Closed dpenkler closed 7 years ago
mpromonet
commented
7 years ago Hi,
Surely it miss a lots of librairies...
Did you tried to run make clean and make ?
If you are right and now webrtc use shared librairy, it needs some modification in the Makefile.
Best Regards,
Michel.
dpenkler
commented
7 years ago Did make clean; make multiple times. It may be caused by them migrating to gn from gyp.
There are only 2 .so files the rest are just .o files in the build directories.
Can you send the complete list of .a files please.
mpromonet
commented
7 years ago Last time I built webrtc it was based on gyp, I am building with gn to see what is missing.
Probably it is needed to link with the .so libraries.
dpenkler
commented
7 years ago It is now almost building.
g++ -o webrtc-serverRelease src/PeerConnectionManager.o src/HttpServerRequestHandler.o src/main.o libWebRTCRelease.a -pthread -lX11 -ldl -lrt
src/PeerConnectionManager.o: In function cricket::FakeVideoCapturer::ResetSupportedFormats(std::vector<cricket::VideoFormat, std::allocator<cricket::VideoFormat> > const&)': /home/dave/webrtc-streamer/../webrtc/src/webrtc/media/base/fakevideocapturer.h:62: undefined reference tocricket::VideoCapturer::SetSupportedFormats(std::vector<cricket::VideoFormat, std::allocatornon-virtual thunk to cricket::VideoCapturer::AddOrUpdateSink(rtc::VideoSinkInterface<cricket::VideoFrame>*, rtc::VideoSinkWants const&)' src/PeerConnectionManager.o:(.rodata._ZTVN7cricket17FakeVideoCapturerE[_ZTVN7cricket17FakeVideoCapturerE]+0xd0): undefined reference tonon-virtual thunk to cricket::VideoCapturer::RemoveSink(rtc::VideoSinkInterfacecricket::VideoFrame_)'
collect2: error: ld returned 1 exit status
Makefile:162: recipe for target 'webrtc-server_Release' failed
make: ** [webrtc-server__Release] Error 1
dave@edgelinegw:~/webrtc-streamer$
` Here is the makefile. CC = $(CROSS)g++ $(foreach sysroot,$(SYSROOT),--sysroot=$(sysroot))
AR = $(CROSS)ar
CFLAGS = -W -pthread -g -std=c++11 -Iinc
LDFLAGS = -pthread
ifneq ($(wildcard $(SYSROOT)/usr/include/liveMedia/liveMedia.hh),) CFLAGS += -DHAVE_LIVE555 CFLAGS += -I $(SYSROOT)/usr/include/liveMedia -I $(SYSROOT)/usr/include/groupsock -I $(SYSROOT)/usr/include/UsageEnvironment -I $(SYSROOT)/usr/include/BasicUsageEnvironment/ LDFLAGS += -lliveMedia -lgroupsock -lUsageEnvironment -lBasicUsageEnvironment endif
WEBRTCROOT?=../webrtc WEBRTCBUILD?=Release WEBRTCLIBPATH=$(WEBRTCROOT)/src/$(GYP_GENERATOR_OUTPUT)/out/$(WEBRTCBUILD) ifneq ($(wildcard $(WEBRTCROOT)/src/webrtc/media/base/yuvframegenerator.h),) CFLAGS += -DHAVE_YUVFRAMEGENERATOR endif
CFLAGS += -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 CFLAGS += -I $(WEBRTCROOT)/src -I $(WEBRTCROOT)/src/chromium/src/third_party/jsoncpp/source/include ifeq ($(WEBRTCBUILD),Debug) CFLAGS += -D_GLIBCXX_DEBUG=1 endif LDFLAGS += -lX11 -ldl -lrt
TARGET = webrtc-server_$(GYP_GENERATOROUTPUT)$(WEBRTCBUILD) all: $(TARGET)
WEBRTCLIB = $(shell find $(WEBRTCLIBPATH) -name '.a') WEBRTC_OBJ = $(shell find \ $(WEBRTCLIBPATH)/obj/webrtc/stats/rtc_stats \ $(WEBRTCLIBPATH)/obj/webrtc/video/video \ $(WEBRTCLIBPATH)/obj/webrtc/rtc_event_log_proto \ $(WEBRTCLIBPATH)/obj/webrtc/rtc_event_log_parser \ $(WEBRTCLIBPATH)/obj/webrtc/system_wrappers/field_trial_default \ $(WEBRTCLIBPATH)/obj/webrtc/system_wrappers/system_wrappers \ $(WEBRTCLIBPATH)/obj/webrtc/system_wrappers/metrics_default \ $(WEBRTCLIBPATH)/obj/webrtc/system_wrappers/cpu_features_linux \ $(WEBRTCLIBPATH)/obj/webrtc/pc/rtc_pc \ $(WEBRTCLIBPATH)/obj/webrtc/api/libjingle_peerconnection \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_capture/video_capture_internal_impl \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_capture/video_capture \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_capture/video_capture_module \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/webrtc_h264 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/video_coding \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/webrtc_i420 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/webrtc_vp8 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/webrtc_vp9 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_coding/video_coding_utility \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_processing/video_processing_sse2 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/video_processing/video_processing \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_device/audio_device \ $(WEBRTCLIBPATH)/obj/webrtc/modules/remote_bitrate_estimator/bwe_simulator \ $(WEBRTCLIBPATH)/obj/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator \ $(WEBRTCLIBPATH)/obj/webrtc/modules/pacing/pacing \ $(WEBRTCLIBPATH)/obj/webrtc/modules/media_file/media_file \ $(WEBRTCLIBPATH)/obj/webrtc/modules/rtp_rtcp/rtp_rtcp \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/RTPtimeshift \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/rtp_analyze \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/ilbc \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/webrtc_opus \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/rent_a_codec \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/neteq_rtpplay \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/g722 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/pcm16b \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/red \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/RTPencode \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/g711 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/builtin_audio_decoder_factory \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/audio_decoder_factory_interface \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/neteq \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/RTPchange \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/audio_coding \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/isac_fix \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/audio_encoder_interface \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/rtc_event_log_source \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/cng \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/audio_decoder_interface \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/rtpcat \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/insert_packet_with_timing \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/RTPjitter \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/isac \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_coding/isac_common \ $(WEBRTCLIBPATH)/obj/webrtc/modules/bitrate_controller/bitrate_controller \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_mixer/audio_mixer \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_conference_mixer \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_conference_mixer/audio_conference_mixer \ $(WEBRTCLIBPATH)/obj/webrtc/modules/congestion_controller/congestion_controller \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_processing/unpack_aecdump \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_processing/audioproc_debug_proto \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_processing/audio_processing \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_processing/audioproc_protobuf_utils \ $(WEBRTCLIBPATH)/obj/webrtc/modules/audio_processing/audio_processing_sse2 \ $(WEBRTCLIBPATH)/obj/webrtc/modules/utility/utility \ $(WEBRTCLIBPATH)/obj/webrtc/common_video/common_video \ $(WEBRTCLIBPATH)/obj/webrtc/voice_engine/level_indicator \ $(WEBRTCLIBPATH)/obj/webrtc/voice_engine/voice_engine \ $(WEBRTCLIBPATH)/obj/webrtc/p2p/libstunprober \ $(WEBRTCLIBPATH)/obj/webrtc/p2p/rtc_p2p \ $(WEBRTCLIBPATH)/obj/webrtc/p2p/stun_prober \ $(WEBRTCLIBPATH)/obj/webrtc/webrtc_common \ $(WEBRTCLIBPATH)/obj/webrtc/common_audio/common_audio_sse2 \ $(WEBRTCLIBPATH)/obj/webrtc/common_audio/common_audio \ $(WEBRTCLIBPATH)/obj/webrtc/audio/audio \ $(WEBRTCLIBPATH)/obj/webrtc/rtc_event_log \ $(WEBRTCLIBPATH)/obj/webrtc/base/rtc_base_approved \ $(WEBRTCLIBPATH)/obj/webrtc/base/rtc_task_queue \ $(WEBRTCLIBPATH)/obj/webrtc/base/rtc_base \ $(WEBRTCLIBPATH)/obj/webrtc/media/rtc_media \ $(WEBRTCLIBPATH)/obj/webrtc/call/call \ $(WEBRTCLIBPATH)/obj/third_party/openmax_dl/dl \ $(WEBRTCLIBPATH)/obj/third_party/openmax_dl/dl/dl \ $(WEBRTCLIBPATH)/obj/third_party/usrsctp/usrsctp \ $(WEBRTCLIBPATH)/obj/third_party/gflags/gflags \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/srtp_test_rand_gen \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/libsrtp \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/rdbx_driver \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/roc_driver \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/rtpw \ $(WEBRTCLIBPATH)/obj/third_party/libsrtp/srtp_driver \ $(WEBRTCLIBPATH)/obj/third_party/jsoncpp/jsoncpp \ $(WEBRTCLIBPATH)/obj/third_party/libyuv/libyuv \ $(WEBRTCLIBPATH)/obj/third_party/libjpeg_turbo/simd_asm \ $(WEBRTCLIBPATH)/obj/third_party/libjpeg_turbo/libjpeg \ $(WEBRTCLIBPATH)/obj/third_party/libjpeg_turbo/simd \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_sse4_1 \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_yasm \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_mmx \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_ssse3 \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_sse2 \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_avx2 \ $(WEBRTCLIBPATH)/obj/third_party/libvpx/libvpx_intrinsics_avx \ $(WEBRTCLIBPATH)/obj/third_party/boringssl/boringssl_asm \ $(WEBRTCLIBPATH)/obj/third_party/boringssl/boringssl \ $(WEBRTCLIBPATH)/obj/third_party/protobuf/protobuf_full \ $(WEBRTCLIBPATH)/obj/third_party/protobuf/protoc_lib \ $(WEBRTCLIBPATH)/obj/third_party/protobuf/protobuf_lite \ $(WEBRTCLIBPATH)/obj/third_party/protobuf/protoc \ $(WEBRTCLIBPATH)/obj/third_party/opus/opus_compare \ $(WEBRTCLIBPATH)/obj/third_party/opus/opus \ $(WEBRTCLIBPATH)/obj/third_party/yasm/genmacro \ $(WEBRTCLIBPATH)/obj/third_party/yasm/re2c \ $(WEBRTCLIBPATH)/obj/third_party/yasm/genperf \ $(WEBRTCLIBPATH)/obj/third_party/yasm/genstring \ $(WEBRTCLIBPATH)/obj/third_party/yasm/genmodule \ $(WEBRTCLIBPATH)/obj/third_party/yasm/genversion \ $(WEBRTCLIBPATH)/obj/third_party/yasm/yasm_utils \ $(WEBRTCLIBPATH)/obj/third_party/yasm/yasm \ $(WEBRTCLIBPATH)/obj/base/thirdparty/libevent/libevent \ -name '.o') libWebRTC_$(GYP_GENERATOROUTPUT)$(WEBRTCBUILD).a: $(WEBRTC_OBJ) $(AR) -rcT $@ $^
src/%.o: src/%.cpp $(CC) -o $@ -c $^ $(CFLAGS)
FILES = $(wildcard src/*.cpp) $(TARGET): $(subst .cpp,.o,$(FILES)) libWebRTC_$(GYP_GENERATOROUTPUT)$(WEBRTCBUILD).a $(CC) -o $@ $^ $(LDFLAGS)
clean: rm -f src/*.o libWebRTC_$(GYP_GENERATOROUTPUT)$(WEBRTCBUILD).a $(TARGET) `
webrtc builds ok. webrtc-streamer compiles ok but fails in link step. Could be caused by some libraries in webrtc being generated as shared libraries with .so suffix. ar -rcT libWebRTC__Default.a ../webrtc/src//out/Default/obj/webrtc/librtc_event_log_proto.a ../webrtc/src//out/Default/obj/webrtc/modules/audio_coding/libneteq_unittest_proto.a ../webrtc/src//out/Default/obj/webrtc/modules/audio_processing/libaudioproc_debug_proto.a ../webrtc/src//out/Default/obj/webrtc/modules/audio_processing/libaudioproc_unittest_proto.a ../webrtc/src//out/Default/obj/webrtc/tools/libgraph_proto.a ../webrtc/src//out/Default/obj/webrtc/base/librtc_base_approved.a ../webrtc/src//out/Default/obj/webrtc/base/librtc_task_queue.a ../webrtc/src//out/Default/obj/webrtc/base/librtc_base.a ../webrtc/src//out/Default/obj/testing/gmock/libgmock_main.a ../webrtc/src//out/Default/obj/testing/gmock/libgmock.a ../webrtc/src//out/Default/obj/testing/gtest/libgtest.a ../webrtc/src//out/Default/obj/third_party/openmax_dl/dl/libdl.a ../webrtc/src//out/Default/obj/third_party/usrsctp/libusrsctp.a ../webrtc/src//out/Default/obj/third_party/libsrtp/libsrtp.a ../webrtc/src//out/Default/obj/third_party/libyuv/libyuv.a ../webrtc/src//out/Default/obj/third_party/libjpeg_turbo/libjpeg.a ../webrtc/src//out/Default/obj/third_party/libjpeg_turbo/libsimd_asm.a ../webrtc/src//out/Default/obj/third_party/libjpeg_turbo/libsimd.a ../webrtc/src//out/Default/obj/third_party/libvpx/libvpx.a ../webrtc/src//out/Default/obj/third_party/libvpx/libvpx_yasm.a ../webrtc/src//out/Default/obj/third_party/protobuf/libprotobuf_full.a ../webrtc/src//out/Default/obj/third_party/protobuf/libprotoc_lib.a ../webrtc/src//out/Default/obj/third_party/opus/libopus.a ../webrtc/src//out/Default/obj/third_party/yasm/libyasm_utils.a ../webrtc/src//out/Default/obj/base/third_party/libevent/libevent.a ../webrtc/src//out/Default/obj/build/config/sanitizers/liboptions_sources.a g++ -o webrtc-serverDefault src/PeerConnectionManager.o src/HttpServerRequestHandler.o src/main.o libWebRTCDefault.a -pthread -lX11 -ldl -lrt src/PeerConnectionManager.o: In function
PeerConnectionManager::PeerConnectionManager(std::string const&)': /home/dave/webrtc-streamer/src/PeerConnectionManager.cpp:42: undefined reference toJson::Value::Value(Json::ValueType)' src/PeerConnectionManager.o: In functionPeerConnectionManager::~PeerConnectionManager()': /home/dave/webrtc-streamer/src/PeerConnectionManager.cpp:46: undefined reference toJson::Value::~Value()'``