WSS File Format: Difference between revisions

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==Introduction==
==Introduction==
WSS files are used since OFP times to store sound data and its file format is pretty much like the [http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html RIFF WAVE file format] used for .wav files.  
WSS files are used since OFP times to store sound data and its file format is pretty much like the [http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html RIFF WAVE file format] used for .wav files.  
For the purposes of description here, 'Sound' in both wav or wss is recorded as PCM data in 16 bit 'samples'. The quality of that sound is determined by the frequency of those 'samples' per second, referred to as the SampleRate.
Given this: a few important paramaters are define here:
*BitsPerSample: 16
*BytesPerSample: 2 (Inferred BitsPerSample/8)


==File Format==
==File Format==
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|4
|4
|ulong
|ulong
|<compression>
|CompressionType
|if <compression> == 8 the PCM data is compressed, otherwise <compression> == 0
|0 == none. 8== compressed PCM data
|-
|-
|8
|8
|ushort
|ushort
|<format tag>
|format
|defines in which format the data is saved. Always 0x0001 in WSS files
|Always 1 (WAVE_FORMAT_PCM)
|-
|-
|10
|10
|ushort
|ushort
|<channels>
|nChannels
|number of channels: 1=mono, 2=stereo
|1=mono, 2=stereo
|-
|-
|12
|12
|ulong
|ulong
|<sample rate>
|SampleRate
|sample rate in Hz (e.g. 44100Hz)
|e.g. 44100Hz
|-
|-
|16
|16
|ulong
|ulong
|<bytes/second>
|BytesPerSecond
|<sample rate> * <block align>
|SampleRate * BlockAlign
|-
|-
|20
|20
|ushort
|ushort
|<block align>
|BlockAlign
|<channels> * (<bits/sample> / 8)
|nChannels * BytesPerSample
|-
|-
|22
|22
|ushort
|ushort
|<bits/sample>
|BitsPerSecond
|usually 0x0010 in WSS files
|usually 16
|-
|-
|24
|24
Line 65: Line 73:




Compression consists of a single encoded bytes versus uncompressed shorts.
Compression consists of single encoded byte 'samples' versus uncompressed short 'samples'.
Each byte is, effectively, extrapolated to a short, thus making the compressed BYTE array, and the resulting decompressed SHORT array the same number of elements(length). The length is the remaining file length (in bytes) after the header.
Each byte is, effectively, extrapolated to a short, thus making the compressed BYTE array, and the resulting decompressed SHORT array the same number of elements(length). The length is the remaining file length (in bytes) after the header.



Revision as of 07:02, 29 January 2010

Introduction

WSS files are used since OFP times to store sound data and its file format is pretty much like the RIFF WAVE file format used for .wav files.

For the purposes of description here, 'Sound' in both wav or wss is recorded as PCM data in 16 bit 'samples'. The quality of that sound is determined by the frequency of those 'samples' per second, referred to as the SampleRate.

Given this: a few important paramaters are define here:

  • BitsPerSample: 16
  • BytesPerSample: 2 (Inferred BitsPerSample/8)


File Format

Offset Datatype Content Description
0 char[4] "WSS0" file signature
4 ulong CompressionType 0 == none. 8== compressed PCM data
8 ushort format Always 1 (WAVE_FORMAT_PCM)
10 ushort nChannels 1=mono, 2=stereo
12 ulong SampleRate e.g. 44100Hz
16 ulong BytesPerSecond SampleRate * BlockAlign
20 ushort BlockAlign nChannels * BytesPerSample
22 ushort BitsPerSecond usually 16
24 ushort <unknown> unknown value
26 byte[fileSize-26] <soundData> here the PCM data of the sound is stored

Decompression

Compression consists of single encoded byte 'samples' versus uncompressed short 'samples'. Each byte is, effectively, extrapolated to a short, thus making the compressed BYTE array, and the resulting decompressed SHORT array the same number of elements(length). The length is the remaining file length (in bytes) after the header.

C++ code

Function returns a short array, same size as it's compressed byte equivalent

#define LOG10	2.3025850929940456840	//ln(10)
#define LOG2	1.4426950408889634070	//log2(e)
#define MAGIC_NUMBER	((LOG10*LOG2)/28.12574042515172)

short* DeCompress(const char *CompressedData,int len)
{
short *snap,*OutputData;

	if (!(snap=OutputData=new short[len]))  return 0;
	short LastVal=0;
	for (;len--;CompressedData++)
	{
		if (*CompressedData)
		{
			double asFloat = abs(*CompressedData) *MAGIC_NUMBER;
			double rnd = Round(asFloat);
			asFloat = pow(2.0, asFloat - rnd) * pow(2, rnd);// mantissa -
			if (*CompressedData < 0) asFloat *= -1;
			int asInt = Round(asFloat)+LastVal;
			if (asInt > SHRT_MAX ) asInt = SHRT_MAX ;
			if (asInt < SHRT_MIN) asInt = SHRT_MIN;
			LastVal=(short)asInt;
		 }
		*OutputData++ = LastVal;
	}
	return snap;
}

C# code

If the <soundData> is compressed the following (C#) code can be used for decompression:

PCMData = new Int16[soundData.Length];
for (int j = 0; j < PCMData.Length; j++)
{
  SByte srcSample = (SByte)soundData[j];
  if (srcSample != 0)
  {
    double asFloat = Math.Abs(srcSample) / 28.12574042515172;
    asFloat *= 2.3025850929940456840; //ln(10)
    asFloat *= 1.4426950408889634070; //log2(e)
    double rnd = Math.Round(asFloat);
    double mantisse = Math.Pow(2.0, asFloat - rnd);
    asFloat = mantisse * Math.Pow(2, rnd);
    if (srcSample < 0) asFloat *= -1;
    Int32 asInt = (int)Math.Round(asFloat);
    asInt = (j == 0) ? asInt : (asInt + PCMData[j - 1]);
    if (asInt > short.MaxValue) asInt = short.MaxValue;
    if (asInt < short.MinValue) asInt = short.MinValue;
    PCMData[j] = (Int16)asInt;
  }
  else PCMData[j] = (j == 0) ? (Int16)0 : PCMData[j - 1];
}