void dumb_source_close(audio_source *src) {
dumb_source *local = source_get_userdata(src);
duh_end_sigrenderer(local->renderer);
unload_duh(local->data);
destroy_sample_buffer(local->sig_samples);
free(local);
DEBUG("Libdumb Source: Closed.");
}
long duh_render(
DUH_SIGRENDERER *sigrenderer,
int bits, int unsign,
float volume, float delta,
long size, void *sptr
)
{
long n;
sample_t **sampptr;
int n_channels;
ASSERT(bits == 8 || bits == 16);
ASSERT(sptr);
if (!sigrenderer)
return 0;
n_channels = duh_sigrenderer_get_n_channels(sigrenderer);
ASSERT(n_channels > 0);
/* This restriction will be removed when need be. At the moment, tightly
* optimised loops exist for exactly one or two channels.
*/
ASSERT(n_channels <= 2);
sampptr = allocate_sample_buffer(n_channels, size);
if (!sampptr)
return 0;
dumb_silence(sampptr[0], n_channels * size);
size = duh_sigrenderer_generate_samples(sigrenderer, volume, delta, size, sampptr);
if (bits == 16) {
int signconv = unsign ? 0x8000 : 0x0000;
for (n = 0; n < size * n_channels; n++) {
CONVERT16(sampptr[0][n], n, signconv);
}
} else {
char signconv = unsign ? 0x80 : 0x00;
for (n = 0; n < size * n_channels; n++) {
CONVERT8(sampptr[0][n], n, signconv);
}
}
destroy_sample_buffer(sampptr);
return size;
}
long duh_render_float(DUH_SIGRENDERER *sigrenderer, sample_t ***sig_samples,
long *sig_samples_size, int bits, float volume,
float delta, long size, void *sptr) {
long n;
sample_t **sampptr;
int n_channels;
ASSERT(bits == 32 || bits == 64);
ASSERT(sptr);
ASSERT(sig_samples);
ASSERT(sig_samples_size);
if (!sigrenderer)
return 0;
n_channels = duh_sigrenderer_get_n_channels(sigrenderer);
ASSERT(n_channels > 0);
/* This restriction will be removed when need be. At the moment, tightly
* optimised loops exist for exactly one or two channels.
*/
ASSERT(n_channels <= 2);
if ((*sig_samples == NULL) || (*sig_samples_size != size)) {
destroy_sample_buffer(*sig_samples);
*sig_samples = allocate_sample_buffer(n_channels, size);
*sig_samples_size = size;
}
sampptr = *sig_samples;
if (!sampptr)
return 0;
dumb_silence(sampptr[0], n_channels * size);
size = duh_sigrenderer_generate_samples(sigrenderer, volume, delta, size,
sampptr);
if (bits == 64) {
for (n = 0; n < size * n_channels; n++) {
CONVERT64F(sampptr[0][n], n);
}
} else if (bits == 32) {
for (n = 0; n < size * n_channels; n++) {
CONVERT32F(sampptr[0][n], n);
}
}
return size;
}
int main(int argc, char *argv[]) {
int retcode = 1;
int nerrors = 0;
streamer_t streamer;
memset(&streamer, 0, sizeof(streamer_t));
// Signal handlers
signal(SIGINT, sig_fn);
signal(SIGTERM, sig_fn);
// Initialize SDL2
if (SDL_Init(SDL_INIT_AUDIO) != 0) {
fprintf(stderr, "%s\n", SDL_GetError());
return 1;
}
// Defaults
streamer.freq = 44100;
streamer.n_channels = 2;
streamer.bits = 16;
streamer.volume = 1.0f;
streamer.quality = DUMB_RQ_CUBIC;
// commandline argument parser options
struct arg_lit *arg_help =
arg_lit0("h", "help", "print this help and exits");
struct arg_dbl *arg_volume =
arg_dbl0("v", "volume", "<volume",
"sets the output volume (-8.0 to +8.0, default 1.0)");
struct arg_int *arg_samplerate = arg_int0(
"s", "samplerate", "<freq>", "sets the sampling rate (default 44100)");
struct arg_int *arg_quality = arg_int0(
"r", "quality", "<quality>", "specify the resampling quality to use");
struct arg_lit *arg_mono =
arg_lit0("m", "mono", "generate mono output instead of stereo");
struct arg_lit *arg_eight =
arg_lit0("8", "eight", "generate 8-bit instead of 16-bit");
struct arg_lit *arg_noprogress =
arg_lit0("n", "noprogress", "hide progress bar");
struct arg_file *arg_output =
arg_file0("o", "output", "<file>", "output file");
struct arg_file *arg_input =
arg_file1(NULL, NULL, "<file>", "input module file");
struct arg_end *arg_fend = arg_end(20);
void *argtable[] = {arg_help, arg_input, arg_volume,
arg_samplerate, arg_quality, arg_mono,
arg_eight, arg_noprogress, arg_fend};
const char *progname = "dumbplay";
// Make sure everything got allocated
if (arg_nullcheck(argtable) != 0) {
fprintf(stderr, "%s: insufficient memory\n", progname);
goto exit_0;
}
// Parse inputs
nerrors = arg_parse(argc, argv, argtable);
// Handle help
if (arg_help->count > 0) {
fprintf(stderr, "Usage: %s", progname);
arg_print_syntax(stderr, argtable, "\n");
fprintf(stderr, "\nArguments:\n");
arg_print_glossary(stderr, argtable, "%-25s %s\n");
goto exit_0;
}
// Handle libargtable errors
if (nerrors > 0) {
arg_print_errors(stderr, arg_fend, progname);
fprintf(stderr, "Try '%s --help' for more information.\n", progname);
goto exit_0;
}
// Handle the switch options
streamer.input = arg_input->filename[0];
if (arg_eight->count > 0) {
streamer.bits = 8;
}
if (arg_mono->count > 0) {
streamer.n_channels = 1;
}
if (arg_noprogress->count > 0) {
streamer.no_progress = true;
}
if (arg_volume->count > 0) {
streamer.volume = arg_volume->dval[0];
if (streamer.volume < -8.0f || streamer.volume > 8.0f) {
fprintf(stderr, "Volume must be between -8.0f and 8.0f.\n");
goto exit_0;
}
}
if (arg_samplerate->count > 0) {
streamer.freq = arg_samplerate->ival[0];
if (streamer.freq < 1 || streamer.freq > 96000) {
fprintf(stderr, "Sampling rate must be between 1 and 96000.\n");
goto exit_0;
}
}
if (arg_quality->count > 0) {
streamer.quality = arg_quality->ival[0];
if (streamer.quality < 0 || streamer.quality >= DUMB_RQ_N_LEVELS) {
fprintf(stderr, "Quality must be between %d and %d.\n", 0,
DUMB_RQ_N_LEVELS - 1);
goto exit_0;
}
}
// Load source file.
dumb_register_stdfiles();
streamer.src = dumb_load_any(streamer.input, 0, 0);
if (!streamer.src) {
fprintf(stderr, "Unable to load file %s for playback!\n",
streamer.input);
goto exit_0;
}
// Set up playback
streamer.renderer =
duh_start_sigrenderer(streamer.src, 0, streamer.n_channels, 0);
streamer.delta = 65536.0f / streamer.freq;
streamer.sbytes = (streamer.bits / 8) * streamer.n_channels;
streamer.ssize = duh_get_length(streamer.src);
// Stop producing samples on module end
DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer(streamer.renderer);
dumb_it_set_loop_callback(itsr, &dumb_it_callback_terminate, NULL);
dumb_it_set_xm_speed_zero_callback(itsr, &dumb_it_callback_terminate, NULL);
dumb_it_set_resampling_quality(itsr, streamer.quality);
// Set up the SDL2 format we want for playback.
SDL_AudioSpec want;
SDL_zero(want);
want.freq = streamer.freq;
want.format = (streamer.bits == 16) ? AUDIO_S16 : AUDIO_S8;
want.channels = streamer.n_channels;
want.samples = SAMPLES;
want.callback = stream_audio;
want.userdata = &streamer;
// Find SDL2 audio device, and request the format we just set up.
// SDL2 will tell us what we got in the "have" struct.
SDL_AudioSpec have;
streamer.dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (streamer.dev == 0) {
fprintf(stderr, "%s\n", SDL_GetError());
goto exit_1;
}
// Make sure we got the format we wanted. If not, stop here.
if (have.format != want.format) {
fprintf(stderr, "Could not get correct playback format.\n");
goto exit_2;
}
// Play file
SDL_PauseAudioDevice(streamer.dev, 0);
// Show initial state of the progress bar (if it is enabled)
int time_start = SDL_GetTicks();
float seek = 0.0f;
int ms_played = 0;
if (!streamer.no_progress) {
show_progress(PROGRESSBAR_LENGTH, seek, ms_played);
}
// Loop while dumb is still giving data. Update progressbar if enabled.
while (!stop_signal && !streamer.ended) {
if (!streamer.no_progress) {
seek = ((float)streamer.spos) / ((float)streamer.ssize);
ms_played = SDL_GetTicks() - time_start;
show_progress(PROGRESSBAR_LENGTH, seek, ms_played);
}
SDL_Delay(100);
}
// We made it this far without crashing, so let's just exit with no error :)
retcode = 0;
// Free up resources and exit.
if (streamer.sig_samples) {
destroy_sample_buffer(streamer.sig_samples);
}
exit_2:
SDL_CloseAudioDevice(streamer.dev);
exit_1:
if (streamer.renderer) {
duh_end_sigrenderer(streamer.renderer);
}
if (streamer.src) {
unload_duh(streamer.src);
}
exit_0:
arg_freetable(argtable, sizeof(argtable) / sizeof(argtable[0]));
SDL_Quit();
return retcode;
}
