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LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
LOCAL_MODULE := gstwebrtc
LOCAL_SRC_FILES := webrtc.c dummy.cpp
LOCAL_SHARED_LIBRARIES := gstreamer_android
LOCAL_LDLIBS := -llog -landroid
include $(BUILD_SHARED_LIBRARY)
ifndef GSTREAMER_ROOT_ANDROID
$(error GSTREAMER_ROOT_ANDROID is not defined!)
endif
ifeq ($(TARGET_ARCH_ABI),armeabi)
GSTREAMER_ROOT := $(GSTREAMER_ROOT_ANDROID)/arm
else ifeq ($(TARGET_ARCH_ABI),armeabi-v7a)
GSTREAMER_ROOT := $(GSTREAMER_ROOT_ANDROID)/armv7
else ifeq ($(TARGET_ARCH_ABI),arm64-v8a)
GSTREAMER_ROOT := $(GSTREAMER_ROOT_ANDROID)/arm64
else ifeq ($(TARGET_ARCH_ABI),x86)
GSTREAMER_ROOT := $(GSTREAMER_ROOT_ANDROID)/x86
else ifeq ($(TARGET_ARCH_ABI),x86_64)
GSTREAMER_ROOT := $(GSTREAMER_ROOT_ANDROID)/x86_64
else
$(error Target arch ABI not supported: $(TARGET_ARCH_ABI))
endif
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_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_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
G_IO_MODULES = openssl
include $(GSTREAMER_NDK_BUILD_PATH)/gstreamer-1.0.mk
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