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Merge pull request #278 from trilene/voip
Fix earlier commit + GStreamer v1.18 released
This commit is contained in:
commit
b37534aa53
1 changed files with 6 additions and 4 deletions
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@ -176,7 +176,7 @@ createAnswer(GstPromise *promise, gpointer webrtc)
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g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
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}
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#if GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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void
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iceGatheringStateChanged(GstElement *webrtc,
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GParamSpec *pspec G_GNUC_UNUSED,
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@ -223,7 +223,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
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{
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nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
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#if GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
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return;
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#else
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@ -233,8 +233,10 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
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return;
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}
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localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
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// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
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// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.17.
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// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18.
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// Use a 100ms timeout in the meantime
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static guint timerid = 0;
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if (timerid)
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@ -472,7 +474,7 @@ WebRTCSession::startPipeline(int opusPayloadType)
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gst_element_set_state(pipe_, GST_STATE_READY);
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g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
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#if GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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// capture ICE gathering completion
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g_signal_connect(
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webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
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