import paulscode.sound.SoundBuffer; //导入依赖的package包/类
@Override
public SoundBuffer read() {
    if (!initialized || streamClosed)
        return null;

    final int limit = SoundSystemConfig.getStreamingBufferSize();
    ByteArrayOutputStream output = new ByteArrayOutputStream(limit);

    try {
        do {
            readBytes(output);
            if (!updateBuffer())
                break;
        } while (!streamClosed && output.size() < limit);
    } catch (Throwable t) {
        Log.warn(t, "Error in stream decoding, aborting");
        streamClosed = true;
    }

    return new SoundBuffer(output.toByteArray(), audioFormat);
}
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
@Override
public SoundBuffer readAll() {
    if (!initialized || streamClosed)
        return null;

    ByteArrayOutputStream output = new ByteArrayOutputStream();

    try {
        do {
            readBytes(output);
            if (!updateBuffer())
                break;
        } while (!streamClosed);
    } catch (Throwable t) {
        Log.warn(t, "Error in stream decoding, aborting");
        streamClosed = true;
    }

    return new SoundBuffer(output.toByteArray(), audioFormat);
}
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
@Override
public SoundBuffer read() {
    if (!initialized || streamClosed)
        return null;

    final int limit = SoundSystemConfig.getStreamingBufferSize();
    ByteArrayOutputStream output = new ByteArrayOutputStream(limit);

    try {
        do {
            output.write(buffer.getData(), 0, buffer.getLen());
            if (!updateBuffer())
                break;
        } while (!streamClosed && output.size() < limit);
    } catch (Throwable t) {
        Log.warn(t, "Error in stream decoding, aborting");
        streamClosed = true;
    }

    return new SoundBuffer(output.toByteArray(), audioFormat);
}
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
@Override
public SoundBuffer readAll() {
    if (!initialized || streamClosed)
        return null;

    ByteArrayOutputStream output = new ByteArrayOutputStream();

    try {
        do {
            output.write(buffer.getData(), 0, buffer.getLen());
            if (!updateBuffer())
                break;
        } while (!streamClosed);
    } catch (Throwable t) {
        Log.warn(t, "Error in stream decoding, aborting");
        streamClosed = true;
    }

    return new SoundBuffer(output.toByteArray(), audioFormat);
}
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Empties the streamBuffers list, shuts the channel down and removes 
* references to all instantiated objects.
*/
   @Override
   public void cleanup()
   {
       if( streamBuffers != null )
       {
           SoundBuffer buf = null;
           while( !streamBuffers.isEmpty() )
           {
               buf = streamBuffers.remove( 0 );
               buf.cleanup();
               buf = null;
           }
           streamBuffers.clear();
       }
       
       clip = null;
       soundBuffer = null;
       sourceDataLine = null;
       streamBuffers.clear();
       myMixer = null;
       myFormat = null;
       streamBuffers = null;
       
       super.cleanup();
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Feeds raw data to the stream.
* @param buffer Buffer containing raw audio data to stream.
* @return Number of prior buffers that have been processed, or -1 if error.
*/
   @Override
   public int feedRawAudioData( byte[] buffer )
   {
       // Stream buffers can only be queued for streaming sources:
       if( errorCheck( channelType != SoundSystemConfig.TYPE_STREAMING,
               "Raw audio data can only be processed by streaming sources." ) )
           return -1;

       if( errorCheck( streamBuffers == null,
                   "StreamBuffers queue null in method 'feedRawAudioData'." ) )
           return -1;
       
       streamBuffers.add( new SoundBuffer( buffer, myFormat ) );
       return buffersProcessed();
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Constructor:  Creates a new source using the specified parameters.
* @param listenerPosition FloatBuffer containing the listener's 3D coordinates.
* @param myBuffer OpenAL IntBuffer sound-buffer identifier to use for a new normal source.
* @param priority Setting this to true will prevent other sounds from overriding this one.
* @param toStream Setting this to true will create a streaming source.
* @param toLoop Should this source loop, or play only once.
* @param sourcename A unique identifier for this source.  Two sources may not use the same sourcename.
* @param filenameURL Filename/URL of the sound file to play at this source.
* @param soundBuffer Buffer containing audio data, or null if not loaded yet.
* @param x X position for this source.
* @param y Y position for this source.
* @param z Z position for this source.
* @param attModel Attenuation model to use.
* @param distOrRoll Either the fading distance or rolloff factor, depending on the value of 'att'.
* @param temporary Whether or not to remove this source after it finishes playing.
*/
   public SourceLWJGLOpenAL( FloatBuffer listenerPosition, IntBuffer myBuffer,
                             boolean priority, boolean toStream,
                             boolean toLoop, String sourcename,
                             FilenameURL filenameURL, SoundBuffer soundBuffer,
                             float x, float y, float z, int attModel,
                             float distOrRoll, boolean temporary )
   {
       super( priority, toStream, toLoop, sourcename, filenameURL, soundBuffer,
              x, y, z, attModel, distOrRoll, temporary );
       if( codec != null )
           codec.reverseByteOrder( true );
       this.listenerPosition = listenerPosition;
       this.myBuffer = myBuffer;
       libraryType = LibraryLWJGLOpenAL.class;
       pitch = 1.0f;
       resetALInformation();
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in one stream buffer worth of audio data.  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return The audio data wrapped into a SoundBuffer context.
*/
   public SoundBuffer read()
   {
       byte[] returnBuffer = null;
       
       while( !endOfStream( GET, XXX ) && ( returnBuffer == null ||
           returnBuffer.length < SoundSystemConfig.getStreamingBufferSize() ) )
       {
           if( returnBuffer == null )
               returnBuffer = readBytes();
           else
               returnBuffer = appendByteArrays( returnBuffer, readBytes() );
       }

       if( returnBuffer == null )
           return null;

       return new SoundBuffer( returnBuffer, audioFormat );
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in all the audio data from the stream (up to the default
* "maximum file size".  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return the audio data wrapped into a SoundBuffer context.
*/
   public SoundBuffer readAll()
   {
       byte[] returnBuffer = null;

       while( !endOfStream( GET, XXX ) )
       {
           if( returnBuffer == null )
               returnBuffer = readBytes();
           else
               returnBuffer = appendByteArrays( returnBuffer, readBytes() );
       }

       if( returnBuffer == null )
           return null;

       return new SoundBuffer( returnBuffer, audioFormat );
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Constructor:  Creates a new source matching the specified source.
* @param listener Handle to information about the listener.
* @param old Source to copy information from.
* @param soundBuffer Sound buffer to use if creating a new normal source.
*/
   public SourceJavaSound( ListenerData listener, Source old,
                           SoundBuffer soundBuffer )
   {
       super( old, soundBuffer );
       libraryType = LibraryJavaSound.class;

       // point handle to the listener information:
       this.listener = listener;
       positionChanged();
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Saves the specified sample data, under the specified identifier.  This
* identifier can be later used in place of 'filename' parameters to reference
* the sample data.
* @param buffer the sample data and audio format to save.
* @param identifier What to call the sample.
* @return True if there weren't any problems.
*/
   @Override
   public boolean loadSound( SoundBuffer buffer, String identifier )
   {
       // Make sure the buffer map exists:
       if( bufferMap == null )
       {
           bufferMap = new HashMap<String, SoundBuffer>();
           importantMessage( "Buffer Map was null in method 'loadSound'" );
       }

       // make sure they gave us an identifier:
       if( errorCheck(identifier == null,
                         "Identifier not specified in method 'loadSound'" ) )
           return false;

       // check if it is already loaded:
       if( bufferMap.get( identifier ) != null )
           return true;

       // save it for later:
       if( buffer != null )
           bufferMap.put( identifier, buffer );
       else
           errorMessage( "Sound buffer null in method 'loadSound'" );

       return true;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Constructor:  takes channelType identifier and a handle to the Mixer as 
* paramaters.  Possible values for channel type can be found in the 
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} class.
* @param type Type of channel (normal or streaming).
* @param mixer Handle to the JavaSound Mixer.
*/
   public ChannelJavaSound( int type, Mixer mixer )
   {
       super( type );
       libraryType = LibraryJavaSound.class;
       
       myMixer = mixer;
       clip = null;
       sourceDataLine = null;
       streamBuffers = new LinkedList<SoundBuffer>();
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Queues up a byte[] buffer of data to be streamed.
* @param buffer The next buffer to be played for a streaming source.
* @return False if an error occurred or if the channel is shutting down.
*/
   @Override
   public boolean queueBuffer( byte[] buffer )
   {
       // Stream buffers can only be queued for streaming sources:
       if( errorCheck( channelType != SoundSystemConfig.TYPE_STREAMING,
                       "Buffers may only be queued for streaming sources." ) )
           return false;
       
       // Make sure we have a SourceDataLine:
       if( errorCheck( sourceDataLine == null,
                       "SourceDataLine null in method 'queueBuffer'." ) )
           return false;
       
       // make sure a format was specified:
       if( errorCheck( myFormat == null,
                       "AudioFormat null in method 'queueBuffer'" ) )
           return false;
       
       // Queue a new buffer:
       streamBuffers.add( new SoundBuffer( buffer, myFormat ) );
       
       // Dequeue a buffer and process it:
       processBuffer();
       
       processed = 0;
       return true;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Plays the next queued byte[] buffer.  This method is run from the seperate 
* {@link paulscode.sound.StreamThread StreamThread}.
* @return False when no more buffers are left to process.
*/
   @Override
   public boolean processBuffer()
   {
       // Stream buffers can only be queued for streaming sources:
       if( errorCheck( channelType != SoundSystemConfig.TYPE_STREAMING,
                       "Buffers are only processed for streaming sources." ) )
           return false;
       
       // Make sure we have a SourceDataLine:
       if( errorCheck( sourceDataLine == null,
                       "SourceDataLine null in method 'processBuffer'." ) )
           return false;
       
       if( streamBuffers == null || streamBuffers.isEmpty() )
           return false;
       
       // Dequeue a buffer and feed it to the SourceDataLine:
       SoundBuffer nextBuffer = streamBuffers.remove( 0 );

       sourceDataLine.write( nextBuffer.audioData, 0,
                             nextBuffer.audioData.length );
       if( !sourceDataLine.isActive() )
           sourceDataLine.start();
       nextBuffer.cleanup();
       nextBuffer = null;
       
       return true;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in one stream buffer worth of audio data.  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return The audio data wrapped into a SoundBuffer context.
*/
   @Override
   public SoundBuffer read()
   {
       if( endOfStream( GET, XXX ) )
           return null;

       if( module == null )
       {
           errorMessage( "Module null in method 'read'" );
           return null;
       }

       // Check to make sure there is an audio format:
       if( myAudioFormat == null )
       {
           errorMessage( "Audio Format null in method 'read'" );
           return null;
       }

       int bufferFrameSize = (int) SoundSystemConfig.getStreamingBufferSize()
                                   / 4;

       int frames = songDuration - playPosition;
       if( frames > bufferFrameSize )
           frames = bufferFrameSize;

       if( frames <= 0 )
       {
           endOfStream( SET, true );
           return null;
       }
	byte[] outputBuffer = new byte[ frames * 4 ];

       ibxm.get_audio( outputBuffer, frames );

       playPosition += frames;
       if( playPosition >= songDuration )
       {
           endOfStream( SET, true );
       }

       // Reverse the byte order if necessary:
       if( reverseBytes )
           reverseBytes( outputBuffer, 0, frames * 4 );

       // Wrap the data into a SoundBuffer:
       SoundBuffer buffer = new SoundBuffer( outputBuffer, myAudioFormat );

       return buffer;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in all the audio data from the stream (up to the default
* "maximum file size".  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return the audio data wrapped into a SoundBuffer context.
*/
   @Override
   public SoundBuffer readAll()
   {
       if( module == null )
       {
           errorMessage( "Module null in method 'readAll'" );
           return null;
       }

       // Check to make sure there is an audio format:
       if( myAudioFormat == null )
       {
           errorMessage( "Audio Format null in method 'readAll'" );
           return null;
       }

       int bufferFrameSize = (int) SoundSystemConfig.getFileChunkSize()
                                   / 4;

	byte[] outputBuffer = new byte[ bufferFrameSize * 4 ];

       // Buffer to contain the audio data:
       byte[] fullBuffer = null;
       // frames of audio data:
       int frames;
       // bytes of audio data:
       int totalBytes = 0;

       while( (!endOfStream(GET, XXX)) &&
              (totalBytes < SoundSystemConfig.getMaxFileSize()) )
       {
           frames = songDuration - playPosition;
           if( frames > bufferFrameSize )
               frames = bufferFrameSize;
           ibxm.get_audio( outputBuffer, frames );
           totalBytes += (frames * 4);

           fullBuffer = appendByteArrays( fullBuffer, outputBuffer,
                                          frames * 4 );

           playPosition += frames;
           if( playPosition >= songDuration )
           {
               endOfStream( SET, true );
           }
       }

       // Reverse the byte order if necessary:
       if( reverseBytes )
           reverseBytes( fullBuffer, 0, totalBytes );

       // Wrap the data into a SoundBuffer:
       SoundBuffer buffer = new SoundBuffer( fullBuffer, myAudioFormat );

       return buffer;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
@Override
public SoundBuffer read()
{
	if(myAudioInputStream == null)
	{
		endOfStream(SET, true);
		return null;
	}

	// Get the format for the audio data:
	AudioFormat audioFormat = myAudioInputStream.getFormat();

	// Check to make sure there is an audio format:
	if(audioFormat == null)
	{
		errorMessage("Audio Format null in method 'read'");
		endOfStream(SET, true);
		return null;
	}

	// Variables used when reading from the audio input stream:
	int bytesRead = 0, cnt = 0;

	// Allocate memory for the audio data:
	byte[] streamBuffer = new byte[SoundSystemConfig.getStreamingBufferSize()];

	try
	{
		// Read until buffer is full or end of stream is reached:
		while((!endOfStream(GET, XXX)) && (bytesRead < streamBuffer.length))
		{
			myAudioInputStream.execute();
			if((cnt = myAudioInputStream.read(streamBuffer, bytesRead, streamBuffer.length
					- bytesRead)) <= 0)
			{
				endOfStream(SET, true);
				break;
			}
			// keep track of how many bytes were read:
			bytesRead += cnt;
		}
	} catch (IOException ioe)
	{

		/*
		 * errorMessage( "Exception thrown while reading from the " +
		 * "AudioInputStream (location #3)." ); printStackTrace( e ); return
		 * null;
		 */// TODO: Figure out why this exceptions is being thrown at end of
			// MP3 files!
		endOfStream(SET, true);
		return null;
	} catch (ArrayIndexOutOfBoundsException e)
	{
		//this exception is thrown at the end of the mp3's
		endOfStream(SET, true);
		return null;
       }

	// Return null if no data was read:
	if(bytesRead <= 0)
	{
		endOfStream(SET, true);
		return null;
	}

	// Insert the converted data into a ByteBuffer:
	// byte[] data = convertAudioBytes(streamBuffer,
	// audioFormat.getSampleSizeInBits() == 16);

	// Wrap the data into a SoundBuffer:
	SoundBuffer buffer = new SoundBuffer(streamBuffer, audioFormat);

	// Return the result:
	return buffer;
}
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in one stream buffer worth of audio data.  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return The audio data wrapped into a SoundBuffer context.
*/
   public SoundBuffer read()
   {
       if( endOfStream( GET, XXX ) )
           return null;

       if( module == null )
       {
           errorMessage( "Module null in method 'read'" );
           return null;
       }

       // Check to make sure there is an audio format:
       if( myAudioFormat == null )
       {
           errorMessage( "Audio Format null in method 'read'" );
           return null;
       }

       int bufferFrameSize = (int) SoundSystemConfig.getStreamingBufferSize()
                                   / 4;

       int frames = songDuration - playPosition;
       if( frames > bufferFrameSize )
           frames = bufferFrameSize;

       if( frames <= 0 )
       {
           endOfStream( SET, true );
           return null;
       }
	byte[] outputBuffer = new byte[ frames * 4 ];

       ibxm.get_audio( outputBuffer, frames );

       playPosition += frames;
       if( playPosition >= songDuration )
       {
           endOfStream( SET, true );
       }

       // Reverse the byte order if necessary:
       if( reverseBytes )
           reverseBytes( outputBuffer, 0, frames * 4 );

       // Wrap the data into a SoundBuffer:
       SoundBuffer buffer = new SoundBuffer( outputBuffer, myAudioFormat );

       return buffer;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Reads in all the audio data from the stream (up to the default
* "maximum file size".  See
* {@link paulscode.sound.SoundSystemConfig SoundSystemConfig} for more
* information about accessing and changing default settings.
* @return the audio data wrapped into a SoundBuffer context.
*/
   public SoundBuffer readAll()
   {
       if( module == null )
       {
           errorMessage( "Module null in method 'readAll'" );
           return null;
       }

       // Check to make sure there is an audio format:
       if( myAudioFormat == null )
       {
           errorMessage( "Audio Format null in method 'readAll'" );
           return null;
       }

       int bufferFrameSize = (int) SoundSystemConfig.getFileChunkSize()
                                   / 4;

	byte[] outputBuffer = new byte[ bufferFrameSize * 4 ];

       // Buffer to contain the audio data:
       byte[] fullBuffer = null;
       // frames of audio data:
       int frames;
       // bytes of audio data:
       int totalBytes = 0;

       while( (!endOfStream(GET, XXX)) &&
              (totalBytes < SoundSystemConfig.getMaxFileSize()) )
       {
           frames = songDuration - playPosition;
           if( frames > bufferFrameSize )
               frames = bufferFrameSize;
           ibxm.get_audio( outputBuffer, frames );
           totalBytes += (frames * 4);

           fullBuffer = appendByteArrays( fullBuffer, outputBuffer,
                                          frames * 4 );

           playPosition += frames;
           if( playPosition >= songDuration )
           {
               endOfStream( SET, true );
           }
       }

       // Reverse the byte order if necessary:
       if( reverseBytes )
           reverseBytes( fullBuffer, 0, totalBytes );

       // Wrap the data into a SoundBuffer:
       SoundBuffer buffer = new SoundBuffer( fullBuffer, myAudioFormat );

       return buffer;
   }
 

import paulscode.sound.SoundBuffer; //导入依赖的package包/类
/**
* Creates a new source and places it into the source map.
* @param priority Setting this to true will prevent other sounds from overriding this one.
* @param toStream Setting this to true will load the sound in pieces rather than all at once.
* @param toLoop Should this source loop, or play only once.
* @param sourcename A unique identifier for this source.  Two sources may not use the same sourcename.
* @param filenameURL Filename/URL of the sound file to play at this source.
* @param x X position for this source.
* @param y Y position for this source.
* @param z Z position for this source.
* @param attModel Attenuation model to use.
* @param distOrRoll Either the fading distance or rolloff factor, depending on the value of "attmodel".
*/
   @Override
   public void newSource( boolean priority, boolean toStream, boolean toLoop,
                          String sourcename, FilenameURL filenameURL, float x,
                          float y, float z, int attModel, float distOrRoll )
   {
       SoundBuffer buffer = null;
       
       if( !toStream )
       {
           // Grab the audio data for this file:
           buffer = bufferMap.get( filenameURL.getFilename() );
           // if not found, try loading it:
           if( buffer == null )
           {
               if( !loadSound( filenameURL ) )
               {
                   errorMessage( "Source '" + sourcename + "' was not created "
                                 + "because an error occurred while loading "
                                 + filenameURL.getFilename() );
                   return;
               }
           }
           // try and grab the sound buffer again:
           buffer = bufferMap.get( filenameURL.getFilename() );
           // see if it was there this time:
           if( buffer == null )
           {
               errorMessage( "Source '" + sourcename + "' was not created "
                             + "because audio data was not found for "
                             + filenameURL.getFilename() );
               return;
           }
       }
       
       if( !toStream && buffer != null )
           buffer.trimData( maxClipSize );
       
       sourceMap.put( sourcename,
                      new SourceJavaSound( listener, priority, toStream,
                                           toLoop, sourcename, filenameURL,
                                           buffer, x, y, z, attModel,
                                           distOrRoll, false ) );
   }