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| − | {{OldWikiEntry}} Maxis eXtendable Audio format from The Sims and SimCity 3000/4. Can use Bil Simser's [[http:''www.simstools.com/xantippe.php Xantippe]] to convert to WAV format.
| + | #REDIRECT [[2026960B]] |
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| − | | + | [[Category:Modding]] |
| − | | + | [[Category:InternalFormats]] |
| − | --------------------------------------
| + | [[Category:FormatsByName]] |
| − | Maxis XA Audio File Format Description 5-01-2002
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| − | --------------------------------------
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| − | | + | |
| − | By Valery V. Anisimovsky (samael@avn.mccme.ru)
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| − | | + | |
| − | In this document I'll try to describe audio file format used in some Maxis
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| − | games, in particlular, SimCity3000 (perhaps, in some other Maxis games) for
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| − | music, speech and sfx files.
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| − | | + | |
| − | The files this document deals with have extension: .XA. Note that the
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| − | extension of audio files of this format may be different from that.
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| − | | + | |
| − | Throughout this document I use C-like notation.
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| − | All numbers in all structures described in this document are stored in files
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| − | using little-endian (Intel) byte order.
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| − | <pre>
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| − | ======
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| − | 1. XA File Header
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| − | ======
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| − | | + | |
| − | The XA file has the following header:
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| − | struct XAHeader
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| − | {
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| − | char szID[4];
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| − | DWORD dwOutSize;
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| − | WORD wTag;
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| − | WORD wChannels;
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| − | DWORD dwSampleRate;
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| − | DWORD dwAvgByteRate;
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| − | WORD wAlign;
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| − | WORD wBits;
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| − | };
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| − | szID -- string ID, which is equal to "XAI\0" (sound/speech) or "XAJ\0" (music).
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| − | | + | |
| − | dwOutSize -- the output size of the audio stream stored in the file (in bytes).
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| − | wTag -- seems to be PCM waveformat tag (0x0001). This corresponds to the
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| − | (decompressed) output audio stream, of course.
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| − | | + | |
| − | wChannels -- number of channels for the file.
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| − | dwSampleRate -- sample rate for the file.
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| − | | + | |
| − | dwAvgByteRate -- average byte rate for the file (equal to (dwSampleRate)*(wAlign)).
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| − | Note that this also corresponds to the decompressed output audio stream.
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| − | | + | |
| − | wAlign -- the sample align value for the file (equal to (wBits/8)*(wChannels)).
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| − | Again, this corresponds to the decompressed output audio stream.
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| − | | + | |
| − | wBits -- resolution of the file (8 (8-bit), 16 (16-bit), etc.).
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| − | | + | |
| − | Note that the part of the header from (wTag) until (wBits) is really
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| − | WAVEFORMATEX structure (the contents of PCM .WAV fmt chunk).
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| − | | + | |
| − | =====
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| − | 2. XA File Data
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| − | =====
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| − | | + | |
| − | Right after the XA header comes the compressed audio stream. The compression
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| − | algorithm used is EA ADPCM (see below). Music files in SimCity3000 are
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| − | stereo 22050 Hz 16-bit, and speech/sfx are mono 22050 Hz 16-bit.
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| − | | + | |
| − | ============
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| − | 3. EA ADPCM Decompression Algorithm
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| − | ============
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| − | During the decompression four LONG variables must be maintained for stereo
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| − | stream: lCurSampleLeft, lCurSampleRight, lPrevSampleLeft, lPrevSampleRight
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| − | and two -- for mono stream: lCurSample, lPrevSample. At the beginning of the
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| − | audio stream you must initialize these variables to zeros.
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| − | Note that LONG here is signed.
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| − | | + | |
| − | The stream is divided into small blocks of 0x1E (stereo) or 0xF (mono) bytes.
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| − | You should process all blocks in their turn. Here's the code which
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| − | decompresses one stereo stream block.
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| − | BYTE InputBuffer[InputBufferSize]; '' buffer containing data for one block
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| − | BYTE bInput;
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| − | DWORD i;
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| − | LONG c1left,c2left,c1right,c2right,left,right;
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| − | BYTE dleft,dright;
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| − | bInput=InputBuffer[0];
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| − | c1left=EATable[HINIBBLE(bInput)]; '' predictor coeffs for left channel
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| − | c2left=EATable[HINIBBLE(bInput)+4];
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| − | dleft=LONIBBLE(bInput)+8; '' shift value for left channel
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| − | | + | |
| − | bInput=InputBuffer[1];
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| − | c1right=EATable[HINIBBLE(bInput)]; '' predictor coeffs for right channel
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| − | c2right=EATable[HINIBBLE(bInput)+4];
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| − | dright=LONIBBLE(bInput)+8; '' shift value for right channel
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| − | | + | |
| − | for (i=2;i<0x1E;i+=2)
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| − | {
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| − | left=HINIBBLE(InputBuffer[i]); '' HIGHER nibble for left channel
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| − | left=(left<<0x1c)>>dleft;
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| − | left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
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| − | left=Clip16BitSample(left);
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| − | lPrevSampleLeft=lCurSampleLeft;
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| − | lCurSampleLeft=left;
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| − | right=HINIBBLE(InputBuffer[i+1]); '' HIGHER nibble for right channel
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| − | right=(right<<0x1c)>>dright;
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| − | right=(right+lCurSampleRight*c1right+lPrevSampleRight*c2right+0x80)>>8;
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| − | right=Clip16BitSample(right);
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| − | lPrevSampleRight=lCurSampleRight;
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| − | lCurSampleRight=right;
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| − | | + | |
| − | '' Now we've got lCurSampleLeft and lCurSampleRight which form one stereo
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| − | '' sample and all is set for the next step...
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| − | Output((SHORT)lCurSampleLeft,(SHORT)lCurSampleRight); '' send the sample to output
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| − | '' now do just the same for LOWER nibbles...
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| − | '' note that nubbles for each channel are packed pairwise into one byte
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| − | left=LONIBBLE(InputBuffer[i]); '' LOWER nibble for left channel
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| − | left=(left<<0x1c)>>dleft;
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| − | left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
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| − | left=Clip16BitSample(left);
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| − | lPrevSampleLeft=lCurSampleLeft;
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| − | lCurSampleLeft=left;
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| − | right=LONIBBLE(InputBuffer[i+1]); '' LOWER nibble for right channel
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| − | right=(right<<0x1c)>>dright;
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| − | right=(right+lCurSampleRight*c1right+lPrevSampleRight*c2right+0x80)>>8;
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| − | right=Clip16BitSample(right);
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| − | lPrevSampleRight=lCurSampleRight;
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| − | lCurSampleRight=right;
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| − | | + | |
| − | '' Now we've got lCurSampleLeft and lCurSampleRight which form one stereo
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| − | '' sample and all is set for the next step...
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| − | Output((SHORT)lCurSampleLeft,(SHORT)lCurSampleRight); '' send the sample to output
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| − | }
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| − | HINIBBLE and LONIBBLE are higher and lower 4-bit nibbles:
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| − | #define HINIBBLE(byte) ((byte) >> 4)
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| − | #define LONIBBLE(byte) ((byte) & 0x0F)
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| − | Note that depending on your compiler you may need to use additional nibble
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| − | separation in these defines, e.g. (((byte) >> 4) & 0x0F).
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| − | EATable is the table given in the next section of this document.
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| − | Output() is just a placeholder for any action you would like to perform for
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| − | decompressed sample value.
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| − | Clip16BitSample is quite evident:
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| − | LONG Clip16BitSample(LONG sample)
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| − | {
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| − | if (sample>32767)
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| − | return 32767;
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| − | else if (sample<-32768)
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| − | return (-32768);
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| − | else
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| − | return sample;
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| − | }
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| − | As to mono sound, it's just analoguous -- you should process the blocks each
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| − | being 0xF bytes long:
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| − | bInput=InputBuffer[0];
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| − | c1=EATable[HINIBBLE(bInput)]; '' predictor coeffs
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| − | c2=EATable[HINIBBLE(bInput)+4];
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| − | d=LONIBBLE(bInput)+8; '' shift value
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| − | | + | |
| − | for (i=1;i<0xF;i++)
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| − | {
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| − | left=HINIBBLE(InputBuffer[i]); '' HIGHER nibble for left channel
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| − | left=(left<<0x1c)>>dleft;
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| − | left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
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| − | left=Clip16BitSample(left);
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| − | lPrevSampleLeft=lCurSampleLeft;
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| − | lCurSampleLeft=left;
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| − | '' Now we've got lCurSampleLeft which is one mono sample and all is set
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| − | '' for the next input nibble...
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| − | Output((SHORT)lCurSampleLeft); '' send the sample to output
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| − | left=LONIBBLE(InputBuffer[i]); '' LOWER nibble for left channel
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| − | left=(left<<0x1c)>>dleft;
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| − | left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
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| − | left=Clip16BitSample(left);
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| − | lPrevSampleLeft=lCurSampleLeft;
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| − | lCurSampleLeft=left;
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| − | '' Now we've got lCurSampleLeft which is one mono sample and all is set
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| − | '' for the next input byte...
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| − | Output((SHORT)lCurSampleLeft); '' send the sample to output
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| − | }
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| − | So, you should process HIGHER nibble of the input byte first and then
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| − | LOWER nibble for mono sound.
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| − | Of course, this decompression routine may be greatly optimized.
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| − | | + | |
| − | ======
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| − | 4. EA ADPCM Table
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| − | ======
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| − | LONG EATable[]=
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| − | {
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| − | 0x00000000,
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| − | 0x000000F0,
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| − | 0x000001CC,
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| − | 0x00000188,
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| − | 0x00000000,
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| − | 0x00000000,
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| − | 0xFFFFFF30,
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| − | 0xFFFFFF24,
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| − | 0x00000000,
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| − | 0x00000001,
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| − | 0x00000003,
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| − | 0x00000004,
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| − | 0x00000007,
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| − | 0x00000008,
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| − | 0x0000000A,
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| − | 0x0000000B,
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| − | 0x00000000,
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| − | 0xFFFFFFFF,
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| − | 0xFFFFFFFD,
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| − | 0xFFFFFFFC
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| − | };
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| − | </pre>
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| − | -----------
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| − | 5. Credits
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| − | -----------
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| − | Dmitry Kirnocenskij (ejt@mail.ru)
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| − | Worked out EA ADPCM decompression algorithm.
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| − | Nicholas Sales (nicsales@mweb.co.za)
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| − | Provided me with SimCity3000 decoding stuff thereby inspired
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| − | me to decode the formats and write the plug-in for GAP.
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| − | | + | |
| − | -------------------------------------------
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| − | | + | |
| − | Valery V. Anisimovsky (samael@avn.mccme.ru)
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| − | http:''bim.km.ru/gap/
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| − | http:''www.anxsoft.newmail.ru
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| − | http:''anx.da.ru
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| − | On these sites you can find my GAP program which can search for XA audio
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| − | files in game resources, extract them, convert them to WAV and play them back.
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| − | There's also complete source code of GAP and all its plug-ins there,
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| − | including XA plug-in, which could be used for further details on how you
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| − | can deal with this format.
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| − | [[Category:Modding]]
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