mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-13 12:51:16 +00:00
Code Issues 1 Projects Releases Wiki Activity

Update to new promise API

Browse source
This commit is contained in:
Mathieu Duponchelle 2017-11-22 13:15:48 +01:00 committed by Matthew Waters
parent 0c5e799952
commit e5c5767298
2 changed files with 18 additions and 14 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

21
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
View file

@ -252,11 +252,13 @@ on_offer_created (GstPromise * promise, const gchar * peer_id)
{
GstElement *webrtc;
GstWebRTCSessionDescription *offer;
const GstStructure *reply;
g_assert_cmpint (app_state, ==, ROOM_CALL_OFFERING);
g_assert_cmpint (promise->result, ==, GST_PROMISE_RESULT_REPLIED);
gst_structure_get (promise->promise, "offer",
g_assert_cmpint (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
gst_promise_unref (promise);
@ -275,12 +277,12 @@ on_offer_created (GstPromise * promise, const gchar * peer_id)
static void
on_negotiation_needed (GstElement * webrtc, const gchar * peer_id)
{
GstPromise *promise = gst_promise_new ();
GstPromise *promise;
app_state = ROOM_CALL_OFFERING;
g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
gst_promise_set_change_callback (promise,
promise = gst_promise_new_with_change_func (
(GstPromiseChangeFunc) on_offer_created, (gpointer) peer_id, NULL);
g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
}
static void
@ -576,11 +578,13 @@ on_answer_created (GstPromise * promise, const gchar * peer_id)
{
GstElement *webrtc;
GstWebRTCSessionDescription *answer;
const GstStructure *reply;
g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING);
g_assert_cmpint (promise->result, ==, GST_PROMISE_RESULT_REPLIED);
gst_structure_get (promise->promise, "answer",
g_assert_cmpint (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
gst_promise_unref (promise);
@ -630,8 +634,7 @@ handle_sdp_offer (const gchar * peer_id, const gchar * text)
gst_promise_unref (promise);
/* Create an answer that we will send back to the peer */
promise = gst_promise_new ();
gst_promise_set_change_callback (promise,
promise = gst_promise_new_with_change_func (
(GstPromiseChangeFunc) on_answer_created, (gpointer) peer_id, NULL);
g_signal_emit_by_name (webrtc, "create-answer", NULL, promise);

11
webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -219,12 +219,14 @@ static void
on_offer_created (GstPromise * promise, gpointer user_data)
{
GstWebRTCSessionDescription *offer = NULL;
const GstStructure *reply;
gchar *desc;
g_assert (app_state == PEER_CALL_NEGOTIATING);
g_assert (promise->result == GST_PROMISE_RESULT_REPLIED);
gst_structure_get (promise->promise, "offer",
g_assert (gst_promise_wait(promise) == GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
gst_promise_unref (promise);
@ -241,12 +243,11 @@ on_offer_created (GstPromise * promise, gpointer user_data)
static void
on_negotiation_needed (GstElement * element, gpointer user_data)
{
GstPromise *promise = gst_promise_new ();
GstPromise *promise;
app_state = PEER_CALL_NEGOTIATING;
promise = gst_promise_new_with_change_func (on_offer_created, user_data, NULL);;
g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
gst_promise_set_change_callback (promise, on_offer_created, user_data,
NULL);
}
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="