如何使用标量手动更改 Android AudioTrack 流的增益级别？

我正在使用 Android AudioTrack 对象来传输 16 位单声道 PCM 文件。我的代码从 DataInputStream 读取双精度值，将其转换为 8 字节，然后将该 8 字节保存到缓冲区以写入 AudioTrack。这很好用。但现在，我尝试通过将双精度乘以标量（例如 0.5）来调整增益。当我这样做时，它会使我的音频严重失真。我尝试使用浮点数而不是双精度数，得到了相同的结果。有没有更好的方法来改变流的增益？我的最终目标是创建一个简单的回声效果，这就是我这样做的原因。

ByteBuffer bb = ByteBuffer.allocate(8);
while(isPlaying){
        try {
            //fill buffer with bytes from file reader
            for(int i=0; i < BUFFSIZE/8; i++){

                //read double from DataInputStream
                double temp = dis.readDouble();

                //save double to ByteBuffer
                bb.putDouble(0, temp * .5);

                // save 8 bytes to array of bytes
                bb.rewind();
                bb.get(buff,i*8,8);
            }

            //write buffer to track to play
            track.write(buff, 0, BUFFSIZE);

        } catch (IOException e) {
            break; //when eof is reached
        }
    }

如果您使用 Android AudioRecord 和 AudioTrack 类进行录制和播放，请记住，16 位单声道音频样本默认为小端字节序。如果您使用 DataInputStream 类的 readDouble() 方法读取字节，则会不加区别地读取 Big endian 中的 64 位，因此会以错误的顺序读取字节。但请记住，即使 readDouble 会以小端方式读取字节，或者如果您的音频采用大端方式，它也会读取 64 位并将它们视为一个大双值，而实际上每个样本都代表一个带符号短值，其值范围为从 -32768 到 +32767（假设它已签名）。如果出于数字信号处理的目的将其转换为双精度型，最好通过除以 32767.0f 将其标准化为 -1.0 到 +1.0 范围（否则，不必费心将其转换为双精度型）第一名）。当您完成 DSP 的双精度运算后，将其乘以 32767 并将其转换回小端。这是一个快速但肮脏的方法：

try{
    for(int offset=0; offset < buff_size; offset+= 2)  //increment index by two because 16bit mono sample is 2 bytes long
    {

        //reads 2 bytes from stream and stores them in buff at offset. returns number of bytes read.
        if (dis.read(buff,offset, 2) > 0){

            //where buff is an array of bytes
            //                 Low Byte               High Byte    Left Shift
            double sample= ((buff[offset + 0]) | (buff[offset + 1] << 8) );
            /*Explanation: First We take the High Byte and shift it 8 Bytes to the left. In Java, this promotes it to an integer(32 bits).
            * Next we merge the int representing High Byte and the Low Byte byte with a bitwise OR operator
            * 00000000 0000000 10101111 00000000
            *                       OR  10110000 =
            * 00000000 0000000 10101111 10110000 Now our bytes are in the Big Endian order required for primitive types */

            //since 2 bytes is a short which has range -32768 to +32767, we divide by 32768 to normalize to -1.0 to +1.0 range for DSP
            sample = sample /32768.0;

            sample = sample * .5;

            //convert it back to small endian
            int nsample = (int) Math.round(sample * 32767.0);//expands it to the range of -32768 to 32767 range of short, round, & truncate
            buff[offset + 1] = (byte) ((nsample >> 8) & 0xFF); //isolate and extract the high byte
            buff[offset + 0] = (byte) (nsample & 0xFF);        //isolate the low byte with MASK
        }

    }

    track.write(buff, 0, buff_size);
    }catch(Exception e){
        //
    }

来源：http://www.jsresources.org/faq_audio.html#samples_organized http://www.jsresources.org/faq%5faudio.html#samples%5forganized