Existe-t-il un moyen de faire Android émettre un son de fréquence arbitraire (c’est-à-dire que je ne veux pas de fichiers son préenregistrés)?
J'ai regardé autour de moi et ToneGenerator était la seule chose que je pouvais trouver qui soit même proche, mais il semble être uniquement capable de restituer les tonalités DTMF standard.
Des idées?
À l'origine, j'ai trouvé cet exemple de code sur un blog, mais il contenait des bugs générant des sons épouvantables. J'ai corrigé les bugs et posté le code résultant ici. Semble bien fonctionner pour moi!
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();
}
}
Améliorer sur le code ci-dessus:
Ajoutez de l'amplitude, augmentez et réduisez progressivement pour éviter les clics.
Ajoutez du code pour déterminer quand le virement de bord est terminé.
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.
J'ai emballé les merveilleuses solutions ci-dessus dans un petit paquet ordonné qui est plus facilement utilisable comme un simple buzzer configurable. Il l'exécute dans un thread d'arrière-plan et dispose de méthodes d'arrêt et de lecture et d'une poignée d'options que vous pouvez définir.
C'est sur JCenter que vous pouvez l'ajouter à votre liste de dépendances comme ceci
compile 'net.mabboud:Android-tone-player:0.2'
et vous l'utilisez comme ceci pour une sonnerie continue
ContinuousBuzzer tonePlayer = new ContinuousBuzzer();
tonePlayer.play();
// just an example don't actually use Thread.sleep in your app
Thread.sleep(1000);
tonePlayer.stop();
ou un buzzer joué qu'une seule fois et vous pouvez régler la fréquence et le volume comme ceci
OneTimeBuzzer buzzer = new OneTimeBuzzer();
buzzer.setDuration(5);
// volume values are from 0-100
buzzer.setVolume(50);
buzzer.setToneFreqInHz(110);
Puisqu'il existe un bogue dans certaines versions plus anciennes Android) qui provoque une fuite de mémoire lors de l'utilisation de MODE_STATIC, j'ai modifié la réponse de Xarph pour utiliser MODE_STREAM. Espérons que cela en aidera quelques-uns.
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.
}
Code modifié basé sur la réponse de Singhaks
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();
}
}
Voici un autre blog montrant un synthé simple et une interface utilisateur
Csound ou pdlib (librairie de données pure) pour Android pourraient également vous intéresser.
voir cette bibliothèque utile
https://github.com/karlotoy/perfectTune
c'est facile à utiliser
ajoutez ceci à vos dépendances
compile 'com.github.karlotoy:perfectTune:1.0.2'
Et vous l'utilisez comme ceci:
PerfectTune perfectTune = new PerfectTune();
perfectTune.setTuneFreq(desire_freq);
perfectTune.playTune();
pour arrêter la mélodie:
perfectTune.stopTune();
Do majeur (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();
}
}
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);
}
// Vous pouvez ajouter une valeur arbitraire dans synth_frequency pour obtenir un changement de son. Par exemple, vous pouvez ajouter une variable aléatoire pour obtenir un son.
Il existe plusieurs programmes pour cela, mais ils sont nuls. J'ai mesuré quelques-uns:
http://www.endolith.com/wordpress/2009/11/24/Android-audio-applications/
Alors ne faites pas ce qu'ils font. :RÉ