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Update WebRTC to branch-heads/52 branch 

WebRTC change
- rtc::scoped_ptr to std::unique_ptr
- rtc::Buffer to rtc::CopyOnWriteBuffer
master
Ryan Lee 2016-06-08 11:45:45 +09:00
parent 00b44bfea0
commit e0d9e63914
6 changed files with 343 additions and 402 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

723
cmake/FindWebRTC.cmake
View File

@ -1,391 +1,332 @@
# The following variables will be defined:
#
# WEBRTC_FOUND
# WEBRTC_DEFINES
# WEBRTC_INCLUDE_DIR
# WEBRTC_LIBRARIES
#
# WEBRTC_BORING_SSL_INCLUDE
# WEBRTC_BORING_SSL_LIBRARIES
#
# ============================================================================
# WebRTC root and default library directory
# ============================================================================
message("Check WebRTC library")
if (DEFINED ENV{WEBRTC_ROOT_DIR})
set(WEBRTC_ROOT_DIR $ENV{WEBRTC_ROOT_DIR})
message("WEBRTC_ROOT_DIR = '${WEBRTC_ROOT_DIR}' from environment variable")
else()
set(WEBRTC_ROOT_DIR
"webrtc-checkout/src"
CACHE PATH
"WebRTC root directory."
)
endif()
if (DEFINED ENV{WEBRTC_LIBRARY_DIR})
set(WEBRTC_LIBRARY_DIR $ENV{WEBRTC_LIBRARY_DIR})
message("WEBRTC_LIBRARY_DIR = '${WEBRTC_LIBRARY_DIR}' from environment variable")
else()
set(WEBRTC_LIBRARY_DIR ${WEBRTC_ROOT_DIR}/out/Release)
endif()
if (DEFINED ENV{WEBRTC_LIBRARY_DIR_DEBUG})
set(WEBRTC_LIBRARY_DIR_DEBUG $ENV{WEBRTC_LIBRARY_DIR_DEBUG})
message("WEBRTC_LIBRARY_DIR = '${WEBRTC_LIBRARY_DIR}' from environment variable")
else()
set(WEBRTC_LIBRARY_DIR_DEBUG ${WEBRTC_ROOT_DIR}/out/Debug)
endif()
# ============================================================================
# Find WebRTC header directory
# ============================================================================
find_path(WEBRTC_INCLUDE_DIR
NAMES
webrtc/config.h
PATHS
${WEBRTC_ROOT_DIR}
)
list(APPEND WEBRTC_INCLUDE_DIR
"${WEBRTC_INCLUDE_DIR}"
"${WEBRTC_INCLUDE_DIR}/webrtc"
"${WEBRTC_INCLUDE_DIR}/third_party"
"${WEBRTC_INCLUDE_DIR}/third_party/webrtc"
"${WEBRTC_INCLUDE_DIR}/third_party/wtl/include"
"${WEBRTC_INCLUDE_DIR}/third_party/jsoncpp/overrides/include"
"${WEBRTC_INCLUDE_DIR}/third_party/jsoncpp/source/include"
)
# ============================================================================
# Find WebRTC libries
# webrtc -> webrtc.lib or libwebrtc.a
# libyuv yuv -> libyuv.lib or libyuv.a
# ...
# ============================================================================
set (WEBRTC_LIBRARIES)
list(APPEND _WEBRTC_LIB_NAMES
"libyuv yuv" # Find one of them
"libjingle_peerconnection"
"field_trial_default"
"jsoncpp"
"rtc_media"
"rtc_base_approved"
"webrtc_common"
"video_render_module"
"webrtc_utility"
"audio_coding_module"
"cng"
"common_audio"
"system_wrappers"
"openmax_dl"
"common_audio_sse2"
"audio_encoder_interface"
"g711"
"pcm16b"
"ilbc"
"webrtc_opus"
"opus"
"g722"
"isac"
"audio_decoder_interface"
"isac_common"
"red"
"rtc_event_log"
"rtc_event_log_proto"
"protobuf_lite"
"neteq"
"rent_a_codec"
"media_file"
"common_video"
"libjpeg"
"webrtc"
"voice_engine"
"audio_conference_mixer"
"audio_processing"
"audioproc_debug_proto"
"audio_processing_sse2"
"audio_device"
"bitrate_controller"
"paced_sender"
"rtp_rtcp"
"remote_bitrate_estimator"
"congestion_controller"
"video_capture_module"
"video_processing"
"video_processing_sse2"
"webrtc_video_coding"
"webrtc_h264"
"webrtc_i420"
"video_coding_utility"
"webrtc_vp8"
"libvpx"
"libvpx_intrinsics_mmx"
"libvpx_intrinsics_sse2"
"libvpx_intrinsics_ssse3"
"libvpx_intrinsics_sse4_1"
"libvpx_intrinsics_avx"
"libvpx_intrinsics_avx2"
"webrtc_vp9"
"rtc_sound"
"rtc_base"
"boringssl"
"metrics_default"
"rtc_xmllite"
"expat"
"rtc_xmpp"
"rtc_p2p"
"usrsctplib"
"video_capture_module_internal_impl"
"directshow_baseclasses"
"video_render_module_internal_impl"
"rtc_pc"
"libsrtp"
)
if (MSVC)
list(APPEND _WEBRTC_LIB_NAMES
"boringssl_asm"
)
endif(MSVC)
foreach (lib ${_WEBRTC_LIB_NAMES})
unset(_WEBRTC_LIB_PATH CACHE)
separate_arguments(lib)
find_library(_WEBRTC_LIB_PATH
NAMES ${lib}
PATHS
${WEBRTC_LIBRARY_DIR}
${WEBRTC_LIBRARY_DIR}/obj/webrtc
${WEBRTC_LIBRARY_DIR}/obj/webrtc/api
${WEBRTC_LIBRARY_DIR}/obj/webrtc/base
${WEBRTC_LIBRARY_DIR}/obj/webrtc/common_audio
${WEBRTC_LIBRARY_DIR}/obj/webrtc/common_video
${WEBRTC_LIBRARY_DIR}/obj/webrtc/libjingle/xmllite
${WEBRTC_LIBRARY_DIR}/obj/webrtc/libjingle/xmpp
${WEBRTC_LIBRARY_DIR}/obj/webrtc/media
${WEBRTC_LIBRARY_DIR}/obj/webrtc/modules
${WEBRTC_LIBRARY_DIR}/obj/webrtc/modules/video_coding/utility
${WEBRTC_LIBRARY_DIR}/obj/webrtc/modules/video_coding/codecs/vp8
${WEBRTC_LIBRARY_DIR}/obj/webrtc/modules/video_coding/codecs/vp9
${WEBRTC_LIBRARY_DIR}/obj/webrtc/p2p
${WEBRTC_LIBRARY_DIR}/obj/webrtc/pc
${WEBRTC_LIBRARY_DIR}/obj/webrtc/sound
${WEBRTC_LIBRARY_DIR}/obj/webrtc/system_wrappers
${WEBRTC_LIBRARY_DIR}/obj/webrtc/voice_engine
${WEBRTC_LIBRARY_DIR}/obj/third_party/boringssl
${WEBRTC_LIBRARY_DIR}/obj/third_party/expat
${WEBRTC_LIBRARY_DIR}/obj/third_party/libjpeg_turbo
${WEBRTC_LIBRARY_DIR}/obj/third_party/libsrtp
${WEBRTC_LIBRARY_DIR}/obj/third_party/libvpx
${WEBRTC_LIBRARY_DIR}/obj/third_party/jsoncpp
${WEBRTC_LIBRARY_DIR}/obj/third_party/openmax_dl/dl
${WEBRTC_LIBRARY_DIR}/obj/third_party/opus
${WEBRTC_LIBRARY_DIR}/obj/third_party/protobuf
${WEBRTC_LIBRARY_DIR}/obj/third_party/usrsctp
${WEBRTC_LIBRARY_DIR}/obj/third_party/winsdk_samples
)
if (_WEBRTC_LIB_PATH)
list(APPEND
WEBRTC_LIBRARIES
optimized ${_WEBRTC_LIB_PATH}
)
else(_WEBRTC_LIB_PATH)
message("\nCurrent WEBRTC_ROOT_DIR is '${WEBRTC_ROOT_DIR}'")
message("Current WEBRTC_LIBRARY_DIR is '${WEBRTC_LIBRARY_DIR}'\n")
message(FATAL_ERROR "WebRTC module '${lib}' was not found. \n"
"Check 'WEBRTC_ROOT_DIR'\n")
endif()
endforeach()
# Optional DEBUG library
foreach (lib ${_WEBRTC_LIB_NAMES})
unset(_WEBRTC_LIB_PATH CACHE)
separate_arguments(lib)
find_library(_WEBRTC_LIB_PATH
NAMES ${lib}
PATHS
${WEBRTC_LIBRARY_DIR_DEBUG}
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/api
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/base
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/common_audio
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/common_video
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/libjingle/xmllite
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/libjingle/xmpp
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/media
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/modules
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/modules/video_coding/utility
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/modules/video_coding/codecs/vp8
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/modules/video_coding/codecs/vp9
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/p2p
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/pc
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/sound
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/system_wrappers
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/webrtc/voice_engine
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/boringssl
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/expat
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/libjpeg_turbo
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/libsrtp
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/libvpx
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/jsoncpp
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/openmax_dl/dl
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/opus
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/protobuf
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/usrsctp
${WEBRTC_LIBRARY_DIR_DEBUG}/obj/third_party/winsdk_samples
)
if (_WEBRTC_LIB_PATH)
list(APPEND
WEBRTC_LIBRARIES
debug ${_WEBRTC_LIB_PATH}
)
endif()
endforeach()
if(WIN32 AND MSVC)
list(APPEND
WEBRTC_LIBRARIES
Secur32.lib Winmm.lib msdmo.lib dmoguids.lib wmcodecdspuuid.lib
wininet.lib dnsapi.lib version.lib ws2_32.lib Strmiids.lib
)
endif()
if(UNIX)
find_package (Threads REQUIRED)
if (APPLE)
find_library(FOUNDATION_LIBRARY Foundation)
find_library(CORE_FOUNDATION_LIBRARY CoreFoundation)
find_library(CORE_SERVICES_LIBRARY CoreServices)
endif()
list(APPEND WEBRTC_LIBRARIES
${CMAKE_THREAD_LIBS_INIT}
${FOUNDATION_LIBRARY}
${CORE_FOUNDATION_LIBRARY}
${CORE_SERVICES_LIBRARY}
)
endif()
# BoringSSL
find_library(_WEBRTC_BORING_SSL_LIBRARY
NAMES boringssl
PATHS
${WEBRTC_LIBRARY_DIR}
${WEBRTC_LIBRARY_DIR}/obj/third_party/boringssl
)
if (_WEBRTC_BORING_SSL_LIBRARY)
list(APPEND WEBRTC_BORING_SSL_LIBRARIES ${_WEBRTC_BORING_SSL_LIBRARY})
else()
message(FATAL_ERROR "BoringSSL library not found in WebRTC")
endif()
if (MSVC)
find_library(_WEBRTC_BORING_SSL_ASM_LIBRARY
NAMES boringssl_asm
PATHS
${WEBRTC_LIBRARY_DIR}
${WEBRTC_LIBRARY_DIR}/obj/third_party/boringssl
)
if (_WEBRTC_BORING_SSL_ASM_LIBRARY)
list(APPEND WEBRTC_BORING_SSL_LIBRARIES ${_WEBRTC_BORING_SSL_ASM_LIBRARY})
else()
message(FATAL_ERROR "BoringSSL asm library not found in WebRTC")
endif()
endif(MSVC)
find_path(WEBRTC_BORING_SSL_INCLUDE
NAMES
openssl/ssl.h
PATHS
${WEBRTC_ROOT_DIR}/third_party/boringssl/src/include
)
# ============================================================================
# Definitions
# ============================================================================
set(WEBRTC_DEFINES
-DV8_DEPRECATION_WARNINGS -DCLD_VERSION=2
-DCHROMIUM_BUILD
-DUSE_LIBJPEG_TURBO=1 -DENABLE_WEBRTC=1
-DENABLE_MEDIA_ROUTER=1 -DENABLE_PEPPER_CDMS
-DENABLE_CONFIGURATION_POLICY -DENABLE_NOTIFICATIONS
-DENABLE_TOPCHROME_MD=1 -DFIELDTRIAL_TESTING_ENABLED
-DENABLE_TASK_MANAGER=1 -DENABLE_EXTENSIONS=1 -DENABLE_PDF=1
-DENABLE_PLUGIN_INSTALLATION=1 -DENABLE_PLUGINS=1
-DENABLE_SESSION_SERVICE=1 -DENABLE_THEMES=1 -DENABLE_AUTOFILL_DIALOG=1
-DENABLE_PRINTING=1 -DENABLE_BASIC_PRINTING=1 -DENABLE_PRINT_PREVIEW=1
-DENABLE_SPELLCHECK=1 -DENABLE_CAPTIVE_PORTAL_DETECTION=1
-DENABLE_APP_LIST=1
-DENABLE_SETTINGS_APP=1 -DENABLE_SUPERVISED_USERS=1 -DENABLE_MDNS=1
-DENABLE_SERVICE_DISCOVERY=1 -DV8_USE_EXTERNAL_STARTUP_DATA
-DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL
-DUSE_LIBPCI=1
-DUSE_OPENSSL=1 -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0
-DUSE_DEFAULT_RENDER_THEME=1
)
if (MSVC)
list(APPEND WEBRTC_DEFINES
-DNOMINMAX -DPSAPI_VERSION=1 -D_CRT_RAND_S
-DCERT_CHAIN_PARA_HAS_EXTRA_FIELDS
-D_ATL_NO_OPENGL -D_SECURE_ATL
# -D_HAS_EXCEPTIONS=0
-D_WINSOCK_DEPRECATED_NO_WARNINGS
-DNO_TCMALLOC -D__STD_C
-D_CRT_SECURE_NO_DEPRECATE -D_SCL_SECURE_NO_DEPRECATE
-D_CRT_NONSTDC_NO_WARNINGS
-DENABLE_CAPTIVE_PORTAL_DETECTION=1
-D_CRT_NONSTDC_NO_DEPRECATE
-D_UNICODE -DUNICODE
-DWEBRTC_WIN
)
elseif(UNIX)
list(APPEND WEBRTC_DEFINES
-DWEBRTC_POSIX
-DCARBON_DEPRECATED=YES
-DDISABLE_DYNAMIC_CAST -D_REENTRANT
-DUSE_BROWSER_SPELLCHECKER=1
)
if(APPLE)
list(APPEND WEBRTC_DEFINES
-DWEBRTC_MAC
)
else()
list(APPEND WEBRTC_DEFINES
-DWEBRTC_LINUX
-DDUI_COMPOSITOR_IMAGE_TRANSPORT
-DUSE_PANGO=1 -DUSE_CAIRO=1
-DUSE_X11=1 -DUSE_GLIB=1 -DUSE_NSS_CERTS=1
)
endif()
endif()
# ============================================================================
# Validation
# ============================================================================
if (WEBRTC_INCLUDE_DIR AND _WEBRTC_LIB_PATH)
message("Found WebRTC library successfully.")
set(WEBRTC_FOUND 1)
else()
set(WEBRTC_FOUND 0)
message(FATAL_ERROR "WebRTC file was not found.\n"
"Please check 'WEBRTC_ROOT_DIR'.\n")
endif()
# The following variables will be defined:
#
# WEBRTC_FOUND
# WEBRTC_DEFINES
# WEBRTC_INCLUDE_DIR
# WEBRTC_LIBRARIES
#
# WEBRTC_BORING_SSL_INCLUDE
# WEBRTC_BORING_SSL_LIBRARIES
#
# ============================================================================
# WebRTC root and default library directory
# ============================================================================
message("Check WebRTC library")
if (DEFINED ENV{WEBRTC_ROOT_DIR})
set(WEBRTC_ROOT_DIR $ENV{WEBRTC_ROOT_DIR})
message("WEBRTC_ROOT_DIR = '${WEBRTC_ROOT_DIR}' from environment variable")
else()
set(WEBRTC_ROOT_DIR
"webrtc-checkout/src"
CACHE PATH
"WebRTC root directory."
)
endif()
if (DEFINED ENV{WEBRTC_LIBRARY_DIR})
set(WEBRTC_LIBRARY_DIR $ENV{WEBRTC_LIBRARY_DIR})
message("WEBRTC_LIBRARY_DIR = '${WEBRTC_LIBRARY_DIR}' from environment variable")
else()
set(WEBRTC_LIBRARY_DIR ${WEBRTC_ROOT_DIR}/out/Release)
endif()
if (DEFINED ENV{WEBRTC_LIBRARY_DIR_DEBUG})
set(WEBRTC_LIBRARY_DIR_DEBUG $ENV{WEBRTC_LIBRARY_DIR_DEBUG})
message("WEBRTC_LIBRARY_DIR = '${WEBRTC_LIBRARY_DIR}' from environment variable")
else()
set(WEBRTC_LIBRARY_DIR_DEBUG ${WEBRTC_ROOT_DIR}/out/Debug)
endif()
# ============================================================================
# Find WebRTC header directory
# ============================================================================
find_path(WEBRTC_INCLUDE_DIR
NAMES
webrtc/config.h
PATHS
${WEBRTC_ROOT_DIR}
)
list(APPEND WEBRTC_INCLUDE_DIR
"${WEBRTC_INCLUDE_DIR}"
"${WEBRTC_INCLUDE_DIR}/webrtc"
"${WEBRTC_INCLUDE_DIR}/third_party"
"${WEBRTC_INCLUDE_DIR}/third_party/webrtc"
"${WEBRTC_INCLUDE_DIR}/third_party/wtl/include"
"${WEBRTC_INCLUDE_DIR}/third_party/jsoncpp/overrides/include"
"${WEBRTC_INCLUDE_DIR}/third_party/jsoncpp/source/include"
)
# ============================================================================
# Find WebRTC libries
# webrtc -> webrtc.lib or libwebrtc.a
# libyuv yuv -> libyuv.lib or libyuv.a
# ...
# ============================================================================
set (WEBRTC_LIBRARIES)
if (MSVC)
list(APPEND _WEBRTC_LIB_NAMES
"obj/webrtc/api/libjingle_peerconnection.lib"
"obj/webrtc/system_wrappers/field_trial_default.lib"
"obj/third_party/jsoncpp/jsoncpp.lib"
"obj/webrtc/media/rtc_media.lib"
"obj/webrtc/base/rtc_base_approved.lib"
"obj/webrtc/webrtc_common.lib"
"obj/webrtc/webrtc.lib"
"obj/webrtc/system_wrappers/system_wrappers.lib"
"obj/webrtc/voice_engine/voice_engine.lib"
"obj/webrtc/common_audio/common_audio.lib"
"obj/third_party/openmax_dl/dl/openmax_dl.lib"
"obj/webrtc/common_audio/common_audio_sse2.lib"
"obj/webrtc/modules/audio_coding_module.lib"
"obj/webrtc/modules/cng.lib"
"obj/webrtc/modules/audio_encoder_interface.lib"
"obj/webrtc/modules/g711.lib"
"obj/webrtc/modules/pcm16b.lib"
"obj/webrtc/modules/ilbc.lib"
"obj/webrtc/modules/webrtc_opus.lib"
"obj/third_party/opus/opus.lib"
"obj/webrtc/modules/g722.lib"
"obj/webrtc/modules/isac.lib"
"obj/webrtc/modules/audio_decoder_interface.lib"
"obj/webrtc/modules/isac_common.lib"
"obj/webrtc/modules/red.lib"
"obj/webrtc/rtc_event_log.lib"
"obj/webrtc/rtc_event_log_proto.lib"
"obj/third_party/protobuf/protobuf_lite.lib"
"obj/webrtc/modules/neteq.lib"
"obj/webrtc/modules/builtin_audio_decoder_factory.lib"
"obj/webrtc/modules/audio_decoder_factory_interface.lib"
"obj/webrtc/modules/rent_a_codec.lib"
"obj/webrtc/modules/audio_conference_mixer.lib"
"obj/webrtc/modules/audio_processing.lib"
"obj/webrtc/modules/audioproc_debug_proto.lib"
"obj/webrtc/modules/audio_processing_sse2.lib"
"obj/webrtc/modules/webrtc_utility.lib"
"obj/webrtc/modules/media_file.lib"
"obj/webrtc/modules/audio_device.lib"
"obj/webrtc/modules/bitrate_controller.lib"
"obj/webrtc/modules/paced_sender.lib"
"obj/webrtc/modules/rtp_rtcp.lib"
"obj/webrtc/modules/remote_bitrate_estimator.lib"
"obj/webrtc/modules/congestion_controller.lib"
"obj/webrtc/common_video/common_video.lib"
"libyuv.lib"
"obj/third_party/libjpeg_turbo/libjpeg.lib"
"obj/webrtc/modules/video_capture_module.lib"
"obj/webrtc/modules/video_processing.lib"
"obj/webrtc/modules/video_processing_sse2.lib"
"obj/webrtc/modules/webrtc_video_coding.lib"
"obj/webrtc/modules/webrtc_h264.lib"
"obj/webrtc/modules/webrtc_i420.lib"
"obj/webrtc/modules/video_coding/utility/video_coding_utility.lib"
"obj/webrtc/modules/video_coding/codecs/vp8/webrtc_vp8.lib"
"obj/third_party/libvpx/libvpx.lib"
"obj/third_party/libvpx/libvpx_intrinsics_mmx.lib"
"obj/third_party/libvpx/libvpx_intrinsics_sse2.lib"
"obj/third_party/libvpx/libvpx_intrinsics_ssse3.lib"
"obj/third_party/libvpx/libvpx_intrinsics_sse4_1.lib"
"obj/third_party/libvpx/libvpx_intrinsics_avx.lib"
"obj/third_party/libvpx/libvpx_intrinsics_avx2.lib"
"obj/webrtc/modules/video_coding/codecs/vp9/webrtc_vp9.lib"
"obj/webrtc/system_wrappers/metrics_default.lib"
"obj/webrtc/libjingle/xmllite/rtc_xmllite.lib"
"obj/webrtc/base/rtc_base.lib"
"obj/third_party/boringssl/boringssl.lib"
"obj/third_party/boringssl/boringssl_asm.lib"
"obj/third_party/expat/expat.lib"
"obj/webrtc/libjingle/xmpp/rtc_xmpp.lib"
"obj/webrtc/p2p/rtc_p2p.lib"
"obj/third_party/usrsctp/usrsctplib.lib"
"obj/webrtc/modules/video_capture_module_internal_impl.lib"
"obj/third_party/winsdk_samples/directshow_baseclasses.lib"
"obj/webrtc/pc/rtc_pc.lib"
"obj/third_party/libsrtp/libsrtp.lib"
)
elseif (APPLE)
elseif (UNIX)
else (MSVC)
message(FATAL_ERROR "Unknown environment. \n")
endif (MSVC)
foreach (lib ${_WEBRTC_LIB_NAMES})
unset(_WEBRTC_LIB_PATH CACHE)
separate_arguments(lib)
find_library(_WEBRTC_LIB_PATH
NAMES ${lib}
PATHS
${WEBRTC_LIBRARY_DIR}
)
if (_WEBRTC_LIB_PATH)
list(APPEND
WEBRTC_LIBRARIES
optimized ${_WEBRTC_LIB_PATH}
)
else(_WEBRTC_LIB_PATH)
message("\nCurrent WEBRTC_ROOT_DIR is '${WEBRTC_ROOT_DIR}'")
message("Current WEBRTC_LIBRARY_DIR is '${WEBRTC_LIBRARY_DIR}'\n")
message(FATAL_ERROR "WebRTC module '${lib}' was not found. \n"
"Check 'WEBRTC_ROOT_DIR'\n")
endif()
endforeach()
# Optional DEBUG library
foreach (lib ${_WEBRTC_LIB_NAMES})
unset(_WEBRTC_LIB_PATH CACHE)
separate_arguments(lib)
find_library(_WEBRTC_LIB_PATH
NAMES ${lib}
PATHS
${WEBRTC_LIBRARY_DIR_DEBUG}
)
if (_WEBRTC_LIB_PATH)
list(APPEND
WEBRTC_LIBRARIES
debug ${_WEBRTC_LIB_PATH}
)
endif()
endforeach()
if(WIN32 AND MSVC)
list(APPEND
WEBRTC_LIBRARIES
Secur32.lib Winmm.lib msdmo.lib dmoguids.lib wmcodecdspuuid.lib
wininet.lib dnsapi.lib version.lib ws2_32.lib Strmiids.lib
)
endif()
if(UNIX)
find_package (Threads REQUIRED)
if (APPLE)
find_library(FOUNDATION_LIBRARY Foundation)
find_library(CORE_FOUNDATION_LIBRARY CoreFoundation)
find_library(CORE_SERVICES_LIBRARY CoreServices)
endif()
list(APPEND WEBRTC_LIBRARIES
${CMAKE_THREAD_LIBS_INIT}
${FOUNDATION_LIBRARY}
${CORE_FOUNDATION_LIBRARY}
${CORE_SERVICES_LIBRARY}
)
endif()
# BoringSSL
find_library(_WEBRTC_BORING_SSL_LIBRARY
NAMES boringssl
PATHS
${WEBRTC_LIBRARY_DIR}
${WEBRTC_LIBRARY_DIR}/obj/third_party/boringssl
)
if (_WEBRTC_BORING_SSL_LIBRARY)
list(APPEND WEBRTC_BORING_SSL_LIBRARIES ${_WEBRTC_BORING_SSL_LIBRARY})
else()
message(FATAL_ERROR "BoringSSL library not found in WebRTC")
endif()
find_library(_WEBRTC_BORING_SSL_ASM_LIBRARY
NAMES boringssl_asm
PATHS
${WEBRTC_LIBRARY_DIR}
${WEBRTC_LIBRARY_DIR}/obj/third_party/boringssl
)
if (_WEBRTC_BORING_SSL_ASM_LIBRARY)
list(APPEND WEBRTC_BORING_SSL_LIBRARIES ${_WEBRTC_BORING_SSL_ASM_LIBRARY})
else()
message(FATAL_ERROR "BoringSSL asm library not found in WebRTC")
endif()
find_path(WEBRTC_BORING_SSL_INCLUDE
NAMES
openssl/ssl.h
PATHS
${WEBRTC_ROOT_DIR}/third_party/boringssl/src/include
)
# ============================================================================
# Definitions
# ============================================================================
if (MSVC)
set(WEBRTC_DEFINES
-DV8_DEPRECATION_WARNINGS
-DNOMINMAX -DPSAPI_VERSION=1 -D_CRT_RAND_S
-DCERT_CHAIN_PARA_HAS_EXTRA_FIELDS -DWIN32_LEAN_AND_MEAN
-D_ATL_NO_OPENGL -D_SECURE_ATL -D_HAS_EXCEPTIONS=0
-D_WINSOCK_DEPRECATED_NO_WARNINGS -DCHROMIUM_BUILD
-DUSE_AURA=1 -DUSE_DEFAULT_RENDER_THEME=1
-DUSE_LIBJPEG_TURBO=1 -DENABLE_WEBRTC=1 -DENABLE_MEDIA_ROUTER=1
-DENABLE_PEPPER_CDMS -DENABLE_NOTIFICATIONS -DENABLE_TOPCHROME_MD=1
-DFIELDTRIAL_TESTING_ENABLED -DNO_TCMALLOC -D__STD_C
-D_CRT_SECURE_NO_DEPRECATE -D_SCL_SECURE_NO_DEPRECATE
-DENABLE_TASK_MANAGER=1
-DENABLE_EXTENSIONS=1 -DENABLE_PDF=1 -DENABLE_PLUGIN_INSTALLATION=1
-DENABLE_PLUGINS=1 -DENABLE_SESSION_SERVICE=1 -DENABLE_THEMES=1
-DENABLE_PRINTING=1 -DENABLE_BASIC_PRINTING=1 -DENABLE_PRINT_PREVIEW=1
-DENABLE_SPELLCHECK=1 -DENABLE_CAPTIVE_PORTAL_DETECTION=1
-DENABLE_APP_LIST=1 -DENABLE_SETTINGS_APP=1 -DENABLE_SUPERVISED_USERS=1
-DENABLE_MDNS=1 -DENABLE_SERVICE_DISCOVERY=1
-DV8_USE_EXTERNAL_STARTUP_DATA -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD
-DSAFE_BROWSING_DB_LOCAL -DWEBRTC_WIN -DUSE_LIBPCI=1
-D_CRT_NONSTDC_NO_WARNINGS -D_CRT_NONSTDC_NO_DEPRECATE
-DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DALLOCATOR_SHIM -D_UNICODE
-DUNICODE
# Removed from origianl peerconnection_client.ninja
# -D_WIN32_WINNT=0x0A00 -DWINVER=0x0A00
# -DWIN32 -D_WINDOWS
# -DCR_CLANG_REVISION=268813-1
# -DNTDDI_VERSION=0x0A000000
# -D_USING_V110_SDK71_
# -DNDEBUG
)
elseif(UNIX)
list(APPEND WEBRTC_DEFINES
-DWEBRTC_POSIX
-DCARBON_DEPRECATED=YES
-DDISABLE_DYNAMIC_CAST -D_REENTRANT
-DUSE_BROWSER_SPELLCHECKER=1
)
if(APPLE)
list(APPEND WEBRTC_DEFINES
-DWEBRTC_MAC
)
else()
list(APPEND WEBRTC_DEFINES
-DWEBRTC_LINUX
-DDUI_COMPOSITOR_IMAGE_TRANSPORT
-DUSE_PANGO=1 -DUSE_CAIRO=1
-DUSE_X11=1 -DUSE_GLIB=1 -DUSE_NSS_CERTS=1
)
endif()
endif()
# ============================================================================
# Validation
# ============================================================================
if (WEBRTC_INCLUDE_DIR AND _WEBRTC_LIB_PATH)
message("Found WebRTC library successfully.")
set(WEBRTC_FOUND 1)
else()
set(WEBRTC_FOUND 0)
message(FATAL_ERROR "WebRTC file was not found.\n"
"Please check 'WEBRTC_ROOT_DIR'.\n")
endif()

2
src/control.h
View File

@ -24,7 +24,7 @@ class Control
public sigslot::has_slots<>,
public rtc::MessageHandler {
public:
typedef std::vector<rtc::scoped_ptr<PeerDataChannelObserver> >
typedef std::vector<std::unique_ptr<PeerDataChannelObserver> >
DataChannelList;
explicit Control();

4
src/fakeaudiocapturemodule.h
View File

@ -20,10 +20,10 @@
#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
#include <memory>
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
@ -247,7 +247,7 @@ class FakeAudioCaptureModule
bool started_;
uint32_t next_frame_time_;
rtc::scoped_ptr<rtc::Thread> process_thread_;
std::unique_ptr<rtc::Thread> process_thread_;
// Buffer for storing samples received from the webrtc::AudioTransport.
char rec_buffer_[kNumberSamples * kNumberBytesPerSample];

10
src/peer.cc
View File

@ -90,7 +90,7 @@ void PeerControl::ClosePeerConnection() {
void PeerControl::OnDataChannel(webrtc::DataChannelInterface* data_channel) {
PeerDataChannelObserver* Observer = new PeerDataChannelObserver(data_channel);
remote_data_channel_ = rtc::scoped_ptr<PeerDataChannelObserver>(Observer);
remote_data_channel_ = std::unique_ptr<PeerDataChannelObserver>(Observer);
Attach(remote_data_channel_.get());
}
@ -139,7 +139,7 @@ void PeerControl::OnIceCandidate(const webrtc::IceCandidateInterface* candidate)
void PeerControl::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
// This callback should take the ownership of |desc|.
rtc::scoped_ptr<webrtc::SessionDescriptionInterface> owned_desc(desc);
std::unique_ptr<webrtc::SessionDescriptionInterface> owned_desc(desc);
std::string sdp;
if (!desc->ToString(&sdp)) return;
@ -224,7 +224,7 @@ bool PeerControl::CreateDataChannel(
void PeerControl::AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
const std::string& candidate) {
rtc::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
peer_connection_->AddIceCandidate(owned_candidate.get());
@ -339,7 +339,7 @@ void PeerDataChannelObserver::OnMessage(const webrtc::DataBuffer& buffer) {
}
bool PeerDataChannelObserver::Send(const char* buffer, const size_t size) {
rtc::Buffer rtcbuffer(buffer, size);
rtc::CopyOnWriteBuffer rtcbuffer(buffer, size);
webrtc::DataBuffer databuffer(rtcbuffer, true);
if (channel_->buffered_amount() >= max_buffer_size_) return false;
@ -347,7 +347,7 @@ bool PeerDataChannelObserver::Send(const char* buffer, const size_t size) {
}
bool PeerDataChannelObserver::SyncSend(const char* buffer, const size_t size) {
rtc::Buffer rtcbuffer(buffer, size);
rtc::CopyOnWriteBuffer rtcbuffer(buffer, size);
webrtc::DataBuffer databuffer(rtcbuffer, true);
std::unique_lock<std::mutex> lock(send_lock_);

5
src/peer.h
View File

@ -9,6 +9,7 @@
#include <condition_variable>
#include <mutex>
#include <memory>
#include "webrtc/api/datachannelinterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
@ -123,8 +124,8 @@ protected:
std::string local_id_;
std::string remote_id_;
rtc::scoped_ptr<PeerDataChannelObserver> local_data_channel_;
rtc::scoped_ptr<PeerDataChannelObserver> remote_data_channel_;
std::unique_ptr<PeerDataChannelObserver> local_data_channel_;
std::unique_ptr<PeerDataChannelObserver> remote_data_channel_;
PeerObserver* observer_;

1
src/signalconnection.cc
View File

@ -37,7 +37,6 @@
#include <list>
#include "signalconnection.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
namespace tn {