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Improving streaming audio
Rank: Advanced Member
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Joined: 7/29/2020(UTC) Posts: 64  Thanks: 1 times
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Wondering if anybody knows of a plugin, series of plugins, technique or idea of how to improve the audio quality coming from a phone, or laptop computer mic used by the guest streaming into Vmix?
Its not like the audio is bad, or unclear, just thin, with little bottom end.....some of this can be cleaned up in post, but I hope to keep post to a minimum....as I will be under time constraints doing numerous shows a week. There is also some level of phasing due fluctuations in streaming quality.
I know you cannot inject frequencies that are not there - but maybe someone has ideas....
Thanks.
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Rank: Advanced Member
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Joined: 4/23/2017(UTC) Posts: 1,143  Location: Germany Thanks: 3 times
Was thanked: 166 time(s) in 148 post(s)
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The nature of phones and the like is that these things focus on the human voice and not an audiophile approach. Voice is arround 500Hz to 2KHz mainly. Some devices/codecs are optimized for this freuqncy range and thus it's no wonder why it might sound a bit flat.
vmixcall uses webRTC which doesn't make it better. SRT could be a solution for you if audio is an important factor for your production. I mean audio is important for all productions but you might have a special focus on cristal clear audio.
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Rank: Advanced Member
Groups: Registered
Joined: 7/29/2020(UTC) Posts: 64 Thanks: 1 times
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Yes, I know about the human voice frequency response....and the trouble lies with the quality of the input device - mostly. SRT may help a little, but the real deal is a 50 cent mic in a laptop....
Thanks for the post - will take a look at SRT later today when I'm back at the studio.
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Rank: Advanced Member
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Joined: 4/23/2017(UTC) Posts: 1,143 Location: Germany Thanks: 3 times
Was thanked: 166 time(s) in 148 post(s)
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You can use your mobile with larix broadcaster into vmix for SRT testing. Keep in mind that you have to open a port on your network that makes it to the vmix machine (NAT). Otherwise SRT won't be able to establish a connection. Each SRT connection needs a specific port. So for 10 inputs you would need ten open ports.
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Rank: Advanced Member
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Joined: 7/1/2015(UTC) Posts: 1,150  Location: Houston TX Thanks: 319 times
Was thanked: 263 time(s) in 233 post(s)
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Let's put some definition to this.
Traditional toll-grade telephone passes 300 Hz - 3.4 KHz based upon 8 KHz sampling rate and G.711 codec. In this case, the channel defines the limits of delivered audio.
HDVoice passes 50 Hz - 7.5 KHz based upon 16 KHz sampling using G.722 (or equiv) and is dramatically better that above. In most cases this is adequate for voice.
WebRTC uses Opus which can pass full bandwidth audio. Source Elements (and many others) use WebRTC to deliver production quality audio.
In those case, the transducers most typically define the deliverable sound. That's the microphone and related signal processing.
Get them to use a better microphone or a headset and it will sound better.
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