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make webrtc audio work
This commit is contained in:
parent
68ad006b73
commit
a0a4337214
6 changed files with 790 additions and 16 deletions
22
trunk/configure
vendored
22
trunk/configure
vendored
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@ -150,6 +150,8 @@ LibSTRoot="${SRS_OBJS_DIR}/st"; LibSTfile="${LibSTRoot}/libst.a"
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if [[ $SRS_SHARED_ST == YES ]]; then LibSTfile="-lst"; fi
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# srtp
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LibSrtpRoot="${SRS_OBJS_DIR}/srtp2/include"; LibSrtpFile="${SRS_OBJS_DIR}/srtp2/lib/libsrtp2.a"
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# ffmpeg
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LibFfmpegRoot="${SRS_OBJS_DIR}/ffmpeg/include"; LibFfmpegFile="${SRS_OBJS_DIR}/ffmpeg/lib/libavcodec.a ${SRS_OBJS_DIR}/ffmpeg/lib/libswresample.a ${SRS_OBJS_DIR}/ffmpeg/lib/libavutil.a ${SRS_OBJS_DIR}/ffmpeg/lib/libopus.a -lpthread"
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# openssl-1.1.0e, for the RTMP complex handshake.
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LibSSLRoot="";LibSSLfile=""
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if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == NO ]]; then
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@ -235,7 +237,7 @@ fi
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if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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MODULE_ID="SERVICE"
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MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL")
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
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MODULE_FILES=("srs_service_log" "srs_service_st" "srs_service_http_client"
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"srs_service_http_conn" "srs_service_rtmp_conn" "srs_service_utility"
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"srs_service_conn")
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@ -248,7 +250,7 @@ fi
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if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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MODULE_ID="APP"
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MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "SERVICE")
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
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MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_source"
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"srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder" "srs_app_http_stream"
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"srs_app_thread" "srs_app_bandwidth" "srs_app_st" "srs_app_log" "srs_app_config"
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@ -259,7 +261,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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"srs_app_mpegts_udp" "srs_app_rtp" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
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"srs_app_caster_flv" "srs_app_process" "srs_app_ng_exec"
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"srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
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"srs_app_coworkers" "srs_app_hybrid")
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"srs_app_coworkers" "srs_app_hybrid" "srs_app_audio_recode")
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DEFINES=""
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# add each modules for app
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for SRS_MODULE in ${SRS_MODULES[*]}; do
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@ -286,7 +288,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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if [[ $SRS_SRT == YES ]]; then
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MODULE_DEPENDS+=("SRT")
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fi
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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if [[ $SRS_SRT == YES ]]; then
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ModuleLibIncs+=("${LibSRTRoot[*]}")
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fi
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@ -299,7 +301,7 @@ fi
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if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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MODULE_ID="MAIN"
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MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "SERVICE")
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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MODULE_FILES=()
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DEFINES=""
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# add each modules for main
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@ -326,13 +328,13 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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done
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#
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# all depends libraries
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibSSLfile} ${LibGperfFile})
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile} ${LibGperfFile})
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if [[ $SRS_SRT == YES ]]; then
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ModuleLibFiles+=("${LibSRTfile[*]}")
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fi
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# all depends objects
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MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${SERVICE_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
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if [[ $SRS_SRT == YES ]]; then
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MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
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fi
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@ -343,7 +345,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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#
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# For modules, without the app module.
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MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${SERVICE_OBJS[@]} ${MAIN_OBJS[@]}"
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibSSLfile} ${LibGperfFile})
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile} ${LibGperfFile})
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#
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for SRS_MODULE in ${SRS_MODULES[*]}; do
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. $SRS_MODULE/config
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@ -363,11 +365,11 @@ if [ $SRS_UTEST = YES ]; then
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MODULE_FILES=("srs_utest" "srs_utest_amf0" "srs_utest_protocol" "srs_utest_kernel" "srs_utest_core"
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"srs_utest_config" "srs_utest_rtmp" "srs_utest_http" "srs_utest_avc" "srs_utest_reload"
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"srs_utest_mp4" "srs_utest_service" "srs_utest_app")
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ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSrtpRoot} ${LibSSLRoot})
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ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${LibSSLRoot})
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if [[ $SRS_SRT == YES ]]; then
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ModuleLibIncs+=("${LibSRTRoot[*]}")
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fi
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibSSLfile})
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ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile})
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if [[ $SRS_SRT == YES ]]; then
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ModuleLibFiles+=("${LibSRTfile[*]}")
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fi
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