public ADPCMStream(RSTMHeader* pRSTM, VoidPtr dataAddr)
{
HEADHeader* pHeader = pRSTM->HEADData;
StrmDataInfo* part1 = pHeader->Part1;
bshort* ynCache = (bshort*)pRSTM->ADPCData->Data;
byte* sPtr;
short[][] coefs;
ADPCMInfo* info;
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;
_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;
}
}
public override bool OnInitialize()
{
if ((_name == null) && (_origPath != null))
{
_name = Path.GetFileNameWithoutExtension(_origPath);
}
base.OnInitialize();
StrmDataInfo *part1 = Header->HEADData->Part1;
_encoding = part1->_format._encoding;
_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;
int offset = ((int)(Header->DATAData->Data) - (int)(Header));
if (offset < WorkingUncompressed.Length)
{
_audioSource = new DataSource(Header->DATAData->Data, WorkingUncompressed.Length - offset);
SetSizeInternal(offset);
}
return(false);
}
public override bool OnInitialize()
{
if (_name == null && _origPath != null)
{
_name = Path.GetFileNameWithoutExtension(_origPath);
}
base.OnInitialize();
if (Header->_header._tag == CSTMHeader.Tag)
{
ShowADPCMConversionWarning();
byte[] brstm_temp = CSTMConverter.ToRSTM((CSTMHeader *)Header);
fixed(byte *ptr = brstm_temp)
{
ReplaceRaw(ptr, brstm_temp.Length);
return(false);
}
}
if (Header->_header._tag == FSTMHeader.Tag)
{
ShowADPCMConversionWarning();
byte[] brstm_temp = FSTMConverter.ToRSTM((FSTMHeader *)Header);
fixed(byte *ptr = brstm_temp)
{
ReplaceRaw(ptr, brstm_temp.Length);
return(false);
}
}
StrmDataInfo *part1 = Header->HEADData->Part1;
_encoding = part1->_format._encoding;
_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;
int offset = Header->DATAData->Data - Header;
if (offset < WorkingUncompressed.Length)
{
_audioSource = new DataSource(Header->DATAData->Data, WorkingUncompressed.Length - offset);
SetSizeInternal(offset);
}
return(false);
}
public static ADPCMStream[] GetStreams(RSTMHeader* pRSTM, VoidPtr dataAddr)
{
HEADHeader* pHeader = pRSTM->HEADData;
StrmDataInfo* part1 = pHeader->Part1;
int c = part1->_format._channels;
ADPCMStream[] streams = new ADPCMStream[c.RoundUpToEven() / 2];
for (int i = 0; i < streams.Length; i++)
{
int x = (i + 1) * 2 <= c ? 2 : 1;
streams[i] = new ADPCMStream(pRSTM, x, i * 2, dataAddr);
}
return streams;
}
public IAudioStream[] CreateStreams()
{
if (Header == null)
{
return(null);
}
StrmDataInfo *info = Header->HEADData->Part1;
if (Header != null)
{
switch ((WaveEncoding)info->_format._encoding)
{
case WaveEncoding.ADPCM:
return(ADPCMStream.GetStreams(Header, _audioSource.Address));
case WaveEncoding.PCM16:
return(PCMStream.GetStreams(Header, _audioSource.Address));
}
}
return(new IAudioStream[] { null });
}
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
StrmDataInfo *part1 = head->Part1;
part1->_format = new AudioFormatInfo(2, (byte)(looped ? 1 : 0), (byte)channels, 0);
part1->_sampleRate = (ushort)sampleRate;
part1->_blockHeaderOffset = 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->_dataInterval = 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()
{
_pad = 0
}
}
;
//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 static unsafe FileMap Encode(IAudioStream stream, IProgressTracker progress, WaveEncoding encoding = WaveEncoding.ADPCM)
#endif
{
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;
int samplesPerBlock = encoding == WaveEncoding.ADPCM ? 0x3800
: encoding == WaveEncoding.PCM16 ? 0x1000
: 0;
if (samplesPerBlock == 0)
{
throw new ArgumentException("Encoding must be ADPCM or PCM16");
}
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 % samplesPerBlock) != 0)
{
loopPadding = samplesPerBlock - 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 + samplesPerBlock - 1) / samplesPerBlock;
//Initialize stream info
if ((tmp = totalSamples % samplesPerBlock) != 0)
{
lbSamples = tmp;
if (encoding == WaveEncoding.ADPCM)
{
lbSize = (lbSamples + 13) / 14 * 8;
}
else if (encoding == WaveEncoding.PCM16)
{
lbTotal = lbSize = lbSamples * 2;
}
else if (encoding == WaveEncoding.PCM8)
{
lbTotal = lbSize = lbSamples;
}
else
{
throw new NotImplementedException();
}
lbTotal = lbSize.Align(0x20);
}
else
{
lbSamples = samplesPerBlock;
lbTotal = lbSize = 0x2000;
}
//Get section sizes
int rstmSize = 0x40;
int headSize = (0x68 + (channels * (encoding == WaveEncoding.ADPCM ? 0x40 : 0x10))).Align(0x20);
int adpcSize = encoding == WaveEncoding.ADPCM
? ((blocks - 1) * 4 * channels + 0x10).Align(0x20)
: 0;
int dataSize = ((blocks - 1) * 0x2000 + lbTotal) * channels + 0x20;
#if RSTMLIB
//Create byte array
byte[] array = new byte[rstmSize + headSize + adpcSize + dataSize];
fixed(byte *address = array)
{
#else
//Create file map
FileMap map = FileMap.FromTempFile(rstmSize + headSize + adpcSize + dataSize);
VoidPtr address = map.Address;
#endif
//Get section pointers
RSTMHeader * rstm = (RSTMHeader *)address;
HEADHeader * head = (HEADHeader *)((byte *)rstm + rstmSize);
ADPCHeader * adpc = (ADPCHeader *)((byte *)head + headSize);
RSTMDATAHeader *data = (RSTMDATAHeader *)((byte *)adpc + adpcSize);
//Initialize sections
rstm->Set(headSize, adpcSize, dataSize);
head->Set(headSize, channels, encoding);
if (adpcSize > 0)
{
adpc->Set(adpcSize);
}
data->Set(dataSize);
//Set HEAD data
StrmDataInfo *part1 = head->Part1;
part1->_format = new AudioFormatInfo((byte)encoding, (byte)(looped ? 1 : 0), (byte)channels, 0);
part1->_sampleRate = (ushort)sampleRate;
part1->_blockHeaderOffset = 0;
part1->_loopStartSample = loopStart;
part1->_numSamples = totalSamples;
part1->_dataOffset = rstmSize + headSize + adpcSize + 0x20;
part1->_numBlocks = blocks;
part1->_blockSize = 0x2000;
part1->_samplesPerBlock = samplesPerBlock;
part1->_lastBlockSize = lbSize;
part1->_lastBlockSamples = lbSamples;
part1->_lastBlockTotal = lbTotal;
part1->_dataInterval = encoding == WaveEncoding.ADPCM ? samplesPerBlock : 0;
part1->_bitsPerSample = encoding == WaveEncoding.ADPCM ? 4 : 0;
if (encoding == WaveEncoding.ADPCM)
{
//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()
{
_pad = 0
}
}
;
//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 x = 0; x < channels; x++)
{
*pyn++ = 0;
*pyn++ = 0;
}
for (int sIndex = 0, bIndex = 1; sIndex < totalSamples; sIndex += samplesPerBlock, bIndex++)
{
int blockSamples = Math.Min(totalSamples - sIndex, samplesPerBlock);
for (int x = 0; x < channels; x++)
{
short *sPtr = channelBuffers[x] + sIndex;
//Set block yn values
if (bIndex != blocks)
{
*pyn++ = sPtr[samplesPerBlock + 1];
*pyn++ = sPtr[samplesPerBlock];
}
//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 % samplesPerBlock) == 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 / samplesPerBlock;
int loopChunk = (loopStart - (loopBlock * samplesPerBlock)) / 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]);
}
}
else if (encoding == WaveEncoding.PCM16)
{
bshort *destPtr = (bshort *)data->Data;
for (int i = 0; i < blocks; i++)
{
int samplesPerChannel = i < blocks - 1
? part1->_samplesPerBlock
: part1->_lastBlockSamples;
int bytesPerChannel = i < blocks - 1
? part1->_blockSize
: part1->_lastBlockTotal;
short[] sampleData = new short[channels * bytesPerChannel / sizeof(short)];
fixed(short *sampleDataPtr = sampleData)
{
int read = 0;
do
{
if (stream.SamplePosition == stream.LoopEndSample && looped)
{
stream.SamplePosition = stream.LoopStartSample;
}
int s = stream.ReadSamples(sampleDataPtr + read, samplesPerChannel - read);
if (s == 0)
{
throw new Exception("No samples could be read from the stream");
}
read += s;
}while (read < samplesPerChannel);
}
for (int j = 0; j < channels; j++)
{
for (int k = j; k < sampleData.Length; k += channels)
{
*(destPtr++) = sampleData[k];
}
}
progress.Update(progress.CurrentValue + (samplesPerChannel * channels * 3));
}
}
if (progress != null)
{
progress.Finish();
}
#if RSTMLIB
}
return(array);
#else
return(map);
#endif
}
internal PCMStream(RSTMHeader *header, int channels, int startChannel, void *audioSource)
{
StrmDataInfo *info = header->HEADData->Part1;
if (channels > 2)
{
throw new NotImplementedException("Cannot load PCM16 audio with more than 2 channels");
}
if (info->_format._channels < channels)
{
throw new Exception("Not enough channels");
}
int size = info->_numSamples * channels * sizeof(short);
_allocatedHGlobal = Marshal.AllocHGlobal(size);
byte *fromL = (byte *)audioSource;
byte *fromR = fromL;
byte *to = (byte *)_allocatedHGlobal;
bool stereo = channels == 2;
for (int block = 0; block < info->_numBlocks; block++)
{
int bs = (block == info->_numBlocks - 1)
? info->_lastBlockSize
: info->_blockSize;
if (block == 0)
{
fromL += (bs * startChannel);
fromR += (bs * startChannel);
}
else
{
fromL += bs * (info->_format._channels - channels);
fromR += bs * (info->_format._channels - channels);
}
if (stereo)
{
fromR += bs;
}
for (int i = 0; i < bs; i += 2)
{
to[0] = fromL[1];
to[1] = fromL[0];
fromL += 2;
to += 2;
if (stereo)
{
to[0] = fromR[1];
to[1] = fromR[0];
fromR += 2;
to += 2;
}
}
if (stereo)
{
fromL += bs;
}
}
_bps = 16;
_numChannels = channels;
_frequency = info->_sampleRate;
_numSamples = info->_numSamples;
_source = (short *)_allocatedHGlobal;
_samplePos = 0;
_looped = info->_format._looped != 0;
_loopStart = info->_loopStartSample;
_loopEnd = _numSamples;
}
internal PCMStream(RSTMHeader *header, int channels, int startChannel, void *audioSource)
{
StrmDataInfo *info = header->HEADData->Part1;
if (info->_format._channels < startChannel + channels)
{
throw new Exception("Not enough channels");
}
List <short[]>[] blocksByChannel = new List <short[]> [info->_format._channels];
for (int i = 0; i < blocksByChannel.Length; i++)
{
blocksByChannel[i] = new List <short[]>();
}
byte *from = (byte *)audioSource;
for (int block = 0; block < info->_numBlocks; block++)
{
int blockSize = (block == info->_numBlocks - 1)
? info->_lastBlockSize
: info->_blockSize;
int blockTotal = (block == info->_numBlocks - 1)
? info->_lastBlockTotal
: info->_blockSize;
for (int channel = 0; channel < info->_format._channels; channel++)
{
short[] b = new short[blockSize / sizeof(short)];
Marshal.Copy((IntPtr)from, b, 0, b.Length);
from += blockTotal;
blocksByChannel[channel].Add(b);
}
}
List <List <short[]> > blocksToUseByChannel = new List <List <short[]> >();
for (int i = 0; i < channels; i++)
{
blocksToUseByChannel.Add(blocksByChannel[startChannel + i]);
}
int size = info->_numSamples * channels * sizeof(short);
//int size2 = blocksToUseByChannel.SelectMany(blocks => blocks.Select(block => block.Length)).Sum() * sizeof(short);
//if (size != size2)
//{
// throw new Exception($"{size} != {size2}");
//}
_allocatedHGlobal = Marshal.AllocHGlobal(size);
short *toPtr = (short *)_allocatedHGlobal;
//short* endPtr = (short*)(_allocatedHGlobal + size);
int blockCount = blocksToUseByChannel.Select(b => b.Count).Distinct().Single();
for (int blockIndex = 0; blockIndex < blockCount; blockIndex++)
{
List <short[]> blocks = blocksToUseByChannel.Select(l => l[blockIndex]).ToList();
int blockShorts = blocks.Select(a => a.Length).Distinct().Single();
for (int i = 0; i < blockShorts; i++)
{
foreach (short[] block in blocks)
{
*toPtr++ = block[i].Reverse();
}
}
}
//if (toPtr != endPtr) throw new Exception($"Not all data filled ({(long)_allocatedHGlobal}, {(long)toPtr}, {(long)endPtr})");
_bps = 16;
_numChannels = channels;
_frequency = info->_sampleRate;
_numSamples = info->_numSamples;
_source = (short *)_allocatedHGlobal;
_samplePos = 0;
_looped = info->_format._looped != 0;
_loopStart = info->_loopStartSample;
_loopEnd = _numSamples;
}
