Java 线不到头
我正在开发的android应用程序使用AudioTrack播放音符。这将创建一个新的线程,在我的应用程序崩溃之前,它不会为每个被按下的音符结束 来自A类:Java 线不到头,java,android,multithreading,Java,Android,Multithreading,我正在开发的android应用程序使用AudioTrack播放音符。这将创建一个新的线程,在我的应用程序崩溃之前,它不会为每个被按下的音符结束 来自A类: AudioGenerator.playSound(AudioGenerator.genTone((tone))); 音频发生器等级: import android.media.AudioFormat; import android.media.AudioManager; import android.media.AudioTrack; p
AudioGenerator.playSound(AudioGenerator.genTone((tone)));
音频发生器等级:
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
public class AudioGenerator {
private static final int sampleRate = 8000;
private static final double duration = 1; // seconds
private static double dnumSamples = duration * sampleRate;
private static final int numSamples = (int) dnumSamples;
private static double sample[] = new double[numSamples];
private static byte generatedSnd[] = new byte[2*numSamples];
static void playSound(byte[] sound){
final byte[] play = sound;
(new Thread(new Runnable() {
@Override
public void run() {
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, 8000, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, numSamples, AudioTrack.MODE_STATIC);
audioTrack.write(play, 0, play.length);
audioTrack.play();
}})).start();
}
static byte[] genTone(double freq){
for (int i = 0; i < numSamples; ++i) sample[i] = Math.sin(freq * 2 * Math.PI * i / (sampleRate));
// 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);
}
return generatedSnd;
}
}
我不知道我错过了什么。请帮助尝试修改playTone和playSamples方法:
private void playTone(final int freq, final double duration)
{
Thread thread = new Thread(new Runnable()
{
public void run()
{
genTone(freq, duration);
}
});
thread.start();
}
private void playSamples(byte[] generatedSound, int sampleRate, int numSamples)
{
AudioTrack audioTrack = null;
try
{
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT,
(int) numSamples * 2, AudioTrack.MODE_STATIC);
// Load the track
audioTrack.write(generatedSound, 0, generatedSound.length);
audioTrack.play();
}
catch (Exception e)
{
Log.e(MainActivity.DEBUG_TAG, "Audio track exception: " + e.getMessage());
}
int x = 0;
do
{ // Montior playback to find when done
if (audioTrack != null)
x = audioTrack.getPlaybackHeadPosition();
else
x = numSamples;
}
while (x < numSamples);
audioTrack.release();
}
最后,添加playSamplesgeneratedSnd、sampleRate、numSamples;
直到你的genTone方法结束 嗯,它会在每次播放声音时创建一个新线程,因为每次播放声音时都会创建一个新线程…是的,但是为什么声音播放后线程不会结束?你怎么知道线程不会结束?当我在eclipse中使用debug和DDMS进行调试时,我每次播放音符时都会看到一个新线程被创建,如果你能解释一下为什么你认为这会解决OP的问题,这会更有用。这很好。它工作得很好。我不明白为什么。我以为这是你打电话给audioTrack.release的那一行,但我把它包括在了我的原稿中,但仍然不起作用。