Clean up Android.mk
Review URL: http://webrtc-codereview.appspot.com/92014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@315 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/Android.mk b/Android.mk
index 3006785..6eaceae 100644
--- a/Android.mk
+++ b/Android.mk
@@ -8,21 +8,6 @@
MY_WEBRTC_ROOT_PATH := $(call my-dir)
-# These defines will apply to all source files
-# Think again before changing it
-MY_WEBRTC_COMMON_DEFS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_CLOCK_TYPE_REALTIME' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES'
-ifeq ($(TARGET_ARCH),arm)
-MY_WEBRTC_COMMON_DEFS += \
- '-DWEBRTC_ARM_INLINE_CALLS' \
- '-DWEBRTC_ARCH_ARM'
-endif
-
# voice
include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/main/source/Android.mk
include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing_library/main/source/Android.mk
@@ -66,15 +51,103 @@
include $(MY_WEBRTC_ROOT_PATH)/libvpx.mk
# build .so
-include $(MY_WEBRTC_ROOT_PATH)/android-webrtc.mk
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_audio_preprocessing
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_spl \
+ libwebrtc_resampler \
+ libwebrtc_apm \
+ libwebrtc_apm_utility \
+ libwebrtc_vad \
+ libwebrtc_ns \
+ libwebrtc_agc \
+ libwebrtc_aec \
+ libwebrtc_aecm \
+ libwebrtc_system_wrappers
+
+LOCAL_STATIC_LIBRARIES := \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
+###
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_system_wrappers \
+ libwebrtc_audio_device \
+ libwebrtc_pcm16b \
+ libwebrtc_cng \
+ libwebrtc_audio_coding \
+ libwebrtc_rtp_rtcp \
+ libwebrtc_media_file \
+ libwebrtc_udp_transport \
+ libwebrtc_utility \
+ libwebrtc_neteq \
+ libwebrtc_audio_conference_mixer \
+ libwebrtc_isac \
+ libwebrtc_ilbc \
+ libwebrtc_isacfix \
+ libwebrtc_g722 \
+ libwebrtc_g711 \
+ libwebrtc_voe_core \
+ libwebrtc_video_render \
+ libwebrtc_video_capture \
+ libwebrtc_i420 \
+ libwebrtc_video_coding \
+ libwebrtc_video_processing \
+ libwebrtc_vp8 \
+ libwebrtc_vie_core \
+ libwebrtc_vplib \
+ libwebrtc_jpeg \
+ libwebrtc_vpx
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport \
+ libjpeg \
+ libGLESv2 \
+ libOpenSLES \
+ libwebrtc_audio_preprocessing
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
# test apps, they're for test only; all these test apps have LOCAL_MODULE_TAGS:=tests
# audio processing test apps
-# include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/process_test/Android.mk
-# include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/unit_test/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/process_test/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/unit_test/Android.mk
# voice engine test apps
-# include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
-# include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/auto_test/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/auto_test/Android.mk
# video engien test apps
-# include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/AutoTest/android/Android.mk
-# include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/android_test/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/AutoTest/android/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/android_test/Android.mk