Real-time, ultra-low-latency audio and sensor processing system for BeagleBone Black
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Playback of large wav files

When dealing with large wav files it is usually not a good idea to load the entire file into memory. This example shows how to use two buffers to continually load chunks of the file, thus allowing playback of very large files. While one buffer is being used for playback the other buffer is being filled with the next chunk of samples.

In order to do this, an AuxiliaryTask is used to load the file into the inactive buffer without interrupting the audio thread. We can set the global variable gDoneLoadingBuffer to 1 each time the buffer has finished loading, allowing us to detect cases were the buffers havn't been filled in time. These cases can usually be mitigated by using a larger buffer size.

Try uploading a large wav file into the project and playing it back. You will need specify the amount of channels (#define NUM_CHANNELS) and the name of the file (gFilename).

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The platform for ultra-low latency audio and sensor processing
A project of the Augmented Instruments Laboratory within the
Centre for Digital Music at Queen Mary University of London.
(c) 2016 Augmented Instruments Laboratory: Andrew McPherson,
Astrid Bin, Liam Donovan, Christian Heinrichs, Robert Jack,
Giulio Moro, Laurel Pardue, Victor Zappi. All rights reserved.
The Bela software is distributed under the GNU Lesser General Public License
(LGPL 3.0), available here:
#include <Bela.h>
#include <libraries/AudioFile/AudioFile.h>
#include <vector>
std::string gFilename = "waves.wav";
int gNumFramesInFile;
// Two buffers for each channel:
// one of them loads the next chunk of audio while the other one is used for playback
std::vector<std::vector<float> > gSampleBuf[2];
// read pointer relative current buffer (range 0-BUFFER_LEN)
// initialise at BUFFER_LEN to pre-load second buffer (see render())
int gReadPtr = BUFFER_LEN;
// read pointer relative to file, increments by BUFFER_LEN (see fillBuffer())
int gBufferReadPtr = 0;
// keeps track of which buffer is currently active (switches between 0 and 1)
int gActiveBuffer = 0;
// this variable will let us know if the buffer doesn't manage to load in time
int gDoneLoadingBuffer = 1;
AuxiliaryTask gFillBufferTask;
void fillBuffer(void*) {
// increment buffer read pointer by buffer length
// reset buffer pointer if it exceeds the number of frames in the file
int endFrame = gBufferReadPtr + BUFFER_LEN;
int zeroPad = 0;
// if reaching the end of the file take note of the last frame index
// so we can zero-pad the rest later
if((gBufferReadPtr+BUFFER_LEN)>=gNumFramesInFile-1) {
endFrame = gNumFramesInFile-1;
zeroPad = 1;
for(int ch=0;ch<gSampleBuf[0].size();ch++) {
// fill (nonactive) buffer
// zero-pad if necessary
if(zeroPad) {
int numFramesToPad = BUFFER_LEN - (endFrame-gBufferReadPtr);
for(int n=0;n<numFramesToPad;n++)
gSampleBuf[!gActiveBuffer][ch][n+(BUFFER_LEN-numFramesToPad)] = 0;
gDoneLoadingBuffer = 1;
//printf("done loading buffer!\n");
bool setup(BelaContext *context, void *userData)
// Initialise auxiliary tasks
if((gFillBufferTask = Bela_createAuxiliaryTask(&fillBuffer, 90, "fill-buffer")) == 0)
return false;
gNumFramesInFile = AudioFileUtilities::getNumFrames(gFilename);
if(gNumFramesInFile <= 0)
return false;
if(gNumFramesInFile <= BUFFER_LEN) {
printf("Sample needs to be longer than buffer size. This example is intended to work with long samples.");
return false;
gSampleBuf[0] = AudioFileUtilities::load(gFilename, BUFFER_LEN, 0);
gSampleBuf[1] = gSampleBuf[0]; // initialise the inactive buffer with the same channels and frames as the first one
return true;
void render(BelaContext *context, void *userData)
for(unsigned int n = 0; n < context->audioFrames; n++) {
// Increment read pointer and reset to 0 when end of file is reached
if(++gReadPtr >= BUFFER_LEN) {
rt_printf("Couldn't load buffer in time :( -- try increasing buffer size!");
gDoneLoadingBuffer = 0;
gReadPtr = 0;
gActiveBuffer = !gActiveBuffer;
for(unsigned int channel = 0; channel < context->audioOutChannels; channel++) {
// Wrap channel index in case there are more audio output channels than the file contains
float out = gSampleBuf[gActiveBuffer][channel%gSampleBuf[0].size()][gReadPtr];
audioWrite(context, n, channel, out);
void cleanup(BelaContext *context, void *userData)