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mmssrc: port to 0.11

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This commit is contained in:
Wim Taymans 2011-10-04 18:54:13 +02:00
parent bd0b3f0761
commit d934d76870
2 changed files with 25 additions and 46 deletions
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2
configure.ac
View file

@ -305,7 +305,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
videomaxrate videomeasure videoparsers videosignal vmnc y4m \
decklink dvb fbdev linsys shm vcd \
voaacenc assrender apexsink bz2 camerabin2 cdaudio celt cog curl dc1394 dirac dts resindvd \
faac gsettings gsm jp2k ladspa libmms modplug mpeg2enc mplex mimic \
faac gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
musepack musicbrainz nas neon ofa rsvg schro sdl sndfile soundtouch timidity \
vp8 wildmidi xvid "
AC_SUBST(GST_PLUGINS_NONPORTED)

69
ext/libmms/gstmms.c
View file

@ -70,42 +70,16 @@ static GstFlowReturn gst_mms_create (GstPushSrc * psrc, GstBuffer ** buf);
static gboolean gst_mms_uri_set_uri (GstURIHandler * handler,
const gchar * uri);
static void
gst_mms_urihandler_init (GType mms_type)
{
static const GInterfaceInfo urihandler_info = {
gst_mms_uri_handler_init,
NULL,
NULL
};
g_type_add_interface_static (mms_type, GST_TYPE_URI_HANDLER,
&urihandler_info);
}
GST_BOILERPLATE_FULL (GstMMS, gst_mms, GstPushSrc, GST_TYPE_PUSH_SRC,
gst_mms_urihandler_init);
static void
gst_mms_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "MMS streaming source",
"Source/Network",
"Receive data streamed via MSFT Multi Media Server protocol",
"Maciej Katafiasz <mathrick@users.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (mmssrc_debug, "mmssrc", 0, "MMS Source Element");
}
#define gst_mms_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstMMS, gst_mms, GST_TYPE_PUSH_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_mms_uri_handler_init));
/* initialize the plugin's class */
static void
gst_mms_class_init (GstMMSClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstBaseSrcClass *gstbasesrc_class = (GstBaseSrcClass *) klass;
GstPushSrcClass *gstpushsrc_class = (GstPushSrcClass *) klass;
@ -125,6 +99,16 @@ gst_mms_class_init (GstMMSClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/* Note: connection-speed is intentionaly limited to G_MAXINT as libmms use int for it */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (gstelement_class,
"MMS streaming source", "Source/Network",
"Receive data streamed via MSFT Multi Media Server protocol",
"Maciej Katafiasz <mathrick@users.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (mmssrc_debug, "mmssrc", 0, "MMS Source Element");
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_mms_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_mms_stop);
@ -144,7 +128,7 @@ gst_mms_class_init (GstMMSClass * klass)
* initialize structure
*/
static void
gst_mms_init (GstMMS * mmssrc, GstMMSClass * g_class)
gst_mms_init (GstMMS * mmssrc)
{
mmssrc->uri_name = NULL;
mmssrc->current_connection_uri_name = NULL;
@ -264,7 +248,7 @@ gst_mms_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
of the segment. */
gst_segment_init (segment, seek_format);
gst_segment_set_seek (segment, rate, seek_format, flags, cur_type, cur,
gst_segment_do_seek (segment, rate, seek_format, flags, cur_type, cur,
stop_type, stop, NULL);
return TRUE;
@ -301,7 +285,7 @@ gst_mms_do_seek (GstBaseSrc * src, GstSegment * segment)
return FALSE;
}
gst_segment_init (segment, GST_FORMAT_BYTES);
gst_segment_set_seek (segment, segment->rate, GST_FORMAT_BYTES,
gst_segment_do_seek (segment, segment->rate, GST_FORMAT_BYTES,
segment->flags, GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE,
segment->stop, NULL);
return TRUE;
@ -339,35 +323,30 @@ gst_mms_create (GstPushSrc * psrc, GstBuffer ** buf)
else
blocksize = mmsx_get_asf_packet_len (mmssrc->connection);
*buf = gst_buffer_try_new_and_alloc (blocksize);
if (!*buf) {
data = g_try_malloc (blocksize);
if (!data) {
GST_ERROR_OBJECT (mmssrc, "Failed to allocate %u bytes", blocksize);
return GST_FLOW_ERROR;
}
data = GST_BUFFER_DATA (*buf);
GST_BUFFER_SIZE (*buf) = 0;
GST_LOG_OBJECT (mmssrc, "reading %d bytes", blocksize);
result = mmsx_read (NULL, mmssrc->connection, (char *) data, blocksize);
/* EOS? */
if (result == 0)
goto eos;
*buf = gst_buffer_new_wrapped (data, result);
GST_BUFFER_OFFSET (*buf) = offset;
GST_BUFFER_SIZE (*buf) = result;
GST_LOG_OBJECT (mmssrc, "Returning buffer with offset %" G_GINT64_FORMAT
" and size %u", GST_BUFFER_OFFSET (*buf), GST_BUFFER_SIZE (*buf));
gst_buffer_set_caps (*buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (mmssrc)));
" and size %u", offset, result);
return GST_FLOW_OK;
eos:
{
GST_DEBUG_OBJECT (mmssrc, "EOS");
gst_buffer_unref (*buf);
g_free (data);
*buf = NULL;
return GST_FLOW_UNEXPECTED;
}
@ -544,13 +523,13 @@ plugin_init (GstPlugin * plugin)
}
static GstURIType
gst_mms_uri_get_type (void)
gst_mms_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static gchar **
gst_mms_uri_get_protocols (void)
gst_mms_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { "mms", "mmsh", "mmst", "mmsu", NULL };