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Conditionally compile against upcoming GStreamer release

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This commit is contained in:
trilene 2020-08-02 22:27:05 -04:00
parent b86711a388
commit 02dfc8039f
1 changed files with 34 additions and 2 deletions
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36
src/WebRTCSession.cpp
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@ -169,6 +169,30 @@ createAnswer(GstPromise *promise, gpointer webrtc)
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise); g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
} }
#if GST_CHECK_VERSION(1, 17, 0)
void
iceGatheringStateChanged(GstElement *webrtc,
GParamSpec *pspec G_GNUC_UNUSED,
gpointer user_data G_GNUC_UNUSED)
{
GstWebRTCICEGatheringState newState;
g_object_get(webrtc, "ice-gathering-state", &newState, nullptr);
if (newState == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
nhlog::ui()->debug("WebRTC: GstWebRTCICEGatheringState -> Complete");
if (isoffering_) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(
WebRTCSession::State::OFFERSENT);
} else {
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(
WebRTCSession::State::ANSWERSENT);
}
}
}
#else
gboolean gboolean
onICEGatheringCompletion(gpointer timerid) onICEGatheringCompletion(gpointer timerid)
{ {
@ -182,6 +206,7 @@ onICEGatheringCompletion(gpointer timerid)
} }
return FALSE; return FALSE;
} }
#endif
void void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
@ -200,13 +225,15 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate}); localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers // GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18. Use a 100ms timeout in // GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.17.
// the meantime // Use a 100ms timeout in the meantime
#if !GST_CHECK_VERSION(1, 17, 0)
static guint timerid = 0; static guint timerid = 0;
if (timerid) if (timerid)
g_source_remove(timerid); g_source_remove(timerid);
timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid); timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
#endif
} }
void void
@ -437,6 +464,11 @@ WebRTCSession::startPipeline(int opusPayloadType)
gst_element_set_state(pipe_, GST_STATE_READY); gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_); g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
#if GST_CHECK_VERSION(1, 17, 0)
// capture ICE gathering completion
g_signal_connect(
webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
#endif
// webrtcbin lifetime is the same as that of the pipeline // webrtcbin lifetime is the same as that of the pipeline
gst_object_unref(webrtc_); gst_object_unref(webrtc_);