cleaned up code structure a bit

1 files changed, 0 insertions, 441 deletions

diff --git a/src/engine/client/snd.c b/src/engine/client/snd.c
deleted file mode 100644
index aa155c0c..00000000
--- a/src/engine/client/snd.c
+++ /dev/null
@@ -1,441 +0,0 @@
-/* copyright (c) 2007 magnus auvinen, see licence.txt for more info */
-#include <engine/system.h>
-#include <engine/interface.h>
-#include <engine/config.h>
-
-#include <engine/external/portaudio/portaudio.h>
-#include <engine/external/wavpack/wavpack.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-enum
-{
-	NUM_SAMPLES = 512,
-	NUM_VOICES = 64,
-	NUM_CHANNELS = 16,
-	
-	MAX_FRAMES = 1024
-};
-
-typedef struct
-{
-	short *data;
-	int num_frames;
-	int rate;
-	int channels;
-	int loop_start;
-	int loop_end;
-} SAMPLE;
-
-typedef struct
-{
-	int vol;
-	int pan;
-} CHANNEL;
-
-typedef struct VOICE_t
-{
-	SAMPLE *snd;
-	CHANNEL *channel;
-	int tick;
-	int vol; /* 0 - 255 */
-	int flags;
-	int x, y;
-} VOICE;
-
-static SAMPLE samples[NUM_SAMPLES] = { {0} };
-static VOICE voices[NUM_VOICES] = { {0} };
-static CHANNEL channels[NUM_CHANNELS] = { {255, 0} };
-
-static LOCK sound_lock = 0;
-static int sound_enabled = 0;
-
-static int center_x = 0;
-static int center_y = 0;
-
-static int mixing_rate = 48000;
-
-void snd_set_channel(int cid, float vol, float pan)
-{
-	channels[cid].vol = (int)(vol*255.0f);
-	channels[cid].pan = (int)(pan*255.0f); /* TODO: this is only on and off right now */
-}
-
-static int play(int cid, int sid, int flags, float x, float y)
-{
-	int vid = -1;
-	int i;
-	
-	lock_wait(sound_lock);
-	
-	/* search for voice */
-	/* TODO: fix this linear search */
-	for(i = 0; i < NUM_VOICES; i++)
-	{
-		if(!voices[i].snd)
-		{
-			vid = i;
-			break;
-		}
-	}
-	
-	/* voice found, use it */
-	if(vid != -1)
-	{
-		voices[vid].snd = &samples[sid];
-		voices[vid].channel = &channels[cid];
-		voices[vid].tick = 0;
-		voices[vid].vol = 255;
-		voices[vid].flags = flags;
-		voices[vid].x = (int)x;
-		voices[vid].y = (int)y;
-	}
-	
-	lock_release(sound_lock);
-	return vid;
-}
-
-int snd_play_at(int cid, int sid, int flags, float x, float y)
-{
-	return play(cid, sid, flags|SNDFLAG_POS, x, y);
-}
-
-int snd_play(int cid, int sid, int flags)
-{
-	return play(cid, sid, flags, 0, 0);
-}
-
-void snd_stop(int vid)
-{
-	/* TODO: a nice fade out */
-	lock_wait(sound_lock);
-	voices[vid].snd = 0;
-	lock_release(sound_lock);
-}
-
-/* TODO: there should be a faster way todo this */
-static short int2short(int i)
-{
-	if(i > 0x7fff)
-		return 0x7fff;
-	else if(i < -0x7fff)
-		return -0x7fff;
-	return i;
-}
-
-static int iabs(int i)
-{
-	if(i<0)
-		return -i;
-	return i;
-}
-
-static void mix(short *final_out, unsigned frames)
-{
-	int mix_buffer[MAX_FRAMES*2] = {0};
-	int i, s;
-
-	/* aquire lock while we are mixing */
-	lock_wait(sound_lock);
-	
-	for(i = 0; i < NUM_VOICES; i++)
-	{
-		if(voices[i].snd)
-		{
-			/* mix voice */
-			VOICE *v = &voices[i];
-			int *out = mix_buffer;
-
-			int step = v->snd->channels; /* setup input sources */
-			short *in_l = &v->snd->data[v->tick*step];
-			short *in_r = &v->snd->data[v->tick*step+1];
-			
-			int end = v->snd->num_frames-v->tick;
-
-			int rvol = v->channel->vol;
-			int lvol = v->channel->vol;
-
-			/* make sure that we don't go outside the sound data */
-			if(frames < end)
-				end = frames;
-			
-			/* check if we have a mono sound */
-			if(v->snd->channels == 1)
-				in_r = in_l;
-
-			/* volume calculation */
-			if(v->flags&SNDFLAG_POS && v->channel->pan)
-			{
-				/* TODO: we should respect the channel panning value */
-				const int range = 1500; /* magic value, remove */
-				int dx = v->x - center_x;
-				int dy = v->y - center_y;
-				int dist = sqrt(dx*dx+dy*dy); /* double here. nasty */
-				int p = iabs(dx);
-				if(dist < range)
-				{
-					/* panning */
-					if(dx > 0)
-						lvol = ((range-p)*lvol)/range;
-					else
-						rvol = ((range-p)*rvol)/range;
-					
-					/* falloff */
-					lvol = (lvol*(range-dist))/range;
-					rvol = (rvol*(range-dist))/range;
-				}
-				else
-				{
-					lvol = 0;
-					rvol = 0;
-				}
-			}
-
-			/* process all frames */
-			for(s = 0; s < end; s++)
-			{
-				*out++ += (*in_l)*lvol;
-				*out++ += (*in_r)*rvol;
-				in_l += step;
-				in_r += step;
-				v->tick++;
-			}
-			
-			/* free voice if not used any more */
-			if(v->tick == v->snd->num_frames)
-				v->snd = 0;
-			
-		}
-	}
-
-	/* release the lock */
-	lock_release(sound_lock);
-
-	{
-		int master_vol = config.snd_volume;
-		
-		/* clamp accumulated values */
-		/* TODO: this seams slow */
-		for(i = 0; i < frames; i++)
-		{
-			int j = i<<1;
-			int vl = ((mix_buffer[j]*master_vol)/101)>>8;
-			int vr = ((mix_buffer[j+1]*master_vol)/101)>>8;
-
-			final_out[j] = int2short(vl);
-			final_out[j+1] = int2short(vr);
-		}
-	}
-}
-
-static int pacallback(const void *in, void *out, unsigned long frames, const PaStreamCallbackTimeInfo* time, PaStreamCallbackFlags status, void *user)
-{
-	mix(out, frames);
-	return 0;
-}
-
-static PaStream *stream;
-
-int snd_init()
-{
-	PaStreamParameters params;
-	PaError err = Pa_Initialize();
-	
-	sound_lock = lock_create();
-	
-	if(!config.snd_enable)
-		return 0;
-	
-	mixing_rate = config.snd_rate;
-
-	params.device = Pa_GetDefaultOutputDevice();
-	if(params.device < 0)
-		return 1;
-	params.channelCount = 2;
-	params.sampleFormat = paInt16;
-	params.suggestedLatency = Pa_GetDeviceInfo(params.device)->defaultLowOutputLatency;
-	params.hostApiSpecificStreamInfo = 0x0;
-
-	err = Pa_OpenStream(
-			&stream,        /* passes back stream pointer */
-			0,              /* no input channels */
-			&params,                /* pointer to parameters */
-			mixing_rate,          /* sample rate */
-			128,            /* frames per buffer */
-			paClipOff,              /* no clamping */
-			pacallback,             /* specify our custom callback */
-			0x0); /* pass our data through to callback */
-	err = Pa_StartStream(stream);
-
-	sound_enabled = 1;
-	return 0;
-}
-
-int snd_shutdown()
-{
-	Pa_StopStream(stream);
-	Pa_Terminate();
-
-	lock_destroy(sound_lock);
-
-	return 0;
-}
-
-int snd_alloc_id()
-{
-	/* TODO: linear search, get rid of it */
-	unsigned sid;
-	for(sid = 0; sid < NUM_SAMPLES; sid++)
-	{
-		if(samples[sid].data == 0x0)
-			return sid;
-	}
-
-	return -1;
-}
-
-static void rate_convert(int sid)
-{
-	SAMPLE *snd = &samples[sid];
-	int num_frames = 0;
-	short *new_data = 0;
-	int i;
-	
-	/* make sure that we need to convert this sound */
-	if(!snd->data || snd->rate == mixing_rate)
-		return;
-
-	/* allocate new data */
-	num_frames = (int)((snd->num_frames/(float)snd->rate)*mixing_rate);
-	new_data = mem_alloc(num_frames*snd->channels*sizeof(short), 1);
-	
-	for(i = 0; i < num_frames; i++)
-	{
-		/* resample TODO: this should be done better, like linear atleast */
-		float a = i/(float)num_frames;
-		int f = (int)(a*snd->num_frames);
-		if(f >= snd->num_frames)
-			f = snd->num_frames-1;
-		
-		/* set new data */
-		if(snd->channels == 1)
-			new_data[i] = snd->data[f];
-		else if(snd->channels == 2)
-		{
-			new_data[i*2] = snd->data[f*2];
-			new_data[i*2+1] = snd->data[f*2+1];
-		}
-	}
-	
-	/* free old data and apply new */
-	mem_free(snd->data);
-	snd->data = new_data;
-	snd->num_frames = num_frames;
-}
-
-
-static FILE *file = NULL;
-
-static int read_data(void *buffer, int size)
-{
-	return fread(buffer, 1, size, file);	
-}
-
-int snd_load_wv(const char *filename)
-{
-	SAMPLE *snd;
-	int sid = -1;
-	char error[100];
-	WavpackContext *context;
-	
-	/* don't waste memory on sound when we are stress testing */
-	if(config.dbg_stress)
-		return -1;
-		
-	/* no need to load sound when we are running with no sound */
-	if(!sound_enabled)
-		return 1;
-
-	file = fopen(filename, "rb"); /* TODO: use system.h stuff for this */
-	if(!file)
-	{
-		dbg_msg("sound/wv", "failed to open %s", filename);
-		return -1;
-	}
-
-	sid = snd_alloc_id();
-	if(sid < 0)
-		return -1;
-	snd = &samples[sid];
-
-	context = WavpackOpenFileInput(read_data, error);
-	if (context)
-	{
-		int samples = WavpackGetNumSamples(context);
-		int bitspersample = WavpackGetBitsPerSample(context);
-		unsigned int samplerate = WavpackGetSampleRate(context);
-		int channels = WavpackGetNumChannels(context);
-		int *data;
-		int *src;
-		short *dst;
-		int i;
-
-		snd->channels = channels;
-		snd->rate = samplerate;
-
-		if(snd->channels > 2)
-		{
-			dbg_msg("sound/wv", "file is not mono or stereo. filename='%s'", filename);
-			return -1;
-		}
-
-		/*
-		if(snd->rate != 44100)
-		{
-			dbg_msg("sound/wv", "file is %d Hz, not 44100 Hz. filename='%s'", snd->rate, filename);
-			return -1;
-		}*/
-		
-		if(bitspersample != 16)
-		{
-			dbg_msg("sound/wv", "bps is %d, not 16, filname='%s'", bitspersample, filename);
-			return -1;
-		}
-
-		data = (int *)mem_alloc(4*samples*channels, 1);
-		WavpackUnpackSamples(context, data, samples); /* TODO: check return value */
-		src = data;
-		
-		snd->data = (short *)mem_alloc(2*samples*channels, 1);
-		dst = snd->data;
-
-		for (i = 0; i < samples*channels; i++)
-			*dst++ = (short)*src++;
-
-		mem_free(data);
-
-		snd->num_frames = samples;
-		snd->loop_start = -1;
-		snd->loop_end = -1;
-	}
-	else
-	{
-		dbg_msg("sound/wv", "failed to open %s: %s", filename, error);
-	}
-
-	fclose(file);
-	file = NULL;
-
-	if(config.debug)
-		dbg_msg("sound/wv", "loaded %s", filename);
-
-	rate_convert(sid);
-	return sid;
-}
-
-void snd_set_listener_pos(float x, float y)
-{
-	center_x = (int)x;
-	center_y = (int)y;
-}