我正在研究一种将wav文件修剪为用户选择的方法。基本思想是使用Javazoom的WaveFile
类来编写一个wav文件,该文件仅包含用户所做选择中的声音数据。问题是,我编写的代码在一半时间内完美地工作,而另一半时间产生静态。它似乎工作,并不在相同的完全相同的情况下工作。另一个时间由MediaPlayer
加载wav文件,并在其他方法中作为输入流加载。这可能是问题的根源吗?我已经尝试关闭流并释放MediaPlayer
,但仍然遇到同样的问题。
public void TrimToSelection(double startTime, double endTime){ // Time in seconds
InputStream wavStream = null; // InputStream to stream the wav to trim
File trimmedSample = null; // File to contain the trimmed down sample
File sampleFile = new File(samplePath); // File pointer to the current wav sample
// If the sample file exists, try to trim it
if (sampleFile.isFile()){
trimmedSample = new File(Environment.getExternalStoragePublicDirectory(Environment.DIRECTORY_MUSIC), "trimmedSample.wav");
if (trimmedSample.isFile()) trimmedSample.delete(); // Delete if already exists
// Trim the sample down and write it to file
try {
wavStream = new BufferedInputStream(new FileInputStream(sampleFile));
// Javazoom class WaveFile is used to write the wav
WaveFile waveFile = new WaveFile();
waveFile.OpenForWrite(trimmedSample.getAbsolutePath(), (int)audioFormat.getSampleRate(), (short)audioFormat.getSampleSizeInBits(), (short)audioFormat.getChannels());
// The number of bytes of wav data to trim off the beginning
long startOffset = (long)(startTime * audioFormat.getSampleSizeInBits() * audioFormat.getSampleRate() / 4);
// The number of bytes to copy
long length = (long)(endTime * audioFormat.getSampleSizeInBits() * audioFormat.getSampleRate() / 4) - startOffset;
wavStream.skip(44); // Skip the header
wavStream.skip(startOffset);
byte[] buffer = new byte[1024 * 16];
int bufferLength;
for (long i = startOffset; i < length + startOffset; i += buffer.length){
bufferLength = wavStream.read(buffer);
short[] shorts = new short[buffer.length / 2];
ByteBuffer.wrap(buffer).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shorts);
waveFile.WriteData(shorts, shorts.length);
}
waveFile.Close(); // Complete writing the wave file
wavStream.close(); // Close the input stream
} catch (IOException e) {e.printStackTrace();}
finally {
try {if (wavStream != null) wavStream.close();} catch (IOException e){}
}
}
// Delete the original wav sample
sampleFile.delete();
// Copy the trimmed wav over to replace the sample
trimmedSample.renameTo(sampleFile);
}
更新:我将long startOffset = (long)(startTime * audioFormat.getSampleSizeInBits() * audioFormat.getSampleRate() / 4);
更改为long startOffset = ((long)startTime * audioFormat.getSampleSizeInBits() * (long)audioFormat.getSampleRate() / 4);
,同样更改为length
。出于某种原因,改变演员阵容的位置似乎已经解决了静态问题(我认为),尽管我不确定原因。现在,我想我需要调整缓冲区循环,因为样本的结尾会被切断。
答案 0 :(得分:0)
我不确定问题的根源,但这可能有助于您弄明白:RingDroid是一个开源铃声创建者。它内置了一个波形编辑器。我曾经下载它并使用Waveform编辑器(并自定义它)来实现一个有趣的项目。这可以帮助您到达您需要去的地方。请享用。
答案 1 :(得分:0)
我的更新中的问题是startTime
的强制转换有效地向下舍入到最接近的秒。下面的代码有效,但我仍然不确定为什么我的原始投射方法不起作用。
public void TrimToSelection(double startTime, double endTime){
InputStream wavStream = null; // InputStream to stream the wav to trim
File trimmedSample = null; // File to contain the trimmed down sample
File sampleFile = new File(samplePath); // File pointer to the current wav sample
// If the sample file exists, try to trim it
if (sampleFile.isFile()){
trimmedSample = new File(Environment.getExternalStoragePublicDirectory(Environment.DIRECTORY_MUSIC), "trimmedSample.wav");
if (trimmedSample.isFile()) trimmedSample.delete();
// Trim the sample down and write it to file
try {
wavStream = new BufferedInputStream(new FileInputStream(sampleFile));
// Javazoom WaveFile class is used to write the wav
WaveFile waveFile = new WaveFile();
waveFile.OpenForWrite(trimmedSample.getAbsolutePath(), (int)audioFormat.getSampleRate(), (short)audioFormat.getSampleSizeInBits(), (short)audioFormat.getChannels());
// The number of bytes of wav data to trim off the beginning
long startOffset = (long)(startTime * audioFormat.getSampleRate()) * audioFormat.getSampleSizeInBits() / 4;
// The number of bytes to copy
long length = ((long)(endTime * audioFormat.getSampleRate()) * audioFormat.getSampleSizeInBits() / 4) - startOffset;
wavStream.skip(44); // Skip the header
wavStream.skip(startOffset);
byte[] buffer = new byte[1024];
int i = 0;
while (i < length){
if (length - i >= buffer.length) {
wavStream.read(buffer);
}
else { // Write the remaining number of bytes
buffer = new byte[(int)length - i];
wavStream.read(buffer);
}
short[] shorts = new short[buffer.length / 2];
ByteBuffer.wrap(buffer).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shorts);
waveFile.WriteData(shorts, shorts.length);
i += buffer.length;
}
waveFile.Close(); // Complete writing the wave file
wavStream.close(); // Close the input stream
} catch (IOException e) {e.printStackTrace();}
finally {
try {if (wavStream != null) wavStream.close();} catch (IOException e){}
}
}
}