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Require GStreamer 1.18 for voip support

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This commit is contained in:
trilene 2021-02-20 17:14:22 -05:00
parent 8ccd2abc6a
commit c461c0aac0
9 changed files with 3 additions and 97 deletions
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46
src/WebRTCSession.cpp
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@ -174,7 +174,6 @@ createAnswer(GstPromise *promise, gpointer webrtc)
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
}
#if GST_CHECK_VERSION(1, 18, 0)
void
iceGatheringStateChanged(GstElement *webrtc,
GParamSpec *pspec G_GNUC_UNUSED,
@ -194,23 +193,6 @@ iceGatheringStateChanged(GstElement *webrtc,
}
}
#else
gboolean
onICEGatheringCompletion(gpointer timerid)
{
*(guint *)(timerid) = 0;
if (WebRTCSession::instance().isOffering()) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
} else {
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
}
return FALSE;
}
#endif
void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
guint mlineIndex,
@ -218,28 +200,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
gpointer G_GNUC_UNUSED)
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
#if GST_CHECK_VERSION(1, 18, 0)
localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
return;
#else
if (WebRTCSession::instance().state() >= State::OFFERSENT) {
emit WebRTCSession::instance().newICECandidate(
{std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
return;
}
localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18.
// Use a 1s timeout in the meantime
static guint timerid = 0;
if (timerid)
g_source_remove(timerid);
timerid = g_timeout_add(1000, onICEGatheringCompletion, &timerid);
#endif
}
void
@ -328,7 +289,6 @@ testPacketLoss(gpointer G_GNUC_UNUSED)
return FALSE;
}
#if GST_CHECK_VERSION(1, 18, 0)
void
setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointer G_GNUC_UNUSED)
{
@ -337,7 +297,6 @@ setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointe
"rtpvp8depay"))
g_object_set(element, "wait-for-keyframe", TRUE, nullptr);
}
#endif
GstElement *
newAudioSinkChain(GstElement *pipe)
@ -537,9 +496,7 @@ addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
// hardware decoding needs investigation; eg rendering fails if vaapi plugin installed
g_object_set(decodebin, "force-sw-decoders", TRUE, nullptr);
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
#if GST_CHECK_VERSION(1, 18, 0)
g_signal_connect(decodebin, "element-added", G_CALLBACK(setWaitForKeyFrame), nullptr);
#endif
gst_bin_add(GST_BIN(pipe), decodebin);
gst_element_sync_state_with_parent(decodebin);
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
@ -810,11 +767,10 @@ WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
#if GST_CHECK_VERSION(1, 18, 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
gst_object_unref(webrtc_);