Is there any way to make Android emit a sound of arbitrary frequency (meaning, I don't want to have pre-recorded sound files)?
I've looked around and ToneGenerator was the only thing I was able to find that was even close, but it seems to only be capable of outputting the standard DTMF tones.
Any ideas?
I originally found this example code on a blog, but it had some bugs in it that generated some horrendous sounds. I've fixed the bugs and posted the resulting code here. Seems to work well for me!
public class PlaySound extends Activity {
// originally from http://marblemice.blogspot.com/2010/04/generate-and-play-tone-in-android.html
// and modified by Steve Pomeroy <[email protected]>
private final int duration = 3; // seconds
private final int sampleRate = 8000;
private final int numSamples = duration * sampleRate;
private final double sample[] = new double[numSamples];
private final double freqOfTone = 440; // hz
private final byte generatedSnd[] = new byte[2 * numSamples];
Handler handler = new Handler();
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
}
@Override
protected void onResume() {
super.onResume();
// Use a new tread as this can take a while
final Thread thread = new Thread(new Runnable() {
public void run() {
genTone();
handler.post(new Runnable() {
public void run() {
playSound();
}
});
}
});
thread.start();
}
void genTone(){
// fill out the array
for (int i = 0; i < numSamples; ++i) {
sample[i] = Math.sin(2 * Math.PI * i / (sampleRate/freqOfTone));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
for (final double dVal : sample) {
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
}
void playSound(){
final AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, generatedSnd.length,
AudioTrack.MODE_STATIC);
audioTrack.write(generatedSnd, 0, generatedSnd.length);
audioTrack.play();
}
}
Fmax = samplerate / 2, so it seems correct. Fmax is approx 22khz. –
Frigidarium 0.0 to 1.0. Multiply by 32767 would convert it into the 16-bit fixed point range. The AudioTrack expects the buffer to be little endian format. Hence the next two line just converts the byte order from big endian into little endian. –
Fornof audioTrack.setLoopPoints(0, generatedSnd.length / 4, -1); –
Clavicle Improving on the above code:
Add amplitude ramp up and ramp down to avoid the clicks.
Add code to determine when the tack has finished playing.
double duration = 1; // seconds
double freqOfTone = 1000; // hz
int sampleRate = 8000; // a number
double dnumSamples = duration * sampleRate;
dnumSamples = Math.ceil(dnumSamples);
int numSamples = (int) dnumSamples;
double sample[] = new double[numSamples];
byte generatedSnd[] = new byte[2 * numSamples];
for (int i = 0; i < numSamples; ++i) { // Fill the sample array
sample[i] = Math.sin(freqOfTone * 2 * Math.PI * i / (sampleRate));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalized.
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
int i = 0 ;
int ramp = numSamples / 20 ; // Amplitude ramp as a percent of sample count
for (i = 0; i< ramp; ++i) { // Ramp amplitude up (to avoid clicks)
double dVal = sample[i];
// Ramp up to maximum
final short val = (short) ((dVal * 32767 * i/ramp));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
for (i = i; i< numSamples - ramp; ++i) { // Max amplitude for most of the samples
double dVal = sample[i];
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
for (i = i; i< numSamples; ++i) { // Ramp amplitude down
double dVal = sample[i];
// Ramp down to zero
final short val = (short) ((dVal * 32767 * (numSamples-i)/ramp ));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
AudioTrack audioTrack = null; // Get audio track
try {
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, (int)numSamples*2,
AudioTrack.MODE_STATIC);
audioTrack.write(generatedSnd, 0, generatedSnd.length); // Load the track
audioTrack.play(); // Play the track
}
catch (Exception e){
RunTimeError("Error: " + e);
return false;
}
int x =0;
do{ // Monitor playback to find when done
if (audioTrack != null)
x = audioTrack.getPlaybackHeadPosition();
else
x = numSamples;
} while (x<numSamples);
if (audioTrack != null) audioTrack.release(); // Track play done. Release track.
I wrapped the above wonderful solutions into a neat little package that's more useable out of the box as a simple configurable buzzer. It runs it in a background thread and has stop and play methods and a handful of options you can set.
It's up on JCenter so you can add it to your dependencies list like this
compile 'net.mabboud:android-tone-player:0.2'
and you use it like this for a continuous buzzer
ContinuousBuzzer tonePlayer = new ContinuousBuzzer();
tonePlayer.play();
// just an example don't actually use Thread.sleep in your app
Thread.sleep(1000);
tonePlayer.stop();
or a buzzer played only once and you can set frequency and volume like this
OneTimeBuzzer buzzer = new OneTimeBuzzer();
buzzer.setDuration(5);
// volume values are from 0-100
buzzer.setVolume(50);
buzzer.setToneFreqInHz(110);
Since there is a bug in some older android versions that causes a memory leak when using MODE_STATIC, I modified Xarph's answer above to use MODE_STREAM. Hopefully it will help some.
public void playTone(double freqOfTone, double duration) {
//double duration = 1000; // seconds
// double freqOfTone = 1000; // hz
int sampleRate = 8000; // a number
double dnumSamples = duration * sampleRate;
dnumSamples = Math.ceil(dnumSamples);
int numSamples = (int) dnumSamples;
double sample[] = new double[numSamples];
byte generatedSnd[] = new byte[2 * numSamples];
for (int i = 0; i < numSamples; ++i) { // Fill the sample array
sample[i] = Math.sin(freqOfTone * 2 * Math.PI * i / (sampleRate));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalized.
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
int i = 0 ;
int ramp = numSamples / 20 ; // Amplitude ramp as a percent of sample count
for (i = 0; i< ramp; ++i) { // Ramp amplitude up (to avoid clicks)
double dVal = sample[i];
// Ramp up to maximum
final short val = (short) ((dVal * 32767 * i/ramp));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
for (i = i; i< numSamples - ramp; ++i) { // Max amplitude for most of the samples
double dVal = sample[i];
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
for (i = i; i< numSamples; ++i) { // Ramp amplitude down
double dVal = sample[i];
// Ramp down to zero
final short val = (short) ((dVal * 32767 * (numSamples-i)/ramp ));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
AudioTrack audioTrack = null; // Get audio track
try {
int bufferSize = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, bufferSize,
AudioTrack.MODE_STREAM);
audioTrack.play(); // Play the track
audioTrack.write(generatedSnd, 0, generatedSnd.length); // Load the track
}
catch (Exception e){
}
if (audioTrack != null) audioTrack.release(); // Track play done. Release track.
}
Here's another blog demoing a simple synth plus some UI
You might also be interested in csound or pdlib (pure data lib) for android.
Modified Code Based on Singhaks' answer
public class MainActivity extends Activity {
private final int duration = 30; // seconds
private final int sampleRate = 8000;
private final int numSamples = duration * sampleRate;
private final double sample[] = new double[numSamples];
private final double freqOfTone = 440; // hz
private final byte generatedSnd[] = new byte[2 * numSamples];
Handler handler = new Handler();
private AudioTrack audioTrack;
private boolean play = false;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
8000, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, numSamples,
AudioTrack.MODE_STREAM);
}
@Override
protected void onResume() {
super.onResume();
// Use a new tread as this can take a while
Thread thread = new Thread(new Runnable() {
public void run() {
handler.post(new Runnable() {
public void run() {
playSound();
genTone();
}
});
}
});
thread.start();
}
void genTone(){
// fill out the array
while(play){
for (int i = 0; i < numSamples; ++i) {
// float angular_frequency =
sample[i] = Math.sin(2 * Math.PI * i / (sampleRate/freqOfTone));
}
int idx = 0;
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
for (double dVal : sample) {
short val = (short) (dVal * 32767);
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
audioTrack.write(generatedSnd, 0, numSamples);
}
}
void playSound(){
play = true;
audioTrack.play();
}
}
Do major (16 notes)
public class MainActivity extends AppCompatActivity {
private double mInterval = 0.125;
private int mSampleRate = 8000;
private byte[] generatedSnd;
private final double mStandardFreq = 440;
Handler handler = new Handler();
private AudioTrack audioTrack;
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
}
@Override
protected void onResume() {
super.onResume();
// Use a new tread as this can take a while
final Thread thread = new Thread(new Runnable() {
public void run() {
byte[] tempByte = new byte[0];
for (int i = 0; i < 16 ; i++ ){
double note = getNoteFrequencies(i);
byte[] tonByteNote = getTone(mInterval, mSampleRate, note);
tempByte = concat(tonByteNote, tempByte);
}
generatedSnd = tempByte;
handler.post(new Runnable() {
public void run() {
playTrack(generatedSnd);
}
});
}
});
thread.start();
}
public byte[] concat(byte[] a, byte[] b) {
int aLen = a.length;
int bLen = b.length;
byte[] c= new byte[aLen+bLen];
System.arraycopy(a, 0, c, 0, aLen);
System.arraycopy(b, 0, c, aLen, bLen);
return c;
}
private double getNoteFrequencies(int index){
return mStandardFreq * Math.pow(2, (double) index/12.0d);
}
private byte[] getTone(double duration, int rate, double frequencies){
int maxLength = (int)(duration * rate);
byte generatedTone[] = new byte[2 * maxLength];
double[] sample = new double[maxLength];
int idx = 0;
for (int x = 0; x < maxLength; x++){
sample[x] = sine(x, frequencies / rate);
}
for (final double dVal : sample) {
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedTone[idx++] = (byte) (val & 0x00ff);
generatedTone[idx++] = (byte) ((val & 0xff00) >>> 8);
}
return generatedTone;
}
private AudioTrack getAudioTrack(int length){
if (audioTrack == null)
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
mSampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, length,
AudioTrack.MODE_STATIC);
return audioTrack;
}
private double sine(int x, double frequencies){
return Math.sin( 2*Math.PI * x * frequencies);
}
void playTrack(byte[] generatedSnd){
getAudioTrack(generatedSnd.length)
.write(generatedSnd, 0, generatedSnd.length);
audioTrack.play();
}
}
see this helpful library
https://github.com/karlotoy/perfectTune
it's easy to use
add this to your dependencies
compile 'com.github.karlotoy:perfectTune:1.0.2'
And you use it like this:
PerfectTune perfectTune = new PerfectTune();
perfectTune.setTuneFreq(desire_freq);
perfectTune.playTune();
to stop the tune:
perfectTune.stopTune();
There are several programs for this, but they suck. I measured a few:
http://www.endolith.com/wordpress/2009/11/24/android-audio-applications/
So don't do whatever they do. :D
float synth_frequency = 440;
int minSize = AudioTrack.getMinBufferSize(SAMPLE_RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT);
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
SAMPLE_RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
minSize,
AudioTrack.MODE_STREAM);
audioTrack.play();
short[] buffer = new short[minSize];
float angle = 0;
while (true)
{
if (play)
{
for (int i = 0; i < buffer.length; i++)
{
float angular_frequency =
(float)(2*Math.PI) * synth_frequency / SAMPLE_RATE;
buffer[i] = (short)(Short.MAX_VALUE * ((float) Math.sin(angle)));
angle += angular_frequency;
}
audioTrack.write(buffer, 0, buffer.length);
}
// You can add arbitrary value in synth_frequency to get change sound for example you can add random variable to get sound
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