简介
库名称：AudioChannel
版本:1.0
由于项目需求录音并base64编码存到服务器中，就顺手改装了一个别人的封装类
原封装类地址:Android AudioRecord音频录制wav文件输出 - 简书 (jianshu.com)
描述：此封装类基于AudioRecord实现wav的音频录制，本封装类对原版进行了以下修改:
1.部分修正
(1).可以看到，原封装类继承Thread，代码逻辑很清晰，因此改动过程也较轻松，单次运行能够正常，但是在二次运行，发现报错:
D/CompatibilityChangeReporter: Compat change id reported: 147798919; UID 10428; state: ENABLED
W/System.err: java.lang.IllegalThreadStateException
W/System.err: at java.lang.Thread.start(Thread.java:960)
at com.yy.audiochannaldemo.AudioChannel.startLive(AudioChannel.java:84)经过跟踪发现，在二次运行的时候，线程的state变为TERMINATED，这意味着线程已经完成了它的执行并且已经退出。一旦线程终止，不能重新启动，因此新版封装类不再继承Thread，而是通过priavate线程重建函数initThread来实现。
(2).首先AudioRecord不能够直接保存录音为wav，因此必须先保存为pcm文件，再通过头部写入数据，转换为wav文件，在这个过程中注意到原封装库，没有对保存pcm的文件进行删除处理，后续可能导致容量过大
(3).构建函数，传入context，以此就无需动态授权外部存储写入权限，也方便后续需要context的操作部分
2.权限控制
在使用过程注意到，原版库并没有处理权限申请，在改版加上了，6.0以上安卓加入了权限控制，另外去除了使用外部存储，需要额外动态授权的情况，直接存入cache
3.功能实现
在原先基础上加入了音高、声音贝计算，并通过onResult接口回调这三个变量，不过db和hz都有一定偏差
一、配置部分
需要先在清单中加入这两项:
<uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />此封装库需要配置的部分就这么多。
需要在build.gradle加入以下依赖添加代码
implementation 'com.github.wendykierp:JTransforms:3.1'二、代码部分
1.PcmToWavUtil.java:Pcm转Wav工具类
public class PcmToWavUtil {
private int mBufferSize; //缓存的音频大小
private int mSampleRate = 8000;// 8000|16000
private int mChannelConfig = AudioFormat.CHANNEL_IN_STEREO; //立体声
private int mChannelCount = 2;
private int mEncoding = AudioFormat.ENCODING_PCM_16BIT;
public PcmToWavUtil() {
this.mBufferSize = AudioRecord.getMinBufferSize(mSampleRate, mChannelConfig, mEncoding);
}
public PcmToWavUtil(int sampleRate, int channelConfig, int channelCount, int encoding) {
this.mSampleRate = sampleRate;
this.mChannelConfig = channelConfig;
this.mChannelCount = channelCount;
this.mEncoding = encoding;
this.mBufferSize = AudioRecord.getMinBufferSize(mSampleRate, mChannelConfig, mEncoding);
}
public void pcmToWav(String inFilename, String outFilename) {
FileInputStream in;
FileOutputStream out;
long totalAudioLen;
long totalDataLen;
long longSampleRate = mSampleRate;
int channels = mChannelCount;
long byteRate = 16 * mSampleRate * channels / 8;
byte[] data = new byte[mBufferSize];
try {
in = new FileInputStream(inFilename);
out = new FileOutputStream(outFilename);
totalAudioLen = in.getChannel().size();
totalDataLen = totalAudioLen + 36;//44-8(RIFF+dadasize(4个字节))
writeWaveFileHeader(out, totalAudioLen, totalDataLen,
longSampleRate, channels, byteRate);
while (in.read(data) != -1) {
out.write(data);
}
in.close();
out.close();
} catch (IOException e) {
e.printStackTrace();
}
}
/**
* 加入wav文件头
*/
private void writeWaveFileHeader(FileOutputStream out, long totalAudioLen,
long totalDataLen, long longSampleRate, int channels, long byteRate)
throws IOException {
byte[] header = new byte[44];
header[0] = 'R'; // RIFF/WAVE header
header[1] = 'I';
header[2] = 'F';
header[3] = 'F';
header[4] = (byte) (totalDataLen & 0xff);
header[5] = (byte) ((totalDataLen >> 8) & 0xff);
header[6] = (byte) ((totalDataLen >> 16) & 0xff);
header[7] = (byte) ((totalDataLen >> 24) & 0xff);
header[8] = 'W'; //WAVE
header[9] = 'A';
header[10] = 'V';
header[11] = 'E';
header[12] = 'f'; // 'fmt ' chunk
header[13] = 'm';
header[14] = 't';
header[15] = ' ';
header[16] = 16; // 4 bytes: size of 'fmt ' chunk
header[17] = 0;
header[18] = 0;
header[19] = 0;
header[20] = 1; // format = 1
header[21] = 0;
header[22] = (byte) channels;
header[23] = 0;
header[24] = (byte) (longSampleRate & 0xff);
header[25] = (byte) ((longSampleRate >> 8) & 0xff);
header[26] = (byte) ((longSampleRate >> 16) & 0xff);
header[27] = (byte) ((longSampleRate >> 24) & 0xff);
header[28] = (byte) (byteRate & 0xff);
header[29] = (byte) ((byteRate >> 8) & 0xff);
header[30] = (byte) ((byteRate >> 16) & 0xff);
header[31] = (byte) ((byteRate >> 24) & 0xff);
header[32] = (byte) (2 * 16 / 8); // block align
header[33] = 0;
header[34] = 16; // bits per sample
header[35] = 0;
header[36] = 'd'; //data
header[37] = 'a';
header[38] = 't';
header[39] = 'a';
header[40] = (byte) (totalAudioLen & 0xff);
header[41] = (byte) ((totalAudioLen >> 8) & 0xff);
header[42] = (byte) ((totalAudioLen >> 16) & 0xff);
header[43] = (byte) ((totalAudioLen >> 24) & 0xff);
out.write(header, 0, 44);
}
}
2.AudioChannel.java:录音封装类主体
package com.yy.audiochannaldemo;
import android.Manifest;
import android.app.Activity;
import android.content.Context;
import android.content.pm.PackageManager;
import android.media.AudioFormat;
import android.media.AudioRecord;
import android.media.MediaRecorder;
import android.util.Log;
import android.widget.Toast;
import androidx.core.app.ActivityCompat;
import androidx.core.content.ContextCompat;
import org.jtransforms.fft.DoubleFFT_1D;
import java.io.File;
import java.io.FileOutputStream;
import java.io.IOException;
import java.text.SimpleDateFormat;
import java.util.Date;
public class AudioChannel {
private int sampleRate;
private int channelConfig;
private int minBufferSize;
private byte[] buffer;
private Thread recordThread;
private AudioRecord audioRecord;
private boolean isRecoding;
private SimpleDateFormat sdf;
String filename;
Context context;
long startTime;
private onResult onResult;
private DoubleFFT_1D fft;
public AudioChannel(int sampleRate, int channels, Context context) {
this.sampleRate = sampleRate;
this.context = context;
channelConfig = channels == 2 ? AudioFormat.CHANNEL_IN_STEREO : AudioFormat.CHANNEL_IN_MONO;
minBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
Log.i("AudioChannel", "minBufferSize: " + minBufferSize);
buffer = new byte[minBufferSize];
sdf = new SimpleDateFormat("yyyy-MM-dd-HH:mm:ss");
fft = new DoubleFFT_1D(minBufferSize/ 2);
}
private double calculateRMS(byte[] audioBuffer) {
double sum = 0.0;
for (byte sample : audioBuffer) {
sum += sample * sample;
}
return Math.sqrt(sum / audioBuffer.length);
}
// 将RMS值转换为分贝值的方法
private double rmsToDB(double rms) {
// 假设参考值为1（通常是最小可听声音的RMS值）
double reference = 1.0;
return 20 * Math.log10(rms / reference);
}
private double calculateHZ(byte[] buffer) {
// Convert byte array to double array for FFT
double[] fftBuffer = new double[buffer.length / 2];
for (int i = 0; i < buffer.length; i += 2) {
short sample = (short) ((buffer[i] << 8) | (buffer[i + 1] & 0xFF));
fftBuffer[i / 2] = sample;
}
// 执行 FFT
fft.realForward(fftBuffer);
double maxAmplitude = 0.0;
int pitchIndex = 0;
for (int i = 0; i < fftBuffer.length-1; i++) {
double amplitude = fftBuffer[i] * fftBuffer[i] + fftBuffer[i + 1] * fftBuffer[i + 1];
if (i < fftBuffer.length / 2 && amplitude > maxAmplitude) {
maxAmplitude = amplitude;
pitchIndex = i;
}
}
//计算hz，不过偏差比较大
double frequency = (double) pitchIndex * sampleRate / (fftBuffer.length / 2) ;
return frequency /100;
}
void initPremission() {
ActivityCompat.requestPermissions((Activity)context, new String[]{Manifest.permission.RECORD_AUDIO}, 169);
}
void initThread() {
this.recordThread=new Thread(){ //开线程
@Override
public void run() {
audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT, minBufferSize);
audioRecord.startRecording();
FileOutputStream writer = null;
Date current = new Date();
String time = sdf.format(current);
byte[] audioBuffer = new byte[minBufferSize]; // 创建一个缓冲区
try {
filename = context.getCacheDir() + "/" + time + ".pcm"; //cache目录不需要权限
writer = new FileOutputStream(filename, true);
while (!Thread.currentThread().isInterrupted() && isRecoding) { //如果线程没有Interrupted而且isRecording变量为True代表在录制状态的情况
if (audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING) {
audioRecord.read(audioBuffer, 0, minBufferSize); // 读取音频数据到缓冲区
double rms = calculateRMS(audioBuffer);
double db = rmsToDB(rms); //db的值
double hz = calculateHZ(audioBuffer);
int seaconds =(int) (System.currentTimeMillis() -startTime) /1000;
if (isRecoding) {
((Activity)context).runOnUiThread(new Runnable() {
@Override
public void run() {
onResult.update(seaconds,db,hz); //如果还在线程运行状态把信息回调出来
}
});
}
writer.write(audioBuffer);
}
}
} catch (IOException e) {
e.printStackTrace();
} finally {
audioRecord.stop();
audioRecord.release();
audioRecord = null;
try {
writer.close();
} catch (IOException e) {
e.printStackTrace();
}
}
new PcmToWavUtil(44100, AudioFormat.CHANNEL_IN_STEREO, 2, AudioFormat.ENCODING_PCM_16BIT).pcmToWav(filename, filename.replace("pcm","wav"));
}
};
}
public void startLive() { //录制
initThread();
initPremission();
if (ContextCompat.checkSelfPermission(context, Manifest.permission.RECORD_AUDIO) ==
PackageManager.PERMISSION_GRANTED) {
isRecoding = true;
recordThread.start();
startTime = System.currentTimeMillis();
} else {
Toast.makeText(context,"没有录音权限",Toast.LENGTH_LONG).show();
}
}
public void stopLive(int mode) { //mode为-1时代表取消，为0代表取消
if (!isRecoding) return;
try {
isRecoding = false;
recordThread.join();
} catch (Exception e){
isRecoding = false;
e.printStackTrace();
}
new File(filename).delete();
switch (mode) {
case 0:
onResult.finish(filename.replace("pcm","wav")); //正常结束后pcm会被转换成wav
break;
case -1:
new File(filename.replace("pcm","wav")).delete();
onResult.cancel(); //取消回调
break;
}
}
public interface onResult { //三个回调
void update(int seaconds,double db,double hz);
void finish(String filename);
void cancel();
}
public void onResult(onResult onResult) { //功能点击
this.onResult = onResult;
}
}三.Demo部分
Demo下载地址:
gitee地址:
AudioChannel/demo · keyxh/AndroidUtils - 码云 - 开源中国 (gitee.com)
csdn地址:【免费】安卓开发:挑战每天发布一个封装类02-Wav录音封装类AudioChannel1.0资源资源-CSDN文库
在Demo中有两个Actvity
1.MainActvity：简易demo，示范调用
MainActvity的案例是普通调用，调用过程会将参数打印出来，结束时会将音频转base64，界面和logcat如下图所示，MainActvity的demo是没有任何交互
第二个demo:PPActvity（音高测试仪）
本来想做音高测试仪的，后来音高频率转换(例如440HZ转A4)没有整出来，后面有空再修改投放gitee，目前最终效果如下:
由福州职业技术学校温辉编写，欢迎搬运帮助更多人，但请带上以上这句。
