* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:rtsp-stream-transport
+ * @short_description: A media stream transport configuration
+ * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
+ *
+ * The #GstRTSPStreamTransport configures the transport used by a
+ * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
+ *
+ * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+ * to handle the RTP and RTCP packets from the stream, for example when they
+ * need to be sent over TCP.
+ *
+ * With gst_rtsp_stream_transport_set_active() the transports are added and
+ * removed from the stream.
+ *
+ * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+ * is received from the client. It will also call
+ * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
#include <string.h>
#include <stdlib.h>
-#include <gst/app/gstappsrc.h>
-#include <gst/app/gstappsink.h>
-
#include "rtsp-stream-transport.h"
+#define GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportPrivate))
+
+struct _GstRTSPStreamTransportPrivate
+{
+ GstRTSPStream *stream;
+
+ GstRTSPSendFunc send_rtp;
+ GstRTSPSendFunc send_rtcp;
+ gpointer user_data;
+ GDestroyNotify notify;
+
+ GstRTSPKeepAliveFunc keep_alive;
+ gpointer ka_user_data;
+ GDestroyNotify ka_notify;
+ gboolean active;
+ gboolean timed_out;
+
+ GstRTSPTransport *transport;
+ GstRTSPUrl *url;
+
+ GObject *rtpsource;
+};
+
enum
{
PROP_0,
{
GObjectClass *gobject_class;
+ g_type_class_add_private (klass, sizeof (GstRTSPStreamTransportPrivate));
+
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_stream_transport_finalize;
static void
gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
{
+ GstRTSPStreamTransportPrivate *priv =
+ GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE (trans);
+
+ trans->priv = priv;
}
static void
gst_rtsp_stream_transport_finalize (GObject * obj)
{
+ GstRTSPStreamTransportPrivate *priv;
GstRTSPStreamTransport *trans;
trans = GST_RTSP_STREAM_TRANSPORT (obj);
+ priv = trans->priv;
/* remove callbacks now */
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
- if (trans->transport)
- gst_rtsp_transport_free (trans->transport);
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
-#if 0
- if (trans->rtpsource)
- g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
-#endif
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
}
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
{
+ GstRTSPStreamTransportPrivate *priv;
GstRTSPStreamTransport *trans;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (tr != NULL, NULL);
trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
- trans->stream = stream;
- trans->transport = tr;
+ priv = trans->priv;
+ priv->stream = stream;
+ priv->transport = tr;
return trans;
}
/**
+ * gst_rtsp_stream_transport_get_stream:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the #GstRTSPStream used when constructing @trans.
+ *
+ * Returns: (transfer none): the stream used when constructing @trans.
+ */
+GstRTSPStream *
+gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->stream;
+}
+
+/**
* gst_rtsp_stream_transport_set_callbacks:
* @trans: a #GstRTSPStreamTransport
* @send_rtp: (scope notified): a callback called when RTP should be sent
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
gpointer user_data, GDestroyNotify notify)
{
- trans->send_rtp = send_rtp;
- trans->send_rtcp = send_rtcp;
- if (trans->notify)
- trans->notify (trans->user_data);
- trans->user_data = user_data;
- trans->notify = notify;
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->send_rtp = send_rtp;
+ priv->send_rtcp = send_rtcp;
+ if (priv->notify)
+ priv->notify (priv->user_data);
+ priv->user_data = user_data;
+ priv->notify = notify;
}
/**
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
- trans->keep_alive = keep_alive;
- if (trans->ka_notify)
- trans->ka_notify (trans->ka_user_data);
- trans->ka_user_data = user_data;
- trans->ka_notify = notify;
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->keep_alive = keep_alive;
+ if (priv->ka_notify)
+ priv->ka_notify (priv->ka_user_data);
+ priv->ka_user_data = user_data;
+ priv->ka_notify = notify;
}
* @trans: a #GstRTSPStreamTransport
* @tr: (transfer full): a client #GstRTSPTransport
*
- * Set @ct as the client transport. This function takes ownership of
- * the passed @tr.
+ * Set @tr as the client transport. This function takes ownership of the
+ * passed @tr.
*/
void
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
GstRTSPTransport * tr)
{
+ GstRTSPStreamTransportPrivate *priv;
+
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
g_return_if_fail (tr != NULL);
+ priv = trans->priv;
+
/* keep track of the transports in the stream. */
- if (trans->transport)
- gst_rtsp_transport_free (trans->transport);
- trans->transport = tr;
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
+ priv->transport = tr;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_transport:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the transport configured in @trans.
+ *
+ * Returns: (transfer none): the transport configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPTransport *
+gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->transport;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_url:
+ * @trans: a #GstRTSPStreamTransport
+ * @url: (transfer none): a client #GstRTSPUrl
+ *
+ * Set @url as the client url.
+ */
+void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
+ const GstRTSPUrl * url)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ /* keep track of the transports in the stream. */
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
+ priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
+}
+
+/**
+ * gst_rtsp_stream_transport_get_url:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the url configured in @trans.
+ *
+ * Returns: (transfer none): the url configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->url;
+}
+
+ /**
+ * gst_rtsp_stream_transport_get_rtpinfo:
+ * @trans: a #GstRTSPStreamTransport
+ * @start_time: a star time
+ *
+ * Get the RTP-Info string for @trans and @start_time.
+ *
+ * Returns: the RTPInfo string for @trans and @start_time or %NULL when
+ * the RTP-Info could not be determined. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
+ GstClockTime start_time)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gchar *url_str;
+ GString *rtpinfo;
+ guint rtptime, seq, clock_rate;
+ GstClockTime running_time = GST_CLOCK_TIME_NONE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ priv = trans->priv;
+
+ if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
+ &running_time))
+ return NULL;
+
+ GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
+ rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
+
+ if (GST_CLOCK_TIME_IS_VALID (running_time)
+ && GST_CLOCK_TIME_IS_VALID (start_time)) {
+ if (running_time > start_time) {
+ rtptime -=
+ gst_util_uint64_scale_int (running_time - start_time, clock_rate,
+ GST_SECOND);
+ } else {
+ rtptime +=
+ gst_util_uint64_scale_int (start_time - running_time, clock_rate,
+ GST_SECOND);
+ }
+ }
+ GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
+ rtptime, GST_TIME_ARGS (start_time));
+
+ rtpinfo = g_string_new ("");
+
+ url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
+
+ return g_string_free (rtpinfo, FALSE);
+}
+
+/**
+ * gst_rtsp_stream_transport_set_active:
+ * @trans: a #GstRTSPStreamTransport
+ * @active: new state of @trans
+ *
+ * Activate or deactivate datatransfer configured in @trans.
+ *
+ * Returns: %TRUE when the state was changed.
+ */
+gboolean
+gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
+ gboolean active)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->active == active)
+ return FALSE;
+
+ if (active)
+ res = gst_rtsp_stream_add_transport (priv->stream, trans);
+ else
+ res = gst_rtsp_stream_remove_transport (priv->stream, trans);
+
+ if (res)
+ priv->active = active;
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ * @timedout: timed out value
+ *
+ * Set the timed out state of @trans to @timedout
+ */
+void
+gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
+ gboolean timedout)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ trans->priv->timed_out = timedout;
+}
+
+/**
+ * gst_rtsp_stream_transport_is_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Check if @trans is timed out.
+ *
+ * Returns: %TRUE if @trans timed out.
+ */
+gboolean
+gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ return trans->priv->timed_out;
}
/**
gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
+ GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
- if (trans->send_rtp)
+ priv = trans->priv;
+
+ if (priv->send_rtp)
res =
- trans->send_rtp (buffer, trans->transport->interleaved.min,
- trans->user_data);
+ priv->send_rtp (buffer, priv->transport->interleaved.min,
+ priv->user_data);
return res;
}
gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
+ GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
- if (trans->send_rtcp)
+ priv = trans->priv;
+
+ if (priv->send_rtcp)
res =
- trans->send_rtcp (buffer, trans->transport->interleaved.max,
- trans->user_data);
+ priv->send_rtcp (buffer, priv->transport->interleaved.max,
+ priv->user_data);
return res;
}
+
+/**
+ * gst_rtsp_stream_transport_keep_alive:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Signal the installed keep_alive callback for @trans.
+ */
+void
+gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ priv = trans->priv;
+
+ if (priv->keep_alive)
+ priv->keep_alive (priv->ka_user_data);
+}
projects / platform / upstream / gstreamer.git / blobdiff