Version 3.2

include/RtAudio.h

@@ -0,0 +1,155 @@
+/******************************************/
+/*
+  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