path: root/src/audio.c

/*
 * audio.c
 *
 * Sound stuff
 *
 * Copyright (c) 2008 Thomas White <taw27@cam.ac.uk>
 *
 * This file is part of Thrust3D - a silly game
 *
 * Thrust3D 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 3 of the License, or
 * (at your option) any later version.
 *
 * Thrust3D 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 Thrust3D.  If not, see <http://www.gnu.org/licenses/>.
 *
 */

#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"

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, &current_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=%li, dpos=%li. "
				       "I am now %.1f seconds ahead of the playing position.\n",
				       idx, (long int)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);
	
}

void audio_pause(AudioContext *a) {
	SDL_PauseAudio(1);
}

void audio_unpause(AudioContext *a) {
	SDL_PauseAudio(0);
}