comparison examples/basic_FFT_phase_vocoder/render.cpp @ 373:3bed6b09223c prerelease

Updated NE10 library to the latest version; needs a corresponding update to the /usr/include/ne10 header files on the SD image. Updated examples to compile against new version, and reordered D-Box channels to account for new PRU-based DAC channel reordering.

 };
 
 float gDryWet = 1; // mix between the unprocessed and processed sound
 float gPlaybackLive = 0.5f; // mix between the file playback and the live audio input
 float gGain = 1; // overall gain
+float *gInputAudio = NULL;
 Midi midi;
+
+
 void midiCallback(MidiChannelMessage message, void* arg){
 	if(message.getType() == kmmNoteOn){
 		if(message.getDataByte(1) > 0){
 			int note = message.getDataByte(0);
 			float frequency = powf(2, (note-69)/12.f)*440;

 	gOutputBufferWritePointer += gHopSize;
 
 	timeDomainIn = (ne10_fft_cpx_float32_t*) NE10_MALLOC (gFFTSize * sizeof (ne10_fft_cpx_float32_t));
 	timeDomainOut = (ne10_fft_cpx_float32_t*) NE10_MALLOC (gFFTSize * sizeof (ne10_fft_cpx_float32_t));
 	frequencyDomain = (ne10_fft_cpx_float32_t*) NE10_MALLOC (gFFTSize * sizeof (ne10_fft_cpx_float32_t));
-	cfg = ne10_fft_alloc_c2c_float32 (gFFTSize);
+	cfg = ne10_fft_alloc_c2c_float32_neon (gFFTSize);
 
 	memset(timeDomainOut, 0, gFFTSize * sizeof (ne10_fft_cpx_float32_t));
 	memset(gOutputBuffer, 0, BUFFER_SIZE * sizeof(float));
+
+	// Allocate buffer to mirror and modify the input
+	gInputAudio = (float *)malloc(context->audioFrames * context->audioChannels * sizeof(float));
+	if(gInputAudio == 0)
+		return false;
 
 	// Allocate the window buffer based on the FFT size
 	gWindowBuffer = (float *)malloc(gFFTSize * sizeof(float));
 	if(gWindowBuffer == 0)
 		return false;

 		if(pointer >= BUFFER_SIZE)
 			pointer = 0;
 	}
 
 	// Run the FFT
-	ne10_fft_c2c_1d_float32_neon (frequencyDomain, timeDomainIn, cfg->twiddles, cfg->factors, gFFTSize, 0);
+	ne10_fft_c2c_1d_float32_neon (frequencyDomain, timeDomainIn, cfg, 0);
 
 	switch (gEffect){
 	  case kRobot :
 	  // Robotise the output
 	    for(int n = 0; n < gFFTSize; n++) {

 	    //bypass
 	    break;
 	  }
 
 	// Run the inverse FFT
-	ne10_fft_c2c_1d_float32_neon (timeDomainOut, frequencyDomain, cfg->twiddles, cfg->factors, gFFTSize, 1);
+	ne10_fft_c2c_1d_float32_neon (timeDomainOut, frequencyDomain, cfg, 1);
 	// Overlap-and-add timeDomainOut into the output buffer
 	pointer = outWritePointer;
 	for(int n = 0; n < gFFTSize; n++) {
 		outBuffer[pointer] += (timeDomainOut[n].r) * gFFTScaleFactor;
 		if(isnan(outBuffer[pointer]))

 // Input and output are given from the audio hardware and the other
 // ADCs and DACs (if available). If only audio is available, numMatrixFrames
 // will be 0.
 void render(BelaContext* context, void* userData)
 {
-	float* audioIn = context->audioIn;
 	float* audioOut = context->audioOut;
 	int numAudioFrames = context->audioFrames;
 	int numAudioChannels = context->audioChannels;
 	// ------ this code internal to the demo; leave as is ----------------
 
 	// Prep the "input" to be the sound file played in a loop
 	for(int n = 0; n < numAudioFrames; n++) {
 		if(gReadPtr < gSampleData.sampleLen)
-			audioIn[2*n] = audioIn[2*n+1] = gSampleData.samples[gReadPtr]*(1-gPlaybackLive) +
+			gInputAudio[2*n] = gInputAudio[2*n+1] = gSampleData.samples[gReadPtr]*(1-gPlaybackLive) +
 			gPlaybackLive*0.5f*(audioRead(context,n,0)+audioRead(context,n,1));
 		else
-			audioIn[2*n] = audioIn[2*n+1] = 0;
+			gInputAudio[2*n] = gInputAudio[2*n+1] = 0;
 		if(++gReadPtr >= gSampleData.sampleLen)
 			gReadPtr = 0;
 	}
 	// -------------------------------------------------------------------
 
 	for(int n = 0; n < numAudioFrames; n++) {
-		gInputBuffer[gInputBufferPointer] = ((audioIn[n*numAudioChannels] + audioIn[n*numAudioChannels+1]) * 0.5);
+		gInputBuffer[gInputBufferPointer] = ((gInputAudio[n*numAudioChannels] + gInputAudio[n*numAudioChannels+1]) * 0.5);
 
 		// Copy output buffer to output
 		for(int channel = 0; channel < numAudioChannels; channel++){
-			audioOut[n * numAudioChannels + channel] = gOutputBuffer[gOutputBufferReadPointer] * gGain * gDryWet + (1 - gDryWet) * audioIn[n * numAudioChannels + channel];
+			audioOut[n * numAudioChannels + channel] = gOutputBuffer[gOutputBufferReadPointer] * gGain * gDryWet + (1 - gDryWet) * gInputAudio[n * numAudioChannels + channel];
 		}
 
 		// Clear the output sample in the buffer so it is ready for the next overlap-add
 		gOutputBuffer[gOutputBufferReadPointer] = 0;
 		gOutputBufferReadPointer++;

 {
 	NE10_FREE(timeDomainIn);
 	NE10_FREE(timeDomainOut);
 	NE10_FREE(frequencyDomain);
 	NE10_FREE(cfg);
+	free(gInputAudio);
 	free(gWindowBuffer);
 }