--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/audio/SDL_wave.c	Thu Apr 26 16:45:43 2001 +0000
@@ -0,0 +1,591 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997, 1998, 1999, 2000, 2001  Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Library General Public
+    License as published by the Free Software Foundation; either
+    version 2 of the License, or (at your option) any later version.
+
+    This library 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
+    Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public
+    License along with this library; if not, write to the Free
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+    Sam Lantinga
+    slouken@devolution.com
+*/
+
+#ifdef SAVE_RCSID
+static char rcsid =
+ "@(#) $Id$";
+#endif
+
+#ifndef DISABLE_FILE
+
+/* Microsoft WAVE file loading routines */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "SDL_error.h"
+#include "SDL_audio.h"
+#include "SDL_wave.h"
+#include "SDL_endian.h"
+
+#ifndef NELEMS
+#define NELEMS(array)	((sizeof array)/(sizeof array[0]))
+#endif
+
+static int ReadChunk(SDL_RWops *src, Chunk *chunk);
+
+struct MS_ADPCM_decodestate {
+	Uint8 hPredictor;
+	Uint16 iDelta;
+	Sint16 iSamp1;
+	Sint16 iSamp2;
+};
+static struct MS_ADPCM_decoder {
+	WaveFMT wavefmt;
+	Uint16 wSamplesPerBlock;
+	Uint16 wNumCoef;
+	Sint16 aCoeff[7][2];
+	/* * * */
+	struct MS_ADPCM_decodestate state[2];
+} MS_ADPCM_state;
+
+static int InitMS_ADPCM(WaveFMT *format)
+{
+	Uint8 *rogue_feel;
+	Uint16 extra_info;
+	int i;
+
+	/* Set the rogue pointer to the MS_ADPCM specific data */
+	MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+	MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+	MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+	MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+	MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+	MS_ADPCM_state.wavefmt.bitspersample =
+					 SDL_SwapLE16(format->bitspersample);
+	rogue_feel = (Uint8 *)format+sizeof(*format);
+	if ( sizeof(*format) == 16 ) {
+		extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	rogue_feel += sizeof(Uint16);
+	MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	rogue_feel += sizeof(Uint16);
+	if ( MS_ADPCM_state.wNumCoef != 7 ) {
+		SDL_SetError("Unknown set of MS_ADPCM coefficients");
+		return(-1);
+	}
+	for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) {
+		MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+		MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	return(0);
+}
+
+static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
+					Uint8 nybble, Sint16 *coeff)
+{
+	const Sint32 max_audioval = ((1<<(16-1))-1);
+	const Sint32 min_audioval = -(1<<(16-1));
+	const Sint32 adaptive[] = {
+		230, 230, 230, 230, 307, 409, 512, 614,
+		768, 614, 512, 409, 307, 230, 230, 230
+	};
+	Sint32 new_sample, delta;
+
+	new_sample = ((state->iSamp1 * coeff[0]) +
+		      (state->iSamp2 * coeff[1]))/256;
+	if ( nybble & 0x08 ) {
+		new_sample += state->iDelta * (nybble-0x10);
+	} else {
+		new_sample += state->iDelta * nybble;
+	}
+	if ( new_sample < min_audioval ) {
+		new_sample = min_audioval;
+	} else
+	if ( new_sample > max_audioval ) {
+		new_sample = max_audioval;
+	}
+	delta = ((Sint32)state->iDelta * adaptive[nybble])/256;
+	if ( delta < 16 ) {
+		delta = 16;
+	}
+	state->iDelta = delta;
+	state->iSamp2 = state->iSamp1;
+	state->iSamp1 = new_sample;
+	return(new_sample);
+}
+
+static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
+{
+	struct MS_ADPCM_decodestate *state[2];
+	Uint8 *freeable, *encoded, *decoded;
+	Sint32 encoded_len, samplesleft;
+	Sint8 nybble, stereo;
+	Sint16 *coeff[2];
+	Sint32 new_sample;
+
+	/* Allocate the proper sized output buffer */
+	encoded_len = *audio_len;
+	encoded = *audio_buf;
+	freeable = *audio_buf;
+	*audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * 
+				MS_ADPCM_state.wSamplesPerBlock*
+				MS_ADPCM_state.wavefmt.channels*sizeof(Sint16);
+	*audio_buf = (Uint8 *)malloc(*audio_len);
+	if ( *audio_buf == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	decoded = *audio_buf;
+
+	/* Get ready... Go! */
+	stereo = (MS_ADPCM_state.wavefmt.channels == 2);
+	state[0] = &MS_ADPCM_state.state[0];
+	state[1] = &MS_ADPCM_state.state[stereo];
+	while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) {
+		/* Grab the initial information for this block */
+		state[0]->hPredictor = *encoded++;
+		if ( stereo ) {
+			state[1]->hPredictor = *encoded++;
+		}
+		state[0]->iDelta = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iDelta = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
+		coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
+
+		/* Store the two initial samples we start with */
+		decoded[0] = state[0]->iSamp2&0xFF;
+		decoded[1] = state[0]->iSamp2>>8;
+		decoded += 2;
+		if ( stereo ) {
+			decoded[0] = state[1]->iSamp2&0xFF;
+			decoded[1] = state[1]->iSamp2>>8;
+			decoded += 2;
+		}
+		decoded[0] = state[0]->iSamp1&0xFF;
+		decoded[1] = state[0]->iSamp1>>8;
+		decoded += 2;
+		if ( stereo ) {
+			decoded[0] = state[1]->iSamp1&0xFF;
+			decoded[1] = state[1]->iSamp1>>8;
+			decoded += 2;
+		}
+
+		/* Decode and store the other samples in this block */
+		samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)*
+					MS_ADPCM_state.wavefmt.channels;
+		while ( samplesleft > 0 ) {
+			nybble = (*encoded)>>4;
+			new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]);
+			decoded[0] = new_sample&0xFF;
+			new_sample >>= 8;
+			decoded[1] = new_sample&0xFF;
+			decoded += 2;
+
+			nybble = (*encoded)&0x0F;
+			new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]);
+			decoded[0] = new_sample&0xFF;
+			new_sample >>= 8;
+			decoded[1] = new_sample&0xFF;
+			decoded += 2;
+
+			++encoded;
+			samplesleft -= 2;
+		}
+		encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
+	}
+	free(freeable);
+	return(0);
+}
+
+struct IMA_ADPCM_decodestate {
+	Sint32 sample;
+	Sint8 index;
+};
+static struct IMA_ADPCM_decoder {
+	WaveFMT wavefmt;
+	Uint16 wSamplesPerBlock;
+	/* * * */
+	struct IMA_ADPCM_decodestate state[2];
+} IMA_ADPCM_state;
+
+static int InitIMA_ADPCM(WaveFMT *format)
+{
+	Uint8 *rogue_feel;
+	Uint16 extra_info;
+
+	/* Set the rogue pointer to the IMA_ADPCM specific data */
+	IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+	IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+	IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+	IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+	IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+	IMA_ADPCM_state.wavefmt.bitspersample =
+					 SDL_SwapLE16(format->bitspersample);
+	rogue_feel = (Uint8 *)format+sizeof(*format);
+	if ( sizeof(*format) == 16 ) {
+		extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	return(0);
+}
+
+static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble)
+{
+	const Sint32 max_audioval = ((1<<(16-1))-1);
+	const Sint32 min_audioval = -(1<<(16-1));
+	const int index_table[16] = {
+		-1, -1, -1, -1,
+		 2,  4,  6,  8,
+		-1, -1, -1, -1,
+		 2,  4,  6,  8
+	};
+	const Sint32 step_table[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
+	};
+	Sint32 delta, step;
+
+	/* Compute difference and new sample value */
+	step = step_table[state->index];
+	delta = step >> 3;
+	if ( nybble & 0x04 ) delta += step;
+	if ( nybble & 0x02 ) delta += (step >> 1);
+	if ( nybble & 0x01 ) delta += (step >> 2);
+	if ( nybble & 0x08 ) delta = -delta;
+	state->sample += delta;
+
+	/* Update index value */
+	state->index += index_table[nybble];
+	if ( state->index > 88 ) {
+		state->index = 88;
+	} else
+	if ( state->index < 0 ) {
+		state->index = 0;
+	}
+
+	/* Clamp output sample */
+	if ( state->sample > max_audioval ) {
+		state->sample = max_audioval;
+	} else
+	if ( state->sample < min_audioval ) {
+		state->sample = min_audioval;
+	}
+	return(state->sample);
+}
+
+/* Fill the decode buffer with a channel block of data (8 samples) */
+static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded,
+	int channel, int numchannels, struct IMA_ADPCM_decodestate *state)
+{
+	int i;
+	Sint8 nybble;
+	Sint32 new_sample;
+
+	decoded += (channel * 2);
+	for ( i=0; i<4; ++i ) {
+		nybble = (*encoded)&0x0F;
+		new_sample = IMA_ADPCM_nibble(state, nybble);
+		decoded[0] = new_sample&0xFF;
+		new_sample >>= 8;
+		decoded[1] = new_sample&0xFF;
+		decoded += 2 * numchannels;
+
+		nybble = (*encoded)>>4;
+		new_sample = IMA_ADPCM_nibble(state, nybble);
+		decoded[0] = new_sample&0xFF;
+		new_sample >>= 8;
+		decoded[1] = new_sample&0xFF;
+		decoded += 2 * numchannels;
+
+		++encoded;
+	}
+}
+
+static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
+{
+	struct IMA_ADPCM_decodestate *state;
+	Uint8 *freeable, *encoded, *decoded;
+	Sint32 encoded_len, samplesleft;
+	int c, channels;
+
+	/* Check to make sure we have enough variables in the state array */
+	channels = IMA_ADPCM_state.wavefmt.channels;
+	if ( channels > NELEMS(IMA_ADPCM_state.state) ) {
+		SDL_SetError("IMA ADPCM decoder can only handle %d channels",
+						NELEMS(IMA_ADPCM_state.state));
+		return(-1);
+	}
+	state = IMA_ADPCM_state.state;
+
+	/* Allocate the proper sized output buffer */
+	encoded_len = *audio_len;
+	encoded = *audio_buf;
+	freeable = *audio_buf;
+	*audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * 
+				IMA_ADPCM_state.wSamplesPerBlock*
+				IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16);
+	*audio_buf = (Uint8 *)malloc(*audio_len);
+	if ( *audio_buf == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	decoded = *audio_buf;
+
+	/* Get ready... Go! */
+	while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) {
+		/* Grab the initial information for this block */
+		for ( c=0; c<channels; ++c ) {
+			/* Fill the state information for this block */
+			state[c].sample = ((encoded[1]<<8)|encoded[0]);
+			encoded += 2;
+			if ( state[c].sample & 0x8000 ) {
+				state[c].sample -= 0x10000;
+			}
+			state[c].index = *encoded++;
+			/* Reserved byte in buffer header, should be 0 */
+			if ( *encoded++ != 0 ) {
+				/* Uh oh, corrupt data?  Buggy code? */;
+			}
+
+			/* Store the initial sample we start with */
+			decoded[0] = state[c].sample&0xFF;
+			decoded[1] = state[c].sample>>8;
+			decoded += 2;
+		}
+
+		/* Decode and store the other samples in this block */
+		samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels;
+		while ( samplesleft > 0 ) {
+			for ( c=0; c<channels; ++c ) {
+				Fill_IMA_ADPCM_block(decoded, encoded,
+						c, channels, &state[c]);
+				encoded += 4;
+				samplesleft -= 8;
+			}
+			decoded += (channels * 8 * 2);
+		}
+		encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
+	}
+	free(freeable);
+	return(0);
+}
+
+SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc,
+		SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
+{
+	int was_error;
+	Chunk chunk;
+	int lenread;
+	int MS_ADPCM_encoded, IMA_ADPCM_encoded;
+	int samplesize;
+
+	/* WAV magic header */
+	Uint32 RIFFchunk;
+	Uint32 wavelen;
+	Uint32 WAVEmagic;
+
+	/* FMT chunk */
+	WaveFMT *format = NULL;
+
+	/* Make sure we are passed a valid data source */
+	was_error = 0;
+	if ( src == NULL ) {
+		was_error = 1;
+		goto done;
+	}
+		
+	/* Check the magic header */
+	RIFFchunk	= SDL_ReadLE32(src);
+	wavelen		= SDL_ReadLE32(src);
+	WAVEmagic	= SDL_ReadLE32(src);
+	if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {
+		SDL_SetError("Unrecognized file type (not WAVE)");
+		was_error = 1;
+		goto done;
+	}
+
+	/* Read the audio data format chunk */
+	chunk.data = NULL;
+	do {
+		if ( chunk.data != NULL ) {
+			free(chunk.data);
+		}
+		lenread = ReadChunk(src, &chunk);
+		if ( lenread < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+	} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );
+
+	/* Decode the audio data format */
+	format = (WaveFMT *)chunk.data;
+	if ( chunk.magic != FMT ) {
+		SDL_SetError("Complex WAVE files not supported");
+		was_error = 1;
+		goto done;
+	}
+	MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
+	switch (SDL_SwapLE16(format->encoding)) {
+		case PCM_CODE:
+			/* We can understand this */
+			break;
+		case MS_ADPCM_CODE:
+			/* Try to understand this */
+			if ( InitMS_ADPCM(format) < 0 ) {
+				was_error = 1;
+				goto done;
+			}
+			MS_ADPCM_encoded = 1;
+			break;
+		case IMA_ADPCM_CODE:
+			/* Try to understand this */
+			if ( InitIMA_ADPCM(format) < 0 ) {
+				was_error = 1;
+				goto done;
+			}
+			IMA_ADPCM_encoded = 1;
+			break;
+		default:
+			SDL_SetError("Unknown WAVE data format: 0x%.4x",
+					SDL_SwapLE16(format->encoding));
+			was_error = 1;
+			goto done;
+	}
+	memset(spec, 0, (sizeof *spec));
+	spec->freq = SDL_SwapLE32(format->frequency);
+	switch (SDL_SwapLE16(format->bitspersample)) {
+		case 4:
+			if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) {
+				spec->format = AUDIO_S16;
+			} else {
+				was_error = 1;
+			}
+			break;
+		case 8:
+			spec->format = AUDIO_U8;
+			break;
+		case 16:
+			spec->format = AUDIO_S16;
+			break;
+		default:
+			was_error = 1;
+			break;
+	}
+	if ( was_error ) {
+		SDL_SetError("Unknown %d-bit PCM data format",
+			SDL_SwapLE16(format->bitspersample));
+		goto done;
+	}
+	spec->channels = (Uint8)SDL_SwapLE16(format->channels);
+	spec->samples = 4096;		/* Good default buffer size */
+
+	/* Read the audio data chunk */
+	*audio_buf = NULL;
+	do {
+		if ( *audio_buf != NULL ) {
+			free(*audio_buf);
+		}
+		lenread = ReadChunk(src, &chunk);
+		if ( lenread < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+		*audio_len = lenread;
+		*audio_buf = chunk.data;
+	} while ( chunk.magic != DATA );
+
+	if ( MS_ADPCM_encoded ) {
+		if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+	}
+	if ( IMA_ADPCM_encoded ) {
+		if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+	}
+
+	/* Don't return a buffer that isn't a multiple of samplesize */
+	samplesize = ((spec->format & 0xFF)/8)*spec->channels;
+	*audio_len &= ~(samplesize-1);
+
+done:
+	if ( format != NULL ) {
+		free(format);
+	}
+	if ( freesrc && src ) {
+		SDL_RWclose(src);
+	}
+	if ( was_error ) {
+		spec = NULL;
+	}
+	return(spec);
+}
+
+/* Since the WAV memory is allocated in the shared library, it must also
+   be freed here.  (Necessary under Win32, VC++)
+ */
+void SDL_FreeWAV(Uint8 *audio_buf)
+{
+	if ( audio_buf != NULL ) {
+		free(audio_buf);
+	}
+}
+
+static int ReadChunk(SDL_RWops *src, Chunk *chunk)
+{
+	chunk->magic	= SDL_ReadLE32(src);
+	chunk->length	= SDL_ReadLE32(src);
+	chunk->data = (Uint8 *)malloc(chunk->length);
+	if ( chunk->data == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
+		SDL_Error(SDL_EFREAD);
+		free(chunk->data);
+		return(-1);
+	}
+	return(chunk->length);
+}
+
+#endif /* ENABLE_FILE */