Files
scummvm-cursorfix/engines/sci/sound/decoders/sol.cpp
2026-02-02 04:50:13 +01:00

400 lines
14 KiB
C++

/* ScummVM - Graphic Adventure Engine
*
* ScummVM is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
#include "audio/audiostream.h"
#include "audio/rate.h"
#include "audio/decoders/raw.h"
#include "common/substream.h"
#include "common/util.h"
#include "sci/sci.h"
#include "sci/engine/features.h"
#include "sci/sound/decoders/sol.h"
#include "sci/resource/resource.h"
namespace Sci {
// Note that the 16-bit version is also used in coktelvideo.cpp
static const uint16 tableDPCM16[128] = {
0x0000, 0x0008, 0x0010, 0x0020, 0x0030, 0x0040, 0x0050, 0x0060, 0x0070, 0x0080,
0x0090, 0x00A0, 0x00B0, 0x00C0, 0x00D0, 0x00E0, 0x00F0, 0x0100, 0x0110, 0x0120,
0x0130, 0x0140, 0x0150, 0x0160, 0x0170, 0x0180, 0x0190, 0x01A0, 0x01B0, 0x01C0,
0x01D0, 0x01E0, 0x01F0, 0x0200, 0x0208, 0x0210, 0x0218, 0x0220, 0x0228, 0x0230,
0x0238, 0x0240, 0x0248, 0x0250, 0x0258, 0x0260, 0x0268, 0x0270, 0x0278, 0x0280,
0x0288, 0x0290, 0x0298, 0x02A0, 0x02A8, 0x02B0, 0x02B8, 0x02C0, 0x02C8, 0x02D0,
0x02D8, 0x02E0, 0x02E8, 0x02F0, 0x02F8, 0x0300, 0x0308, 0x0310, 0x0318, 0x0320,
0x0328, 0x0330, 0x0338, 0x0340, 0x0348, 0x0350, 0x0358, 0x0360, 0x0368, 0x0370,
0x0378, 0x0380, 0x0388, 0x0390, 0x0398, 0x03A0, 0x03A8, 0x03B0, 0x03B8, 0x03C0,
0x03C8, 0x03D0, 0x03D8, 0x03E0, 0x03E8, 0x03F0, 0x03F8, 0x0400, 0x0440, 0x0480,
0x04C0, 0x0500, 0x0540, 0x0580, 0x05C0, 0x0600, 0x0640, 0x0680, 0x06C0, 0x0700,
0x0740, 0x0780, 0x07C0, 0x0800, 0x0900, 0x0A00, 0x0B00, 0x0C00, 0x0D00, 0x0E00,
0x0F00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
};
// Each 4-bit nibble indexes into this table to refer to one of the 16 delta values.
// deDPCM8Nibble() currently uses the first 8 values, with logic to order the negative
// deltas differently depending on the exact encoding used.
// deDPCM8NibbleWithRepair() uses the whole table, since it matches the order of "old"
// encoding as-is. This saves a tiny bit of computation in the more complex function.
static const int8 tableDPCM8[16] = {
0, 1, 2, 3, 6, 10, 15, 21, -21, -15, -10, -6, -3, -2, -1, -0
};
/**
* Decompresses one channel of 16-bit DPCM compressed audio.
*/
static void deDPCM16Channel(int16 *out, int16 &sample, uint8 delta) {
int32 nextSample = sample;
if (delta & 0x80) {
nextSample -= tableDPCM16[delta & 0x7f];
} else {
nextSample += tableDPCM16[delta];
}
// Emulating x86 16-bit signed register overflow
if (nextSample > 32767) {
nextSample -= 65536;
} else if (nextSample < -32768) {
nextSample += 65536;
}
*out = sample = nextSample;
}
/**
* Decompresses 16-bit DPCM compressed audio. Each byte read
* outputs one sample into the decompression buffer.
*/
static void deDPCM16Mono(int16 *out, Common::ReadStream &audioStream, const uint32 numBytes, int16 &sample) {
for (uint32 i = 0; i < numBytes; ++i) {
const uint8 delta = audioStream.readByte();
deDPCM16Channel(out++, sample, delta);
}
}
// Used by Robot
void deDPCM16Mono(int16 *out, const byte *in, const uint32 numBytes, int16 &sample) {
for (uint32 i = 0; i < numBytes; ++i) {
const uint8 delta = *in++;
deDPCM16Channel(out++, sample, delta);
}
}
static void deDPCM16Stereo(int16 *out, Common::ReadStream &audioStream, const uint32 numBytes, int16 &sampleL, int16 &sampleR) {
assert((numBytes % 2) == 0);
for (uint32 i = 0; i < numBytes / 2; ++i) {
deDPCM16Channel(out++, sampleL, audioStream.readByte());
deDPCM16Channel(out++, sampleR, audioStream.readByte());
}
}
/**
* Decompresses one half of an 8-bit DPCM compressed audio
* byte.
*/
template <bool OLD>
static void deDPCM8Nibble(int16 *out, uint8 &sample, uint8 delta) {
const uint8 lastSample = sample;
if (delta & 8) {
sample -= tableDPCM8[OLD ? (7 - (delta & 7)) : (delta & 7)];
} else {
sample += tableDPCM8[delta & 7];
}
*out = ((lastSample + sample) << 7) ^ 0x8000;
}
/**
* Decompresses one half of an 8-bit DPCM compressed audio
* byte. Attempts to repair overflows on the fly.
*/
static void deDPCM8NibbleWithRepair(int16 *const out, uint8 &sample, const uint8 delta,
uint8 &repairState, uint8 &preRepairSample) {
const uint8 lastSample = sample;
// In Gabriel Knight: Sins of the Fathers CD, the DPCM8-encoded speech contains overflows.
// Overflows wrap from positive to negative, or negative to positive. This continues
// until the wave settles at a new (wrapped) zero DC offset.
// We can't look ahead to see where the wave "reconnects" to valid data, so we
// decay the wave on an artificial slope until it does. This seems to take on average
// around 9-10 samples. The slope value below was chosen through analysing spectrographs
// of the decoded/repaired wave data to find the slope which removed the pops most
// cleanly across a test selection of game speech.
#define REPAIR_SLOPE 12
switch (repairState) {
case 0: {
const int16 newSampleOverflow = (int16)sample + tableDPCM8[delta & 15];
if (newSampleOverflow > 255) {
// Positive overflow has occurred; begin artificial negative slope.
repairState = 1;
sample = lastSample - REPAIR_SLOPE;
// We also begin tracking the un-repaired waveform, so we can tell when to stop.
preRepairSample = (uint8)newSampleOverflow;
debugC(1, kDebugLevelSound, "DPCM8 OVERFLOW (+)");
} else if (newSampleOverflow < 0) {
// Negative overflow has occurred; begin artificial positive slope.
repairState = 2;
sample = lastSample + REPAIR_SLOPE;
// We also begin tracking the un-repaired waveform, so we can tell when to stop.
preRepairSample = (uint8)newSampleOverflow;
debugC(1, kDebugLevelSound, "DPCM8 OVERFLOW (-)");
} else {
sample = (uint8)newSampleOverflow;
}
} break;
case 1: {
// Check for a slope wrap. This circumstance should never happen in reality;
// the unrepaired wave would somehow need to be stuck near minimum
// value over the entire course of the slope.
if (lastSample < REPAIR_SLOPE)
warning("Negative slope wrap!");
const uint8 slopeSample = lastSample - REPAIR_SLOPE;
preRepairSample += tableDPCM8[delta & 15];
// Stop the repair if the artificial slope has intersected with real data.
if (preRepairSample >= slopeSample) {
// Return to real data.
repairState = 0;
sample = preRepairSample;
} else {
sample = slopeSample;
}
} break;
case 2: {
// Check for a slope wrap. This circumstance should never happen in reality;
// the unrepaired wave would somehow need to be stuck near maximum
// value over the entire course of the slope.
if (lastSample > (255 - REPAIR_SLOPE))
warning("Positive slope wrap!");
const uint8 slopeSample = lastSample + REPAIR_SLOPE;
preRepairSample += tableDPCM8[delta & 15];
// Stop the repair if the artificial slope has intersected with real data.
if (preRepairSample <= slopeSample) {
// Return to real data.
repairState = 0;
sample = preRepairSample;
} else {
sample = slopeSample;
}
} break;
default:
warning("Invalid repair state!");
repairState = 0;
break;
}
*out = ((lastSample + sample) << 7) ^ 0x8000;
}
/**
* Decompresses 8-bit DPCM compressed audio. Each byte read
* outputs two samples into the decompression buffer.
*/
template <bool OLD>
static void deDPCM8Mono(int16 *out, Common::ReadStream &audioStream, const uint32 numBytes, uint8 &sample,
const bool popfixEnabled, uint8 &repairState, uint8 &preRepairSample) {
if (popfixEnabled) {
for (uint32 i = 0; i < numBytes; ++i) {
const uint8 delta = audioStream.readByte();
deDPCM8NibbleWithRepair(out++, sample, delta >> 4, repairState, preRepairSample);
deDPCM8NibbleWithRepair(out++, sample, delta & 0xf, repairState, preRepairSample);
}
} else {
for (uint32 i = 0; i < numBytes; ++i) {
const uint8 delta = audioStream.readByte();
deDPCM8Nibble<OLD>(out++, sample, delta >> 4);
deDPCM8Nibble<OLD>(out++, sample, delta & 0xf);
}
}
}
static void deDPCM8Stereo(int16 *out, Common::ReadStream &audioStream, uint32 numBytes, uint8 &sampleL, uint8 &sampleR) {
for (uint32 i = 0; i < numBytes; ++i) {
const uint8 delta = audioStream.readByte();
deDPCM8Nibble<false>(out++, sampleL, delta >> 4);
deDPCM8Nibble<false>(out++, sampleR, delta & 0xf);
}
}
# pragma mark -
template<bool STEREO, bool S16BIT, bool OLDDPCM8>
SOLStream<STEREO, S16BIT, OLDDPCM8>::SOLStream(Common::SeekableReadStream *stream, const DisposeAfterUse::Flag disposeAfterUse, const uint16 sampleRate, const int32 rawDataSize) :
_stream(stream, disposeAfterUse),
_sampleRate(sampleRate),
// SSCI aligns the size of SOL data to 32 bits
_rawDataSize(rawDataSize & ~3),
// The pop fix is only verified with (relevant to?) "old" DPCM8, so we enforce that here.
_popfixDPCM8(g_sci->_features->useAudioPopfix() && OLDDPCM8) {
if (S16BIT) {
_dpcmCarry16.l = _dpcmCarry16.r = 0;
} else {
_dpcmCarry8.l = _dpcmCarry8.r = 0x80;
}
const uint8 compressionRatio = 2;
const uint8 numChannels = STEREO ? 2 : 1;
const uint8 bytesPerSample = S16BIT ? 2 : 1;
_length = ((uint64)_rawDataSize * compressionRatio * 1000) / (_sampleRate * numChannels * bytesPerSample);
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
bool SOLStream<STEREO, S16BIT, OLDDPCM8>::seek(const Audio::Timestamp &where) {
if (where != 0) {
// In order to seek in compressed SOL files, all previous bytes must be
// known since it uses differential compression. Therefore, only seeking
// to the beginning is supported now (SSCI does not offer seeking
// anyway)
return false;
}
if (S16BIT) {
_dpcmCarry16.l = _dpcmCarry16.r = 0;
} else {
_dpcmCarry8.l = _dpcmCarry8.r = 0x80;
}
return _stream->seek(0, SEEK_SET);
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
Audio::Timestamp SOLStream<STEREO, S16BIT, OLDDPCM8>::getLength() const {
return _length;
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
int SOLStream<STEREO, S16BIT, OLDDPCM8>::readBuffer(int16 *buffer, const int numSamples) {
// Reading an odd number of 8-bit samples will result in a loss of samples
// since one byte represents two samples and we do not store the second
// nibble in this case; it should never happen in reality
assert(S16BIT || (numSamples % 2) == 0);
const int samplesPerByte = S16BIT ? 1 : 2;
int32 bytesToRead = numSamples / samplesPerByte;
if (_stream->pos() + bytesToRead > _rawDataSize) {
bytesToRead = _rawDataSize - _stream->pos();
}
if (S16BIT) {
if (STEREO) {
deDPCM16Stereo(buffer, *_stream, bytesToRead, _dpcmCarry16.l, _dpcmCarry16.r);
} else {
deDPCM16Mono(buffer, *_stream, bytesToRead, _dpcmCarry16.l);
}
} else {
if (STEREO) {
deDPCM8Stereo(buffer, *_stream, bytesToRead, _dpcmCarry8.l, _dpcmCarry8.r);
} else {
deDPCM8Mono<OLDDPCM8>(buffer, *_stream, bytesToRead, _dpcmCarry8.l,
_popfixDPCM8.enabled, _popfixDPCM8.state, _popfixDPCM8.preRepairSample);
}
}
const int samplesRead = bytesToRead * samplesPerByte;
return samplesRead;
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
bool SOLStream<STEREO, S16BIT, OLDDPCM8>::isStereo() const {
return STEREO;
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
int SOLStream<STEREO, S16BIT, OLDDPCM8>::getRate() const {
return _sampleRate;
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
bool SOLStream<STEREO, S16BIT, OLDDPCM8>::endOfData() const {
return _stream->eos() || _stream->pos() >= _rawDataSize;
}
template <bool STEREO, bool S16BIT, bool OLDDPCM8>
bool SOLStream<STEREO, S16BIT, OLDDPCM8>::rewind() {
return seek(0);
}
Audio::SeekableAudioStream *makeSOLStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse) {
int32 initialPosition = stream->pos();
byte header[6];
if (stream->read(header, sizeof(header)) != sizeof(header)) {
stream->seek(initialPosition, SEEK_SET);
return nullptr;
}
if ((header[0] & 0x7f) != kResourceTypeAudio || READ_BE_UINT32(header + 2) != MKTAG('S', 'O', 'L', 0)) {
stream->seek(initialPosition, SEEK_SET);
return nullptr;
}
const uint8 headerSize = header[1] + kResourceHeaderSize;
const uint16 sampleRate = stream->readUint16LE();
const byte flags = stream->readByte();
const uint32 dataSize = stream->readUint32LE();
initialPosition += headerSize;
if (flags & kCompressed) {
if (flags & kStereo && flags & k16Bit) {
return new SOLStream<true, true, false>(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), DisposeAfterUse::YES, sampleRate, dataSize);
} else if (flags & kStereo) {
if (getSciVersion() < SCI_VERSION_2_1_EARLY) {
error("SCI2 and earlier did not support stereo SOL audio");
}
return new SOLStream<true, false, false>(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), DisposeAfterUse::YES, sampleRate, dataSize);
} else if (flags & k16Bit) {
return new SOLStream<false, true, false>(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), DisposeAfterUse::YES, sampleRate, dataSize);
} else {
if (getSciVersion() < SCI_VERSION_2_1_EARLY) {
return new SOLStream<false, false, true>(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), DisposeAfterUse::YES, sampleRate, dataSize);
} else {
return new SOLStream<false, false, false>(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), DisposeAfterUse::YES, sampleRate, dataSize);
}
}
}
byte rawFlags = Audio::FLAG_LITTLE_ENDIAN;
if (flags & k16Bit) {
rawFlags |= Audio::FLAG_16BITS;
} else {
rawFlags |= Audio::FLAG_UNSIGNED;
}
if (flags & kStereo) {
rawFlags |= Audio::FLAG_STEREO;
}
return Audio::makeRawStream(new Common::SeekableSubReadStream(stream, initialPosition, initialPosition + dataSize, disposeAfterUse), sampleRate, rawFlags, disposeAfterUse);
}
}