我已经阅读了两篇有关从AudioInputStream中提取样本并将其转换为dB的文章。

https://stackoverflow.com/a/26576548/8428414

https://stackoverflow.com/a/26824664/8428414

据我了解byte[] bytes;具有如下结构:

Index 0: Sample 0 (Left Channel)
Index 1: Sample 0 (Right Channel)
Index 2: Sample 1 (Left Channel)
Index 3: Sample 1 (Right Channel)
Index 4: Sample 2 (Left Channel)
Index 5: Sample 2 (Right Channel)

在第一篇文章中,它展示了如何从一个通道(单声道)获取样本。

所以,我的问题是我想分别为右声道和左声道分别获取样本,以便计算右声道和左声道的dB。

这是代码。如何更改它以分别获得左右声道?
我不明白索引i的变化...
final byte[] buffer = new byte[2048];

float[] samples = new float[buffer.length / 2];

for (int n = 0; n != -1; n = in.read(buffer, 0, buffer.length)) {
    line.write(buffer, 0, n);

    for (int i = 0, sampleIndex = 0; i < n; ) {
        int sample = 0;

        sample |= buffer[i++] & 0xFF; // (reverse these two lines
        sample |= buffer[i++] << 8;   //  if the format is big endian)

        // normalize to range of +/-1.0f
        samples[sampleIndex++] = sample / 32768f;
    }

    float rms = 0f;
    for (float sample : samples) {
        rms += sample * sample;
    }

    rms = (float) Math.sqrt(rms / samples.length);

希望你能帮助我。先感谢您。

最佳答案

立体声信号保存的格式称为interleaved。即,正如您正确描述的那样,它是LLRRLLRRLLRR...。因此,您首先需要阅读一个左样本,然后是一个右样本,依此类推。

我已经编辑了您的代码以反射(reflect)这一点。但是,通过refactoring仍有一些改进的空间。

注意:代码更改仅处理交织。我还没有检查其余的代码。

final byte[] buffer = new byte[2048];

// create two buffers. One for the left, one for the right channel.
float[] leftSamples = new float[buffer.length / 4];
float[] rightSamples = new float[buffer.length / 4];

for (int n = 0; n != -1; n = in.read(buffer, 0, buffer.length)) {
    line.write(buffer, 0, n);

    for (int i = 0, sampleIndex = 0; i < n; ) {
        int sample = 0;

        leftSample |= buffer[i++] & 0xFF; // (reverse these two lines
        leftSample |= buffer[i++] << 8;   //  if the format is big endian)

        rightSample |= buffer[i++] & 0xFF; // (reverse these two lines
        rightSample |= buffer[i++] << 8;   //  if the format is big endian)

        // normalize to range of +/-1.0f
        leftSamples[sampleIndex] = leftSample / 32768f;
        rightSamples[sampleIndex] = rightSample / 32768f;

        sampleIndex++;
    }

    // now compute RMS for left
    float leftRMS = 0f;
    for (float sample : leftSamples) {
        leftRMS += sample * sample;
    }

    leftRMS = (float) Math.sqrt(leftRMS / leftSamples.length);

    // ...and right
    float rightRMS = 0f;
    for (float sample : rightSamples) {
        rightRMS += sample * sample;
    }

    rightRMS = (float) Math.sqrt(rightRMS / rightSamples.length);
}

关于java - 计算两个声道的音频电平/幅度/分贝,我们在Stack Overflow上找到一个类似的问题:https://stackoverflow.com/questions/61149086/

10-11 03:29