An Audio Synthesis Textbook For Musicians, Digital Artists and Programmers by Mike Krzyzaniak
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  1. //WavetableSynthesis.c
  2. //gcc MKAiff.c WavetableSynthesis.c -o WavetableSynthesis
  5. #include "MKAiff.h"
  6. #include <math.h>
  8. #define PRERECORDED_AUDIO_SAMPLE_PATH "demo.aif"
  10. #define SAMPLE_RATE 44100
  11. #define BITS_PER_SAMPLE 16
  12. #define NUM_SECONDS 7
  13. #define PLAYBACK_PITCH 70
  15. void linearInterpolateBuffer(float* previousFrame, int numChannels, float* input, int inNumFrames, float* output, int outNumFrames);
  17. int main()
  18. {
  19. MKAiff* audioSample = aiffWithContentsOfFile(PRERECORDED_AUDIO_SAMPLE_PATH);
  20. if((audioSample == NULL) || (!aiffHasInstrumentInfo(audioSample))) return 1;
  21. const int NUM_CHANNELS = aiffNumChannels(audioSample);
  22. const int numSamples = NUM_SECONDS * NUM_CHANNELS * SAMPLE_RATE;
  24. //get the original wavetable
  25. int wavetableNumSamples = aiffDurationInSamples(audioSample);
  26. float* wavetable = malloc(wavetableNumSamples * sizeof(*wavetable));
  27. aiffRewindPlayheadToBeginning(audioSample);
  28. aiffReadFloatingPointSamplesAtPlayhead(audioSample, wavetable, numSamples);
  30. //get the instrument info and adjust it for interpolation
  31. uint32_t numSamplesInAttack=0, numSamplesInSustain, numSamplesUntilRelease, numSamplesInRelease;
  32. aiffPositionInFramesOfMarkerWithID(audioSample, aiffInstrumentSustainLoopStartMarkerID(audioSample), &numSamplesInAttack);
  33. aiffPositionInFramesOfMarkerWithID (audioSample, aiffInstrumentSustainLoopEndMarkerID(audioSample), &numSamplesUntilRelease);
  34. double playbackSpeed = pow(pow(2, (aiffInstrumentBaseNote(audioSample) - PLAYBACK_PITCH)), (1/12.0));
  35. numSamplesInAttack *= NUM_CHANNELS * playbackSpeed;
  36. numSamplesUntilRelease *= NUM_CHANNELS * playbackSpeed;
  37. numSamplesInSustain = numSamplesUntilRelease - numSamplesInAttack;
  38. numSamplesInRelease = wavetableNumSamples*playbackSpeed - numSamplesUntilRelease;
  40. //interpolate the wavetable
  41. int i;
  42. int numInterpolatedSamples = wavetableNumSamples * playbackSpeed ;
  43. numInterpolatedSamples -= (numInterpolatedSamples % NUM_CHANNELS);
  44. float *interpolatedWavetable = malloc(numInterpolatedSamples * sizeof(*interpolatedWavetable));
  45. float previousFrame [NUM_CHANNELS]; for(i=0; i<NUM_CHANNELS; previousFrame[i++]=0);
  46. linearInterpolateBuffer(previousFrame, NUM_CHANNELS, wavetable, wavetableNumSamples/NUM_CHANNELS, interpolatedWavetable, numInterpolatedSamples/NUM_CHANNELS);
  48. //perform the synthesis
  49. float* nextSample = interpolatedWavetable;
  50. int shouldRelease = aiffNo, forwardSustain = aiffYes;
  51. float* audioBuffer = malloc(sizeof(*audioBuffer) * numSamples);
  53. for(i=0; i<numSamples; i++)
  54. {
  55. //trigger the release
  56. if(i >= (numSamples - (2*numSamplesInSustain + numSamplesInRelease)))
  57. shouldRelease = aiffYes;
  58. if((!shouldRelease) && forwardSustain && (nextSample >= (interpolatedWavetable + numSamplesUntilRelease)))
  59. forwardSustain = aiffNo;
  60. if((!forwardSustain) && (nextSample <= interpolatedWavetable + numSamplesInAttack))
  61. forwardSustain = aiffYes;
  62. if(nextSample >= interpolatedWavetable + numInterpolatedSamples)
  63. audioBuffer[i] = 0;
  64. else
  65. {
  66. audioBuffer[i] = *nextSample;
  67. nextSample += (forwardSustain) ? 1 : -1;
  68. }
  69. }
  71. MKAiff* aiff = aiffWithDurationInSeconds(NUM_CHANNELS, SAMPLE_RATE, BITS_PER_SAMPLE, NUM_SECONDS);
  72. if(aiff == NULL) return 1;
  73. aiffAppendFloatingPointSamples(aiff, audioBuffer, numSamples, aiffFloatSampleType);
  74. aiffSaveWithFilename(aiff, "WavetableSynthesis.aif");
  76. aiffDestroy(aiff);
  77. free(interpolatedWavetable);
  78. free(wavetable);
  80. return 0;
  81. }
  83. void linearInterpolateBuffer(float* previousFrame, int numChannels, float* input, int inNumFrames, float* output, int outNumFrames)
  84. {
  85. int i, j, prevIndex;
  86. double distance, prevValue, nextValue;
  88. for(i=0; i<outNumFrames; i++)
  89. {
  90. for(j=0; j<numChannels; j++)
  91. {
  92. distance = i * (inNumFrames / (double)outNumFrames) - 1;
  93. prevIndex = ((int)distance) * numChannels + j;
  94. nextValue = distance < 0 ? input[j] : input[prevIndex + numChannels];
  95. prevValue = distance < 0 ? previousFrame[j] : input[prevIndex];
  96. distance -= (int)distance;
  97. if(distance < 0) distance += 1;
  98. output[i*numChannels+j] = (nextValue-prevValue) * distance + prevValue;
  99. }
  100. }
  101. for(j=0; j<numChannels; j++)
  102. previousFrame[j] = input[(inNumFrames - 1) * numChannels + j];
  103. }

Output:

Downloads

Download Code Download Audio Download MKAiff

Builds On

AIFF Template Linear Interpolation How To Use Interpolation Preparing Samples Forward Backward Pitch

Explanation of the Concepts

This example is a relatively complete wavetable synthesis algorithm. It reads the "Attack Buffer", "Sustain Loop", and "Release Buffer" from the audio-sample "demo.aif". It transposes these using linear interpolation, and fills a new audio file with a single note by looping the sustain portion of the note.

In the previous examples, the wavetables were copied into an aiff file using commands that are specific to MKAiff. This examples attempts to more closely emulate a realtime environment by copying the wavetable into a final audio buffer that is attached to the aiff file only at the very end. The final audio buffer wastes a good deal of space, but it emulates a realtime output stream closely enough that the actual synthesis algorithm no longer relies upon MKAiff. Here, the release portion of the note is triggered by the approach of the end of the file. In the context of a larger program, this could be triggered by something external, like the key of a keyboard being released.

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