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webrtcbin: fix check_negotiation computing on caps event
It seems logical that check_negotiation be true if received_caps is *not* equal to the new caps. Also clean up handling of transceivers' ssrc events, as this patch triggered a leaky code path. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
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commit
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2 changed files with 11 additions and 12 deletions
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@ -269,10 +269,11 @@ gst_webrtc_bin_pad_update_ssrc_event (GstWebRTCBinPad * wpad)
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WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans;
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GstPad *pad = GST_PAD (wpad);
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trans->ssrc_event =
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gst_event_take (&trans->ssrc_event,
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gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
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gst_structure_new ("GstWebRtcBinUpdateTos", "ssrc", G_TYPE_UINT,
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trans->current_ssrc, NULL));
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trans->current_ssrc, NULL)));
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gst_pad_send_event (pad, gst_event_ref (trans->ssrc_event));
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}
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}
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@ -302,7 +303,7 @@ gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
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gst_event_parse_caps (event, &caps);
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check_negotiation = (!wpad->received_caps
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|| gst_caps_is_equal (wpad->received_caps, caps));
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|| !gst_caps_is_equal (wpad->received_caps, caps));
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gst_caps_replace (&wpad->received_caps, caps);
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GST_DEBUG_OBJECT (parent,
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@ -468,8 +469,7 @@ gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
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G_ADD_PRIVATE (GstWebRTCBin)
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
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"webrtcbin element");
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);
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"webrtcbin element"););
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enum
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{
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@ -1489,8 +1489,8 @@ _check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
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/* If connection has created any RTCDataChannel's, and no m= section has
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* been negotiated yet for data, return "true". */
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if (webrtc->priv->data_channels->len > 0) {
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if (_message_get_datachannel_index (webrtc->
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current_local_description->sdp) >= G_MAXUINT) {
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if (_message_get_datachannel_index (webrtc->current_local_description->
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sdp) >= G_MAXUINT) {
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GST_LOG_OBJECT (webrtc,
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"no data channel media section and have %u " "transports",
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webrtc->priv->data_channels->len);
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@ -32,7 +32,8 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
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GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
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GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
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"webrtctransceiver", 0, "webrtctransceiver"););
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"webrtctransceiver", 0, "webrtctransceiver");
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);
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#define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
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#define DEFAULT_DO_NACK FALSE
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@ -157,9 +158,7 @@ webrtc_transceiver_finalize (GObject * object)
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gst_caps_replace (&trans->last_configured_caps, NULL);
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if (trans->ssrc_event)
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gst_event_unref (trans->ssrc_event);
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trans->ssrc_event = NULL;
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gst_event_replace (&trans->ssrc_event, NULL);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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