scummvm/sound/adpcm.cpp
2006-09-21 20:12:16 +00:00

362 lines
10 KiB
C++

/* ScummVM - Scumm Interpreter
* Copyright (C) 2005-2006 The ScummVM project
*
* 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 2
* 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, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*
*/
#include "common/stdafx.h"
#include "common/endian.h"
#include "sound/adpcm.h"
#include "sound/audiostream.h"
namespace Audio {
// TODO: Switch from a SeekableReadStream to a plain ReadStream. This requires
// some internal refactoring but is definitely possible and will increase the
// flexibility of this code.
class ADPCMInputStream : public AudioStream {
private:
Common::SeekableReadStream *_stream;
uint32 _endpos;
int _channels;
typesADPCM _type;
uint32 _blockAlign;
uint32 _blockPos;
int _blockLen;
int _rate;
struct ADPCMChannelStatus {
byte predictor;
int16 delta;
int16 coeff1;
int16 coeff2;
int16 sample1;
int16 sample2;
};
struct adpcmStatus {
// IMA
int32 last;
int32 stepIndex;
// MS ADPCM
ADPCMChannelStatus ch[2];
} _status;
int16 stepAdjust(byte);
int16 decodeOKI(byte);
int16 decodeMSIMA(byte);
int16 decodeMS(ADPCMChannelStatus *c, byte);
public:
ADPCMInputStream(Common::SeekableReadStream *stream, uint32 size, typesADPCM type, int rate, int channels = 2, uint32 blockAlign = 0);
~ADPCMInputStream() {};
int readBuffer(int16 *buffer, const int numSamples);
int readBufferOKI(int16 *buffer, const int numSamples);
int readBufferMSIMA1(int16 *buffer, const int numSamples);
int readBufferMSIMA2(int16 *buffer, const int numSamples);
int readBufferMS(int channels, int16 *buffer, const int numSamples);
bool endOfData() const { return (_stream->eos() || _stream->pos() >= _endpos); }
bool isStereo() const { return false; }
int getRate() const { return _rate; }
};
// Routines to convert 12 bit linear samples to the
// Dialogic or Oki ADPCM coding format aka VOX.
// See also <http://www.comptek.ru/telephony/tnotes/tt1-13.html>
//
// In addition, also MS IMA ADPCM is supported. See
// <http://wiki.multimedia.cx/index.php?title=Microsoft_IMA_ADPCM>.
ADPCMInputStream::ADPCMInputStream(Common::SeekableReadStream *stream, uint32 size, typesADPCM type, int rate, int channels, uint32 blockAlign)
: _stream(stream), _channels(channels), _type(type), _blockAlign(blockAlign), _rate(rate) {
_status.last = 0;
_status.stepIndex = 0;
memset(_status.ch, 0, sizeof(_status.ch));
_endpos = stream->pos() + size;
_blockLen = 0;
_blockPos = _blockAlign; // To make sure first header is read
if (type == kADPCMMSIma && blockAlign == 0)
error("ADPCMInputStream(): blockAlign isn't specifiled for MS IMA ADPCM");
if (type == kADPCMMS && blockAlign == 0)
error("ADPCMInputStream(): blockAlign isn't specifiled for MS ADPCM");
}
int ADPCMInputStream::readBuffer(int16 *buffer, const int numSamples) {
switch (_type) {
case kADPCMOki:
return readBufferOKI(buffer, numSamples);
case kADPCMMSIma:
if (_channels == 1)
return readBufferMSIMA1(buffer, numSamples);
else
return readBufferMSIMA2(buffer, numSamples);
case kADPCMMS:
return readBufferMS(_channels, buffer, numSamples);
default:
error("Unsupported ADPCM encoding");
break;
}
return 0;
}
int ADPCMInputStream::readBufferOKI(int16 *buffer, const int numSamples) {
int samples;
byte data;
assert(numSamples % 2 == 0);
for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
data = _stream->readByte();
buffer[samples] = TO_LE_16(decodeOKI((data >> 4) & 0x0f));
buffer[samples + 1] = TO_LE_16(decodeOKI(data & 0x0f));
}
return samples;
}
int ADPCMInputStream::readBufferMSIMA1(int16 *buffer, const int numSamples) {
int samples;
byte data;
assert(numSamples % 2 == 0);
samples = 0;
while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
if (_blockPos == _blockAlign) {
// read block header
_status.last = _stream->readSint16LE();
_status.stepIndex = _stream->readSint16LE();
_blockPos = 4;
}
for (; samples < numSamples && _blockPos < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
data = _stream->readByte();
_blockPos++;
buffer[samples] = TO_LE_16(decodeMSIMA(data & 0x0f));
buffer[samples + 1] = TO_LE_16(decodeMSIMA((data >> 4) & 0x0f));
}
}
return samples;
}
// Microsoft as usual tries to implement it differently. This method
// is used for stereo data.
int ADPCMInputStream::readBufferMSIMA2(int16 *buffer, const int numSamples) {
int samples;
uint32 data;
int nibble;
for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos;) {
for (int channel = 0; channel < 2; channel++) {
data = _stream->readUint32LE();
for (nibble = 0; nibble < 8; nibble++) {
byte k = ((data & 0xf0000000) >> 28);
buffer[samples + channel + nibble * 2] = TO_LE_16(decodeMSIMA(k));
data <<= 4;
}
}
samples += 16;
}
return samples;
}
static const int MSADPCMAdaptCoeff1[] = {
256, 512, 0, 192, 240, 460, 392
};
static const int MSADPCMAdaptCoeff2[] = {
0, -256, 0, 64, 0, -208, -232
};
int ADPCMInputStream::readBufferMS(int channels, int16 *buffer, const int numSamples) {
int samples;
byte data;
int stereo = channels - 1; // We use it in index
samples = 0;
while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
if (_blockPos == _blockAlign) {
// read block header
_status.ch[0].predictor = CLIP(_stream->readByte(), (byte)0, (byte)6);
_status.ch[0].coeff1 = MSADPCMAdaptCoeff1[_status.ch[0].predictor];
_status.ch[0].coeff2 = MSADPCMAdaptCoeff2[_status.ch[0].predictor];
if (stereo) {
_status.ch[1].predictor = CLIP(_stream->readByte(), (byte)0, (byte)6);
_status.ch[1].coeff1 = MSADPCMAdaptCoeff1[_status.ch[1].predictor];
_status.ch[1].coeff2 = MSADPCMAdaptCoeff2[_status.ch[1].predictor];
}
_status.ch[0].delta = _stream->readSint16LE();
if (stereo)
_status.ch[1].delta = _stream->readSint16LE();
_status.ch[0].sample1 = _stream->readSint16LE();
if (stereo)
_status.ch[1].sample1 = _stream->readSint16LE();
buffer[samples++] = _status.ch[0].sample2 = _stream->readSint16LE();
if (stereo)
buffer[samples++] = _status.ch[1].sample2 = _stream->readSint16LE();
buffer[samples++] = _status.ch[0].sample1;
if (stereo)
buffer[samples++] = _status.ch[1].sample1;
_blockPos = channels * 7;
}
for (; samples < numSamples && _blockPos < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
data = _stream->readByte();
_blockPos++;
buffer[samples] = TO_LE_16(decodeMS(&_status.ch[0], (data >> 4) & 0x0f));
buffer[samples + 1] = TO_LE_16(decodeMS(&_status.ch[stereo], data & 0x0f));
}
}
return samples;
}
static const int MSADPCMAdaptationTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
int16 ADPCMInputStream::decodeMS(ADPCMChannelStatus *c, byte code) {
int32 predictor;
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
predictor += (signed)((code & 0x08) ? (code - 0x10) : (code)) * c->delta;
if (predictor < -0x8000)
predictor = -0x8000;
else if (predictor > 0x7fff)
predictor = 0x7fff;
c->sample2 = c->sample1;
c->sample1 = predictor;
c->delta = (MSADPCMAdaptationTable[(int)code] * c->delta) >> 8;
if (c->delta < 16)
c->delta = 16;
return (int16)predictor;
}
// adjust the step for use on the next sample.
int16 ADPCMInputStream::stepAdjust(byte code) {
static const int16 adjusts[] = {-1, -1, -1, -1, 2, 4, 6, 8};
return adjusts[code & 0x07];
}
static const int16 okiStepSize[49] = {
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552
};
// Decode Linear to ADPCM
int16 ADPCMInputStream::decodeOKI(byte code) {
int16 diff, E, samp;
E = (2 * (code & 0x7) + 1) * okiStepSize[_status.stepIndex] / 8;
diff = (code & 0x08) ? -E : E;
samp = _status.last + diff;
// Clip the values to +/- 2^11 (supposed to be 12 bits)
if (samp > 2047)
samp = 2047;
if (samp < -2048)
samp = -2048;
_status.last = samp;
_status.stepIndex += stepAdjust(code);
if (_status.stepIndex < 0)
_status.stepIndex = 0;
if (_status.stepIndex > ARRAYSIZE(okiStepSize) - 1)
_status.stepIndex = ARRAYSIZE(okiStepSize) - 1;
// * 16 effectively converts 12-bit input to 16-bit output
return samp * 16;
}
static const uint16 imaStepTable[89] = {
7, 8, 9, 10, 11, 12, 13, 14,
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
7132, 7845, 8630, 9493,10442,11487,12635,13899,
15289,16818,18500,20350,22385,24623,27086,29794,
32767
};
int16 ADPCMInputStream::decodeMSIMA(byte code) {
int32 diff, E, samp;
E = (2 * (code & 0x7) + 1) * imaStepTable[_status.stepIndex] / 8;
diff = (code & 0x08) ? -E : E;
samp = _status.last + diff;
if (samp < -0x8000)
samp = -0x8000;
else if (samp > 0x7fff)
samp = 0x7fff;
_status.last = samp;
_status.stepIndex += stepAdjust(code);
if (_status.stepIndex < 0)
_status.stepIndex = 0;
if (_status.stepIndex > ARRAYSIZE(imaStepTable) - 1)
_status.stepIndex = ARRAYSIZE(imaStepTable) - 1;
return samp;
}
AudioStream *makeADPCMStream(Common::SeekableReadStream *stream, uint32 size, typesADPCM type, int rate, int channels, uint32 blockAlign) {
return new ADPCMInputStream(stream, size, type, rate, channels, blockAlign);
}
} // End of namespace Audio