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GStreamer v1.18.0 released

This commit is contained in:
trilene 2020-09-10 14:34:10 -04:00
parent 7d2844b2b0
commit b6563d9ffe
1 changed files with 4 additions and 4 deletions
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8
src/WebRTCSession.cpp
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@ -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,
@ -223,7 +223,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
#if GST_CHECK_VERSION(1, 17, 0)
#if GST_CHECK_VERSION(1, 18, 0)
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
return;
#else
@ -236,7 +236,7 @@ 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
static guint timerid = 0;
if (timerid)
@ -474,7 +474,7 @@ 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);