rework audio system to be able to call it from anywhere, rather than having to have access to the audio data.

src/io/audio.c

@@ -11,22 +11,28 @@
 
 #include "../error.h"
 #include "../util/compat.h"
+#include "../util/types.h"
 
 struct audioplayer {
-	uint8_t const* buf;
+	u8 const* buf;
 	int len;
 };
 
-struct audiodevice {
+static struct audiodevice {
 	struct audioplayer audio[AUDIO_MAX];
-	unsigned id;
-	int freq;
-	SDL_AudioFormat fmt;
-	uint8_t ch; // audio channels
+	u32 id;
+	SDL_AudioSpec spec;
 } dev;
 
+struct audiodata audio_dat[AUDIO_ID_COUNT] = {0}; // contains pointers to audio buffers.
+
+static char const* const audio_path[AUDIO_ID_COUNT] = {
+	"korobeiniki.wav",
+	"place.wav",
+};
+
 /* mixes the audio output stream, using the different audio as sources */
-static void audiomixer(void* const userdata, uint8_t* const stream, int const len) {
+static void audiomixer(void* const userdata, u8* const stream, int const len) {
 	(void)userdata;
 	memset(stream, 0, len); // clear the playing audio
 
@@ -34,7 +40,7 @@ static void audiomixer(void* const userdata, uint8_t* const stream, int const le
 	for (unsigned i = 0; i < AUDIO_MAX; i++) {
 		if (dev.audio[i].len > 0) {
 			unsigned mixlen = SDL_min(len, dev.audio[i].len);
-			SDL_MixAudioFormat(stream, dev.audio[i].buf, dev.fmt, mixlen, SDL_MIX_MAXVOLUME);
+			SDL_MixAudioFormat(stream, dev.audio[i].buf, dev.spec.format, mixlen, SDL_MIX_MAXVOLUME);
 			dev.audio[i].len -= mixlen;
 			dev.audio[i].buf += mixlen;
 		}
@@ -45,12 +51,12 @@ static void audiomixer(void* const userdata, uint8_t* const stream, int const le
  * `len` is a pointer to the current size, the new size will be written to this location.
  * returns the pointer to the audio buffer to use, or NULL, when something went wrong.
  * NULL will never be returned after the conversion */
-static uint8_t* audio_cvt(SDL_AudioSpec const* spec, uint8_t* bufptr, unsigned* len) {
+static u8* audio_cvt(SDL_AudioSpec const* spec, u8* bufptr, unsigned* len) {
 	if (!bufptr) return NULL;
 
 	// init the converter
 	SDL_AudioCVT cvt;
-	if (SDL_BuildAudioCVT(&cvt, spec->format, spec->channels, spec->freq, dev.fmt, dev.ch, dev.freq) < 0) {
+	if (SDL_BuildAudioCVT(&cvt, spec->format, spec->channels, spec->freq, dev.spec.format, dev.spec.channels, dev.spec.freq) < 0) {
 		error("could not build the audio converter! SDL Error: %s", SDL_GetError());
 		return NULL;
 	}
@@ -78,72 +84,68 @@ static uint8_t* audio_cvt(SDL_AudioSpec const* spec, uint8_t* bufptr, unsigned*
 	return bufptr ? bufptr : cvt.buf;
 }
 
-int audio_device_init(int freq, SDL_AudioFormat fmt, uint8_t channels, uint16_t samples) {
-	// define the audio specification
-	SDL_AudioSpec spec = {freq, fmt, channels, 0, samples, 0, 0, NULL, NULL};
-	spec.callback = audiomixer;
-	spec.userdata = NULL;
-
-	unsigned id = SDL_OpenAudioDevice(NULL, 0, &spec, NULL, 0);
-
-	// create the audio device
-	dev = (struct audiodevice){
-		{0},
-		id,
-		freq,
-		fmt,
-		channels,
-	};
-
-	if (id) {
-		// default state of the device is paused, so we unpause it here
-		SDL_PauseAudioDevice(id, 0);
-		return 0;
-	} else return 1;
+/* computes the time in milliseconds of the audio fragment by dividing
+ * the audio byte length by the format's bit size, which is divided by
+ * the audio device's channels and frequency */
+static inline u32 audio_btoms(u32 len) {
+	return (((1000 * len) / (SDL_AUDIO_BITSIZE(dev.spec.format) / 8)) / dev.spec.channels / dev.spec.freq);
 }
 
-void audio_device_free(void) {
-	SDL_CloseAudioDevice(dev.id);
-	dev = (struct audiodevice){0};
-}
-
-void audio_play(audiodata const* audio) {
-	if (!dev.id) return;
-	if (!audio->buf) return;
-
-	unsigned i = 0;
-	while (i < AUDIO_MAX && dev.audio[i].len > 0) i++;
-	if (i < AUDIO_MAX)
-		dev.audio[i] = (struct audioplayer){audio->buf, audio->len};
-}
-
-audiodata audio_wav_load(char const* fpath) {
-	debug("loading audio file '%s'...", fpath);
-
-	if (faccess(fpath, FA_R)) {
-		error("%s:%u audio file either isn't readable or doesn't exist. path: '%s'!", __FILE_NAME__, __LINE__, fpath);
-		return (struct audiodata){0};
+/* loads a `struct audiodata` from `fpat` to `out`. */
+static void audio_wav_load(char const* restrict fpat, struct audiodata* restrict out) {
+	debug("loading audio file '%s'...", fpat);
+	if (faccess(fpat, FA_R)) {
+		error("audio file either isn't readable or doesn't exist. path: '%s'!", fpat);
+		return;
 	}
 
 	// load the audio
+	u32 len;
+	u8 *ptr, *tmp;
 	SDL_AudioSpec spec;
-	audiodata audio;
-	SDL_LoadWAV(fpath, &spec, &audio.buf, &audio.len);
-	if (!audio.buf) return audio;
+	SDL_LoadWAV(fpat, &spec, &ptr, &len);
 
 	// convert the audio data to the format reflecting dev
-	void* ptr = audio_cvt(&spec, audio.buf, &audio.len);
-	if (!ptr) free(audio.buf); // free the buffer if NULL was returned; failure
-	audio.buf = ptr;
+	tmp = audio_cvt(&spec, ptr, &len);
+	if (!tmp) free(ptr); // free the buffer if NULL was returned; failure
 
-	// calculate the time in milliseconds of the audio fragment
-	// by dividing the audio bytelength by the format's bitsize, by the audio device's channels and the audio device's frequency
-	audio.ms = (((1000 * audio.len) / (SDL_AUDIO_BITSIZE(dev.fmt) / 8)) / dev.ch / dev.freq);
-
-	return audio;
+	*out = (struct audiodata){tmp, len, audio_btoms(len)};
 }
 
-void audio_wav_unload(audiodata* audio) {
-	free(audio->buf);
-	*audio = (audiodata){0}; // zero out all audio data
+/* loads the audio data into the buffer */
+static inline void audio_load(void) {
+	for (size_t i = 0; i < AUDIO_ID_COUNT; i++)
+		audio_wav_load(audio_path[i], &audio_dat[i]);
+}
+
+void audio_play(u32 audio_id) {
+	if (!dev.id) return;
+	size_t i = 0;
+	while (i < AUDIO_MAX && dev.audio[i].len > 0) i++;
+	if (i >= AUDIO_MAX) return;
+
+	dev.audio[i] = (struct audioplayer){
+		audio_dat[audio_id].buf,
+		audio_dat[audio_id].len,
+	};
+}
+
+int audio_init(int freq, SDL_AudioFormat fmt, u8 ch, u16 samples) {
+	// initialise the audio device + specification
+	SDL_AudioSpec spec = {freq, fmt, ch, 0, samples, 0, 0, audiomixer, NULL};
+	unsigned id = SDL_OpenAudioDevice(NULL, 0, &spec, &spec, 0);
+	dev = (struct audiodevice){{0}, id, spec};
+
+	if (!id) return 1;
+	SDL_PauseAudioDevice(id, 0);
+	audio_load();
+	return 0;
+}
+
+void audio_free(void) {
+	SDL_CloseAudioDevice(dev.id);
+	dev = (struct audiodevice){0};
+
+	for (size_t i = 0; i < AUDIO_ID_COUNT; i++)
+		free((void*)audio_dat[i].buf);
 }

src/io/audio.h

@@ -3,18 +3,31 @@
 #include <SDL_audio.h>
 #include <stdint.h>
 
+#include "../util/types.h"
+
 #define AUDIO_MAX 4 // maximum number of sound effects that are allowed to play at once
 
 struct audiodata {
-	uint8_t* buf; // pointer to the audio buffer
-	uint32_t len; // length in bytes of the audio buffer
-	uint32_t ms;  // length in miliseconds of the audio buffer
+	u8 const* buf; // pointer to the audio buffer
+	u32 len;       // length in bytes of the audio buffer
+	u32 ms;        // length in miliseconds of the audio buffer
 };
-typedef struct audiodata audiodata;
 
-int audio_device_init(int, SDL_AudioFormat, uint8_t, uint16_t);
-void audio_device_free(void);
-void audio_play(audiodata const*);
+enum audio_id {
+	AUDIO_ID_MUSIC,
+	AUDIO_ID_PLACE,
 
-audiodata audio_wav_load(char const*);
-void audio_wav_unload(audiodata*);
+	// leave at end, will contain count
+	AUDIO_ID_COUNT,
+};
+
+extern struct audiodata audio_dat[AUDIO_ID_COUNT];
+
+/* loads the audio to be played, unless `AUDIO_MAX` has been reached, then this call will fail */
+void audio_play(u32);
+
+/* initialises the audio device, then loads the audio data */
+int audio_init(int freq, SDL_AudioFormat fmt, u8 ch, u16 samples);
+
+/* frees up resources held by the audio device and audio buffers */
+void audio_free(void);

src/io/window.c

@@ -17,7 +17,6 @@
 
 static SDL_Window* win = NULL;
 static bool close = false;
-static audiodata music;
 
 void window_init(struct gamedata const* gdat) {
 	assert(!win && !close);
@@ -31,8 +30,7 @@ void window_init(struct gamedata const* gdat) {
 		fatal(ERROR_SDL_RENDERING_INIT, "Window failed to be created! SDL Error: %s", SDL_GetError());
 
 	render_init(win, gdat);
-	audio_device_init(32000, AUDIO_S16, 1, 4096);
-	music = audio_wav_load("korobeiniki.wav");
+	audio_init(32000, AUDIO_S16, 1, 4096);
 }
 
 void window_free(void) {
@@ -42,8 +40,7 @@ void window_free(void) {
 	win = NULL;
 	close = false;
 
-	audio_wav_unload(&music);
-	audio_device_free();
+	audio_free();
 }
 
 void window_open(void) {
@@ -53,8 +50,8 @@ void window_open(void) {
 		render_update();
 
 		static time_t timeout = 0;
-		if (time_poll(time, music.ms, &timeout))
-			audio_play(&music);
+		if (time_poll(time, audio_dat[AUDIO_ID_MUSIC].ms, &timeout))
+			audio_play(AUDIO_ID_MUSIC);
 	}
 }