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; }