];    // Playback and record, respectively.
        unsigned long latency[2];         // Playback and record, respectively.
        RtAudioFormat userFormat;
        RtAudioFormat deviceFormat[2];    // Playback and record, respectively.
        StreamMutex mutex;
        CallbackInfo callbackInfo;
        ConvertInfo convertInfo[2];
        double streamTime;         // Number of elapsed seconds since the stream started.
    };

    static unsigned int formatBytes(RtAudioFormat format);
    static void convertBuffer(const RtApi::RtApiStream stream_,
                              char *outBuffer,
                              const char *inBuffer,
                              RtApi::ConvertInfo info,
                              unsigned int samples,
                              RtApi::StreamMode mode);
    static void byteSwapBuffer(char* buffer, unsigned int samples, RtAudioFormat format);
    static void setConvertInfo(RtApi::StreamMode mode, RtApi::RtApiStream& stream_);
};


#include <sstream>

class RTAUDIO_DLL_PUBLIC ErrorBase {
public:
    virtual ~ErrorBase() {}

    void setErrorCallback(RtAudioErrorCallback errorCallback) { errorCallback_ = errorCallback; }
    void showWarnings(bool value) { showWarnings_ = value; }
    const std::string& getErrorText(void) const;

    RtAudioErrorType error(RtAudioErrorType type, const std::string& message);
protected:
    std::ostringstream errorStream_;
    RtAudioErrorCallback errorCallback_ = nullptr;
    RtAudioErrorType errorThread(RtAudioErrorType type, const std::string& message);
    RtAudioErrorType error(RtAudioErrorType type);
private:
    std::string errorText_;
    bool showWarnings_ = false;
};

class RTAUDIO_DLL_PUBLIC RtApiEnumerator : public ErrorBase {
public:
    virtual RtAudio::Api getCurrentApi(void) = 0;
    virtual std::vector<RtAudio::DeviceInfoPartial> listDevices(void) = 0;
    virtual std::string getDefaultDevice(RtApi::StreamMode mode) { return {}; }
};

class RTAUDIO_DLL_PUBLIC RtApiProber : public ErrorBase {
public:
    RtApiProber() {}
    virtual ~RtApiProber() {}
    virtual RtAudio::Api getCurrentApi(void) = 0;
    virtual std::optional<RtAudio::DeviceInfo> probeDevice(const std::string& busId) = 0;
};

class RTAUDIO_DLL_PUBLIC RtApiSystemCallback : public ErrorBase {
public:
    RtApiSystemCallback() {}
    virtual ~RtApiSystemCallback() {}
    virtual RtAudio::Api getCurrentApi(void) = 0;
    virtual bool hasError() const { return false; }
};

class RTAUDIO_DLL_PUBLIC RtApiStreamClass : public ErrorBase {
public:
    RtApiStreamClass(RtApi::RtApiStream stream);
    virtual ~RtApiStreamClass();
    virtual RtAudio::Api getCurrentApi(void) = 0;

    virtual RtAudioErrorType startStream(void) = 0;
    virtual RtAudioErrorType stopStream(void) = 0;

    bool isStreamRunning();

    double getStreamTime(void) const { return stream_.streamTime; }
    void tickStreamTime(void) { stream_.streamTime += (stream_.bufferSize * 1.0 / stream_.sampleRate); }
    unsigned int getBufferSize(void) const;
protected:
    RtAudioErrorType startStreamCheck();
    RtAudioErrorType stopStreamCheck();

    RtApi::RtApiStream stream_;
};

struct CreateStreamParams {
    std::string busId;
    RtApi::StreamMode mode = RtApi::StreamMode::UNINITIALIZED;
    unsigned int channelsInput = 0;
    unsigned int channelsOutput = 0;
    unsigned int sampleRate = 0;
    RtAudioFormat format = 0;
    unsigned int bufferSize = 0;
    RtAudioCallback callback = nullptr;
    void* userData = nullptr;
    RtAudio::StreamOptions* options = nullptr;
};
class RTAUDIO_DLL_PUBLIC RtApiStreamClassFactory : public ErrorBase {
public:
    RtApiStreamClassFactory() {}
    virtual ~RtApiStreamClassFactory() {}
    virtual RtAudio::Api getCurrentApi(void) = 0;
    virtual std::shared_ptr<RtApiStreamClass> createStream(CreateStreamParams params) = 0;

protected:
    bool setupStreamCommon(RtApi::RtApiStream& stream_);
    bool setupStreamWithParams(RtApi::RtApiStream& stream_, const CreateStreamParams& params);
};
#endif