[calls] sip: media-pipeline: Take srtp into account when determing pipeline state

[Evangelos Ribeiro Tzaras <devrtz src gnome org>]

commit 47d4164a0970844be9e3a9bdb69461e8c257c77e
Author: Evangelos Ribeiro Tzaras <devrtz fortysixandtwo eu>
Date:   Thu Apr 7 15:29:39 2022 +0200

    sip: media-pipeline: Take srtp into account when determing pipeline state
    
    If we're using srtp we should also consider the state of srtpenc and srtpdec
    elements when determining the state of the whole pipeline.

 plugins/sip/calls-sip-media-pipeline.c | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)
---
diff --git a/plugins/sip/calls-sip-media-pipeline.c b/plugins/sip/calls-sip-media-pipeline.c
index 7cfb4f7d..1e58294f 100644
--- a/plugins/sip/calls-sip-media-pipeline.c
+++ b/plugins/sip/calls-sip-media-pipeline.c
@@ -206,22 +206,25 @@ set_state (CallsSipMediaPipeline  *self,
 static void
 check_element_maps (CallsSipMediaPipeline *self)
 {
+  uint all_rtp_elements;
+
   g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
 
-  /* TODO take encryption into account */
-  if (self->element_map_playing == EL_ALL_RTP) {
+  all_rtp_elements = self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP;
+
+  if (self->element_map_playing == all_rtp_elements) {
     g_debug ("All pipeline elements are playing");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
     return;
   }
 
-  if (self->element_map_paused == EL_ALL_RTP) {
+  if (self->element_map_paused == all_rtp_elements) {
     g_debug ("All pipeline elements are paused");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
     return;
   }
 
-  if (self->element_map_stopped == EL_ALL_RTP) {
+  if (self->element_map_stopped == all_rtp_elements) {
     g_debug ("All pipeline elements are stopped");
     set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
     return;
@@ -319,13 +322,10 @@ on_bus_message (GstBus     *bus,
     else if (message->src == GST_OBJECT (self->rtcp_sink))
       element_id = EL_RTCP_SINK;
 
-    /* TODO srtp encryption
-       else if (message->src == GST_OBJECT (self->srtpenc))
-       element_id = EL_SRTP_ENCODER;
-       else if (message->src == GST_OBJECT (self->srtpdec))
-       element_id = EL_SRTP_DECODER;
-     */
-
+    else if (message->src == GST_OBJECT (self->srtpenc))
+      element_id = EL_SRTP_ENCODER;
+    else if (message->src == GST_OBJECT (self->srtpdec))
+      element_id = EL_SRTP_DECODER;
 
     else if (message->src == GST_OBJECT (self->audio_src))
       element_id = EL_AUDIO_SRC;