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/*
* audio.c
*
* Sound stuff
*
* (c) 2008 Thomas White <taw27@cam.ac.uk>
*
* thrust3d - a silly game
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/errno.h>
#include <SDL.h>
#include <SDL_audio.h>
#include <vorbis/vorbisfile.h>
#include <vorbis/codec.h>
#include <pthread.h>
#include <unistd.h>
#include "types.h"
int ppp = 0;
static void audio_mix(void *data, Uint8 *stream8, int len) {
AudioContext *a = data;
int i, j, clip_count;
Sint16 *stream = (Sint16 *)stream8;
len /= 2; /* Number of samples to write */
/* Zero the buffer */
for ( j=0; j<len; j++ ) {
stream[j] = 0;
}
/* For each currently playing sound... */
clip_count = 0;
for ( i=0; i<AUDIO_MAX_SOUNDS; i++ ) {
/* Playing? */
if ( !a->sounds[i].playing ) continue;
for ( j=0; j<len; j++ ) {
Sint16 samp;
Sint32 test;
if ( (a->sounds[i].decode_pos == 0) || (a->sounds[i].dpos < a->sounds[i].decode_pos) ) {
samp = a->sounds[i].data[a->sounds[i].dpos++];
} else {
if ( a->debug ) printf("AU: Channel %i: Decoding thread underran\n", i);
samp = 0;
}
test = stream[j] + samp*a->sounds[i].volume;
if ( test > 32767 ) {
if ( stream[j] != 32767 ) {
clip_count++;
}
stream[j] = 32767;
} else if ( test < -32767 ) {
if ( stream[j] != -32767 ) {
clip_count++;
}
stream[j] = -32767;
} else {
stream[j] += samp * a->sounds[i].volume;
}
if ( a->sounds[i].dpos == a->sounds[i].dlen ) {
if ( a->sounds[i].repeat ) {
a->sounds[i].dpos = 0;
if ( a->debug ) printf("AU: Channel %i: Looping at offset %i/%i\n", i, j, len);
} else {
a->sounds[i].inuse = 0;
a->sounds[i].playing = 0;
free(a->sounds[i].data);
if ( a->debug ) printf("AU: Channel %i: End of data\n", i);
break;
}
}
}
}
/* Fast ramp-up of volume when starting program, avoids 'click' */
if ( a->startup ) {
for ( j=0; j<len; j++ ) {
stream[j] *= a->startup_volume;
a->startup_volume += 0.001;
if ( a->startup_volume > 1.0 ) {
if ( a->debug && a->startup ) printf("AU: Finished startup ramp.\n");
a->startup = 0;
a->startup_volume = 1.0;
}
}
}
if ( a->debug && (clip_count > 0) ) printf("AU: Clipped %i samples.\n", clip_count);
}
static void *audio_play_wav(void *add_void) {
AudioDispatchData *add = add_void;
AudioContext *a = add->audiocontext;
char *filename = add->filename;
int idx;
SDL_AudioSpec wave;
Uint8 *data;
Uint32 dlen;
SDL_AudioCVT cvt;
/* Look for an empty sound slot */
pthread_mutex_lock(&a->sounds_mutex);
for ( idx=0; idx<AUDIO_MAX_SOUNDS; idx++ ) {
if ( !a->sounds[idx].inuse ) break;
}
if ( idx == AUDIO_MAX_SOUNDS ) {
fprintf(stderr, "Not enough audio channels to play '%s'\n", filename);
pthread_mutex_unlock(&a->sounds_mutex);
goto out;
}
a->sounds[idx].inuse = 1;
pthread_mutex_unlock(&a->sounds_mutex);
if ( a->debug ) printf("AU: Channel %i: Selected this channel for sound '%s'\n", idx, filename);
/* Load the sound file and convert it to 16-bit stereo at 44.1 kHz */
if ( SDL_LoadWAV(filename, &wave, &data, &dlen) == NULL ) {
fprintf(stderr, "Couldn't load %s: (channel %i) %s\n", filename, idx, SDL_GetError());
a->sounds[idx].inuse = 0;
goto out;
}
SDL_BuildAudioCVT(&cvt, wave.format, wave.channels, wave.freq, AUDIO_S16, 2, 44100);
cvt.buf = malloc(dlen*cvt.len_mult);
if ( cvt.buf == NULL ) {
fprintf(stderr, "Not enough memory to convert audio (channel %i)\n", idx);
a->sounds[idx].inuse = 0;
SDL_FreeWAV(data);
free(cvt.buf);
goto out;
}
memcpy(cvt.buf, data, dlen);
cvt.len = dlen;
SDL_ConvertAudio(&cvt);
SDL_FreeWAV(data);
/* Put the sound data in the slot */
a->sounds[idx].data = (Sint16 *)cvt.buf;
/* Not entirely sure whose fault it is that the last (stereo) sample is wrong */
a->sounds[idx].dlen = cvt.len_cvt / 2 - 2; /* Convert bytes to samples */
a->sounds[idx].dpos = 0;
a->sounds[idx].repeat = add->repeat;
a->sounds[idx].volume = add->volume;
a->sounds[idx].decode_pos = 0;
a->sounds[idx].playing = 1; /* Must be done last - tell the mixer thread it can use this */
out:
free(add->filename);
if ( a->debug ) printf("AU: Channel %i: WAV dispatch thread completed.\n", idx);
a->dispatch_threads_inuse[add->idx] = 0;
free(add);
return NULL;
}
static void *audio_play_vorbis(void *add_void) {
AudioDispatchData *add = add_void;
AudioContext *a = add->audiocontext;
char *filename = add->filename;
int idx;
char *data;
int len;
OggVorbis_File vf;
vorbis_info *vi;
int finished, err, current_section, decode_done, decode_samples_done;
size_t offs;
SDL_AudioCVT cvt;
/* Look for an empty sound slot */
pthread_mutex_lock(&a->sounds_mutex);
for ( idx=0; idx<AUDIO_MAX_SOUNDS; idx++ ) {
if ( !a->sounds[idx].inuse ) break;
}
if ( idx == AUDIO_MAX_SOUNDS ) {
fprintf(stderr, "Not enough audio channels to play '%s'\n", filename);
pthread_mutex_unlock(&a->sounds_mutex);
goto out;
}
a->sounds[idx].inuse = 1;
pthread_mutex_unlock(&a->sounds_mutex);
if ( a->debug ) printf("AU: Channel %i: Selected this channel for sound '%s'\n", idx, filename);
err = ov_fopen(filename, &vf);
if ( err != 0 ) {
fprintf(stderr, "Couldn't open Vorbis file '%s' (channel %i, code %i,%i)\n", filename, idx, err, errno);
a->sounds[idx].inuse = 0;
goto out;
}
len = ov_pcm_total(&vf, -1); /* Length in samples 'per channel' */
if ( a->debug ) printf("AU: Channel %i: Length is %i samples 'per channel'\n", idx, len);
vi = ov_info(&vf,-1);
if ( a->debug ) printf("AU: Channel %i: %i channels, %li Hz\n", idx, vi->channels, vi->rate);
/* Decode ten seconds of audio every 8 seconds until the job's done */
if ( a->debug ) printf("AU: Channel %i: Started decoding Vorbis stream\n", idx);
/* Do this now because cvt.len_mult is needed to work out the size of 'data' */
SDL_BuildAudioCVT(&cvt, AUDIO_S16, vi->channels, vi->rate, AUDIO_S16, 2, 44100);
data = malloc(10*vi->rate*2*vi->channels*cvt.len_mult);
if ( data == NULL ) {
fprintf(stderr, "Not enough memory to decode Vorbis stream (channel %i)\n", idx);
a->sounds[idx].inuse = 0;
ov_clear(&vf);
goto out;
}
decode_done = 0;
decode_samples_done = 0; /* Number of samples from the Vorbis file (i.e. BEFORE CONVERSION) */
a->sounds[idx].decode_pos = 0; /* Position (in sounds[idx].data) at which to write the next block */
while ( !decode_done ) {
long decode_block_samples; /* Number of samples BEFORE CONVERSION */
long decode_block_length;
int w;
int i;
Sint16 *cvtbuf16;
/* Decide how much data to shovel this time */
decode_block_samples = 10*vi->rate; /* 10 seconds in samples */
if ( decode_samples_done + decode_block_samples > len ) {
/* This is the last block. Yay! */
decode_block_samples = len - decode_samples_done;
decode_done = 1;
}
decode_block_length = vi->channels*2*decode_block_samples;
decode_samples_done += decode_block_samples;
/* Read the chosen amount of data */
offs = 0; finished = 0;
while ( finished == 0 ) {
long rval;
rval = ov_read(&vf, data+offs, decode_block_length-offs, 0, 2, 1, ¤t_section);
if ( rval < 0 ) {
fprintf(stderr, "Vorbis stream error (channel %i)\n", idx);
} else {
offs += rval;
}
if ( offs == decode_block_length ) finished = 1;
}
/* Convert to 44.1 kHz */
cvt.buf = (Uint8 *)data;
cvt.len = decode_block_length;
SDL_ConvertAudio(&cvt);
/* Paste this seamlessly into the playing thread's buffer */
if ( !a->sounds[idx].playing ) {
a->sounds[idx].data = malloc(len*vi->channels*2);
}
cvtbuf16 = (Sint16 *)cvt.buf;
for ( i=0; i<decode_block_samples*2; i++ ) {
a->sounds[idx].data[i+a->sounds[idx].decode_pos] = cvtbuf16[i];
}
a->sounds[idx].decode_pos += decode_block_samples*2;
/* It's safe to start playing at this point */
if ( !a->sounds[idx].playing ) {
a->sounds[idx].dlen = 2*(len-1);
a->sounds[idx].dpos = 0;
a->sounds[idx].repeat = add->repeat;
a->sounds[idx].volume = add->volume;
a->sounds[idx].playing = 1; /* Must be done last - tell the mixer thread it can use this */
}
if ( a->debug ) printf("AU: Channel %i: decode_pos=%i, dpos=%li. "
"I am now %.1f seconds ahead of the playing position.\n",
idx, a->sounds[idx].decode_pos, a->sounds[idx].dpos,
((double)a->sounds[idx].decode_pos-a->sounds[idx].dpos)/(44100*2));
/* Sleep for eight seconds while periodically checking for shutdown */
for ( w=0; w<80; w++ ) {
if ( a->shutdown ) {
ov_clear(&vf);
free(data);
if ( a->debug ) printf("AU: Channel %i: Dispatch thread aborting.\n", idx);
goto out;
}
usleep(100000); /* 0.1 seconds */
}
}
if ( a->debug ) printf("AU: Channel %i: Finished decoding the Vorbis stream.\n", idx);
free(data);
/* Needed until now */
ov_clear(&vf);
out:
free(add->filename);
if ( a->debug ) printf("AU: Channel %i: Vorbis dispatch thread completed.\n", idx);
a->dispatch_threads_inuse[add->idx] = 0;
free(add);
return NULL;
}
void audio_play(AudioContext *a, char *name, float volume, int repeat) {
char filename[128];
struct stat statbuf;
if ( a == NULL ) return;
/* Try to find an Ogg/Vorbis file */
snprintf(filename, 127, "%s%s.ogg", DATADIR"/sound/", name);
if ( stat(filename, &statbuf) == 0 ) {
int rval, idx;
AudioDispatchData *add;
pthread_attr_t attr;
add = malloc(sizeof(AudioDispatchData));
add->audiocontext = a;
add->filename = strdup(filename);
add->volume = volume;
add->repeat = repeat;
/* Claim a dispatch thread slot */
for ( idx=0; idx<AUDIO_MAX_SOUNDS; idx++ ) {
if ( !a->dispatch_threads_inuse[idx] ) break;
}
if ( idx == AUDIO_MAX_SOUNDS ) {
fprintf(stderr, "Not enough dispatch thread space to play '%s'\n", filename);
free(add->filename);
free(add);
return;
}
a->dispatch_threads_inuse[idx] = 1;
add->idx = idx;
/* Start the dispatch thread */
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
rval = pthread_create(&a->dispatch_threads[idx], &attr, audio_play_vorbis, add);
pthread_attr_destroy(&attr);
if ( rval != 0 ) {
fprintf(stderr, "Couldn't create audio dispatch thread (%i)\n", rval);
free(add->filename);
free(add);
a->dispatch_threads_inuse[idx] = 0;
return;
}
goto done;
}
/* Try to find a WAV file */
snprintf(filename, 127, "%s%s.wav", DATADIR"/sound/", name);
if ( stat(filename, &statbuf) == 0 ) {
int rval, idx;
AudioDispatchData *add;
pthread_attr_t attr;
add = malloc(sizeof(AudioDispatchData));
add->audiocontext = a;
add->filename = strdup(filename);
add->volume = volume;
add->repeat = repeat;
/* Claim a dispatch thread slot */
for ( idx=0; idx<AUDIO_MAX_SOUNDS; idx++ ) {
if ( !a->dispatch_threads_inuse[idx] ) break;
}
if ( idx == AUDIO_MAX_SOUNDS ) {
fprintf(stderr, "Not enough dispatch thread space to play '%s'\n", filename);
free(add->filename);
free(add);
return;
}
a->dispatch_threads_inuse[idx] = 1;
add->idx = idx;
/* Start the dispatch thread */
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
rval = pthread_create(&a->dispatch_threads[idx], &attr, audio_play_wav, add);
pthread_attr_destroy(&attr);
if ( rval != 0 ) {
fprintf(stderr, "Couldn't create audio dispatch thread (%i)\n", rval);
free(add->filename);
free(add);
a->dispatch_threads_inuse[idx] = 0;
return;
}
goto done;
}
/* Still not found */
fprintf(stderr, "Couldn't find file for sound '%s'\n", name);
return;
done: /* Success */
if ( a->paused == 1 ) {
SDL_PauseAudio(0);
a->paused = 0;
}
return;
}
/* SDL audio initial setup */
AudioContext *audio_setup(int debug, int no_music) {
AudioContext *a;
SDL_AudioSpec fmt;
int i;
/* Create audio context */
a = malloc(sizeof(AudioContext));
if ( a == NULL ) return NULL;
/* Initialise audio context */
a->debug = debug;
a->startup = 1;
a->startup_volume = 0.0;
a->paused = 1;
a->shutdown = 0;
pthread_mutex_init(&a->sounds_mutex, NULL);
for ( i=0; i<AUDIO_MAX_SOUNDS; i++ ) {
a->sounds[i].inuse = 0;
a->sounds[i].playing = 0;
a->dispatch_threads[i] = 0;
a->dispatch_threads_inuse[i] = 0;
}
/* 16-bit stereo audio at 44.1 kHz */
fmt.freq = 44100;
fmt.format = AUDIO_S16;
fmt.channels = 2;
fmt.samples = 512;
fmt.callback = audio_mix;
fmt.userdata = a;
fmt.silence = 0;
if ( SDL_OpenAudio(&fmt, NULL) < 0 ) {
fprintf(stderr, "Unable to open audio: %s\n", SDL_GetError());
free(a);
return NULL;
}
if ( !no_music ) {
audio_play(a, "music", 0.2, 1);
} else {
if ( a->debug ) printf("AU: Music disabled.\n");
}
audio_play(a, "moan", 0.2, 1);
return a;
}
void audio_shutdown(AudioContext *a) {
int i;
if ( a == NULL ) return;
a->shutdown = 1;
SDL_CloseAudio();
/* Wait for dispatch threads */
for ( i=0; i<AUDIO_MAX_SOUNDS; i++ ) {
if ( a->dispatch_threads_inuse[i] ) {
pthread_join(a->dispatch_threads[i], NULL);
}
}
/* Now that all the dispatch threads are done, and the mixer callback
is no longer getting called, the sound system is in a consistent
state. i.e. 'inuse' means memory is allocated and can be freed */
for ( i=0; i<AUDIO_MAX_SOUNDS; i++ ) {
if ( a->sounds[i].inuse ) {
free(a->sounds[i].data);
}
}
/* Now this can be freed */
free(a);
}
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