diff --git a/webrtc/android/app/src/main/jni/Android.mk b/webrtc/android/app/src/main/jni/Android.mk
index 326781d6af..a3b94da8a8 100644
--- a/webrtc/android/app/src/main/jni/Android.mk
+++ b/webrtc/android/app/src/main/jni/Android.mk
@@ -31,12 +31,13 @@ GSTREAMER_NDK_BUILD_PATH  := $(GSTREAMER_ROOT)/share/gst-android/ndk-build/
 
 include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
 
-GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume
+GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc audiotestsrc volume autodetect
 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_SYS)
+                             $(GSTREAMER_PLUGINS_SYS) \
+                             $(GSTREAMER_PLUGINS_PLAYBACK)
 
 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
 GSTREAMER_EXTRA_LIBS      := -liconv