Question: Can a console perform the same function as a summing mixer in a hybrid system? How is a console different from a summing system?

Yes it can . A console is different in that it can provide aux sends, eqs, and sub grouping as well as a two bus for summing.

Question: My plan would be to test the following signal chain: DAW->DA->Small Mixing Board->-AD->DAW while monitoring via the "control room out" on the small mixing board.

Would this signal chain give me a fair assessment of what using analog summing is about? (Obvious caveats given for the quality of the mixing board itself etc)

Yes it would. However the better your mixer, the better the mix's would sound. Using a mixer with with a cheese-o mix bus / power supply might not reveal much of a difference.

Question: Is there something inherently tricky about panning that prevents a pan rotary from being present on many summing mixers? For the DIY-inclined out there, is there something about panning that makes a rotary knob not work in a passive summing mixer?

Pan knobs require an active circuit. Anyway a pan knob should not be necessary as you should be assigning DAW outs in pairs to the summing network. Do your pans in DAW land.

Thanks for the quick and clear response!

If I do the panning in DAW land I am left with half as many channels to sum or I have to create stems/submixes. From reading on the various posts about this topic this seems to be a common/normal/great practice. In my case, having an 8 channel AD/DA converter, it would mean creating 4 stems/submixes and sending those out for analog summing.

My understanding is that I would want to group like material in those submixes (all the guitars in one, all the drums in one for example).

This raises two more questions for me related to using summing mixers:

Question: When choosing "like material" for creating a submix/stem to send out for summing what parameters should be considered? Pitch range? Attack/decay? Overtones? Or is this more of an arbitrary thing that is dependent on a great number of factors/taste/preference?

and then the inevitable...

Question: If groups of instruments are being mixed together in the DAW and then sent to the summing mixer as a stereo pair, isn't that the same as doing the mixing ITB in the first place? Or is this ITB mixing less problematic due to the nature of the "like material" selected via parameters outlined in the previous question (or more likely the answer to the previous question)?

Thanks!

Question: When choosing "like material" for creating a submix/stem to send out for summing what parameters should be considered? Pitch range? Attack/decay? Overtones? Or is this more of an arbitrary thing that is dependent on a great number of factors/taste/preference?

Either way. I would group Drums L/R, bass and maybe kick on one mono channel, Lead Vox on another mono channel, keys and guitars together, and solos. There's 8 channels. But really it's a matter of personal preference / what ever works for you.

I want to add another advantage to doing your panning in the DAW is of course, automated pans.

Question: If groups of instruments are being mixed together in the DAW and then sent to the summing mixer as a stereo pair, isn't that the same as doing the mixing ITB in the first place? Or is this ITB mixing less problematic due to the nature of the "like material" selected via parameters outlined in the previous question (or more likely the answer to the previous question)?

Not so much "like material" ... but the use of more than one stereo bus.

Think of the stereo bus in a typical DAW as a bottleneck. The more bottlenecks you use, the more easily the audio will "flow" into the summing network. 8 is better than 2 and 16 would be better than 8. Most of us are going 16.

I have to add (even though you said not to) With the advant of 64 bit processing, DAWs like Studio One2 and PT 10 XHD have come a long way towards solving the problem of "choking" in the 2-bus. I have a feeling the whole summing and hybrid thing is going to go away in the future. Finally, we have what we have all been wanting. Mixing ITB sans a mixer or other hardware even though mixing with a mouse still sucks!.

I have a feeling the whole summing and hybrid thing is going to go away in the future. Finally, we have what we have all been wanting. Mixing ITB sans a mixer or other hardware even though mixing with a mouse still sucks!.

BUT! its not just about headroom, its about sonic variation and analog colour you get from very cool gear plus the unique space and imaging you get including a really pleasing distortion famous with analog. Its all relevant :) the better ITB gets, the better analog will get. The two worlds together, done right, are gold. That's how I hear it.

You just won't need a summing solution. Even ITB you can use analog processing (compressors, reverbs, gates, etc) by using your sends and returns . I do this all the time in lieu of purchasing plug ins that only get me 98% of the way there. I think that's where we are going to see things wind up. Of course everyone has their own preferences and no matter what if I could afford a nice API console, I would never mix in the box if I didn't have to.

From the beginning, the main promise of DAW's was the ability to record and mix without a lot of expensive hardware and the associated maintenance costs other than converters and computers. I myself was very disappointed when I heard what those mixes sounded like and by then I had unfortunately sold off my analog console. BUMMER!

Kurt: Just so I can make sure I understand, is the problem (assuming there is one) of mixing ITB one of mashing more channels into a stereo bus? For example is the following statement true? In mixing ITB, mashing 32 channels into one stereo bus tends to result in something that sounds different than mashing 32 channels into 8 stereo buses.

And I appreciate your insight into how changing technology if effecting this issue. Especially because you give a clear source for your suspicion--64 bit processing. It at least gives me something to go on when researching further.

As for mixing with a mouse, well that's what I've been doing for the most part but there are USB controllers which mimic the haptics of a console and so on. I think the psychological impact of having nobs that are readily tweakable has an impact on decision-making.

My income is generated as an online business consultant with a heavy focus on analytics--I'm the guy who tracks what people click on and know what/how/when to tweak it for different results. My research in that realm absolutely supports the idea that people will make different decisions based on what options are available to them onscreen (blue button, red button, on the left or right or up or down etc). Often these decisions people make are tied to large sums of money and/or important life decisions. Even in these instances what they see and how many clicks it takes to get someplace influences their behavior.

I have to believe that the same is true of professionals as well. Screen real estate is by nature limited. Shunting some of the interface out of the screen and into a hardware controller (whether analog or USB connected to the DAW) allows a greater set of options to be available in the mind as well as in the hand.

To be honest, this is one of the reasons I'm looking to cycle my recording setup from a completely DAW configuration to one which mixes in more analog stuff. Ultimately I would like to have a recording setup that takes advantage of DAW characteristics but requires an absolute minimum use of my eyes on a screen. This is primarily due to cognition stuff. The brain will prioritize visual stimuli over auditory stimili so in order to free up some cycles in my limited CPU (cognitive processing unit) I want to decrease visual requirements while mixing.

In this instance, even if ITB results in exactly the same auditory experience as a hybrid solution I would choose to be hybrid.

So that's the crazy underbelly of what I'm exploring here. On the surface though, I just want to develop a system and process that sound good. And since I don't know what I don't know I'm here asking these questions. And very grateful for the answers given.

I have no problem discussing advantages/disadvantages of analog summing. But so many of these threads have devolved into statements of taste/fashion which then quickly enflamed into episodes of individuals establishing status/dominance in relation to their taste/fashion. Since I don't make my living from audio engineering it's unfair for me to weigh in on these sorts of things; it's people's livelihoods we're discussing at some point.

I avoid going back into the same box as you suggest Kurt. I would never do that after having a straight line like the Dangerous and SPL hybrid gear. I use the DAW for mixing and some plug-ins but sum OTB and never go back to the same DAW once its analog. Using a MixDream for its headroom and inserts is a glorious sound. I do the hybrid thing with a MixDream and Dangerous Master together and master to a second computer so there is no SRC. The magic happen OTB and I do my best to save it on a second system albeit DSD or DAW.

audiokid: Sorry for this basic question but what is SRC? And in your configuration that second recording medium could be anything right? Tape deck, DAT, wax cylinder, DAW, whatever?

Sounds like two approaches (please correct me if I've got this wrong):

Kurt is advocating gaining some analog fun by sending items from the channel strips and back in the mixing. Then at the end summing the whole thing down ITB. Do you bother with stems etc in this process?

audiokid is advocating sending stems to the MixDream summing mixer (which has inserts for each channel and the master) for summing, then on to a completely different device to record that final output. Are there benefits to this approach beyond the lack of SRC?

Thank you both for helping me with this. I'll be updating the top post once I have clearer understanding of what's what.

I agree with what Chis is saying as far as going back to the box with DAWs of the past, due to these same choking / bottle neck problems but I truly believe that this will be fully addressed in the near future as more and more DAW designers jump on the 64 bit bandwagon.

On the other hand there will always be those who like the sound of a particular piece of gear as Chris seems to feel about his Mix Dream. I get it and that's fine with me. Like I said, If I had an API or for that matter even my old JH 636, I would be jumping out of the box at the first opportunity. But the whole idea of computer recording is supposed to be to get away from expensive hardware, not to jump to using different expensive hardware.

I still suspect we will see a lot of summing box's on E Bay going for cheap in the next year or so. It's the nature of digital audio. Things change every 16 months.

Bottom line for me is at the present moment it appears that investing in a DAW with higher bit rate processing may provide a more efective way of improving your audio. And I would add, just try to sell analog gear. I have quite a few very nice pieces that I almost can't even give away.

shoot me again! I'm still here.

For the OP,

from all my reading on forums for the last 13 years and doing extensive research on hybrid, a lot of people who have tried it claim it is no better but when you dig for what they did or used, its really obvious they are not going about it correctly and are also pretending to know more about it than they do. Most people do not want it to be better than ITB because it is expensive and out of reach so they spread incorrect information.

I've read many threads where people claim to have even done comparisons and wonder how they could truly have done this ( in this economy to add) without investing in at least $2000 in cable right off the bat, $1000 for a proper interface, $4,000 in adequate conversion, $3,000 for a proper summing amp before you even get started lol. Then if you really want to get serious add $60,000 in nice hardware.
Most people that give opinions have never even used the gear I describe so how could they know I ask.
And most of the guys that used a console in the past and try hybrid are not the best consoles. They are noisy and have bad routing in comparison to something like a MixDream. A console is not the same thing. But, if we are talking about a great console like and API, Neve to name two, well thats another story. But its still not like what I have. But I'm certain either will produce true excellence.

I'm not saying you need tons of outboard gear to get some vibe happening either but it sure helps. Once you pass the basic set-up, and hear the results its addicting.

Even something like the Bricasti at the end of my analog chain is to die for. That sound isn't even possible ITB. There aren't CP big enough to handle all this processing. So, we will always be passing the meat to analog. And when there is one day where CP are so powerful ( I'm still waiting lol) , the gear that has the glorious colour will always have a place to help that super clean digital track or stem out. Hybrid is the best of both worlds. Will it make you more money? doubtful in this stupid economy but it should if you are a good business man and an audio engineer. .
Hybrid done right is sure a lot more fun and way easier getting a great sound. Its a rich mans game and its definitely for the upper crust in this industry.

Kurt: In your system where whatever analog stuff you do comes in at the track level via inserts do you also do stem mixes in order to avoid the bottleneck thing? Only on 32 bit machines but not on 64 bit machines? Or do you always do stem mixes anyway regardless because that's your normal process and the technology doesn't matter?

Hi Kurt,

The MixDream dosen't have a sound. Its transparent in a good way but it has tons of headroom making it able to handle a punch. It does have a optional Lundhal tranny but I rarely use it. What the MD does is allow gear to be inserted in a really logical way. So quiet and so smart. Its the gear that you add that adds the colour. The MixDream is the hub. Hybrid without added gear is okay but not really worth it. However, mastering to a second CP via hybrid is definitely worth it. And, a really good monitor control system is essential.

audiokid:

I've read many of the forum posts and have admired your consistent message and approach: use good gear, use the right method. What I'm trying to do in this post is clarify things a bit. That way the next person like me, who is looking for the on ramp to hybrid, can get things straight without reading a lot of armchair opinions and false starts.

There are a couple questions that you could help out with greatly, I think:

Question: What is the difference between a summing mixer and a console? Is it sound color? Coloration flexibility? Workflow? Something else entirely?

Question: What is the bare minimum technology pieces required to do hybrid well? I'm not looking for brand names (though that's nice) but instead I'm looking for function.

Question: In those functional items, what are the parameters of quality to be looking for?

In many of the hybrid and summing mixer threads I've read there is a strong focus on improving DA/AD conversion since a hybrid method will, by it's nature, include this more than a straight ITB method. By reading the tech specs there is little to differentiate a $200 AD/DA converter from a $2000 AD/DA converter--they all take x channels and convert them into 44.1/48/98 16bit/24bit digital files. Obviously there must be some difference beyond marketing and I hope the difference isn't simply 24 karat gold connections for data.

Regarding whether the investment in gear will result in a positive return on investment is probably far more closely linked to one's business skills than one's engineering skills. But as I've noted earlier, the food on my table isn't provided for through my efforts as an audio engineer. It's unfair of me to engage deeply in that conversation.

If possible, I'd like this thread to stay focused on issues and aspects on the borders of the digital and analog world--especially items useful to someone just beginning to approach the topic.

audiokid: Looks like some of the items are approached in your post above this one. I'd like to add another question though:

Question: What are the essentials of good monitoring system in a hybrid setup that are different from ITB? (i.e. good speakers and headphones and positioning and room and all of that are common to both)

Also what was the SRC that is removed when you're using a second machine at the end of your hybrid setup. That was one that was missed earlier.

Thanks for your patience with all of this.

Gahlord, post: 393346 wrote: Kurt: In your system where whatever analog stuff you do comes in at the track level via inserts do you also do stem mixes in order to avoid the bottleneck thing? Only on 32 bit machines but not on 64 bit machines? Or do you always do stem mixes anyway regardless because that's your normal process and the technology doesn't matter?

I have 16 ins and outs so what I do is not what I would call "mixing stems". It really depends on what I am working on, how many tracks there are. I have done mixes with 16 tracks mono or just a couple of sub - "stems" plus the 2-bus and everywhere in between. No rules carved in stone as far as I am concerned. It all depends on what sounds good (enough).

Thanks Kurt. I suppose in some ways what you are describing--using inserts at the channel level--solves one of my initial issues. It would allow me to apply analog stuff to each of my channels without having to create stereo stems. In my system where I have only 8in/8out I could apply analog stuff to 8 different individual channels instead and handle the panning in the DAW. For the process involving a summing mixer I could apply analog stuff to only 4 stereo subs at a time.

What I'm hearing is that process/machines used are arbitrary/taste-driven at the 64-bit level. At the 32-bit level there are constraints with bottlenecks on summing a large number of tracks down to a single stereo bus. This would lead me to believe a 32-bit system would favor the summing mixer approach while a 64-bit system would favor the insert/send approach.

Thanks again for taking the time to answer these questions. This is starting to clear up a bit for me. Which of course only means maybe I'll think of more interesting questions.

Thanks for the compliment, Fun Topic and also fun having you in on this one Kurt!

Question: What is the difference between a summing mixer and a console? Is it sound color? Coloration flexibility? Workflow? Something else entirely?

It took me a year to understand it all and I still didn't get it all until I bought the gear and starting testing things. The summing amps IMHO should be transparent and have mass headroom. A console is very different as noted above in a previous post.

Question: What is the bare minimum technology pieces required to do hybrid well? I'm not looking for brand names (though that's nice) but instead I'm looking for function.

I chose a more complete system so I can't be sure if less will produce similar results but even summing otb and adding a nice compression for flavour more than actual compression can be enough to glue things more. I wrote a long post on why I felt it was important to sum OTB but in a nut shell, I am 100% convinced that digital music sounds so different to real music that is actually doesn't sound right unless you help bridge them closer together. People call this glue but they don't tell you what it is. I don't think a lot of people understand why things sound better but thats how I hear it.

I've been into digital music since 1980 and thought it was the greatest thing until I sold my analog keyboards and bought a $12,000 Sampler ( well actually a few of them duh).
The first 6 months of having my first one , I was so overwhelmed with what it did that I never really listened to the sound of it. I mean it had all these libraries available but they all sound sterile and eventually I started noticing my mixes where never the way I sounded when I was all analog. I spend all my time trying to get vocals and guitars to blend with the samplers.

18 years later I bought a Pro Tool system. Yup, another $50 thousand invested in digital gear. Sept 2000 I started a thread on the DUC and predicted by 2005 most of the BIG Studio would be bankrupt and Pro Tools would take over the industry. The thread got so ugly it was locked and disappeared. Many big studios chimed in on that attacking me.

11 years later I am now convinced that the reason our music have little real sound in it are partly because its so difficult to get stuff to blend with digital music. Yes, we do it but to my ears real acoustic music, before it is digitized sounds out of place. Yes, I'm sure there are thousands of engineers that disagree but I hear it man. And todays music is in serious need for more real sounds.

Digital music if is awesome. I love it! I love how full and clean it is. I LOVE drum machines and processed drums. I love synth bass and keyboards. I loved pop music but its starting to get a bit to club and boring because its produced all ITB. There is no doubt to me that if you are using both worlds in a mix, you get there with a big sound via hybrid.

Enough said on that.

Question: In those functional items, what are the parameters of quality to be looking for?

You mean what quality of summing gear are you looking for? Doesn't matter. Its all about adding grit to clean and clean to grit so they blend and open up in a more fluid sound-scape.

Question: What are the essentials of good monitoring system in a hybrid setup that are different from ITB? (i.e. good speakers and headphones and positioning and room and all of that are common to both)

I own an SPL 2381 MTC and a Dangerous Monitor ST and all I can say is wow! Having the ability to compare and hearing your mix unscathed in the analog domain, via a monitor control surface engineers for a hybrid system, just before you master it to a second DAW is paramount. No more guessing. It is exactly how it sounds to the lost drop (Well at least super close).

Anyone who rely s on just a headphone out , lol. Good luck trying to be a pro is all I have to say and no wonder you aren't hearing everything.
The Dangerous ST is awesome. Watch a few youtube videos on that.

Also what was the SRC that is removed when you're using a second machine at the end of your hybrid setup. That was one that was missed earlier.

It just sounds right. And because I am already OTB and glowing with vibe... for some reason ( I'm not a technical guru and don't even want be) mastering to a second box just sound great. This may change in time but I will probably do it this way until I die now.

Thanks audiokid for your responses here (and everywhere on this board). I'll try to find the post you mention on the importance of OTB sum, I must have read it and misplaced it in my mind--a lot of posts about summing read in the past week.

Re: second machine summing I wonder if the quality of your 2 channel AD converter at the end plays more of a role or if increased availability of CPU cycles has an impact on your perception of improved sound in that method. I'm definitely curious to try it out. Do you use the same AD/DA converters at all three conversion points in your system?

Re: second machine summing I wonder if the quality of your 2 channel AD converter at the end plays more of a role or if increased availability of CPU cycles has an impact on your perception of improved sound in that method. I'm definitely curious to try it out. Do you use the same AD/DA converters at all three conversion points in your system?

No, I use different tracking converters on the second recorder but listen to everything via the Dangerous ST.

Just found these questions from you so I'll address those.

Gahlord, post: 393343 wrote: audiokid: Sorry for this basic question but what is SRC? And in your configuration that second recording medium could be anything right? Tape deck, DAT, wax cylinder, DAW, whatever?

Yes, and SRC mean Sample Rate Conversion

Sounds like two approaches (please correct me if I've got this wrong):

audiokid is advocating sending stems to the MixDream summing mixer (which has inserts for each channel and the master) for summing, then on to a completely different device to record that final output. Are there benefits to this approach beyond the lack of SRC?

No idea. It just sounds great! I am able to crank my analog mix and control it so well , thus, sending it off to the second recorder. Its easy and it works. Most people I follow that try this, never go back.

Thank you both for helping me with this. I'll be updating the top post once I have clearer understanding of what's what.

Speaking of SPL / Mixdream, the SPL / Neos 1010 looks very cool. :cool: [[url=http://[/URL]="http://spl.info/ind…"]In Short: Sound Performance Lab[/]="http://spl.info/ind…"]In Short: Sound Performance Lab[/]

Faders and inserts and compression ! Oh my!

I don't think we're in Kansas any more Toto!

This is an interesting thread! I'll chime in here to clarify one aspect of what is being talked about, and that's SRC (sample rate conversion).

SRC is needed only when the recording sample rate is different from the output target sample rate. If you are recording at 44.1KHz with the aim of producing a CDs at the end of it, then SRC does not feature in the process, and I do many recordings of demos and other material at 44.1KHz. However, my preferred method for high-quality recordings is to capture all tracks at 96KHz, usually on a pair of Alesis HD24XRs. I will then replay these 96KHz tracks into an analog mixer and capture the 2-track mix both at 96KHz back to a couple of spare HD24XR tracks but also split to a conventional audio interface into a computer at 44.1KHz.

Using this method, the mix is done in analog from the 96KHz tracks, and I need no digital SRC to go down from 96KHz to 44.1KHz. I have not found a way of generating better CD-quality stereo results, whatever the quality of the individual pieces of gear used. But ahead of actual gear, there are two things to which I attribute the sound quality of the method:

(1) the avoidance of a digital SRC. None the SRCs I have tried are completely transparent. I think Chris reported that he immediately noticed an improvement in his mixes when he adopted this source-mix-capture method.

(2) Not mixing at 44.1KHz. The addition of tracks that all have brick-wall anti-aliaising filters at around 20KHz produces a tiring top octave. This effect is not dissimilar from the bedroom-recordists results of tracking everything using a single bright condenser microphone. When the top octave is 20-40 KHz, the 10-20KHz region is much cleaner, and having just the stereo 20KHz filter for the mix capture is a tolerable necessity.

Nicely put, Bos and thank you for making me aware of this when I first got interested in hybrid!

I just got a [[url=http://[/URL]="http://korg.com/MR2…"]Korg MR2000SBK[/]="http://korg.com/MR2…"]Korg MR2000SBK[/] for this. Boswell, can you explain more about DSD and why we aren't seeing larger scale, 8 and 16 channel DSD recorders like this?

Bos, have you tried and compared numerous analog consoles and found one over another more favourable? Has headroom been a factor? I find threads where people try summing otb, who use an older analog console and they eventually succumb to ITB. As noted in a few threads back, I personally trust they are not going about it well but also wonder if their console lacks headroom too.

The MixDream and Dangerous Master have high headroom. I've never actually tried to push them to the limit.

Kurt, the NEOS would be stunning! There is a long thread on it where people joke about it being overkill. Understandably why, it goes right over most peoples heads. I would love to have that. It would be the ultimate summing system. oohhh la la.

audiokid, post: 393372 wrote: I just got a [[url=http://[/URL]="http://korg.com/MR2…"]Korg MR2000SBK[/]="http://korg.com/MR2…"]Korg MR2000SBK[/] for this. Boswell, can you explain more about DSD and why we aren't seeing larger scale, 8 and 16 channel DSD recorders like this?

I could be cynical and say that because editing DSD streams is not supported by PTHD, there's no market for DSD recording devices. However, this just masks the fact that DSD edits are difficult and complex. I had a hope a couple of years back when DSD started to make its mark that a compromise editing scheme would be worked out where you made PCM versions of your DSD input tracks and worked out all the edits, effects and mix that you wanted using the PCM tracks. You would then set off an overnight batch job that built a stereo DSD mix from the original DSD inputs using the parameter set you had decided upon from your work on the PCM tracks. Sadly, this does not seem to have happened.

audiokid, post: 393372 wrote: Bos, have you tried and compared numerous analog consoles and found one over another more favourable? Has headroom been a factor? I find threads where people try summing otb, who use an older analog console and they eventually succumb to ITB. As noted in a few threads back, I personally trust they are not going about it well but also wonder if their console lacks headroom too. The MixDream and Dangerous Master have high headroom.

Headroom you can sometimes work around, usually at the expense of noise floor. However, purpose-built analog high-level mixers like the two you mention are in a different league.

For me, the big difference between using nice-sounding conventional analog mixers over cooking varieties for mixdowns is in things like the musicality of the EQ circuits and whether the line inputs are constrained to go through the pre-amps. This sort of thing does not need to be very expensive, although it's not in the bottom-level price bracket. For example, I get great results from an A+H Zed-R16, ironically used purely in its analog mode, and from some of the older Midas boards like the Venice and Verona. All these are excellent-sounding products for analog mixdowns as long as you take care to avoid the pre-amps by using the insert returns for your input signals where the line inputs are not separately routed to the main mix.

Boswell: Thank you for adding that!

Just to I can make sure I understand what's happening, your signal chain looks something like:

sounds->preamps->AD at 96khz->DAW1->DA at 96khz-->Analog stuff-->Split to: AD at 44.1 on different recording device/DAW2 for CD --and-- AD 96khz on DAW1? or DAW3? for archival

Question: How many recording devices are you looking at here? For example-- DAW1 is the ITB mix. DAW2 could be anything (tape/DAT/wax cylinder/Logic/PT/etc) so long as the AD being sent to it makes sense. DAW3 could be either non-existent (sending the stereo sum back to the original DAW1) or else it could be anything.

It seems a waste to be running three full computers for this given that two of them are simply catching a stereo track. But I suppose computers for that purpose could be relatively inexpensive.

The purpose of the final 96khz stereo track is so that you can then re-output it to DA 96khz and re-capture it at whatever sample rate you need for different distribution formats, right? Sort of like "re-amping... for masters"?

Question: Is the headroom you and audiokid discuss the same kind of headroom I concern myself with on my bass--having 1600 watt power stage but never needing to turn up even half way--to get a clear sound? In other words, is one element of quality in the summing mixer configuration dealing with having a lot of available gain?

Thanks again for your insights here. This thread is really clearing things up for me.

Also, glad you are liking this thread too Boswell. My hope is that it becomes the go-to thread for people who express an interest in moving towards an OTB process. I am continually updating the top post so people don't have to trawl through the entire dialogue to find the highlights.

The main point of the recording/mixing chain that I described I use for high-quality mixes is that no DAW is involved. The recording is done to Alesis HD24XRs, which are 12/24 channel 24-bit digital hard disk recorders that can be stacked in blocks of 12 or 24 channels (12 at 96KHz, 24 at 4.1/48KHz).

For mixdown, I replay the HD24XRs via their D-A converters into the analog desk and then capture the two-track analog mix out. I usually capture the mix at both the original higher rate (96KHz/24-bit) to spare tracks on one of the HD24XRs, and simultaneously at the target rate, which is most often CD standard 44.1KHz/16-bit but could be video standard 48KHz/24-bit.

The CD-rate capture can be done on something like an Alesis Masterlink, but I usually use an RME FireFace800 and a simple data capture program.

I can transfer the higher-rate mix to DVD-audio format, and then use it for blowing the socks off the people I know who have Hi-Fi systems incorporating a DVD player with DVD-audio replay capability. Through hearing DVD-audio like this, they have come to realise that commercial CD-quality leaves a lot to be desired. Since I don't do commercial mastering, the higher-rate 24-bit mix is also useful as a target for the mastering houses who are tasked with squashing the CD-mix to something they regard as releasable.

Thanks Boswell. So in the chain above DAW1 and DAW3 would, in your case, be the same device Alesis HD24XR (or a few of them). But the same device may be both playing audio out to the console and receiving the stereo pair back from the console. I'm assuming you run the CD-rate from a monitor out or other out from the console that is the same signal the HD24XR is receiving?

Super helpful. And great setup eliminating the screen entirely but keeping digital as a medium. I have to ask though, do you use the Alesis HD24XR instead of tape simply to avoid the maintenance and supplies issues of tape? Or because of other advantages in keeping the tracks in a digital medium?

Also regarding previous post about a DSD edit path: That is exactly how many higher end film edit stations work. All of the editing on screen is done using lower resolution files which creates something called an EDL (edit decision list). That EDL is simply an XML file that gets run against the full resolution file when the time comes to make a print. It's a workflow that should very easily transfer over to audio files--could probably hack film editing software to do it with a few lines of code. I don't know anything about DSD but will have to investigate now. Lol.

Yes, that's the chain. The mix simply produces a single 2-channel (stereo) analog out; I split this to a pair of HD24XR inputs and to the FF800 or whatever device is capturing the 44.1KHz stream.

I know only a little about audio processing for film, but was aware of the use of EDLs. The big difference between film and pure audio is the emphasis on editing and sync rather than on levels, effects and dynamics. These latter things play a part in film sound, of course, but you can get away with a great deal more when there is a screen to watch, provided the timing is accurate.

When I first started this game in the 1960s, tape was what it was all about, but I rarely use tape now. I have shelves of 10.5" boxes that I know I should go through and transfer to digital format, but somehow the time to do it is difficult to find...

Hi guys! I'm new to OTB and I just posted a new thread before reading this one. Can you guys share your calibration routine? I'm using Logic Pro and a MOTU 828mk3 Hybrid to output 4 stems to a passive summing mixer. Then from the mixer back to pre's through a compressor and then back into the MOTU. I'm trying to decide on a cal level that works best for me.

Cal to what? Studio equipment, since it was invented, from the beginning of its existence, calibration reference level of 1.23 V across 600 ohms a.k.a. +4 Dbm, is still the standard in real studio equipment. Everything else was just everything else. That's a normal operating level, where your peaks will be +10 to 15 DB and beyond. And only the professional stuff can do that. So when asking about the best reference level, that is the best reference level. So your preamps, along with your compressors and your passive summing mixer. That should have an active output, are all most likely intended to be used with each other at +4 DB average working level outputs and inputs.

Now if you are asking where to set the knobs on your various pieces of equipment, there are only general guidelines. Some devices have a marker about two thirds of the way up on some volume controls. General guidelines with rotary controls put that marker to the two o'clock position. This is where most input sources, will equally equal output, a.k.a. unity gain. While at the same time, being able to provide variable gain. From the gain trim, which are two different things. Gain trim affects the internal operating parameters of the amplifier, where the variable gain actually refers to a passive control at the output of the amplifier. Both are interactive. Utilizing them the right way is always good. Utilizing them the wrong way, is frequently good. But that means that you have to understand how to do it right before how you can do it wrong the right way. Wrong the right way is what we refer to as saturation and overdrive. Done right, it can present a more upfront and/or aggressive quality to the sound tonality. Doing it wrong means. It's choking in noise and unlistenable distortion. Sometimes even that can be right. For instance, you certainly wouldn't want the same kind of guitar overdrive plug-in for use on an operatic soprano. Unless you really thought she was that awful and wanted to make sure she couldn't get a head, but rather need to give it?

The same can be said of your compressors. When you push certain dynamic range devices beyond their rated norm, other cool things can come from it. Otherwise you crushed the crap out of everything leaving your self nowhere to go. And each type of compressor imparts its own unique coloration. If it's an optical unit, peak sensing, average or RMS sensing type and whether it has variable attack and release times or no adjustments for that. They'll create their own magic or, crash and burns. What peak limiter is good for the goose ain't good for the RMS Gander. Or maybe it is? Only you can find out.

If I told you how to get good sound... I'd have to kill you.
Mx. Remy Ann David

Good point. Sometimes I overcomplicate the theory.

Overthinking the theory is actually a good thing to do. You need to think that you are a sound wave and don't know how to go home. So you have to put yourself into those little wavelets and pretend you are swimming through your electronics to find the out door. It's sort of like, LET'S MAKE A DEAL TV show, where you get your choice of what's behind door number one, door number two, or door number three. One's a very nice prize and the other two are just stupid funny. I find a lot of our recording equipment today to be just stupid funny. And they are trying to sell you stupid funny telling you, honey, they want your money. And then trying to rationalize their own stupidity in what they are trying to sell you. That's another reason why I like attending the AES shows. They give you all of this technical blather that really has nothing to do with good sound. Then you can just laugh and walk away from them. Knowing that you know better. Let's face it, no one's willing to give a sucker an even break.

I like Tootsie Pops best. Because in the middle there is something you can dig your teeth into. Not just a stick.
Mx. Remy Ann David

Boswell, post: 393366 wrote:

SRC is needed only when the recording sample rate is different from the output target sample rate. If you are recording at 44.1KHz with the aim of producing a CDs at the end of it, then SRC does not feature in the process, and I do many recordings of demos and other material at 44.1KHz. However, my preferred method for high-quality recordings is to capture all tracks at 96KHz, usually on a pair of Alesis HD24XRs. I will then replay these 96KHz tracks into an analog mixer and capture the 2-track mix both at 96KHz back to a couple of spare HD24XR tracks but also split to a conventional audio interface into a computer at 44.1KHz.

Using this method, the mix is done in analog from the 96KHz tracks, and I need no digital SRC to go down from 96KHz to 44.1KHz. I have not found a way of generating better CD-quality stereo results, whatever the quality of the individual pieces of gear used. But ahead of actual gear, there are two things to which I attribute the sound quality of the method:

(1) the avoidance of a digital SRC. None the SRCs I have tried are completely transparent. I think Chris reported that he immediately noticed an improvement in his mixes when he adopted this source-mix-capture method.

(2) Not mixing at 44.1KHz. The addition of tracks that all have brick-wall anti-aliaising filters at around 20KHz produces a tiring top octave. This effect is not dissimilar from the bedroom-recordists results of tracking everything using a single bright condenser microphone. When the top octave is 20-40 KHz, the 10-20KHz region is much cleaner, and having just the stereo 20KHz filter for the mix capture is a tolerable necessity.

This is a great thread I am learning a lot from seeing as I'm about to go Hybrid for the first time as indicated in my first post here. Many points have been cleared up for me already. Boswell, I'm very interested in this capturing method but would you mind explaining to me how it would be different from capturing the St master out from a console into a daw as a 96KHz file and then playing that 96KHz recorded file back through the desk and re-capturing it as a 44.1KHz file.

Thanks in advance.

Boswell, post: 393375 wrote:

For me, the big difference between using nice-sounding conventional analog mixers over cooking varieties for mixdowns is in things like the musicality of the EQ circuits and whether the line inputs are constrained to go through the pre-amps. This sort of thing does not need to be very expensive, although it's not in the bottom-level price bracket. For example, I get great results from an A+H Zed-R16, ironically used purely in its analog mode, and from some of the older Midas boards like the Venice and Verona. All these are excellent-sounding products for analog mixdowns as long as you take care to avoid the pre-amps by using the insert returns for your input signals where the line inputs are not separately routed to the main mix.

Hi another question from me if you don't mind. The ZR16 is one of my considerations as a summing mixer / new audio interface. It's reading very well indeed and it would seem you confirm that is is a very good sounding unit used in a specific way by avoiding the pre amps & using insert returns for inputs. Would you mind elaborating on that a little more please baring in mind I have no console experience? For info, all my sounds apart fro my vocals and guitars are generated ITB at the moment and I already own a very good pre amp for capturing these. Thanks.

Kevin Big Jam, post: 394343 wrote: This is a great thread I am learning a lot from seeing as I'm about to go Hybrid for the first time as indicated in my first post here. Many points have been cleared up for me already. Boswell, I'm very interested in this capturing method but would you mind explaining to me how it would be different from capturing the St master out from a console into a daw as a 96KHz file and then playing that 96KHz recorded file back through the desk and re-capturing it as a 44.1KHz file.

The simultaneous capture at two different rates results in better audio quality at 44.1KHz, as it saves one set of A-D-A conversions. If it were just a stereo track, you would not in any case need to put it back through a desk. Simple replay at one rate directly captured at another is all that would be needed.

Kevin Big Jam, post: 394343 wrote: Hi another question from me if you don't mind. The ZR16 is one of my considerations as a summing mixer / new audio interface. It's reading very well indeed and it would seem you confirm that is is a very good sounding unit used in a specific way by avoiding the pre amps & using insert returns for inputs. Would you mind elaborating on that a little more please baring in mind I have no console experience? For info, all my sounds apart fro my vocals and guitars are generated ITB at the moment and I already own a very good pre amp for capturing these. Thanks.

The pre-amps in the Zed-R16 are excellent, especially given that you get 16 of them in a mid-price console. However, as with many live consoles of this type, the line inputs are attenuated and put through the microphone pre-amps. The insert returns by-pass the pre-amps, but are unbalanced, so you have to deal with that in the wiring or use something like high-quality bal-un transformers. I try to stick to a rule of not feeding line-level signals through pre-amps of any sort unless either it's unavoidable in routing terms or I'm going for a special sonic character such as transformer saturation.

It's a bit ironic using the Zed-R16 as a purely analog console by ignoring its A-D and D-A converters, but it does perform well in this role, and also the digital side of it is not designed to run at the higher conversion rates.

Boswell, post: 394346 wrote: The simultaneous capture at two different rates results in better audio quality at 44.1KHz, as it saves one set of A-D-A conversions. If it were just a stereo track, you would not in any case need to put it back through a desk. Simple replay at one rate directly captured at another is all that would be needed.

Thanks for the swift reply Boswell. Yes of course. My morning head had not even considered that! Funny how we try to over complicate things at times.

Boswell, post: 394346 wrote: The pre-amps in the Zed-R16 are excellent, especially given that you get 16 of them in a mid-price console. However, as with many live consoles of this type, the line inputs are attenuated and put through the microphone pre-amps. The insert returns by-pass the pre-amps, but are unbalanced, so you have to deal with that in the wiring or use something like high-quality bal-un transformers. I try to stick to a rule of not feeding line-level signals through pre-amps of any sort unless either it's unavoidable in routing terms or I'm going for a special sonic character such as transformer saturation.

It's a bit ironic using the Zed-R16 as a purely analog console by ignoring its A-D and D-A converters, but it does perform well in this role, and also the digital side of it is not designed to run at the higher conversion rates.

Understood, are the firewire inputs into the ZR16 from the Daw processed through the mic pre-amps as you explained? As an audio interface, I thought the zR16 ran upto 96KhZ? You have already been helpful, however if you get a chance please offer your thoughts on my upgrade options

https://recording.o…

Thank you yet again.

Yes, the D-A converters come in at the same level as the insert points, i.e. after the pre-amps.

I was a bit terse in describing the digital side of the Zed-R16 as not running at higher rates. It's slightly complicated and restricted, and you should study pages 24-25 of the manual. Basically, the FireWire 2-bus interface and all the ADAT I/O are limited to 44.1/48KHz operation, whereas the individual channel A-D and D-A will run via FireWire up to 96KHz. So you can use it as a mixer with 96KHz digital sources coming in via FireWire but with an analog 2-bus result, or as a multitrack digital recorder via FireWire at up to 96KHz.

The Alesis HD24XRs that I use for recording have analog and ADAT I/O up to 96KHz, but no FireWire. Their A-D and D-A converters are good, and so the easiest and best-sounding way of working for me is to have the HD24XRs produce high-bandwidth analog outputs that are mixed in analog by the Zed-R16. I could use the Zed-R16 FireWire host to give me a stereo 44.1KHz result, but actually I choose to split the main 2-bus analog outputs to go back to an HD24XR at 96KHz and also to a computer via an RME FF800 or via another independent box at 44.1KHz for the CD-rate capture.

Hey Boswell, thank you again. I read the manual regarding this and I understand the limitations. They are fine for my planned usage of the unit at this stage. I would be sending 16 x 96KHz audio stems / instruments from logic via the 16 firewire channels to the unit to complete the final mix in the analog domain. Capturing the final mix file on a device like the HD24XR is a very good piece of advice for a transparent final file. A second bounce to that machine could also produce a 44.1KHz version avoiding any SRC inside a DAW at any time. My main query about this unit is of the D/A converters sound as good as other high end audio interfaces like the Apogee Ensemble or Apollo.
What's worth noting, especially for those producing there own masters, is that you can supply 96KHz masters to I-tunes for distribution as they have optimised there conversion software to create the AAC and MP3 files from this resolution. There is a very good article about that here http://images.apple.com/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf including info on how to use there software to test how your tracks will sound after the conversions.

Kevin Big Jam, post: 394382 wrote: Hey Boswell, thank you again. I read the manual regarding this and I understand the limitations. They are fine for my planned usage of the unit at this stage. I would be sending 16 x 96KHz audio stems / instruments from logic via the 16 firewire channels to the unit to complete the final mix in the analog domain. Capturing the final mix file on a device like the HD24XR is a very good piece of advice for a transparent final file. A second bounce to that machine could also produce a 44.1KHz version avoiding any SRC inside a DAW at any time. My main query about this unit is whether the D/A converters sound as good as other high end audio interfaces like the Apogee Ensemble or Apollo. What's worth noting, especially for those producing their own masters, is that you can supply 96KHz masters to I-tunes for distribution, as they have optimised their conversion software to create the AAC and MP3 files from this resolution. There is a very good article about that here http://images.apple.com/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf including info on how to use their software to test how your tracks will sound after the conversions.

I would certainly rate the HD24XR (A-D and D-A) converters alongside the Ensemble and similar quality units. They are not in the very top league of stereo mastering converters, but on a per-channel basis are very good.

I know your plan is to source from the D-A converters in a Zed-R16 at 96KHz, but if you wanted to use HD24XRs instead, you would need two HD24XR units, as each has the capability of 24 channels at standard rates or 12 channels at higher rates.

If using an HD24XR to re-capture a 44.1KHz stereo mix, it's advisable to clock the HD24 from an external source, as its internal 44.1KHz clock is stable but slightly inaccurate. This HD24XR can't, of course, be the same HD24XR as you are using as a 96KHz source device.