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android, mp-webrtc-sendrecv, sendonly: cleanup
authorCosta Shulyupin <costa@MakeLinux.net>
Tue, 14 Apr 2020 17:13:37 +0000 (20:13 +0300)
committerMathieu Duponchelle <mathieu.duponchelle@opencreed.com>
Thu, 16 Apr 2020 15:34:11 +0000 (17:34 +0200)
webrtc-unidirectional-h264.c: removed empty lines

android: removed unused var

webrtc/android/app/src/main/jni/webrtc.c patch | blob | history
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c patch | blob | history
webrtc/sendonly/webrtc-unidirectional-h264.c patch | blob | history

diff --git a/webrtc/android/app/src/main/jni/webrtc.c b/webrtc/android/app/src/main/jni/webrtc.c
index bfcd8b2..6bfd16a 100644 (file)
--- a/webrtc/android/app/src/main/jni/webrtc.c
+++ b/webrtc/android/app/src/main/jni/webrtc.c
@@ -280,7 +280,6 @@ on_offer_created (GstPromise * promise, WebRTC * webrtc)
 {
   GstWebRTCSessionDescription *offer = NULL;
   const GstStructure *reply;
-  gchar *desc;
 
   g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING);
 
diff --git a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
index 6e66fbd..1c5b0ec 100644 (file)
--- a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
+++ b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
@@ -404,7 +404,6 @@ incoming_call_from_peer (const gchar * peer_id)
 
 #define STR(x) #x
 #define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x)
-#define RTP_CAPS_VP8(x) "application/x-rtp,media=video,encoding-name=VP8,payload=" STR(x)
 
 static gboolean
 start_pipeline (void)
@@ -902,7 +901,6 @@ check_plugins (void)
 {
   int i;
   gboolean ret;
-  GstPlugin *plugin;
   GstRegistry *registry;
   const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp",
     "rtpmanager", "audiotestsrc", NULL
@@ -911,6 +909,7 @@ check_plugins (void)
   registry = gst_registry_get ();
   ret = TRUE;
   for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
+    GstPlugin *plugin;
     plugin = gst_registry_find_plugin (registry, needed[i]);
     if (!plugin) {
       g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
diff --git a/webrtc/sendonly/webrtc-unidirectional-h264.c b/webrtc/sendonly/webrtc-unidirectional-h264.c
index e6f7a59..bb9ca8e 100644 (file)
--- a/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -11,14 +11,10 @@
 #include <json-glib/json-glib.h>
 #include <string.h>
 
-
-
 #define RTP_PAYLOAD_TYPE "96"
 #define SOUP_HTTP_PORT 57778
 #define STUN_SERVER "stun.l.google.com:19302"
 
-
-
 typedef struct _ReceiverEntry ReceiverEntry;
 
 ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
@@ -48,9 +44,6 @@ static gchar *get_string_from_json_object (JsonObject * object);
 
 gboolean exit_sighandler (gpointer user_data);
 
-
-
-
 struct _ReceiverEntry
 {
   SoupWebsocketConnection *connection;
@@ -59,8 +52,6 @@ struct _ReceiverEntry
   GstElement *webrtcbin;
 };
 
-
-
 const gchar *html_source = " \n \
 <html> \n \
   <head> \n \
@@ -166,9 +157,6 @@ const gchar *html_source = " \n \
 </html> \n \
 ";
 
-
-
-
 ReceiverEntry *
 create_receiver_entry (SoupWebsocketConnection * connection)
 {
Domain: Multimedia / Media Common;