diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index 2248fb1a..e9822f7d 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -176,7 +176,7 @@ createAnswer(GstPromise *promise, gpointer webrtc)
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
}
-#if GST_CHECK_VERSION(1, 17, 0)
+#if GST_CHECK_VERSION(1, 18, 0)
void
iceGatheringStateChanged(GstElement *webrtc,
GParamSpec *pspec G_GNUC_UNUSED,
@@ -223,6 +223,10 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
+#if GST_CHECK_VERSION(1, 18, 0)
+ localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
+ return;
+#else
if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
emit WebRTCSession::instance().newICECandidate(
{"audio", (uint16_t)mlineIndex, candidate});
@@ -232,9 +236,8 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
- // GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.17.
+ // GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18.
// Use a 100ms timeout in the meantime
-#if !GST_CHECK_VERSION(1, 17, 0)
static guint timerid = 0;
if (timerid)
g_source_remove(timerid);
@@ -282,11 +285,11 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
+ gst_element_link_many(queue, convert, resample, sink, nullptr);
gst_element_sync_state_with_parent(queue);
gst_element_sync_state_with_parent(convert);
gst_element_sync_state_with_parent(resample);
gst_element_sync_state_with_parent(sink);
- gst_element_link_many(queue, convert, resample, sink, nullptr);
queuepad = gst_element_get_static_pad(queue, "sink");
}
@@ -423,8 +426,12 @@ WebRTCSession::acceptICECandidates(
for (const auto &c : candidates) {
nhlog::ui()->debug(
"WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
- g_signal_emit_by_name(
- webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
+ if (!c.candidate.empty()) {
+ g_signal_emit_by_name(webrtc_,
+ "add-ice-candidate",
+ c.sdpMLineIndex,
+ c.candidate.c_str());
+ }
}
}
}
@@ -471,7 +478,7 @@ WebRTCSession::startPipeline(int opusPayloadType)
gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
-#if GST_CHECK_VERSION(1, 17, 0)
+#if GST_CHECK_VERSION(1, 18, 0)
// capture ICE gathering completion
g_signal_connect(
webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
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