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Is it possible to play as background music the same soundtrack to all members, controller by the event organizer, and switch sometimes between soundtracks? "Send / share" with other members upfront, keep high music quality, avoid lagging. It would be awesame for online trainings held in these days.
Looking something for web.
I think your requirements would be best suited and served by Cisco Webex meetings, and it is made free now up till certain users.
The host has all the controls required for the purposes you just mentioned.
play same soundtrack to all
send messages/share screen to all
Also, now it has an additional way to share multimedia and using this approach you can share online media in a much better way. It is really smooth experience.
This doesn't meet the requirement of sending to them upfront, but with Zoom you can share the audio that is playing on the computer at the moment.
When do you share screen, you can check the "Share computer sound" box.
The one caveat to this is that if you stop sharing, the music you're playing will stop.
For high quality audio I would recommend to use a dedicated streaming tool that does not rely on a web browser for encoding. Reason is that most audio processing and encoding built into modern web browsers is tuned for speech, not music.
Instead, I would use the use the new draft internet standard WHIP for this.
Concretely, I would use Janus Gateway as backend for the video room, e.g. with web frontend Jangouts, and then run command-line tool simple-whip-client to feed the output of a local music player into the video room.
Opus - one of the audio codecs mandatory with WebRTC - is quite flexible, and can optimize for lower-than GSM narrowband speech, or for higher-than-MP3 wideband music: https://en.wikipedia.org/wiki/Opus_(audio_format)#Quality_comparison_and_low-latency_performance
Benefit of using a dedicated encoder, possible thanks to the WHIP protocol - is that it can be told explicitly to encode using music-oriented Opus encoding (not speech-oriented Opus encoding which is the default with most WebRTC implementations).
