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# adpcm encoding currently doesn't work well, so this isn't being integrated into the assets scripts
# however, this can be used to convert all ast files to a wav file and to encode a wav file to a pcm16 ast file
import struct
import os
import sys
import dataclasses
import json
DSPADPCM_FILTER = [
[0, 0],
[2048, 0],
[0, 2048],
[1024, 1024],
[4096, -2048],
[3584, -1536],
[3072, -1024],
[4608, -2560],
[4200, -2248],
[4800, -2300],
[5120, -3072],
[2048, -2048],
[1024, -1024],
[-1024, 1024],
[-1024, 0],
[-2048, 0],
]
@dataclasses.dataclass
class AST_Header:
identifier: int
fileSize: int
format: int # 0 = ADPCM, 1 = PCM16
bitDepth: int
numChannels: int
unk_1: int
sampleRate: int
totalSamples: int
loopStartSample: int
loopEndSample: int
blockSize: int
unk_2: int
unk_3: int
@dataclasses.dataclass
class AST_Json:
blockSize: int
loopStartSample: int
loopEndSample: int
encodingFormat: int
loops: bool
def testWriteFunction(name, data):
open(name, "wb").write(data)
def clampSample(sample):
if sample > 32767:
sample = 32767
elif sample < -32767:
sample = -32767
return sample
def convert_to_wav(name, astData, writeFunction):
header = struct.unpack(">IIHHHHIIIIIII", astData[:44])
astHeader = AST_Header(*header)
print(astHeader)
if astHeader.bitDepth != 16:
print("Error: libast only supports 16 bit audio!")
return None
nextBlockPosition = 0x40
audioData = []
for channel in range(astHeader.numChannels):
audioData.append([])
while nextBlockPosition < astHeader.fileSize + 0x40:
currentBlockPosition = nextBlockPosition
blockHeader = struct.unpack(
">II", astData[currentBlockPosition : currentBlockPosition + 8]
)
if blockHeader[0] != 1112294219:
print("Block not found at position " + hex(currentBlockPosition))
break
blockSize = blockHeader[1]
nextBlockPosition = (
currentBlockPosition + 32 + (blockSize * astHeader.numChannels)
)
print(
"current is "
+ str(currentBlockPosition)
+ " next is "
+ str(nextBlockPosition)
+ " full is "
+ str(astHeader.fileSize + 0x40)
+ " size is "
+ str(blockSize)
)
for channel in range(astHeader.numChannels):
blockDataIndex = currentBlockPosition + 32 + (blockSize * channel)
if astHeader.format == 0: # adcpm
endIndex = blockDataIndex + ((blockSize // 9) * 9)
for i in range(blockDataIndex, endIndex, 9):
delta = 1 << ((astData[i] >> 4) & 0b1111)
id = astData[i] & 0b1111
data = []
for j in range(8):
data.append((astData[i + 1 + j] >> 4) & 0b1111)
data.append((astData[i + 1 + j]) & 0b1111)
for nibble in data:
if nibble >= 8:
nibble = nibble - 16
hist = 0
hist2 = 0
if len(audioData[channel]) == 1:
hist = audioData[channel][-1]
elif len(audioData[channel]) > 1:
hist = audioData[channel][-1]
hist2 = audioData[channel][-2]
sample = (delta * nibble) << 11
sample = sample + (
(hist * DSPADPCM_FILTER[id][0])
+ (hist2 * DSPADPCM_FILTER[id][1])
)
sample = sample >> 11
sample = clampSample(sample)
audioData[channel].append(sample)
elif astHeader.format == 1: # pcm16
for i in range(blockDataIndex, blockDataIndex + blockSize, 2):
audioData[channel].append(
struct.unpack(">h", astData[i : i + 2])[0]
)
if nextBlockPosition == astHeader.fileSize + 0x40:
break
# done parsing data
subChunk2Size = len(audioData[0]) * 2 * astHeader.numChannels
wavHeader = struct.pack(
"<ccccIccccccccIHHIIHHccccI",
bytes("R", "ascii"),
bytes("I", "ascii"),
bytes("F", "ascii"),
bytes("F", "ascii"),
36 + subChunk2Size,
bytes("W", "ascii"),
bytes("A", "ascii"),
bytes("V", "ascii"),
bytes("E", "ascii"),
bytes("f", "ascii"),
bytes("m", "ascii"),
bytes("t", "ascii"),
bytes(" ", "ascii"),
16,
1,
astHeader.numChannels,
astHeader.sampleRate,
2 * astHeader.numChannels * astHeader.sampleRate,
2 * astHeader.numChannels,
16,
bytes("d", "ascii"),
bytes("a", "ascii"),
bytes("t", "ascii"),
bytes("a", "ascii"),
subChunk2Size,
)
wavFile = bytearray(wavHeader)
print("writing")
for i in range(len(audioData[0])):
if i % 100000 == 0:
print(i)
for channel in audioData:
wavFile.append(channel[i] & 0xFF)
wavFile.append((channel[i] >> 8) & 0xFF)
loops = False
if astHeader.unk_1 == 0xFFFF:
loops = True
jsonData = AST_Json(
astHeader.blockSize,
astHeader.loopStartSample,
astHeader.loopEndSample,
astHeader.format,
loops,
)
jsonDataString = json.dumps(dataclasses.asdict(jsonData))
split = os.path.splitext(name)
open(split[0] + ".json", "w").write(jsonDataString)
writeFunction(split[0] + ".wav", wavFile)
def wav_to_ast(sourceWav, convertFunction):
wavData = open(sourceWav, "rb").read()
jsonDataString = open(os.path.splitext(sourceWav)[0] + ".json", "r").read()
jsonData = json.loads(jsonDataString)
print(jsonData)
headerData = struct.unpack("<HI", wavData[0x16:0x1C])
blockSize = jsonData["blockSize"]
loopStartSample = jsonData["loopStartSample"]
loopEndSample = jsonData["loopEndSample"]
encodingFormat = jsonData["encodingFormat"]
numChannels = headerData[0]
sampleRate = headerData[1]
astData = bytearray(20)
wavDataPos = 0x2C
loops = 0
if jsonData["loops"] == True:
loops = 0xFFFF
sampleNum = 0
if encodingFormat == 1: # pcm16
exitLoop = False
while exitLoop == False:
thisBlockSize = blockSize
if len(wavData) - wavDataPos < (blockSize * numChannels):
thisBlockSize = (len(wavData) - wavDataPos) // numChannels
exitLoop = True
astData.append(ord("B"))
astData.append(ord("L"))
astData.append(ord("C"))
astData.append(ord("K"))
# doing this is faster than adding two bytearrays together
astData.append((thisBlockSize >> 24) & 0xFF)
astData.append((thisBlockSize >> 16) & 0xFF)
astData.append((thisBlockSize >> 8) & 0xFF)
astData.append((thisBlockSize) & 0xFF) # Write as big endian
print(thisBlockSize)
for i in range(24):
astData.append(0)
sampleNum = sampleNum + (thisBlockSize // 2)
for channel in range(numChannels):
for i in range(
0, thisBlockSize * numChannels, 2 * numChannels
): # convert little to big endian
astData.append(wavData[wavDataPos + i + 1 + (channel * 2)])
astData.append(wavData[wavDataPos + i + (channel * 2)])
wavDataPos = wavDataPos + (thisBlockSize * numChannels)
elif encodingFormat == 0: # adpcm
exitLoop = False
sampleList = []
for channel in range(numChannels):
sampleList.append([])
for i in range(
0, len(wavData) - wavDataPos, 2 * numChannels
): # convert little to big endian
sampleList[channel].append(
(wavData[i + 1 + (channel * 2)] << 8) | (wavData[i + (channel * 2)])
)
for i in range(16 - (len(sampleList) % 16)):
sampleList[channel].append(0)
last = 0
penult = 0
while exitLoop == False:
thisBlockSize = blockSize
if loopEndSample - sampleNum <= ((blockSize // 9) * 16):
thisBlockSize = ((loopEndSample - sampleNum) // 16) * 9
exitLoop = True
astData.append(ord("B"))
astData.append(ord("L"))
astData.append(ord("C"))
astData.append(ord("K"))
# doing this is faster than adding two bytearrays together
astData.append((thisBlockSize >> 24) & 0xFF)
astData.append((thisBlockSize >> 16) & 0xFF)
astData.append((thisBlockSize >> 8) & 0xFF)
astData.append((thisBlockSize) & 0xFF) # Write as big endian
print(thisBlockSize)
for i in range(24):
astData.append(0)
sampleNum = sampleNum + ((thisBlockSize // 9) * 16)
print(sampleNum)
for channel in range(numChannels):
for sample in range(
sampleNum, sampleNum + ((thisBlockSize // 9) * 16), 16
):
def getSample(sampleIndex):
if len(sampleList[channel]) > sampleIndex:
return sampleList[channel][sampleIndex]
else:
return 0
# code ported from https://github.com/XAYRGA/wsystool/blob/37eb357b11b2e2b80f50784d4e993713e5e00508/wsystool/bananapeel/banan_flaaf.cs#L393
isBlank = True
for i in range(16):
if getSample(sample + i) != 0:
isBlank = False
break
if isBlank:
for i in range(9):
astData.append(0)
continue
coeffIndex = -1
minerror = 0xFFFFFFFF
for coeff in range(16):
lastCoeff = DSPADPCM_FILTER[coeff][0]
penultCoeff = DSPADPCM_FILTER[coeff][1]
foundScale = -1
coeff_error = 0
for current_scale in range(16):
step = 1 << current_scale
nibbleCoeff = 2048 << current_scale
success = True
coeff_error = 0
_last = lastCoeff
_penult = penultCoeff
for i in range(16):
prediction = clampSample(
((lastCoeff * _last) + (penultCoeff * _penult))
>> 11
)
difference = -(prediction - getSample(sample + i))
nibble = difference // step
if nibble < -8 or nibble > 7:
success = False
break
nibbleSample = nibble * nibbleCoeff
decoded = clampSample(
(
(nibbleSample)
+ (lastCoeff * _last)
+ (penultCoeff * _penult)
)
>> 11
)
_penult = _last
_last = decoded
coeff_error = coeff_error + abs(difference)
if success:
foundScale = current_scale
break
if foundScale < 0:
continue
if coeff_error < minerror:
minerror = coeff_error
coeffIndex = coeff
scale = foundScale
lastCoeff = DSPADPCM_FILTER[coeffIndex][0]
penultCoeff = DSPADPCM_FILTER[coeffIndex][1]
step = 1 << scale
nibbles = [int] * 16
for i in range(16):
prediction = clampSample(
((lastCoeff * last) + (penultCoeff * penult)) >> 11
)
difference = -(prediction - getSample(sample + i))
nibbles[i] = difference // step
nibbleSample = nibbles[i] * (2048 << scale)
decoded = clampSample(
nibbleSample + (lastCoeff * last) + (penultCoeff * penult)
>> 11
)
penult = last
last = decoded
astData.append(((scale & 0xF) << 4) | coeffIndex)
for i in range(8):
astData.append(
((nibbles[i * 2] & 0xF) << 4)
| ((nibbles[(i * 2) + 1] & 0xF))
)
astHeader = struct.pack(
">IIHHHHIIIIIII",
1398035021,
len(astData) - 20,
encodingFormat,
16,
numChannels,
loops,
sampleRate,
loopEndSample,
loopStartSample,
loopEndSample,
blockSize,
0,
0x7F000000,
)
astFile = astHeader + astData
split = os.path.splitext(sourceWav)
convertFunction(split[0] + ".ast", astFile)
if __name__ == "__main__":
if len(sys.argv) >= 2:
split = os.path.splitext(sys.argv[1])
if split[1] == ".ast":
convert_to_wav(
sys.argv[1], open(sys.argv[1], "rb").read(), testWriteFunction
)
elif split[1] == ".wav":
wav_to_ast(sys.argv[1], testWriteFunction)
|