Java – convert audio stereo to audio bytes
I'm trying to do some audio processing. I'm really stuck in stereo to mono conversion I checked the stereo to mono conversion on the Internet
As far as I know, I can take the left channel and the right channel, add them and divide them by 2 But when I convert the result to WAV file again, I get a lot of foreground noise I know that there may be noise when processing data, and there will be some overflow of byte variables
This is the class to retrieve byte [] data block from my MP3 file:
public class InputSoundDecoder {
private int BUFFER_SIZE = 128000; private String _inputFileName; private File _soundFile; private AudioInputStream _audioInputStream; private AudioFormat _audioInputFormat; private AudioFormat _decodedFormat; private AudioInputStream _audioInputDecodedStream; public InputSoundDecoder(String fileName) throws UnsuportedSampleRateException{ this._inputFileName = fileName; this._soundFile = new File(this._inputFileName); try{ this._audioInputStream = AudioSystem.getAudioInputStream(this._soundFile); } catch (Exception e){ e.printStackTrace(); System.err.println("Could not open file: " + this._inputFileName); System.exit(1); } this._audioInputFormat = this._audioInputStream.getFormat(); this._decodedFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,44100,16,2,1,false); this._audioInputDecodedStream = AudioSystem.getAudioInputStream(this._decodedFormat,this._audioInputStream); /** Supported sample rates */ switch((int)this._audioInputFormat.getSampleRate()){ case 22050: this.BUFFER_SIZE = 2304; break; case 44100: this.BUFFER_SIZE = 4608; break; default: throw new UnsuportedSampleRateException((int)this._audioInputFormat.getSampleRate()); } System.out.println ("# Channels: " + this._decodedFormat.getChannels()); System.out.println ("Sample size (bits): " + this._decodedFormat.getSampleSizeInBits()); System.out.println ("Frame size: " + this._decodedFormat.getFrameSize()); System.out.println ("Frame rate: " + this._decodedFormat.getFrameRate()); } public byte[] getSamples(){ byte[] abData = new byte[this.BUFFER_SIZE]; int bytesRead = 0; try{ bytesRead = this._audioInputDecodedStream.read(abData,abData.length); } catch (Exception e){ e.printStackTrace(); System.err.println("Error getting samples from file: " + this._inputFileName); System.exit(1); } if (bytesRead > 0) return abData; else return null; }
}
This means that every time getsamples is called, it returns an array, such as:
buff = {Lchannel,Rchannel,Lchannel,Rchannel ……}
The processing procedure for converting to mono is as follows:
byte[] buff = null; while( (buff = _input.getSamples()) != null ){ /** Convert to mono */ byte[] mono = new byte[buff.length/2]; for (int i = 0 ; i < mono.length/2; ++i){ int left = (buff[i * 4] << 8) | (buff[i * 4 + 1] & 0xff); int right = (buff[i * 4 + 2] <<8) | (buff[i * 4 + 3] & 0xff); int avg = (left + right) / 2; short m = (short)avg; /*Mono is an average between 2 channels (stereo)*/ mono[i * 2] = (byte)((short)(m >> 8)); mono[i * 2 + 1] = (byte)(m & 0xff); }
}
And write the wav file using the following methods:
public static void writeWav(byte [] theResult,int samplerate,File outfile) { // Now convert theResult into a wav file // probably should use a file if samplecount is too big! int theSize = theResult.length; InputStream is = new ByteArrayInputStream(theResult); //Short2InputStream sis = new Short2InputStream(theResult); AudioFormat audioF = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,samplerate,// channels 2,// framesize samplerate,false ); AudioInputStream ais = new AudioInputStream(is,audioF,theSize); try { AudioSystem.write(ais,AudioFileFormat.Type.WAVE,outfile); } catch (IOException ioe) { System.err.println("IO Exception; probably just done with file"); return; } }
Take 44100 as the sampling rate
Remember, in fact, I have obtained that the byte [] array is PCM, so the MP3 – > PCM conversion is completed by specifying
Solution
As pointed out in the comments, the byte order may be wrong In addition, converting to a signed short circuit and shifting it may result in the first byte being 0xff
Try:
int HI = 0; int LO = 1; int left = (buff[i * 4 + HI] << 8) | (buff[i * 4 + LO] & 0xff); int right = (buff[i * 4 + 2 + HI] << 8) | (buff[i * 4 + 2 + LO] & 0xff); int avg = (left + right) / 2; mono[i * 2 + HI] = (byte)((avg >> 8) & 0xff); mono[i * 2 + LO] = (byte)(avg & 0xff);
Then switch the values of hi and lo to see if it gets better