[Feature] add to http api bitrate info for video/audio for CBR/VBR detection and better observability

[felipewd]

Hello!

When probing for data using the http api, I get something like:

{
  "code": 0,
  "server": "vid-6ou26s1",
  "service": "91h157e8",
  "pid": "617435",
  "stream": {
    "id": "vid-x4em786",
    "name": "test",
    "vhost": "vid-593zr55",
    "app": "live",
    "tcUrl": "srt://xx.xx.xx.xx./live",
    "url": "/live/test",
    "live_ms": 1762283099995,
    "clients": 3,
    "frames": 0,
    "send_bytes": 18876990888,
    "recv_bytes": 9501504208,
    "kbps": {
      "recv_30s": 10197,
      "send_30s": 20395
    },
    "publish": {
      "active": true,
      "cid": "s6n9b808"
    },
    "video": {
      "codec": "H264",
      "profile": "Main",
      "level": "4",
      "width": 1920,
      "height": 1080
    },
    "audio": {
      "codec": "AAC",
      "sample_rate": 44100,
      "channel": 2,
      "profile": "LC"
    }
  }
}

However, nginx-rtmp can report the bitrate of audio and video, which provides for greater observability for streams using CBR, VBR, etc and can help detect issues.

nginx provides this as bw_video and bw_audio, as such:

<name>nginx_test</name>
<time>6004423</time><bw_in>5758864</bw_in>
<bytes_in>3851548334</bytes_in>
<bw_out>5758864</bw_out>
<bytes_out>3843815196</bytes_out>
<bw_audio>147048</bw_audio> # <=== Audio bitrate in bytes
<bw_video>5611816</bw_video> # <==== Video bitrate in bytes

It'd be great if srs can provide this as well!

[winlinvip]

It make sense, but it's too complex for SRS to support it. Nginx-RTMP support it because it only support RTMP. For SRS, difficult to provide this audio and video bandwidth for all protocols, for example, for HLS. While for WebRTC, there may be 2 video streams.

[winlinvip]

The frames is number of video frames for RTMP, or video packets for SRT:

{
  "streams": [
    {
      "name": "livestream",
      "tcUrl": "rtmp://localhost:1935/live",
      "clients": 1,
      "frames": 0, // video frames.
      "send_bytes": 4330,
      "recv_bytes": 28938,
      "kbps": {
        "recv_30s": 0,
        "send_30s": 0
      }
    }
  ]
}

So it's good to have the audio packets either.

{
  "streams": [
    {
      "name": "livestream",
      "tcUrl": "rtmp://localhost:1935/live",
      "clients": 1,
      "frames": 3, // total frames.
      "video_frames": 1, // video frames.
      "audio_frames": 2, // audio frames.
      "send_bytes": 4330,
      "recv_bytes": 28938,
      "kbps": {
        "recv_30s": 0,
        "send_30s": 0
      }
    }
  ]
}

With the audio/video frames, you are somehow know the quality of audio/video stream, just like the audio/video bandwidth.