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Add support for 16-bit, stereo, and MACE 3:1 sound import.

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This commit is contained in:
elasota
2020-06-20 01:03:41 -04:00
parent 1c15ea5940
commit 1981320afe
1 changed files with 111 additions and 17 deletions
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128
gpr2gpa/gpr2gpa.cpp
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@@ -738,7 +738,7 @@ void PadAlignWave(std::vector<uint8_t> &outWAV)
outWAV.push_back(0);
}
bool DecompressSound(int compressionID, int channelCount, const void *sndData, size_t numFrames, std::vector<uint8_t> &decompressed)
bool DecompressSound(int compressionID, int channelCount, const void *sndData, size_t sndDataSize, size_t numFrames, std::vector<uint8_t> &decompressed)
{
if (compressionID == AudioCompressionCodecID_ThreeToOne)
{
@@ -748,8 +748,27 @@ bool DecompressSound(int compressionID, int channelCount, const void *sndData, s
return false;
}
fprintf(stderr, "Unimplemented audio codec\n");
return false;
if (sndDataSize < numFrames * 2)
{
fprintf(stderr, "Sound data is too small\n");
return false;
}
MaceChannelDecState state;
memset(&state, 0, sizeof(state));
const uint8_t *packets = static_cast<const uint8_t*>(sndData);
for (size_t i = 0; i < numFrames; i++)
{
for (size_t subPkt = 0; subPkt < 2; subPkt++)
{
uint8_t samples[3];
DecodeMACE3(&state, packets[i * 2 + subPkt], samples);
for (int s = 0; s < 3; s++)
decompressed.push_back(samples[s]);
}
}
}
else if (compressionID = AudioCompressionCodecID_SixToOne)
{
@@ -759,6 +778,12 @@ bool DecompressSound(int compressionID, int channelCount, const void *sndData, s
return false;
}
if (sndDataSize < numFrames)
{
fprintf(stderr, "Sound data is too small\n");
return false;
}
MaceChannelDecState state;
memset(&state, 0, sizeof(state));
@@ -781,7 +806,7 @@ bool DecompressSound(int compressionID, int channelCount, const void *sndData, s
return true;
}
bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize)
bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize, int resID)
{
// Glider PRO has a hard-coded expectation that the sound will have exactly 20 bytes of prefix.
// The resource format can have more than that, we'll just follow this base expectation
@@ -861,8 +886,11 @@ bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize
memcpy(&header, sndBufferData, sizeof(header));
std::vector<uint8_t> decompressedSound;
std::vector<uint8_t> resampledSound;
std::vector<uint8_t> downmixedSound;
uint32_t dataLength = 0;
uint32_t inputDataLength = 0;
uint32_t outputDataLength = 0;
if (header.m_encoding == 0xfe)
{
if (inSize < sizeof(CmpHeader))
@@ -874,10 +902,10 @@ bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize
sndBufferData += sizeof(ExtHeader);
inSize -= sizeof(ExtHeader);
if (!DecompressSound(cmpHeader.m_compressionID, cmpHeader.m_channelCount, sndBufferData, cmpHeader.m_numFrames, decompressedSound))
if (!DecompressSound(cmpHeader.m_compressionID, cmpHeader.m_channelCount, sndBufferData, inSize, cmpHeader.m_numFrames, decompressedSound))
return false;
dataLength = decompressedSound.size();
outputDataLength = decompressedSound.size();
if (decompressedSound.size() > 0)
sndBufferData = &decompressedSound[0];
}
@@ -900,34 +928,98 @@ bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize
uint16_t bitsPerSample = extHeader.m_sampleSize;
if (bitsPerSample != 8)
if (bitsPerSample != 8 && bitsPerSample != 16)
{
fprintf(stderr, "Sound had unexpected bit rate\n");
return false;
}
if (extHeader.m_channelCount != 1)
if (extHeader.m_channelCount != 1 && extHeader.m_channelCount != 2)
{
fprintf(stderr, "Sound had unexpected channel count\n");
return false;
}
dataLength = extHeader.m_numSamples * extHeader.m_channelCount * bitsPerSample / 8;
inputDataLength = extHeader.m_numSamples * extHeader.m_channelCount * (bitsPerSample / 8);
outputDataLength = extHeader.m_numSamples;
sndBufferData += sizeof(ExtHeader);
inSize -= sizeof(ExtHeader);
if (dataLength > inSize)
if (inputDataLength > inSize)
return false;
if (inputDataLength == 0)
return false;
int channelCount = extHeader.m_channelCount;
if (channelCount == 2)
{
fprintf(stderr, "WARNING: Downmixing sound resource %i to mono\n", resID);
if (bitsPerSample == 16)
{
const size_t numSamples = extHeader.m_numSamples;
downmixedSound.resize(numSamples * 2);
for (size_t i = 0; i < numSamples; i++)
{
BEInt16_t channels[2];
memcpy(channels, sndBufferData + i * 4, 4);
const BEInt16_t downmixedSample = BEInt16_t(static_cast<int16_t>((static_cast<int32_t>(channels[0]) + static_cast<int32_t>(channels[1]) + 1) >> 1));
memcpy(&downmixedSound[i * 2], &downmixedSample, 2);
}
sndBufferData = &downmixedSound[0];
inputDataLength = downmixedSound.size();
}
else
{
const size_t numSamples = extHeader.m_numSamples;
downmixedSound.resize(numSamples);
for (size_t i = 0; i < numSamples; i++)
{
const uint8_t *channels = sndBufferData + i * 2;
const uint8_t downmixedSample = static_cast<uint8_t>((channels[0] + channels[1] + 1) >> 1);
downmixedSound[i] = downmixedSample;
}
sndBufferData = &downmixedSound[0];
inputDataLength = downmixedSound.size();
}
}
if (bitsPerSample == 16)
{
fprintf(stderr, "WARNING: Downsampling sound resource %i to 8 bit\n", resID);
const size_t numSamples = extHeader.m_numSamples * extHeader.m_channelCount;
resampledSound.resize(numSamples);
for (size_t i = 0; i < numSamples; i++)
{
const uint16_t sample16BitSigned = ((static_cast<int16_t>(sndBufferData[i * 2]) << 8) | sndBufferData[i * 2 + 1]) ^ 0x8000;
const uint8_t sample8Bit = (static_cast<uint32_t>(sample16BitSigned) * 2 + 257) / 514;
resampledSound[i] = sample8Bit;
}
sndBufferData = &resampledSound[0];
inputDataLength = resampledSound.size();
}
}
else
{
dataLength = header.m_length;
inputDataLength = outputDataLength = header.m_length;
sndBufferData += sizeof(header);
inSize -= sizeof(header);
if (dataLength > inSize)
if (inputDataLength > inSize)
return false;
}
@@ -942,7 +1034,7 @@ bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize
const size_t fmtTagSize = sizeof(PortabilityLayer::RIFFTag);
const size_t fmtContentSize = PadRiffChunk(sizeof(formatChunk));
const size_t dataTagSize = sizeof(PortabilityLayer::RIFFTag);
const size_t dataContentSize = PadRiffChunk(dataLength);
const size_t dataContentSize = PadRiffChunk(outputDataLength);
// Structure:
// riffTag
@@ -981,11 +1073,11 @@ bool ImportSound(std::vector<uint8_t> &outWAV, const void *inData, size_t inSize
PortabilityLayer::RIFFTag dataTag;
dataTag.m_tag = PortabilityLayer::WaveConstants::kDataChunkID;
dataTag.m_chunkSize = dataLength;
dataTag.m_chunkSize = outputDataLength;
VectorAppend(outWAV, reinterpret_cast<const uint8_t*>(&dataTag), sizeof(dataTag));
VectorAppend(outWAV, sndBufferData, dataLength);
VectorAppend(outWAV, sndBufferData, outputDataLength);
PadAlignWave(outWAV);
return true;
@@ -1408,8 +1500,10 @@ int ConvertSingleFile(const char *resPath, const PortabilityLayer::CombinedTimes
entry.m_name = resName;
if (ImportSound(entry.m_uncompressedContents, resData, resSize))
if (ImportSound(entry.m_uncompressedContents, resData, resSize, res.m_resID))
contents.push_back(entry);
else
fprintf(stderr, "Failed to import snd res %i\n", static_cast<int>(res.m_resID));
}
else if (typeList.m_resType == indexStringTypeID)
{