コード例 #1
0
ファイル: acsound.cpp プロジェクト: smarinel/ags-web
void al_duh_set_loop(AL_DUH_PLAYER *dp, int loop) {
  DUH_SIGRENDERER *sr = al_duh_get_sigrenderer(dp);
  DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer(sr);
  if (itsr == NULL)
    return;

  if (loop)
    dumb_it_set_loop_callback(itsr, NULL, NULL);
  else
    dumb_it_set_loop_callback(itsr, dumb_it_callback_terminate, itsr);
}
コード例 #2
0
ファイル: cdumb.c プロジェクト: saivert/deadbeef
static int
cdumb_read (DB_fileinfo_t *_info, char *bytes, int size) {
    trace ("cdumb_read req %d\n", size);
    dumb_info_t *info = (dumb_info_t *)_info;
    int samplesize = (_info->fmt.bps >> 3) * _info->fmt.channels;
    int length = size / samplesize;
    long ret;

    DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer (info->renderer);
    if (conf_play_forever && info->can_loop)
        dumb_it_set_loop_callback (itsr, &cdumb_it_callback_loop_forever, NULL);
    else
        dumb_it_set_loop_callback (itsr, &dumb_it_callback_terminate, NULL);

    ret = duh_render (info->renderer, _info->fmt.bps, 0, 1, 65536.f / _info->fmt.samplerate, length, bytes);
    _info->readpos += ret / (float)_info->fmt.samplerate;
    trace ("cdumb_read %d\n", ret*samplesize);
    return ret*samplesize;
}
コード例 #3
0
ファイル: cdumb.c プロジェクト: saivert/deadbeef
static int
cdumb_startrenderer (DB_fileinfo_t *_info) {
    dumb_info_t *info = (dumb_info_t *)_info;
    // reopen
    if (info->renderer) {
        duh_end_sigrenderer (info->renderer);
        info->renderer = NULL;
    }
    info->renderer = duh_start_sigrenderer (info->duh, 0, 2, 0);
    if (!info->renderer) {
        return -1;
    }

    DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer (info->renderer);
    dumb_it_set_loop_callback (itsr, &dumb_it_callback_terminate, NULL);

    int q = conf_resampling_quality;
    if (q < 0) {
        q = 0;
    }
    else if (q >= DUMB_RQ_N_LEVELS) {
        q = DUMB_RQ_N_LEVELS - 1;
    }

    dumb_it_set_resampling_quality (itsr, q);
    dumb_it_set_xm_speed_zero_callback (itsr, &dumb_it_callback_terminate, NULL);
    dumb_it_set_global_volume_zero_callback (itsr, &dumb_it_callback_terminate, NULL);

    int rq = conf_ramping_style;
    if (rq < 0) {
        rq = 0;
    }
    else if (rq > 2) {
        rq = 2;
    }
    dumb_it_set_ramp_style(itsr, rq);

    dumb_it_sr_set_global_volume (itsr, conf_global_volume);
    return 0;
}
コード例 #4
0
ファイル: dumbplay.c プロジェクト: JoeyZheng/deadbeef
int main(int argc, const char *const *argv) /* I'm const-crazy! */
{
	DUH *duh;          /* Encapsulates the music file. */
	AL_DUH_PLAYER *dp; /* Holds the current playback state. */

	/* Initialise Allegro */
	if (allegro_init())
		return EXIT_FAILURE;

	/* Check that we have one argument (plus the executable name). */
	if (argc != 2)
		usage(argv[0]);

	/* Tell Allegro where to find configuration data. This means you can
	 * put any settings for Allegro in dumb.ini. See Allegro's
	 * documentation for more information.
	 */
	set_config_file("dumb.ini");

	/* Initialise Allegro's keyboard input. */
	if (install_keyboard()) {
		allegro_message("Failed to initialise keyboard driver!\n");
		return EXIT_FAILURE;
	}

	/* This function call is appropriate for a program that will play one
	 * sample or one audio stream at a time. If you have sound effects
	 * too, you may want to increase the parameter. See Allegro's
	 * documentation for details on what the parameter means. Note that
	 * newer versions of Allegro act as if set_volume_per_voice() was
	 * called with parameter 1 initially, while older versions behave as
	 * if -1 was passed, so you should call the function if you want
	 * consistent behaviour.
	 */
	set_volume_per_voice(0);

	/* Initialise Allegro's sound output system. */
	if (install_sound(DIGI_AUTODETECT, MIDI_NONE, NULL)) {
		allegro_message("Failed to initialise sound driver!\n%s\n", allegro_error);
		return EXIT_FAILURE;
	}

	/* dumb_exit() is a function defined by DUMB. This operation arranges
	 * for dumb_exit() to be called last thing before the program exits.
	 * dumb_exit() does a bit of cleaning up for you. atexit() is
	 * declared in stdlib.h.
	 */
	atexit(&dumb_exit);

	/* DUMB defines its own wrappers for file input. There is a struct
	 * called DUMBFILE that holds function pointers for the various file
	 * operations needed by DUMB. You can decide whether to use stdio
	 * FILE objects, Allegro's PACKFILEs or something else entirely. No
	 * wrapper is installed initially, so you must call this or
	 * dumb_register_stdfiles() or set up your own before trying to load
	 * modules by file name. (If you are using another method, such as
	 * loading an Allegro datafile with modules embedded in it, then DUMB
	 * never opens a file by file name so this doesn't apply.)
	 */
	dumb_register_packfiles();

	/* Load the module file into a DUH object. Quick and dirty: try the
	 * loader for each format until one succeeds. Note that 15-sample
	 * mods have no identifying features, so dumb_load_mod() may succeed
	 * on files that aren't mods at all. We therefore try that one last.
	 */
	duh = dumb_load_it(argv[1]);
	if (!duh) {
		duh = dumb_load_xm(argv[1]);
		if (!duh) {
			duh = dumb_load_s3m(argv[1]);
			if (!duh) {
				duh = dumb_load_mod(argv[1]);
				if (!duh) {
					allegro_message("Failed to load %s!\n", argv[1]);
					return EXIT_FAILURE;
				}
			}
		}
	}

	/* Read the quality values from the config file we told Allegro to
	 * use. You may want to hardcode these or provide a more elaborate
	 * interface via which the user can control them.
	 */
	dumb_resampling_quality = get_config_int("sound", "dumb_resampling_quality", 4);
	dumb_it_max_to_mix = get_config_int("sound", "dumb_it_max_to_mix", 128);

	/* If we're not in DOS, show a window and register our close hook
	 * function.
	 */
#	ifndef ALLEGRO_DOS
		{
			const char *fn = get_filename(argv[1]);
			gfx_half_width = strlen(fn);
			if (gfx_half_width < 22) gfx_half_width = 22;
			gfx_half_width = (gfx_half_width + 2) * 4;

			/* set_window_title() is not const-correct (yet). */
			set_window_title((char *)"DUMB Music Player");

			if (set_gfx_mode(GFX_DUMB_MODE, gfx_half_width*2, 80, 0, 0) == 0) {
				acquire_screen();
				textout_centre(screen, font, fn, gfx_half_width, 20, 14);
				textout_centre(screen, font, "Press any key to exit.", gfx_half_width, 52, 11);
				release_screen();
			} else
				gfx_half_width = 0;
		}

		/* Silly check to get around the fact that someone stupidly removed
		 * an old function from Allegro instead of deprecating it. The old
		 * function was put back a version later, but we may as well use the
		 * new one if it's there!
		 */
#		if ALLEGRO_VERSION*10000 + ALLEGRO_SUB_VERSION*100 + ALLEGRO_WIP_VERSION >= 40105
			set_close_button_callback(&closehook);
#		else
			set_window_close_hook(&closehook);
#		endif

#	endif

	/* We want to continue running if the user switches to another
	 * application.
	 */
	set_display_switch_mode(SWITCH_BACKGROUND);

	/* We have the music loaded, but it isn't playing yet. This starts it
	 * playing. We construct a second object, the AL_DUH_PLAYER, to
	 * represent the playing music. This means you can play the music
	 * twice at the same time should you want to!
	 *
	 * Specify the number of channels (2 for stereo), which 'signal' to
	 * play (always 0 for modules), the volume (1.0f for default), the
	 * buffer size (4096 generally works well) and the sampling frequency
	 * (ideally match the final output frequency Allegro is using). An
	 * Allegro audio stream will be started.
	 */
	dp = al_start_duh(duh, 2, 0, 1.0f,
		get_config_int("sound", "buffer_size", 4096),
		get_config_int("sound", "sound_freq", 44100));

	/* Register our callback functions so that they are called when the
	 * music loops or stops. See docs/howto.txt for more information.
	 * There is no threading issue: DUMB will only process playback
	 * in al_poll_duh(), which we call below.
	 */
	{
		DUH_SIGRENDERER *sr = al_duh_get_sigrenderer(dp);
		DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer(sr);
		dumb_it_set_loop_callback(itsr, &loop_callback, NULL);
		dumb_it_set_xm_speed_zero_callback(itsr, &xm_speed_zero_callback, NULL);
	}

	/* Main loop. */
	for (;;) {
		/* Check for keys in the buffer. If we get one, discard it
		 * and exit the main loop.
		 */
		if (keypressed()) {
			readkey();
			break;
		}

		/* Poll the music. We exit the loop if al_poll_duh() has
		 * returned nonzero (music finished) or the window has been
		 * closed. al_poll_duh() might return nonzero if you have set
		 * up a callback that tells the music to stop.
		 */
		if (al_poll_duh(dp) || closed)
			break;

		/* Give other threads a look-in, or allow the processor to
		 * sleep for a bit. YIELD() is defined further up in this
		 * file.
		 */
		YIELD();
	}

	/* Remove the audio stream and deallocate the memory being used for
	 * the playback state. We set dp to NULL to emphasise that the object
	 * has gone.
	 */
	al_stop_duh(dp);
	dp = NULL;

	/* Free the DUH object containing the actual music data. */
	unload_duh(duh);
	duh = NULL;

	/* All done! */
	return EXIT_SUCCESS;
}
コード例 #5
0
ファイル: dumbplay.c プロジェクト: codefoco/dumb
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;
}
コード例 #6
0
ファイル: modplay.c プロジェクト: jangler/modplay
void init_sigrenderer(DUH_SIGRENDERER *sr) {
	DUMB_IT_SIGRENDERER *itsr = duh_get_it_sigrenderer(sr);
	dumb_it_set_loop_callback(itsr, &loop_callback, NULL);
	dumb_it_set_xm_speed_zero_callback(itsr, &dumb_it_callback_terminate,
			NULL);
}