예제 #1
0
bool
audio_format_parse(struct audio_format *dest, const char *src,
		   bool mask, GError **error_r)
{
	uint32_t rate;
	enum sample_format sample_format;
	uint8_t channels;

	audio_format_clear(dest);

	/* parse sample rate */

#if GCC_CHECK_VERSION(4,7)
	/* workaround -Wmaybe-uninitialized false positive */
	rate = 0;
#endif

	if (!parse_sample_rate(src, mask, &rate, &src, error_r))
		return false;

	if (*src++ != ':') {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Sample format missing");
		return false;
	}

	/* parse sample format */

#if GCC_CHECK_VERSION(4,7)
	/* workaround -Wmaybe-uninitialized false positive */
	sample_format = SAMPLE_FORMAT_UNDEFINED;
#endif

	if (!parse_sample_format(src, mask, &sample_format, &src, error_r))
		return false;

	if (*src++ != ':') {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Channel count missing");
		return false;
	}

	/* parse channel count */

	if (!parse_channel_count(src, mask, &channels, &src, error_r))
		return false;

	if (*src != 0) {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Extra data after channel count: %s", src);
		return false;
	}

	audio_format_init(dest, rate, sample_format, channels);
	assert(mask ? audio_format_mask_valid(dest)
	       : audio_format_valid(dest));

	return true;
}
예제 #2
0
/**
 * After the decoder has been started asynchronously, wait for the
 * "START" command to finish.  The decoder may not be initialized yet,
 * i.e. there is no audio_format information yet.
 *
 * The player lock is not held.
 */
static bool
player_wait_for_decoder(struct player *player)
{
	struct player_control *pc = player->pc;
	struct decoder_control *dc = player->dc;

	assert(player->queued || pc->command == PLAYER_COMMAND_SEEK);
	assert(pc->next_song != NULL);

	player->queued = false;

	GError *error = dc_lock_get_error(dc);
	if (error != NULL) {
		player_lock(pc);
		pc_set_error(pc, PLAYER_ERROR_DECODER, error);

		song_free(pc->next_song);
		pc->next_song = NULL;

		player_unlock(pc);

		return false;
	}

	if (player->song != NULL)
		song_free(player->song);

	player->song = pc->next_song;
	player->elapsed_time = 0.0;

	/* set the "starting" flag, which will be cleared by
	   player_check_decoder_startup() */
	player->decoder_starting = true;

	player_lock(pc);

	/* update player_control's song information */
	pc->total_time = song_get_duration(pc->next_song);
	pc->bit_rate = 0;
	audio_format_clear(&pc->audio_format);

	/* clear the queued song */
	pc->next_song = NULL;

	player_unlock(pc);

	/* call syncPlaylistWithQueue() in the main thread */
	event_pipe_emit(PIPE_EVENT_PLAYLIST);

	return true;
}
예제 #3
0
bool
audio_format_parse(struct audio_format *dest, const char *src,
		   bool mask, GError **error_r)
{
	uint32_t rate;
	enum sample_format sample_format;
	uint8_t channels;

	audio_format_clear(dest);

	/* parse sample rate */

	if (!parse_sample_rate(src, mask, &rate, &src, error_r))
		return false;

	if (*src++ != ':') {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Sample format missing");
		return false;
	}

	/* parse sample format */

	if (!parse_sample_format(src, mask, &sample_format, &src, error_r))
		return false;

	if (*src++ != ':') {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Channel count missing");
		return false;
	}

	/* parse channel count */

	if (!parse_channel_count(src, mask, &channels, &src, error_r))
		return false;

	if (*src != 0) {
		g_set_error(error_r, audio_parser_quark(), 0,
			    "Extra data after channel count: %s", src);
		return false;
	}

	audio_format_init(dest, rate, sample_format, channels);
	assert(mask ? audio_format_mask_valid(dest)
	       : audio_format_valid(dest));

	return true;
}
예제 #4
0
/**
 * After the decoder has been started asynchronously, wait for the
 * "START" command to finish.  The decoder may not be initialized yet,
 * i.e. there is no audio_format information yet.
 *
 * The player lock is not held.
 */
static bool
player_wait_for_decoder(struct player *player)
{
	struct decoder_control *dc = player->dc;

	assert(player->queued || pc.command == PLAYER_COMMAND_SEEK);
	assert(pc.next_song != NULL);

	player->queued = false;

	if (decoder_lock_has_failed(dc)) {
		player_lock();
		pc.errored_song = dc->song;
		pc.error = PLAYER_ERROR_FILE;
		pc.next_song = NULL;
		player_unlock();

		return false;
	}

	player->song = pc.next_song;
	player->elapsed_time = 0.0;

	/* set the "starting" flag, which will be cleared by
	   player_check_decoder_startup() */
	player->decoder_starting = true;

	player_lock();

	/* update player_control's song information */
	pc.total_time = song_get_duration(pc.next_song);
	pc.bit_rate = 0;
	audio_format_clear(&pc.audio_format);

	/* clear the queued song */
	pc.next_song = NULL;

	player_unlock();

	/* call syncPlaylistWithQueue() in the main thread */
	event_pipe_emit(PIPE_EVENT_PLAYLIST);

	return true;
}
예제 #5
0
파일: output_init.c 프로젝트: ion1/mpd
bool
audio_output_init(struct audio_output *ao, const struct config_param *param,
		  GError **error_r)
{
	const struct audio_output_plugin *plugin = NULL;
	GError *error = NULL;

	if (param) {
		const char *p;

		p = config_get_block_string(param, AUDIO_OUTPUT_TYPE, NULL);
		if (p == NULL) {
			g_set_error(error_r, audio_output_quark(), 0,
				    "Missing \"type\" configuration");
			return false;
		}

		plugin = audio_output_plugin_get(p);
		if (plugin == NULL) {
			g_set_error(error_r, audio_output_quark(), 0,
				    "No such audio output plugin: %s", p);
			return false;
		}

		ao->name = config_get_block_string(param, AUDIO_OUTPUT_NAME,
						   NULL);
		if (ao->name == NULL) {
			g_set_error(error_r, audio_output_quark(), 0,
				    "Missing \"name\" configuration");
			return false;
		}

		p = config_get_block_string(param, AUDIO_OUTPUT_FORMAT,
						 NULL);
		if (p != NULL) {
			bool success =
				audio_format_parse(&ao->config_audio_format,
						   p, true, error_r);
			if (!success)
				return false;
		} else
			audio_format_clear(&ao->config_audio_format);
	} else {
		g_warning("No \"%s\" defined in config file\n",
			  CONF_AUDIO_OUTPUT);

		plugin = audio_output_detect(error_r);
		if (plugin == NULL)
			return false;

		g_message("Successfully detected a %s audio device",
			  plugin->name);

		ao->name = "default detected output";

		audio_format_clear(&ao->config_audio_format);
	}

	ao->plugin = plugin;
	ao->always_on = config_get_block_bool(param, "always_on", false);
	ao->enabled = config_get_block_bool(param, "enabled", true);
	ao->really_enabled = false;
	ao->open = false;
	ao->pause = false;
	ao->fail_timer = NULL;

	pcm_buffer_init(&ao->cross_fade_buffer);

	/* set up the filter chain */

	ao->filter = filter_chain_new();
	assert(ao->filter != NULL);

	/* create the replay_gain filter */

	const char *replay_gain_handler =
		config_get_block_string(param, "replay_gain_handler",
					"software");

	if (strcmp(replay_gain_handler, "none") != 0) {
		ao->replay_gain_filter = filter_new(&replay_gain_filter_plugin,
						    param, NULL);
		assert(ao->replay_gain_filter != NULL);

		ao->replay_gain_serial = 0;

		ao->other_replay_gain_filter = filter_new(&replay_gain_filter_plugin,
							  param, NULL);
		assert(ao->other_replay_gain_filter != NULL);

		ao->other_replay_gain_serial = 0;
	} else {
		ao->replay_gain_filter = NULL;
		ao->other_replay_gain_filter = NULL;
	}

	/* create the normalization filter (if configured) */

	if (config_get_bool(CONF_VOLUME_NORMALIZATION, false)) {
		struct filter *normalize_filter =
			filter_new(&normalize_filter_plugin, NULL, NULL);
		assert(normalize_filter != NULL);

		filter_chain_append(ao->filter,
				    autoconvert_filter_new(normalize_filter));
	}

	filter_chain_parse(ao->filter,
	                   config_get_block_string(param, AUDIO_FILTERS, ""),
	                   &error
	);

	// It's not really fatal - Part of the filter chain has been set up already
	// and even an empty one will work (if only with unexpected behaviour)
	if (error != NULL) {
		g_warning("Failed to initialize filter chain for '%s': %s",
			  ao->name, error->message);
		g_error_free(error);
	}

	ao->thread = NULL;
	ao->command = AO_COMMAND_NONE;
	ao->mutex = g_mutex_new();
	ao->cond = g_cond_new();

	ao->data = ao_plugin_init(plugin,
				  &ao->config_audio_format,
				  param, error_r);
	if (ao->data == NULL)
		return false;

	ao->mixer = audio_output_load_mixer(ao->data, param,
					    plugin->mixer_plugin,
					    ao->filter, &error);
	if (ao->mixer == NULL && error != NULL) {
		g_warning("Failed to initialize hardware mixer for '%s': %s",
			  ao->name, error->message);
		g_error_free(error);
	}

	/* use the hardware mixer for replay gain? */

	if (strcmp(replay_gain_handler, "mixer") == 0) {
		if (ao->mixer != NULL)
			replay_gain_filter_set_mixer(ao->replay_gain_filter,
						     ao->mixer, 100);
		else
			g_warning("No such mixer for output '%s'", ao->name);
	} else if (strcmp(replay_gain_handler, "software") != 0 &&
		   ao->replay_gain_filter != NULL) {
		g_set_error(error_r, audio_output_quark(), 0,
			    "Invalid \"replay_gain_handler\" value");
		return false;
	}

	/* the "convert" filter must be the last one in the chain */

	ao->convert_filter = filter_new(&convert_filter_plugin, NULL, NULL);
	assert(ao->convert_filter != NULL);

	filter_chain_append(ao->filter, ao->convert_filter);

	/* done */

	return true;
}