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webrtc: Use properties to access the inside of the transceiver object

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This will allow hiding the insides from unsafe application access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/36>
This commit is contained in:
Olivier Crête 2021-04-21 16:27:38 -04:00
parent 5f9ba620ce
commit ba079092f8
1 changed files with 18 additions and 6 deletions
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24
webrtc/sendonly/webrtc-unidirectional-h264.c
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@ -259,22 +259,34 @@ create_receiver_entry (SoupWebsocketConnection * connection)
&transceivers);
g_assert (transceivers != NULL && transceivers->len > 1);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
if (video_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (video_priority);
if (priority)
gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
if (priority) {
GstWebRTCRTPSender *sender;
g_object_get (trans, "sender", &sender, NULL);
gst_webrtc_rtp_sender_set_priority (sender, priority);
g_object_unref (sender);
}
}
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
if (audio_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (audio_priority);
if (priority)
gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
if (priority) {
GstWebRTCRTPSender *sender;
g_object_get (trans, "sender", &sender, NULL);
gst_webrtc_rtp_sender_set_priority (sender, priority);
g_object_unref (sender);
}
}
g_array_unref (transceivers);