Version 3.2

src/RtWvIn.cpp

@@ -0,0 +1,146 @@
+/*******************************************/
+/*  RtWvIn Input Class,                    */
+/*  by Gary P. Scavone, 1999-2000          */
+/*                                         */
+/*  This object inherits from WvIn and is  */
+/*  used to read in realtime 16-bit data   */
+/*  from a computer's audio port.          */
+/*                                         */
+/*  NOTE: This object is NOT intended for  */
+/*  use in achieving simultaneous realtime */
+/*  audio input/output (together with      */
+/*  RtWvOut). Under certain circumstances  */
+/*  such a scheme is possible, though you  */
+/*  should definitely know what you are    */
+/*  doing before trying.  For safer "full- */
+/*  duplex" operation, use the RtDuplex    */
+/*  class.                                 */
+/*******************************************/
+
+#include "RtWvIn.h"
+
+RtWvIn :: RtWvIn(int chans, MY_FLOAT srate, int device)
+{
+  chunking = 1;
+  looping = 0;
+  sound_dev = new RtAudio(chans, srate, "record", device);
+  channels = chans;
+  bufferSize = RT_BUFFER_SIZE;
+  data = 0;
+  rtdata = (INT16 *) new INT16[(bufferSize+1)*channels];
+
+  lastSamples = (INT16 *) new INT16[channels];
+  for (int i=0;i<channels;i++) {
+    lastSamples[i] = (INT16) 0.0;
+  }
+
+  this->getData(0);
+
+  rate = (MY_FLOAT) srate / SRATE;
+  if (fmod(rate, 1.0) > 0.0) interpolate = 1;
+  else interpolate = 0;
+  phaseOffset = (MY_FLOAT) 0.0;
+  lastOutput = (MY_FLOAT *) new MY_FLOAT[channels];
+  this->reset();
+
+  gain = 0.00003052;
+#if (defined(__STK_REALTIME_) && defined(__OS_IRIX_))
+  // This is necessary under IRIX because it scales the input by 0.5
+  // when using single-channel input.
+  if (channels == 1) gain *= 2;
+#endif
+}
+
+RtWvIn :: ~RtWvIn()
+{
+  delete sound_dev;
+  if (rtdata) {
+    delete [ ] rtdata;
+    rtdata = 0;
+  }
+  if (lastSamples) {
+    delete [ ] lastSamples;
+    lastSamples = 0;
+  }
+}
+
+void RtWvIn :: setRate(MY_FLOAT aRate)
+{
+  // Negative rates not allowed for realtime input
+  rate = fabs(aRate);
+
+  if (fmod(rate, 1.0) != 0.0) interpolate = 1;
+  else interpolate = 0;
+}
+
+void RtWvIn :: addTime(MY_FLOAT aTime)   
+{
+  // Negative time shift no allowed for realtime input
+  time += fabs(aTime);
+}
+
+void RtWvIn :: setLooping(int aLoopStatus)
+{
+  // No looping for realtime data.
+  looping = 0;
+}
+
+long RtWvIn :: getSize()
+{
+  return bufferSize;
+}
+
+void RtWvIn :: getData(long index)
+{
+  static long temp = RT_BUFFER_SIZE*channels;
+  sound_dev->recordBuffer(&rtdata[channels],temp);
+
+  /* Fill in the extra sample frame for interpolation.
+   * We do this by pre-pending the last sample frame
+   * from the previous input buffer to the current one.
+   */
+  for (int i=0;i<channels;i++) {
+    rtdata[i] = lastSamples[i];
+    lastSamples[i] = rtdata[temp+i];
+  }
+}
+
+int RtWvIn :: informTick()
+{
+  static MY_FLOAT alpha;
+  static long index;
+
+  if (time >= bufferSize) {
+    this->getData(0);
+    while (time >= bufferSize)
+      time -= bufferSize;
+  }
+
+  // integer part of time address
+  index = (long) time;
+
+  if (interpolate) {
+    // fractional part of time address
+    alpha = time - (MY_FLOAT) index;
+    index *= channels;
+    for (int i=0;i<channels;i++) {
+      //  Do linear interpolation
+      lastOutput[i] = (MY_FLOAT) rtdata[index];
+      lastOutput[i] += alpha * ((MY_FLOAT) rtdata[index+channels] - lastOutput[i]);
+      lastOutput[i] *= gain;
+      index++;
+    }
+  }
+  else {
+    index *= channels;
+    for (int i=0;i<channels;i++) {
+      lastOutput[i] = rtdata[index++];
+      lastOutput[i] *= gain;
+    }
+  }
+
+  // increment time, which can be negative
+  time += rate;
+
+  return finished;
+}