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webrtcbin: When creating a new transceiver as part of creating the answer also take its codec preferences into account

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
This commit is contained in:
Sebastian Dröge 2021-06-08 11:39:27 +03:00 committed by GStreamer Marge Bot
parent b7951fb897
commit 4efdb40f43
1 changed files with 29 additions and 13 deletions
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42
ext/webrtc/gstwebrtcbin.c
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@ -3706,34 +3706,50 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
g_assert (answer_caps != NULL);
} else {
/* if no transceiver, then we only receive that stream and respond with
* the exact same caps */
/* FIXME: how to validate that subsequent elements can actually receive
* this payload/format */
* the intersection with the transceivers codec preferences caps */
answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
answer_caps = gst_caps_ref (offer_caps);
}
if (gst_caps_is_empty (answer_caps)) {
GST_WARNING_OBJECT (webrtc, "Could not create caps for media");
if (rtp_trans)
gst_object_unref (rtp_trans);
gst_caps_unref (answer_caps);
gst_caps_unref (offer_caps);
goto rejected;
}
seen_transceivers = g_list_prepend (seen_transceivers, rtp_trans);
if (!rtp_trans) {
GstCaps *trans_caps;
trans = _create_webrtc_transceiver (webrtc, answer_dir, i);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
" for mline %u", trans, i);
trans_caps = _find_codec_preferences (webrtc, rtp_trans, i, error);
if (*error) {
gst_caps_unref (offer_caps);
goto rejected;
}
GST_TRACE_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT
" and %" GST_PTR_FORMAT, offer_caps, trans_caps);
/* FIXME: technically this is a little overreaching as some fields we
* we can deal with not having and/or we may have unrecognized fields
* that we cannot actually support */
if (trans_caps) {
answer_caps = gst_caps_intersect (offer_caps, trans_caps);
gst_caps_unref (trans_caps);
} else {
answer_caps = gst_caps_ref (offer_caps);
}
} else {
trans = WEBRTC_TRANSCEIVER (rtp_trans);
}
if (gst_caps_is_empty (answer_caps)) {
GST_WARNING_OBJECT (webrtc, "Could not create caps for media");
gst_caps_unref (answer_caps);
gst_caps_unref (offer_caps);
goto rejected;
}
if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps))
GST_WARNING_OBJECT (webrtc,
"Trying to change transceiver %d kind from %d to %d",