mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-29 23:32:33 +00:00
Rework the way we handle transports for streams
Make the media accept an array of transports for the streams that we have configured for the play/pause requests. Implement server states for a client and its media. Require 0.10.22.1 (git HEAD) of gstreamer.
This commit is contained in:
Wim Taymans
parent
f303eef9bb
commit
d5a00f1f23
6 changed files with 208 additions and 115 deletions
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@ -37,8 +37,8 @@ AC_SUBST(GST_MAJORMINOR)
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AM_PROG_LIBTOOL
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=0.10.20
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GSTPB_REQ=0.10.20
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GST_REQ=0.10.22.1
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GSTPB_REQ=0.10.22.1
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dnl export for .pc files
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AC_SUBST([GST_REQ])
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@ -287,8 +287,12 @@ handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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if (!media)
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goto not_found;
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/* the session state must be playing or recording */
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if (media->state != GST_RTSP_STATE_PLAYING &&
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media->state != GST_RTSP_STATE_RECORDING)
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goto invalid_state;
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gst_rtsp_session_media_pause (media);
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g_object_unref (session);
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/* construct the response now */
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code = GST_RTSP_STS_OK;
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@ -296,6 +300,10 @@ handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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send_response (client, &response);
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/* the state is now READY */
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media->state = GST_RTSP_STATE_READY;
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g_object_unref (session);
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return FALSE;
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/* ERRORS */
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@ -308,6 +316,11 @@ not_found:
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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return FALSE;
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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return FALSE;
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}
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}
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static gboolean
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@ -329,6 +342,11 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *req
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if (!media)
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goto not_found;
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/* the session state must be playing or ready */
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if (media->state != GST_RTSP_STATE_PLAYING &&
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media->state != GST_RTSP_STATE_READY)
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goto invalid_state;
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/* grab RTPInfo from the payloaders now */
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rtpinfo = g_string_new ("");
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@ -362,6 +380,8 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *req
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/* start playing after sending the request */
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gst_rtsp_session_media_play (media);
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media->state = GST_RTSP_STATE_PLAYING;
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g_object_unref (session);
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return FALSE;
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@ -377,6 +397,11 @@ not_found:
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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return FALSE;
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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return FALSE;
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}
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}
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static gboolean
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@ -508,10 +533,23 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
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g_free (trans_str);
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g_object_unref (session);
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send_response (client, &response);
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/* update the state */
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switch (media->state) {
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case GST_RTSP_STATE_PLAYING:
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case GST_RTSP_STATE_RECORDING:
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case GST_RTSP_STATE_READY:
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/* no state change */
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break;
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default:
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media->state = GST_RTSP_STATE_READY;
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break;
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}
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g_object_unref (session);
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return TRUE;
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/* ERRORS */
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@ -538,100 +538,125 @@ state_failed:
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}
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}
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gboolean
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gst_rtsp_media_stream_add (GstRTSPMediaStream *stream, GstRTSPTransport *ct)
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{
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g_return_val_if_fail (stream != NULL, FALSE);
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g_return_val_if_fail (ct != NULL, FALSE);
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g_return_val_if_fail (stream->prepared, FALSE);
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g_message ("adding %s:%d", ct->destination, ct->client_port.min);
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g_signal_emit_by_name (stream->udpsink[0], "add", ct->destination, ct->client_port.min, NULL);
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g_signal_emit_by_name (stream->udpsink[1], "add", ct->destination, ct->client_port.max, NULL);
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return TRUE;
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}
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gboolean
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gst_rtsp_media_stream_remove (GstRTSPMediaStream *stream, GstRTSPTransport *ct)
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{
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g_return_val_if_fail (stream != NULL, FALSE);
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g_return_val_if_fail (ct != NULL, FALSE);
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g_return_val_if_fail (stream->prepared, FALSE);
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g_message ("removing %s:%d", ct->destination, ct->client_port.min);
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g_signal_emit_by_name (stream->udpsink[0], "remove", ct->destination, ct->client_port.min, NULL);
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g_signal_emit_by_name (stream->udpsink[1], "remove", ct->destination, ct->client_port.max, NULL);
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return TRUE;
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}
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/**
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* gst_rtsp_media_play:
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* @media: a #GstRTSPMedia
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* @transports: a GArray of #GstRTSPMediaTrans pointers
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*
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* Tell the @media to start playing and streaming to the client.
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* Start playing @media for to the transports in @transports.
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*
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* Returns: a #GstStateChangeReturn
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* Returns: %TRUE on success.
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*/
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GstStateChangeReturn
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gst_rtsp_media_play (GstRTSPMedia *media)
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gboolean
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gst_rtsp_media_play (GstRTSPMedia *media, GArray *transports)
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{
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gint i;
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GstStateChangeReturn ret;
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
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g_return_val_if_fail (transports != NULL, FALSE);
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g_return_val_if_fail (media->prepared, FALSE);
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for (i = 0; i < transports->len; i++) {
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GstRTSPMediaTrans *tr;
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GstRTSPMediaStream *stream;
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GstRTSPTransport *trans;
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/* we need a non-NULL entry in the array */
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tr = g_array_index (transports, GstRTSPMediaTrans *, i);
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if (tr == NULL)
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continue;
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/* we need a transport */
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if (!(trans = tr->transport))
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continue;
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/* get the stream and add the destinations */
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stream = gst_rtsp_media_get_stream (media, tr->idx);
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g_message ("adding %s:%d-%d", trans->destination, trans->client_port.min, trans->client_port.max);
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g_signal_emit_by_name (stream->udpsink[0], "add", trans->destination, trans->client_port.min, NULL);
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g_signal_emit_by_name (stream->udpsink[1], "add", trans->destination, trans->client_port.max, NULL);
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}
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g_message ("playing");
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ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
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return ret;
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return TRUE;
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}
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/**
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* gst_rtsp_media_pause:
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* @media: a #GstRTSPMedia
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* @transports: a array of #GstRTSPTransport pointers
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*
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* Tell the @media to pause.
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* Pause playing @media for to the transports in @transports.
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*
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* Returns: a #GstStateChangeReturn
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* Returns: %TRUE on success.
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*/
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GstStateChangeReturn
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gst_rtsp_media_pause (GstRTSPMedia *media)
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gboolean
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gst_rtsp_media_pause (GstRTSPMedia *media, GArray *transports)
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{
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gint i;
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GstStateChangeReturn ret;
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
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g_return_val_if_fail (transports != NULL, FALSE);
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g_return_val_if_fail (media->prepared, FALSE);
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g_message ("paused");
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for (i = 0; i < transports->len; i++) {
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GstRTSPMediaTrans *tr;
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GstRTSPMediaStream *stream;
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GstRTSPTransport *trans;
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/* we need a non-NULL entry in the array */
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tr = g_array_index (transports, GstRTSPMediaTrans *, i);
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if (tr == NULL)
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continue;
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/* we need a transport */
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if (!(trans = tr->transport))
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continue;
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/* get the stream and add the destinations */
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stream = gst_rtsp_media_get_stream (media, tr->idx);
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g_message ("removing %s:%d-%d", trans->destination, trans->client_port.min, trans->client_port.max);
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g_signal_emit_by_name (stream->udpsink[0], "remove", trans->destination, trans->client_port.min, NULL);
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g_signal_emit_by_name (stream->udpsink[1], "remove", trans->destination, trans->client_port.max, NULL);
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}
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g_message ("pause");
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ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
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return ret;
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return TRUE;
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}
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/**
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* gst_rtsp_media_stop:
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* gst_rtsp_media_stream_stop:
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* @media: a #GstRTSPMedia
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* @transports: a GArray of #GstRTSPMediaTrans pointers
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*
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* Tell the @media to stop playing. After this call the media
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* cannot be played or paused anymore
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* Stop playing @media for to the transports in @transports.
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*
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* Returns: a #GstStateChangeReturn
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* Returns: %TRUE on success.
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*/
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GstStateChangeReturn
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gst_rtsp_media_stop (GstRTSPMedia *media)
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gboolean
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gst_rtsp_media_stop (GstRTSPMedia *media, GArray *transports)
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{
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GstStateChangeReturn ret;
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
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g_return_val_if_fail (transports != NULL, FALSE);
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g_return_val_if_fail (media->prepared, FALSE);
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gst_rtsp_media_pause (media, transports);
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g_message ("stop");
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ret = gst_element_set_state (media->pipeline, GST_STATE_NULL);
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return ret;
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return TRUE;
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}
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@ -38,6 +38,20 @@ G_BEGIN_DECLS
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typedef struct _GstRTSPMediaStream GstRTSPMediaStream;
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typedef struct _GstRTSPMedia GstRTSPMedia;
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typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
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typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
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/**
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* GstRTSPMediaTrans:
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* @idx: a stream index
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* @transport: a transport description
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*
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* A Transport description for stream @idx
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*/
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struct _GstRTSPMediaTrans {
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guint idx;
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GstRTSPTransport *transport;
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};
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/**
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* GstRTSPMediaStream:
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@ -136,13 +150,9 @@ gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
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guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
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GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
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/* add destinations to a stream */
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gboolean gst_rtsp_media_stream_add (GstRTSPMediaStream *stream, GstRTSPTransport *ct);
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gboolean gst_rtsp_media_stream_remove (GstRTSPMediaStream *stream, GstRTSPTransport *ct);
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GstStateChangeReturn gst_rtsp_media_play (GstRTSPMedia *media);
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GstStateChangeReturn gst_rtsp_media_pause (GstRTSPMedia *media);
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GstStateChangeReturn gst_rtsp_media_stop (GstRTSPMedia *media);
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gboolean gst_rtsp_media_play (GstRTSPMedia *media, GArray *trans);
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gboolean gst_rtsp_media_pause (GstRTSPMedia *media, GArray *trans);
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gboolean gst_rtsp_media_stop (GstRTSPMedia *media, GArray *trans);
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G_END_DECLS
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@ -42,10 +42,10 @@ gst_rtsp_session_init (GstRTSPSession * session)
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}
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static void
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gst_rtsp_session_free_stream (GstRTSPSessionStream *stream, GstRTSPSessionMedia *media)
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gst_rtsp_session_free_stream (GstRTSPSessionStream *stream)
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{
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if (stream->client_trans)
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gst_rtsp_transport_free (stream->client_trans);
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if (stream->trans.transport)
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gst_rtsp_transport_free (stream->trans.transport);
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g_free (stream);
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}
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@ -53,9 +53,19 @@ gst_rtsp_session_free_stream (GstRTSPSessionStream *stream, GstRTSPSessionMedia
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static void
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gst_rtsp_session_free_media (GstRTSPSessionMedia *media, GstRTSPSession *session)
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{
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g_list_foreach (media->streams, (GFunc) gst_rtsp_session_free_stream,
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media);
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g_list_free (media->streams);
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guint size, i;
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size = media->streams->len;
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for (i = 0; i < size; i++) {
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GstRTSPSessionStream *stream;
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stream = g_array_index (media->streams, GstRTSPSessionStream *, i);
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if (stream)
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gst_rtsp_session_free_stream (stream);
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}
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g_array_free (media->streams, TRUE);
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if (media->url)
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gst_rtsp_url_free (media->url);
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@ -100,10 +110,22 @@ gst_rtsp_session_manage_media (GstRTSPSession *sess, const GstRTSPUrl *uri,
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GstRTSPMedia *media)
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{
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GstRTSPSessionMedia *result;
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guint n_streams;
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g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
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g_return_val_if_fail (uri != NULL, NULL);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
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g_return_val_if_fail (media->prepared, NULL);
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result = g_new0 (GstRTSPSessionMedia, 1);
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result->media = media;
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result->url = gst_rtsp_url_copy ((GstRTSPUrl *)uri);
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result->state = GST_RTSP_STATE_INIT;
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/* prealloc the streams now, filled with NULL */
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n_streams = gst_rtsp_media_n_streams (media);
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result->streams = g_array_sized_new (FALSE, TRUE, sizeof (GstRTSPSessionStream *), n_streams);
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g_array_set_size (result->streams, n_streams);
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sess->medias = g_list_prepend (sess->medias, result);
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@ -158,29 +180,25 @@ gst_rtsp_session_media_get_stream (GstRTSPSessionMedia *media, guint idx)
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{
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GstRTSPSessionStream *result;
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GstRTSPMediaStream *media_stream;
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GList *walk;
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g_return_val_if_fail (media != NULL, NULL);
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g_return_val_if_fail (media->media != NULL, NULL);
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media_stream = gst_rtsp_media_get_stream (media->media, idx);
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if (media_stream == NULL)
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goto no_media;
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if (idx >= media->streams->len)
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return NULL;
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result = NULL;
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for (walk = media->streams; walk; walk = g_list_next (walk)) {
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result = (GstRTSPSessionStream *) walk->data;
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if (result->media_stream == media_stream)
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break;
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result = NULL;
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}
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result = g_array_index (media->streams, GstRTSPSessionStream *, idx);
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if (result == NULL) {
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media_stream = gst_rtsp_media_get_stream (media->media, idx);
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if (media_stream == NULL)
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goto no_media;
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result = g_new0 (GstRTSPSessionStream, 1);
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result->trans.idx = idx;
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result->trans.transport = NULL;
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result->media_stream = media_stream;
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media->streams = g_list_prepend (media->streams, result);
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g_array_insert_val (media->streams, idx, result);
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}
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return result;
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@ -232,9 +250,9 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream *stream,
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st->client_port = ct->client_port;
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/* keep track of the transports */
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if (stream->client_trans)
|
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gst_rtsp_transport_free (stream->client_trans);
|
||||
stream->client_trans = ct;
|
||||
if (stream->trans.transport)
|
||||
gst_rtsp_transport_free (stream->trans.transport);
|
||||
stream->trans.transport = ct;
|
||||
|
||||
st->server_port.min = stream->media_stream->server_port.min;
|
||||
st->server_port.max = stream->media_stream->server_port.max;
|
||||
|
|
@ -242,28 +260,20 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream *stream,
|
|||
return st;
|
||||
}
|
||||
|
||||
|
||||
/**
|
||||
* gst_rtsp_session_media_play:
|
||||
* @media: a #GstRTSPSessionMedia
|
||||
*
|
||||
* Tell the media object @media to start playing and streaming to the client.
|
||||
*
|
||||
* Returns: a #GstStateChangeReturn
|
||||
* Returns: %TRUE on success.
|
||||
*/
|
||||
GstStateChangeReturn
|
||||
gboolean
|
||||
gst_rtsp_session_media_play (GstRTSPSessionMedia *media)
|
||||
{
|
||||
GstStateChangeReturn ret;
|
||||
GstRTSPSessionStream *stream;
|
||||
GList *walk;
|
||||
gboolean ret;
|
||||
|
||||
for (walk = media->streams; walk; walk = g_list_next (walk)) {
|
||||
stream = (GstRTSPSessionStream *) walk->data;
|
||||
|
||||
gst_rtsp_media_stream_add (stream->media_stream, stream->client_trans);
|
||||
}
|
||||
ret = gst_rtsp_media_play (media->media);
|
||||
ret = gst_rtsp_media_play (media->media, media->streams);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
|
@ -274,14 +284,14 @@ gst_rtsp_session_media_play (GstRTSPSessionMedia *media)
|
|||
*
|
||||
* Tell the media object @media to pause.
|
||||
*
|
||||
* Returns: a #GstStateChangeReturn
|
||||
* Returns: %TRUE on success.
|
||||
*/
|
||||
GstStateChangeReturn
|
||||
gboolean
|
||||
gst_rtsp_session_media_pause (GstRTSPSessionMedia *media)
|
||||
{
|
||||
GstStateChangeReturn ret;
|
||||
gboolean ret;
|
||||
|
||||
ret = gst_rtsp_media_pause (media->media);
|
||||
ret = gst_rtsp_media_pause (media->media, media->streams);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
|
@ -293,14 +303,14 @@ gst_rtsp_session_media_pause (GstRTSPSessionMedia *media)
|
|||
* Tell the media object @media to stop playing. After this call the media
|
||||
* cannot be played or paused anymore
|
||||
*
|
||||
* Returns: a #GstStateChangeReturn
|
||||
* Returns: %TRUE on success.
|
||||
*/
|
||||
GstStateChangeReturn
|
||||
gboolean
|
||||
gst_rtsp_session_media_stop (GstRTSPSessionMedia *media)
|
||||
{
|
||||
GstStateChangeReturn ret;
|
||||
gboolean ret;
|
||||
|
||||
ret = gst_rtsp_media_stop (media->media);
|
||||
ret = gst_rtsp_media_stop (media->media, media->streams);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -45,17 +45,18 @@ typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
|
|||
|
||||
/**
|
||||
* GstRTSPSessionStream:
|
||||
* @trans: the media transport
|
||||
* @media_stream: the controlled media stream
|
||||
*
|
||||
* Configuration of a stream. A stream is an audio or video stream related to a
|
||||
* media.
|
||||
*/
|
||||
struct _GstRTSPSessionStream
|
||||
{
|
||||
GstRTSPMediaTrans trans;
|
||||
|
||||
/* the stream of the media */
|
||||
GstRTSPMediaStream *media_stream;
|
||||
|
||||
/* client and server transports */
|
||||
GstRTSPTransport *client_trans;
|
||||
};
|
||||
|
||||
/**
|
||||
|
|
@ -71,8 +72,11 @@ struct _GstRTSPSessionMedia
|
|||
/* the pipeline for the media */
|
||||
GstRTSPMedia *media;
|
||||
|
||||
/* the server state */
|
||||
GstRTSPState state;
|
||||
|
||||
/* configuration for the different streams */
|
||||
GList *streams;
|
||||
GArray *streams;
|
||||
};
|
||||
|
||||
/**
|
||||
|
|
@ -96,21 +100,27 @@ struct _GstRTSPSessionClass {
|
|||
|
||||
GType gst_rtsp_session_get_type (void);
|
||||
|
||||
/* create a new session */
|
||||
GstRTSPSession * gst_rtsp_session_new (const gchar *sessionid);
|
||||
|
||||
/* handle media in a session */
|
||||
GstRTSPSessionMedia * gst_rtsp_session_manage_media (GstRTSPSession *sess,
|
||||
const GstRTSPUrl *uri,
|
||||
GstRTSPMedia *media);
|
||||
/* get media in a session */
|
||||
GstRTSPSessionMedia * gst_rtsp_session_get_media (GstRTSPSession *sess,
|
||||
const GstRTSPUrl *uri);
|
||||
|
||||
GstStateChangeReturn gst_rtsp_session_media_play (GstRTSPSessionMedia *media);
|
||||
GstStateChangeReturn gst_rtsp_session_media_pause (GstRTSPSessionMedia *media);
|
||||
GstStateChangeReturn gst_rtsp_session_media_stop (GstRTSPSessionMedia *media);
|
||||
/* control media */
|
||||
gboolean gst_rtsp_session_media_play (GstRTSPSessionMedia *media);
|
||||
gboolean gst_rtsp_session_media_pause (GstRTSPSessionMedia *media);
|
||||
gboolean gst_rtsp_session_media_stop (GstRTSPSessionMedia *media);
|
||||
|
||||
/* get stream config */
|
||||
GstRTSPSessionStream * gst_rtsp_session_media_get_stream (GstRTSPSessionMedia *media,
|
||||
guint idx);
|
||||
|
||||
/* configure transport */
|
||||
GstRTSPTransport * gst_rtsp_session_stream_set_transport (GstRTSPSessionStream *stream,
|
||||
GstRTSPTransport *ct);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue