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用xugler实现java音频转换

我正在尝试用Java中的Xuggler将aac/wav/wma音频文件转换为mp3

不幸的是,我的质量损失很大。我的输入文件大小约为7MB,输出文件大小仅为1,5MB

采样率设置为44100 Hz,是否有其他参数需要设置

谢谢你的回答

  if (args.length <= 1)
        throw new IllegalArgumentException("must pass an input filename and output filename as argument");

    IMediaWriter writer = ToolFactory.makeWriter(args[1]);

    String filename = args[0];

    // Create a Xuggler container object
    IContainer container = IContainer.make();

    // Open up the container
    if (container.open(filename, IContainer.Type.READ, null) < 0)
        throw new IllegalArgumentException("could not open file: " + filename);

    // query how many streams the call to open found
    int numStreams = container.getNumStreams();

    // and iterate through the streams to find the first audio stream
    int audioStreamId = -1;
    IStreamCoder audioCoder = null;
    for(int i = 0; i < numStreams; i++)
    {
        // Find the stream object
        IStream stream = container.getStream(i);
        // Get the pre-configured decoder that can decode this stream;
        IStreamCoder coder = stream.getStreamCoder();

        if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_AUDIO)
        {
            audioStreamId = i;
            audioCoder = coder;
            audioCoder.setBitRate(container.getBitRate());

            break;
        }
    }

    if (audioStreamId == -1)
        throw new RuntimeException("could not find audio stream in container: "+filename);

    /* We read only AAC file for the moment */
    if(audioCoder.getCodecID() != ICodec.ID.CODEC_ID_AAC 
        && audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WAVPACK 
        && audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAV1
        && audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAV2
        && audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAPRO
        && audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAVOICE)
    {
        System.out.println("Read only AAC, WAV or WMA files");
        System.exit(1);
    }

    audioCoder.setSampleFormat(IAudioSamples.Format.FMT_S16);
    /*
     * Now we have found the audio stream in this file.  Let's open up our decoder so it can
     * do work.
     */
    if (audioCoder.open() < 0)
        throw new RuntimeException("could not open audio decoder for container: "+filename);

    int streamIndex = writer.addAudioStream(0, 0, audioCoder.getChannels(), audioCoder.getSampleRate());


    System.out.println("audio Frame size : "+audioCoder.getAudioFrameSize());


    /*
     * Now, we start walking through the container looking at each packet.
     */
    IPacket packet = IPacket.make();

    while(container.readNextPacket(packet) >= 0)
    {
        /*
         * Now we have a packet, let's see if it belongs to our audio stream
         */
        if (packet.getStreamIndex() == audioStreamId)
        {
            /*
             * We allocate a set of samples with the same number of channels as the
             * coder tells us is in this buffer.
             * 
             * We also pass in a buffer size (1024 in our example), although Xuggler
             * will probably allocate more space than just the 1024 (it's not important why).
             */

            IAudioSamples samples = IAudioSamples.make(512, audioCoder.getChannels(),IAudioSamples.Format.FMT_S16 );

            /*
             * A packet can actually contain multiple sets of samples (or frames of samples
             * in audio-decoding speak).  So, we may need to call decode audio multiple
             * times at different offsets in the packet's data.  We capture that here.
             */
            int offset = 0;

            /*
             * Keep going until we've processed all data
             */         

            while(offset < packet.getSize())
            {
                int bytesDecoded = audioCoder.decodeAudio(samples, packet, offset);
                if (bytesDecoded < 0)
                    throw new RuntimeException("got error decoding audio in: " + filename);

                offset += bytesDecoded;

                /*
                 * Some decoder will consume data in a packet, but will not be able to construct
                 * a full set of samples yet.  Therefore you should always check if you
                 * got a complete set of samples from the decoder
                 */                                     
                if (samples.isComplete())
                {
                    writer.encodeAudio(streamIndex, samples);   
                }
            }
        }
        else
        {
            /*
             * This packet isn't part of our audio stream, so we just silently drop it.
             */
            do {} while(false);
        }
    }

共 (3) 个答案

  1. # 1 楼答案

    当心当你有一个带封面的mp3(png)时,你可能会因为试图将视频png流发送到音频流而出错。。通过使用ISamples并通过if(packet.getStreamIndex()==audioStreamId){}读取数据包,可以更好地控制所使用的流。 检查我的完整代码:

    private static void streamToSource( OutputStream source, Path path ) throws IOException {
    
        byte[] buffer = new byte[4096];
        PipedInputStream pis = new PipedInputStream( );
        PipedOutputStream pos = new PipedOutputStream( pis );
        convertToMP3Xuggler( path, pos );
    
        System.out.println( "start streaming" );
        int nRead = 0;
        while ( ( nRead = pis.read( buffer ) ) != -1 ) {
            source.write( buffer,0 , nRead );
        }
        pis.close( );
    
        System.out.println( "end : " + path );
    
    }
    
    private static void convertToMP3Xuggler( Path path, PipedOutputStream pos ) throws FileNotFoundException {
    
        // create a media reader
        // final IMediaReader mediaReader = ToolFactory.makeReader( XugglerIO.map( new FileInputStream( path.toFile( ) ) ) );
    
        // create a media writer
        // IMediaWriter mediaWriter = ToolFactory.makeWriter( XugglerIO.map( XugglerIO.generateUniqueName( os, ".mp3" ), os ), mediaReader );
        IMediaWriter mediaWriter = ToolFactory.makeWriter( XugglerIO.map( pos ) );
        // manually set the container format (because it can't detect it by filename anymore)
    
    
        IContainerFormat containerFormat = IContainerFormat.make( );
        containerFormat.setOutputFormat( "mp3", null, "audio/mp3" );
        mediaWriter.getContainer( ).setFormat( containerFormat );
    
        System.out.println( "file = " + path.toFile( ).toString( ) );
    
        IContainer audioContainer = IContainer.make( );
        audioContainer.open( path.toFile( ).toString( ), IContainer.Type.READ, null );
    
        System.out.println( "streams= " + audioContainer.getNumStreams( ) );
        System.out.println( "# Duration (ms): " + ( ( audioContainer.getDuration( ) == Global.NO_PTS ) ? "unknown" : "" + audioContainer.getDuration( ) / 1000 ) );
        System.out.println( "# File size (bytes): " + audioContainer.getFileSize( ) );
        System.out.println( "# Bit rate: " + audioContainer.getBitRate( ) );
        int audioStreamId = -1;
    
    
        for ( int i = 0; i < audioContainer.getNumStreams( ); i++ ) {
            // Find the stream object
            IStream stream = audioContainer.getStream( i );
            // Get the pre-configured decoder that can decode this stream;
            IStreamCoder coder = stream.getStreamCoder( );
            if ( coder.getCodecType( ) == ICodec.Type.CODEC_TYPE_AUDIO ) {
                audioStreamId = i;
                break;
            }
        }
        if ( audioStreamId < 0 ) {
            throw new IllegalArgumentException( "cannot find audio stream in the current file : " + path.toString( ) );
        }
        System.out.println( "found audio stream = " + audioStreamId );
    
        IStreamCoder coderAudio = audioContainer.getStream( audioStreamId ).getStreamCoder( );
    
        if ( coderAudio.open( null, null ) < 0 ) {
            throw new RuntimeException( "Cant open audio coder" );
        }
        coderAudio.setSampleFormat( IAudioSamples.Format.FMT_S16 );
    
        System.out.println( "bitrate from reading = " + audioContainer.getBitRate( ) );
        System.out.println( "bitrate from reading = " + coderAudio.getBitRate( ) );
    
        int streamIndex = mediaWriter.addAudioStream( 0, 0, coderAudio.getChannels( ), coderAudio.getSampleRate( ) );
        IStreamCoder writerCoder = mediaWriter.getContainer( ).getStream( streamIndex ).getStreamCoder( );
        writerCoder.setFlag( IStreamCoder.Flags.FLAG_QSCALE, false );
        writerCoder.setBitRate( BITRATE * 1000 );
        writerCoder.setBitRateTolerance( 0 );
        System.out.println( "bitrate for output = " + writerCoder.getBitRate( ) );
    
        IPacket packet = IPacket.make( );
    
        runInThread( path, pos, mediaWriter, audioContainer, audioStreamId, coderAudio, streamIndex, packet );
    
    }
    
    private static void runInThread( Path path, PipedOutputStream pos, IMediaWriter mediaWriter, IContainer audioContainer, int audioStreamId, IStreamCoder coderAudio, int streamIndex, IPacket packet ) {
    
        new Thread( ) {
            @Override
            public void run( ) {
    
                while ( audioContainer.readNextPacket( packet ) >= 0 ) {
                    /*
                     * Now we have a packet, let's see if it belongs to our audio stream
                     */
                    if ( packet.getStreamIndex( ) == audioStreamId ) {
                        /*
                         * We allocate a set of samples with the same number of channels as the
                         * coder tells us is in this buffer.
                         * We also pass in a buffer size (4096 in our example), although Xuggler
                         * will probably allocate more space than just the 4096 (it's not important why).
                         */
    
                        IAudioSamples samples = IAudioSamples.make( 4096, coderAudio.getChannels( ), IAudioSamples.Format.FMT_S16 );
    
                        /*
                         * A packet can actually contain multiple sets of samples (or frames of samples
                         * in audio-decoding speak). So, we may need to call decode audio multiple
                         * times at different offsets in the packet's data. We capture that here.
                         */
                        int offset = 0;
    
                        /*
                         * Keep going until we've processed all data
                         */
    
                        while ( offset < packet.getSize( ) ) {
                            int bytesDecoded = coderAudio.decodeAudio( samples, packet, offset );
                            if ( bytesDecoded < 0 ) {
                                System.out.println( "decode error in : " + path + " bytesDecoded =" + bytesDecoded + " offset=" + offset + " packet=" + packet );
                                break;
                                //                                throw new RuntimeException( "got error decoding audio in: " + path );
                            }
    
                            offset += bytesDecoded;
    
                            //                            System.out.println( "pktSize = " + packet.getSize( ) + "  offset = " + offset + " samplesComplete = " + samples.isComplete( ) );
    
                            /*
                             * Some decoder will consume data in a packet, but will not be able to construct
                             * a full set of samples yet. Therefore you should always check if you
                             * got a complete set of samples from the decoder
                             */
                            if ( samples.isComplete( ) ) {
                                mediaWriter.encodeAudio( streamIndex, samples );
                            }
                        }
                    }
                }
                coderAudio.close( );
                audioContainer.close( );
                mediaWriter.close( );
                try {
                    pos.close( );
                } catch ( IOException e ) {
                    e.printStackTrace( );
                }
            }
    
        }.start( );
    }
    
  2. # 2 楼答案

    我会这样做:

    public void convertToMP3(File input, File output, int kbps) { //modify on your convenience
        // create a media reader
        IMediaReader mediaReader = ToolFactory.makeReader(input.getPath());
    
        // create a media writer
        IMediaWriter mediaWriter = ToolFactory.makeWriter(output.getPath(), mediaReader);
    
        // add a writer to the reader, to create the output file
        mediaReader.addListener(mediaWriter);
    
        // add a IMediaListner to the writer to change bit rate
        mediaWriter.addListener(new MediaListenerAdapter() {
            @Override
            public void onAddStream(IAddStreamEvent event) {
                IStreamCoder streamCoder = event.getSource().getContainer().getStream(event.getStreamIndex()).getStreamCoder();
                streamCoder.setFlag(IStreamCoder.Flags.FLAG_QSCALE, false);
                streamCoder.setBitRate(kbps);
                streamCoder.setBitRateTolerance(0);
                }
            });
    
        // read and decode packets from the source file and
        // and dispatch decoded audio and video to the writer
        while (mediaReader.readPacket() == null);
    }
    

    输入是要转换的文件(aac/wav/wma),输出是新文件。mp3文件(Xugler通过扩展名计算转换)

    您可以提高质量,增加kbps(即,对于320 kbps,您需要通过320000)

    希望有帮助:-)

    仅供参考:对于Java项目,如果尚未导入,则需要导入以下内容:

    import com.xuggle.mediatool.MediaListenerAdapter;
    import com.xuggle.mediatool.event.IAddStreamEvent;
    import com.xuggle.xuggler.IStreamCoder;
    
  3. # 3 楼答案

    我不确定具体的选项和它们的作用,但看看javadoc for IStreamCoder。你可能还想玩其他各种选择。如果想要完全控制,甚至可以使用setFlags()方法直接在ffmpeg上设置标志(Xugler在下面使用)