path: root/src/audio/SDL_wave.c

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2006 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 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
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Sam Lantinga
    slouken@libsdl.org
*/

#ifndef DISABLE_FILE

/* Microsoft WAVE file loading routines */

#include "SDL_audio.h"
#include "SDL_wave.h"


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 *)SDL_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;
	}
	SDL_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 > SDL_arraysize(IMA_ADPCM_state.state) ) {
		SDL_SetError("IMA ADPCM decoder can only handle %d channels",
					SDL_arraysize(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 *)SDL_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;
	}
	SDL_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 = 0;
	Uint32 WAVEmagic;
	Uint32 headerDiff = 0;

	/* 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;
	}
	headerDiff += sizeof(Uint32); // for WAVE

	/* Read the audio data format chunk */
	chunk.data = NULL;
	do {
		if ( chunk.data != NULL ) {
			SDL_free(chunk.data);
		}
		lenread = ReadChunk(src, &chunk);
		if ( lenread < 0 ) {
			was_error = 1;
			goto done;
		}
		// 2 Uint32's for chunk header+len, plus the lenread
		headerDiff += lenread + 2 * sizeof(Uint32);
	} 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;
	}
	SDL_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 ) {
			SDL_free(*audio_buf);
		}
		lenread = ReadChunk(src, &chunk);
		if ( lenread < 0 ) {
			was_error = 1;
			goto done;
		}
		*audio_len = lenread;
		*audio_buf = chunk.data;
		if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32);
	} while ( chunk.magic != DATA );
	headerDiff += 2 * sizeof(Uint32); // for the data chunk and len

	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 ) {
		SDL_free(format);
	}
	if ( freesrc && src ) {
		SDL_RWclose(src);
	}
	else {
		// seek to the end of the file (given by the RIFF chunk)
		SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
	}
	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 ) {
		SDL_free(audio_buf);
	}
}

static int ReadChunk(SDL_RWops *src, Chunk *chunk)
{
	chunk->magic	= SDL_ReadLE32(src);
	chunk->length	= SDL_ReadLE32(src);
	chunk->data = (Uint8 *)SDL_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);
		SDL_free(chunk->data);
		return(-1);
	}
	return(chunk->length);
}

#endif /* ENABLE_FILE */