有没有办法让 Android 发出任意频率的声音（也就是说，我不想使用预先录制的声音文件）？
我查了一下，ToneGenerator 是我能找到的唯一接近的东西，但它似乎只能输出标准的 DTMF 音调。
有什么想法吗？
10个回答
117
我最初在博客上找到了这个示例代码，但它存在一些错误，会产生一些可怕的声音。我已经修复了这些错误并在此发布了修复后的代码。对我来说似乎很有效！
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 <steve@staticfree.info>
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();
}
}
- Steve Pomeroy
24
2这行代码正确吗？audioTrack.write(generatedSnd, 0, numSamples); 还是应该是numSamples * 2，因为每个样本有2个字节。另外，write方法还接受一个shorts数组，那么创建一个中间字节数组的优点是什么？ - Damian Kennedy
2这确实是一个很好的例子，非常感谢。但是我发现另一个讨厌的 bug（如果你扩展这段代码），即：audioTrack.write(generatedSnd, 0, numSamples) 应该是 audioTrack.write(generatedSnd, 0, 2*numSamples) 或者更好的 audioTrack.write(generatedSnd, 0, generatedSnd.length); - AudioDroid
6在 AudioTrack 构造函数中，应使用 generatedSnd.length 而非 numSamples，因为第五个参数是“缓冲区大小（以字节为单位）”。该示例仅播放音调的前半部分。 - Torben
5这段内容的意思是：样本使用浮点数创建，幅度范围从
0.0到1.0。将其乘以32767将其转换为16位固定点范围。AudioTrack期望缓冲区采用小端格式。因此下面的两行代码只是将字节顺序从大端变成了小端。 - ains2使用私有静态常量int sampleRate = 192000;，我能够播放超声波。 - DataYoda
显示剩余19条评论
32
改进上述代码：
添加振幅逐渐增加和减小的代码以避免噪音。
添加代码以确定曲目何时播放结束。
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.
- Xarph
6
1主要的改变是振幅的逐渐增加和减小。原始代码在开始和结束时都是最大振幅，这会在音调开头和结尾产生噪声。新代码在前20%的样本中将振幅从0逐渐增加到最大振幅，然后在最后20%的样本中将振幅从最大值逐渐降低到0。这样音调更加平滑，更加愉悦。另一个改变是监控音调的播放，并在音调播放完成之前不继续进行。 - Xarph
我无法让它运行。我能够运行第一个，但实际上不知道如何修改为您所做的那样。如果能帮忙解决这个问题将非常有帮助，因为我想消除点击声。 - Coder
3+1，但是这个答案中的代码根本无法编译。我在这里正确地实现了它：https://gist.github.com/SuspendedPhan/7596139
只需用我的genTone()方法替换Steve的即可获得逐渐变化的效果。 - Dylan P
由于MODE_STATIC存在内存泄漏问题，我修改了代码以在下面使用MODE_STREAM。 - extreme
2从API开始，可以使用setVolume()来进行坡道。这使得仅循环一个非常小的样本甚至播放动态长度的声音成为可能（例如，当用户按住按钮时）。代码示例：https://github.com/stefanhaustein/android-tone-generator/blob/master/core/src/main/java/org/kobjects/atg/ToneGenerator.java - Stefan Haustein
我使用了这段代码，它在声音的开头进行了斜坡处理，但在结尾仍然听到了点击声...有人能帮我解决吗？ - ArMot
9
我将上述出色的解决方案打包成了一个简单易用的可配置蜂鸣器。它在后台线程中运行，并具有停止和播放方法以及一些可以设置的选项。
它已经上传到JCenter，所以你可以像这样将其添加到依赖列表中：
compile 'net.mabboud:android-tone-player:0.2'
对于连续蜂鸣器，您可以像这样使用它
ContinuousBuzzer tonePlayer = new ContinuousBuzzer();
tonePlayer.play();
// just an example don't actually use Thread.sleep in your app
Thread.sleep(1000);
tonePlayer.stop();
或者是仅播放一次的蜂鸣器，您可以像这样设置频率和音量
OneTimeBuzzer buzzer = new OneTimeBuzzer();
buzzer.setDuration(5);
// volume values are from 0-100
buzzer.setVolume(50);
buzzer.setToneFreqInHz(110);
- meese
1
@Melchester，现在已经修复了。感谢您的提醒，对此我们深表歉意。 - meese
5
由于一些旧版的Android存在使用MODE_STATIC时会导致内存泄漏的bug，因此我修改了Xarph上面的答案，改用MODE_STREAM。希望这能对一些人有所帮助。
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.
}
- extreme
3
进行大调（16个音符）
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();
}
}
- Vahe Gharibyan
3
基于Singhaks的答案修改的代码
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();
}
}
- Larry Lo
3
这是另一个演示简单合成器和一些UI的博客：
您可能还对适用于Android的csound或pdlib（纯数据库）感兴趣。
- simou
3
看看这个有用的库
https://github.com/karlotoy/perfectTune
它很容易使用
将此添加到您的依赖项中
compile 'com.github.karlotoy:perfectTune:1.0.2'
你可以像这样使用它：
PerfectTune perfectTune = new PerfectTune();
perfectTune.setTuneFreq(desire_freq);
perfectTune.playTune();
停止音乐：
perfectTune.stopTune();
- shinta
2
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);
}
// 你可以在synth_frequency中添加任意值来改变声音，例如，你可以添加随机变量来获取声音。
- Singhak
1
最终你将所有内容都转换为short。没有理由将角度作为浮点数处理。使用double数学运算速度相同，而且不需要大量的强制类型转换。 - Tatarize
2
有几个程序可以做到这一点，但它们很差。我测试了几个：
http://www.endolith.com/wordpress/2009/11/24/android-audio-applications/
所以不要做他们所做的。 :D- endolith
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