Binaural beats in C++ - c++

Question

I am trying to build a brain device using EEG input and outputting light pulses and binaural beats in "close to real-time" on a Raspberry PI. The light-output is no problem using WiringPi, but any audio output seems to be a major hurdle. The math of calculating a sinewave for a buffer is straightforward, but playing two frequencies on two channels via any standard libraries seems to be a very complicated process, and I can't come up with any relevant examples. I have successfully opened and closed an ALSA device thanks to this tutorial which complicates my fairly simple code tremendously but appears necessary for ALSA. I would be incredibly grateful if someone could show me the easiest method for playing two different calculated tones on left and right channels. The code below is the simplest ALSA playback example I could find.
#include <alsa/asoundlib.h>
#include <iostream>
using namespace std;
// Globals are generally a bad idea in code. We're using one here to keep it simple.
snd_pcm_t * _soundDevice;
bool Init(const char *name)
{
int i;
int err;
snd_pcm_hw_params_t *hw_params;
if( name == NULL )
{
// Try to open the default device
err = snd_pcm_open( &_soundDevice, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0 );
}
else
{
// Open the device we were told to open.
err = snd_pcm_open (&_soundDevice, name, SND_PCM_STREAM_PLAYBACK, 0);
}
// Check for error on open.
if( err < 0 )
{
cout << "Init: cannot open audio device " << name << " (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device opened successfully." << endl;
}
// Allocate the hardware parameter structure.
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{
cout << "Init: cannot allocate hardware parameter structure (" << snd_strerror (err) << ")" << endl;
return false;
}
if ((err = snd_pcm_hw_params_any (_soundDevice, hw_params)) < 0)
{
cout << "Init: cannot initialize hardware parameter structure (" << snd_strerror (err) << ")" << endl;
return false;
}
// Enable resampling.
unsigned int resample = 1;
err = snd_pcm_hw_params_set_rate_resample(_soundDevice, hw_params, resample);
if (err < 0)
{
cout << "Init: Resampling setup failed for playback: " << snd_strerror(err) << endl;
return err;
}
// Set access to RW interleaved.
if ((err = snd_pcm_hw_params_set_access (_soundDevice, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
cout << "Init: cannot set access type (" << snd_strerror (err) << ")" << endl;
return false;
}
if ((err = snd_pcm_hw_params_set_format (_soundDevice, hw_params, SND_PCM_FORMAT_S16_LE)) < 0)
{
cout << "Init: cannot set sample format (" << snd_strerror (err) << ")" << endl;
return false;
}
// Set channels to stereo (2).
if ((err = snd_pcm_hw_params_set_channels (_soundDevice, hw_params, 2)) < 0)
{
cout << "Init: cannot set channel count (" << snd_strerror (err) << ")" << endl;
return false;
}
// Set sample rate.
unsigned int actualRate = 44100;
if ((err = snd_pcm_hw_params_set_rate_near (_soundDevice, hw_params, &actualRate, 0)) < 0)
{
cout << "Init: cannot set sample rate to 44100. (" << snd_strerror (err) << ")" << endl;
return false;
}
if( actualRate < 44100 )
{
cout << "Init: sample rate does not match requested rate. (" << "44100 requested, " << actualRate << " acquired)" << endl;
}
// Apply the hardware parameters that we've set.
if ((err = snd_pcm_hw_params (_soundDevice, hw_params)) < 0)
{
cout << "Init: cannot set parameters (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device parameters have been set successfully." << endl;
}
// Get the buffer size.
snd_pcm_uframes_t bufferSize;
snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );
// If we were going to do more with our sound device we would want to store
// the buffer size so we know how much data we will need to fill it with.
cout << "Init: Buffer size = " << bufferSize << " frames." << endl;
// Display the bit size of samples.
cout << "Init: Significant bits for linear samples = " << snd_pcm_hw_params_get_sbits(hw_params) << endl;
// Free the hardware parameters now that we're done with them.
snd_pcm_hw_params_free (hw_params);
// Prepare interface for use.
if ((err = snd_pcm_prepare (_soundDevice)) < 0)
{
cout << "Init: cannot prepare audio interface for use (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device has been prepared for use." << endl;
}
return true;
}
bool UnInit()
{
snd_pcm_close (_soundDevice);
cout << "Audio device has been uninitialized." << endl;
return true;
}
int main( char *argc, int argv )
{
Init(NULL);
UnInit();
return 0;
}

Answer 1

Use some up-to-date example, like this:
#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
static const char *device = "default";
unsigned short buffer[2 * 24000];
int main(void)
{
int err;
snd_pcm_t *handle;
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
printf("open error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
if ((err = snd_pcm_set_params(handle,
SND_PCM_FORMAT_S16,
SND_PCM_ACCESS_RW_INTERLEAVED,
2, /* channels */
48000, /* rate */
1,
500000)) < 0) { /* buffer: 0.5 sec */
printf("open error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
for (;;) {
for (int i = 0; i < 24000; i++) {
buffer[2 * i + 0] = 32767 * sin(...); /* left channel */
buffer[2 * i + 1] = 32767 * sin(...); /* right channel */
}
snd_pcm_sframes_t frames = snd_pcm_writei(handle, buffer, 24000);
if (frames < 0)
frames = snd_pcm_recover(handle, frames, 0);
if (frames < 0) {
printf("snd_pcm_writei failed: %s\n", snd_strerror(err));
break;
}
}
snd_pcm_close(handle);
return 0;
}