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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-04 01:32:42 +00:00
rtsp-stream: added helper function for creating the sender/receiver parts
Code refactoring: introduced helper function for creating the receiver and the sender parts of the streaming pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=757488
This commit is contained in:
Patricia Muscalu
Sebastian Dröge
parent
457c00dc0a
commit
89bc8009dd
1 changed files with 166 additions and 127 deletions
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@ -2112,132 +2112,26 @@ on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
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gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
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}
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/**
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* gst_rtsp_stream_join_bin:
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* @stream: a #GstRTSPStream
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* @bin: (transfer none): a #GstBin to join
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* @rtpbin: (transfer none): a rtpbin element in @bin
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* @state: the target state of the new elements
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*
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* Join the #GstBin @bin that contains the element @rtpbin.
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*
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* @stream will link to @rtpbin, which must be inside @bin. The elements
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* added to @bin will be set to the state given in @state.
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*
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* Returns: %TRUE on success.
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*/
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gboolean
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gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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GstElement * rtpbin, GstState state)
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/* must be called with lock */
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static void
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create_sender_part (GstRTSPStream * stream, GstBin * bin,
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GstState state)
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{
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GstRTSPStreamPrivate *priv;
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gint i;
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guint idx;
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gchar *name;
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GstPad *pad, *sinkpad = NULL, *selpad;
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GstPadLinkReturn ret;
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GstPad *pad, *sinkpad = NULL;
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gboolean is_tcp = FALSE, is_udp = FALSE;
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g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
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g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
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g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
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gint i;
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priv = stream->priv;
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g_mutex_lock (&priv->lock);
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if (priv->is_joined)
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goto was_joined;
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/* create a session with the same index as the stream */
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idx = priv->idx;
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GST_INFO ("stream %p joining bin as session %u", stream, idx);
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is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
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is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
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(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
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if (is_udp && !alloc_ports (stream))
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goto no_ports;
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/* update the dscp qos field in the sinks */
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update_dscp_qos (stream);
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if (priv->profiles & GST_RTSP_PROFILE_SAVP
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|| priv->profiles & GST_RTSP_PROFILE_SAVPF) {
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/* For SRTP */
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g_signal_connect (rtpbin, "request-rtp-encoder",
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(GCallback) request_rtp_encoder, stream);
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g_signal_connect (rtpbin, "request-rtcp-encoder",
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(GCallback) request_rtcp_encoder, stream);
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g_signal_connect (rtpbin, "request-rtp-decoder",
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(GCallback) request_rtp_rtcp_decoder, stream);
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g_signal_connect (rtpbin, "request-rtcp-decoder",
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(GCallback) request_rtp_rtcp_decoder, stream);
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}
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if (priv->sinkpad) {
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g_signal_connect (rtpbin, "request-pt-map",
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(GCallback) request_pt_map, stream);
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}
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/* get pads from the RTP session element for sending and receiving
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* RTP/RTCP*/
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if (priv->srcpad) {
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/* get a pad for sending RTP */
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name = g_strdup_printf ("send_rtp_sink_%u", idx);
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priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* link the RTP pad to the session manager, it should not really fail unless
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* this is not really an RTP pad */
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ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
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if (ret != GST_PAD_LINK_OK)
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goto link_failed;
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name = g_strdup_printf ("send_rtp_src_%u", idx);
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priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
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g_free (name);
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} else {
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/* Need to connect our sinkpad from here */
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g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
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/* EOS */
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g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
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name = g_strdup_printf ("recv_rtp_sink_%u", idx);
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priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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}
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name = g_strdup_printf ("send_rtcp_src_%u", idx);
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priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
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priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* get the session */
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g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
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g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
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stream);
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g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
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stream);
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g_signal_connect (priv->session, "on-ssrc-active",
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(GCallback) on_ssrc_active, stream);
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g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
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stream);
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g_signal_connect (priv->session, "on-bye-timeout",
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(GCallback) on_bye_timeout, stream);
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g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
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stream);
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/* signal for sender ssrc */
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g_signal_connect (priv->session, "on-new-sender-ssrc",
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(GCallback) on_new_sender_ssrc, stream);
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g_signal_connect (priv->session, "on-sender-ssrc-active",
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(GCallback) on_sender_ssrc_active, stream);
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if (is_udp)
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/* update the dscp qos field in the sinks */
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update_dscp_qos (stream);
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for (i = 0; i < 2; i++) {
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GstPad *teepad, *queuepad;
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@ -2261,7 +2155,6 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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* and link the udpsink (for UDP) or appsink (for TCP) directly to
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* the session.
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*/
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/* Only link the RTP send src if we're going to send RTP, link
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* the RTCP send src always */
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if (priv->srcpad || i == 1) {
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@ -2347,6 +2240,37 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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}
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}
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/* check if we need to set to a special state */
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if (state != GST_STATE_NULL) {
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if (priv->udpsink[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->udpsink[i], state);
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if (priv->appsink[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->appsink[i], state);
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if (priv->appqueue[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->appqueue[i], state);
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if (priv->udpqueue[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->udpqueue[i], state);
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if (priv->tee[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->tee[i], state);
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}
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}
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}
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/* must be called with lock */
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static void
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create_receiver_part (GstRTSPStream * stream, GstBin * bin,
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GstState state)
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{
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GstRTSPStreamPrivate *priv;
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GstPad *pad, *selpad;
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gboolean is_tcp;
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gint i;
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priv = stream->priv;
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is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
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for (i = 0; i < 2; i++) {
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/* Only connect recv RTP sink if we expect to receive RTP. Connect recv
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* RTCP sink always */
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if (priv->sinkpad || i == 1) {
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@ -2421,22 +2345,137 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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/* check if we need to set to a special state */
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if (state != GST_STATE_NULL) {
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if (priv->udpsink[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->udpsink[i], state);
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if (priv->appsink[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->appsink[i], state);
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if (priv->appqueue[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->appqueue[i], state);
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if (priv->udpqueue[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->udpqueue[i], state);
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if (priv->tee[i] && (priv->srcpad || i == 1))
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gst_element_set_state (priv->tee[i], state);
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if (priv->funnel[i] && (priv->sinkpad || i == 1))
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gst_element_set_state (priv->funnel[i], state);
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if (priv->appsrc[i] && (priv->sinkpad || i == 1))
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gst_element_set_state (priv->appsrc[i], state);
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}
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}
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}
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/**
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* gst_rtsp_stream_join_bin:
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* @stream: a #GstRTSPStream
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* @bin: (transfer none): a #GstBin to join
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* @rtpbin: (transfer none): a rtpbin element in @bin
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* @state: the target state of the new elements
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*
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* Join the #GstBin @bin that contains the element @rtpbin.
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*
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* @stream will link to @rtpbin, which must be inside @bin. The elements
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* added to @bin will be set to the state given in @state.
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*
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* Returns: %TRUE on success.
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*/
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gboolean
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gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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GstElement * rtpbin, GstState state)
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{
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GstRTSPStreamPrivate *priv;
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guint idx;
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gchar *name;
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GstPadLinkReturn ret;
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gboolean is_udp;
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g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
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g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
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g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
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priv = stream->priv;
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g_mutex_lock (&priv->lock);
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if (priv->is_joined)
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goto was_joined;
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/* create a session with the same index as the stream */
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idx = priv->idx;
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GST_INFO ("stream %p joining bin as session %u", stream, idx);
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is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
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(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
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if (is_udp && !alloc_ports (stream))
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goto no_ports;
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if (priv->profiles & GST_RTSP_PROFILE_SAVP
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|| priv->profiles & GST_RTSP_PROFILE_SAVPF) {
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/* For SRTP */
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g_signal_connect (rtpbin, "request-rtp-encoder",
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(GCallback) request_rtp_encoder, stream);
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g_signal_connect (rtpbin, "request-rtcp-encoder",
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(GCallback) request_rtcp_encoder, stream);
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g_signal_connect (rtpbin, "request-rtp-decoder",
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(GCallback) request_rtp_rtcp_decoder, stream);
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g_signal_connect (rtpbin, "request-rtcp-decoder",
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(GCallback) request_rtp_rtcp_decoder, stream);
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}
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if (priv->sinkpad) {
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g_signal_connect (rtpbin, "request-pt-map",
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(GCallback) request_pt_map, stream);
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}
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/* get pads from the RTP session element for sending and receiving
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* RTP/RTCP*/
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if (priv->srcpad) {
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/* get a pad for sending RTP */
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name = g_strdup_printf ("send_rtp_sink_%u", idx);
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priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* link the RTP pad to the session manager, it should not really fail unless
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* this is not really an RTP pad */
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ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
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if (ret != GST_PAD_LINK_OK)
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goto link_failed;
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name = g_strdup_printf ("send_rtp_src_%u", idx);
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priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
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g_free (name);
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} else {
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/* Need to connect our sinkpad from here */
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g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
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/* EOS */
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g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
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name = g_strdup_printf ("recv_rtp_sink_%u", idx);
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priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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}
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name = g_strdup_printf ("send_rtcp_src_%u", idx);
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priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
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priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* get the session */
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g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
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g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
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stream);
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g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
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stream);
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g_signal_connect (priv->session, "on-ssrc-active",
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(GCallback) on_ssrc_active, stream);
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g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
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stream);
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g_signal_connect (priv->session, "on-bye-timeout",
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(GCallback) on_bye_timeout, stream);
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g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
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stream);
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/* signal for sender ssrc */
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g_signal_connect (priv->session, "on-new-sender-ssrc",
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(GCallback) on_new_sender_ssrc, stream);
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g_signal_connect (priv->session, "on-sender-ssrc-active",
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(GCallback) on_sender_ssrc_active, stream);
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create_sender_part (stream, bin, state);
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create_receiver_part (stream, bin, state);
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if (priv->srcpad) {
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/* be notified of caps changes */
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