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android, mp-webrtc-sendrecv, sendonly: cleanup
webrtc-unidirectional-h264.c: removed empty lines android: removed unused var
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parent
133a1593ee
commit
8c4345da7d
3 changed files with 1 additions and 15 deletions
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@ -280,7 +280,6 @@ on_offer_created (GstPromise * promise, WebRTC * webrtc)
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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 (webrtc->app_state == PEER_CALL_NEGOTIATING);
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@ -404,7 +404,6 @@ incoming_call_from_peer (const gchar * peer_id)
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#define STR(x) #x
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#define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x)
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#define RTP_CAPS_VP8(x) "application/x-rtp,media=video,encoding-name=VP8,payload=" STR(x)
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static gboolean
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start_pipeline (void)
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@ -902,7 +901,6 @@ check_plugins (void)
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{
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int i;
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gboolean ret;
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GstPlugin *plugin;
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GstRegistry *registry;
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const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp",
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"rtpmanager", "audiotestsrc", NULL
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@ -911,6 +909,7 @@ check_plugins (void)
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registry = gst_registry_get ();
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ret = TRUE;
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for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
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GstPlugin *plugin;
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plugin = gst_registry_find_plugin (registry, needed[i]);
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if (!plugin) {
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g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
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@ -11,14 +11,10 @@
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#include <json-glib/json-glib.h>
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#include <string.h>
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#define RTP_PAYLOAD_TYPE "96"
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#define SOUP_HTTP_PORT 57778
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#define STUN_SERVER "stun.l.google.com:19302"
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typedef struct _ReceiverEntry ReceiverEntry;
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ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
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@ -48,9 +44,6 @@ static gchar *get_string_from_json_object (JsonObject * object);
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gboolean exit_sighandler (gpointer user_data);
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struct _ReceiverEntry
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{
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SoupWebsocketConnection *connection;
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@ -59,8 +52,6 @@ struct _ReceiverEntry
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GstElement *webrtcbin;
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};
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const gchar *html_source = " \n \
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<html> \n \
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<head> \n \
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@ -166,9 +157,6 @@ const gchar *html_source = " \n \
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</html> \n \
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";
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ReceiverEntry *
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create_receiver_entry (SoupWebsocketConnection * connection)
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{
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