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core/audio: Build up the rest of the pipeline 

Signed-off-by: Anna Schumaker <Anna@NoWheyCreamery.com>
This commit is contained in:
Anna Schumaker 2017-04-05 09:23:55 -04:00
parent 9c6a9f7759
commit 76e400e156
1 changed files with 38 additions and 15 deletions
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53
core/audio.c
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@ -16,10 +16,13 @@ static int audio_pause_count = -1;
static GstElement *audio_player = NULL;
static guint audio_old_id = 0;
static GstElement *audio_pipeline = NULL;
static GstElement *audio_source = NULL;
static GstElement *audio_decoder = NULL;
static guint audio_bus_id = 0;
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;
@ -47,6 +50,7 @@ static void __audio_gst_load(struct track *track, GstState state)
gst_element_set_state(audio_player, GST_STATE_NULL);
g_object_set(G_OBJECT(audio_player), "uri", uri, NULL);
gst_element_set_state(audio_pipeline, GST_STATE_READY);
g_object_set(G_OBJECT(audio_source), "location", path, NULL);
__audio_change_state(state);
@ -87,6 +91,15 @@ static struct track *__audio_next(GstState state)
return __audio_load(playlist_next(), state);
}
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)
{
switch (GST_MESSAGE_TYPE(message)) {
@ -133,14 +146,21 @@ void audio_init(int *argc, char ***argv, struct audio_callbacks *callbacks)
audio_old_id = gst_bus_add_watch(bus, __audio_message, NULL);
gst_object_unref(bus);
audio_pipeline = gst_pipeline_new("pipeline");
audio_source = gst_element_factory_make("filesrc", "source");
audio_decoder = gst_element_factory_make("decodebin", "decoder");
bus = gst_pipeline_get_bus(GST_PIPELINE(audio_pipeline));
audio_bus_id = gst_bus_add_watch(bus, __audio_message, NULL);
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, 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))
@ -160,11 +180,14 @@ void audio_deinit()
gst_object_unref(GST_ELEMENT(audio_pipeline));
g_source_remove(audio_bus_id);
audio_player = NULL;
audio_pipeline = NULL;
audio_source = NULL;
audio_decoder = NULL;
audio_track = NULL;
audio_player = NULL;
audio_pipeline = NULL;
audio_source = NULL;
audio_decoder = NULL;
audio_converter = NULL;
audio_volume = NULL;
audio_sink = NULL;
audio_track = NULL;
gst_deinit();
}