JavaEncoderpublic class JavaEncoder extends BufferedEncoder RTP GSM encoder plugin wrapper, which uses Java methods to do the encoding. |
Fields Summary |
---|
public static final String | a_copyright_notice
Licensed Materials - Property of IBM
"Restricted Materials of IBM"
5648-B81
(c) Copyright IBM Corporation 1997,1999 All Rights Reserved
US Government Users Restricted Rights - Use, duplication or
disclosure restricted by GSA ADP Schedule Contract with
IBM Corporation. | protected GsmEncoder | encoder | private int | sample_count | private long | currentSeq | private long | timestamp | byte[] | pendingBuffer |
Constructors Summary |
---|
public JavaEncoder()
////////////////////////////////////////////////////////////////////////////
// Methods
supportedInputFormats = new AudioFormat[] {
new AudioFormat(AudioFormat.LINEAR,
Format.NOT_SPECIFIED,
16,
1,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
Format.NOT_SPECIFIED,
Format.NOT_SPECIFIED,
Format.byteArray) };
defaultOutputFormats = new AudioFormat[] {
new AudioFormat(AudioFormat.GSM) /*,
new AudioFormat(AudioFormat.GSM_RTP)*/};
PLUGIN_NAME="GSM Encoder";
historySize = 320;
pendingFrames = 0;
// for RTP, set default packetsize to 160 audiosamples for a
// default ms/packet of 20ms. This works out to 33 octets.
//note if this value changes in setPacketSize(), sample count
// needs to be updated as well.
packetSize = 33;
|
Methods Summary |
---|
protected int | calculateFramesNumber(int inputSize)
return inputSize / 320;
| protected int | calculateOutputSize(int inputSize)
return calculateFramesNumber(inputSize) * 33 ;
| public void | close()
encoder=null;
| protected boolean | codecProcess(byte[] inpData, int readPtr, byte[] outData, int writePtr, int inpLength, int[] readBytes, int[] writeBytes, int[] frameNumber, int[] regions, int[] regionsTypes)
int inCount = 0;
int outCount = 0;
int channels=inputFormat.getChannels();
boolean isStereo = ( channels == 2);
int frames = inpLength/(320);
regions[0]=writePtr;
for (int frameCounter = 0; frameCounter<frames ; frameCounter++) {
encoder.gsm_encode_frame(inpData, readPtr , outData,writePtr);
readPtr += 320;
inCount += 320;
outCount += 33;
writePtr += 33;
//regions [frameCounter+1]= outCount + writePtr;
regions [frameCounter +1] = writePtr;
regionsTypes[frameCounter ]= 0;
//System.out.println(inCount+" "+outCount+" "+inpLength);
}
readBytes [0]=inCount;
writeBytes[0]=outCount;
frameNumber[0]=frames;
return true;
| public void | codecReset()
encoder.gsm_encoder_reset();
| protected javax.media.Format[] | getMatchingOutputFormats(javax.media.Format in)
AudioFormat af =(AudioFormat) in;
supportedOutputFormats = new AudioFormat[] {
new AudioFormat(
AudioFormat.GSM,
af.getSampleRate(),
16,
af.getChannels(),
Format.NOT_SPECIFIED,
Format.NOT_SPECIFIED,
264,
Format.NOT_SPECIFIED,
Format.byteArray
)/*,
new AudioFormat(
AudioFormat.GSM_RTP,
af.getSampleRate(),
Format.NOT_SPECIFIED,
af.getChannels(),
Format.NOT_SPECIFIED,
Format.NOT_SPECIFIED,
264,
Format.NOT_SPECIFIED,
Format.byteArray
) */
};
return supportedOutputFormats;
| public void | open()
encoder=new GsmVadEncoder();
encoder.gsm_encoder_reset();
| public int | process(javax.media.Buffer inputBuffer, javax.media.Buffer outputBuffer)
// let the buffered encoder process the input buffer. this will
// encode the data for us.Only RTP packetization will be desired
// at this time.
int retVal = super.process(inputBuffer, outputBuffer);
// if the output format is not RTP, just return the from here.
if (!outputFormat.getEncoding().equals(AudioFormat.GSM_RTP))
return retVal;
// if it is RTP, packetize the data.
if (outputFormat.getEncoding().equals(AudioFormat.GSM_RTP)){
// before we proceed for packetization, check for failure in
// encoding and EOM
if (retVal == BUFFER_PROCESSED_FAILED)
return retVal;
if (isEOM(inputBuffer) ) {
propagateEOM(outputBuffer);
return BUFFER_PROCESSED_OK;
}
// Now, if there are no pending frames, we are beginning
// packetization of a new buffer.get a handle over the buffer to
// be packetized
if (pendingFrames == 0)
pendingBuffer = (byte[])outputBuffer.getData();
// start packetizing one frame at a time (160 samples)
// the size of outputdata depends on the packet size set.
byte[] outData = new byte[packetSize];
outputBuffer.setData(outData);
updateOutput(outputBuffer, outputFormat,packetSize, 0);
outputBuffer.setSequenceNumber(currentSeq++);
outputBuffer.setTimeStamp(timestamp);
timestamp+=sample_count;
System.arraycopy(pendingBuffer,
regions[pendingFrames],
outData,
0,
packetSize);
if (pendingFrames + 1== frameNumber[0]){
pendingFrames = 0;
pendingBuffer = null;
return BUFFER_PROCESSED_OK;
}else
pendingFrames++;
return INPUT_BUFFER_NOT_CONSUMED;
}//end of GSM_RTP
return retVal;
|
|