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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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0837e1e495
Original commit message from CVS: Modified a lot of plugins to use the caps system. Modified the caps of audio/raw to our agreed properties. Added the multidisksrc plugin of Dominic Ludlam Renamed audiosink/src to osssink/src and updated all the examples using the old name. Moved oss specific plugins in an oss directory. removed the old audiosink from the elements/ dir. removed audioraw.h metadata header files since we now use the properties. There are still a few plugins that won't build because they include the old audioraw.h header file. This will be fixed soon. Make sure the caps are set in the plugins as described by their padtemplates (this should solve problems with gstmediaplay with various media files). *please don't panic when some plugins won't build, just cd manually into the plugin dirs* This will be fixed soon.
116 lines
3.1 KiB
C
116 lines
3.1 KiB
C
#include <glib.h>
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#include <gst/gst.h>
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#include <ghttp.h>
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void eof(GstSrc *src) {
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g_print("have eof, quitting\n");
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exit(0);
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}
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int main(int argc,char *argv[]) {
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guint16 mp3type;
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GList *factories;
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GstElementFactory *parsefactory;
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GstElement *bin, *src, *parse, *osssink;
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GList *padlist;
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ghttp_request *pls;
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guchar *plsbuf;
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gint plsbuflen,parsedlen = 0;
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guchar *url,*local;
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GSList *urls = NULL;
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pls = ghttp_request_new();
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if (argc >= 2)
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ghttp_set_uri(pls,argv[1]);
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else
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ghttp_set_uri(pls,"http://209.127.18.4:9000");
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ghttp_prepare(pls);
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ghttp_process(pls);
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plsbuf = ghttp_get_body(pls);
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plsbuflen = ghttp_get_body_len(pls);
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while (parsedlen < plsbuflen) {
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local = plsbuf + parsedlen;
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if ((*local != '[') && (*local != '\n')) { /* t/v pair */
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if (!strncmp(local,"File1=",4)) { /* if file */
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url = strchr(local,'=') + 1; /* url after = */
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local = strchr(url,'\n'); /* ffwd after = */
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*(local)++ = 0; /* nullz url */
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g_print("prepending '%s' to list\n",url);
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urls = g_slist_prepend(urls,g_strdup(url));
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/* local should point to next line now */
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} else {
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local = strchr(local,'\n') + 1; /* skip line */
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}
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} else {
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local = strchr(local,'\n') + 1; /* skip line */
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}
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/* we can consider that line parsed... */
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parsedlen = local - plsbuf;
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}
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if (urls == NULL) {
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g_print("couldn't find any streams\n");
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exit(1);
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}
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ghttp_request_destroy(pls);
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gst_init(&argc,&argv);
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gst_plugin_load_all();
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bin = gst_bin_new("bin");
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src = gst_httpsrc_new("src");
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if (argc == 3) {
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int i;
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for (i=1;i<atoi(argv[2]);i++)
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urls = g_slist_next(urls);
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}
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g_print("loading shoutcast server %s\n",urls->data);
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gtk_object_set(GTK_OBJECT(src),"location",urls->data,NULL);
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g_print("created src\n");
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/* now it's time to get the parser */
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mp3type = gst_type_find_by_mime("audio/mpeg");
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factories = gst_type_get_sinks(mp3type);
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if (factories != NULL)
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parsefactory = GST_ELEMENTFACTORY(factories->data);
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else {
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g_print("sorry, can't find anyone registered to sink 'mp3'\n");
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return 1;
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}
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parse = gst_elementfactory_create(parsefactory,"parser");
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if (parse == NULL) {
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g_print("sorry, couldn't create parser\n");
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return 1;
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}
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osssink = gst_osssink_new("osssink");
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gtk_signal_connect(GTK_OBJECT(src),"eof",
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GTK_SIGNAL_FUNC(eof),NULL);
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/* add objects to the main pipeline */
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gst_bin_add(GST_BIN(bin),GST_OBJECT(src));
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gst_bin_add(GST_BIN(bin),GST_OBJECT(parse));
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gst_bin_add(GST_BIN(bin),GST_OBJECT(osssink));
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/* connect src to sink */
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gst_pad_connect(gst_element_get_pad(src,"src"),
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gst_element_get_pad(parse,"sink"));
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gst_pad_connect(gst_element_get_pad(parse,"src"),
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gst_element_get_pad(osssink,"sink"));
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sleep(5); /* to let the network buffer fill a bit */
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while(1) {
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g_print("calling gst_httpsrc_push\n");
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gst_httpsrc_push(GST_SRC(src));
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}
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gst_object_destroy(GST_OBJECT(osssink));
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gst_object_destroy(GST_OBJECT(parse));
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gst_object_destroy(GST_OBJECT(src));
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gst_object_destroy(GST_OBJECT(bin));
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}
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