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;
int blockLen, samplesPerBlock;
if (looped)
{
loopStart = stream.LoopStartSample;
samples = stream.LoopEndSample; //Set sample size to end sample. That way the audio gets cut off when encoding.
blockLen = (samples.Align(14) / 14 * 8);
samplesPerBlock = blockLen / 8 * 14;
//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 = 0;
loopStart = 0;
totalSamples = samples = stream.Samples;
blockLen = (samples.Align(14) / 14 * 8);
samplesPerBlock = blockLen / 8 * 14;
}
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;
lbSize = (lbSamples + 13) / 14 * 8;
lbTotal = lbSize.Align(0x20);
}
else
{
lbSamples = samplesPerBlock;
lbTotal = lbSize = blockLen;
}
//Get section sizes
int headerSize = RWAV.Size,
infoSize = 8,
waveSize = 0x1C,
tableSize = channels * 4,
channelSize = channels * 0x1C,
adpcmInfoSize = channels * 0x30,
entrySize = (infoSize + waveSize + tableSize + channelSize + adpcmInfoSize).Align(0x20) - 8,
dataSize = (((blocks - 1) * blockLen + lbTotal) * channels) + 8;
//Create file map
FileMap map = FileMap.FromTempFile(headerSize + entrySize + 8 + dataSize);
//Get section pointers
RWAV *header = (RWAV *)map.Address;
header->_header._tag = RWAV.Tag;
header->_header.Endian = Endian.Big;
header->_header._version = 0x102;
header->_header._length = map.Length;
header->_header._firstOffset = 0x20;
header->_header._numEntries = 2;
header->_infoOffset = 0x20;
header->_infoLength = entrySize + 8;
header->_dataOffset = 0x20 + entrySize + 8;
header->_dataLength = dataSize;
RWAVInfo *infoBlock = header->Info;
infoBlock->_header._tag = RWAVInfo.Tag;
infoBlock->_header._length = entrySize + 8;
WaveInfo *wave = &infoBlock->_info;
wave->_format = new AudioFormatInfo(2, (byte)(looped ? 1 : 0), (byte)channels, 0);
wave->_sampleRate = (ushort)sampleRate;
wave->_channelInfoTableOffset = 0x1C;
wave->_dataLocation = (uint)(header->Data->Data - map.Address);
wave->LoopSample = loopStart;
wave->NumSamples = totalSamples;
RWAVData *dataBlock = header->Data;
dataBlock->_header._tag = RWAVData.Tag;
dataBlock->_header._length = dataSize;
//Create one ChannelInfo for each channel
buint * table = (buint *)((VoidPtr)wave + 0x1C);
ChannelInfo *channelInfo = (ChannelInfo *)((VoidPtr)wave + waveSize + tableSize);
for (int i = 0; i < channels; i++)
{
table[i] = (uint)&channelInfo[i] - (uint)wave;
channelInfo[i] = new ChannelInfo()
{
_volBackLeft = 1,
_volBackRight = 1,
_volFrontLeft = 1,
_volFrontRight = 1,
_adpcmInfoOffset = waveSize + tableSize + channelSize + i * 0x30
};
}
//Create one ADPCMInfo for each channel
int * adpcData = stackalloc int[channels];
ADPCMInfo **pAdpcm = (ADPCMInfo **)adpcData;
for (int i = 0; i < channels; i++)
{
*(pAdpcm[i] = wave->GetADPCMInfo(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 *)dataBlock->Data;
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++)
{
channelInfo[x]._channelDataOffset = (int)(dPtr - (byte *)dataBlock->Data);
short *sPtr = channelBuffers[x] + sIndex;
//Set block yn values
if (bIndex != blocks)
{
pAdpcm[x]->_yn1 = sPtr[samplesPerBlock + 1];
pAdpcm[x]->_yn2 = 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 += blockLen;
}
}
if (progress != null && (sIndex % samplesPerBlock) == 0)
{
progress.Update(progress.CurrentValue + (samplesPerBlock * 2 * 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 *)dataBlock->Data + (loopBlock * blockLen * channels) + (loopChunk * 8);
tmp = (loopBlock == blocks - 1) ? lbTotal : blockLen;
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)
{
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
}