wrtc combine

README.txt

https://docs.google.com/document/d/1J6rcqV5KWpYCZlhWv4vt8Ilrh_f08QC2KA1jbkSBo9s/edit?usp=sharing

buildtest.sh

# ninja -C ./out/release boringssl protobuf_lite p2p base jsoncpp -t clean
# ninja -C ./out/release boringssl protobuf_lite p2p base jsoncpp
# ninja -C ./out/release -t clean
# ninja -C ./out/release
# rm ~/dev/webrtc-builds/src/out/release/libwebrtc_full.a
# rm -rf ~/dev/webrtc-builds/src/include
# bash compile.sh
# sh buildtest.sh
# ./simpleTest

# see http://ipop-project.org/wiki/Build-WebRTC-Libraries-for-Raspberry-Pi-Zero-(Cross-compile-on-Ubuntu)
# ninja -C ./out/release boringssl protobuf_lite p2p base jsoncpp
file ./out/release/libwebrtc_full.a

#  RUN BEFORE IT bash compile.sh
# https://www.smwenku.com/a/5b8830292b71775d1cdaf404/
# SEE https://github.com/Palakis/webrtc-builds/blob/master/resource/pkgconfig/libwebrtc_full.pc.in
g++ -DWEBRTC_POSIX -std=gnu++11 -o simpleTest test1.cpp -Iinclude -Iinclude/webrtc -Ithird_party/abseil-cpp ./out/release/libwebrtc_full.a  ./out/release/obj/api/libjingle_peerconnection_api.a ./out/release/obj/third_party/boringssl/libboringssl.a -lssl -lcrypto -lpthread -lX11 -ldl -lexpat -Ithird_party/jsoncpp/source/include -Ithird_party/libyuv/include -fno-rtti -fno-exceptions

# READ https://stackoverflow.com/questions/39748072/linking-native-webrtc-application-with-visual-studio
# USED find "$(cd out/release; pwd)" -name "*.a"
#g++ -DWEBRTC_POSIX -std=gnu++11 -o simpleTest test1.cpp -Iinclude -Iinclude/webrtc -Ithird_party/abseil-cpp -lssl -lcrypto -lpthread -lX11 -ldl -lexpat -Ithird_party/jsoncpp/source/include -Ithird_party/libyuv/include -fno-rtti -fno-exceptions /home/denis/dev/webrtc-builds/src/out/release/obj/testing/gtest/libgtest.a /home/denis/dev/webrtc-builds/src/out/release/obj/call/libcall.a /home/denis/dev/webrtc-builds/src/out/release/obj/libwebrtc.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libsrtp/libsrtp.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/usrsctp/libusrsctp.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/protobuf/libprotoc_lib.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/protobuf/libprotobuf_lite.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/protobuf/libprotobuf_full.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libvpx/libvpx.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libvpx/libvpx_yasm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libyuv/libyuv_internal.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/opus/libopus.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libjpeg_turbo/libsimd.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libjpeg_turbo/libjpeg.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/libjpeg_turbo/libsimd_asm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/yasm/libyasm_utils.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/openh264/libopenh264_processing_yasm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/openh264/libopenh264_common_yasm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/openh264/libopenh264_encoder_yasm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/zlib/libchrome_zlib.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/zlib/libzlib_x86_simd.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/ffmpeg/libffmpeg_yasm.a /home/denis/dev/webrtc-builds/src/out/release/obj/third_party/ffmpeg/libffmpeg_internal.a /home/denis/dev/webrtc-builds/src/out/release/obj/p2p/libstunprober.a /home/denis/dev/webrtc-builds/src/out/release/obj/p2p/librtc_p2p.a /home/denis/dev/webrtc-builds/src/out/release/obj/base/third_party/libevent/libevent.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/remote_bitrate_estimator/libremote_bitrate_estimator.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/remote_bitrate_estimator/libbwe_simulator_lib.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libwebrtc_vp9_helpers.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libnack_module.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libwebrtc_vp9.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libwebrtc_h264.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libwebrtc_vp8.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libwebrtc_multiplex.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libencoded_frame.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libvideo_coding.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_coding/libpacket.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/bitrate_controller/libbitrate_controller.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_capture/libvideo_capture_module.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_capture/libvideo_capture.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/rtp_rtcp/librtp_rtcp.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/rtp/libcongestion_controller.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/rtp/libtransport_feedback.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/libpcc.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/libutility_function.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/libmonitor_interval.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/libbitrate_controller.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/libpcc_controller.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/pcc/librtt_tracker.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/libcongestion_controller.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/bbr/libbbr.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/libtransport_feedback.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/congestion_controller/goog_cc/libgoog_cc.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_processing/libvideo_processing_sse2.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/video_processing/libvideo_processing.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/libruntime_settings_protobuf_utils.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/test/conversational_speech/lib.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/libaudioproc_debug_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/aec3/libaec3.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/libaudioproc_unittest_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/vad/libvad.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/vad/libvad_unittests.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/libaudioproc_protobuf_utils.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_processing/libaudio_processing.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/desktop_capture/libprimitives.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/desktop_capture/libdesktop_capture_generic.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/desktop_capture/libdesktop_capture_differ_sse2.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libisac_common.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libred.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/librent_a_codec.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libneteq.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/codecs/opus/test/libtest.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libcng.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/librtc_event_log_source.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libneteq_unittest_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libwebrtc_opus.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libaudio_coding.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libisac.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libaudio_network_adaptor.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libg711.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libilbc.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libpcm16b.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libisac_c.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libaudio_format_conversion.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libana_config_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libana_debug_dump_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/liblegacy_encoded_audio_frame.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libisac_fix.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libaudio_network_adaptor_config.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_coding/libg722.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/utility/libutility.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/pacing/libpacing.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_mixer/libaudio_frame_manipulator.a /home/denis/dev/webrtc-builds/src/out/release/obj/modules/audio_mixer/libaudio_mixer_impl.a /home/denis/dev/webrtc-builds/src/out/release/obj/video/libvideo.a /home/denis/dev/webrtc-builds/src/out/release/obj/media/librtc_constants.a /home/denis/dev/webrtc-builds/src/out/release/obj/media/librtc_media_base.a /home/denis/dev/webrtc-builds/src/out/release/obj/media/librtc_internal_video_codecs.a /home/denis/dev/webrtc-builds/src/out/release/obj/media/librtc_data.a /home/denis/dev/webrtc-builds/src/out/release/obj/media/librtc_audio_video.a /home/denis/dev/webrtc-builds/src/out/release/obj/common_audio/libcommon_audio.a /home/denis/dev/webrtc-builds/src/out/release/obj/common_audio/libcommon_audio_sse2.a /home/denis/dev/webrtc-builds/src/out/release/obj/common_video/libcommon_video.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/video_codecs/librtc_software_fallback_wrappers.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/video_codecs/libbuiltin_video_decoder_factory.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/video_codecs/libbuiltin_video_encoder_factory.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/transport/libgoog_cc.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/transport/libnetwork_control.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/libjingle_peerconnection_api.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/L16/libaudio_decoder_L16.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/L16/libaudio_encoder_L16.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/libbuiltin_audio_decoder_factory.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/opus/libaudio_decoder_opus.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/opus/libaudio_encoder_opus_config.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/g711/libaudio_encoder_g711.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/g711/libaudio_decoder_g711.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/ilbc/libaudio_decoder_ilbc.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/ilbc/libaudio_encoder_ilbc.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/isac/libaudio_encoder_isac_float.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/isac/libaudio_encoder_isac_fix.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/isac/libaudio_decoder_isac_fix.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/isac/libaudio_decoder_isac_float.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/g722/libaudio_encoder_g722.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/g722/libaudio_decoder_g722.a /home/denis/dev/webrtc-builds/src/out/release/obj/api/audio_codecs/libbuiltin_audio_encoder_factory.a /home/denis/dev/webrtc-builds/src/out/release/obj/pc/librtc_pc_base.a /home/denis/dev/webrtc-builds/src/out/release/obj/pc/libcreate_pc_factory.a /home/denis/dev/webrtc-builds/src/out/release/obj/pc/libpeerconnection.a /home/denis/dev/webrtc-builds/src/out/release/obj/system_wrappers/libsystem_wrappers.a /home/denis/dev/webrtc-builds/src/out/release/obj/system_wrappers/libcpu_features_linux.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/librtc_numerics.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/librtc_base_generic.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/libweak_ptr.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/librtc_base.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/libjitter_upper_bound_experiment.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/libfield_trial_parser.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/libcongestion_controller_experiment.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/librtt_mult_experiment.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/libquality_scaling_experiment.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_base/experiments/libalr_experiment.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/network_tester/libnetwork_tester.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/network_tester/libnetwork_tester_config_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/network_tester/libnetwork_tester_packet_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libvideo_file_writer.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libvideo_quality_analysis.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libevent_log_visualizer_utils.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libchart_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libframe_editing_lib.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libcommand_line_parser.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libreference_less_video_analysis_lib.a /home/denis/dev/webrtc-builds/src/out/release/obj/rtc_tools/libvideo_file_reader.a /home/denis/dev/webrtc-builds/src/out/release/obj/logging/librtc_event_log_impl_base.a /home/denis/dev/webrtc-builds/src/out/release/obj/logging/librtc_event_log_parser.a /home/denis/dev/webrtc-builds/src/out/release/obj/logging/librtc_event_log_impl_encoder.a /home/denis/dev/webrtc-builds/src/out/release/obj/logging/librtc_event_log2_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/logging/librtc_event_log_proto.a /home/denis/dev/webrtc-builds/src/out/release/obj/stats/librtc_stats.a /home/denis/dev/webrtc-builds/src/out/release/obj/audio/libaudio.a /home/denis/dev/webrtc-builds/src/out/release/obj/audio/utility/libaudio_frame_operations.a
#C:\Libs\webrtc\001\webrtc-checkout\src\out\Debug\obj\webrtc\api\libjingle_peerconnection_api.lib
#C:\Libs\webrtc\001\webrtc-checkout\src\out\Debug\obj\webrtc\audio\audio.lib

combine.sh

# SEE https://github.com/vsimon/webrtcbuilds/blob/master/util.sh#L232


# This function compile and combine build artifact objects into one library.
# $1 the output directory, 'Debug', 'Release'
# $2 additional gn arguments
# After check: find ../ -name 'libwebrtc_full*'
function compile-unix() {
  local outputdir="$1"

  local blacklist="unittest|examples|/yasm|protobuf_lite|main.o|\
video_capture_external.o|device_info_external.o"

  pushd $outputdir >/dev/null

  #rm -f libwebrtc_full.a || true
  # Produce an ordered objects list by parsing .ninja_deps for strings
  # matching .o files.
  local objlist=$(strings .ninja_deps | grep -o '.*\.o')
  echo "$objlist" | tr ' ' '\n' | grep -v -E $blacklist >libwebrtc_full.list
  # various intrinsics aren't included by default in .ninja_deps
  local extras=$(find \
    ./obj/third_party/libvpx/libvpx_* \
    ./obj/third_party/libjpeg_turbo/simd_asm \
    ./obj/third_party/boringssl/boringssl_asm -name '*.o')
  echo "$extras" | tr ' ' '\n' >>libwebrtc_full.list
  # generate the archive
  cat libwebrtc_full.list | xargs ar -crs libwebrtc_full.a
  # generate an index list
  ranlib libwebrtc_full.a
  popd >/dev/null
}

# BASED ON http://technicaladventure.blogspot.com/2017/10/compiling-webrtc-on-ubuntu.html
function combine-includes() {
	mkdir include
	mkdir include/webrtc
	find ./ -name *.h -exec cp --parents '{}' include/webrtc ';'
	# SEE https://github.com/vsimon/webrtcbuilds/blob/master/util.sh#L345
	find . -path './third_party*' -prune -o -name '*.h' -exec cp --parents '{}' include/webrtc ';'
}

compile-unix "out/release" "$common_args $target_args"
combine-includes

# USAGE https://github.com/BrandonMakin/Godot-Module-WebRTC/blob/master/godot/modules/webrtc/config.py
# USAGE https://github.com/BrandonMakin/Godot-Module-WebRTC/blob/master/godot/modules/webrtc/SCsub

extra.txt

sudo apt-get install libssl-dev
gn gen out/release --args='target_os="linux" enable_iterator_debugging=false is_component_build=false is_debug=false use_custom_libcxx=false proprietary_codecs=true use_custom_libcxx_for_host=false'
export GYP_DEFINES="target_arch=x64 host_arch=x64 build_with_chromium=0 rtc_build_ssl=0 use_openssl=0 use_gtk=0 use_x11=0 include_examples=0 include_tests=0 fastbuild=1 remove_webcore_debug_symbols=1 include_pulse_audio=0 include_internal_video_render=0 clang=0 "
sudo apt-get install libcurl4
find "$(cd .; pwd)" -name "*.a"

test1.cpp

#include <iostream>
#include <unistd.h>
/*
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/jsep.h"
#include "webrtc/api/datachannelinterface.h"
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/pc/test/mockpeerconnectionobservers.h"
#include "webrtc/base/logging.h"
#include "webrtc/rtc_base/ssladapter.h"
#include "webrtc/rtc_base/sslstreamadapter.h"
#include "webrtc/rtc_base/thread.h"
//#include "webrtc/p2p/base/common.h"
//#include "webrtc/base/basictypes.h"
#include <functional>         // std::function
#include <iostream>           // remove eventually
//#include "os/mutex.h"         // mutex // necessary to include this? it seemed to work find without including
//#include "ustring.h"          // String
//#include "reference.h"
//#include "io/packet_peer.h"
//#include "ring_buffer.h"      // RingBuffer
#include "api/peerconnectioninterface.h"
#include "media/base/mediaengine.h"       // needed for CreateModularPeerConnectionFactory
//using rtc::scoped_ptr;
using rtc::scoped_refptr;
using webrtc::MediaStreamInterface;
using webrtc::CreatePeerConnectionFactory;
using webrtc::DataChannelInterface;
using webrtc::MockDataChannelObserver;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
//using webrtc::PortAllocatorFactoryInterface;
//using webrtc::VideoSourceInterface;
using webrtc::VideoTrackInterface;
*/
#include "api/peerconnectioninterface.h"
int main()
{
    fprintf(stderr,"Hello World1\n");
    fprintf(stderr,"Initializing SSL\n");

    //rtc::InitializeSSL();

    fprintf(stderr,"Initialized SSL\n");
    fprintf(stderr,"Closing SSL\n");

    //rtc::CleanupSSL();

  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory;
  rtc::Thread* signaling_thread;

  signaling_thread = new rtc::Thread;
  signaling_thread->Start();
  pc_factory = webrtc::CreateModularPeerConnectionFactory(
    nullptr, // rtc::Thread* network_thread,
    nullptr, // rtc::Thread* worker_thread,
    signaling_thread,
    nullptr, // std::unique_ptr<cricket::MediaEngineInterface> media_engine,
    nullptr, // std::unique_ptr<CallFactoryInterface> call_factory,
    nullptr  // std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory
  );

    fprintf(stderr,"Closed SSL\n");

    return 0;
}