From 473293b6a5d06e670065dd4d35af0e456621e9e6 Mon Sep 17 00:00:00 2001 From: trilene Date: Sat, 22 Aug 2020 08:18:42 -0400 Subject: Under GStreamer >= 1.17 gather all candidates before sending offer/answer --- src/WebRTCSession.cpp | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'src/WebRTCSession.cpp') diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index 2248fb1a..b4e7eeb3 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -223,18 +223,19 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, { nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate); +#if GST_CHECK_VERSION(1, 17, 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}); return; } - 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. // Use a 100ms timeout in the meantime -#if !GST_CHECK_VERSION(1, 17, 0) static guint timerid = 0; if (timerid) g_source_remove(timerid); @@ -447,6 +448,7 @@ WebRTCSession::startPipeline(int opusPayloadType) g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr); } + for (const auto &uri : turnServers_) { nhlog::ui()->info("WebRTC: setting TURN server: {}", uri); gboolean udata; -- cgit 1.5.1 From 67a6ab401b90445169ae35a4107a9973474f8073 Mon Sep 17 00:00:00 2001 From: trilene Date: Fri, 28 Aug 2020 10:49:39 -0400 Subject: Link GStreamer elements before syncing state --- src/WebRTCSession.cpp | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'src/WebRTCSession.cpp') diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index b4e7eeb3..f5dc49d8 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -283,11 +283,11 @@ 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"); } @@ -448,7 +448,6 @@ WebRTCSession::startPipeline(int opusPayloadType) g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr); } - for (const auto &uri : turnServers_) { nhlog::ui()->info("WebRTC: setting TURN server: {}", uri); gboolean udata; -- cgit 1.5.1