1. windows10下編譯32位和64位webrtc(m77)靜態庫
省略掛代理下載depot_tools以及webrtc程式碼的過程。。。
可參考webrtc編譯
務必在 cmd 終端環境下進入到 webrtc\src 目錄，再執行以下操作！
1.1. 環境配置
- 在系統環境變數下編輯PATH，將depot_tools所在路徑放在PATH變數最前面。
- 設定環境變數
DEPOT_TOOLS_WIN_TOOLCHAIN=0
1.2. 編譯流程
-
檢出m77版本的webrtc
git checkout -b m77 remotes/branch-heads/m77gclient sync -Dm77版本對應詳情：
commit ad73985e75684cb4ac4dadb9d3d86ad0d66612a0 (HEAD -> m77, branch-heads/m77) Author: Henrik Boström <hbos@webrtc.org> Date: Wed Aug 21 12:09:51 2019 +0200 Add implemented-but-missing members to RTCMediaStreamTrackStats::Members ... -
配置編譯引數
gn args out\m77在彈出的
out\m77\args.gn文字檔案中輸入編譯引數，或者直接以命令列引數形式攜帶。 -
生成ninja編譯指令碼，並生成vs2017工程檔案
gn gen out\m77 --ide=vs2017 -
使用ninja編譯，將過程日誌記錄到檔案
ninja -C out\m77 -d stats >> out\m77\comp.log
1.3. 先說遇到的問題
-
ImportError: No module named win32file
[236/2777] LIB obj/media/rtc_constants.lib FAILED: obj/media/rtc_constants.lib e:/google/depot_tools/bootstrap-2@3_8_10_chromium_23_bin/python/bin/python.exe ../../build/toolchain/win/tool_wrapper.py link-wrapper environment.x64 False lib.exe /nologo /ignore:4221 /OUT:obj/media/rtc_constants.lib @obj/media/rtc_constants.lib.rsp Traceback (most recent call last): File "../../build/toolchain/win/tool_wrapper.py", line 31, in <module> import win32file # pylint: disable=import-error ImportError: No module named win32file [237/2777] CXX obj/logging/rtc_stream_config/rtc_stream_config.obj參考Build issue with M76 version of webrtc on Windows
使用以下命令
python -m pip install pywin32最好保證系統環境變數路徑中沒有其他版本的python影響。
然後繼續執行編譯流程中第4步。
-
fatal error C1189: #error: "See: bugs.webrtc.
這個問題在編譯引數
is_clang=false時才會出現[1210/2535] CXX obj/modules/video_coding/webrtc_h264/h264_encoder_impl.obj FAILED: obj/modules/video_coding/webrtc_h264/h264_encoder_impl.obj ... E:\google\webrtc\src\modules/video_coding/codecs/h264/h264_encoder_impl.h(21): fatal error C1189: #error: "See: bugs.webrtc.org/9213#c13."註釋掉該行之後繼續執行編譯流程中第4步，發現後續還會出現類似的報錯
E:\google\webrtc\src\modules/video_coding/codecs/h264/h264_color_space.h(20): fatal error C1189: #error: "See: bugs.webrtc.org/9213#c13."E:\google\webrtc\src\modules/video_coding/codecs/h264/h264_decoder_impl.h(21): fatal error C1189: #error: "See: bugs.webrtc.org/9213#c13."，所以一次性註釋掉三個檔案中將會報錯的行：
See: bugs.webrtc.org/9213#c13.- modules/video_coding/codecs/h264/h264_decoder_impl.h(21)
- modules/video_coding/codecs/h264/h264_encoder_impl.h(21)
- modules/video_coding/codecs/h264/h264_color_space.h(20)
然後繼續執行編譯流程中第4步。
-
error C2059: 語法錯誤:“字串”
[13/1314] CC obj/third_party/ffmpeg/ffmpeg_internal/pcm.obj FAILED: obj/third_party/ffmpeg/ffmpeg_internal/pcm.obj ... ../../third_party/ffmpeg/libavcodec/pcm.c(629): error C2059: 語法錯誤:“字串”同樣註釋掉報錯這一行，然後繼續執行編譯流程中第4步。
-
關於應用程式在webrtc.lib連結靜態庫時報錯
Linker can't find CreatePeerConnectionFactory after M77 update
修改方法
148073: Add missing dependencies to the static library
然後重新生成會增加：create_peerconnection_factory.lib
同時更新：webrtc.lib再次編譯後報另外的連線錯誤：
1>webrtc.lib(http_common.obj) : error LNK2019: 無法解析的外部符號 _AcquireCredentialsHandleA@36，該符號在函式 "enum rtc::HttpAuthResult... 中被引用 1>webrtc.lib(http_common.obj) : error LNK2019: 無法解析的外部符號 __imp__FreeCredentialsHandle@4，該符號在函式 "public: virtual __thiscall rtc::`anonymous namespace'::NegotiateAuthContext::~NegotiateAuthContext(void)"... 中被引用 1>webrtc.lib(http_common.obj) : error LNK2019: 無法解析的外部符號 _InitializeSecurityContextA@48，該符號在函式 "enum rtc::HttpAuthResult... 中被引用 1>webrtc.lib(http_common.obj) : error LNK2019: 無法解析的外部符號 _CompleteAuthToken@8，該符號在函式 "enum rtc::HttpAuthResult __cdecl rtc::HttpAuthenticate... 中被引用 1>webrtc.lib(http_common.obj) : error LNK2019: 無法解析的外部符號 __imp__DeleteSecurityContext@4，該符號在函式 "public: virtual __thiscall rtc::`anonymous namespace'::NegotiateAuthContext::~NegotiateAuthContext(void)"... 中被引用 1>... : fatal error LNK1120: 5 個無法解析的外部命令解決方法：
連結依賴庫中增加Secur32.lib
微軟的解釋：包含了security.h的標頭檔案要加#pragma comment(lib,"Secur32.lib")
AcquireCredentialsHandleA function (sspi.h)在這兩個檔案中都有對security.h的引用：
rtc_base/http_common.cc:19:#include <security.h>
rtc_base/socket_adapters.cc:27:#include <security.h> -
無法解析的外部符號 avpriv_emms_asm
這個問題在
is_clang=false編譯64位庫之後連結webrtc靜態庫時出現雖然生成了webrtc.lib，但在應用程式或者dll在連結webrtc時會報錯
4> 正在建立庫 ... 和物件 ... 4>webrtc.lib(autorename_libavcodec_utils.obj) : error LNK2019: 無法解析的外部符號 avpriv_emms_asm，該符號在函式 avcodec_default_execute 中被引用 4>webrtc.lib(decode.obj) : error LNK2001: 無法解析的外部符號 avpriv_emms_asm 4>webrtc.lib(vp3.obj) : error LNK2001: 無法解析的外部符號 avpriv_emms_asm 4>webrtc.lib(pcm.obj) : error LNK2001: 無法解析的外部符號 avpriv_emms_asm 4>E:\gitlab\sdk\zkms_client\build-win64\sdk\Debug\zkmsclient.dll : fatal error LNK1120: 1 個無法解析的外部命令根據網上有限的資料顯示，可能是X64不支援大多數彙編指令，webrtc原始碼的第三方目錄ffmpeg裡面有彙編程式碼。用clang編譯沒這個問題，但是clang編譯的庫vc不能引用。只能替換ffmpeg原始碼或者在windows上編譯 好ffmpeg，然後將ffmpeg的include和lib檔案新增到webrtc專案。
但是筆者後來使用clang編譯的庫在vc下仍能正常使用。
參考：
webrtc 支援openh264
在win10上編譯webRTC（問題篇）
windows上編譯webrtc_m84支援h.264編解碼遇到的問題總結
webrtc 4577 version Build error，unresoved symbol avpriv_emms_asm, build with vs2017，x64
1.4. 編譯引數
在 gn args out\m77 命令執行之後彈出的文字檔案中就是編譯時需要的所有引數，關於全部引數及說明可以使用命令 gn args out\m77 --list 檢視。
1.4.1. 32位編譯引數示例
# Build arguments go here.
# See "gn args <out_dir> --list" for available build arguments.
is_clang = true
is_debug = true
proprietary_codecs = true
rtc_build_examples = false
rtc_build_tools = false
rtc_include_internal_audio_device = false
rtc_include_pulse_audio = false
rtc_include_tests = false
rtc_libvpx_build_vp9 = false
rtc_use_gtk = false
rtc_use_h264 = true
target_cpu = "x86"
treat_warnings_as_errors = false
use_aura = false
use_custom_libcxx = false
use_gold = true
use_lld = false
use_ozone = true
use_rtti = true
1.4.2. 64位編譯引數示例
只需將上述 target_cpu 的值換成 "x64" 即可。
1.5. 生成目標
如果一切順利，最終生成一大堆lib檔案，實際上我們只需要 webrtc.lib 檔案足矣。
展開/摺疊
out/m77/obj/api/audio_codecs/builtin_audio_decoder_factory.lib out/m77/obj/api/audio_codecs/builtin_audio_encoder_factory.lib out/m77/obj/api/audio_codecs/g711/audio_decoder_g711.lib out/m77/obj/api/audio_codecs/g711/audio_encoder_g711.lib out/m77/obj/api/audio_codecs/g722/audio_decoder_g722.lib out/m77/obj/api/audio_codecs/g722/audio_encoder_g722.lib out/m77/obj/api/audio_codecs/ilbc/audio_decoder_ilbc.lib out/m77/obj/api/audio_codecs/ilbc/audio_encoder_ilbc.lib out/m77/obj/api/audio_codecs/isac/audio_decoder_isac_float.lib out/m77/obj/api/audio_codecs/isac/audio_encoder_isac_float.lib out/m77/obj/api/audio_codecs/L16/audio_decoder_L16.lib out/m77/obj/api/audio_codecs/L16/audio_encoder_L16.lib out/m77/obj/api/audio_codecs/opus/audio_decoder_multiopus.lib out/m77/obj/api/audio_codecs/opus/audio_decoder_opus.lib out/m77/obj/api/audio_codecs/opus/audio_encoder_opus_config.lib out/m77/obj/api/libjingle_peerconnection_api.lib out/m77/obj/api/transport/goog_cc.lib out/m77/obj/api/transport/network_control.lib out/m77/obj/api/video/builtin_video_bitrate_allocator_factory.lib out/m77/obj/api/video_codecs/builtin_video_decoder_factory.lib out/m77/obj/api/video_codecs/builtin_video_encoder_factory.lib out/m77/obj/api/video_codecs/rtc_software_fallback_wrappers.lib out/m77/obj/api/video_codecs/vp8_temporal_layers_factory.lib out/m77/obj/audio/audio.lib out/m77/obj/audio/utility/audio_frame_operations.lib out/m77/obj/call/call.lib out/m77/obj/common_audio/common_audio.lib out/m77/obj/common_audio/common_audio_sse2.lib out/m77/obj/common_video/common_video.lib out/m77/obj/logging/rtc_event_log_impl_encoder.lib out/m77/obj/media/rtc_audio_video.lib out/m77/obj/media/rtc_constants.lib out/m77/obj/media/rtc_data.lib out/m77/obj/media/rtc_encoder_simulcast_proxy.lib out/m77/obj/media/rtc_internal_video_codecs.lib out/m77/obj/media/rtc_media_base.lib out/m77/obj/media/rtc_simulcast_encoder_adapter.lib out/m77/obj/modules/audio_coding/audio_coding.lib out/m77/obj/modules/audio_coding/audio_coding_opus_common.lib out/m77/obj/modules/audio_coding/audio_encoder_cng.lib out/m77/obj/modules/audio_coding/audio_network_adaptor.lib out/m77/obj/modules/audio_coding/audio_network_adaptor_config.lib out/m77/obj/modules/audio_coding/g711.lib out/m77/obj/modules/audio_coding/g722.lib out/m77/obj/modules/audio_coding/ilbc.lib out/m77/obj/modules/audio_coding/isac.lib out/m77/obj/modules/audio_coding/isac_c.lib out/m77/obj/modules/audio_coding/isac_common.lib out/m77/obj/modules/audio_coding/legacy_encoded_audio_frame.lib out/m77/obj/modules/audio_coding/neteq.lib out/m77/obj/modules/audio_coding/pcm16b.lib out/m77/obj/modules/audio_coding/webrtc_cng.lib out/m77/obj/modules/audio_coding/webrtc_multiopus.lib out/m77/obj/modules/audio_coding/webrtc_opus.lib out/m77/obj/modules/audio_mixer/audio_frame_manipulator.lib out/m77/obj/modules/audio_mixer/audio_mixer_impl.lib out/m77/obj/modules/audio_processing/aec3/aec3.lib out/m77/obj/modules/audio_processing/audio_buffer.lib out/m77/obj/modules/audio_processing/audio_processing.lib out/m77/obj/modules/audio_processing/config.lib out/m77/obj/modules/audio_processing/vad/vad.lib out/m77/obj/modules/bitrate_controller/bitrate_controller.lib out/m77/obj/modules/congestion_controller/congestion_controller.lib out/m77/obj/modules/congestion_controller/goog_cc/goog_cc.lib out/m77/obj/modules/congestion_controller/rtp/transport_feedback.lib out/m77/obj/modules/desktop_capture/desktop_capture_differ_sse2.lib out/m77/obj/modules/desktop_capture/desktop_capture_generic.lib out/m77/obj/modules/desktop_capture/primitives.lib out/m77/obj/modules/pacing/pacing.lib out/m77/obj/modules/remote_bitrate_estimator/remote_bitrate_estimator.lib out/m77/obj/modules/rtp_rtcp/rtp_rtcp.lib out/m77/obj/modules/utility/utility.lib out/m77/obj/modules/video_capture/video_capture_module.lib out/m77/obj/modules/video_coding/encoded_frame.lib out/m77/obj/modules/video_coding/nack_module.lib out/m77/obj/modules/video_coding/packet.lib out/m77/obj/modules/video_coding/video_coding.lib out/m77/obj/modules/video_coding/webrtc_h264.lib out/m77/obj/modules/video_coding/webrtc_multiplex.lib out/m77/obj/modules/video_coding/webrtc_vp8.lib out/m77/obj/modules/video_coding/webrtc_vp8_temporal_layers.lib out/m77/obj/modules/video_coding/webrtc_vp9.lib out/m77/obj/modules/video_coding/webrtc_vp9_helpers.lib out/m77/obj/modules/video_processing/video_processing.lib out/m77/obj/modules/video_processing/video_processing_sse2.lib out/m77/obj/p2p/libstunprober.lib out/m77/obj/p2p/rtc_p2p.lib out/m77/obj/pc/peerconnection.lib out/m77/obj/pc/rtc_pc_base.lib out/m77/obj/rtc_base/experiments/alr_experiment.lib out/m77/obj/rtc_base/experiments/audio_allocation_settings.lib out/m77/obj/rtc_base/experiments/balanced_degradation_settings.lib out/m77/obj/rtc_base/experiments/cpu_speed_experiment.lib out/m77/obj/rtc_base/experiments/field_trial_parser.lib out/m77/obj/rtc_base/experiments/jitter_upper_bound_experiment.lib out/m77/obj/rtc_base/experiments/keyframe_interval_settings_experiment.lib out/m77/obj/rtc_base/experiments/normalize_simulcast_size_experiment.lib out/m77/obj/rtc_base/experiments/quality_scaler_settings.lib out/m77/obj/rtc_base/experiments/quality_scaling_experiment.lib out/m77/obj/rtc_base/experiments/rate_control_settings.lib out/m77/obj/rtc_base/experiments/rtt_mult_experiment.lib out/m77/obj/rtc_base/rtc_base.lib out/m77/obj/rtc_base/rtc_numerics.lib out/m77/obj/rtc_base/weak_ptr.lib out/m77/obj/stats/rtc_stats.lib out/m77/obj/system_wrappers/system_wrappers.lib out/m77/obj/third_party/boringssl/boringssl.lib out/m77/obj/third_party/boringssl/boringssl_asm.lib out/m77/obj/third_party/ffmpeg/ffmpeg_internal.lib out/m77/obj/third_party/ffmpeg/ffmpeg_nasm.lib out/m77/obj/third_party/libjpeg_turbo/libjpeg.lib out/m77/obj/third_party/libjpeg_turbo/simd.lib out/m77/obj/third_party/libjpeg_turbo/simd_asm.lib out/m77/obj/third_party/libsrtp/libsrtp.lib out/m77/obj/third_party/libvpx/libvpx.lib out/m77/obj/third_party/libvpx/libvpx_yasm.lib out/m77/obj/third_party/openh264/openh264_common_yasm.lib out/m77/obj/third_party/openh264/openh264_encoder_yasm.lib out/m77/obj/third_party/openh264/openh264_processing_yasm.lib out/m77/obj/third_party/opus/opus.lib out/m77/obj/third_party/pffft/pffft.lib out/m77/obj/third_party/usrsctp/usrsctp.lib out/m77/obj/third_party/yasm/yasm_utils.lib out/m77/obj/video/video.lib out/m77/obj/webrtc.lib out/m77/win_clang_x64/obj/third_party/libyuv/libyuv_internal.lib1.6. 提取標頭檔案和庫檔案
1.6.1. 庫檔案提取
新建批處理檔案 gen-webrtc-lib.bat，鍵入以下內容。
echo off
:: 定義源目錄
set sourcePath=E:\google\webrtc\src\out\m77\obj
:: 定義目標路徑
set resulePath=E:\google\webrtc\lib
xcopy %sourcePath%\*.lib %resulePath%\ /s /c /y /h /r /f
pause
執行批處理檔案 gen-webrtc-lib.bat，webrtc的所有編譯好的庫檔案會複製到目標路徑下。
1.6.2. 標頭檔案提取
新建批處理檔案 gen-webrtc-inc.bat，鍵入以下內容。
echo off
:: 定義源目錄
set sourcePath=E:\google\webrtc\src
:: 定義目標路徑
set resulePath=E:\google\webrtc\include
robocopy %sourcePath% %resulePath% *.h *.hpp *.hxx ^
/s /mt /log:robocopy-log.txt /fp /ndl ^
/xd ^
"%sourcePath%\.git" ^
"%sourcePath%\build" ^
"%sourcePath%\build_overrides" ^
"%sourcePath%\buildtools" ^
"%sourcePath%\data" ^
"%sourcePath%\examples" ^
"%sourcePath%\out" ^
"%sourcePath%\rtc_tools" ^
"%sourcePath%\resources" ^
"%sourcePath%\test" ^
"%sourcePath%\testing" ^
"%sourcePath%\tools" ^
"%sourcePath%\tools_webrtc" ^
"%sourcePath%\third_party\blink" ^
"%sourcePath%\third_party\depot_tools" ^
"%sourcePath%\third_party\catapult"
pause
執行批處理檔案 gen-webrtc-inc.bat，webrtc的標頭檔案會以源目錄結構形式複製到目標路徑下。
有關robocopy的使用，參考robocopy