protected override bool OnInitialize()
{
if ((_name == null) && (_origPath != null))
{
_name = Path.GetFileNameWithoutExtension(_origPath);
}
HEADPart1 *part1 = Header->HEADData->Part1;
_channels = part1->_format._channels;
_looped = part1->_format._looped != 0;
_sampleRate = part1->_sampleRate;
_loopStart = part1->_loopStartSample;
_numSamples = part1->_numSamples;
_dataOffset = part1->_dataOffset;
_numBlocks = part1->_numBlocks;
_blockSize = part1->_blockSize;
_bps = part1->_bitsPerSample;
return(false);
}
public static unsafe FileMap Encode(IAudioStream stream, IProgressTracker progress)
{
int tmp;
bool looped = stream.IsLooping;
int channels = stream.Channels;
int samples;
int blocks;
int sampleRate = stream.Frequency;
int lbSamples, lbSize, lbTotal;
int loopPadding, loopStart, totalSamples;
short *tPtr;
if (looped)
{
loopStart = stream.LoopStartSample;
samples = stream.LoopEndSample; //Set sample size to end sample. That way the audio gets cut off when encoding.
//If loop point doesn't land on a block, pad the stream so that it does.
if ((tmp = loopStart % 0x3800) != 0)
{
loopPadding = 0x3800 - tmp;
loopStart += loopPadding;
}
else
{
loopPadding = 0;
}
totalSamples = loopPadding + samples;
}
else
{
loopPadding = loopStart = 0;
totalSamples = samples = stream.Samples;
}
if (progress != null)
{
progress.Begin(0, totalSamples * channels * 3, 0);
}
blocks = (totalSamples + 0x37FF) / 0x3800;
//Initialize stream info
if ((tmp = totalSamples % 0x3800) != 0)
{
lbSamples = tmp;
lbSize = (lbSamples + 13) / 14 * 8;
lbTotal = lbSize.Align(0x20);
}
else
{
lbSamples = 0x3800;
lbTotal = lbSize = 0x2000;
}
//Get section sizes
int rstmSize = 0x40;
int headSize = (0x68 + (channels * 0x40)).Align(0x20);
int adpcSize = ((blocks - 1) * 4 * channels + 0x10).Align(0x20);
int dataSize = ((blocks - 1) * 0x2000 + lbTotal) * channels + 0x20;
//Create file map
FileMap map = FileMap.FromTempFile(rstmSize + headSize + adpcSize + dataSize);
//Get section pointers
RSTMHeader * rstm = (RSTMHeader *)map.Address;
HEADHeader * head = (HEADHeader *)((int)rstm + rstmSize);
ADPCHeader * adpc = (ADPCHeader *)((int)head + headSize);
RSTMDATAHeader *data = (RSTMDATAHeader *)((int)adpc + adpcSize);
//Initialize sections
rstm->Set(headSize, adpcSize, dataSize);
head->Set(headSize, channels);
adpc->Set(adpcSize);
data->Set(dataSize);
//Set HEAD data
HEADPart1 *part1 = head->Part1;
part1->_format = new AudioFormatInfo(2, (byte)(looped ? 1 : 0), (byte)channels, 0);
part1->_sampleRate = (ushort)sampleRate;
part1->_unk1 = 0;
part1->_loopStartSample = loopStart;
part1->_numSamples = totalSamples;
part1->_dataOffset = rstmSize + headSize + adpcSize + 0x20;
part1->_numBlocks = blocks;
part1->_blockSize = 0x2000;
part1->_samplesPerBlock = 0x3800;
part1->_lastBlockSize = lbSize;
part1->_lastBlockSamples = lbSamples;
part1->_lastBlockTotal = lbTotal;
part1->_unk8 = 0x3800;
part1->_bitsPerSample = 4;
//Create one ADPCMInfo for each channel
int * adpcData = stackalloc int[channels];
ADPCMInfo **pAdpcm = (ADPCMInfo **)adpcData;
for (int i = 0; i < channels; i++)
{
*(pAdpcm[i] = head->GetChannelInfo(i)) = new ADPCMInfo();
}
//Create buffer for each channel
int * bufferData = stackalloc int[channels];
short **channelBuffers = (short **)bufferData;
int bufferSamples = totalSamples + 2; //Add two samples for initial yn values
for (int i = 0; i < channels; i++)
{
channelBuffers[i] = tPtr = (short *)Marshal.AllocHGlobal(bufferSamples * 2); //Two bytes per sample
//Zero padding samples and initial yn values
for (int x = 0; x < (loopPadding + 2); x++)
{
*tPtr++ = 0;
}
}
//Fill buffers
stream.SamplePosition = 0;
short *sampleBuffer = stackalloc short[channels];
for (int i = 2; i < bufferSamples; i++)
{
if (stream.SamplePosition == stream.LoopEndSample && looped)
{
stream.SamplePosition = stream.LoopStartSample;
}
stream.ReadSamples(sampleBuffer, 1);
for (int x = 0; x < channels; x++)
{
channelBuffers[x][i] = sampleBuffer[x];
}
}
//Calculate coefs
for (int i = 0; i < channels; i++)
{
AudioConverter.CalcCoefs(channelBuffers[i] + 2, totalSamples, (short *)pAdpcm[i], progress);
}
//Encode blocks
byte * dPtr = (byte *)data->Data;
bshort *pyn = (bshort *)adpc->Data;
for (int sIndex = 0, bIndex = 1; sIndex < totalSamples; sIndex += 0x3800, bIndex++)
{
int blockSamples = Math.Min(totalSamples - sIndex, 0x3800);
for (int x = 0; x < channels; x++)
{
short *sPtr = channelBuffers[x] + sIndex;
//Set block yn values
if (bIndex != blocks)
{
*pyn++ = sPtr[0x3801];
*pyn++ = sPtr[0x3800];
}
//Encode block (include yn in sPtr)
AudioConverter.EncodeBlock(sPtr, blockSamples, dPtr, (short *)pAdpcm[x]);
//Set initial ps
if (bIndex == 1)
{
pAdpcm[x]->_ps = *dPtr;
}
//Advance output pointer
if (bIndex == blocks)
{
//Fill remaining
dPtr += lbSize;
for (int i = lbSize; i < lbTotal; i++)
{
*dPtr++ = 0;
}
}
else
{
dPtr += 0x2000;
}
}
if (progress != null)
{
if ((sIndex % 0x3800) == 0)
{
progress.Update(progress.CurrentValue + (0x7000 * channels));
}
}
}
//Reverse coefs
for (int i = 0; i < channels; i++)
{
short *p = pAdpcm[i]->_coefs;
for (int x = 0; x < 16; x++, p++)
{
*p = p->Reverse();
}
}
//Write loop states
if (looped)
{
//Can't we just use block states?
int loopBlock = loopStart / 0x3800;
int loopChunk = (loopStart - (loopBlock * 0x3800)) / 14;
dPtr = (byte *)data->Data + (loopBlock * 0x2000 * channels) + (loopChunk * 8);
tmp = (loopBlock == blocks - 1) ? lbTotal : 0x2000;
for (int i = 0; i < channels; i++, dPtr += tmp)
{
//Use adjusted samples for yn values
tPtr = channelBuffers[i] + loopStart;
pAdpcm[i]->_lps = *dPtr;
pAdpcm[i]->_lyn2 = *tPtr++;
pAdpcm[i]->_lyn1 = *tPtr;
}
}
//Free memory
for (int i = 0; i < channels; i++)
{
Marshal.FreeHGlobal((IntPtr)channelBuffers[i]);
}
if (progress != null)
{
progress.Finish();
}
return(map);
}
public ADPCMStream(RSTMHeader *pRSTM)
{
HEADHeader *pHeader = pRSTM->HEADData;
HEADPart1 * part1 = pHeader->Part1;
bshort * ynCache = (bshort *)pRSTM->ADPCData->Data;
byte * sPtr;
short[][] coefs;
ADPCMInfo *info;
VoidPtr dataAddr;
int loopBlock, loopChunk;
short yn1 = 0, yn2 = 0;
_numChannels = part1->_format._channels;
_isLooped = part1->_format._looped != 0;
_sampleRate = part1->_sampleRate;
_numSamples = part1->_numSamples;
_numBlocks = part1->_numBlocks;
_blockLen = part1->_blockSize;
_loopStartSample = part1->_loopStartSample;
_lastBlockSamples = part1->_lastBlockSamples;
_lastBlockSize = part1->_lastBlockTotal;
_samplesPerBlock = part1->_samplesPerBlock;
_loopEndSample = _numSamples;
//Init();
dataAddr = pRSTM->DATAData->Data;
_blockStates = new ADPCMState[_numChannels, _numBlocks];
_currentStates = new ADPCMState[_numChannels];
_loopStates = new ADPCMState[_numChannels];
coefs = new short[_numChannels][];
loopBlock = _loopStartSample / _samplesPerBlock;
loopChunk = (_loopStartSample - (loopBlock * _samplesPerBlock)) / 14;
sPtr = (byte *)dataAddr + (loopBlock * _blockLen * _numChannels) + (loopChunk * 8);
//Get channel info
for (int i = 0; i < _numChannels; i++)
{
info = pHeader->GetChannelInfo(i);
//Get channel coefs
coefs[i] = info->Coefs;
//Fill loop state
_loopStates[i] = new ADPCMState(sPtr, info->_lps, info->_lyn1, info->_lyn2, coefs[i]);
//Advance source pointer for next channel
sPtr += _blockLen;
}
//Fill block states in a linear fashion
sPtr = (byte *)dataAddr;
for (int sIndex = 0, bIndex = 0; sIndex < _numSamples; sIndex += _samplesPerBlock, bIndex++)
{
for (int cIndex = 0; cIndex < _numChannels; cIndex++)
{
if (bIndex > 0) //yn values will be zero if first block
{
yn1 = *ynCache++;
yn2 = *ynCache++;
}
//Get block state
_blockStates[cIndex, bIndex] = new ADPCMState(sPtr, *sPtr, yn1, yn2, coefs[cIndex]); //Use ps from data stream
//Advance address
sPtr += (bIndex == _numBlocks - 1) ? _lastBlockSize : _blockLen;
}
}
}
