#include "schannel.h"
#include "chimara-glk-private.h"
#include "gi_dispa.h"
+#include "gi_blorb.h"
+#include "resource.h"
extern GPrivate *glk_data_key;
+#ifdef GSTREAMER_SOUND
+/* Stop any currently playing sound on this channel, and remove any
+ format-specific GStreamer elements from the channel. */
+static void
+clean_up_after_playing_sound(schanid_t chan)
+{
+ if(!gst_element_set_state(chan->pipeline, GST_STATE_NULL))
+ WARNING_S(_("Could not set GstElement state to"), "NULL");
+ if(chan->demux)
+ {
+ gst_bin_remove(GST_BIN(chan->pipeline), chan->demux);
+ chan->demux = NULL;
+ }
+ if(chan->decode)
+ {
+ gst_bin_remove(GST_BIN(chan->pipeline), chan->decode);
+ chan->decode = NULL;
+ }
+}
+
+/* This signal is thrown whenever the GStreamer pipeline generates a message.
+ Most messages are harmless. */
+static void
+on_pipeline_message(GstBus *bus, GstMessage *message, schanid_t s)
+{
+ /* g_printerr("Got %s message\n", GST_MESSAGE_TYPE_NAME(message)); */
+
+ GError *err;
+ gchar *debug_message;
+
+ switch(GST_MESSAGE_TYPE(message)) {
+ case GST_MESSAGE_ERROR:
+ {
+ gst_message_parse_error(message, &err, &debug_message);
+ IO_WARNING(_("GStreamer error"), err->message, debug_message);
+ g_error_free(err);
+ g_free(debug_message);
+ clean_up_after_playing_sound(s);
+ }
+ break;
+ case GST_MESSAGE_WARNING:
+ {
+ gst_message_parse_warning(message, &err, &debug_message);
+ IO_WARNING(_("GStreamer warning"), err->message, debug_message);
+ g_error_free(err);
+ g_free(debug_message);
+ }
+ break;
+ case GST_MESSAGE_INFO:
+ {
+ gst_message_parse_info(message, &err, &debug_message);
+ g_message("GStreamer info \"%s\": %s", err->message, debug_message);
+ g_error_free(err);
+ g_free(debug_message);
+ }
+ break;
+ case GST_MESSAGE_EOS:
+ /* end-of-stream */
+ clean_up_after_playing_sound(s);
+ break;
+ default:
+ /* unhandled message */
+ break;
+ }
+}
+
+/* This signal is thrown when the OGG demuxer element has decided what kind of
+ outputs it will output. We connect the decoder element dynamically. */
+static void
+on_ogg_demuxer_pad_added(GstElement *demux, GstPad *pad, schanid_t s)
+{
+ GstPad *sinkpad;
+
+ /* We can now link this pad with the vorbis-decoder sink pad */
+ sinkpad = gst_element_get_static_pad(s->decode, "sink");
+ if(gst_pad_link(pad, sinkpad) != GST_PAD_LINK_OK)
+ WARNING(_("Could not link OGG demuxer with Vorbis decoder"));
+ gst_object_unref(sinkpad);
+}
+
+/* This signal is thrown when the typefinder element has found the type of its
+ input. Now that we know what kind of input stream we have, we can connect the
+ proper demuxer/decoder elements. */
+static void
+on_type_found(GstElement *typefind, guint probability, GstCaps *caps, schanid_t s)
+{
+ gchar *type = gst_caps_to_string(caps);
+ if(strcmp(type, "application/ogg") == 0)
+ {
+ s->demux = gst_element_factory_make("oggdemux", NULL);
+ s->decode = gst_element_factory_make("vorbisdec", NULL);
+ if(!s->demux || !s->decode)
+ {
+ WARNING(_("Could not create one or more GStreamer elements"));
+ goto finally;
+ }
+ gst_bin_add_many(GST_BIN(s->pipeline), s->demux, s->decode, NULL);
+ if(!gst_element_link(s->typefind, s->demux) || !gst_element_link(s->decode, s->convert))
+ {
+ WARNING(_("Could not link GStreamer elements"));
+ goto finally;
+ }
+ /* We link the demuxer and decoder together dynamically, since the
+ demuxer doesn't know what source pads it will have until it starts
+ demuxing the stream */
+ g_signal_connect(s->demux, "pad-added", G_CALLBACK(on_ogg_demuxer_pad_added), s);
+ }
+ else if(strcmp(type, "audio/x-aiff") == 0)
+ {
+ s->decode = gst_element_factory_make("aiffparse", NULL);
+ if(!s->decode)
+ {
+ WARNING(_("Could not create 'aiffparse' GStreamer element"));
+ goto finally;
+ }
+ gst_bin_add(GST_BIN(s->pipeline), s->decode);
+ if(!gst_element_link_many(s->typefind, s->decode, s->convert, NULL))
+ {
+ WARNING(_("Could not link GStreamer elements"));
+ goto finally;
+ }
+ }
+ else
+ {
+ WARNING_S(_("Unexpected audio type in blorb"), type);
+ }
+
+finally:
+ g_free(type);
+}
+#endif /* GSTREAMER_SOUND */
+
/**
* glk_schannel_create:
* @rock: The rock value to give the new sound channel.
s->pipeline = gst_pipeline_new(pipeline_name);
g_free(pipeline_name);
+ /* Watch for messages from the pipeline */
+ GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(s->pipeline));
+ gst_bus_add_signal_watch(bus);
+ g_signal_connect(bus, "message", G_CALLBACK(on_pipeline_message), s);
+ gst_object_unref(bus);
+
/* Create GStreamer elements to put in the pipeline */
- s->source = gst_element_factory_make("audiotestsrc", NULL);
+ s->source = gst_element_factory_make("giostreamsrc", NULL);
+ s->typefind = gst_element_factory_make("typefind", NULL);
+ s->convert = gst_element_factory_make("audioconvert", NULL);
s->filter = gst_element_factory_make("volume", NULL);
s->sink = gst_element_factory_make("autoaudiosink", NULL);
- if(!s->source || !s->filter || !s->sink) {
- WARNING("Could not create one or more GStreamer elements");
+ if(!s->source || !s->typefind || !s->convert || !s->filter || !s->sink) {
+ WARNING(_("Could not create one or more GStreamer elements"));
goto fail;
}
-
- gst_bin_add_many(GST_BIN(s->pipeline), s->source, s->filter, s->sink, NULL);
- if(!gst_element_link_many(s->source, s->filter, s->sink, NULL)) {
- WARNING("Could not link GStreamer elements");
+
+ /* Put the elements in the pipeline and link as many together as we can
+ without knowing the type of the audio stream */
+ gst_bin_add_many(GST_BIN(s->pipeline), s->source, s->typefind, s->convert, s->filter, s->sink, NULL);
+ /* Link elements: Source -> typefinder -> ??? -> Converter -> Volume filter -> Sink */
+ if(!gst_element_link(s->source, s->typefind) || !gst_element_link_many(s->convert, s->filter, s->sink, NULL)) {
+ WARNING(_("Could not link GStreamer elements"));
goto fail;
}
+ g_signal_connect(s->typefind, "have-type", G_CALLBACK(on_type_found), s);
return s;
(*glk_data->unregister_obj)(chan, gidisp_Class_Schannel, chan->disprock);
chan->disprock.ptr = NULL;
}
-
+
+ /* This also frees all the objects inside the pipeline */
if(chan->pipeline)
gst_object_unref(chan->pipeline);
{
VALID_SCHANNEL(chan, return 0);
#ifdef GSTREAMER_SOUND
+ ChimaraGlkPrivate *glk_data = g_private_get(glk_data_key);
+
+ if(!glk_data->resource_map) {
+ if(!glk_data->resource_load_callback) {
+ WARNING(_("No resource map has been loaded yet."));
+ return 0;
+ }
+ WARNING(_("Loading sound resources from alternative location not yet supported."));
+ return 0;
+ }
+
+ giblorb_result_t resource;
+ giblorb_err_t result = giblorb_load_resource(glk_data->resource_map, giblorb_method_Memory, &resource, giblorb_ID_Snd, snd);
+ if(result != giblorb_err_None) {
+ WARNING_S( _("Error loading resource"), giblorb_get_error_message(result) );
+ return 0;
+ }
+ GInputStream *stream = g_memory_input_stream_new_from_data(resource.data.ptr, resource.length, NULL);
+ g_object_set(chan->source, "stream", stream, NULL);
+
if(!gst_element_set_state(chan->pipeline, GST_STATE_PLAYING)) {
WARNING_S(_("Could not set GstElement state to"), "PLAYING");
return 0;
{
VALID_SCHANNEL(chan, return);
#ifdef GSTREAMER_SOUND
- if(!gst_element_set_state(chan->pipeline, GST_STATE_READY))
- WARNING_S(_("Could not set GstElement state to"), "READY");
+ clean_up_after_playing_sound(chan);
#endif
}
projects / projects / chimara / chimara.git / blobdiff