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webrtc: correctly use the pad template

Browse source 
GstHarness uses this for releasing request pads correctly. Fixes
numerous leaks in the webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
This commit is contained in:
Matthew Waters 2020-05-05 17:14:46 +10:00 committed by GStreamer Merge Bot
parent 46176fbcc7
commit d552c6556c
1 changed files with 36 additions and 12 deletions
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48
ext/webrtc/gstwebrtcbin.c
View file

@ -100,6 +100,16 @@ static void _update_need_negotiation (GstWebRTCBin * webrtc);
#define GST_CAT_DEFAULT gst_webrtc_bin_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp"));
enum
{
PROP_PAD_TRANSCEIVER = 1,
@ -284,9 +294,20 @@ gst_webrtc_bin_pad_init (GstWebRTCBinPad * pad)
static GstWebRTCBinPad *
gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
{
GstWebRTCBinPad *pad =
GstWebRTCBinPad *pad;
GstPadTemplate *template;
if (direction == GST_PAD_SINK)
template = gst_static_pad_template_get (&sink_template);
else if (direction == GST_PAD_SRC)
template = gst_static_pad_template_get (&src_template);
else
g_assert_not_reached ();
pad =
g_object_new (gst_webrtc_bin_pad_get_type (), "name", name, "direction",
direction, NULL);
direction, "template", template, NULL);
gst_object_unref (template);
gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
@ -310,16 +331,6 @@ G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
GstWebRTCBinPad * pad);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp"));
enum
{
SIGNAL_0,
@ -4438,6 +4449,12 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
continue;
}
if (!pad->trans) {
GST_LOG_OBJECT (pad, "doesn't have a transceiver");
tmp = tmp->next;
continue;
}
new_dir = pad->trans->direction;
if (new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY &&
new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
@ -5711,11 +5728,18 @@ gst_webrtc_bin_release_pad (GstElement * element, GstPad * pad)
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad);
GST_DEBUG_OBJECT (webrtc, "Releasing %" GST_PTR_FORMAT, webrtc_pad);
/* remove the transceiver from the pad so that subsequent code doesn't use
* a possibly dead transceiver */
PC_LOCK (webrtc);
if (webrtc_pad->trans)
gst_object_unref (webrtc_pad->trans);
webrtc_pad->trans = NULL;
PC_UNLOCK (webrtc);
_remove_pad (webrtc, webrtc_pad);
PC_LOCK (webrtc);
_update_need_negotiation (webrtc);
PC_UNLOCK (webrtc);