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Increase audio bitrate in vMixCall
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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Hi again!
after hours playing with our new toy, I seem to find audio quality quite poor in vMixCall. So far, I noticed that best bitrate for audio was around 64kbits (variable from 60 to 70ish and could be around 32kbps from a c920 thru webpage) and never produced a clear sound even when streaming an excellent source from vMix to vMix.
Maybe you guys did have better results... I'd like to hear from you on this point.
Based on my experience, I believe that audio is more important than video in a point to point transmission with remote guests. I did a lot of events with Skype calls and so far when the audio is good (even if the video is crappy) my customers are happy and we can cope with it sometimes only having audio on air with a simple picture of the guest. But if image is great and not audio, then there's no way to use the feed and usually my customers decide to shorten or cancel the guest appearance (and they're not happy).
So the request would be to increase audio bitrates a little, maybe up to 96 or 128kbps when the connection is really good but especially also when the bandwidth is limited and to give priority to audio over video, so that the sound is always as good as possible in any situation.
Thank you Guillaume
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Joined: 1/13/2010(UTC) Posts: 5,210 Location: Gold Coast, Australia Was thanked: 4292 time(s) in 1522 post(s)
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The audio codec used by vMix Call is Opus which coincidentally is based on the same codec used by Skype.
Bitrate is determined by the browser when used and for vMix to vMix is fixed at 64kbps. (But can go lower during network congestion)
I guess some audio samples would be helpful as Opus @ 64kbps is generally "lossless" for speech and quite close to MP3 128kbps for music.
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1 user thanked admin for this useful post.
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Rank: Advanced Member
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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Thanks for this technical feedback Martin...
I guess the only solution is to try to improve audio quality by processing with external equipments/softwares then. It's always very challenging because in general remote guests don't use professional mikes. The difference we had this morning with your DPA and guests laptop mikes was huge.
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Joined: 7/1/2015(UTC) Posts: 1,151 Location: Houston TX Thanks: 319 times Was thanked: 263 time(s) in 233 post(s)
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Opus is in theory a fantastic codec. It works really well in many cases, but implementations vary widely. Successful interop between browsers and hardware end-points are rare.
With respect to the Chrome web browser it may not be the codec that's causing the problem. It could be the echo canceller.
The current EC in Chrome suffers doubletalk badly. That is, when there's audio incoming to the participant the echo canceller can mangle their outgoing audio. It nearly always degrades the outbound audio. It's a question of how much time the participants is in doubletalk. The problem is not objectionable if conversation is ping-ponging back and forth normally.
I work for a company that is engaged in delivering simultaneous interpretation. We use webrtc as a way to connect interpreters to our system. An interpreter delivering SI is continuously in doubletalk. Chrome mangles their audio so badly that we are forced to make the interpreter wear a headset, then we disable the echo canceller when setting up the connection. This is what they call a "constraint" on the getusermedia call.
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Rank: Advanced Member
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I don't think EC is an issue here as vMix does the mix minus. Besides I was disappointed by audio in a vMix to vMix situation where no browser is involved at all and no cheap microphone either : vMix 1 was capturing FullHD digital TV and sending it to vMix 2 thru vMixCall.
I could easily record both sides to compare but I fully agree and understand that vMixCall main interest is not here. I wished it could be used not only for remote guests but also for point to point transmissions where quality really matters.
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DWAM wrote:I don't think EC is an issue here as vMix does the mix minus. What I've described is entirely separate from the mix-minus handling in vMix. If the audio is passing through Chrome I can assure you that EC is an issue. It's on by default. It has an impact. I do think that high bitrate audio would be nice. 32 kbps is ok, but in many cases higher bit rate would not impact bandwidth requirement in a meaningful way (32 kbps vs 64 or 96 kbps is inconsequential)...and could deliver real benefit.
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Rank: Advanced Member
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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Thanks for the explanation.
I'm very familiar with video conferencing quality and also VoiP/IPBX (I use Asterisk for nearly 15 years), so I'm not surprised by telephony audio codecs quality like G723, G729, GSM or uLaw/aLaw. I don't find Opus really better especially when used with cheap mikes. But I understand that AEC is a cause of quality reduction.
I was expecting/hoping that WebRTC integration into vMix would push audio quality to a broadcast level (or closer at least). And I really don't mind using more bandwidth to get a clearer sound 'cos I do hear a difference between a 64kbits or a 128kbps audio stream. I won't use vMixCall for chatting with my friends but to produce broadcast quality videos.
I understand the limitations. I'm a little disappointed that's all...
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Joined: 8/2/2013(UTC) Posts: 1,072 Location: Fairhope, Alabama USA Thanks: 553 times Was thanked: 200 time(s) in 166 post(s)
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+1
I will pretend that I understood everything you guys posted above.
I just want to be able to choose 64K, 128k, 192k, etc. for supreme audio quality.
- Tom
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Joined: 1/13/2010(UTC) Posts: 5,210 Location: Gold Coast, Australia Was thanked: 4292 time(s) in 1522 post(s)
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4 users thanked admin for this useful post.
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Rank: Advanced Member
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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Yeah I already read that after you first mentionned Opus as the audio codec, Martin... I believe AEC is the real problem then, even when used with good mikes thru the browser as MJGraves explained.
I did more testing for vMix to vMix calls with a perfect audio source and I agree audio quality is pretty good in this context.
Thanks Guillaume
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Joined: 7/1/2015(UTC) Posts: 1,151 Location: Houston TX Thanks: 319 times Was thanked: 263 time(s) in 233 post(s)
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The new, completely rewritten AEC for Chrome is known as AEC3 and planned for initial release in Chrome 58. This is purportedly not feature complete, but still better than the existing approach.
Because it's an entirely new AEC developers must stipulate its use in their apps. That is, they must enable AEC3 explicitly.
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OPUS is really good codec, but for VMIX-to-VMIX connection 128K audio can be very usefull - of course if we have enough bandwith. But this is a typical broadcast usecase, and in this situation the bandwith is not too problematic.
My main usecase is the VMIX-to-VMIX connection, for example connect two different venue. In this situation can be usefull the higher bitrate: 1080p@5-10Mbit with 128K audio. In this usecase the PGM out going to the projector with a really big screen...
But all in all, vmix call feature is a REALLY GREAT thing, and I like it. Thanks, Martin!
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Rank: Advanced Member
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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Quote:My main usecase is the VMIX-to-VMIX connection, for example connect two different venue. In this situation can be usefull a higher bitrate: 1080p@5-10Mbit with 128K audio. In this usecase the PGM out going to a projector and to a really big screen... Yep! That's my point exactly. 1080p at 8/10Mbits with 128kbits audio would be perfect for this.
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Joined: 10/31/2016(UTC) Posts: 25 Location: New York
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We used Vmix Call for the first time during a live show today. I would love to be able to select higher bitrate for the audio quality. I know this is going to be a work in progress and its amazing how far its come. With that said I really would love to be able to select a higher audio Bitrate :)
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Joined: 3/7/2017(UTC) Posts: 60
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Agreed. I noticed today while having a guest on, sometimes their audio sounded a little crunchy/digital when they began to speak.
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Rank: Administration
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Joined: 1/13/2010(UTC) Posts: 5,210 Location: Gold Coast, Australia Was thanked: 4292 time(s) in 1522 post(s)
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Could you post an example? it is likely something other than the bitrate. Opus is indistinguishable from raw audio for speech at its set bitrate in vMix Call.
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Joined: 5/31/2018(UTC) Posts: 11 Location: Muenster
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Is there a chance to put this request back on the list?
Besides streaming live I´m using vMix and especially vMixCall for 1:1 trainingsessions with students showing and listenning to recordings in my DAW (digital audio workstation). ATM vMixCall has a better sound quality than skype, teamspeak, and all the others, but it would be great to have the option to increase the bitrate to 192 or even better 256 kbit. This would also be great for my livestreaming-sessions on youtube.
So please, put it back on the list for your next update :-)
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Increase audio bitrate in vMixCall
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