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Added system library option for ffmpeg, srtp, srt libraries. v5.0.193 (#3846)
--------- Co-authored-by: john <hondaxiao@tencent.com>
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parent
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commit
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5 changed files with 62 additions and 7 deletions
15
trunk/configure
vendored
15
trunk/configure
vendored
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@ -152,6 +152,10 @@ if [[ $SRS_SHARED_ST == YES ]]; then LibSTfile="-L${LibSTRoot} -lst"; fi
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# srtp
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if [[ $SRS_RTC == YES ]]; then
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LibSrtpRoot="${SRS_OBJS}/srtp2/include"; LibSrtpFile="${SRS_OBJS}/srtp2/lib/libsrtp2.a"
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if [[ $SRS_USE_SYS_SRTP == YES ]]; then
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LibSrtpRoot=""; LibSrtpFile="libsrtp2.a"
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if [[ $SRS_SHARED_SRTP == YES ]]; then LibSrtpFile="-lsrtp2"; fi
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fi
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fi
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# FFMPEG for WebRTC transcoding, such as aac to opus.
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@ -167,6 +171,13 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
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LibFfmpegFile="$LibFfmpegFile -L${SRS_OBJS}/opus/lib -lopus"
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fi
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fi
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if [[ $SRS_USE_SYS_FFMPEG == YES ]]; then
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LibFfmpegRoot=""
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LibFfmpegFile="libavcodec.a libswresample.a libavutil.a libopus.a";
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if [[ $SRS_SHARED_FFMPEG == YES ]]; then
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LibFfmpegFile="-lavcodec -lswresample -lavutil -lopus"
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fi
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fi
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fi
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# openssl-1.1.0e, for the RTMP complex handshake.
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@ -188,6 +199,10 @@ fi
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if [[ $SRS_SRT == YES ]]; then
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LibSRTRoot="${SRS_OBJS}/srt/include"; LibSRTfile="${SRS_OBJS}/srt/lib/libsrt.a"
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if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-L${SRS_OBJS}/srt/lib -lsrt"; fi
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if [[ $SRS_USE_SYS_SRT == YES ]]; then
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LibSRTRoot=""; LibSRTfile="libsrt.a"
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if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-lsrt"; fi
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fi
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fi
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# the link options, always use static link
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