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Author: Adam
Adam is a professional photographer, videographer and audio engineer. He started Real Home Recording back in 2011 and in 2017 launched Don't Go to Recording School.
Is this $500 software compressor worth the price? Watch the video review here:
As usual, here are the script notes:
Softube, 1:1 digital code translation. $10,000 hardware Weiss can be purchased for $419 at EveryPlugin.com DS1 sounds great on the mix buss. Sounds natural/transparent. Works well on tracks as well, particularly vocals. Just automate breaths! Missing AAX DSP. You can get the limiter by itself cheaper. It’s a de-esser, compressor and limiter.
Linear phase crossover filters
Touchscreen Controls
Upward expansion (helps with over-compressed signals)
Mid/Side
External Sidechain
Parallel Compression button
A/B comparison
Limiter gain reduction meter
Waveform Display View
3 Limiters: Hardware, Type 1 (highest RMS) and Type 2 (True Peak)
The easier to use MM-1 limiter is included
Knee adjustment
Bob Katz presets
3 page Options menu
Mouseover descriptions
Bob Ludwig mode…parameter settings are always displayed. Unfortunately they are in the shadow.
What makes this processor unique and work well is its release measurements/settings. You have four settings that control the release’s behavior. Average, fast, slow and delay.
Ganged mode? When stereo processing you usually keep this on. In
Every studio should have at least one nice hardware compressor. The SA-76 could be that piece. Here’s my review of Stam Audio’s new 1176 Rev. A clone:
And here are the script notes:
In this video I’ll be reviewing Stam Audio’s SA-76 feedback style FET compressor. It’s a hand assembled replica of the iconic 1967 vintage Urei 1176 Revision A.
Thanks to Stam Audio for making this review possible. They kindly sent me the compressor to review and keep. Therefore this is not an Audio Skeptics Society review but the SA-76 will be honestly evaluated. The 1176 Rev A, nicknamed the blue stripe, is perhaps the most sought after compressor in the world. Engineers love how it evens out audio volume levels and brings track details out with added mid range frequencies. It even has its own Wikipedia page!
1176 style compressors are also famous for the British mode or all buttons in trick which can work well on drum room mics, bass guitar and certain vocal styles.
Vintage units will run you $2,000 or more. Some have sold for over $10,000. Stam Audio sells theirs for a little under $700 plus shipping. The SA-76’s front panel features Input and Output knobs. Four ratio buttons along with an analog VU meter. Meter display buttons can be found on the far right side. On the back is a power switch, fuse and ground connector. Balanced XLR and ¼” phone inputs and outputs round out the back panel. Stam’s web site they talk about Transformer replicas, Vishay capacitors, Phillips capacitors and New Old Stock Carbon resistors. I honestly don’t know what any of this means other than the parts are combined to closely match the original hardware blueprint. But enough talk, let’s listen to some samples! Reviewing the SA-76 was part fun and part learning experience. Since my only previous hardware compressor experience was with FMR Audio’s Really Nice Compressor and an Aphex 320D Compellor, I didn’t know what to expect going in. Pictures don’t do this thing justice. It looks so small on a computer screen but in real life it is big. 3 ½ inches tall and 19” wide.
The attack and release knobs were not as expected. On the original hardware, the fastest settings are the highest numbers. Joshua, Stam Audio’s owner, on Gearslutz wrote that by request buyers can have their units set to the old way. Speaking of attack, it goes from very fast to ridiculously fast. The slowest is 800 microseconds. There are one million microseconds in a second…800 microseconds is FAST! The release is slower, at 50 milliseconds through its slowest 1.1 seconds timing.
Getting the routing in REAPER set up was easier than expected but still required more steps than a regular plugin. The Reainsert plugin is very useful for gainstaging and track alignment purposes. Another thing I learned was don’t trust the meter. Don’t even look at the meters. Many if not most software meters are nothing like real hardware. You can hear compression happening even when the gain reduction needle isn’t moving. Before unplugging it, turn your speakers off and then press the meter off button.
Tweaking an EQ plugin before compression, while the SA-76 was processing audio, is how I got things to sound good. The importance of feeding the compressor good quality audio cannot be understated.
Gainstaging is also very important, so I tried to stay under -8 dBFS as much as possible on the output and less than -12 dBFS on the way back into the audio interface.
Find the loudest part of your track to check for unwanted distortion.
With hardware you can use two hands at once! I read that the Revision A was noisy. The SA-76’s noise floor is quite low, even at the highest Input level settings.
There are no hardware presets. You twist the knobs until it sounds good. Then twist them until it sounds better. You have to use your ears, it is as simple as that.
Engineers love the Release on fastest. Particularly vocals. Try 3 and 7 (5 and 1 on the SA-76)
All Buttons in, try Attack slowest, Release fastest. Also try Release fastest (1) and Attack at 5 (3 on the SA76) Just as everyone said all along, this compressor works best with vocals, bass guitar and drums. I didn’t like it on piano at all.
Tone shaper. Especially with kick drums.
The Bad:
Sharp edges. Once it’s racked this isn’t an issue. Fingerprint magnet. I tried using Windex to clean it up before getting the product shots but that didn’t work.
No extended warranty option available.
Worth it over plugins? This is ultimately up to you and your budget. If you run a studio people will take you more seriously when they see pictures of your control room with rack gear.
More gain reduction can be applied vs. plugins. Using the compressor while recording will save time later on. Just a few decibels of gain reduction, slowest attack and medium release. If you mix it will only process in real time and obviously uses more electricity than just your computer. 10W isn’t too bad though…it’s less than some LED bulbs.
Twisting real knobs is a more gratifying experience than moving a mouse scrollwheel.
I believe that Acustica Audio plugins are the pinnacle of in the box equalizers and saturation plugins right now. Unfortunately their installation process isn’t exactly straight forward. So, here’s how to make it happen so you can try out some of the best EQs out there:
The script notes: After long last here is my review of the Audient iD14 audio interface.
I have been using the iD14 since February 2016. I know it very well at this point. It is has constantly been plugged in for over two years and I have rarely had problems with driver crashes on my Windows 7 machine.
The iD 14 features Two Class A Audient console preamps with balanced Neutrik brand XLR-¼” combo connectors Each preamp can deliver up to 56 dB of gain plus an additional 10 dB of software A Class A JFET instrument input Burr Brown Converters Selectable phantom power on each input channel Optical input for up to 8 additional inputs ¼” balanced Line level outputs Headphone output Big knob silver knob is called the encoder. Pressing it in quickly mutes outputs. Holding it down temporarily mutes it.. 8 segment LEDs display output level and volume settings. Assignable function to the iD button. ScrollControl, Sum to Mono, Sum/Difference, -15 dB Dim and Talkback Power supply with US, EU, UK and Australia socket adapters. Class compliant, so it can work with mobile devices. There’s also a Kensington lock, to prevent theft. You didn’t click on this video to hear me talk about features though. You want audio samples…so here are a few… If you want to find out more product specs, go to Audient’s web site. The rest of this video is going to be my experience and opinion after using the iD14 for over two years. Power Supply build quality feels cheap, as do the speaker outs. The wall wart uses adapters so you can use it in different countries. This cost savings is one reason the iD14 costs less than $300. Would be nice if power cable locked into place because it can be easily yanked out by accident. Headphone output issue. Seems to be a common problem. You have to twist the headphone connector around for both channels to come out. Drivers failed on me about four times in the past two years. A restart fixed the problem. Warranty: One year from the sale date for faulty parts and workmanship. Gain knob range is too broad from 7 o c’lock to 12 o ‘clock and then super tight from 3 o clock to 5 o’ clock. If an updated iD14 is released I would love to see the gain knob have a more even range. If you record virtual instruments, be prepared to bounce tracks. The low buffer settings only work effectively when CPU usage is under about 15-20%. Otherwise, gremlins appear. This may be updated with new drivers in the future, which are slated to give lower latency capability on Windows plus a new control panel interface. Right now RME interfaces are still your best bet for the interface with the best low latency and high CPU stability. That’s the bad. There’s a lot more good. Full color nicely designed PDF manual available for download. It explains every hardware and software control in detail. Installed the drivers no problem. Plugged it in and the interface was detected without issue. It’s powered off USB if you use microphones that don’t require phantom power. I did have to download/install firmware, it’s the first thing I checked in the control panel software under Help>Check for updates . The upgrade process was very easy…just point to the .bin file and click next. Close out of all other programs. In windows you must change playback devices Advanced options to 24/96 otherwise it will revert to 44.1 kHz by default. Does not have DSP effects. Although you can get a little more volume out of the main mix, cue mix is nearly real time.
– 10/10 on tech support. Customer service rep Tom sent me detailed emails on how to fix a problem I was having with the interface. REAPER puts out a higher amount of signal in ASIO mode. The problem was resolved with a suggestion to switch to Direct Sound mode and then I got a few more extra tips to help with speaker calibration and output gain staging. I didn’t think it would work with Camtasia for screen capturing. This allowed me to finally sell my old mixing board. A very sturdy feel. Doesn’t feel cheap at all with one exception. The ¼” phono line outputs have an inner ring which appears to be plastic that wiggles a bit when you insert and remove cables. If Audient makes a new version of the iD14 I would recommend to them to use the same ¼” connector type that the headphone output and DI input uses. I don’t think it’s a major issue because you aren’t pulling line output cables in and out as much as headphones or guitar cables. Noise specs are very nice. Here are some tests I did a few weeks after receiving the iD14. And here are some more recent test results. Power cable length: A little over six feet.
Only +12 dBu of headroom…so Audient advises peaking no more than -12 dBFS for a clean input. Running a -12 dBFS sine wave out to it showed it perfectly at -12 on the Controlling the headphone and speaker volume in the software is awesome.. I also really like that you can rename the inputs and outputs. The font looks handwritten, which is a nice touch. The control panel software is one of my favorite things about the iD14. It’s easy to use and not as cluttered as some other software I’ve used. The important controls are upfront and you can hide panels. Lesser used options are hidden away. I think software preamp boosts are silly/possibly confusing for newbies. It’s a feature I do use when playing video games, however. The first test I did was to record a voice over with a Shure SM7b. Sounded quite nice and the preamp did have enough juice to boost it. Glenn Fricker says the Audient preamps remind him of API 512 preamps. A Sweetwater reviewer said it sounds like a Focusrite ISA with less noise. Another guy on Zenpro Audio said the preamps are equal quality of his Shinybox Si4 preamps. Whatever the case is, they have a detailed, slightly bright sound that isn’t harsh. They are the best sounding preamps I have personally used.
Converter wise this interface has chips that are normally out of its price range. Burr Brown PCM4202 for the ADC and Burr Brown PCM4104 for the DAC officially. Some interfaces that use the PCM4202 are the Metric Halo 8×192, Mytek Stereo 96 ADC version 6, and the Mytek 8×192 ADC, Interfaces that use the PCM4104: RME Fireface UFX, RME Babyface, Presonus Firestudio Mobile Source and the Allen & Heath Zed-R16: Source: https://www.gearslutz.com/board/product-alerts-older-than-2-months/1009761-audient-id14-audio-interface-available-now.html That doesn’t mean that this sounds just like those other interfaces. As important as the converter chips are, the wordclock, capacitors, switching supply, op amps and the overall design of the circuitry is what affects the sonic quality of interfaces. I wish it was in the budget for me to crack this thing open and take a look at its guts but that’s not going to happen. First Impressions from 2 years ago: – Installed the interface as my main soundcard on Saturday Feb 20, 2016. I don’t know if it’s a placebo effect but I did notice a quality difference. Not night and day but it sounds like there is more low end detail and the high end is smoother. The sound also appears to not sound stuck to the speakers and therefore the phantom center and areas in between sound more 3D. And again I don’t know if this is a placebo effect or not but I started noticing subtle acoustic reflections more often. Most importantly though, I can turn my speaker volume knob all the way up and there isn’t any noise unlike my built in sound card. That’s incredible!
I would have liked a marker arrow or indentation on the preamp knob. The instrument input holds up nicely to my standalone direct boxes. While I won’t be shelving the J48 or BigAmp, I don’t feel the quality suffers. You can watch electric guitar and bass guitar direct box shoot out videos which are linked below. Headphone amp has plenty of clean gain. When you turn it up very loud it doesn’t distort much. Audio-Technica ATH-M50 as my reference headphones, you couldn’t hear any noise. However, with Sony MDR-7506 headphones I could. This is with USB power. Quiet pops when switching sample rates No power switch
All metal construction means you may have issues with static electric build up when touching the preamps and plugging in microphones. – Gain knob bunching or top heaviness. Instead of evenly spaced gain steps the last 3 o clock to 5 o clock area has a large variance in gain level. That’s the major difference between the preamps on their large consoles and their smaller products.
Volume resets to zero when you shut your computer off. Unless you leave the power cable in (test this). This may be a positive though.
No MIDI. That’s OK, get the iConnectivity mio instead for about $35.
LEDs do not display input levels. You need to look at your computer screen to see that. That’s just as well, since software meters are better/more sensitive and precise anyway, in my opinion. Centrance ASIO Latency Test Utility 3.7 round trip latency results (rounded numbers): 44.1 at 64 samples: 362 samples / 8 ms 48 at 64 samples: 347 samples / 7 ms 88.2 at 128 samples: Could not test 96 at 128 samples: 573 samples / 6 ms The Cue Mix is quieter than the Main Mix due the extra slider headroom. So, when recording it may be better to set speakers to Cue mix and headphones to Main Mix.
Line in test. A concern of mine was that they go through the preamps. I ran a bunch of songs that I am familiar with from a variety of music genres through the line ins. I was very happy with the results. You can run tracks through them for about passes before quality goes down. You can watch the analog generation loss test video. You can download a copy of those tests…the link is on RealHomeRecording.com The only real negative is that you only get two inputs and two outputs. That’s not really a negative to me though due to the ADAT input. In fact, the ADAT input sealed the deal for me. If you guys recall my video about two high quality inputs there’s a lot that you can do with just two inputs. When you want 10 inputs for recording a whole drum kit or whatever you have that option. Unfortunately you are limited to six total inputs at 24/96 or 10 inputs at 24/48. That’s a limitation of the SMUX format. Wrap up: In closing, I want to say that I’m glad I waited about seven years to buy a new audio interface. You better believe that whenever I saw new interfaces on Gearslutz or whatever web site I salivated over interfaces with nice built in amps and high end converters. I never thought I’d be able to get that level of quality for under $300 but here in 2015 Audient changed the game. Two years later and Audient is still supporting it with upcoming driver updates and through their third party ARC Creative Club add-ons. Cubase LE, Cubasis LE2, Eventide Reverb and Ultrachannel plugins, 10 LANDR masters and two free Producertech.com audio courses are included as of March 2018 Please watch some of the companion videos and listen to the raw audio files which I’ll provide links to in the video description.
I was going to do a tutorial on Top Down mixing but after playing around a bit I found a better mix strategy Another mix workflow tutorial.
If you’re not familiar with the concept of top down mixing, there is a school of thought that the less digital signal processing the better. Purity of the signal is preserved with the less processors that are used.
Saves CPU power
Saves time
Top Down Mixing its name from the signal flow hierarchy. The master buss is at the top then group busses and then individual tracks. After experimenting with the top down mixing strategy, I discovered that I prefer to start with busses first. Here’s how to do it.
If you have Hornet’s VU Meter plugin, run that first. This will get your initial gain staging in check and makes the second step easier. It’s a very cheap plugin that I recommend everyone buy. Next, let the song play and adjust anything that is obviously too loud or too quiet using track faders. The third step is to create your group busses. Put EQ, saturation and compressor plugins on each then adjust them. If you are a fan of processor heavy software like Acustica Audio, this is a great place to take advantage of the CPU savings.
After the busses are sounding good, go to your master buss. Add saturation, EQ and buss compressor plugins. Maybe a little bit of room reverb as well. Last, a brickwall limiter to protect speakers. Don’t go overboard and be mindful of your gain staging.
Next, set up your group busses. Again, EQ and saturation plugins and maybe compression.
1. Save session to appropriate folder.
2. Drop in just the lead vocals.
3. Let Hornet VU Meter do its thing if gainstaging is bad.
4. A crappy brickwall limiter set to bring levels up about +10 decibels and Sonarworks Reference on master channel
5. EQ cuts on lead vocal track. If there is more than one track, make a buss and EQ on it.
6. Lead vocals compression + tape and console saturation. At this point, drop speakers volume down pretty low and add pre-FX volume automation. Automate the lead vocals until you can hear every word at that low volume.
7. Add background vocals.
8. Mute Lead vocals, for now.
9. Gain stage and create a buss.
10. EQ and compress the buss. Add tape/console saturation as well.
11. Unmute lead vocals. If necessary, thin out background vocals (the 3 to 5 kHz range is usually effective here by cutting)
12. Add second most important element of mix. For me, this is usually the drums or lead guitar.
13. Repeat the usual steps. Clean EQ for cuts, compression then character EQ for boosting. After that, tape and console saturation.
14. Drums usually get the 1176 all buttons in effect.
15. After everything is sounding pretty good overall throughout the whole song, it’s time to start panning and adding delays/reverb. I like a pan law of -4.5.
16. Master buss processing is last. Console saturation plugin first then usually an EQ then compression. Follow that up sometimes with a VERY small amount of room reverb. Finally, 1/2″ tape saturation.
17. At this point, adjust the brickwall limiter’s threshold slider. -10 dB RMS is a good compromise for too quiet vs. too compressed.
More and more I am releasing music where I want to avoid the brickwall limiter at all costs. But, the customer gets the final say on that. The important thing with all of this for me is to keep headroom. The brickwall limiter setting at +10 dB from the beginning helps keep everything in check.
I use Ozone on IRC I mode with the Character slider on 0.00 because if my levels are going over -10 dBFS I want to hear it! IRC III mode on clipping also sounds bad but uses more CPU.
One caveat with bussing everything is it makes archiving tougher. I think I will save compression (except drums) for individual tracks. So, clean (IIEQ Pro, IK White Channel or Fabfilter Pro Q2) cuts on individual tracks + compression then routed to the busses for “character” EQ like PINK, Navy, Purple/Ruby/whatever.
The way I work with my own recordings is that Mix on the Go strategy: https://www.youtube.com/watch?v=1SC7J_HdyQk That may not work for everyone but it does for me because I’d use real hardware EQs/compressors on the way in before hitting the A/D converter but I don’t have $20,000 to spare for music production. So, “Mix on the Go” it is!
I usually save master buss processing for last. A lot of engineers like to mix into a compressor but I never liked doing that. Compared to some tracks that I’ve seen on Sound on Sound, I use a lot less plugins than other engineers with the exception of vocals. Vocals are a pain in the ass and require a lot of automation and processing to sit on top of the mix. Especially if a condenser microphone is used!
The product page claims that One click sets up your system and the only thing you have to care about is the fine tuning and gentle saturation of your sound. You save your time and insert slots and CPU power.
User reviews on the web site were solid, so I was really looking forward to trying Drum Enhancer out. For this video version 1.0.2 was used.
Level in – Input trim. Set so that the signal is around 0. So, peaks are happening in the yellow lights.
Noise gate, compressor, EQ, saturation,
By default “0” on the meters is -9 dBFS. This can be changed.
Go through the controls.
EQ Position – Before or after compressor
Saturation: Presence is enhanced upper mids.
Vintage – Smoother high end, even harmonics
Brown – Even frequency response except a touch of high end enhancement
White – Enhanced highs
Lofi – Less lows and highs
When you change the drum type, under the hood a bunch of parameters are changing. Compressor and EQs. EQ frequencies and bandwidths. Compressor attack and release times. Knee shapes and sidechain filters. High and low pass frequencies. Noise gate attack/release times.
The quickest way to set the plugin up is Change default calibration level to peaks around 0. Then choose a preset that sounds good.
Advanced Method:
– Input trim so peaks are hitting around “0”.
– Choose Drum Type
– Phase inverse if needed
– Noise gate controls
– Compressor controlers
– EQ controls
– Compressor Makeup (this is pre-saturation module)
– Saturation flavor and then amount
Input/output control link?
Not easy to cycle through presets.
Inaudible signal when track was paused.
Low CPU bothers me. There are multiple modules happening. On other plugins, this would add up to more CPU. Not a fan of the EQ. Saturation seemed good and whatever is enhancing worked but the comp was also not that great either.
Fine for quick mixes but I would recommend Waves Signature Series Bass and Drums over this. Right now it’s on sale for $89 they are cheaper and better than DW Drum Enhancer. But as always I suggest trying all the products out though and deciding for yourself where to spend your hard earned money.
This seems to be a common question for new audio engineers. This is what comes after mix prep. I have an in depth tutorial on how to prepare a mix already, which is linked down below.
First, make sure that your session sample rate matches your files. Then, save your project. Next, turn your audio interface software faders up. When mixing, you should not peak past around
-8 dBFS on the meters. So, they’ll need to be louder than when mixing to mastered music or internet videos. Otherwise, you will lose headroom.
Instead of adjusting your software faders a trick I do is to put Ozone on IRC I mode or LoudMax on the master and set the threshold control to -10 dB. This way, the overall volume of the mix is
brought up by 10 decibels. Then, I just make sure my track peaks don’t go past about -8. It’s important to use a limiter that doesn’t sound too good, so IRC mode I is what I go with.
The kick drum is usually the track hitting the highest peak. You save the volume raising for the final steps.
Reference 4 or ARC is then put as the last plugin. I could put it on the monitoring FX panel but it’s easier to disable it on the master channel.
Everything is done in mono at first with no spatial effects. Reference’s mono switch works in this case. Some people swear by disabling one speaker and moving the active speaker to the center spot. I’m usually fine with dual mono though.
At this point, drop your vocal track in. I’ve found that doing tracks one by one is less intimidating than dropping say 40 tracks into a session all at once. It keeps you focused on the task at hand. I prefer to start with lead vocals because they are the most important element in most mixes. They should be the center of attention. People tend to sing along to songs more than they play air guitar or air drums.
Trim the volume down if necessary then EQ for clarity, a couple compressors plus tape and console emulation plugins. Reverb, delay and other plugins like that will be added later.
After the vocals are sounding good. Enable volume (pre-FX) automation. Lower the software volume control so that you just barely hear most of the lyrics. The goal is to be able to hear every word at a low volume. Add background vocals next and give them the same treatment. Solo’d at first and then add the lead vocals in.
Next, add the second most important element of the song. If you or the musicians you’re mixing for can’t decide then go with rhythm guitars or piano. That’s because I’ve found these elements tend to have frequency clashes with vocals the most. I like to solo this track at first and then add vocals in once I get EQ and compression sounding good. Start at the final chorus or crescendo of the mix where most if not all tracks will be playing simultaneously. Again, we’re only going to use equalizers, dynamics and saturation processing for now. The goal is to be able to hear every note along with the vocals.
Your third most important element goes next. In a typical mix, only three elements can usually be heard loud and clear. For a rock mix, lead guitar usually comes next. It can clash with the rhythm guitars and vocals pretty easily. Again, at first I’ll solo the track, EQ and compress to get it sounding good and then I’ll bring the rest of the tracks in one by one starting with vocals. Bass comes next. Once again, start solo’d then fade it up with the rest of the mix. This reinforces the rhythm section and tends to stay out of the way. You should be able to hear most bass notes without a subwoofer. If not, check out my tutorial on how to make this happen linked below. If you have synth parts I’ll usually put these next.
Next, drums. Maybe put a sidechain on the bass that is triggered off the kick drum. Make a bus for the all buttons in 1176 trick (Slate Digital Monster is a free effect that simulates this quite nicely).
Everything else. Again, start at the busiest part of the mix which is usually the final chorus. Everything should sound pretty good when completely dry. This ensures the greatest amount of track separation. When the dry mix sounds good in mono, it’s time to start adding reverb, delay and panning to the equation. Post effects fader adjustments will be necessary. Automation is the last step. This should take about 1-2 more hours and it adds life to the mix. It adds variety. It lets parts stand out. When less tracks are playing, they should sound bigger.
If you go to school for a degree that won’t pay immediate dividends upon graduation than you are a cow. A cash cow. Coach Red Pill explains why in his video thorough video:
I’ve layed out the arguments as to why–unless you are going for a degree in medicine or similar profession that has many job openings–you shouldn’t go to college. Especially for audio engineering. But, if you still are not convinced then watch this video:
