【发布时间】：2017-08-23 03:05:24
【问题描述】：
注意：我正在使用 android studio，目前我正在使用最新的tarsos audio library，它应该与 android 兼容，实际上已经成功地将库添加到我的 android studio 项目中。我之前尝试过使用 JTransforms 和 Minim 库，但没有成功。 2017 年 8 月 23 日编辑：发现并修复了一些错误，重新发布了当前代码，但仍然没有任何进展，实际问题总结如下：
总结：在我发布的第 5 个代码块中，在被注释掉的第 15 行，我需要知道如何让该行工作而不抛出编译错误
我要做的是从麦克风录制，并在录制时使用 tarsos 库中的 dsp BandPass 过滤器并将结果输出到 .wav 文件。我可以通过使用 android.media 导入成功地将麦克风流式传输到this tutorial 之后的 .wav 文件，但这不允许我添加带通过滤器，并且使用 tarsos 导入函数不允许我使用另存为that tutorial has 的.wav 方法，我知道我遗漏了一些东西和/或做错了什么，但我已经用谷歌搜索了将近一个星期，还没有找到可行的解决方案，我只找到了链接到库中的 java 文件，这没有帮助，因为我找不到有关如何正确使用它们的教程。我究竟做错了什么？这是我尝试使用的 tarsos 方法的相关代码：
相关的导入和“全局”变量
import android.media.AudioRecord;
import android.media.MediaRecorder;
import android.media.AudioFormat;
import android.media.AudioTrack;
import be.tarsos.dsp.AudioDispatcher;
import be.tarsos.dsp.AudioProcessor;
import be.tarsos.dsp.filters.BandPass;
import be.tarsos.dsp.io.android.AudioDispatcherFactory;
//start the class
AudioRecord alteredRecord = null;
AudioDispatcher dispatcher;
float freqChange;
float tollerance;
private static final int RECORDER_BPP = 16;
private static final String AUDIO_RECORDER_FOLDER = "Crowd_Speech";
private static final String AUDIO_RECORDER_TEMP_FILE = "record_temp.raw";
private static final int RECORDER_SAMPLERATE = 44100;
private static final int RECORDER_CHANNELS = AudioFormat.CHANNEL_IN_MONO;
private static final int RECORDER_AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT;
private int bufferSize = 1024;
private Thread recordingThread = null;
//set the min buffer size in onCreate event
bufferSize = AudioRecord.getMinBufferSize(RECORDER_SAMPLERATE,
RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING)*4;
这会在 onClick 方法中启动麦克风录音，通过注释/取消注释 2 个“运行”变量值之一，我可以在调用 startRecording 方法时在过滤器或无过滤器（android 或 tarsos 函数）之间切换
if(crowdFilter && running==0 && set==0){//crowd speech mode, start talking
Icons(2,"");
running=4;//start recording from mic, apply bandpass filter and save as wave file using TARSOS import
//running=5;//start recording from mic, no filter, save as wav file using android media import
freqChange = Globals.minFr[Globals.curUser];
tollerance = 40;
set=1;
startRecording();
}
开始录音方式：
private void startRecording() {
if (running == 5) {//start recording from mic, no filter, save as wav file using android media library
alteredRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, RECORDER_SAMPLERATE, RECORDER_CHANNELS,RECORDER_AUDIO_ENCODING, bufferSize);
alteredRecord.startRecording();
isRecording = true;
recordingThread = new Thread(new Runnable() {
@Override
public void run() {
writeAudioDataToFile();
}
}, "Crowd_Speech Thread");
recordingThread.start();
}
if (running == 4) {//start recording from mic, apply bandpass filter and save as wave file using TARSOS library
dispatcher = AudioDispatcherFactory.fromDefaultMicrophone(RECORDER_SAMPLERATE, bufferSize, 0);
AudioProcessor p = new BandPass(freqChange, tollerance, RECORDER_SAMPLERATE);
dispatcher.addAudioProcessor(p);
isRecording = true;
dispatcher.run();
recordingThread = new Thread(new Runnable() {
@Override
public void run() {
writeAudioDataToFile();
}
}, "Crowd_Speech Thread");
recordingThread.start();
}
}
onClick 方法中的停止录制按钮
if(crowdFilter && (running==4 || running==5) && set==0) {//crowd speech finished talking
Icons(1, "");
stopRecording();
set = 1;
}
到目前为止，这两种情况都很好，如果 running==4（应用了 tarsos dsp 过滤器）程序会崩溃。如果我使用 running==5 （没有过滤器的 android.media 方式），其余的工作正常并保存文件，但没有应用带通效果。如果我尝试用 tarsos dispatcher = AudioDispatcherFactory... （例如 dispatcher = new AudioRecord...）交换 changedRecord = new AudioRecord... ，它们是不兼容的，甚至不会考虑编译。 （这就是为什么下面方法中的第15行被注释掉了）
private void writeAudioDataToFile(){
byte data[] = new byte[bufferSize];
String filename = getTempFilename();
FileOutputStream os = null;
try {
os = new FileOutputStream(filename);
} catch (FileNotFoundException e) {
e.printStackTrace();
}
int read = 0;
if(null != os){
while(isRecording){
if(running==4)
{
//read = dispatcher.(data, 0, bufferSize);
}
if(running==5)
{
read = alteredRecord.read(data, 0, bufferSize);
}
if(AudioRecord.ERROR_INVALID_OPERATION != read){
try {
os.write(data);
} catch (IOException e) {
e.printStackTrace();
}
}
}
try {
os.close();
} catch (IOException e) {
e.printStackTrace();
}
}
}
private void stopRecording(){
if(null != alteredRecord) {
isRecording = false;
int i = alteredRecord.getState();
if (i == 1) {
running = 0;
alteredRecord.stop();
alteredRecord.release();
alteredRecord = null;
recordingThread = null;
}
}
if(null !=dispatcher){
isRecording = false;
running = 0;
dispatcher.stop();
recordingThread = null;
}
copyWaveFile(getTempFilename(),getFilename());
deleteTempFile();
}
private void deleteTempFile() {
File file = new File(getTempFilename());
file.delete();
}
private void copyWaveFile(String inFilename,String outFilename){
FileInputStream in = null;
FileOutputStream out = null;
long totalAudioLen = 0;
long totalDataLen = totalAudioLen + 36;
long longSampleRate = RECORDER_SAMPLERATE;
int channels = 1;
long byteRate = RECORDER_BPP * RECORDER_SAMPLERATE * channels/8;
byte[] data = new byte[bufferSize];
try {
in = new FileInputStream(inFilename);
out = new FileOutputStream(outFilename);
totalAudioLen = in.getChannel().size();
totalDataLen = totalAudioLen + 36;
WriteWaveFileHeader(out, totalAudioLen, totalDataLen,
longSampleRate, channels, byteRate);
while(in.read(data) != -1){
out.write(data);
}
in.close();
out.close();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
}
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';header[1] = 'I'; header[2] = 'F';header[3] = 'F';// RIFF/WAVE header
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';header[9] = 'A';header[10] = 'V';header[11] = 'E';header[12] = 'f';header[13] = 'm';header[14] = 't';header[15] = ' ';// 'fmt ' chunk
header[16] = 16;header[17] = 0;header[18] = 0;header[19] = 0;// 4 bytes: size of 'fmt ' chunk
header[20] = 1;header[21] = 0;header[22] = (byte) channels;header[23] = 0;// format = 1
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);header[33] = 0;// block align
header[34] = RECORDER_BPP;header[35] = 0;header[36] = 'd';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);// bits per sample
out.write(header, 0, 44);
}
【问题讨论】：
标签： java android audio streaming tarsosdsp