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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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rtmp: port to 0.11
This commit is contained in:
Wim Taymans
parent
c16cc4698a
commit
9ddfdfe60c
2 changed files with 78 additions and 92 deletions
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@ -75,63 +75,50 @@ static gboolean gst_rtmp_sink_stop (GstBaseSink * sink);
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static gboolean gst_rtmp_sink_start (GstBaseSink * sink);
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static GstFlowReturn gst_rtmp_sink_render (GstBaseSink * sink, GstBuffer * buf);
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static void
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_do_init (GType gtype)
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{
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static const GInterfaceInfo urihandler_info = {
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gst_rtmp_sink_uri_handler_init,
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NULL,
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NULL
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};
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g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info);
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GST_DEBUG_CATEGORY_INIT (gst_rtmp_sink_debug, "rtmpsink", 0,
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"RTMP server element");
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}
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GST_BOILERPLATE_FULL (GstRTMPSink, gst_rtmp_sink, GstBaseSink,
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GST_TYPE_BASE_SINK, _do_init);
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static void
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gst_rtmp_sink_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_set_details_simple (element_class,
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"RTMP output sink",
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"Sink/Network", "Sends FLV content to a server via RTMP",
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"Jan Schmidt <thaytan@noraisin.net>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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}
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#define gst_rtmp_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTMPSink, gst_rtmp_sink, GST_TYPE_BASE_SINK,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtmp_sink_uri_handler_init));
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/* initialize the plugin's class */
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static void
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gst_rtmp_sink_class_init (GstRTMPSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gobject_class->set_property = gst_rtmp_sink_set_property;
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gobject_class->get_property = gst_rtmp_sink_get_property;
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gst_element_class_install_std_props (gstelement_class,
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"location", PROP_LOCATION, G_PARAM_READWRITE, NULL);
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gst_element_class_set_details_simple (gstelement_class,
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"RTMP output sink",
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"Sink/Network", "Sends FLV content to a server via RTMP",
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"Jan Schmidt <thaytan@noraisin.net>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_sink_start);
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gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_sink_stop);
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gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_rtmp_sink_render);
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gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
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"location", PROP_LOCATION, G_PARAM_READWRITE, NULL);
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GST_DEBUG_CATEGORY_INIT (gst_rtmp_sink_debug, "rtmpsink", 0,
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"RTMP server element");
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}
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/* initialize the new element
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* initialize instance structure
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*/
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static void
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gst_rtmp_sink_init (GstRTMPSink * sink, GstRTMPSinkClass * klass)
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gst_rtmp_sink_init (GstRTMPSink * sink)
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{
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}
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@ -194,6 +181,8 @@ gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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{
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GstRTMPSink *sink = GST_RTMP_SINK (bsink);
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GstBuffer *reffed_buf = NULL;
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guint8 *data;
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gsize size;
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if (sink->first) {
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/* open the connection */
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@ -214,7 +203,7 @@ gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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/* FIXME: Parse the first buffer and see if it contains a header plus a packet instead
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* of just assuming it's only the header */
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GST_LOG_OBJECT (sink, "Caching first buffer of size %d for concatenation",
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GST_BUFFER_SIZE (buf));
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gst_buffer_get_size (buf));
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gst_buffer_replace (&sink->cache, buf);
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sink->first = FALSE;
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return GST_FLOW_OK;
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@ -222,40 +211,48 @@ gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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if (sink->cache) {
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GST_LOG_OBJECT (sink, "Joining 2nd buffer of size %d to cached buf",
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GST_BUFFER_SIZE (buf));
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gst_buffer_get_size (buf));
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gst_buffer_ref (buf);
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reffed_buf = buf = gst_buffer_join (sink->cache, buf);
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sink->cache = NULL;
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}
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GST_LOG_OBJECT (sink, "Sending %d bytes to RTMP server",
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GST_BUFFER_SIZE (buf));
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gst_buffer_get_size (buf));
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if (!RTMP_Write (sink->rtmp,
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(char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf))) {
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GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
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if (reffed_buf)
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gst_buffer_unref (reffed_buf);
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return GST_FLOW_ERROR;
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}
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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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if (!RTMP_Write (sink->rtmp, (char *) data, size))
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goto write_failed;
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gst_buffer_unmap (buf, data, size);
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if (reffed_buf)
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gst_buffer_unref (reffed_buf);
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return GST_FLOW_OK;
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/* ERRORS */
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write_failed:
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{
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GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
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gst_buffer_unmap (buf, data, size);
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if (reffed_buf)
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gst_buffer_unref (reffed_buf);
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return GST_FLOW_ERROR;
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}
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}
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/*
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* URI interface support.
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*/
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static GstURIType
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gst_rtmp_sink_uri_get_type (void)
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gst_rtmp_sink_uri_get_type (GType type)
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{
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return GST_URI_SINK;
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}
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static gchar **
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gst_rtmp_sink_uri_get_protocols (void)
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gst_rtmp_sink_uri_get_protocols (GType type)
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{
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static gchar *protocols[] =
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{ (char *) "rtmp", (char *) "rtmpt", (char *) "rtmps", (char *) "rtmpe",
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@ -88,47 +88,21 @@ static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc,
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GstBuffer ** buffer);
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static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
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static void
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_do_init (GType gtype)
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{
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static const GInterfaceInfo urihandler_info = {
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gst_rtmp_src_uri_handler_init,
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NULL,
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NULL
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};
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g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info);
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GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
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}
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GST_BOILERPLATE_FULL (GstRTMPSrc, gst_rtmp_src, GstPushSrc, GST_TYPE_PUSH_SRC,
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_do_init);
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static void
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gst_rtmp_src_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_set_details_simple (element_class,
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"RTMP Source",
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"Source/File",
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"Read RTMP streams",
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"Bastien Nocera <hadess@hadess.net>, "
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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}
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#define gst_rtmp_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTMPSrc, gst_rtmp_src, GST_TYPE_PUSH_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtmp_src_uri_handler_init));
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static void
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gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstPushSrcClass *gstpushsrc_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
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@ -137,9 +111,19 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
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gobject_class->get_property = gst_rtmp_src_get_property;
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/* properties */
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gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
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gst_element_class_install_std_props (gstelement_class,
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"location", PROP_LOCATION, G_PARAM_READWRITE, NULL);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_set_details_simple (gstelement_class,
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"RTMP Source",
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"Source/File",
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"Read RTMP streams",
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"Bastien Nocera <hadess@hadess.net>, "
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start);
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gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop);
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gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
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@ -148,10 +132,12 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
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gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_rtmp_src_do_seek);
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gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
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GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
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}
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static void
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gst_rtmp_src_init (GstRTMPSrc * rtmpsrc, GstRTMPSrcClass * klass)
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gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
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{
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rtmpsrc->cur_offset = 0;
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rtmpsrc->last_timestamp = 0;
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@ -175,13 +161,13 @@ gst_rtmp_src_finalize (GObject * object)
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*/
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static GstURIType
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gst_rtmp_src_uri_get_type (void)
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gst_rtmp_src_uri_get_type (GType type)
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{
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return GST_URI_SRC;
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}
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static gchar **
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gst_rtmp_src_uri_get_protocols (void)
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gst_rtmp_src_uri_get_protocols (GType type)
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{
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static gchar *protocols[] =
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{ (char *) "rtmp", (char *) "rtmpt", (char *) "rtmps", (char *) "rtmpe",
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@ -286,8 +272,9 @@ gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
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{
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GstRTMPSrc *src;
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GstBuffer *buf;
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guint8 *data;
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guint8 *data, *bdata;
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guint todo;
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gsize bsize;
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int read;
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int size;
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@ -300,15 +287,15 @@ gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
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GST_DEBUG ("reading from %" G_GUINT64_FORMAT
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", size %u", src->cur_offset, size);
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buf = gst_buffer_try_new_and_alloc (size);
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buf = gst_buffer_new_allocate (NULL, size, 0);
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if (G_UNLIKELY (buf == NULL)) {
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GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
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return GST_FLOW_ERROR;
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}
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todo = size;
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data = GST_BUFFER_DATA (buf);
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read = 0;
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bsize = todo = size;
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bdata = data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
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read = bsize = 0;
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while (todo > 0) {
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read = RTMP_Read (src->rtmp, (char *) data, todo);
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@ -316,7 +303,6 @@ gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
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if (G_UNLIKELY (read == 0 && todo == size)) {
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goto eos;
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} else if (G_UNLIKELY (read == 0)) {
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GST_BUFFER_SIZE (buf) -= todo;
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todo = 0;
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break;
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}
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@ -325,13 +311,16 @@ gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
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goto read_failed;
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if (read < todo) {
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data = &data[read];
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data += read;
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todo -= read;
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bsize += read;
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} else {
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todo = 0;
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bsize += todo;
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}
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GST_LOG (" got size %d", read);
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}
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gst_buffer_unmap (buf, bdata, bsize);
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if (src->discont) {
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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@ -465,7 +454,7 @@ gst_rtmp_src_prepare_seek_segment (GstBaseSrc * basesrc, GstEvent * event,
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}
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gst_segment_init (segment, GST_FORMAT_TIME);
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gst_segment_set_seek (segment, rate, format, flags, cur_type, cur, stop_type,
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gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type,
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stop, NULL);
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return TRUE;
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