Hi Chris,
There will always be latency when using Virtual Cables and Stereo Tool. In general, Stereo Tool should be used further down the pipeline beyond the point when you are monitoring the audio from the software.
I am not sure you point on the last paragraph - why would you be using a virtual cable system if you want to have full control over the audio using mixer faders. In the scenario, you should be taking the output of the mixer and processing this through Stereo Tool - not using virtual cables.
Thanks Jason.
Completely agree, taking the master output of my mixer is exactly what I want to do, i.e. have the output from Playit Live and all other products processed the same.
I just haven't been able to work out the right combination of I/O settings in Stereo Tool to make this happen without either no sound at all or feedback loops, which is why I resorted to the Virtual Cable idea, but as you rightly say, this wasn't the right solution for my setup.
As I use multiple sound cards for input to different faders, would I be right in thinking that the master output from my mixer would be the one I have set to master input in Windows Sound Settings ?
If so, there's no sound through Stereo Tool when I do this.
Thanks
Chris
Chris,
I have a more "old school" arrangement of a physical (analogue) mixer and a similar arrangement of processing post mixer.
If you have a "physical" mixer (you mention "multiple sound cards for input to different faders" so I think you do) then it's the output of that mixer that needs to go to a soundcard input for stereo tool and you want to listen to the "physical" mixer rather than the processed feed.
Depending on what is downstream of "stereo tool" you may want different arrangements but a USB sound acrd dedicated to "stereo tool" might be a good start,
Thanks Mark.
Yes indeed mine is an old school hardware oriented type setup. I like it this way to have 'hands on' fader control to all sources.
My main sound card is an M-Audio M-Track Eight. The TRS Outputs are mapped to :
1/2 = Playiit Live Decks 1/3/5/7 (In To Mixer 2/Fader 4) - Radio Music Playout
3/4 = Playit Live Decks 2/4/6/8 (In To Mixer 2/Fader 5) - Radio Music Playout
5/6 = Windows Media Player (In To Mixer 2/Fader 6) - Personal Listening (Playlists etc)
7/8 = Adobe Audition (In To Mixer 2/Fader 7) - Audio Editing
The only reason I use 3 other Sound Cards (2 x Behringer UFO 202 + Mixer's Internal Sound Card), is to give myself 3 x more sets of Stereo Inputs for mixer, which are mapped as follows :
Mixer 2's Internal Sound Card = PlayitLive Quick Carts (In To Mixer 2/Fader 3) - Radio Carts Playout
Behringer UFO 202 = Zoom (In To Mixer 1/Fader 5) - Zoom AV Conf Output
Behringer UFO 202 = Google Chrome Internet Browser (In To Mixer 1/Fader 6) - Internet Radio etc
Yes I use 2 x physical mixers connected together (both Behringer DX2000USB).
Mixer 1 is the slave, and Mixer 2 the master, the master output of which goes to my main sound card M-Audio M-Track Eight.
As Jason suggests above, I would just like to take that master input from all my audio sources, and have it all processed by Stereo Tool (don't have any need to separate any particular audio 'not' to be processed, except Adobe Audition, in which case I could probably just set Stereo Tool to 'bypass' mode while audio editing).
The problem I have is that I just don't seem to be able to get the right combination of I/O Settings in Stereo Tool to get that master signal in and out of it, without ending up with silence at one extreme, or feedback loops at the other.
Hope that helps to explain in detail what i'm trying to achieve, I seem to be almost there, hopefully !
Many Thanks
Chris
Thanks Chris for the detailed explanation.
I used to have a Behringer DX2000 (the silver version) and it was a very decent mixer for the price and very "radioish" for a DJ mixer.
You are exceeding what your DX2000's can really do but to address your problem I'll tell you what I do,
1) I have a similar 4 channel USB sound card and I use all 4 channels for Playit Live (3 players on 1, 2 & 3 and a combined feed of PFL and carts on the 4th channel).
I also have a "PC" input on a different channel (for Zoom & similar) but that actually uses the built in sound card on the PC. The audio for Zoom etc. gets a "clean feed" or "mix minus" of all the other channels (I assume you just do that with mics and use the FX send on your Behringer).
My suggestion for the processing is to use the Behringer's "Main Inserts" for that and a dedicated Stereo Tool soundcard (one of your UFO 202's). You will need 2 stereo jack to phono leads and use one for left and the other for right (see the DX2000 manual for the insert wiring).
You might also think about trying a Behringer MINICOM 800 at that point in the system.
I think the USB out on the Behringer is probably post main fader (I can't find a block diagram), the Behringer "way" is for the USB to be at a similar point to the phono "2 track" connections. That could be your processed "main" output rather than your multitrack sound card in if needed..
As a more general observation, I replaced my DX2000 with an original version of the D&R Airmate which I managed to get at a reasonable price and that does what you (and I) want very well.
The (non USB) D&R Airmate which you can see second hand from time to time would fit your application very well (but be very careful about the condition of a second hand unit).
Thanks very much Mark.
Sounds like we have very similar setups and working preferences.
Totally agree that the DX1000/2000s are the best pro-sumer desks i've found for radio, but get frustrating after a while due to their lack of robustness and lack of radio specific functions like speaker muting and a proper Cleanfeed for TBU's. (I was sort of achieving this with the FX send as you mention, but for some odd reason it's just stopped working in that the caller can no longer hear me).
I do hope to move on to a pro-make desk such as the D&R you mention, but sadly even the second-hand ones are out of reach for the moment. I like the D&R ones for the fact that all the channels tend to have every kind of connection so it's easy to move things around, but sadly they don't have Pan on the individual channels, which i've often wanted to correct a rogue signal on the fly and had to use the master output faders to do it which is messy.
I think i'm following your general idea if maybe not the intricacies too closely. If i'm understanding correctly you're saying that trying to use the same primary sound card for Mixer I/O and Stereo Tool I/O is introducing too much confusion to the signal chain, so yes in that case having a dedicated sound card for Stereo Tool does sound a great idea, and something I hadn't thought of.
I haven't come across the Behringer MINICOM 800 before, looks very interesting and pre-set based, which might suit my lack of close familiarity with exactly what the number ranges on each compression scale will achieve. Do you suggest this instead of, or as well as, Stereo Tool ? (Hardware Control is always my preference really but Stereo Tool seems to be geared well to the kind of aspects i'm trying to deal with).
At present i'm using the Master Inserts from the mixer into a Behringer MDX2100 Compressor, which i'd hoped would be a total solution, but rather has proved to be excellent for the mic input, but next to useless for the music, despite a lot of adjustment.
Would it be possible then to route just the mic through the MDX2100, and, as you've suggested use the Inserts for a separate sound card dedicated to Stereo Tool I/O ?
Thanks very much for your help, be great to chat if you'd like, my e-mail is [email protected].
Cheers
Chris
Well Chris.
If you have tried your Behringer MDX2100 in the main inserts and that hasn't really worked then I'm not sure the Minicom 800 would be a massive improvement.
What would be (once again if you can find a decent one at a reasonable price) is the Behringer Ultra-Dyne Pro DSP 9024 (essentially a poor man's Optimod). That could be just downstream of the mixer main outputs and would definitely do the job for a music based programme.
It is essentially a hardware equivalent of StereoTool.
However I really can't see why StereoTool connected to the mixer main insert points using a dedicated sound card won't work fine.
As for the Behringer MDX2100, that could be used as an insert in each of the two mic channels on the DX2000 and would probably work well there.
As for the FX output (used for a sort of clean feed), if you have the send buttons pressed do you now hear nothing on the FX send socket (plug something like a powered speaker in there)? I'm surprised it has broken, it sounds more like a cable or config issue.
Hi Mark,
All really helpful and relevant information thanks. Much appreciate how you help me and others on this forum to achieve the things they want in a technically sound and cost-efficient way (as some just think you should do their thing instead !)
Two great options there and I think I will be very tempted to try the hardware processor if I can get one.
Have been reading other posts and see that you have a similar interest in levels and processing, and to give you a brief explanation of what i'm trying to achieve, it is basically to try and level out the music loudness but in a more nuanced way than using normalizing utilities on the files or relying on post-processing necessarily, which is what all the 'radio' people advocate, although i've always found your assertion that it's important that the audio files are in the right ball park in the first place to be absolutely right. Interestingly I started out using mp3gain which I know you recommend, and it was OK for anything wildly out, but a bit of a blunt instrument in broadly setting the whole file to a certain gain level. The other thing which I find odd about it is the scale used. It looks like dBSPL, but I don't follow either why this scale would be used (as opposed to dBFS, PPM, VU, LUFS etc), nor how sound pressure could be measured on an audio file by software without any environmental factors?
The bottom line really is that I currently spend daft amounts of time manually editing music files to sound more like they would from a 70's/80's type music station, and do a lot of old school fader riding and PFL during the show, and I would like to eliminate all these manual interventions, so that I can start the track and just be confident that the regular peaks are tamed (I like about -3.0 dBFS), as well as an overall hard limit (I like -2.5 dBFS) and the dynamic range is reduced only where needed during the track 'radio style' (e.g. for quiet intros and loud crescendos) but not dramatically so; and (without getting into the whole Loudness/LUFS debates), to ensure that both the regular and overall loudness give a bright and energetic but comfortable experience to the listener, but are also appropriate to the genre and style of the track. Hope this all makes sense !
Re the TBU situation, even when it did work, as you know it's not a 'proper' cleanfeed, just one which enables the two signals to co-exist without feedback, but it was better than nothing.
To answer your question, yes definitely had the FX button on, and have done the test you suggested, and also as many other cross-checks as I can think of with other ports/cables etc, and still can't hear the caller. As it was working fine and stopped working between uses without any change in config/cabling (and is the same on both mixers), I suspect its probably the TBU (D&R Hybrid 1 - Original Version), so i'm hoping to try a sonifex one or something similar when I can.
Some good points and questions there Chris.
If you consider what 1980s radio had to work with, the "sound" was achieved by taking the music (from vinyl or CD) and using a processor (perhaps an Optimod for AM and an Inovonics for FM).
Level control was essentially manual and judged using a PPM, so the source level variation was controlled (or should have been) by the time the mixer output hit the processing.
I would thus argue that the "70's/80's type music station" sound was the original source mixes plus the chosen processor.
I'm fairly sure that the current stations playing 1970s & 1980s music will have remastered the (limited) selection of tracks they play to the tastes of the station management (probably reducing the dynamic range of the original material).
In your case, using MP3 gain will get MP3 files in the right ballpark. I don't know what loudness modelling it uses but I've found it gets MP3 files in roughly the right place.
I also like the PlayIt Live "Audio Processing" plugin for the automatic gain control feature (I don't use anything else as I've got the Behringer DSP 9024 after the mixer).
The combination of MP3 gain and the Audio Processing plug in gives the processor a fighting chance (probably as good as the 1980s MCR engineer riding the fader!).
As for your TBU question "still can't hear the caller" suggests a problem with the return from the TBU. I assume you have checked with an incoming call from a mobile, that audio comes out of the "To Mic Input" output of the Hybrid 1 and also the Hybrid 1 "holds the line" (answers the call) when the connect button is pressed.
I wasn't aware of the D&R Hybrid 1 and I must say, having had a peruse of the circuit diagram I rather like it. Essentially it works like a phone but takes the mixer connections as the mouthpiece and earpiece via transformers.
In your setup I assume you have the presenter mic on your Behringer mic channel 1 and the Hybrid 1 on mic channel 2. With the FX send (only) from mic 1 I think this would work OK if the caller only needs to hear you.
Famous last words, but I would say the Hybrid 1 is too simple to break (!), it's got to be leads or plugging.
More great thoughts thanks Mark.
I may not have described my station sound preferences too well as it is a hard thing to define, other than to say that I know it when I hear it (https://xoradio.uk/ is an example), but generally involves a certain amount of compression to make the sound bright and a little punchy but without being overpowering, and while preserving some dynamic range to keep the music authentic, stops amplitude changes between sections of tracks being 'grinding', i.e. 'listener continually reaches for the volume control' syndrome.
Re MP3 Gain, yes I did find it useful many years ago at the outset of my catalogue development, but despite how complex the Replay Gain algorithms are, my understanding is that the upshot action of their analysis is still to just lift or lower the whole track to the gain point they decide on (presumably using Peak Normalisation ?), thus having no effect on any dynamics which don't work well on radio, or on tracks which have been badly mastered. Although bizarrely neither of us knows what scale MP3 Gain is using, for what it's worth, I started out years ago by using what I felt was a good reference track and cross-referencing with several similar ones and my preferred monitoring volume, and settled on "Target Normal Volume" 95.0 dB.
Very interested in your thought about stations remastering tracks at source, as I originally had the same thought, but every station I have asked without exception has always said that they don't touch files and their entire sound is down to post-processing, be it hardware, software, or both.
I've tried the Playit Live AGC and didn't find it useful for my purposes, although I am using the Loudness Analysis Service which does help a little, although it is effectively doing the same as MP3 Gain, but using LUFS scaling instead.
I'm very pleased to have found and purchased a Behringer DSP 9024, as it sounds ideal for what I want to do, although i'm not relishing the thought of the configuration, as presumably I will somehow need to mirror my Stereo Tool settings, which were kindly donated by a station who's sound I like, as i'll be totally honest, I have no idea how to use Stereo Tool 's finite controls in relation to the overall effects I want.
I spent countless hours on the TBU problem a while ago and lost the will. Yes I do have it setup exactly as you describe (my own notes attached), and I totally agree with you that there is little in the TBU itself to go wrong, but so far no change. I have checked physically checked all the connections again today and going to do some call tests over the weekend.
MP3gain (from the explaination) "does some statistical analysis to determine how loud the file actually sounds to the human ear". ( MP3Gain (sourceforge.net) ). The gain is then applied to the whole file.
Thus there is no processing and actually any target loudness will do (assuming that it can be acheved on the file without clipping). Personally I settle for the default 89dB (whatever it means) and then line up the feed to the mixer in the analogue domain.
In combination with the PlayIt Live AGC it achieves my desired result of a correct level presented to the mixer channel.
You will have plenty of fun adjusting the Behringer although there are lots of presets to try first.
As for the TBU, you may have the manual but Canford kindly provide it.
D&R Hybrid 1 Manual v1.07.pdf (canford.co.uk)
Note page 6 & 7. Both input to the TBU (output from the mixer) and output from the TBU (input to the mixer MIC channel) are balanced (using ring & tip on what would otherwise be a stereo jack.
Your notes say
FROM : Telephone Interface “TO MIC INPUT” (¼ Jack)
TO : Mixer Channel 2 -
from which the Caller’s Telephone Audio will be heard in the mix (LINE ¼ Jack)
Two comments.
1) The line input may not have enough gain for what is actually more a mic level signal (although quite hot for a mic level)
2) Try the XLR mic input of channel 2 with a lead as shown in page 6 of the TBU manual.
Note also that the "Effects Send" is a mono jack (tip & sleeve) so will need a "funny" lead to feed the hybrid as that needs audio on tip & ring.
I absolutely understand why D&R have done this the way they have (it allows the hybrid to be unpowered) but the wiring can be a bit confusing.
If you get the cables sorted I'm sure it will work.
Once you have this sorted, I note the Behringer manual says that the line out phonos on the top of the mixer only have the music channels, so with some sort of external mixing it would be possible to get a feed to the hybrid of everything but one mic channel (obviously leave that until the basic setup works however).
Lots of brilliant suggestions, very much appreciated.
Wil report back in due course, may be a while as a I can feel a studio re-org coming on
Hi Mark,
(Still have to try out the Behringer unit, and have purchased another sound card to try Stereo Tool with it as an alternative, so no news on those as yet).
Re the TBU, I've tried some cable tests, and looks like you are on the right track, but I don't seem to be able to get the combination of 1/4 jack connections right. These are the latest results..
I may have unintentionally put across that there was a problem with the LINE Input FROM the TBU, which has actually always been fine, i.e.
FROM : Telephone Interface “TO MIC INPUT” (¼ Jack 2 Ring Plug) TRS I think ?
TO : Mixer Channel 2 - (¼ Jack 2 Ring Plug) TRS I think ?
The actual problem (and apologies if I wasn't clear on this before) is that :
- I can hear the caller (in the mix and level is fine)
- My Mic audio is heard in the mix as normal through Channel 1 XLR at Mic Level
- My Mic audio can barely be heard by the caller through Channel 1 FX ON
I've tried the following 1/4 jack cabling alternatives for sending my audio back to the TBU :
FROM : Telephone Interface “TO CLEANFEED OUTPUT” (¼ Jack 2 Ring Plug) TRS I think ?
TO : Mixer “SEND” on the back of the Mixer (¼ Jack 2 Ring Plug) TRS I think ?
Result = I can hear the caller (perfect level) + They can barely hear me
FROM : Telephone Interface “TO CLEANFEED OUTPUT” (¼ Jack 2 Ring) TRS I think ?
TO : Mixer “SEND” on the back of the Mixer (¼ Jack 1 Ring Plug)
N.B. I used 1 plug of a L/R unbalanced pair for MONO you suggested here
Result = No sound at all, i.e. I can't hear caller + they can't hear me
FROM : Telephone Interface “TO CLEANFEED OUTPUT” (¼ Jack 1 Ring Plug)
TO : Mixer “SEND” on the back of the Mixer (¼ Jack 1 Ring Plug)
Result = I can barely hear the caller + they can barely hear me
My reading of your notes was that I would need a Mono (1 Ring) 1/4 jack plug at the mixer end ("SEND") and a Stereo (2 Ring) jack plug at the TBU end ("TO CLEANFEED OUTPUT”)
.. but as I don't have such a cable, I tried a set with a 2 Ring Plug on one end (TRS for TBU), and one of the 1 Ring Jack plugs from the other end (As Mono for Mixer SEND).
Maybe this won't work and I need to ask for a cable to be made up with 2 Ring Jack one end and 1 Ring Jack the other ? Can't see any like that on the usual eBay cable sites, which is why I thought it would have been OK to use the Left (White) or Red (Right) as a Mono, but appreciate that my understanding of this may not be correct.
Many Thanks
Chris
Hello Chris, you are making progress.
FROM : Telephone Interface “TO CLEANFEED OUTPUT” (¼ Jack 2 Ring Plug) TRS
TO : Mixer “SEND” on the back of the Mixer (¼ Jack 2 Ring Plug) TRS
Yes, that works because the mixer input jack is balanced.
TRS = Tip, Ring, Sleeve jack (sometimes called a stereo jack although in this case it is mono balanced).
From my previous comment " the "Effects Send" is a mono jack (tip & sleeve) so will need a "funny" lead to feed the hybrid as that needs audio on tip & ring".
If getting a "funny" lead for the mixer to TBU made is difficult, try a mono jack lead (tip & sleeve) as that might work with the jack at the D&R TBU input. It's worth a try. You are unbalancing the transformer but in this instance I don't see a problem with this.
The plastic stereo jacks that D&R use are usually OK picking up a sleeve contact with a mono jack.
Chris Pitchford
With much gratitude to Alex James for the vital advice on his post, I have now got Stereo Tool working with Playit Live, using the Virtual Cable (from https://vb-audio.com/Cable/index.htm), and it sounds great, but there are just a couple of issues remaining, and I'd be most grateful for your thoughts on these.
Firstly there's roughly a second's latency between firing the audio in Playit Live and the sound coming out of Stereo Tool, and despite tweaking as many buffer settings as I can find in both products and my sound card, I don't seem to be able to fix it. (I've found it safest not to go below 200ms buffer setting in Playit Live).
There is also a restriction with the Virtual Cable System which doesn't suit the way I work, and I would like to know if there is a system which uses multiple virtual cables, as the one mentioned above comes as a single pipeline, so all the decks from Playit Live have to be routed through this one channel, I/O Stereo Tool; as a full mix, wheras i'm used to having carts and each separate playout deck on separate mixer faders for manual control/fading/mixing etc.
Many Thanks