android: Switch to the camera for input

diff --git a/webrtc/android/app/src/main/AndroidManifest.xml b/webrtc/android/app/src/main/AndroidManifest.xml
index e4baa4f..54d92cf 100644 (file)
--- a/webrtc/android/app/src/main/AndroidManifest.xml
+++ b/webrtc/android/app/src/main/AndroidManifest.xml
@@ -3,9 +3,11 @@
       package="org.freedesktop.gstreamer.webrtc">
 
     <uses-permission android:name="android.permission.INTERNET"/>
-    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
     <uses-permission android:name="android.permission.WAKE_LOCK"/>
     <uses-permission android:name="android.permission.RECORD_AUDIO" />
+    <uses-permission android:name="android.permission.CAMERA" />
+
+    <uses-feature android:name="android.hardware.camera"  android:required="false" />
     <uses-feature android:glEsVersion="0x00020000"/>
 
     <application android:label="@string/app_name">

diff --git a/webrtc/android/app/src/main/jni/Android.mk b/webrtc/android/app/src/main/jni/Android.mk
index 469bcd7..db8bfca 100644 (file)
--- a/webrtc/android/app/src/main/jni/Android.mk
+++ b/webrtc/android/app/src/main/jni/Android.mk
@@ -30,20 +30,14 @@ endif
 GSTREAMER_NDK_BUILD_PATH  := $(GSTREAMER_ROOT)/share/gst-android/ndk-build/
 
 include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
-GSTREAMER_PLUGINS         := $(GSTREAMER_PLUGINS_CORE)      \
-                             $(GSTREAMER_PLUGINS_PLAYBACK)  \
-                             $(GSTREAMER_PLUGINS_NET)       \
-                             $(GSTREAMER_PLUGINS_SYS)       \
-                             $(GSTREAMER_PLUGINS_CODECS_RESTRICTED) \
-                             $(GSTREAMER_CODECS_GPL)        \
+
+GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume 
+GSTREAMER_PLUGINS_CODECS_CUSTOM := videoparsersbad vpx opus audioparsers opusparse androidmedia
+GSTREAMER_PLUGINS_NET_CUSTOM := tcp rtsp rtp rtpmanager udp srtp webrtc dtls nice
+GSTREAMER_PLUGINS         := $(GSTREAMER_PLUGINS_CORE_CUSTOM) $(GSTREAMER_PLUGINS_CODECS_CUSTOM) $(GSTREAMER_PLUGINS_NET_CUSTOM) \
                              $(GSTREAMER_PLUGINS_ENCODING)  \
-                             $(GSTREAMER_PLUGINS_VIS)       \
-                             $(GSTREAMER_PLUGINS_EFFECTS)   \
-                             $(GSTREAMER_PLUGINS_NET_RESTRICTED) \
-                             subparse ogg theora vorbis opus ivorbisdec alaw apetag audioparsers auparse avi dv flac flv flxdec icydemux id3demux isomp4 jpeg lame matroska mpg123 mulaw multipart png speex taglib vpx wavenc wavpack wavparse y4menc adpcmdec adpcmenc dashdemux dvbsuboverlay dvdspu hls id3tag kate midi mxf openh264 opusparse pcapparse pnm rfbsrc siren smoothstreaming subenc videoparsersbad y4mdec jpegformat gdp rsvg openjpeg spandsp sbc \
-                             nice androidmedia
+                             $(GSTREAMER_PLUGINS_SYS)
 
-#                             $(GSTREAMER_PLUGINS_CODECS)
 GSTREAMER_EXTRA_DEPS      := gstreamer-webrtc-1.0 gstreamer-sdp-1.0 gstreamer-video-1.0 libsoup-2.4 json-glib-1.0 glib-2.0
 
 G_IO_MODULES = gnutls

diff --git a/webrtc/android/app/src/main/jni/webrtc.c b/webrtc/android/app/src/main/jni/webrtc.c
index 9e4c4b7..6176266 100644 (file)
--- a/webrtc/android/app/src/main/jni/webrtc.c
+++ b/webrtc/android/app/src/main/jni/webrtc.c
@@ -322,8 +322,8 @@ start_pipeline (WebRTC * webrtc)
 
   webrtc->pipe =
           gst_parse_launch ("webrtcbin name=sendrecv "
-                            "videotestsrc pattern=ball ! queue ! vp8enc deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit ! "
-                            "queue ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
+                            "ahcsrc ! queue max-size-buffers=2 ! videoconvert ! vp8enc deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit ! "
+                            "queue max-size-buffers=1 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
                             "openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! "
                             "queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ",
                             &error);
@@ -497,6 +497,9 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
     JsonNode *root;
     JsonObject *object;
     JsonParser *parser = json_parser_new ();
+
+    g_print ("Got server message %s", text);
+
     if (!json_parser_load_from_data (parser, text, -1, NULL)) {
       g_printerr ("Unknown message '%s', ignoring", text);
       g_object_unref (parser);
@@ -520,7 +523,10 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
 
       g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING);
 
+      object = json_object_get_object_member (object, "sdp");
+
       g_assert (json_object_has_member (object, "type"));
+
       /* In this example, we always create the offer and receive one answer.
        * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to
        * handle offers from peers and reply with answers using webrtcbin. */
@@ -799,7 +805,7 @@ native_class_init (JNIEnv * env, jclass klass)
     (*env)->ThrowNew (env, exception_class, message);
   }
 
-  gst_debug_set_threshold_from_string ("gl*:7", FALSE);
+  //gst_debug_set_threshold_from_string ("gl*:7", FALSE);
 }
 
 static void