ocarina/core/audio.c

/*
 * Copyright 2013 (c) Anna Schumaker.
 */
#include <core/audio.h>
#include <core/idle.h>
#include <core/playlist.h>
#include <core/settings.h>

#define LOAD_PLAYING (1 << 0) /* Begin playback after loading */
#define LOAD_HISTORY (1 << 1) /* Add the track to the history */
#define LOAD_DEFAULT (LOAD_PLAYING | LOAD_HISTORY)

static const char *SETTINGS_TRACK  = "core.audio.cur";
static const char *SETTINGS_VOLUME = "core.audio.volume";

static struct file   audio_file        = FILE_INIT_DATA("", "cur_track", 0);
static struct track *audio_track       = NULL;
static int           audio_pause_count = -1;

static GstElement *audio_pipeline  = NULL;
static GstElement *audio_source    = NULL;
static GstElement *audio_decoder   = NULL;
static GstElement *audio_converter = NULL;
static GstElement *audio_volume    = NULL;
static GstElement *audio_sink      = NULL;
static guint       audio_bus_id    = 0;

static struct audio_callbacks *audio_cb = NULL;


static bool __audio_change_state(GstState state)
{
	if (audio_cur_state() == state)
		return false;
	return gst_element_set_state(audio_pipeline, state) != GST_STATE_CHANGE_FAILURE;
}

static struct track *__audio_load(struct track *track, unsigned int flags)
{
	struct track *prev = audio_track;
	gchar *path;

	if (!track)
		return NULL;

	audio_track = track;
	path  = track_path(track);
	if (audio_cur_state() != GST_STATE_NULL)
		gst_element_set_state(audio_pipeline, GST_STATE_READY);
	g_object_set(G_OBJECT(audio_source), "location", path, NULL);
	gst_element_set_state(audio_pipeline, flags & LOAD_PLAYING ?
					GST_STATE_PLAYING : GST_STATE_PAUSED);

	playlist_played(prev);
	if (prev && TRACK_IS_EXTERNAL(prev))
		track_free_external(prev);

	playlist_selected(track);
	if (flags & LOAD_HISTORY && !TRACK_IS_EXTERNAL(track))
		playlist_add(playlist_lookup(PL_SYSTEM, "History"), track);
	if (audio_cb)
		audio_cb->audio_cb_load(track);

	audio_save();
	g_free(path);
	return track;
}

static void __audio_pad_added(GstElement *element, GstPad *pad, gpointer data)
{
	GstPad *sink = gst_element_get_static_pad(audio_decoder, "sink");

	gst_element_link(element, audio_converter);
	gst_pad_link(pad, sink);
	gst_object_unref(sink);
}

static gboolean __audio_message(GstBus *bus, GstMessage *message, gpointer data)
{
	GstObject *source = GST_OBJECT(GST_MESSAGE_SRC(message));
	gchar     *debug  = NULL;
	GError    *error  = NULL;
	GstState old, state, next;
	unsigned int load_flags = LOAD_DEFAULT;

	switch (GST_MESSAGE_TYPE(message)) {
	case GST_MESSAGE_ERROR:
		gst_message_parse_error(message, &error, &debug);
		g_printerr("ERROR from element %s: %s\n",
			    GST_OBJECT_NAME(source), error->message);
		g_printerr("DEBUG details: %s\n", debug ? debug : "none");
		g_error_free(error);
		g_free(debug);

		if (audio_cur_state() != GST_STATE_PLAYING)
			load_flags = LOAD_HISTORY;
		__audio_load(playlist_next(), load_flags);
		break;

	case GST_MESSAGE_EOS:
		track_played(audio_track);
		if (audio_pause_count >= 0) {
			audio_pause_after(audio_pause_count - 1);
			if (audio_pause_count == -1)
				load_flags = LOAD_HISTORY;
		}
		__audio_load(playlist_next(), load_flags);
		break;

	case GST_MESSAGE_STATE_CHANGED:
		if (!audio_cb || source != GST_OBJECT(audio_pipeline))
			break;

		gst_message_parse_state_changed(message, &old, &state, &next);
		if (state == GST_STATE_PLAYING || state == GST_STATE_PAUSED) {
			if (next == GST_STATE_VOID_PENDING)
				audio_cb->audio_cb_state_change(state);
		}

	default:
		break;
	}

	return true;
}

static bool __audio_init_idle(void *data)
{
	unsigned int track;

	if (settings_has(SETTINGS_TRACK)) {
		track = settings_get(SETTINGS_TRACK);
		__audio_load(track_get(track), LOAD_HISTORY);
	} else if (file_open(&audio_file, OPEN_READ)) {
		track = file_readu(&audio_file);
		file_close(&audio_file);
		file_remove(&audio_file);
		__audio_load(track_get(track), LOAD_HISTORY);
	}

	return true;
}


void audio_init(int *argc, char ***argv, struct audio_callbacks *callbacks)
{
	unsigned int volume = 100;
	GstBus *bus;

	gst_init(argc, argv);
	audio_cb        = callbacks;
	audio_pipeline  = gst_pipeline_new("pipeline");
	audio_source    = gst_element_factory_make("filesrc",       "source");
	audio_decoder   = gst_element_factory_make("decodebin",     "decoder");
	audio_converter = gst_element_factory_make("audioconvert",  "converter");
	audio_volume    = gst_element_factory_make("volume",        "volume");
	audio_sink      = gst_element_factory_make("autoaudiosink", "sink");
	bus             = gst_pipeline_get_bus(GST_PIPELINE(audio_pipeline));
	audio_bus_id    = gst_bus_add_watch(bus, __audio_message, NULL);

	gst_bin_add_many(GST_BIN(audio_pipeline), audio_source,    audio_decoder,
						  audio_converter, audio_volume,
						  audio_sink, NULL);
	gst_element_link(audio_source, audio_decoder);
	gst_element_link_many(audio_converter, audio_volume, audio_sink, NULL);
	g_signal_connect(audio_decoder, "pad-added", G_CALLBACK(__audio_pad_added), NULL);
	gst_object_unref(bus);

	if (settings_has(SETTINGS_VOLUME))
		volume = settings_get(SETTINGS_VOLUME);
	audio_set_volume(volume);

	idle_schedule(IDLE_SYNC, __audio_init_idle, NULL);
}

void audio_deinit()
{
	gst_element_set_state(audio_pipeline, GST_STATE_NULL);
	gst_object_unref(GST_ELEMENT(audio_pipeline));
	g_source_remove(audio_bus_id);

	audio_pipeline  = NULL;
	audio_source    = NULL;
	audio_decoder   = NULL;
	audio_converter = NULL;
	audio_volume    = NULL;
	audio_sink      = NULL;
	audio_track     = NULL;

	gst_deinit();
}

void audio_save()
{
	if (audio_track && !TRACK_IS_EXTERNAL(audio_track))
		settings_set(SETTINGS_TRACK, track_index(audio_track));
}

bool audio_load(struct track *track)
{
	if (track == audio_track)
		return false;
	return __audio_load(track, LOAD_DEFAULT) != NULL;
}

bool audio_load_filepath(const gchar *filepath)
{
	struct track *track;

	if (!filepath)
		return false;
	track = track_lookup(filepath);
	if (!track)
		track = track_alloc_external(filepath);
	return audio_load(track);
}

struct track *audio_cur_track()
{
	return audio_track;
}

GstState audio_cur_state()
{
	GstState cur = GST_STATE_NULL;
	if (audio_pipeline)
		gst_element_get_state(audio_pipeline,
				      &cur, NULL,
				      GST_CLOCK_TIME_NONE);
	return cur;
}

void audio_set_volume(unsigned int volume)
{
	gdouble vol;

	if (volume > 100)
		volume = 100;
	vol = (gdouble)volume / 100;

	settings_set(SETTINGS_VOLUME, volume);
	g_object_set(G_OBJECT(audio_volume), "volume", vol, NULL);
}

unsigned int audio_get_volume()
{
	gdouble volume;
	g_object_get(G_OBJECT(audio_volume), "volume", &volume, NULL);
	return volume * 100;
}

bool audio_play()
{
	if (!audio_track)
		return false;
	return __audio_change_state(GST_STATE_PLAYING);
}

bool audio_pause()
{
	if (!audio_track)
		return false;
	return __audio_change_state(GST_STATE_PAUSED);
}

bool audio_seek(gint64 offset)
{
	if (!audio_track)
		return false;
	return gst_element_seek_simple(audio_pipeline,
				       GST_FORMAT_TIME,
				       GST_SEEK_FLAG_FLUSH,
				       offset);
}

gint64 audio_position()
{
	gint64 position;
	if (gst_element_query_position(audio_pipeline,
				       GST_FORMAT_TIME,
				       &position))
		return position;
	return 0;
}

gint64 audio_duration()
{
	gint64 duration;
	if (gst_element_query_duration(audio_pipeline,
				       GST_FORMAT_TIME,
				       &duration))
		return duration;
	if (audio_track)
		return audio_track->tr_length * GST_SECOND;
	return 0;
}

struct track *audio_next()
{
	return __audio_load(playlist_next(), LOAD_DEFAULT);
}

struct track *audio_prev()
{
	return __audio_load(playlist_prev(), LOAD_PLAYING);
}

bool audio_pause_after(int n)
{
	if (n >= -1 && n != audio_pause_count) {
		audio_pause_count = n;
		if (audio_cb)
			audio_cb->audio_cb_config_pause(audio_pause_count);
		return true;
	}
	return false;
}

int audio_get_pause_count(void)
{
	return audio_pause_count;
}

#ifdef CONFIG_TESTING
void test_audio_eos()
{
	GstMessage *message = gst_message_new_eos(GST_OBJECT(audio_pipeline));
	__audio_message(NULL, message, NULL);
	gst_message_unref(message);
}

void test_audio_error(GError *error, gchar *debug)
{
	GstMessage *message = gst_message_new_error(
				GST_OBJECT(audio_pipeline), error, debug);
	__audio_message(NULL, message, NULL);
	gst_message_unref(message);
}

GstElement *test_audio_pipeline()
{
	return audio_pipeline;
}
#endif /* CONFIG_TESTING */