diff examples/basic_FFT_phase_vocoder/render.cpp @ 375:768acdeea362 prerelease

Merge

--- a/examples/basic_FFT_phase_vocoder/render.cpp	Fri Jun 10 00:02:48 2016 +0100
+++ b/examples/basic_FFT_phase_vocoder/render.cpp	Fri Jun 10 00:35:18 2016 +0100
@@ -1,3 +1,12 @@
+/*
+ ____  _____ _        _    
+| __ )| ____| |      / \   
+|  _ \|  _| | |     / _ \  
+| |_) | |___| |___ / ___ \ 
+|____/|_____|_____/_/   \_\.io
+
+ */
+
 /*
  * render.cpp
  *
@@ -5,6 +14,17 @@
  *      Author: parallels
  */
 
+/**
+\example 4_audio_FFT_phase_vocoder
+
+Phase Vocoder
+----------------------
+
+This sketch shows an implementation of a phase vocoder and builds on the previous FFT example.
+Again it uses the NE10 library, included at the top of the file (line 11).
+
+Read the documentation on the NE10 library [here](http://projectne10.github.io/Ne10/doc/annotated.html).
+*/
 
 #include <Bela.h>
 #include <rtdk.h>
@@ -60,7 +80,10 @@
 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){
@@ -113,11 +136,16 @@
 	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)
@@ -157,7 +185,7 @@
 	}
 
 	// 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 :
@@ -182,7 +210,7 @@
 	  }
 
 	// 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++) {
@@ -206,7 +234,6 @@
 // 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;
@@ -215,21 +242,21 @@
 	// 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
@@ -267,5 +294,6 @@
 	NE10_FREE(timeDomainOut);
 	NE10_FREE(frequencyDomain);
 	NE10_FREE(cfg);
+	free(gInputAudio);
 	free(gWindowBuffer);
 }