/*
* Copyright(c) 1997-2001 Id Software, Inc.
* Copyright(c) 2002 The Quakeforge Project.
* Copyright(c) 2006 Quetoo.
*
* 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., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
*/

/*
* sound cacheing
*/

#include "client.h"

int cache_full_cycle;

byte *S_Alloc(int size);

/*
ResampleSfx
*/
void ResampleSfx(sfx_t *sfx, int inrate, int inwidth, byte *data){
	int outcount;
	int srcsample;
	float stepscale;
	int i;
	int sample, samplefrac, fracstep;
	sfxcache_t *sc;
	
	sc = sfx->cache;
	if(!sc)
		return;
		
	stepscale = (float)inrate / dma.rate;  // this is usually 0.5, 1, or 2
	
	outcount = sc->length / stepscale;
	sc->length = outcount;
	if(sc->loopstart != -1)
		sc->loopstart = sc->loopstart / stepscale;
		
	sc->rate = dma.rate;
	if(s_loadas8bit->value)
		sc->width = 1;
	else
		sc->width = inwidth;
	sc->stereo = 0;
	
	// resample / decimate to the current source rate
	
	if(stepscale == 1 && inwidth == 1 && sc->width == 1){
		// fast special case
		for(i = 0; i < outcount; i++)
			((signed char *)sc->data)[i]
			= (int)((unsigned char)(data[i]) - 128);
	} else {
		// general case
		samplefrac = 0;
		fracstep = stepscale * 256;
		for(i = 0; i < outcount; i++){
			srcsample = samplefrac >> 8;
			samplefrac += fracstep;
			if(inwidth == 2)
				sample = LittleShort(((short *)data)[srcsample]);
			else
				sample =(int)((unsigned char)(data[srcsample]) - 128) << 8;
			if(sc->width == 2)
				((short *)sc->data)[i] = sample;
			else
				((signed char *)sc->data)[i] = sample >> 8;
		}
	}
}


/*
S_LoadSound
*/
sfxcache_t *S_LoadSound(sfx_t *s){
	char namebuffer[MAX_QPATH];
	byte *data;
	wavinfo_t info;
	int len;
	float stepscale;
	sfxcache_t *sc;
	int size;
	char *name;
	
	if(s->name[0] == '*')
		return NULL;
		
	// see if still in memory
	sc = s->cache;
	if(sc)
		return sc;
		
	// load it in
	if(s->truename)
		name = s->truename;
	else
		name = s->name;
		
	if(name[0] == '#')
		strcpy(namebuffer, &name[1]);
	else
		Com_sprintf(namebuffer, sizeof(namebuffer), "sound/%s", name);
	
	size = FS_LoadFile(namebuffer, (void **)(char *)&data);
	if(!data){
		Com_DPrintf("Couldn't load %s\n", namebuffer);
		return NULL;
	}
	
	info = GetWavinfo(s->name, data, size);
	if(info.channels != 1){
		Com_Printf("%s is a stereo sample\n", s->name);
		FS_FreeFile(data);
		return NULL;
	}
	
	stepscale =(float)info.rate / dma.rate;
	len = info.samples / stepscale;
	
	len = len * info.width * info.channels;
	
	sc = s->cache = Z_Malloc(len + sizeof(sfxcache_t));
	if(!sc){
		FS_FreeFile(data);
		return NULL;
	}
	
	sc->length = info.samples;
	sc->loopstart = info.loopstart;
	sc->rate = info.rate;
	sc->width = info.width;
	sc->stereo = info.channels;
	
	ResampleSfx(s, sc->rate, sc->width, data + info.dataofs);
	
	FS_FreeFile(data);
	
	return sc;
}

/*
WAV loading
*/

byte *data_p;
byte *iff_end;
byte *last_chunk;
byte *iff_data;
int iff_chunk_len;


short GetLittleShort(void){
	short val = 0;
	val = *data_p;
	val = val +(*(data_p + 1) << 8);
	data_p += 2;
	return val;
}

int GetLittleLong(void){
	int val = 0;
	val = *data_p;
	val = val +(*(data_p + 1) << 8);
	val = val +(*(data_p + 2) << 16);
	val = val +(*(data_p + 3) << 24);
	data_p += 4;
	return val;
}

void FindNextChunk(char *name){
	while(1){
		data_p = last_chunk;
		
		if(data_p >= iff_end){  // didn't find the chunk
			data_p = NULL;
			return;
		}
		
		data_p += 4;
		iff_chunk_len = GetLittleLong();
		if(iff_chunk_len < 0){
			data_p = NULL;
			return;
		}
		
		data_p -= 8;
		last_chunk = data_p + 8 +((iff_chunk_len + 1) & ~1);
		if(!strncmp((char *)data_p, name, 4))
			return;
	}
}

void FindChunk(char *name){
	last_chunk = iff_data;
	FindNextChunk(name);
}


void DumpChunks(void){
	char str[5];
	
	str[4] = 0;
	data_p = iff_data;
	do {
		memcpy(str, data_p, 4);
		data_p += 4;
		iff_chunk_len = GetLittleLong();
		Com_Printf("0x%p : %s(%d)\n", data_p - 4, str, iff_chunk_len);
		data_p +=(iff_chunk_len + 1) & ~1;
	} while(data_p < iff_end);
}

/*
GetWavinfo
*/
wavinfo_t GetWavinfo(char *name, byte *wav, int wavlength){
	wavinfo_t info;
	int i;
	int format;
	int samples;
	
	memset(&info, 0, sizeof(info));
	
	if(!wav)
		return info;
		
	iff_data = wav;
	iff_end = wav + wavlength;
	
	// find "RIFF" chunk
	FindChunk("RIFF");
	if(!(data_p && !strncmp((char *)data_p + 8, "WAVE", 4))){
		Com_Printf("Missing RIFF/WAVE chunks\n");
		return info;
	}
	
	// get "fmt " chunk
	iff_data = data_p + 12;
	
	FindChunk("fmt ");
	if(!data_p){
		Com_Printf("Missing fmt chunk\n");
		return info;
	}
	data_p += 8;
	format = GetLittleShort();
	if(format != 1){
		Com_Printf("Microsoft PCM format only\n");
		return info;
	}
	
	info.channels = GetLittleShort();
	info.rate = GetLittleLong();
	data_p += 4 + 2;
	info.width = GetLittleShort() / 8;
	
	// get cue chunk
	FindChunk("cue ");
	if(data_p){
		data_p += 32;
		info.loopstart = GetLittleLong();
		
		// if the next chunk is a LIST chunk, look for a cue length marker
		FindNextChunk("LIST");
		if(data_p){
			if(!strncmp((char *)data_p + 28, "mark", 4)){  // this is not a proper parse, but it works with cooledit...
				data_p += 24;
				i = GetLittleLong();  // samples in loop
				info.samples = info.loopstart + i;
			}
		}
	} else
		info.loopstart = -1;
		
	// find data chunk
	FindChunk("data");
	if(!data_p){
		Com_Printf("Missing data chunk\n");
		return info;
	}
	
	data_p += 4;
	samples = GetLittleLong() / info.width;
	
	if(info.samples){
		if(samples < info.samples)
			Com_Error(ERR_DROP, "Sound %s has a bad loop length", name);
	} else
		info.samples = samples;
		
	info.dataofs = data_p - wav;
	
	return info;
}