new mixer. sample rate conversion

This commit is contained in:
Magnus Auvinen 2007-10-08 20:34:11 +00:00
parent 51f25fdfeb
commit 4bd0274a2e
4 changed files with 252 additions and 441 deletions

View file

@ -10,60 +10,110 @@
enum
{
NUM_SOUNDS = 512,
NUM_SAMPLES = 512,
NUM_VOICES = 64,
NUM_CHANNELS = 4,
NUM_CHANNELS = 16,
MAX_FRAMES = 1024
};
static LOCK sound_lock = 0;
static struct sound
typedef struct
{
short *data;
int num_samples;
int num_frames;
int rate;
int channels;
int loop_start;
int loop_end;
} sounds[NUM_SOUNDS] = { {0x0, 0, 0, 0, -1, -1} };
} SAMPLE;
static struct voice
typedef struct
{
volatile struct sound *sound;
int vol;
int pan;
} CHANNEL;
typedef struct VOICE_t
{
SAMPLE *snd;
CHANNEL *channel;
int tick;
int stop;
int loop;
float vol;
float pan;
float x;
float y;
volatile struct voice *prev;
volatile struct voice *next;
} voices[NUM_VOICES] = { {0x0, 0, -1, -1, 1.0f, 0.0f, 0.0f, 0.0f, 0x0, 0x0} };
#define CHANNEL_POSITION_VOLUME 1
#define CHANNEL_POSITION_PAN 2
static struct channel
{
volatile struct voice *first_voice;
float vol;
float pan;
int vol; /* 0 - 255 */
int flags;
} channels[NUM_CHANNELS] = { {0x0, 1.0f, 0.0f, 0} };
int x, y;
} VOICE;
static float center_x = 0.0f;
static float center_y = 0.0f;
static float master_vol = 1.0f;
static float master_pan = 0.0f;
static float pan_deadzone = 256.0f;
static float pan_falloff = 1.0f;
static float volume_deadzone = 256.0f;
static float volume_falloff = 1.0f;
static SAMPLE samples[NUM_SAMPLES] = { {0} };
static VOICE voices[NUM_VOICES] = { {0} };
static CHANNEL channels[NUM_CHANNELS] = { {255, 0} };
static inline short int2short(int i)
static LOCK sound_lock = 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);
}
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);
}
/* there should be a faster way todo this */
static short int2short(int i)
{
if(i > 0x7fff)
return 0x7fff;
@ -72,194 +122,98 @@ static inline short int2short(int i)
return i;
}
static inline float sgn(float f)
static int iabs(int i)
{
if(f < 0.0f)
return -1.0f;
return 1.0f;
if(i<0)
return -i;
return i;
}
static void reset_voice(struct voice *v)
static void mix(short *final_out, unsigned frames)
{
v->sound = 0x0;
v->tick = 0;
v->stop = -1;
v->loop = -1;
v->vol = 1.0f;
v->pan = 0.0f;
v->x = 0.0f;
v->y = 0.0f;
v->next = 0x0;
v->prev = 0x0;
}
int mix_buffer[MAX_FRAMES*2] = {0};
int i, s;
static inline void fill_mono(int *out, unsigned frames, struct voice *v, float fvol, float fpan)
{
int ivol = (int) (31.0f * fvol);
int ipan = (int) (31.0f * ipan);
unsigned i;
for(i = 0; i < frames; i++)
/* aquire lock while we are mixing */
lock_wait(sound_lock);
for(i = 0; i < NUM_VOICES; i++)
{
unsigned j = i<<1;
int val = v->sound->data[v->tick] * ivol;
out[j] += val;
out[j+1] += val;
v->tick++;
}
}
static inline void fill_stereo(int *out, unsigned frames, struct voice *v, float fvol, float fpan)
{
int ivol = (int) (31.0f * fvol);
int ipan = (int) (31.0f * ipan);
unsigned i;
for(i = 0; i < frames; i++)
{
unsigned j = i<<1;
out[j] += v->sound->data[v->tick] * ivol;
out[j+1] += v->sound->data[v->tick+1] * ivol;
v->tick += 2;
}
}
static void mix(short *out, unsigned frames)
{
static int main_buffer[MAX_FRAMES*2];
unsigned locked = 0;
unsigned i;
unsigned cid;
dbg_assert(frames <= MAX_FRAMES, "too many frames to fill");
for(i = 0; i < frames; i++)
{
unsigned j = i<<1;
main_buffer[j] = 0;
main_buffer[j+1] = 0;
}
for(cid = 0; cid < NUM_CHANNELS; cid++)
{
struct channel *c = &channels[cid];
struct voice *v = (struct voice*)c->first_voice;
while(v)
if(voices[i].snd)
{
unsigned filled = 0;
unsigned step = 1;
/* mix voice */
VOICE *v = &voices[i];
int *out = mix_buffer;
while(v && v->sound && filled < frames)
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)
{
/* calculate maximum frames to fill */
unsigned frames_left = (v->sound->num_samples - v->tick) >> (v->sound->channels-1);
unsigned long to_fill = frames>frames_left?frames_left:frames;
float vol = 1.0f;
float pan = 0.0f;
/* clamp to_fill if voice should stop */
if(v->stop >= 0)
to_fill = (unsigned)v->stop>frames_left?frames:v->stop;
/* clamp to_fill if we are about to loop */
if(v->loop >= 0 && v->sound->loop_start >= 0)
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)
{
unsigned tmp = v->sound->loop_end - v->tick;
to_fill = tmp>to_fill?to_fill:tmp;
}
/* calculate voice volume and delta */
if(c->flags & CHANNEL_POSITION_VOLUME)
{
float dx = v->x - center_x;
float dy = v->y - center_y;
float dist = dx*dx + dy*dy;
if(dist < volume_deadzone*volume_deadzone)
vol = master_vol * c->vol;
/* panning */
if(dx > 0)
lvol = ((range-p)*lvol)/range;
else
vol = master_vol * c->vol / ((dist - volume_deadzone*volume_deadzone)*volume_falloff); /*TODO: use some fast 1/x^2 */
rvol = ((range-p)*rvol)/range;
/* falloff */
lvol = (lvol*(range-dist))/range;
rvol = (rvol*(range-dist))/range;
}
else
{
vol = master_vol * c->vol * v->vol;
}
/* calculate voice pan and delta */
if(c->flags & CHANNEL_POSITION_PAN)
{
float dx = v->x - center_x;
if(fabs(dx) < pan_deadzone)
pan = master_pan + c->pan;
else
pan = master_pan + c->pan + sgn(dx)*(fabs(dx) - pan_deadzone)/pan_falloff;
}
else
{
pan = master_pan + c->pan + v->pan;
}
/* fill the main buffer */
if(v->sound->channels == 1)
fill_mono(&main_buffer[filled], to_fill, v, vol, pan);
else
fill_stereo(&main_buffer[filled], to_fill, v, vol, pan);
/* reset tick of we hit loop point */
if(v->loop >= 0 &&
v->sound->loop_start >= 0 &&
v->tick >= v->sound->loop_end)
v->tick = v->sound->loop_start;
/* stop sample if nessecary */
if(v->stop >= 0)
v->stop -= to_fill;
if(v->tick >= v->sound->num_samples || v->stop == 0)
{
struct voice *vn = (struct voice *)v->next;
if(!locked)
{
lock_wait(sound_lock);
locked = 1;
}
if(v->next)
v->next->prev = v->prev;
if(v->prev)
v->prev->next = v->next;
else
channels[cid].first_voice = v->next;
dbg_msg("snd", "sound stopped");
reset_voice(v);
step = 0;
v = vn;
}
filled += to_fill;
}
if(step)
v = (struct voice*)v->next;
/* 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;
}
}
if(locked)
lock_release(sound_lock);
/* release the lock */
lock_release(sound_lock);
/* clamp accumulated values */
for(i = 0; i < frames; i++)
{
int j = i<<1;
int vl = main_buffer[j];
int vr = main_buffer[j+1];
int vl = mix_buffer[j]>>8;
int vr = mix_buffer[j+1]>>8;
out[j] = int2short(vl>>5);
out[j+1] = int2short(vr>>5);
}
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)
@ -274,6 +228,8 @@ int snd_init()
{
PaStreamParameters params;
PaError err = Pa_Initialize();
mixing_rate = config.snd_rate;
sound_lock = lock_create();
@ -290,7 +246,7 @@ int snd_init()
&stream, /* passes back stream pointer */
0, /* no input channels */
&params, /* pointer to parameters */
44100, /* sample rate */
mixing_rate, /* sample rate */
128, /* frames per buffer */
paClipOff, /* no clamping */
pacallback, /* specify our custom callback */
@ -310,26 +266,59 @@ int snd_shutdown()
return 0;
}
void snd_set_center(int x, int y)
{
center_x = x;
center_y = y;
}
int snd_alloc_id()
{
/* TODO: linear search, get rid of it */
unsigned sid;
for(sid = 0; sid < NUM_SOUNDS; sid++)
for(sid = 0; sid < NUM_SAMPLES; sid++)
{
if(sounds[sid].data == 0x0)
{
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)
@ -339,7 +328,7 @@ static int read_data(void *buffer, int size)
int snd_load_wv(const char *filename)
{
struct sound *snd;
SAMPLE *snd;
int sid = -1;
char error[100];
WavpackContext *context;
@ -347,7 +336,7 @@ int snd_load_wv(const char *filename)
sid = snd_alloc_id();
if(sid < 0)
return -1;
snd = &sounds[sid];
snd = &samples[sid];
file = fopen(filename, "rb"); /* TODO: use system.h stuff for this */
@ -396,7 +385,7 @@ int snd_load_wv(const char *filename)
mem_free(data);
snd->num_samples = samples;
snd->num_frames = samples;
snd->loop_start = -1;
snd->loop_end = -1;
}
@ -411,218 +400,17 @@ int snd_load_wv(const char *filename)
if(config.debug)
dbg_msg("sound/wv", "loaded %s", filename);
rate_convert(sid);
return sid;
}
#if 0
int snd_load_wav(const char *filename)
void snd_set_master_volume(float vol)
{
/* open file for reading */
IOHANDLE file;
struct sound *snd;
int sid = -1;
int state = 0;
file = io_open(filename, IOFLAG_READ);
if(!file)
{
dbg_msg("sound/wav", "failed to open file. filename='%s'", filename);
return -1;
}
sid = snd_alloc_id();
if(sid < 0)
return -1;
snd = &sounds[sid];
while(1)
{
/* read chunk header */
unsigned char head[8];
int chunk_size;
if(io_read(file, head, sizeof(head)) != 8)
{
break;
}
chunk_size = head[4] | (head[5]<<8) | (head[6]<<16) | (head[7]<<24);
head[4] = 0;
if(state == 0)
{
unsigned char type[4];
/* read the riff and wave headers */
if(head[0] != 'R' || head[1] != 'I' || head[2] != 'F' || head[3] != 'F')
{
dbg_msg("sound/wav", "not a RIFF file. filename='%s'", filename);
return -1;
}
io_read(file, type, 4);
if(type[0] != 'W' || type[1] != 'A' || type[2] != 'V' || type[3] != 'E')
{
dbg_msg("sound/wav", "RIFF file is not a WAVE. filename='%s'", filename);
return -1;
}
state++;
}
else if(state == 1)
{
/* read the format chunk */
if(head[0] == 'f' && head[1] == 'm' && head[2] == 't' && head[3] == ' ')
{
unsigned char fmt[16];
if(io_read(file, fmt, sizeof(fmt)) != sizeof(fmt))
{
dbg_msg("sound/wav", "failed to read format. filename='%s'", filename);
return -1;
}
/* decode format */
int compression_code = fmt[0] | (fmt[1]<<8);
snd->channels = fmt[2] | (fmt[3]<<8);
snd->rate = fmt[4] | (fmt[5]<<8) | (fmt[6]<<16) | (fmt[7]<<24);
if(compression_code != 1)
{
dbg_msg("sound/wav", "file is not uncompressed. filename='%s'", filename);
return -1;
}
if(snd->channels > 2)
{
dbg_msg("sound/wav", "file is not mono or stereo. filename='%s'", filename);
return -1;
}
if(snd->rate != 44100)
{
dbg_msg("sound/wav", "file is %d Hz, not 44100 Hz. filename='%s'", snd->rate, filename);
return -1;
}
int bps = fmt[14] | (fmt[15]<<8);
if(bps != 16)
{
dbg_msg("sound/wav", "bps is %d, not 16, filname='%s'", bps, filename);
return -1;
}
/* next state */
state++;
}
else
io_skip(file, chunk_size);
}
else if(state == 2)
{
/* read the data */
if(head[0] == 'd' && head[1] == 'a' && head[2] == 't' && head[3] == 'a')
{
snd->data = (short*)mem_alloc(chunk_size, 1);
io_read(file, snd->data, chunk_size);
snd->num_samples = chunk_size/(2);
#if defined(CONF_ARCH_ENDIAN_BIG)
swap_endian(snd->data, sizeof(short), snd->num_samples);
#endif
snd->loop_start = -1;
snd->loop_end = -1;
state++;
}
else
io_skip(file, chunk_size);
}
else if(state == 3)
{
if(head[0] == 's' && head[1] == 'm' && head[2] == 'p' && head[3] == 'l')
{
unsigned char smpl[36];
unsigned char loop[24];
if(config.debug)
dbg_msg("sound/wav", "got sustain");
io_read(file, smpl, sizeof(smpl));
unsigned num_loops = (smpl[28] | (smpl[29]<<8) | (smpl[30]<<16) | (smpl[31]<<24));
unsigned skip = (smpl[32] | (smpl[33]<<8) | (smpl[34]<<16) | (smpl[35]<<24));
if(num_loops > 0)
{
io_read(file, loop, sizeof(loop));
unsigned start = (loop[8] | (loop[9]<<8) | (loop[10]<<16) | (loop[11]<<24));
unsigned end = (loop[12] | (loop[13]<<8) | (loop[14]<<16) | (loop[15]<<24));
snd->loop_start = start * snd->channels;
snd->loop_end = end * snd->channels;
}
if(num_loops > 1)
io_skip(file, (num_loops-1) * sizeof(loop));
io_skip(file, skip);
state++;
}
else
io_skip(file, chunk_size);
}
else
io_skip(file, chunk_size);
}
if(config.debug)
dbg_msg("sound/wav", "loaded %s", filename);
return sid;
}
#endif
int snd_play(int cid, int sid, int loop, float x, float y)
{
int vid;
dbg_msg("snd", "try adding sound");
for(vid = 0; vid < NUM_VOICES; vid++)
{
if(voices[vid].sound == 0x0)
{
voices[vid].tick = 0;
voices[vid].x = x;
voices[vid].y = y;
voices[vid].sound = &sounds[sid];
if(loop == SND_LOOP)
voices[vid].loop = voices[vid].sound->loop_end;
else
voices[vid].loop = -1;
lock_wait(sound_lock);
dbg_msg("snd", "sound added");
voices[vid].next = channels[cid].first_voice;
if(channels[cid].first_voice)
channels[cid].first_voice->prev = &voices[vid];
channels[cid].first_voice = &voices[vid];
lock_release(sound_lock);
return vid;
}
}
dbg_msg("snd", "failed");
return -1;
}
void snd_set_master_volume(float val)
{
master_vol = val;
}
void snd_stop(int vid)
{
/*TODO: lerp volume to 0*/
voices[vid].stop = 0;
/*master_vol = vol;*/
}
void snd_set_listener_pos(float x, float y)
{
center_x = x;
center_y = y;
center_x = (int)x;
center_y = (int)y;
}

View file

@ -20,6 +20,8 @@ MACRO_CONFIG_INT(b_filter_pw, 0, 1, 0)
MACRO_CONFIG_INT(b_sort, 0, 0, 0)
MACRO_CONFIG_INT(b_max_requests, 10, 0, 0)
MACRO_CONFIG_INT(snd_rate, 48000, 0, 0)
MACRO_CONFIG_INT(gfx_screen_width, 800, 0, 0)
MACRO_CONFIG_INT(gfx_screen_height, 600, 0, 0)
MACRO_CONFIG_INT(gfx_fullscreen, 1, 0, 1)

View file

@ -25,6 +25,9 @@ enum
MASK_SET,
MASK_ZERO,
SNDFLAG_LOOP=1,
SNDFLAG_POS=2,
SNDFLAG_ALL=3,
CLIENTSTATE_OFFLINE=0,
CLIENTSTATE_CONNECTING,
@ -366,21 +369,21 @@ void gfx_quads_draw_freeform(
void gfx_quads_text(float x, float y, float size, const char *text);
/* sound (client) */
enum
{
SND_PLAY_ONCE = 0,
SND_LOOP
};
int snd_init();
float snd_get_master_volume();
void snd_set_master_volume(float val);
int snd_load_wav(const char *filename);
void snd_set_channel(int cid, float vol, float pan);
int snd_load_wv(const char *filename);
int snd_play(int cid, int sid, int loop, float x, float y);
int snd_play_at(int cid, int sid, int flags, float x, float y);
int snd_play(int cid, int sid, int flags);
void snd_stop(int id);
void snd_set_vol(int id, float vol);
void snd_set_listener_pos(float x, float y);
int snd_shutdown();
/*

View file

@ -16,6 +16,14 @@ extern "C" {
#include "data.h"
#include "menu.h"
// sound channels
enum
{
CHN_GUI=0,
CHN_MUSIC,
CHN_WORLD,
};
data_container *data = 0x0;
static int charids[16] = {2,10,0,4,12,6,9,1,3,15,13,11,7,5,8,14};
@ -52,7 +60,7 @@ struct client_data
inline float frandom() { return rand()/(float)(RAND_MAX); }
void snd_play_random(int setid, float vol, float pan)
void snd_play_random(int chn, int setid, float vol, vec2 pos)
{
soundset *set = &data->sounds[setid];
@ -61,7 +69,7 @@ void snd_play_random(int setid, float vol, float pan)
if(set->num_sounds == 1)
{
snd_play(0, set->sounds[0].id, SND_PLAY_ONCE, 0, 0);
snd_play_at(chn, set->sounds[0].id, 0, pos.x, pos.y);
return;
}
@ -70,7 +78,7 @@ void snd_play_random(int setid, float vol, float pan)
do {
id = rand() % set->num_sounds;
} while(id == set->last);
snd_play(0, set->sounds[id].id, SND_PLAY_ONCE, 0, 1);
snd_play_at(chn, set->sounds[id].id, 0, pos.x, pos.y);
set->last = id;
}
@ -465,6 +473,11 @@ static void render_loading(float percent)
extern "C" void modc_init()
{
// setup sound channels
snd_set_channel(CHN_GUI, 1.0f, 0.0f);
snd_set_channel(CHN_MUSIC, 1.0f, 0.0f);
snd_set_channel(CHN_WORLD, 1.0f, 1.0f);
// load the data container
data = load_data_from_memory(internal_data);
@ -653,14 +666,14 @@ static void process_events(int s)
int soundid = ev->sound; //(ev->sound & SOUND_MASK);
//bool bstartloop = (ev->sound & SOUND_LOOPFLAG_STARTLOOP) != 0;
//bool bstoploop = (ev->sound & SOUND_LOOPFLAG_STOPLOOP) != 0;
float vol, pan;
sound_vol_pan(p, &vol, &pan);
//float vol, pan;
//sound_vol_pan(p, &vol, &pan);
if(soundid >= 0 && soundid < NUM_SOUNDS)
{
// TODO: we need to control the volume of the diffrent sounds
// depening on the category
snd_play_random(soundid, vol, pan);
snd_play_random(CHN_WORLD, soundid, 1.0f, p);
}
}
}
@ -1833,6 +1846,9 @@ void render_game()
local_player_pos = mix(predicted_prev_player.pos, predicted_player.pos, client_intrapredtick());
if(local_player && local_player->team == -1)
spectate = true;
// set listner pos
snd_set_listener_pos(local_player_pos.x, local_player_pos.y);
animstate idlestate;
anim_eval(&data->animations[ANIM_BASE], 0, &idlestate);
@ -2412,7 +2428,9 @@ extern "C" void modc_render()
else // if (client_state() != CLIENTSTATE_CONNECTING && client_state() != CLIENTSTATE_LOADING)
{
if (music_menu_id == -1)
music_menu_id = snd_play(0, music_menu, SND_LOOP, 0, 0);
{
music_menu_id = snd_play(CHN_MUSIC, music_menu, SNDFLAG_LOOP);
}
//netaddr4 server_address;
if(modmenu_render(false) == -1)
@ -2446,9 +2464,9 @@ extern "C" void modc_message(int msg)
chat_add_line(cid, team, message);
if(cid >= 0)
snd_play(0, data->sounds[SOUND_CHAT_CLIENT].sounds[0].id, SND_PLAY_ONCE, 0, 0);
snd_play(CHN_GUI, data->sounds[SOUND_CHAT_CLIENT].sounds[0].id, 0);
else
snd_play(0, data->sounds[SOUND_CHAT_SERVER].sounds[0].id, SND_PLAY_ONCE, 0, 0);
snd_play(CHN_GUI, data->sounds[SOUND_CHAT_SERVER].sounds[0].id, 0);
}
else if(msg == MSG_SETNAME)
{