EightSVXAudioClip (MonteCC repack 1.0.1 API)

java.lang.Object
- org.monte.media.eightsvx.EightSVXAudioClip

All Implemented Interfaces:

AudioClip, LoopableAudioClip
```
public class EightSVXAudioClip
extends Object
implements LoopableAudioClip
```
Represents an audio sample of type IFF 8SVX.
Supported audio formats:
8 bit linear and fibonacci encoded data samples.
All sample rates
Stereo and Mono
Unsupported features:
Attack and Release information is ignored.
Multi octave samples are not handled.
Known Issues
This class has been implemented with JDK 1.1 in mind. JDK 1.1 does not have a public API for Sound. This class will thus work only on a small number of Java VMS.
Poor sound qualitiy: All data is being converted to U-Law 8000 Hertz, since this is the only kind of audio data that JDK 1.1 supports (As far as I know).
Stereo sound is converted to mono. As far as I know there is now stereo support built in JDK 1.1.

Version:

1.1 2003-04-05 Revised.
1.0 1999-10-19

Author:

Werner Randelshofer, Hausmatt 10, CH-6405 Goldau, Switzerland

Field Summary

Fields
Modifier and Type Field and Description

static int LEFT

static int RIGHT

static int S_CMP_FIB_DELTA

static int S_CMP_NONE

static int STEREO
- Fields inherited from interface org.monte.media.eightsvx.LoopableAudioClip
  LOOP_CONTINUOUSLY

Fields
Modifier and Type	Field and Description
`static int`	`LEFT`
`static int`	`RIGHT`
`static int`	`S_CMP_FIB_DELTA`
`static int`	`S_CMP_NONE`
`static int`	`STEREO`

Constructor Summary

Constructors
Constructor and Description

EightSVXAudioClip()

Constructors
Constructor and Description
`EightSVXAudioClip()`

Method Summary

Methods
Modifier and Type	Method and Description
`static double`	`computeStereoVolumeCorrection(byte[] stereo)` This finds the volume correction needed when converting this stereo sample to mono.
`LoopableAudioClip`	`createAudioClip()`
`LoopableAudioClip`	`createAudioClip(int sampleRate, int volume, float pan)`
`LoopableAudioClip`	`createJDK10AudioClip(int sampleRate)`
`LoopableAudioClip`	`createJDK13AudioClip(int sampleRate, int volume, float pan)`
`byte[]`	`get8SVXBody()`
`protected String`	`getAuthor()`
`protected String`	`getCopyright()`
`int`	`getCtOctave()`
`protected String`	`getName()`
`long`	`getOneShotHiSamples()`
`protected String`	`getRemark()`
`long`	`getRepeatHiSamples()`
`int`	`getSampleRate()`
`long`	`getSamplesPerHiCycle()`
`long`	`getSampleType()`
`int`	`getSCompression()`
`int`	`getVolume()`
`static byte[]`	`linear16ToULaw(int[] linear16)` Converts a buffer of signed 8bit samples to uLaw.
`static byte[]`	`linear8StereoToMono(byte[] stereo, double volumeCorrection)` This converts a stereo sample to mono.
`static byte[]`	`linear8ToULaw(byte[] linear8)` Halfband lowpass.
`void`	`loop()`
`void`	`loop(int count)` Starts looping playback from the current position.
`void`	`play()`
`void`	`prepare()` Make this clip ready for playback.
`static byte[]`	`resample(byte[] input, int inputSampleRate, int outputSampleRate)` Resamples audio data to match the given sample rate and applies a lowpass filter if necessary.
`void`	`set8SVXBody(byte[] value)`
`protected void`	`setAuthor(String value)`
`protected void`	`setCopyright(String value)`
`void`	`setCtOctave(int value)`
`protected void`	`setName(String value)`
`void`	`setOneShotHiSamples(long value)`
`protected void`	`setRemark(String value)`
`void`	`setRepeatHiSamples(long value)`
`void`	`setSampleRate(int value)`
`void`	`setSamplesPerHiCycle(long value)`
`void`	`setSampleType(int value)`
`void`	`setSCompression(int value)`
`void`	`setVolume(int value)`
`void`	`stop()`
`String`	`toString()`
`static byte[]`	`unpackFibonacciDeltaCompression(byte[] source)` Unpack Fibonacci-delta encoded data from n byte source buffer into 2*(n-2) byte dest buffer.
`static void`	`writeSunAudioHeader(OutputStream outfile, int dataSize, int sampleRate, int sampleType)` Write a "standard" sun header.
`static void`	`wrulong(OutputStream outfile, int ulong)` Write an unsigned long (Motorola 68000 CPU format).

Methods inherited from class java.lang.Object
clone, equals, finalize, getClass, hashCode, notify, notifyAll, wait, wait, wait

- Field Detail
  - S_CMP_NONE
```
public static final int S_CMP_NONE
```
    See Also:
    Constant Field Values
  - S_CMP_FIB_DELTA
```
public static final int S_CMP_FIB_DELTA
```
    See Also:
    Constant Field Values
  - RIGHT
```
public static final int RIGHT
```
    See Also:
    Constant Field Values
  - LEFT
```
public static final int LEFT
```
    See Also:
    Constant Field Values
  - STEREO
```
public static final int STEREO
```
    See Also:
    Constant Field Values
- Constructor Detail
  - EightSVXAudioClip
```
public EightSVXAudioClip()
```
- Method Detail
  - setName
```
protected void setName(String value)
```
  - getName
```
protected String getName()
```
  - setAuthor
```
protected void setAuthor(String value)
```
  - getAuthor
```
protected String getAuthor()
```
  - setCopyright
```
protected void setCopyright(String value)
```
  - getCopyright
```
protected String getCopyright()
```
  - setRemark
```
protected void setRemark(String value)
```
  - getRemark
```
protected String getRemark()
```
  - set8SVXBody
```
public void set8SVXBody(byte[] value)
```
  - get8SVXBody
```
public byte[] get8SVXBody()
```
  - setOneShotHiSamples
```
public void setOneShotHiSamples(long value)
```
  - setRepeatHiSamples
```
public void setRepeatHiSamples(long value)
```
  - setSamplesPerHiCycle
```
public void setSamplesPerHiCycle(long value)
```
  - setSampleType
```
public void setSampleType(int value)
```
  - setSampleRate
```
public void setSampleRate(int value)
```
  - setCtOctave
```
public void setCtOctave(int value)
```
  - setSCompression
```
public void setSCompression(int value)
```
  - setVolume
```
public void setVolume(int value)
```
  - getOneShotHiSamples
```
public long getOneShotHiSamples()
```
  - getRepeatHiSamples
```
public long getRepeatHiSamples()
```
  - getSamplesPerHiCycle
```
public long getSamplesPerHiCycle()
```
  - getSampleType
```
public long getSampleType()
```
  - getSampleRate
```
public int getSampleRate()
```
  - getCtOctave
```
public int getCtOctave()
```
  - getVolume
```
public int getVolume()
```
  - getSCompression
```
public int getSCompression()
```
  - toString
```
public String toString()
```
    Overrides:
    
    toString in class Object
  - createAudioClip
```
public LoopableAudioClip createAudioClip()
```
  - createAudioClip
```
public LoopableAudioClip createAudioClip(int sampleRate,
                                int volume,
                                float pan)
```
  - createJDK13AudioClip
```
public LoopableAudioClip createJDK13AudioClip(int sampleRate,
                                     int volume,
                                     float pan)
```
  - createJDK10AudioClip
```
public LoopableAudioClip createJDK10AudioClip(int sampleRate)
```
  - play
```
public void play()
```
    Specified by:
    
    play in interface AudioClip
  - loop
```
public void loop()
```
    Specified by:
    
    loop in interface AudioClip
  - stop
```
public void stop()
```
    Specified by:
    
    stop in interface AudioClip
  - prepare
```
public void prepare()
```
    Make this clip ready for playback.
  - computeStereoVolumeCorrection
```
public static double computeStereoVolumeCorrection(byte[] stereo)
```
    This finds the volume correction needed when converting this stereo sample to mono.
    
    Parameters:
    stereo - Stereo data linear 8. The first half of the array contains the sound for the left speaker, the second half the sound for the right speaker.
    
    Returns:
    volumeCorrection Combining the two channels into one increases the sound volume. This can exceed the maximum volume that can be represented by the linear8 sample model. To avoid this, the volume must be corrected to fit into the sample model.
  - linear8StereoToMono
```
public static byte[] linear8StereoToMono(byte[] stereo,
                         double volumeCorrection)
```
    This converts a stereo sample to mono.
    
    Parameters:
    stereo - Stereo data linear 8. The first half of the array contains the sound for the left speaker, the second half the sound for the right speaker.
    volumeCorrection - Combining the two channels into one increases the sound volume. This can exceed the maximum volume that can be represented by the linear8 sample model. To avoid this, the volume must be corrected to fit into the sample model.
  - resample
```
public static byte[] resample(byte[] input,
              int inputSampleRate,
              int outputSampleRate)
```
    Resamples audio data to match the given sample rate and applies a lowpass filter if necessary.
    
    Parameters:
    input - Linear8 encoded audio data.
    inputSampleRate - The sample rate of the input data
    outputSampleRate - The sample rate of the output data.
    
    Returns:
    Linear8 encoded audio data.
  - linear8ToULaw
```
public static byte[] linear8ToULaw(byte[] linear8)
```
    Halfband lowpass. / public static byte[] halfbandLowpass(byte[] input) { /* Algorithm taken from "Yehar's digital sound processing tutorial for the braindead!" version 2001.05.02 http://www.student.oulu.fi/~oniemita/DSP/DSPSTUFF.TXT Halfband lowpass *** b1 = 0.641339 b10 = -0.0227141 b2 = -3.02936 b3 = 1.65298 a0a12 = 0.008097 b4 = -3.4186 a1a11 = 0.048141 b5 = 1.50021 a2a10 = 0.159244 b6 = -1.73656 a3a9 = 0.365604 b7 = 0.554138 a4a8 = 0.636780 b8 = -0.371742 a5a7 = 0.876793 b9 = 0.0671787 a6 = 0.973529 output(t) = a0a12*(input(t ) + input(t-12)) + a1a11*(input(t-1) + input(t-11)) + a2a10*(input(t-2) + input(t-10)) + a3a9* (input(t-3) + input(t-9 )) + a4a8* (input(t-4) + input(t-8 )) + a5a7* (input(t-5) + input(t-7 )) + a6*input(t-6) + b1*output(t-1) + b2*output(t-2) + b3*output(t-3) + b4*output(t-4) + b5*output(t-5) + b6*output(t-6) + b7*output(t-7) + b8*output(t-8) + b9*output(t-9) + b10*output(t-10) / double b1 = 0.641339, b10 = -0.0227141, b2 = -3.02936, b3 = 1.65298, a0a12 = 0.008097, b4 = -3.4186 , a1a11 = 0.048141, b5 = 1.50021, a2a10 = 0.159244, b6 = -1.73656, a3a9 = 0.365604, b7 = 0.554138, a4a8 = 0.636780, b8 = -0.371742 , a5a7 = 0.876793, b9 = 0.0671787 , a6 = 0.973529; // Create the output array byte[] output = new byte[input.length]; for (int t=12; t < input.length; t++) { output[t] = (byte) ( a0a12*(input[t] + input[t-12]) + a1a11*(input[t-1] + input[t-11]) + a2a10*(input[t-2] + input[t-10]) + a3a9* (input[t-3] + input[t-9 ]) + a4a8* (input[t-4] + input[t-8 ]) + a5a7* (input[t-5] + input[t-7 ]) + a6*input[t-6] + b1*output[t-1] + b2*output[t-2] + b3*output[t-3] + b4*output[t-4] + b5*output[t-5] + b6*output[t-6] + b7*output[t-7] + b8*output[t-8] + b9*output[t-9] + b10*output[t-10] ); } return output; } /** Converts a buffer of signed 8bit samples to uLaw. The uLaw bytes overwrite the original 8 bit values. The first byte-offset of the uLaw bytes is byteOffset. It will be written sampleCount bytes.
  - linear16ToULaw
```
public static byte[] linear16ToULaw(int[] linear16)
```
    Converts a buffer of signed 8bit samples to uLaw. The uLaw bytes overwrite the original 8 bit values. The first byte-offset of the uLaw bytes is byteOffset. It will be written sampleCount bytes.
  - unpackFibonacciDeltaCompression
```
public static byte[] unpackFibonacciDeltaCompression(byte[] source)
```
    Unpack Fibonacci-delta encoded data from n byte source buffer into 2*(n-2) byte dest buffer. Source buffer has a pad byte, an 8-bit initial value, followed by n bytes comprising 2*(n) 4-bit encoded samples.
  - writeSunAudioHeader
```
public static void writeSunAudioHeader(OutputStream outfile,
                       int dataSize,
                       int sampleRate,
                       int sampleType)
                                throws IOException
```
    Write a "standard" sun header.
    
    Parameters:
    sampleType - Specify STEREO, LEFT or RIGHT.
    
    Throws:
    
    IOException
  - wrulong
```
public static void wrulong(OutputStream outfile,
           int ulong)
                    throws IOException
```
    Write an unsigned long (Motorola 68000 CPU format).
    
    Throws:
    
    IOException
  - loop
```
public void loop(int count)
```
    Starts looping playback from the current position. Playback will continue to the loop's end point, then loop back to the loop start point count times, and finally continue playback to the end of the clip.
    If the current position when this method is invoked is greater than the loop end point, playback simply continues to the end of the clip without looping.
    A count value of 0 indicates that any current looping should cease and playback should continue to the end of the clip. The behavior is undefined when this method is invoked with any other value during a loop operation.
    If playback is stopped during looping, the current loop status is cleared; the behavior of subsequent loop and start requests is not affected by an interrupted loop operation.
    
    Specified by:
    
    loop in interface LoopableAudioClip
    
    Parameters:
    count - the number of times playback should loop back from the loop's end position to the loop's start position, or LoopableAudioClip.LOOP_CONTINUOUSLY to indicate that looping should continue until interrupted

Class EightSVXAudioClip

Field Summary

Fields inherited from interface org.monte.media.eightsvx.LoopableAudioClip

Constructor Summary

Method Summary

Methods inherited from class java.lang.Object

Field Detail

S_CMP_NONE

S_CMP_FIB_DELTA

RIGHT

LEFT

STEREO

Constructor Detail

EightSVXAudioClip

Method Detail

setName

getName

setAuthor

getAuthor

setCopyright

getCopyright

setRemark

getRemark

set8SVXBody

get8SVXBody

setOneShotHiSamples

setRepeatHiSamples

setSamplesPerHiCycle

setSampleType

setSampleRate

setCtOctave

setSCompression

setVolume

getOneShotHiSamples

getRepeatHiSamples

getSamplesPerHiCycle

getSampleType

getSampleRate

getCtOctave

getVolume

getSCompression

toString

createAudioClip

createAudioClip

createJDK13AudioClip

createJDK10AudioClip

play

loop

stop

prepare

computeStereoVolumeCorrection

linear8StereoToMono

resample

linear8ToULaw

linear16ToULaw

unpackFibonacciDeltaCompression

writeSunAudioHeader

wrulong

loop