Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes.
BUG=1980 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -6,15 +6,81 @@
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# in the file PATENTS. All contributing project authors may
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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# be found in the AUTHORS file in the root of the source tree.
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{
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{
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'variables': {
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# NOTE: This laundry list of libs comes from jni/Android.mk and should be
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# kept in sync with that location. Also note that the explicit library
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# names must be used (instead of the names of their gyp targets) because
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# these are fed into the 'inputs' attribute of the action below, not to
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# 'dependencies' because these need to be real (build-time) dependencies,
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# not just order-only dependencies
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# (https://code.google.com/p/webrtc/issues/detail?id=1980).
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'android_mk_common_libs': [
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'<(PRODUCT_DIR)/libvoice_engine.a',
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'<(PRODUCT_DIR)/libvideo_engine_core.a',
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'<(PRODUCT_DIR)/libvideo_processing.a',
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'<(PRODUCT_DIR)/libwebrtc_video_coding.a',
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'<(PRODUCT_DIR)/libvideo_render_module.a',
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'<(PRODUCT_DIR)/libvideo_capture_module.a',
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'<(PRODUCT_DIR)/libaudio_coding_module.a',
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'<(PRODUCT_DIR)/libaudio_processing.a',
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'<(PRODUCT_DIR)/libPCM16B.a',
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'<(PRODUCT_DIR)/libCNG.a',
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'<(PRODUCT_DIR)/libNetEq.a',
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'<(PRODUCT_DIR)/libG722.a',
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'<(PRODUCT_DIR)/libiSAC.a',
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'<(PRODUCT_DIR)/libG711.a',
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'<(PRODUCT_DIR)/libiLBC.a',
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'<(PRODUCT_DIR)/libiSACFix.a',
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'<(PRODUCT_DIR)/libwebrtc_opus.a',
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'<(PRODUCT_DIR)/libopus.a',
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'<(PRODUCT_DIR)/libcommon_audio.a',
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'<(PRODUCT_DIR)/libbitrate_controller.a',
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'<(PRODUCT_DIR)/libcommon_video.a',
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'<(PRODUCT_DIR)/libcpu_features_android.a',
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'<(PRODUCT_DIR)/libaudio_device.a',
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'<(PRODUCT_DIR)/libremote_bitrate_estimator.a',
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'<(PRODUCT_DIR)/librtp_rtcp.a',
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'<(PRODUCT_DIR)/libmedia_file.a',
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'<(PRODUCT_DIR)/libchannel_transport.a',
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'<(PRODUCT_DIR)/libwebrtc_utility.a',
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'<(PRODUCT_DIR)/libaudio_conference_mixer.a',
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'<(PRODUCT_DIR)/libyuv.a',
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'<(PRODUCT_DIR)/libwebrtc_i420.a',
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'<(PRODUCT_DIR)/libwebrtc_vp8.a',
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'<(PRODUCT_DIR)/libvideo_coding_utility.a',
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'<(PRODUCT_DIR)/libsystem_wrappers.a',
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'<(PRODUCT_DIR)/libjpeg_turbo.a',
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'<(PRODUCT_DIR)/libaudioproc_debug_proto.a',
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'<(PRODUCT_DIR)/libprotobuf_lite.a',
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'<(PRODUCT_DIR)/libpaced_sender.a',
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'<(PRODUCT_DIR)/libvpx.a',
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],
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'android_mk_x86_libs': [
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'<(PRODUCT_DIR)/libvideo_processing_sse2.a',
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'<(PRODUCT_DIR)/libaudio_processing_sse2.a',
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'<(PRODUCT_DIR)/libcommon_audio_sse2.a',
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'<(PRODUCT_DIR)/libvpx_intrinsics_mmx.a',
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'<(PRODUCT_DIR)/libvpx_intrinsics_sse2.a',
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'<(PRODUCT_DIR)/libvpx_intrinsics_ssse3.a',
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],
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'android_mk_arm_libs': [
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'<(PRODUCT_DIR)/libaudio_processing_neon.a',
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'<(PRODUCT_DIR)/libisac_neon.a',
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'<(PRODUCT_DIR)/libcommon_audio_neon.a',
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'<(PRODUCT_DIR)/libvpx_arm_neon.a',
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],
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},
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'targets': [
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'targets': [
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{
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{
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'target_name': 'video_demo_apk',
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'target_name': 'video_demo_apk',
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'type': 'none',
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'type': 'none',
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'dependencies': [
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'conditions': [
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'<(webrtc_root)/modules/modules.gyp:*',
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['target_arch=="x86"', {
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'<(webrtc_root)/test/test.gyp:channel_transport',
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'inputs': [ '<@(android_mk_x86_libs)' ],
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'<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
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}, {
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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'inputs': [ '<@(android_mk_arm_libs)' ],
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}],
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],
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],
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'actions': [
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'actions': [
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{
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{
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@ -33,6 +99,7 @@
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/IViEAndroidCallback.java',
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/IViEAndroidCallback.java',
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java',
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java',
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/WebRTCDemo.java',
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'<(webrtc_root)/video_engine/test/android/src/org/webrtc/videoengineapp/WebRTCDemo.java',
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'<@(android_mk_common_libs)',
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],
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],
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'outputs': ['<(webrtc_root)'],
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'outputs': ['<(webrtc_root)'],
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'action': ['python',
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'action': ['python',
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@ -43,4 +110,3 @@
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},
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},
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],
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],
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}
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}
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@ -386,7 +386,6 @@ LOCAL_WHOLE_STATIC_LIBRARIES := \
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libremote_bitrate_estimator \
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libremote_bitrate_estimator \
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librtp_rtcp \
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librtp_rtcp \
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libmedia_file \
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libmedia_file \
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libudp_transport \
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libchannel_transport \
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libchannel_transport \
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libwebrtc_utility \
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libwebrtc_utility \
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libaudio_conference_mixer \
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libaudio_conference_mixer \
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