objs/srs_bench -sfu live --help 案例都是无媒体传输，是不是本部分测试不支持推流测试

[pethrowilo]

Usage: objs/srs_bench [Options]
Options:
   -sfu    The target server that can be rtc, live, janus, or gb28181. Default: rtc
               rtc/srs: SRS WebRTC SFU server, for WebRTC/WHIP/WHEP.
               live: SRS live streaming server, for RTMP/HTTP-FLV/HLS.
               janus: Janus WebRTC SFU server, for janus private protocol.
   -sn     The number of streams to simulate. Variable: %d. Default: 1
   -delay  The start delay in ms for each client or stream to simulate. Default: 50
   -stat   [Optional] The stat server API listen port.
Publisher:
   -pr     The url to publish. If sn exceed 1, auto append variable %d.
   -cap    Whether to close connection after publish. Default: false

例如，1个推流，无媒体传输:
   objs/srs_bench -pr=rtmp://localhost/live/livestream -cap=true

例如，2个推流，无媒体传输：
   objs/srs_bench -pr=rtmp://localhost/live/livestream_%d -sn=2 -cap=true

Usage: objs/srs_bench [Options]
Options:
   -sfu    The target server that can be rtc, live, janus, or gb28181. Default: rtc
               rtc/srs: SRS WebRTC SFU server, for WebRTC/WHIP/WHEP.
               live: SRS live streaming server, for RTMP/HTTP-FLV/HLS.
               janus: Janus WebRTC SFU server, for janus private protocol.
   -sn     The number of streams to simulate. Variable: %d. Default: 1
   -delay  The start delay in ms for each client or stream to simulate. Default: 50
   -stat   [Optional] The stat server API listen port.
Publisher:
   -pr     The url to publish. If sn exceed 1, auto append variable %d.
   -cap    Whether to close connection after publish. Default: false

例如，1个推流，无媒体传输:
   objs/srs_bench -pr=rtmp://localhost/live/livestream -cap=true

例如，2个推流，无媒体传输：
   objs/srs_bench -pr=rtmp://localhost/live/livestream_%d -sn=2 -cap=true

添加 -sv -sa 对应文件时也显示 flag provided but not defined: -sa, rtmp 测试是没有必要还是暂未实现？