From 8c4345da7d86686974e37b0cecd753fb6e20f3ac Mon Sep 17 00:00:00 2001 From: Costa Shulyupin Date: Tue, 14 Apr 2020 20:13:37 +0300 Subject: [PATCH] android, mp-webrtc-sendrecv, sendonly: cleanup webrtc-unidirectional-h264.c: removed empty lines android: removed unused var --- webrtc/android/app/src/main/jni/webrtc.c | 1 - webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c | 3 +-- webrtc/sendonly/webrtc-unidirectional-h264.c | 12 ------------ 3 files changed, 1 insertion(+), 15 deletions(-) diff --git a/webrtc/android/app/src/main/jni/webrtc.c b/webrtc/android/app/src/main/jni/webrtc.c index bfcd8b24b2..6bfd16a556 100644 --- 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 6e66fbd162..1c5b0ecc23 100644 --- 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 e6f7a59ae0..bb9ca8e44b 100644 --- a/webrtc/sendonly/webrtc-unidirectional-h264.c +++ b/webrtc/sendonly/webrtc-unidirectional-h264.c @@ -11,14 +11,10 @@ #include #include - - #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 \ \n \ \n \ @@ -166,9 +157,6 @@ const gchar *html_source = " \n \ \n \ "; - - - ReceiverEntry * create_receiver_entry (SoupWebsocketConnection * connection) {