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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.
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commit
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1 changed files with 11 additions and 11 deletions
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@ -206,22 +206,25 @@ set_state (CallsSipMediaPipeline *self,
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static void
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static void
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check_element_maps (CallsSipMediaPipeline *self)
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check_element_maps (CallsSipMediaPipeline *self)
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{
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{
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uint all_rtp_elements;
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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/* TODO take encryption into account */
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all_rtp_elements = self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP;
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if (self->element_map_playing == EL_ALL_RTP) {
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if (self->element_map_playing == all_rtp_elements) {
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g_debug ("All pipeline elements are playing");
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g_debug ("All pipeline elements are playing");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
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return;
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return;
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}
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}
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if (self->element_map_paused == EL_ALL_RTP) {
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if (self->element_map_paused == all_rtp_elements) {
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g_debug ("All pipeline elements are paused");
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g_debug ("All pipeline elements are paused");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
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return;
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return;
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}
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}
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if (self->element_map_stopped == EL_ALL_RTP) {
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if (self->element_map_stopped == all_rtp_elements) {
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g_debug ("All pipeline elements are stopped");
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g_debug ("All pipeline elements are stopped");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
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return;
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return;
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@ -319,13 +322,10 @@ on_bus_message (GstBus *bus,
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else if (message->src == GST_OBJECT (self->rtcp_sink))
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else if (message->src == GST_OBJECT (self->rtcp_sink))
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element_id = EL_RTCP_SINK;
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element_id = EL_RTCP_SINK;
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/* TODO srtp encryption
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else if (message->src == GST_OBJECT (self->srtpenc))
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else if (message->src == GST_OBJECT (self->srtpenc))
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element_id = EL_SRTP_ENCODER;
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element_id = EL_SRTP_ENCODER;
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else if (message->src == GST_OBJECT (self->srtpdec))
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else if (message->src == GST_OBJECT (self->srtpdec))
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element_id = EL_SRTP_DECODER;
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element_id = EL_SRTP_DECODER;
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*/
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else if (message->src == GST_OBJECT (self->audio_src))
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else if (message->src == GST_OBJECT (self->audio_src))
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element_id = EL_AUDIO_SRC;
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element_id = EL_AUDIO_SRC;
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