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webrtc: indent sources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
This commit is contained in:
Matthew Waters
parent
e1c3dad258
commit
204945b902
3 changed files with 140 additions and 123 deletions
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@ -43,24 +43,25 @@ GST_DEBUG_CATEGORY_STATIC (debug_category);
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#define GET_CUSTOM_DATA(env, thiz, fieldID) (WebRTC *)(gintptr)(*env)->GetLongField (env, thiz, fieldID)
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#define SET_CUSTOM_DATA(env, thiz, fieldID, data) (*env)->SetLongField (env, thiz, fieldID, (jlong)(gintptr)data)
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enum AppState {
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APP_STATE_UNKNOWN = 0,
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APP_STATE_ERROR = 1, /* generic error */
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SERVER_CONNECTING = 1000,
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SERVER_CONNECTION_ERROR,
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SERVER_CONNECTED, /* Ready to register */
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SERVER_REGISTERING = 2000,
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SERVER_REGISTRATION_ERROR,
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SERVER_REGISTERED, /* Ready to call a peer */
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SERVER_CLOSED, /* server connection closed by us or the server */
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PEER_CONNECTING = 3000,
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PEER_CONNECTION_ERROR,
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PEER_CONNECTED,
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PEER_CALL_NEGOTIATING = 4000,
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PEER_CALL_STARTED,
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PEER_CALL_STOPPING,
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PEER_CALL_STOPPED,
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PEER_CALL_ERROR,
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enum AppState
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{
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APP_STATE_UNKNOWN = 0,
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APP_STATE_ERROR = 1, /* generic error */
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SERVER_CONNECTING = 1000,
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SERVER_CONNECTION_ERROR,
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SERVER_CONNECTED, /* Ready to register */
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SERVER_REGISTERING = 2000,
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SERVER_REGISTRATION_ERROR,
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SERVER_REGISTERED, /* Ready to call a peer */
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SERVER_CLOSED, /* server connection closed by us or the server */
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PEER_CONNECTING = 3000,
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PEER_CONNECTION_ERROR,
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PEER_CONNECTED,
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PEER_CALL_NEGOTIATING = 4000,
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PEER_CALL_STARTED,
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PEER_CALL_STOPPING,
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PEER_CALL_STOPPED,
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PEER_CALL_ERROR,
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};
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typedef struct _WebRTC
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@ -115,7 +116,7 @@ cleanup_and_quit_loop (WebRTC * webrtc, const gchar * msg, enum AppState state)
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return G_SOURCE_REMOVE;
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}
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static gchar*
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static gchar *
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get_string_from_json_object (JsonObject * object)
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{
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JsonNode *root;
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@ -135,8 +136,8 @@ get_string_from_json_object (JsonObject * object)
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}
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static GstElement *
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handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
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const char * sink_name)
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handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
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const char *sink_name)
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{
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GstPad *qpad;
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GstElement *q, *conv, *sink;
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@ -176,14 +177,14 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
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static void
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on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
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WebRTC * webrtc)
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WebRTC * webrtc)
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{
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GstCaps *caps;
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const gchar *name;
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if (!gst_pad_has_current_caps (pad)) {
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g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
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GST_PAD_NAME (pad));
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GST_PAD_NAME (pad));
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return;
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}
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@ -191,11 +192,13 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
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name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
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if (g_str_has_prefix (name, "video")) {
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GstElement *sink = handle_media_stream (pad, webrtc->pipe, "videoconvert", "glimagesink");
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GstElement *sink =
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handle_media_stream (pad, webrtc->pipe, "videoconvert", "glimagesink");
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if (webrtc->video_sink == NULL) {
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webrtc->video_sink = sink;
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if (webrtc->native_window)
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gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (sink), (gpointer) webrtc->native_window);
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gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (sink),
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(gpointer) webrtc->native_window);
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}
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} else if (g_str_has_prefix (name, "audio")) {
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handle_media_stream (pad, webrtc->pipe, "audioconvert", "autoaudiosink");
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@ -216,21 +219,22 @@ on_incoming_stream (GstElement * webrtcbin, GstPad * pad, WebRTC * webrtc)
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decodebin = gst_element_factory_make ("decodebin", NULL);
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g_signal_connect (decodebin, "pad-added",
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G_CALLBACK (on_incoming_decodebin_stream), webrtc);
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G_CALLBACK (on_incoming_decodebin_stream), webrtc);
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gst_bin_add (GST_BIN (webrtc->pipe), decodebin);
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gst_element_sync_state_with_parent (decodebin);
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gst_element_link (webrtcbin, decodebin);
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}
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static void
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send_ice_candidate_message (GstElement * webrtcbin G_GNUC_UNUSED, guint mlineindex,
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gchar * candidate, WebRTC * webrtc)
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send_ice_candidate_message (GstElement * webrtcbin G_GNUC_UNUSED,
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guint mlineindex, gchar * candidate, WebRTC * webrtc)
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{
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gchar *text;
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JsonObject *ice, *msg;
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if (webrtc->app_state < PEER_CALL_NEGOTIATING) {
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cleanup_and_quit_loop (webrtc, "Can't send ICE, not in call", APP_STATE_ERROR);
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cleanup_and_quit_loop (webrtc, "Can't send ICE, not in call",
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APP_STATE_ERROR);
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return;
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}
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@ -253,7 +257,8 @@ send_sdp_offer (WebRTC * webrtc, GstWebRTCSessionDescription * offer)
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JsonObject *msg, *sdp;
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if (webrtc->app_state < PEER_CALL_NEGOTIATING) {
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cleanup_and_quit_loop (webrtc, "Can't send offer, not in call", APP_STATE_ERROR);
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cleanup_and_quit_loop (webrtc, "Can't send offer, not in call",
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APP_STATE_ERROR);
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return;
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}
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@ -283,14 +288,15 @@ on_offer_created (GstPromise * promise, WebRTC * webrtc)
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g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING);
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g_assert (gst_promise_wait(promise) == GST_PROMISE_RESULT_REPLIED);
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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_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
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gst_promise_unref (promise);
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promise = gst_promise_new ();
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g_signal_emit_by_name (webrtc->webrtcbin, "set-local-description", offer, promise);
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g_signal_emit_by_name (webrtc->webrtcbin, "set-local-description", offer,
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promise);
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gst_promise_interrupt (promise);
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gst_promise_unref (promise);
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@ -333,13 +339,12 @@ start_pipeline (WebRTC * webrtc)
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GstPad *pad;
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webrtc->pipe =
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gst_parse_launch ("webrtcbin name=sendrecv "
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"ahcsrc device-facing=front ! video/x-raw,width=[320,1280] ! queue max-size-buffers=1 ! videoconvert ! "
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"vp8enc keyframe-max-dist=30 deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit mtu=1300 ! "
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"queue max-size-time=300000000 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
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"openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! "
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"queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ",
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&error);
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gst_parse_launch ("webrtcbin name=sendrecv "
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"ahcsrc device-facing=front ! video/x-raw,width=[320,1280] ! queue max-size-buffers=1 ! videoconvert ! "
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"vp8enc keyframe-max-dist=30 deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit mtu=1300 ! "
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"queue max-size-time=300000000 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
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"openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! "
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"queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ", &error);
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if (error) {
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g_printerr ("Failed to parse launch: %s\n", error->message);
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@ -354,15 +359,15 @@ start_pipeline (WebRTC * webrtc)
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/* This is the gstwebrtc entry point where we create the offer and so on. It
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* will be called when the pipeline goes to PLAYING. */
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g_signal_connect (webrtc->webrtcbin, "on-negotiation-needed",
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G_CALLBACK (on_negotiation_needed), webrtc);
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G_CALLBACK (on_negotiation_needed), webrtc);
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/* We need to transmit this ICE candidate to the browser via the websockets
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* signalling server. Incoming ice candidates from the browser need to be
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* added by us too, see on_server_message() */
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g_signal_connect (webrtc->webrtcbin, "on-ice-candidate",
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G_CALLBACK (send_ice_candidate_message), webrtc);
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G_CALLBACK (send_ice_candidate_message), webrtc);
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/* Incoming streams will be exposed via this signal */
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g_signal_connect (webrtc->webrtcbin, "pad-added", G_CALLBACK (on_incoming_stream),
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webrtc);
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g_signal_connect (webrtc->webrtcbin, "pad-added",
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G_CALLBACK (on_incoming_stream), webrtc);
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/* Lifetime is the same as the pipeline itself */
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gst_object_unref (webrtc->webrtcbin);
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@ -425,8 +430,7 @@ register_with_server (WebRTC * webrtc)
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}
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static void
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on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
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WebRTC * webrtc)
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on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED, WebRTC * webrtc)
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{
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webrtc->app_state = SERVER_CLOSED;
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cleanup_and_quit_loop (webrtc, "Server connection closed", 0);
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@ -435,7 +439,7 @@ on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
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/* One mega message handler for our asynchronous calling mechanism */
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static void
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on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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GBytes * message, WebRTC * webrtc)
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GBytes * message, WebRTC * webrtc)
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{
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gsize size;
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gchar *text, *data;
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@ -443,14 +447,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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switch (type) {
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case SOUP_WEBSOCKET_DATA_BINARY:
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g_printerr ("Received unknown binary message, ignoring\n");
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g_bytes_unref (message);
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return;
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g_bytes_unref (message);
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return;
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case SOUP_WEBSOCKET_DATA_TEXT:
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data = g_bytes_unref_to_data (message, &size);
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/* Convert to NULL-terminated string */
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text = g_strndup (data, size);
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g_free (data);
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break;
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/* Convert to NULL-terminated string */
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text = g_strndup (data, size);
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g_free (data);
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break;
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default:
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g_assert_not_reached ();
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}
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@ -458,22 +462,23 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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/* Server has accepted our registration, we are ready to send commands */
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if (g_strcmp0 (text, "HELLO") == 0) {
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if (webrtc->app_state != SERVER_REGISTERING) {
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cleanup_and_quit_loop (webrtc, "ERROR: Received HELLO when not registering",
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APP_STATE_ERROR);
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cleanup_and_quit_loop (webrtc,
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"ERROR: Received HELLO when not registering", APP_STATE_ERROR);
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goto out;
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}
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webrtc->app_state = SERVER_REGISTERED;
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g_print ("Registered with server\n");
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/* Ask signalling server to connect us with a specific peer */
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if (!setup_call (webrtc)) {
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cleanup_and_quit_loop (webrtc, "ERROR: Failed to setup call", PEER_CALL_ERROR);
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cleanup_and_quit_loop (webrtc, "ERROR: Failed to setup call",
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PEER_CALL_ERROR);
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goto out;
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}
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/* Call has been setup by the server, now we can start negotiation */
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} else if (g_strcmp0 (text, "SESSION_OK") == 0) {
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if (webrtc->app_state != PEER_CONNECTING) {
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cleanup_and_quit_loop (webrtc, "ERROR: Received SESSION_OK when not calling",
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PEER_CONNECTION_ERROR);
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cleanup_and_quit_loop (webrtc,
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"ERROR: Received SESSION_OK when not calling", PEER_CONNECTION_ERROR);
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goto out;
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}
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@ -481,23 +486,23 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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/* Start negotiation (exchange SDP and ICE candidates) */
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if (!start_pipeline (webrtc))
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cleanup_and_quit_loop (webrtc, "ERROR: failed to start pipeline",
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PEER_CALL_ERROR);
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PEER_CALL_ERROR);
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/* Handle errors */
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} else if (g_str_has_prefix (text, "ERROR")) {
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switch (webrtc->app_state) {
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case SERVER_CONNECTING:
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webrtc->app_state = SERVER_CONNECTION_ERROR;
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break;
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break;
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case SERVER_REGISTERING:
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webrtc->app_state = SERVER_REGISTRATION_ERROR;
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break;
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break;
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case PEER_CONNECTING:
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webrtc->app_state = PEER_CONNECTION_ERROR;
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break;
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break;
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case PEER_CONNECTED:
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case PEER_CALL_NEGOTIATING:
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webrtc->app_state = PEER_CALL_ERROR;
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break;
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break;
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default:
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webrtc->app_state = APP_STATE_ERROR;
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}
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@ -541,7 +546,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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* See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to
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* handle offers from peers and reply with answers using webrtcbin. */
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g_assert_cmpstr (json_object_get_string_member (object, "type"), ==,
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"answer");
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"answer");
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text = json_object_get_string_member (object, "sdp");
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@ -554,14 +559,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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g_assert (ret == GST_SDP_OK);
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answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
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sdp);
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sdp);
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g_assert (answer);
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/* Set remote description on our pipeline */
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{
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GstPromise *promise = gst_promise_new ();
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g_signal_emit_by_name (webrtc->webrtcbin, "set-remote-description", answer,
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promise);
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g_signal_emit_by_name (webrtc->webrtcbin, "set-remote-description",
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answer, promise);
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gst_promise_interrupt (promise);
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gst_promise_unref (promise);
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}
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@ -577,25 +582,25 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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sdpmlineindex = json_object_get_int_member (ice, "sdpMLineIndex");
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/* Add ice candidate sent by remote peer */
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g_signal_emit_by_name (webrtc->webrtcbin, "add-ice-candidate", sdpmlineindex,
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candidate);
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g_signal_emit_by_name (webrtc->webrtcbin, "add-ice-candidate",
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sdpmlineindex, candidate);
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} else {
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g_printerr ("Ignoring unknown JSON message:\n%s\n", text);
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}
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g_object_unref (parser);
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}
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out:
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out:
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g_free (text);
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}
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static void
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on_server_connected (SoupSession * session, GAsyncResult * res,
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WebRTC * webrtc)
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on_server_connected (SoupSession * session, GAsyncResult * res, WebRTC * webrtc)
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{
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GError *error = NULL;
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webrtc->ws_conn = soup_session_websocket_connect_finish (session, res, &error);
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webrtc->ws_conn =
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soup_session_websocket_connect_finish (session, res, &error);
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if (error) {
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cleanup_and_quit_loop (webrtc, error->message, SERVER_CONNECTION_ERROR);
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g_error_free (error);
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@ -607,8 +612,10 @@ on_server_connected (SoupSession * session, GAsyncResult * res,
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webrtc->app_state = SERVER_CONNECTED;
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g_print ("Connected to signalling server\n");
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g_signal_connect (webrtc->ws_conn, "closed", G_CALLBACK (on_server_closed), webrtc);
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g_signal_connect (webrtc->ws_conn, "message", G_CALLBACK (on_server_message), webrtc);
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g_signal_connect (webrtc->ws_conn, "closed", G_CALLBACK (on_server_closed),
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webrtc);
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g_signal_connect (webrtc->ws_conn, "message", G_CALLBACK (on_server_message),
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webrtc);
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/* Register with the server so it knows about us and can accept commands */
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register_with_server (webrtc);
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@ -623,16 +630,16 @@ connect_to_websocket_server_async (WebRTC * webrtc)
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SoupLogger *logger;
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SoupMessage *message;
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SoupSession *session;
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const char *https_aliases[] = {"wss", NULL};
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const char *https_aliases[] = { "wss", NULL };
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const gchar *ca_certs;
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ca_certs = g_getenv("CA_CERTIFICATES");
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ca_certs = g_getenv ("CA_CERTIFICATES");
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g_assert (ca_certs != NULL);
|
||||
g_print ("ca-certificates %s", ca_certs);
|
||||
session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, FALSE,
|
||||
// SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
|
||||
SOUP_SESSION_SSL_CA_FILE, ca_certs,
|
||||
SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
|
||||
// SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
|
||||
SOUP_SESSION_SSL_CA_FILE, ca_certs,
|
||||
SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
|
||||
|
||||
logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1);
|
||||
soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger));
|
||||
|
|
@ -644,7 +651,7 @@ connect_to_websocket_server_async (WebRTC * webrtc)
|
|||
|
||||
/* Once connected, we will register */
|
||||
soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
|
||||
(GAsyncReadyCallback) on_server_connected, webrtc);
|
||||
(GAsyncReadyCallback) on_server_connected, webrtc);
|
||||
webrtc->app_state = SERVER_CONNECTING;
|
||||
|
||||
return G_SOURCE_REMOVE;
|
||||
|
|
@ -708,7 +715,7 @@ native_end_call (JNIEnv * env, jobject thiz)
|
|||
if (webrtc->loop) {
|
||||
GThread *thread = webrtc->thread;
|
||||
|
||||
GST_INFO("Ending current call");
|
||||
GST_INFO ("Ending current call");
|
||||
cleanup_and_quit_loop (webrtc, NULL, 0);
|
||||
webrtc->thread = NULL;
|
||||
g_mutex_unlock (&webrtc->lock);
|
||||
|
|
@ -729,14 +736,15 @@ static gpointer
|
|||
_call_thread (WebRTC * webrtc)
|
||||
{
|
||||
GMainContext *context = NULL;
|
||||
JNIEnv *env = attach_current_thread();
|
||||
JNIEnv *env = attach_current_thread ();
|
||||
|
||||
g_mutex_lock (&webrtc->lock);
|
||||
|
||||
context = g_main_context_new ();
|
||||
webrtc->loop = g_main_loop_new (context, FALSE);
|
||||
g_main_context_invoke (context, (GSourceFunc) _unlock_mutex, &webrtc->lock);
|
||||
g_main_context_invoke (context, (GSourceFunc) connect_to_websocket_server_async, webrtc);
|
||||
g_main_context_invoke (context,
|
||||
(GSourceFunc) connect_to_websocket_server_async, webrtc);
|
||||
g_main_context_push_thread_default (context);
|
||||
g_cond_broadcast (&webrtc->cond);
|
||||
g_main_loop_run (webrtc->loop);
|
||||
|
|
@ -748,7 +756,7 @@ _call_thread (WebRTC * webrtc)
|
|||
}
|
||||
|
||||
static void
|
||||
native_call_other_party(JNIEnv * env, jobject thiz)
|
||||
native_call_other_party (JNIEnv * env, jobject thiz)
|
||||
{
|
||||
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
|
||||
|
|
@ -758,9 +766,9 @@ native_call_other_party(JNIEnv * env, jobject thiz)
|
|||
if (webrtc->thread)
|
||||
native_end_call (env, thiz);
|
||||
|
||||
GST_INFO("calling other party");
|
||||
GST_INFO ("calling other party");
|
||||
|
||||
webrtc->thread = g_thread_new("webrtc", (GThreadFunc) _call_thread, webrtc);
|
||||
webrtc->thread = g_thread_new ("webrtc", (GThreadFunc) _call_thread, webrtc);
|
||||
g_mutex_lock (&webrtc->lock);
|
||||
while (!webrtc->loop)
|
||||
g_cond_wait (&webrtc->cond, &webrtc->lock);
|
||||
|
|
@ -814,14 +822,13 @@ native_class_init (JNIEnv * env, jclass klass)
|
|||
__android_log_print (ANDROID_LOG_ERROR, "GstPlayer", "%s", message);
|
||||
(*env)->ThrowNew (env, exception_class, message);
|
||||
}
|
||||
|
||||
//gst_debug_set_threshold_from_string ("gl*:7", FALSE);
|
||||
}
|
||||
|
||||
static void
|
||||
native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
|
||||
{
|
||||
WebRTC *webrtc= GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
ANativeWindow *new_native_window;
|
||||
|
||||
if (!webrtc)
|
||||
|
|
@ -829,7 +836,7 @@ native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
|
|||
|
||||
new_native_window = surface ? ANativeWindow_fromSurface (env, surface) : NULL;
|
||||
GST_DEBUG ("Received surface %p (native window %p)", surface,
|
||||
new_native_window);
|
||||
new_native_window);
|
||||
|
||||
if (webrtc->native_window) {
|
||||
ANativeWindow_release (webrtc->native_window);
|
||||
|
|
@ -837,36 +844,39 @@ native_set_surface (JNIEnv * env, jobject thiz, jobject surface)
|
|||
|
||||
webrtc->native_window = new_native_window;
|
||||
if (webrtc->video_sink)
|
||||
gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (webrtc->video_sink), (guintptr) new_native_window);
|
||||
gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (webrtc->video_sink),
|
||||
(guintptr) new_native_window);
|
||||
}
|
||||
|
||||
static void
|
||||
native_set_signalling_server (JNIEnv * env, jobject thiz, jstring server) {
|
||||
WebRTC *webrtc= GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
const gchar *s;
|
||||
|
||||
if (!webrtc)
|
||||
return;
|
||||
|
||||
s = (*env)->GetStringUTFChars(env, server, NULL);
|
||||
if (webrtc->signalling_server)
|
||||
g_free (webrtc->signalling_server);
|
||||
webrtc->signalling_server = g_strdup (s);
|
||||
(*env)->ReleaseStringUTFChars(env, server, s);
|
||||
}
|
||||
|
||||
static void
|
||||
native_set_call_id(JNIEnv * env, jobject thiz, jstring peer_id) {
|
||||
native_set_signalling_server (JNIEnv * env, jobject thiz, jstring server)
|
||||
{
|
||||
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
const gchar *s;
|
||||
|
||||
if (!webrtc)
|
||||
return;
|
||||
|
||||
s = (*env)->GetStringUTFChars(env, peer_id, NULL);
|
||||
s = (*env)->GetStringUTFChars (env, server, NULL);
|
||||
if (webrtc->signalling_server)
|
||||
g_free (webrtc->signalling_server);
|
||||
webrtc->signalling_server = g_strdup (s);
|
||||
(*env)->ReleaseStringUTFChars (env, server, s);
|
||||
}
|
||||
|
||||
static void
|
||||
native_set_call_id (JNIEnv * env, jobject thiz, jstring peer_id)
|
||||
{
|
||||
WebRTC *webrtc = GET_CUSTOM_DATA (env, thiz, native_webrtc_field_id);
|
||||
const gchar *s;
|
||||
|
||||
if (!webrtc)
|
||||
return;
|
||||
|
||||
s = (*env)->GetStringUTFChars (env, peer_id, NULL);
|
||||
g_free (webrtc->peer_id);
|
||||
webrtc->peer_id = g_strdup (s);
|
||||
(*env)->ReleaseStringUTFChars(env, peer_id, s);
|
||||
(*env)->ReleaseStringUTFChars (env, peer_id, s);
|
||||
}
|
||||
|
||||
/* List of implemented native methods */
|
||||
|
|
|
|||
|
|
@ -183,8 +183,8 @@ const gchar *html_source = " \n \
|
|||
";
|
||||
|
||||
static void
|
||||
handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
|
||||
const char * sink_name)
|
||||
handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
|
||||
const char *sink_name)
|
||||
{
|
||||
GstPad *qpad;
|
||||
GstElement *q, *conv, *resample, *sink;
|
||||
|
|
@ -250,7 +250,8 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
|
|||
}
|
||||
|
||||
static void
|
||||
on_incoming_stream (GstElement * webrtc, GstPad * pad, ReceiverEntry *receiver_entry)
|
||||
on_incoming_stream (GstElement * webrtc, GstPad * pad,
|
||||
ReceiverEntry * receiver_entry)
|
||||
{
|
||||
GstElement *decodebin;
|
||||
GstPad *sinkpad;
|
||||
|
|
@ -287,10 +288,11 @@ create_receiver_entry (SoupWebsocketConnection * connection)
|
|||
G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
|
||||
|
||||
error = NULL;
|
||||
receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
|
||||
receiver_entry->pipeline =
|
||||
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
|
||||
STUN_SERVER " "
|
||||
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
|
||||
"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. "
|
||||
, &error);
|
||||
"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. ", &error);
|
||||
if (error != NULL) {
|
||||
g_error ("Could not create WebRTC pipeline: %s\n", error->message);
|
||||
g_error_free (error);
|
||||
|
|
@ -302,18 +304,24 @@ create_receiver_entry (SoupWebsocketConnection * connection)
|
|||
g_assert (receiver_entry->webrtcbin != NULL);
|
||||
|
||||
/* Incoming streams will be exposed via this signal */
|
||||
g_signal_connect (receiver_entry->webrtcbin, "pad-added", G_CALLBACK (on_incoming_stream),
|
||||
receiver_entry);
|
||||
g_signal_connect (receiver_entry->webrtcbin, "pad-added",
|
||||
G_CALLBACK (on_incoming_stream), receiver_entry);
|
||||
|
||||
#if 0
|
||||
GstElement *rtpbin = gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin");
|
||||
GstElement *rtpbin =
|
||||
gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin");
|
||||
g_object_set (rtpbin, "latency", 40, NULL);
|
||||
gst_object_unref (rtpbin);
|
||||
#endif
|
||||
|
||||
// Create a 2nd transceiver for the receive only video stream
|
||||
video_caps = gst_caps_from_string ("application/x-rtp,media=video,encoding-name=H264,payload=" RTP_PAYLOAD_TYPE ",clock-rate=90000,packetization-mode=(string)1, profile-level-id=(string)42c016");
|
||||
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, NULL, &trans);
|
||||
video_caps =
|
||||
gst_caps_from_string
|
||||
("application/x-rtp,media=video,encoding-name=H264,payload="
|
||||
RTP_PAYLOAD_TYPE
|
||||
",clock-rate=90000,packetization-mode=(string)1, profile-level-id=(string)42c016");
|
||||
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver",
|
||||
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, NULL, &trans);
|
||||
gst_caps_unref (video_caps);
|
||||
gst_object_unref (trans);
|
||||
|
||||
|
|
|
|||
|
|
@ -525,7 +525,7 @@ on_offer_set (GstPromise * promise, gpointer user_data)
|
|||
}
|
||||
|
||||
static void
|
||||
on_offer_received (GstSDPMessage *sdp)
|
||||
on_offer_received (GstSDPMessage * sdp)
|
||||
{
|
||||
GstWebRTCSessionDescription *offer = NULL;
|
||||
GstPromise *promise;
|
||||
|
|
@ -536,8 +536,7 @@ on_offer_received (GstSDPMessage *sdp)
|
|||
/* Set remote description on our pipeline */
|
||||
{
|
||||
promise = gst_promise_new_with_change_func (on_offer_set, NULL, NULL);
|
||||
g_signal_emit_by_name (webrtc1, "set-remote-description", offer,
|
||||
promise);
|
||||
g_signal_emit_by_name (webrtc1, "set-remote-description", offer, promise);
|
||||
}
|
||||
gst_webrtc_session_description_free (offer);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue