Sound_Sample *Sound_create_stream(char *file)
{
int n_ext=6;
char *ext[6]={".WAV",".OGG",".MP3",".wav",".ogg",".mp3"};
char name[256],name2[256];
int i;
Sound_AudioInfo inf;
inf.format=AUDIO_S16;
inf.channels=2;
inf.rate=44100;
if (sound_enabled) {
for(i=0;i<n_ext;i++) {
strcpy(name,file);
strcat(name,ext[i]);
sprintf(name2,"%s%s",s_path,name);
if (file_check(name2)) return Sound_NewSampleFromFile(name2,&inf,AUDIO_BUFFER);
// if (file_check(name2)) return Sound_NewSampleFromFile(name2,0,AUDIO_BUFFER);
} /* for */
for(i=0;i<n_ext;i++) {
strcpy(name,file);
strcat(name,ext[i]);
sprintf(name2,"%s%s",default_s_path,name);
if (file_check(name2)) return Sound_NewSampleFromFile(name2,&inf,AUDIO_BUFFER);
// if (file_check(name2)) return Sound_NewSampleFromFile(name2,0,AUDIO_BUFFER);
} /* for */
fprintf(stderr,"ERROR in Sound_create_stream(): Could not load sound file: %s%s.(wav|ogg|mp3)\n",s_path, file);
exit(1);
} else {
return 0;
} /* if */
} /* Sound_create_stream */
int openAudio_platform(AudioInstance* instance)
{
char path[512];
char fullPath[512];
getcwd(path, sizeof(path));
snprintf(fullPath, sizeof(fullPath)-1, "%s\\data\\%s", path, instance->mClip->mPath);
char* slash = strchr(fullPath, '/');
while(slash)
{
*slash = '\\';
slash = strchr(slash+1, '/');
}
instance->mAudio.mSample = Sound_NewSampleFromFile(fullPath, NULL, SDL_BUFFER_SIZE);
if(instance->mAudio.mSample == NULL)
{
dprintf("%s", Sound_GetError());
return -1;
}
dprintf("Opened as:\n channels: %d\n format: 0x%x\n rate: %d",
instance->mAudio.mSample->actual.channels,
instance->mAudio.mSample->actual.format,
instance->mAudio.mSample->actual.rate
);
int bytes;
switch(instance->mAudio.mSample->actual.channels)
{
case 1:
if((instance->mAudio.mSample->actual.format & 0x0008) == 0x0008)
{
instance->mFormat = AL_FORMAT_MONO8;
bytes = 1;
}
else
{
instance->mFormat = AL_FORMAT_MONO16;
bytes = 2;
}
break;
case 2:
if((instance->mAudio.mSample->actual.format & 0x0008) == 0x0008)
{
instance->mFormat = AL_FORMAT_STEREO8;
bytes = 1;
}
else
{
instance->mFormat = AL_FORMAT_STEREO16;
bytes = 2;
}
break;
default:
dprintf("WTF");
}
instance->mSampleRate = instance->mAudio.mSample->actual.rate;
instance->mAudio.mReadFreq = ((float)SDL_BUFFER_SIZE/(float)(instance->mSampleRate * bytes)) * 1000000.0f;
instance->mAudio.mReadFreq -= 500000;
instance->mAudio.mNextRead = getTime();
return 0;
}
int
main (int argc, char **argv)
{
Sound_AudioInfo info;
Sound_Sample *stream;
SDL_AudioSpec spec;
/* Handle command line options. */
if (options (argc, argv) < 0) {
puts ("Usage: amaranth OPTIONS FILE\n\n");
puts ("Options\n");
puts (" -r RATE sampling rate bits/second (default: 441000)\n");
puts (" -f FMT sample format (default: signed 16-bit little-endian)\n");
puts (" -c COUNT number of channels (default: 2)\n");
puts (" -s SIZE audio buffer size in samples (default: 4096)\n");
puts ("\nFormats accepted by -f option:\n");
puts (" S16LSB signed 16-bit little-endian\n");
return 1;
}
/* Initialize SDL audio subsystem. */
if (SDL_Init (SDL_INIT_AUDIO) < 0) {
fprintf (stderr, "amaranth: cannot initialize SDL audio: %s\n",
SDL_GetError ());
return 1;
}
/* Initialize SDL_sound library. */
if (Sound_Init () < 0) {
fprintf (stderr, "amaranth: cannot initialize SDL_sound: %s\n",
Sound_GetError ());
return 1;
}
/* Open the sound stream. */
info.format = sample_format;
info.channels = channel_count;
info.rate = sample_rate;
stream = Sound_NewSampleFromFile (filename, &info, 4096);
if (stream == NULL) {
fprintf (stderr, "amaranth: cannot open sound stream %s: %s\n",
filename, Sound_GetError ());
return 1;
}
/* Open the audio device. */
spec.freq = sample_rate;
spec.format = sample_format;
spec.channels = channel_count;
spec.samples = samples;
spec.callback = stream_callback;
spec.userdata = (void *) stream;
if (SDL_OpenAudio (&spec, NULL) < 0) {
fprintf (stderr, "amaranth: cannot open audio device: %s\n",
SDL_GetError ());
return 1;
}
/* Start playing audio and engage infinite loop. */
SDL_PauseAudio (0);
for (;;) SDL_Delay (10000);
/* NOT REACHED */
return 0;
}
static void playOneSoundFile(const char *fname)
{
PlaysoundAudioCallbackData data;
memset(&data, '\0', sizeof (PlaysoundAudioCallbackData));
data.sample = Sound_NewSampleFromFile(fname, NULL, 65536);
if (data.sample == NULL)
{
fprintf(stderr, "Couldn't load '%s': %s.\n", fname, Sound_GetError());
return;
} /* if */
/*
* Open device in format of the the sound to be played.
* We open and close the device for each sound file, so that SDL
* handles the data conversion to hardware format; this is the
* easy way out, but isn't practical for most apps. Usually you'll
* want to pick one format for all the data or one format for the
* audio device and convert the data when needed. This is a more
* complex issue than I can describe in a source code comment, though.
*/
data.devformat.freq = data.sample->actual.rate;
data.devformat.format = data.sample->actual.format;
data.devformat.channels = data.sample->actual.channels;
data.devformat.samples = 4096; /* I just picked a largish number here. */
data.devformat.callback = audio_callback;
data.devformat.userdata = &data;
if (SDL_OpenAudio(&data.devformat, NULL) < 0)
{
fprintf(stderr, "Couldn't open audio device: %s.\n", SDL_GetError());
Sound_FreeSample(data.sample);
return;
} /* if */
printf("Now playing [%s]...\n", fname);
SDL_PauseAudio(0); /* SDL audio device is "paused" right after opening. */
global_done_flag = 0; /* the audio callback will flip this flag. */
while (!global_done_flag)
SDL_Delay(10); /* just wait for the audio callback to finish. */
/* at this point, we've played the entire audio file. */
SDL_PauseAudio(1); /* so stop the device. */
/*
* Sleep two buffers' worth of audio before closing, in order
* to allow the playback to finish. This isn't always enough;
* perhaps SDL needs a way to explicitly wait for device drain?
* Most apps don't have this issue, since they aren't explicitly
* closing the device as soon as a sound file is done playback.
* As an alternative for this app, you could also change the callback
* to write silence for a call or two before flipping global_done_flag.
*/
SDL_Delay(2 * 1000 * data.devformat.samples / data.devformat.freq);
/* if there was an error, tell the user. */
if (data.sample->flags & SOUND_SAMPLEFLAG_ERROR)
fprintf(stderr, "Error decoding file: %s\n", Sound_GetError());
Sound_FreeSample(data.sample); /* clean up SDL_Sound resources... */
SDL_CloseAudio(); /* will reopen with next file's format. */
} /* playOneSoundFile */
