Version 4.0

This commit is contained in:
Gary Scavone
2013-09-25 14:50:19 +02:00
committed by Stephen Sinclair
parent 3f126af4e5
commit 81475b04c5
473 changed files with 36355 additions and 28396 deletions

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@@ -1,155 +1,434 @@
/******************************************/
/*
RtAudio.cpp
Realtime Sound I/O Object for STK
by Gary P. Scavone, 1998-2000.
The sound output sections of this object
were originally based in part on code by
Doug Scott (SGI), Tim Stilson (Linux),
Bill Putnam (Win Wav), and R. Marsanyi
(DirectSound). The latest DirectSound
code was re-written by Dave Chisholm at
CCRMA.
This object provides a standard API
across all platforms for STK realtime
audio input/output. Multi-channel
support is supported when provided by
the soundcard.
Only 16-bit integer input/output
routines are written for the moment
though it would be simple to overload
the methods for other data types.
*/
/******************************************/
#if !defined(__RtAudio_h)
#define __RtAudio_h
#include "Object.h"
#include "StkError.h"
#if defined(__OS_IRIX_)
#include <dmedia/audio.h>
#include <unistd.h>
#include <errno.h>
#elif defined(__ALSA_API_)
#include <sys/ioctl.h>
#include <sys/asoundlib.h>
#elif defined(__OSS_API_)
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <errno.h>
#elif defined(__OS_Win_)
#include <windows.h>
#include <dsound.h>
#include <winsock.h>
#include "mmsystem.h"
// this is how often we check for new audio input (milliseconds)
#define TIMER_PERIOD 20
// the resolution which we tell windows we are willing to tolerate (milliseconds)
#define TIMER_RESOLUTION 5
// in seconds, doesn't have a real effect on latency
#define DS_CAPTURE_BUFFER_SIZE 2
// this controls the inherent latency of the output ... more fragments produce
// a more stable, though slower, response
#define NUM_FRAGMENTS 10
#define MAX_DEVICES 10
typedef struct DeviceInfo {
LPGUID guid;
char* description;
char* moduleName;
} DeviceInfo;
#endif
class RtAudio : public Object
{
protected:
#if (defined(__STK_REALTIME_) && defined(__OS_IRIX_))
int stk_chans;
ALport audio_port_in;
ALport audio_port_out;
#elif (defined(__STK_REALTIME_) && defined(__OSS_API_))
int audio_fd;
#elif (defined(__STK_REALTIME_) && defined(__ALSA_API_))
snd_pcm_t *ohandle;
snd_pcm_t *ihandle;
int stk_chans; // the number of channels we want to use
int dev_ichans; // the number of input channels the device needs
int dev_ochans; // the number of output channels the device needs
int ifragsize;
int ofragsize;
int bytes_per_sample;
unsigned int direction;
unsigned char *outbuf;
unsigned char *inbuf;
#elif (defined(__STK_REALTIME_) && defined(__OS_Win_) )
DeviceInfo devices[MAX_DEVICES];
int numDevices;
char errormsg[256];
long inputBufferSize;
BYTE *inputBuffer;
UINT nextRecordRead, nextRecordWrite;
MY_FLOAT sampleRate;
//these are the variable relating to direct sound output
LPDIRECTSOUND directSoundObject;
LPDIRECTSOUNDBUFFER directSoundBuffer;
DWORD directSoundBufferSize;
UINT nextWritePos;
//direct sound input
LPDIRECTSOUNDCAPTURE directSoundCaptureObject;
LPDIRECTSOUNDCAPTUREBUFFER directSoundCaptureBuffer;
DWORD directSoundCaptureBufferSize;
// our periodic function will set this flag if something goes wrong
bool internalError;
bool playing, recording;
UINT timerID;
static void CALLBACK PeriodicCallbackFn(UINT uID, UINT uMsg, DWORD dwUser,
DWORD dw1, DWORD dw2);
static bool CALLBACK SoundDeviceEnumCallback(LPGUID lpguid,
LPCSTR lpcstrDescription,
LPCSTR lpcstrModule,
LPVOID lpContext);
void addDevice(LPGUID guid, char* description, char* moduleName);
void getInputSamples();
static char* getErrorMessage(int code);
#endif
public:
RtAudio(int channels, MY_FLOAT srate, const char *mode, int device = -1);
~RtAudio();
int playBuffer(INT16 *buf, int bufsize);
int recordBuffer(INT16 *buf, int bufsize);
#if (defined(__STK_REALTIME_) && defined(__OS_Win_) )
// Sets the pointer to its own internal buffer, returning
// amount of data available ... slightly more efficient.
int recordBuffer(INT16**);
void stopPlay();
void startPlay();
void stopRecord();
void startRecord();
#endif
};
#endif
/******************************************/
/*
RtAudio - realtime sound I/O C++ class
by Gary P. Scavone, 2001-2002.
*/
/******************************************/
#if !defined(__RTAUDIO_H)
#define __RTAUDIO_H
#include <map>
#if defined(__LINUX_ALSA__)
#include <alsa/asoundlib.h>
#include <pthread.h>
#include <unistd.h>
#define THREAD_TYPE
typedef snd_pcm_t *AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
#elif defined(__LINUX_OSS__)
#include <pthread.h>
#include <unistd.h>
#define THREAD_TYPE
typedef int AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
#elif defined(__WINDOWS_DS__)
#include <windows.h>
#include <process.h>
// The following struct is used to hold the extra variables
// specific to the DirectSound implementation.
typedef struct {
void * object;
void * buffer;
UINT bufferPointer;
} AUDIO_HANDLE;
#define THREAD_TYPE __stdcall
typedef LPGUID DEVICE_ID;
typedef unsigned long THREAD_HANDLE;
typedef CRITICAL_SECTION MUTEX;
#elif defined(__IRIX_AL__)
#include <dmedia/audio.h>
#include <pthread.h>
#include <unistd.h>
#define THREAD_TYPE
typedef ALport AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
#endif
// *************************************************** //
//
// RtError class declaration.
//
// *************************************************** //
class RtError
{
public:
enum TYPE {
WARNING,
DEBUG_WARNING,
UNSPECIFIED,
NO_DEVICES_FOUND,
INVALID_DEVICE,
INVALID_STREAM,
MEMORY_ERROR,
INVALID_PARAMETER,
DRIVER_ERROR,
SYSTEM_ERROR,
THREAD_ERROR
};
protected:
char error_message[256];
TYPE type;
public:
//! The constructor.
RtError(const char *p, TYPE tipe = RtError::UNSPECIFIED);
//! The destructor.
virtual ~RtError(void);
//! Prints "thrown" error message to stdout.
virtual void printMessage(void);
//! Returns the "thrown" error message TYPE.
virtual const TYPE& getType(void) { return type; }
//! Returns the "thrown" error message string.
virtual const char *getMessage(void) { return error_message; }
};
// *************************************************** //
//
// RtAudio class declaration.
//
// *************************************************** //
class RtAudio
{
public:
// Support for signed integers and floats. Audio data fed to/from
// the tickStream() routine 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.
typedef unsigned long RTAUDIO_FORMAT;
static const RTAUDIO_FORMAT RTAUDIO_SINT8;
static const RTAUDIO_FORMAT RTAUDIO_SINT16;
static const RTAUDIO_FORMAT RTAUDIO_SINT24; /*!< Upper 3 bytes of 32-bit integer. */
static const RTAUDIO_FORMAT RTAUDIO_SINT32;
static const RTAUDIO_FORMAT RTAUDIO_FLOAT32; /*!< Normalized between plus/minus 1.0. */
static const RTAUDIO_FORMAT RTAUDIO_FLOAT64; /*!< Normalized between plus/minus 1.0. */
//static const int MAX_SAMPLE_RATES = 14;
enum { MAX_SAMPLE_RATES = 14 };
typedef int (*RTAUDIO_CALLBACK)(char *buffer, int bufferSize, void *userData);
typedef struct {
char name[128];
DEVICE_ID id[2]; /*!< No value reported by getDeviceInfo(). */
bool probed; /*!< true if the device capabilities were successfully probed. */
int maxOutputChannels;
int maxInputChannels;
int maxDuplexChannels;
int minOutputChannels;
int minInputChannels;
int minDuplexChannels;
bool hasDuplexSupport; /*!< true if device supports duplex mode. */
int nSampleRates; /*!< Number of discrete rates or -1 if range supported. */
int sampleRates[MAX_SAMPLE_RATES]; /*!< Supported rates or (min, max) if range. */
RTAUDIO_FORMAT nativeFormats; /*!< Bit mask of supported data formats. */
} RTAUDIO_DEVICE;
//! The default constructor.
/*!
Probes the system to make sure at least one audio
input/output device is available and determines
the api-specific identifier for each device found.
An RtError error can be thrown if no devices are
found or if a memory allocation error occurs.
*/
RtAudio();
//! A constructor which can be used to open a stream during instantiation.
/*!
The specified output and/or input device identifiers correspond
to those enumerated via the getDeviceInfo() method. If device =
0, the default or first available devices meeting the given
parameters is selected. If an output or input channel value is
zero, the corresponding device value is ignored. When a stream is
successfully opened, its identifier is returned via the "streamId"
pointer. An RtError can be thrown if no devices are found
for the given parameters, if a memory allocation error occurs, or
if a driver error occurs. \sa openStream()
*/
RtAudio(int *streamId,
int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RTAUDIO_FORMAT format, int sampleRate,
int *bufferSize, int numberOfBuffers);
//! The destructor.
/*!
Stops and closes any open streams and devices and deallocates
buffer and structure memory.
*/
~RtAudio();
//! A public method for opening a stream with the specified parameters.
/*!
If successful, the opened stream ID is returned. Otherwise, an
RtError is thrown.
\param outputDevice: If equal to 0, the default or first device
found meeting the given parameters is opened. Otherwise, the
device number should correspond to one of those enumerated via
the getDeviceInfo() method.
\param outputChannels: The desired number of output channels. If
equal to zero, the outputDevice identifier is ignored.
\param inputDevice: If equal to 0, the default or first device
found meeting the given parameters is opened. Otherwise, the
device number should correspond to one of those enumerated via
the getDeviceInfo() method.
\param inputChannels: The desired number of input channels. If
equal to zero, the inputDevice identifier is ignored.
\param format: An RTAUDIO_FORMAT specifying the desired sample data format.
\param sampleRate: The desired sample rate (sample frames per second).
\param *bufferSize: A pointer value indicating the desired internal buffer
size in sample frames. The actual value used by the device is
returned via the same pointer. A value of zero can be specified,
in which case the lowest allowable value is determined.
\param numberOfBuffers: A value which can be used to help control device
latency. More buffers typically result in more robust performance,
though at a cost of greater latency. A value of zero can be
specified, in which case the lowest allowable value is used.
*/
int openStream(int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RTAUDIO_FORMAT format, int sampleRate,
int *bufferSize, int numberOfBuffers);
//! A public method which sets a user-defined callback function for a given stream.
/*!
This method assigns a callback function to a specific,
previously opened stream for non-blocking stream functionality. A
separate process is initiated, though the user function is called
only when the stream is "running" (between calls to the
startStream() and stopStream() methods, respectively). The
callback process remains active for the duration of the stream and
is automatically shutdown when the stream is closed (via the
closeStream() method or by object destruction). The callback
process can also be shutdown and the user function de-referenced
through an explicit call to the cancelStreamCallback() method.
Note that a single stream can use only blocking or callback
functionality at the same time, though it is possible to alternate
modes on the same stream through the use of the
setStreamCallback() and cancelStreamCallback() methods (the
blocking tickStream() method can be used before a callback is set
and/or after a callback is cancelled). An RtError will be
thrown for an invalid device argument.
*/
void setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData);
//! A public method which cancels a callback process and function for a given stream.
/*!
This method shuts down a callback process and de-references the
user function for a specific stream. Callback functionality can
subsequently be restarted on the stream via the
setStreamCallback() method. An RtError will be thrown for an
invalid device argument.
*/
void cancelStreamCallback(int streamId);
//! A public method which returns the number of audio devices found.
int getDeviceCount(void);
//! Fill a user-supplied RTAUDIO_DEVICE structure for a specified device.
/*!
Any device between 0 and getDeviceCount()-1 is valid. If a
device is busy or otherwise unavailable, the structure member
"probed" has a value of "false". The system default input and
output devices are referenced by device identifier = 0. On
systems which allow dynamic default device settings, the default
devices are not identified by name (specific device enumerations
are assigned device identifiers > 0). An RtError will be
thrown for an invalid device argument.
*/
void getDeviceInfo(int device, RTAUDIO_DEVICE *info);
//! A public method which returns a pointer to the buffer for an open stream.
/*!
The user should fill and/or read the buffer data in interleaved format
and then call the tickStream() method. An RtError will be
thrown for an invalid stream identifier.
*/
char * const getStreamBuffer(int streamId);
//! Public method used to trigger processing of input/output data for a stream.
/*!
This method blocks until all buffer data is read/written. An
RtError will be thrown for an invalid stream identifier or if
a driver error occurs.
*/
void tickStream(int streamId);
//! Public method which closes a stream and frees any associated buffers.
/*!
If an invalid stream identifier is specified, this method
issues a warning and returns (an RtError is not thrown).
*/
void closeStream(int streamId);
//! Public method which starts a stream.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
*/
void startStream(int streamId);
//! Stop a stream, allowing any samples remaining in the queue to be played out and/or read in.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
*/
void stopStream(int streamId);
//! Stop a stream, discarding any samples remaining in the input/output queue.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
*/
void abortStream(int streamId);
//! Queries a stream to determine whether a call to the tickStream() method will block.
/*!
A return value of 0 indicates that the stream will NOT block. A positive
return value indicates the number of sample frames that cannot yet be
processed without blocking.
*/
int streamWillBlock(int streamId);
protected:
private:
static const unsigned int SAMPLE_RATES[MAX_SAMPLE_RATES];
enum { FAILURE, SUCCESS };
enum STREAM_MODE {
PLAYBACK,
RECORD,
DUPLEX,
UNINITIALIZED = -75
};
enum STREAM_STATE {
STREAM_STOPPED,
STREAM_RUNNING
};
typedef struct {
int device[2]; // Playback and record, respectively.
STREAM_MODE mode; // PLAYBACK, RECORD, or DUPLEX.
AUDIO_HANDLE handle[2]; // Playback and record handles, respectively.
STREAM_STATE state; // STOPPED or RUNNING
char *userBuffer;
char *deviceBuffer;
bool doConvertBuffer[2]; // Playback and record, respectively.
bool deInterleave[2]; // Playback and record, respectively.
bool doByteSwap[2]; // Playback and record, respectively.
int sampleRate;
int bufferSize;
int nBuffers;
int nUserChannels[2]; // Playback and record, respectively.
int nDeviceChannels[2]; // Playback and record channels, respectively.
RTAUDIO_FORMAT userFormat;
RTAUDIO_FORMAT deviceFormat[2]; // Playback and record, respectively.
bool usingCallback;
THREAD_HANDLE thread;
MUTEX mutex;
RTAUDIO_CALLBACK callback;
void *userData;
} RTAUDIO_STREAM;
typedef signed short INT16;
typedef signed int INT32;
typedef float FLOAT32;
typedef double FLOAT64;
char message[256];
int nDevices;
RTAUDIO_DEVICE *devices;
std::map<int, void *> streams;
//! Private error method to allow global control over error handling.
void error(RtError::TYPE type);
/*!
Private method to count the system audio devices, allocate the
RTAUDIO_DEVICE structures, and probe the device capabilities.
*/
void initialize(void);
//! Private method to clear an RTAUDIO_DEVICE structure.
void clearDeviceInfo(RTAUDIO_DEVICE *info);
/*!
Private method which attempts to fill an RTAUDIO_DEVICE
structure for a given device. If an error is encountered during
the probe, a "warning" message is reported and the value of
"probed" remains false (no exception is thrown). A successful
probe is indicated by probed = true.
*/
void probeDeviceInfo(RTAUDIO_DEVICE *info);
/*!
Private method which attempts to open a device with the given parameters.
If an error is encountered during the probe, a "warning" message is
reported and FAILURE is returned (no exception is thrown). A
successful probe is indicated by a return value of SUCCESS.
*/
bool probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
STREAM_MODE mode, int channels,
int sampleRate, RTAUDIO_FORMAT format,
int *bufferSize, int numberOfBuffers);
/*!
Private common method used to check validity of a user-passed
stream ID. When the ID is valid, this method returns a pointer to
an RTAUDIO_STREAM structure (in the form of a void pointer).
Otherwise, an "invalid identifier" exception is thrown.
*/
void *verifyStream(int streamId);
/*!
Private method used to perform format, channel number, and/or interleaving
conversions between the user and device buffers.
*/
void convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode);
//! Private method used to perform byte-swapping on buffers.
void byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format);
//! Private method which returns the number of bytes for a given format.
int formatBytes(RTAUDIO_FORMAT format);
};
// Uncomment the following definition to have extra information spewed to stderr.
//#define RTAUDIO_DEBUG
#endif