[Scummvm-cvs-logs] SF.net SVN: scummvm:[47421] tools/trunk
mthreepwood at users.sourceforge.net
mthreepwood at users.sourceforge.net
Thu Jan 21 21:43:07 CET 2010
Revision: 47421
http://scummvm.svn.sourceforge.net/scummvm/?rev=47421&view=rev
Author: mthreepwood
Date: 2010-01-21 20:42:45 +0000 (Thu, 21 Jan 2010)
Log Message:
-----------
Remove unused sound-based code from engines/mohawk/utils.
Modified Paths:
--------------
tools/trunk/Makefile.common
Removed Paths:
-------------
tools/trunk/engines/mohawk/utils/adpcm.cpp
tools/trunk/engines/mohawk/utils/adpcm.h
tools/trunk/engines/mohawk/utils/audiostream.cpp
tools/trunk/engines/mohawk/utils/audiostream.h
tools/trunk/engines/mohawk/utils/voc.cpp
tools/trunk/engines/mohawk/utils/voc.h
tools/trunk/engines/mohawk/utils/wave.cpp
tools/trunk/engines/mohawk/utils/wave.h
Modified: tools/trunk/Makefile.common
===================================================================
--- tools/trunk/Makefile.common 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/Makefile.common 2010-01-21 20:42:45 UTC (rev 47421)
@@ -141,10 +141,6 @@
common/memorypool.o \
common/str.o \
engines/mohawk/utils/file.o \
- engines/mohawk/utils/adpcm.o \
- engines/mohawk/utils/audiostream.o \
- engines/mohawk/utils/voc.o \
- engines/mohawk/utils/wave.o \
common/md5.o \
common/util.o
@@ -177,10 +173,6 @@
common/memorypool.o \
common/str.o \
engines/mohawk/utils/file.o \
- engines/mohawk/utils/adpcm.o \
- engines/mohawk/utils/audiostream.o \
- engines/mohawk/utils/voc.o \
- engines/mohawk/utils/wave.o \
common/md5.o \
common/util.o
Deleted: tools/trunk/engines/mohawk/utils/adpcm.cpp
===================================================================
--- tools/trunk/engines/mohawk/utils/adpcm.cpp 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/adpcm.cpp 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,354 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/adpcm.cpp $
- * $Id: adpcm.cpp 40868 2009-05-24 15:19:28Z lordhoto $
- *
- */
-
-#include "adpcm.h"
-#include "util.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;
- _blockPos = _blockLen = 0;
-
- 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);
- break;
- case kADPCMMSIma:
- if (_channels == 1)
- return readBufferMSIMA1(buffer, numSamples);
- else
- return readBufferMSIMA2(buffer, numSamples);
- break;
- case kADPCMMS:
- return readBufferMS(_channels, buffer, numSamples);
- break;
- 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();
- WRITE_LE_UINT16(buffer + samples, decodeOKI((data >> 4) & 0x0f));
- WRITE_LE_UINT16(buffer + samples + 1, 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++;
- WRITE_LE_UINT16(buffer + samples, decodeMSIMA(data & 0x0f));
- WRITE_LE_UINT16(buffer + samples + 1, 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);
- WRITE_LE_UINT16(buffer + samples + channel + nibble * 2, 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 = Common::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 = Common::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();
-
- buffer[samples++] = _status.ch[0].sample1 = _stream->readSint16LE();
- if (stereo)
- buffer[samples++] = _status.ch[1].sample1 = _stream->readSint16LE();
-
- buffer[samples++] = _status.ch[0].sample2 = _stream->readSint16LE();
- if (stereo)
- buffer[samples++] = _status.ch[1].sample2 = _stream->readSint16LE();
-
- _blockPos = channels * 7;
- }
-
-
- for (; samples < numSamples && _blockPos < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
- data = _stream->readByte();
- _blockPos++;
- WRITE_LE_UINT16(buffer + samples, decodeMS(&_status.ch[0], (data >> 4) & 0x0f));
- WRITE_LE_UINT16(buffer + samples + 1, decodeMS(&_status.ch[stereo], data & 0x0f));
- }
- }
-
- return samples;
-}
-
-
-// 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 > 2048)
- samp = 2048;
- 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;
-}
-
-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;
-}
-
-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
Deleted: tools/trunk/engines/mohawk/utils/adpcm.h
===================================================================
--- tools/trunk/engines/mohawk/utils/adpcm.h 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/adpcm.h 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,43 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/adpcm.h $
- * $Id: adpcm.h 23138 2006-06-15 15:44:06Z h00ligan $
- *
- */
-
-#ifndef SOUND_ADPCM_H
-#define SOUND_ADPCM_H
-
-#include "audiostream.h"
-#include "stream.h"
-
-namespace Audio {
-
-class AudioStream;
-
-enum typesADPCM {
- kADPCMOki,
- kADPCMMSIma,
- kADPCMMS
-};
-
-AudioStream *makeADPCMStream(Common::SeekableReadStream *stream, uint32 size, typesADPCM type, int rate = 22050, int channels = 2, uint32 blockAlign = 0);
-
-} // End of namespace Audio
-
-#endif
Deleted: tools/trunk/engines/mohawk/utils/audiostream.cpp
===================================================================
--- tools/trunk/engines/mohawk/utils/audiostream.cpp 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/audiostream.cpp 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,162 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/audiostream.cpp $
- * $Id: audiostream.cpp 23138 2006-06-15 15:44:06Z h00ligan $
- *
- */
-
-#include "audiostream.h"
-
-
-namespace Audio {
-
-struct StreamFileFormat {
- /** Decodername */
- const char* decoderName;
- const char* fileExtension;
- /**
- * Pointer to a function which tries to open a file of type StreamFormat.
- * Return NULL in case of an error (invalid/nonexisting file).
- */
- AudioStream* (*openStreamFile)(FILE *file, uint32 size);
-};
-
-
-#pragma mark -
-#pragma mark --- LinearMemoryStream ---
-#pragma mark -
-
-
-/**
- * A simple raw audio stream, purely memory based. It operates on a single
- * block of data, which is passed to it upon creation.
- * Optionally supports looping the sound.
- *
- * Design note: This code tries to be as optimized as possible (without
- * resorting to assembly, that is). To this end, it is written as a template
- * class. This way the compiler can actually create optimized code for each
- * special code. This results in a total of 12 versions of the code being
- * generated.
- */
-template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
-class LinearMemoryStream : public AudioStream {
-protected:
- const byte *_ptr;
- const byte *_end;
- const byte *_loopPtr;
- const byte *_loopEnd;
- const int _rate;
- const byte *_origPtr;
-
- inline bool eosIntern() const { return _ptr >= _end; };
-public:
- LinearMemoryStream(int rate, const byte *ptr, uint32 len, uint32 loopOffset, uint32 loopLen, bool autoFreeMemory)
- : _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0), _rate(rate) {
-
- // Verify the buffer sizes are sane
- if (is16Bit && stereo)
- assert((len & 3) == 0 && (loopLen & 3) == 0);
- else if (is16Bit || stereo)
- assert((len & 1) == 0 && (loopLen & 1) == 0);
-
- if (loopLen) {
- _loopPtr = _ptr + loopOffset;
- _loopEnd = _loopPtr + loopLen;
- }
- if (stereo) // Stereo requires even sized data
- assert(len % 2 == 0);
-
- _origPtr = autoFreeMemory ? ptr : 0;
- }
- ~LinearMemoryStream() {
- free(const_cast<byte *>(_origPtr));
- }
- int readBuffer(int16 *buffer, const int numSamples);
-
- bool isStereo() const { return stereo; }
- bool endOfData() const { return eosIntern(); }
-
- int getRate() const { return _rate; }
-};
-
-template<bool stereo, bool is16Bit, bool isUnsigned, bool isLE>
-int LinearMemoryStream<stereo, is16Bit, isUnsigned, isLE>::readBuffer(int16 *buffer, const int numSamples) {
- int samples = 0;
- while (samples < numSamples && !eosIntern()) {
- const int len = Common::MIN(numSamples, samples + (int)(_end - _ptr) / (is16Bit ? 2 : 1));
- while (samples < len) {
- *buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, _ptr, isLE);
- _ptr += (is16Bit ? 2 : 1);
- samples++;
- }
- // Loop, if looping was specified
- if (_loopPtr && eosIntern()) {
- _ptr = _loopPtr;
- _end = _loopEnd;
- }
- }
- return samples;
-}
-
-
-#pragma mark -
-#pragma mark --- Input stream factory ---
-#pragma mark -
-
-/* In the following, we use preprocessor / macro tricks to simplify the code
- * which instantiates the input streams. We used to use template functions for
- * this, but MSVC6 / EVC 3-4 (used for WinCE builds) are extremely buggy when it
- * comes to this feature of C++... so as a compromise we use macros to cut down
- * on the (source) code duplication a bit.
- * So while normally macro tricks are said to make maintenance harder, in this
- * particular case it should actually help it :-)
- */
-
-#define MAKE_LINEAR(STEREO, UNSIGNED) \
- if (is16Bit) { \
- if (isLE) \
- return new LinearMemoryStream<STEREO, true, UNSIGNED, true>(rate, ptr, len, loopOffset, loopLen, autoFree); \
- else \
- return new LinearMemoryStream<STEREO, true, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree); \
- } else \
- return new LinearMemoryStream<STEREO, false, UNSIGNED, false>(rate, ptr, len, loopOffset, loopLen, autoFree)
-
-AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32 len, uint32 loopOffset, uint32 loopLen) {
- const bool isStereo = (flags & Audio::Mixer::FLAG_STEREO) != 0;
- const bool is16Bit = (flags & Audio::Mixer::FLAG_16BITS) != 0;
- const bool isUnsigned = (flags & Audio::Mixer::FLAG_UNSIGNED) != 0;
- const bool isLE = (flags & Audio::Mixer::FLAG_LITTLE_ENDIAN) != 0;
- const bool autoFree = (flags & Audio::Mixer::FLAG_AUTOFREE) != 0;
-
- if (isStereo) {
- if (isUnsigned) {
- MAKE_LINEAR(true, true);
- } else {
- MAKE_LINEAR(true, false);
- }
- } else {
- if (isUnsigned) {
- MAKE_LINEAR(false, true);
- } else {
- MAKE_LINEAR(false, false);
- }
- }
-}
-
-
-} // End of namespace Audio
Deleted: tools/trunk/engines/mohawk/utils/audiostream.h
===================================================================
--- tools/trunk/engines/mohawk/utils/audiostream.h 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/audiostream.h 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,153 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/audiostream.h $
- * $Id: audiostream.h 23138 2006-06-15 15:44:06Z h00ligan $
- *
- */
-
-#ifndef SOUND_AUDIOSTREAM_H
-#define SOUND_AUDIOSTREAM_H
-
-#include "../util.h"
-#include "util.h"
-
-namespace Audio {
-
-class Mixer {
-public:
- enum {
- /** unsigned samples (default: signed) */
- FLAG_UNSIGNED = 1 << 0,
-
- /** sound is 16 bits wide (default: 8bit) */
- FLAG_16BITS = 1 << 1,
-
- /** sample is little endian (default: big endian) */
- FLAG_LITTLE_ENDIAN = 1 << 2,
-
- /** sound is in stereo (default: mono) */
- FLAG_STEREO = 1 << 3,
-
- /** reverse the left and right stereo channel */
- FLAG_REVERSE_STEREO = 1 << 4,
-
- /** sound buffer is freed automagically at the end of playing */
- FLAG_AUTOFREE = 1 << 5,
-
- /** loop the audio */
- FLAG_LOOP = 1 << 6
- };
-};
-
-/**
- * Generic input stream for the resampling code.
- */
-class AudioStream {
-public:
- virtual ~AudioStream() {}
-
- /**
- * Fill the given buffer with up to numSamples samples.
- * Returns the actual number of samples read, or -1 if
- * a critical error occured (note: you *must* check if
- * this value is less than what you requested, this can
- * happen when the stream is fully used up).
- *
- * Data has to be in native endianess, 16 bit per sample, signed.
- * For stereo stream, buffer will be filled with interleaved
- * left and right channel samples, starting with a left sample.
- * Furthermore, the samples in the left and right are summed up.
- * So if you request 4 samples from a stereo stream, you will get
- * a total of two left channel and two right channel samples.
- */
- virtual int readBuffer(int16 *buffer, const int numSamples) = 0;
-
- /** Is this a stereo stream? */
- virtual bool isStereo() const = 0;
-
- /**
- * End of data reached? If this returns true, it means that at this
- * time there is no data available in the stream. However there may be
- * more data in the future.
- * This is used by e.g. a rate converter to decide whether to keep on
- * converting data or stop.
- */
- virtual bool endOfData() const = 0;
-
- /**
- * End of stream reached? If this returns true, it means that all data
- * in this stream is used up and no additional data will appear in it
- * in the future.
- * This is used by the mixer to decide whether a given stream shall be
- * removed from the list of active streams (and thus be destroyed).
- * By default this maps to endOfData()
- */
- virtual bool endOfStream() const { return endOfData(); }
-
- /** Sample rate of the stream. */
- virtual int getRate() const = 0;
-};
-
-/**
- * A simple AudioStream which represents a 'silent' stream,
- * containing the specified number of zero samples.
- */
-class ZeroInputStream : public AudioStream {
-private:
- int _len;
-public:
- ZeroInputStream(uint32 len) : _len(len) { }
- int readBuffer(int16 *buffer, const int numSamples) {
- int samples = Common::MIN(_len, numSamples);
- memset(buffer, 0, samples * 2);
- _len -= samples;
- return samples;
- }
- bool isStereo() const { return false; }
- bool eos() const { return _len <= 0; }
-
- int getRate() const { return -1; }
-};
-
-AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32 len, uint32 loopOffset, uint32 loopLen);
-
-/**
- * An audio stream to which additional data can be appended on-the-fly.
- * Used by SMUSH, iMuseDigital, and the Kyrandia 3 VQA player.
- */
-class AppendableAudioStream : public Audio::AudioStream {
-public:
- virtual void append(const byte *data, uint32 len) = 0;
- virtual void finish() = 0;
-};
-
-AppendableAudioStream *makeAppendableAudioStream(int rate, byte _flags, uint32 len);
-
-
-// This used to be an inline template function, but
-// buggy template function handling in MSVC6 forced
-// us to go with the macro approach. So far this is
-// the only template function that MSVC6 seemed to
-// compile incorrectly. Knock on wood.
-#define READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, ptr, isLE) \
- ((is16Bit ? (isLE ? READ_LE_UINT16(ptr) : READ_BE_UINT16(ptr)) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0))
-
-
-} // End of namespace Audio
-
-#endif
Deleted: tools/trunk/engines/mohawk/utils/voc.cpp
===================================================================
--- tools/trunk/engines/mohawk/utils/voc.cpp 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/voc.cpp 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,140 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/voc.cpp $
- * $Id: voc.cpp 30977 2008-02-26 19:41:33Z fingolfin $
- *
- */
-
-#include "voc.h"
-#include "stream.h"
-#include "audiostream.h"
-
-
-namespace Audio {
-
-int getSampleRateFromVOCRate(int vocSR) {
- if (vocSR == 0xa5 || vocSR == 0xa6) {
- return 11025;
- } else if (vocSR == 0xd2 || vocSR == 0xd3) {
- return 22050;
- } else {
- int sr = 1000000L / (256L - vocSR);
- // inexact sampling rates occur e.g. in the kitchen in Monkey Island,
- // very easy to reach right from the start of the game.
- //warning("inexact sample rate used: %i (0x%x)", sr, vocSR);
- return sr;
- }
-}
-
-byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate, int &loops, int &begin_loop, int &end_loop) {
- VocFileHeader fileHeader;
-
- if (stream.read(&fileHeader, 8) != 8)
- goto invalid;
-
- if (!memcmp(&fileHeader, "VTLK", 4)) {
- if (stream.read(&fileHeader, sizeof(VocFileHeader)) != sizeof(VocFileHeader))
- goto invalid;
- } else if (!memcmp(&fileHeader, "Creative", 8)) {
- if (stream.read(((byte *)&fileHeader) + 8, sizeof(VocFileHeader) - 8) != sizeof(VocFileHeader) - 8)
- goto invalid;
- } else {
- invalid:;
- warning("loadVOCFromStream: Invalid header");
- return NULL;
- }
-
- if (memcmp(fileHeader.desc, "Creative Voice File", 19) != 0)
- error("loadVOCFromStream: Invalid header");
- if (fileHeader.desc[19] != 0x1A)
- debug(3, "loadVOCFromStream: Partially invalid header");
-
- int32 offset = READ_LE_UINT16(&fileHeader.datablock_offset);
- int16 version = READ_LE_UINT16(&fileHeader.version);
- int16 code = READ_LE_UINT16(&fileHeader.id);
- assert(offset == sizeof(VocFileHeader));
- // 0x100 is an invalid VOC version used by German version of DOTT (Disk) and
- // French version of Simon the Sorcerer 2 (CD)
- assert(version == 0x010A || version == 0x0114 || version == 0x0100);
- assert(code == ~version + 0x1234);
-
- int len;
- byte *ret_sound = 0;
- size = 0;
- begin_loop = 0;
- end_loop = 0;
-
- while ((code = stream.readByte())) {
- len = stream.readByte();
- len |= stream.readByte() << 8;
- len |= stream.readByte() << 16;
-
- switch(code) {
- case 1: {
- int time_constant = stream.readByte();
- int packing = stream.readByte();
- len -= 2;
- rate = getSampleRateFromVOCRate(time_constant);
- debug(9, "VOC Data Block: %d, %d, %d", rate, packing, len);
- if (packing == 0) {
- if (size) {
- ret_sound = (byte *)realloc(ret_sound, size + len);
- } else {
- ret_sound = (byte *)malloc(len);
- }
- stream.read(ret_sound + size, len);
- size += len;
- begin_loop = size;
- end_loop = size;
- } else {
- warning("VOC file packing %d unsupported", packing);
- }
- } break;
- case 6: // begin of loop
- assert(len == 2);
- loops = stream.readUint16LE();
- break;
- case 7: // end of loop
- assert(len == 0);
- break;
- default:
- warning("Invalid code in VOC file : %d", code);
- return ret_sound;
- }
- }
- debug(4, "VOC Data Size : %d", size);
- return ret_sound;
-}
-
-byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate) {
- int loops, begin_loop, end_loop;
- return loadVOCFromStream(stream, size, rate, loops, begin_loop, end_loop);
-}
-
-AudioStream *makeVOCStream(Common::ReadStream &stream) {
- int size, rate;
- byte *data = loadVOCFromStream(stream, size, rate);
-
- if (!data)
- return 0;
-
- return makeLinearInputStream(rate, Audio::Mixer::FLAG_AUTOFREE | Audio::Mixer::FLAG_UNSIGNED, data, size, 0, 0);
-}
-
-
-} // End of namespace Audio
Deleted: tools/trunk/engines/mohawk/utils/voc.h
===================================================================
--- tools/trunk/engines/mohawk/utils/voc.h 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/voc.h 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,83 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/voc.h $
- * $Id: voc.h 29430 2007-11-06 10:24:46Z fingolfin $
- *
- */
-
-#ifndef SOUND_VOC_H
-#define SOUND_VOC_H
-
-#include "../util.h"
-
-namespace Common { class ReadStream; }
-
-namespace Audio {
-
-class AudioStream;
-
-#include "common/pack-start.h" /* START STRUCT PACKING */
-
-struct VocFileHeader {
- uint8 desc[20];
- uint16 datablock_offset;
- uint16 version;
- uint16 id;
-} GCC_PACK;
-
-struct VocBlockHeader {
- uint8 blocktype;
- uint8 size[3];
- uint8 sr;
- uint8 pack;
-} GCC_PACK;
-
-#include "common/pack-end.h" /* END STRUCT PACKING */
-
-/**
- * Take a sample rate parameter as it occurs in a VOC sound header, and
- * return the corresponding sample frequency.
- *
- * This method has special cases for the standard rates of 11025 and 22050 kHz,
- * which due to limitations of the format, cannot be encoded exactly in a VOC
- * file. As a consequence, many game files have sound data sampled with those
- * rates, but the VOC marks them incorrectly as 11111 or 22222 kHz. This code
- * works around that and "unrounds" the sampling rates.
- */
-extern int getSampleRateFromVOCRate(int vocSR);
-
-/**
- * Try to load a VOC from the given seekable stream. Returns a pointer to memory
- * containing the PCM sample data (allocated with malloc). It is the callers
- * responsibility to dellocate that data again later on! Currently this
- * function only supports uncompressed raw PCM data.
- */
-extern byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate);
-
-/**
- * Try to load a VOC from the given seekable stream and create an AudioStream
- * from that data. Currently this function only supports uncompressed raw PCM
- * data. Looping is not supported.
- *
- * This function uses loadVOCFromStream() internally.
- */
-AudioStream *makeVOCStream(Common::ReadStream &stream);
-
-} // End of namespace Audio
-
-#endif
Deleted: tools/trunk/engines/mohawk/utils/wave.cpp
===================================================================
--- tools/trunk/engines/mohawk/utils/wave.cpp 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/wave.cpp 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,204 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/wave.cpp $
- * $Id: wave.cpp 41558 2009-06-15 21:12:51Z fingolfin $
- *
- */
-
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/wave.cpp $
- * $Id: wave.cpp 41558 2009-06-15 21:12:51Z fingolfin $
- *
- */
-
-#include "stream.h"
-#include "audiostream.h"
-#include "adpcm.h"
-
-namespace Audio {
-
-bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, byte &flags, uint16 *wavType, int *blockAlign_) {
- const uint32 initialPos = stream.pos();
- byte buf[4+1];
-
- buf[4] = 0;
-
- stream.read(buf, 4);
- if (memcmp(buf, "RIFF", 4) != 0) {
- warning("getWavInfo: No 'RIFF' header");
- return false;
- }
-
- uint32 wavLength = stream.readUint32LE();
-
- stream.read(buf, 4);
- if (memcmp(buf, "WAVE", 4) != 0) {
- warning("getWavInfo: No 'WAVE' header");
- return false;
- }
-
- stream.read(buf, 4);
- if (memcmp(buf, "fmt ", 4) != 0) {
- warning("getWavInfo: No 'fmt' header");
- return false;
- }
-
- uint32 fmtLength = stream.readUint32LE();
- if (fmtLength < 16) {
- // A valid fmt chunk always contains at least 16 bytes
- warning("getWavInfo: 'fmt' header is too short");
- return false;
- }
-
- // Next comes the "type" field of the fmt header. Some typical
- // values for it:
- // 1 -> uncompressed PCM
- // 17 -> IMA ADPCM compressed WAVE
- // See <http://www.saettler.com/RIFFNEW/RIFFNEW.htm> for a more complete
- // list of common WAVE compression formats...
- uint16 type = stream.readUint16LE(); // == 1 for PCM data
- uint16 numChannels = stream.readUint16LE(); // 1 for mono, 2 for stereo
- uint32 samplesPerSec = stream.readUint32LE(); // in Hz
- uint32 avgBytesPerSec = stream.readUint32LE(); // == SampleRate * NumChannels * BitsPerSample/8
-
- uint16 blockAlign = stream.readUint16LE(); // == NumChannels * BitsPerSample/8
- uint16 bitsPerSample = stream.readUint16LE(); // 8, 16 ...
- // 8 bit data is unsigned, 16 bit data signed
-
-
- if (wavType != 0)
- *wavType = type;
-
- if (blockAlign_ != 0)
- *blockAlign_ = blockAlign;
-#if 0
- printf("WAVE information:\n");
- printf(" total size: %d\n", wavLength);
- printf(" fmt size: %d\n", fmtLength);
- printf(" type: %d\n", type);
- printf(" numChannels: %d\n", numChannels);
- printf(" samplesPerSec: %d\n", samplesPerSec);
- printf(" avgBytesPerSec: %d\n", avgBytesPerSec);
- printf(" blockAlign: %d\n", blockAlign);
- printf(" bitsPerSample: %d\n", bitsPerSample);
-#endif
-
- if (type != 1 && type != 2 && type != 17) {
- warning("getWavInfo: only PCM, MS ADPCM or IMA ADPCM data is supported (type %d)", type);
- return false;
- }
-
- if (blockAlign != numChannels * bitsPerSample / 8 && type != 2) {
- debug(0, "getWavInfo: blockAlign is invalid");
- }
-
- if (avgBytesPerSec != samplesPerSec * blockAlign && type != 2) {
- debug(0, "getWavInfo: avgBytesPerSec is invalid");
- }
-
- // Prepare the return values.
- rate = samplesPerSec;
-
- flags = 0;
- if (bitsPerSample == 8) // 8 bit data is unsigned
- flags |= Audio::Mixer::FLAG_UNSIGNED;
- else if (bitsPerSample == 16) // 16 bit data is signed little endian
- flags |= (Audio::Mixer::FLAG_16BITS | Audio::Mixer::FLAG_LITTLE_ENDIAN);
- else if (bitsPerSample == 4 && type == 17) // MS IMA ADPCM compressed. We decompress it
- flags |= (Audio::Mixer::FLAG_16BITS | Audio::Mixer::FLAG_LITTLE_ENDIAN);
- else if (bitsPerSample == 4 && type == 2) // MS ADPCM compressed. We decompress it
- flags |= (Audio::Mixer::FLAG_16BITS | Audio::Mixer::FLAG_LITTLE_ENDIAN);
- else {
- warning("getWavInfo: unsupported bitsPerSample %d", bitsPerSample);
- return false;
- }
-
- if (numChannels == 2)
- flags |= Audio::Mixer::FLAG_STEREO;
- else if (numChannels != 1) {
- warning("getWavInfo: unsupported number of channels %d", numChannels);
- return false;
- }
-
- // It's almost certainly a WAV file, but we still need to find its
- // 'data' chunk.
-
- // Skip over the rest of the fmt chunk.
- int offset = fmtLength - 16;
-
- do {
- stream.seek(offset, SEEK_CUR);
- if (stream.pos() >= initialPos + wavLength + 8) {
- warning("getWavInfo: Cannot find 'data' chunk");
- return false;
- }
- stream.read(buf, 4);
- offset = stream.readUint32LE();
-
-#if 0
- printf(" found a '%s' tag of size %d\n", buf, offset);
-#endif
- } while (memcmp(buf, "data", 4) != 0);
-
- // Stream now points at 'offset' bytes of sample data...
- size = offset;
-
- return true;
-}
-
-AudioStream *makeWAVStream(Common::SeekableReadStream &stream) {
- int size, rate;
- byte flags;
- uint16 type;
- int blockAlign;
-
- if (!loadWAVFromStream(stream, size, rate, flags, &type, &blockAlign))
- return 0;
-
- if (type == 17) // MS IMA ADPCM
- return makeADPCMStream(&stream, size, kADPCMMSIma, rate, (flags & Audio::Mixer::FLAG_STEREO) ? 2 : 1);
- if (type == 2) // MS ADPCM
- return makeADPCMStream(&stream, size, kADPCMMS, rate, (flags & Audio::Mixer::FLAG_STEREO) ? 2 : 1, blockAlign);
-
- byte *data = (byte *)malloc(size);
- assert(data);
- stream.read(data, size);
-
- // Since we allocated our own buffer for the data, we must set the autofree flag.
- flags |= Audio::Mixer::FLAG_AUTOFREE;
-
- return makeLinearInputStream(rate, flags, data, size, 0, 0);
-}
-
-} // End of namespace Audio
Deleted: tools/trunk/engines/mohawk/utils/wave.h
===================================================================
--- tools/trunk/engines/mohawk/utils/wave.h 2010-01-21 18:31:46 UTC (rev 47420)
+++ tools/trunk/engines/mohawk/utils/wave.h 2010-01-21 20:42:45 UTC (rev 47421)
@@ -1,53 +0,0 @@
-/* Scumm Tools
- * Copyright (C) 2004-2006 The ScummVM Team
- *
- * 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: https://scummvm.svn.sourceforge.net/svnroot/scummvm/tools/branches/gsoc2009-gui/utils/wave.h $
- * $Id: wave.h 23138 2006-06-15 15:44:06Z h00ligan $
- *
- */
-
-#ifndef SOUND_WAVE_H
-#define SOUND_WAVE_H
-
-
-namespace Common { class SeekableReadStream; }
-
-namespace Audio {
-
-class AudioStream;
-
-/**
- * Try to load a WAVE from the given seekable stream. Returns true if
- * successful. In that case, the stream's seek position will be set to the
- * start of the audio data, and size, rate and flags contain information
- * necessary for playback. Currently this function only supports uncompressed
- * raw PCM data as well as IMA ADPCM.
- */
-extern bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, byte &flags, uint16 *wavType = 0, int *blockAlign = 0);
-
-/**
- * Try to load a WAVE from the given seekable stream and create an AudioStream
- * from that data. Currently this function only supports uncompressed raw PCM
- * data as well as IMA ADPCM.
- *
- * This function uses loadWAVFromStream() internally.
- */
-AudioStream *makeWAVStream(Common::SeekableReadStream &stream);
-
-} // End of namespace Audio
-
-#endif
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