java – 将音频立体声转换为音频字节
我正在尝试进行一些音频处理,我真的陷入了立体声到单声道转换.我在网上查看了立体声到单声道转换.
据我所知,我可以采用左声道,右声道,将它们相加并除以2.但是当我再次将结果转换为WAV文件时,我得到了很多前景噪声.我知道处理数据时可能会产生噪声,字节变量会有一些溢出. 这是从我的MP3文件中检索byte []数据块的类: 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; } } 这意味着,每次调用getSamples时,它都返回一个数组,如: buff = {Lchannel,Rchannel,Lchannel,Rchannel ……} 转换为单声道的处理程序如下所示: 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); } } 并使用以下方法写入wav文件: 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; } } 以44100作为采样率. 请记住,实际上我已经得到它的byte []数组已经是pcm,所以mp3 – > pcm转换是通过指定完成的
解决方法
正如评论中指出的那样,字节顺序可能是错误的.此外,转换为带符号的短路并将其移位可能会导致第一个字节为0xFF.
尝试: 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); 然后切换HI和LO的值以查看它是否变好. (编辑:李大同) 【声明】本站内容均来自网络,其相关言论仅代表作者个人观点,不代表本站立场。若无意侵犯到您的权利,请及时与联系站长删除相关内容! |