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webrtcbin: Simplify answer_caps intersection code a little

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-04-12 17:54:31 -04:00
parent 474c4bf08f
commit d67dcb2227
1 changed files with 10 additions and 16 deletions
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26
ext/webrtc/gstwebrtcbin.c
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@ -3540,25 +3540,19 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
* that we cannot actually support */ * that we cannot actually support */
if (trans_caps) { if (trans_caps) {
answer_caps = gst_caps_intersect (offer_caps, trans_caps); answer_caps = gst_caps_intersect (offer_caps, trans_caps);
if (answer_caps && !gst_caps_is_empty (answer_caps)) { gst_caps_unref (trans_caps);
GST_LOG_OBJECT (webrtc, if (answer_caps) {
"found compatible transceiver %" GST_PTR_FORMAT if (!gst_caps_is_empty (answer_caps)) {
" for offer media %u", rtp_trans, i); GST_LOG_OBJECT (webrtc,
if (trans_caps) "found compatible transceiver %" GST_PTR_FORMAT
gst_caps_unref (trans_caps); " for offer media %u", rtp_trans, i);
break; break;
} else {
if (answer_caps) {
gst_caps_unref (answer_caps);
answer_caps = NULL;
} }
if (trans_caps) gst_caps_unref (answer_caps);
gst_caps_unref (trans_caps); answer_caps = NULL;
rtp_trans = NULL;
} }
} else {
rtp_trans = NULL;
} }
rtp_trans = NULL;
} }
} }