MAD
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MAD video is the format of Simcity4's videos. Not really anything else to say about it. Its taken us a long time to really get started on this though, but here we are
There are video and audio blocks The video is EA specific type The audio is EA ASF STR Audio The whole format is really a one block video one block audio repeat.
It has the following structure
------------------------------
Video chunks - Hayly,Darkmatter,Karybdis
------------------------------
Header:
DWORD - MAD* - Identifier of chunk where * is a unique chunk type identifier :
DWORD - Size of the frame chunk
DWORD - Unknown (0)
WORD - Max Data rate per Sec. (ABAA normally. Max bytes needed to present the video properly per sec)
WORD - Delay MicroSec Per Frame (standard is 66ms or 15 frames a second.) (indicates file timing)
WORD - Pixel width of the frame chunk
WORD - Pixel height of the frame chunk
BYTE - Unknown
BYTE - Unknown
BYTE - In TS2 movies, chunk type. 02 = B-frame, 00 = P-frame, 01 = I-frame. In SC4 movies, blank
BYTE - Unknown (0)
--MAD TYPES
---k = Intra Coded Keyframes (I-Frame) (estimated size, 30 byte block)
---m = Predictive coded (previous) frames (P-Frame) (estimated 18 byte block)
---e = Bidirectionally-predictive coded (both previous and following) frames (B-frame) (estimated 18 byte block)
-----Note: E blocks are only present in TS2 movies and not SC4
SC4 uses a KMMMM KMMMM format with a keyframe every 5 frames
TS2 uses a KEMEMEMEMEMEME KEMEMEMEMEMEME format with a keyframe every 14 frames
----
Video Data - Each video chunk has readable data blocks with a size dependant on its identifier (k,m,e). The codec breakdown is unknown, so we cant view the movies yet
========
1. .ASF/.STR Music Files (MAD Audio chunks) By Valery V. Anisimovsky (samael@avn.mccme.ru)
========
The music in many new Electronic Arts games is in .ASF stand-alone files
(sometimes ASF files have extension .STR). These files have the block
structure analoguous to RIFF. Namely, these files are divided into blocks
(without any global file header like RIFFs have). Each block has the
following header:
struct ASFBlockHeader
{
char szBlockID[4];
DWORD dwSize;
};
szBlockID -- string ID for the block.
dwSize -- size of the block (in bytes) INCLUDING this header.
Further I'll describe the contents of blocks of all block types in .ASF file.
When I say "block begins with..." that means "the contents of that
block (which begin just after ASFBlockHeader) begin with...".
Quoted strings are block IDs.
"SCHl": header block. This is the first block in ASF.
In the most of files this block begins with the ID string "PT\0\0" (or number
0x50540000). Further goes the PT header data which describes audio data in
the file. This PT header should be parsed rather than just read as a simple
structure. Here I give the parsing code. These functions use fread() and fseek()
stdio functions.
'' first of all, we need a function which reads a small (variable) number
'' bytes and composes a DWORD of them. Note that such DWORD will be a kind
'' of big-endian (Motorola) stored, e.g. 3 consecutive bytes 0x12 0x34 0x56
'' will give a DWORD 0x00123456.
DWORD ReadBytes(FILE* file, BYTE count)
{
BYTE i, byte;
DWORD result;
result=0L;
for (i=0;i<count;i++)
{
fread(&byte,sizeof(BYTE),1,file);
result<<=8;
result+=byte;
}
return result;
}
'' these will be set by ParsePTHeader
DWORD dwSampleRate;
DWORD dwChannels;
DWORD dwCompression;
DWORD dwNumSamples;
DWORD dwDataStart;
DWORD dwLoopOffset;
DWORD dwLoopLength;
DWORD dwBytesPerSample;
BYTE bSplit;
BYTE bSplitCompression;
'' Here goes the parser itself
'' This function assumes that the current file pointer is set to the
'' start of PT header data, that is, just after PT string ID "PT\0\0"
void ParsePTHeader(FILE* file)
{
BYTE byte;
BOOL bInHeader, bInSubHeader;
bInHeader=TRUE;
while (bInHeader)
{
fread(&byte,sizeof(BYTE),1,file);
switch (byte) '' parse header code
{
case 0xFF: '' end of header
bInHeader=FALSE;
case 0xFE: '' skip
case 0xFC: '' skip
break;
case 0xFD: '' subheader starts...
bInSubHeader=TRUE;
while (bInSubHeader)
{
fread(&byte,sizeof(BYTE),1,file);
switch (byte) '' parse subheader code
{
case 0x82:
fread(&byte,sizeof(BYTE),1,file);
dwChannels=ReadBytes(file,byte);
break;
case 0x83:
fread(&byte,sizeof(BYTE),1,file);
dwCompression=ReadBytes(file,byte);
break;
case 0x84:
fread(&byte,sizeof(BYTE),1,file);
dwSampleRate=ReadBytes(file,byte);
break;
case 0x85:
fread(&byte,sizeof(BYTE),1,file);
dwNumSamples=ReadBytes(file,byte);
break;
case 0x86:
fread(&byte,sizeof(BYTE),1,file);
dwLoopOffset=ReadBytes(file,byte);
break;
case 0x87:
fread(&byte,sizeof(BYTE),1,file);
dwLoopLength=ReadBytes(file,byte);
break;
case 0x88:
fread(&byte,sizeof(BYTE),1,file);
dwDataStart=ReadBytes(file,byte);
break;
case 0x92:
fread(&byte,sizeof(BYTE),1,file);
dwBytesPerSample=ReadBytes(file,byte);
break;
case 0x80: '' ???
fread(&byte,sizeof(BYTE),1,file);
bSplit=ReadBytes(file,byte);
break;
case 0xA0: '' ???
fread(&byte,sizeof(BYTE),1,file);
bSplitCompression=ReadBytes(file,byte);
break;
case 0xFF:
subflag=FALSE;
flag=FALSE;
break;
case 0x8A: '' end of subheader
bInSubHeader=FALSE;
default: '' ???
fread(&byte,sizeof(BYTE),1,file);
fseek(file,byte,SEEK_CUR);
}
}
break;
default:
fread(&byte,sizeof(BYTE),1,file);
if (byte==0xFF)
fseek(file,4,SEEK_CUR);
fseek(file,byte,SEEK_CUR);
}
}
}
dwSampleRate -- sample rate for the file. Note that headers of most of
ASFs/MUSes I've seen DO NOT contain sample rate subheader section. Currently
I just set sample rate for such files to the default: 22050 Hz. It seems to
work okay.
dwChannels -- number of channels for the file: 1 for mono, 2 for stereo.
If this is NOT set by ParsePTHeader, then you may use the default: stereo.
dwCompression -- Compression tag. If this is 0x00, then no compression is
used and audio data is signed 16-bit PCM. If this is 0x07, the audio data is
compressed with EA ADPCM algorithm. Please read the next section for the
description of EA ADPCM decompression scheme. In some files this tag is
omitted -- I use 0x00 (no compression) for them.
dwNumSamples -- number of samples in the file.
dwDataStart -- in ASF files this's not used.
dwLoopOffset -- offset when looping (from start of sound part).
dwLoopLength -- length when looping.
dwBytesPerSample -- bytes per sample (Default is 2). Divide this by
dwChannels to get resolution of sound data.
bSplit -- this looks like to be 0x01 for files using "split" SCDl blocks
(see below). If this subheader field is absent, the file uses "normal"
(interleaved) SCDl blocks.
bSplitCompression -- this looks like to be 0x08 for files using non-compressed
"split" SCDl blocks. If this subheader field is absent in the file using
"split" SCDls, the file uses EA ADPCM compression. This subheader field
should not appear in a file using "normal" (interleaved) SCDls.
"SCCl": count block. This block goes after "SCHl" and contains one DWORD
value which is a number of "SCDl" data blocks in ASF file.
"SCDl": data block. These blocks contain audio data. Depending on the
parameters set in the header (see above) SCDl block may contain compressed
(by EA ADPCM or IMA ADPCM) or non-compressed audio data and the data itself
may be interleaved or split (see below).
If no compression is used and the file does not use "split" SCDl blocks,
SCDl block begins with a DWORD value which is the number of samples in this
block and after that comes signed 16-bit PCM data, in the interleaved form:
LRLR...LR (L and R are 16-bit sample values for left and right channels).
Hereafter by "chunk" I mean the audio data in the "SCDl" data block, that is,
compressed/non-compressed data which starts after chunk header.
In the newer EA games (NHL'2000/NBA'2000/FIFA'99'2000/NFS5) non-compressed
"split" SCDl blocks are used. These blocks begin with a chunk header:
struct ASFSplitPCMChunkHeader
{
DWORD dwOutSize;
DWORD dwLeftChannelOffset;
DWORD dwRightChannelOffset;
}
dwOutSize -- size of audio data in this chunk (in samples).
dwLeftChannelOffset, dwRightChannelOffset -- offsets to PCM data for
left and right channels, relative to the byte which immediately follows
ASFSplitPCMChunkHeader structure. E.g. for left channel this offset is zero
-- the data starts immediately after this structure.
After this structure comes PCM data for stereo wavestream and it's not
interleaved (LRLRLR...), but it's split: first go sample values for left
channel, then -- for right channel, that is the layout is LL...LRR...R.
If EA ADPCM (or IMA ADPCM) compression is used, but the file does not use
"split" SCDls, each SCDl block begins with a chunk header:
struct ASFChunkHeader
{
DWORD dwOutSize;
SHORT lCurSampleLeft;
SHORT lPrevSampleLeft;
SHORT lCurSampleRight;
SHORT lPrevSampleRight;
};
dwOutSize -- size of decompressed audio data in this chunk (in samples).
lCurSampleLeft, lCurSampleRight, lPrevSampleLeft, lPrevSampleRight are initial
values for EA ADPCM decompression routine for this data block (for left and right
channels respectively). I'll describe the usage of these further when I get to
EA ADPCM decompression scheme.
Note that the structure above is ONLY for stereo files. For mono there're
just no lCurSampleRight, lPrevSampleRight fields.
If IMA ADPCM compression is used, the meanings of some chunk header fields
are different -- see my EA-ASF.TXT specs for details.
After this chunk header the compressed data comes. See the next section for
EA ADPCM decompression scheme description.
If EA ADPCM (or IMA ADPCM) compression is used and the file uses "split" SCDls,
each SCDl block begins with a different chunk header:
struct ASFSplitChunkHeader
{
DWORD dwOutSize;
DWORD dwLeftChannelOffset;
DWORD dwRightChannelOffset;
};
SHORT lCurSampleLeft;
SHORT lPrevSampleLeft;
BYTE bLeftChannelData[]; '' compressed data for left channel goes here...
SHORT lCurSampleRight;
SHORT lPrevSampleRight;
BYTE bRightChannelData[]; '' compressed data for right channel goes here...
dwOutSize -- size of decompressed audio data in this chunk (in samples).
dwLeftChannelOffset, dwRightChannelOffset -- offsets to compressed data for
left and right channels, relative to the byte which immediately follows
ASFSplitChunkHeader structure. E.g. for left channel this offset is zero --
the data starts immediately after this structure.
lCurSampleLeft, lCurSampleRight, lPrevSampleLeft, lPrevSampleRight have the
same meaning as above, but note that these values are SHORTs.
So, use mono decoder for each channel data and then create normal LRLR...
stereo waveform before outputting.
Such (newer) files may be separated from the others by presence of 0x80 type
section in PT header (the value stored in the section is 0x01 for such files).
Some of such files also do not contain compression type (0x83) section in
their PT header.
"SCLl": loop block. This block defines looping point for the song. It
contains only DWORD value, which is the looping jump position (in samples)
relative to the start of the song. You should make the jump just when you
encounter this block.
"SCEl": end block. This block indicates the end of audio stream.
Note that in some games audio files are contained within game resources. As
a rule, such resources are not compressed/encrypted, so you may just search
for ASF file signature (e.g. "SCHl") and this will mark the beginning of audio
stream, while "SCEl" block marks the end of that stream.
============
2. EA ADPCM Decompression Algorithm
============
During the decompression four LONG variables must be maintained for stereo
stream: lCurSampleLeft, lCurSampleRight, lPrevSampleLeft, lPrevSampleRight
and two -- for mono stream: lCurSample, lPrevSample. At the beginning of each
"SCDl" data block you must initialize these variables using the values in
ASFChunkHeader.
Note that LONG here is signed.
Here's the code which decompresses one "SCDl" block of EA ADPCM compressed
stereo stream.
BYTE InputBuffer[InputBufferSize]; '' buffer containing audio data of "SCDl" block
BYTE bInput;
DWORD dwOutSize; '' outsize value from the ASFChunkHeader
DWORD i, bCount, sCount;
LONG c1left,c2left,c1right,c2right,left,right;
BYTE dleft,dright;
DWORD dwSubOutSize=0x1c;
i=0;
'' process integral number of (dwSubOutSize) samples
for (bCount=0;bCount<(dwOutSize/dwSubOutSize);bCount++)
{
bInput=InputBuffer[i++];
c1left=EATable[HINIBBLE(bInput)]; '' predictor coeffs for left channel
c2left=EATable[HINIBBLE(bInput)+4];
c1right=EATable[LONIBBLE(bInput)]; '' predictor coeffs for right channel
c2right=EATable[LONIBBLE(bInput)+4];
bInput=InputBuffer[i++];
dleft=HINIBBLE(bInput)+8; '' shift value for left channel
dright=LONIBBLE(bInput)+8; '' shift value for right channel
for (sCount=0;sCount<dwSubOutSize;sCount++)
{
bInput=InputBuffer[i++];
left=HINIBBLE(bInput); '' HIGHER nibble for left channel
right=LONIBBLE(bInput); '' LOWER nibble for right channel
left=(left<<0x1c)>>dleft;
right=(right<<0x1c)>>dright;
left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
right=(right+lCurSampleRight*c1right+lPrevSampleRight*c2right+0x80)>>8;
left=Clip16BitSample(left);
right=Clip16BitSample(right);
lPrevSampleLeft=lCurSampleLeft;
lCurSampleLeft=left;
lPrevSampleRight=lCurSampleRight;
lCurSampleRight=right;
'' Now we've got lCurSampleLeft and lCurSampleRight which form one stereo
'' sample and all is set for the next input byte...
Output((SHORT)lCurSampleLeft,(SHORT)lCurSampleRight); '' send the sample to output
}
}
'' process the rest (if any)
if ((dwOutSize % dwSubOutSize) != 0)
{
bInput=InputBuffer[i++];
c1left=EATable[HINIBBLE(bInput)]; '' predictor coeffs for left channel
c2left=EATable[HINIBBLE(bInput)+4];
c1right=EATable[LONIBBLE(bInput)]; '' predictor coeffs for right channel
c2right=EATable[LONIBBLE(bInput)+4];
bInput=InputBuffer[i++];
dleft=HINIBBLE(bInput)+8; '' shift value for left channel
dright=LONIBBLE(bInput)+8; '' shift value for right channel
for (sCount=0;sCount<(dwOutSize % dwSubOutSize);sCount++)
{
bInput=InputBuffer[i++];
left=HINIBBLE(bInput); '' HIGHER nibble for left channel
right=LONIBBLE(bInput); '' LOWER nibble for right channel
left=(left<<0x1c)>>dleft;
right=(right<<0x1c)>>dright;
left=(left+lCurSampleLeft*c1left+lPrevSampleLeft*c2left+0x80)>>8;
right=(right+lCurSampleRight*c1right+lPrevSampleRight*c2right+0x80)>>8;
left=Clip16BitSample(left);
right=Clip16BitSample(right);
lPrevSampleLeft=lCurSampleLeft;
lCurSampleLeft=left;
lPrevSampleRight=lCurSampleRight;
lCurSampleRight=right;
'' Now we've got lCurSampleLeft and lCurSampleRight which form one stereo
'' sample and all is set for the next input byte...
Output((SHORT)lCurSampleLeft,(SHORT)lCurSampleRight); '' send the sample to output
}
}
HINIBBLE and LONIBBLE are higher and lower 4-bit nibbles:
#define HINIBBLE(byte) ((byte) >> 4)
#define LONIBBLE(byte) ((byte) & 0x0F)
Note that depending on your compiler you may need to use additional nibble
separation in these defines, e.g. (((byte) >> 4) & 0x0F).
EATable is the table given in the next section of this document.
Output() is just a placeholder for any action you would like to perform for
decompressed sample value.
Clip16BitSample is quite evident:
LONG Clip16BitSample(LONG sample)
{
if (sample>32767)
return 32767;
else if (sample<-32768)
return (-32768);
else
return sample;
}
As to mono sound, it's just analoguous: dwSubOutSize=0x0E for mono and
you should get predictor coeffs and shift from one byte:
bInput=InputBuffer[i++];
c1=EATable[HINIBBLE(bInput)]; '' predictor coeffs
c2=EATable[HINIBBLE(bInput)+4];
d=LONIBBLE(bInput)+8; '' shift value
And also you should process HIGHER nibble of the input byte first and then
LOWER nibble for mono sound.
Of course, this decompression routine may be greatly optimized.
======
3. EA ADPCM Table
======
LONG EATable[]=
{
0x00000000,
0x000000F0,
0x000001CC,
0x00000188,
0x00000000,
0x00000000,
0xFFFFFF30,
0xFFFFFF24,
0x00000000,
0x00000001,
0x00000003,
0x00000004,
0x00000007,
0x00000008,
0x0000000A,
0x0000000B,
0x00000000,
0xFFFFFFFF,
0xFFFFFFFD,
0xFFFFFFFC
};
