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webrtcbin: fix check_negotiation computing on caps event

Browse source 
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>
This commit is contained in:
Mathieu Duponchelle 2021-10-27 00:20:57 +02:00 committed by GStreamer Marge Bot
parent be0b5c54fd
commit 303c8025c6
2 changed files with 11 additions and 12 deletions
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16
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -269,10 +269,11 @@ gst_webrtc_bin_pad_update_ssrc_event (GstWebRTCBinPad * wpad)
WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans;
GstPad *pad = GST_PAD (wpad);
trans->ssrc_event =
gst_event_take (&trans->ssrc_event,
gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
gst_structure_new ("GstWebRtcBinUpdateTos", "ssrc", G_TYPE_UINT,
trans->current_ssrc, NULL));
gst_structure_new ("GstWebRtcBinUpdateTos", "ssrc", G_TYPE_UINT,
trans->current_ssrc, NULL)));
gst_pad_send_event (pad, gst_event_ref (trans->ssrc_event));
}
}
@ -302,7 +303,7 @@ gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
gst_event_parse_caps (event, &caps);
check_negotiation = (!wpad->received_caps
|| gst_caps_is_equal (wpad->received_caps, caps));
|| !gst_caps_is_equal (wpad->received_caps, caps));
gst_caps_replace (&wpad->received_caps, caps);
GST_DEBUG_OBJECT (parent,
@ -468,8 +469,7 @@ gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
G_ADD_PRIVATE (GstWebRTCBin)
GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
"webrtcbin element");
);
"webrtcbin element"););
enum
{
@ -1489,8 +1489,8 @@ _check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
/* If connection has created any RTCDataChannel's, and no m= section has
* been negotiated yet for data, return "true". */
if (webrtc->priv->data_channels->len > 0) {
if (_message_get_datachannel_index (webrtc->
current_local_description->sdp) >= G_MAXUINT) {
if (_message_get_datachannel_index (webrtc->current_local_description->
sdp) >= G_MAXUINT) {
GST_LOG_OBJECT (webrtc,
"no data channel media section and have %u " "transports",
webrtc->priv->data_channels->len);

7
subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c
View file

@ -32,7 +32,8 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
"webrtctransceiver", 0, "webrtctransceiver"););
"webrtctransceiver", 0, "webrtctransceiver");
);
#define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
#define DEFAULT_DO_NACK FALSE
@ -157,9 +158,7 @@ webrtc_transceiver_finalize (GObject * object)
gst_caps_replace (&trans->last_configured_caps, NULL);
if (trans->ssrc_event)
gst_event_unref (trans->ssrc_event);
trans->ssrc_event = NULL;
gst_event_replace (&trans->ssrc_event, NULL);
G_OBJECT_CLASS (parent_class)->finalize (object);
}