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Update to new promise API
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0c5e799952
commit
e5c5767298
2 changed files with 18 additions and 14 deletions
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@ -252,11 +252,13 @@ on_offer_created (GstPromise * promise, const gchar * peer_id)
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{
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GstElement *webrtc;
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GstWebRTCSessionDescription *offer;
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const GstStructure *reply;
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g_assert_cmpint (app_state, ==, ROOM_CALL_OFFERING);
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g_assert_cmpint (promise->result, ==, GST_PROMISE_RESULT_REPLIED);
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gst_structure_get (promise->promise, "offer",
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g_assert_cmpint (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
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reply = gst_promise_get_reply (promise);
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gst_structure_get (reply, "offer",
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GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
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gst_promise_unref (promise);
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@ -275,12 +277,12 @@ on_offer_created (GstPromise * promise, const gchar * peer_id)
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static void
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on_negotiation_needed (GstElement * webrtc, const gchar * peer_id)
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{
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GstPromise *promise = gst_promise_new ();
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GstPromise *promise;
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app_state = ROOM_CALL_OFFERING;
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g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
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gst_promise_set_change_callback (promise,
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promise = gst_promise_new_with_change_func (
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(GstPromiseChangeFunc) on_offer_created, (gpointer) peer_id, NULL);
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g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
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}
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static void
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@ -576,11 +578,13 @@ on_answer_created (GstPromise * promise, const gchar * peer_id)
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{
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GstElement *webrtc;
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GstWebRTCSessionDescription *answer;
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const GstStructure *reply;
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g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING);
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g_assert_cmpint (promise->result, ==, GST_PROMISE_RESULT_REPLIED);
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gst_structure_get (promise->promise, "answer",
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g_assert_cmpint (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
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reply = gst_promise_get_reply (promise);
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gst_structure_get (reply, "answer",
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GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
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gst_promise_unref (promise);
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@ -630,8 +634,7 @@ handle_sdp_offer (const gchar * peer_id, const gchar * text)
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gst_promise_unref (promise);
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/* Create an answer that we will send back to the peer */
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promise = gst_promise_new ();
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gst_promise_set_change_callback (promise,
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promise = gst_promise_new_with_change_func (
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(GstPromiseChangeFunc) on_answer_created, (gpointer) peer_id, NULL);
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g_signal_emit_by_name (webrtc, "create-answer", NULL, promise);
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@ -219,12 +219,14 @@ static void
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on_offer_created (GstPromise * promise, gpointer user_data)
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{
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GstWebRTCSessionDescription *offer = NULL;
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const GstStructure *reply;
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gchar *desc;
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g_assert (app_state == PEER_CALL_NEGOTIATING);
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g_assert (promise->result == GST_PROMISE_RESULT_REPLIED);
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gst_structure_get (promise->promise, "offer",
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g_assert (gst_promise_wait(promise) == GST_PROMISE_RESULT_REPLIED);
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reply = gst_promise_get_reply (promise);
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gst_structure_get (reply, "offer",
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GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
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gst_promise_unref (promise);
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@ -241,12 +243,11 @@ on_offer_created (GstPromise * promise, gpointer user_data)
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static void
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on_negotiation_needed (GstElement * element, gpointer user_data)
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{
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GstPromise *promise = gst_promise_new ();
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GstPromise *promise;
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app_state = PEER_CALL_NEGOTIATING;
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promise = gst_promise_new_with_change_func (on_offer_created, user_data, NULL);;
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g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
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gst_promise_set_change_callback (promise, on_offer_created, user_data,
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NULL);
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}
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#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
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