/*

    SDL - Simple DirectMedia Layer

    Copyright (C) 1997-2004 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@libsdl.org

*/

#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 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);

	if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */

		WAVEmagic = wavelen;

		wavelen   = RIFFchunk;

		RIFFchunk = RIFF;

	} else {

		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 */