Monday, September 15, 2014
Drawmer Introduces 1973 FET Stereo Compressor, Distributed In U.S. By TransAudio Group
Includes three independent compressor sections with two variable-frequency 6 dB/octave crossovers to separate them into low, middle, and high frequency sections
Ivor Drawmer, maker of analog—and now digital—signal processing equipment, drew on his 30-plus years behind the soldering iron to create the Drawmer 1973, a new 3-band FET stereo compressor. It is being distributed in the U.S. by TransAudio Group, available now at a price of $1,825.
The Drawmer 1973 includes three independent compressor sections with two variable-frequency 6 dB/octave crossovers to separate them into low, middle, and high frequency compression sections.
Each section contains familiar threshold, gain, attack, and release controls, along with gain-reduction metering. Moreover, each section can be independently muted or bypassed for confusion-free setup and monitoring.
The low section possesses a “Big” switch for enhanced low-end, whereas the high section possesses an “Air” switch for enhanced high-end.
The three sections are recombined to form the “wet” signal, which can be mixed to variable degree with the dry signal for easy parallel compression. Illuminated VU meters make monitoring compression and output intuitive and, yes, fun.
“Certainly, the Drawmer 1973 owes some of its sound and functionality to Ivor’s experience designing the classic Drawmer 1960 and 1968 compressors, as well as to the Drawmer S3 signature series multiband tube compressor,” says Brad Lunde, president of TransAudio Group. “But it also has a sound and operation all its own. It’s capable of solving problems single-band compressors simply cannot, such as compressing only the low end, raising its average level relative to everything else, and giving your mix a bit more bass without changing the overall level.
“It has a sound quality that cannot be matched by other analog processors, never mind plug-ins,” he continues. “It will be popular among mixers and EDM mixers alike. The 1973’s layout is impressive. Unlike most other multiband compressors, the 1973’s controls are easy to understand at a glance and work to inspire creative use.
“The real news here may be the 1973’s affordable price. Those in need of stereo multiband compression with Drawmer’s quality can have it for the cost of Drawmer’s famous single-band stereo tube compressor, the 1968.”
Sunday, September 14, 2014
In The Studio: Mastering Your Songs In Six Steps
When I began writing the latest 3rd edition of The Mastering Engineer’s Handbook, one of the things that I wanted to find out from some of the mastering greats was how they approached a project.
In other words, what were the steps they took to make sure that a project was mastered properly. Interestingly, the majority of them follow six primary steps, some consciously followed and some unconsciously. Here’s an excerpt from The Mastering Engineer’s Handbook that outlines the technique.
If you were to ask a number of the best mastering engineers what their general approach to mastering was, you’d get mostly the same answer.
1. Listen to all the tracks. If you’re listening to a collection of tracks such as an album, the first thing to do is listen to brief durations of each song (10 to 20 seconds should be enough) to find out which sounds are louder than the others, which ones are mixed better, and which ones have better frequency balances. By doing this you can tell which songs sound similar and which ones stick out.
Inevitably, you’ll find that unless you’re working on a compilation album where all the songs were done by different production teams, the majority of the songs will have a similar feel to them, and these are the ones to begin with. After you feel pretty good about how these feel, you’ll find it will be easier to get the outliers to sound like the majority than the other way around.
2. Listen to the mix as a whole, instead of hearing the individual parts. Don’t listen like a mixer, don’t listen like an arrangement and don’t listen like a songwriter. Good mastering engineers have the ability to divorce themselves from the inner workings of the song and hear it as a whole, just like the listening public does.
3. Find the most important element. On most modern radio-oriented songs, the vocal is the most important element, unless the song is an instrumental. That means that one of your jobs is trying to make sure that the vocal can be distinguished clearly.
4. Have an idea of where you want to go. Before you go twisting parameter controls, try to have an idea of what you’d like the track to sound like when your finished. Ask yourself the following questions:
—Is there a frequency that seems to be sticking out?
—Are there frequencies that seem to be missing?
—Is the track punchy enough?
—Is the track loud enough?
—Can you hear the lead element distinctly?
5. Raise the level first. Unless you’re extremely confident that you can hear a wide frequency spectrum on your monitors (especially the low end), concentrate on raising the volume instead EQing. You’ll keep yourself out of trouble that way. If you feel that you must EQ, refer to the section of the EQing later in the chapter.
6. Adjust the song levels so they match. One of the most important jobs in mastering is to take a collection of songs like an album, and make sure they each have the same relative level. Remember that you want to be sure that all the songs sound about the same level at their loudest. Do this by listening back and forth to all the songs and making small adjustments in level as necessary.”
Following these steps just like the mastering greats do will ensure that not only will your project sound better, but you’ll avoid some of the pitfalls of mastering your own material as well.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and go here for more info and to acquire a copy of The Mastering Engineer’s Handbook.
Friday, September 12, 2014
V3.0 Update For Yamaha CL/QL Consoles Further Expands Feature Sets For Live Sound & Broadcast
For live sound applications, V3.0 will add the benefit of a new 8-band parametric EQ and real time analyzer
Version 3.0 updates, available in early 2015, have been announced for Yamaha Commercial Audio CL and QL Series digital consoles, providing increased capabilities in both live sound and broadcast applications.
Note that there won’t be a QL Series version 2.0 so that the CL and QL Series Consoles will be at the same V3.0 level. CL and QL files are interchangeable, making it easy to choose and combine models to accommodate front of house and monitoring as well as broadcast relay and recording, from basic to large-scale applications.
For live sound applications, V3.0 will add the benefit of a new 8-band parametric EQ and real time analyzer. Specifically, an 8-band PEQ in the GEQ Rack and Effect Rack making it possible to select 8-band Parametric EQ in the GEQ RACK and EFFECT RACK.
With the real time analyzer (RTA) to support room tuning and other operations, the frequency spectrum of cued channels can be shown in the new RTA display as well as in the PEQ or GEQ display to provide visual feedback while adjusting EQ.
Also of note for live sound applications, V3.0 will provide four banks of enhanced User Defined Keys, not only reducing the possibility of running out of keys, but also allowing keys to be grouped by function for improved efficiency.
DCA Assignment Selection for Scene Management has been added to the parameters for Recall Safe, Focus Recall, and Global Paste. Rather than only being selectable with all parameters during scene memory management, DCA assignment is now an independent setting. This allows more refined control, such as specifying only Channel Name, Fader, and DCA Assign to be Recall Safe, for example.
For broadcast applications, V3.0 delivers 5.1 panning and monitoring for surround broadcasts and a newly developed bus compressor for insertion in the stereo mix bus.
Pan positioning can be set via the touch panel or knobs. Mix to Matrix can be used for international feed production, and Mix to Stereo can be used for stereo mix down. In addition to surround mixing, V3.0 adds basic surround monitoring. Monitor alignment capability is also provided, with adjustment of relative loudspeaker levels and delays.
Also for broadcast applications, Dan Dugan Sound Design automatic microphone mixers already included in QL Series is now included in CL Series consoles. Gain distribution for up to 16 speech microphone channels is automatically optimized in real time, achieving smooth, natural level control.
Yamaha R&D has created an accurate VCM (Virtual Circuitry Modeling) model of an popular bus compressor utilized in broadcast and recording studios. The new Buss Comp 369 is specifically designed for inserting on the stereo mix bus for increased loudness, more uniform levels, and warm overall sound.
Mix minus, considered important for relay broadcast applications, previously available for CL consoles, is now available in the QL consoles. The signal from a specified channel can be easily omitted from a specific bus, making it easy to quickly create a mix for the location reporter, for example, that may not include his/her own mic feed.
The new Frame Delay capability enables the audio engineer to delay the audio at the console in order to achieve proper synchronization, a necessity in broadcast and in live applications where video is being used. The feature makes it possible to set that delay in frame increments for easy sync with a wide variety of video formats.
“We’re very pleased to brings these additional upgrades that have been requested by our users,” states Marc Lopez, marketing manager, Yamaha Commercial Audio Systems. “Our loyal sound reinforcement engineers will benefit from several great tools whether they are mixing on CL or the new QL Series as well as a whole new set of amazing features for our broadcast users. In addition to the new features in V3.0, we are currently working on a number of additional refinements that will provide even greater operability and stability and will continue to deliver products and updates that contribute to the most efficient and comfortable working environment for our customers.”
CL/QL V3.0 firmware will be available as a free download from the Yamaha Commercial Audio website, www.yamahaca.com, in early 2015.
Yamaha Commercial Audio
Riedel Announces New Tango Networked & Expandable Platform With Its Own Intercom Application
Platform can be turned into a flexible, high-end solution for a variety of communications scenario
Riedel Communications has announced the release of Tango TNG-200, the company’s first fully networked platform based on the AES67 and AVB standards.
With its own dedicated intercom application, the platform can be turned into a flexible, high-end solution for a variety of communications scenarios.
Riedel’s Tango TNG-200 is an efficient solution equipped with a high-resolution, full-color thin-film-transistor display that ensures readability at all times. The unit’s intuitive front-panel controls simplify the recall of presets and adjustment of audio levels.
Along with powerful processing capabilities, the Tango TNG-200 offers two integrated Riedel Digital Partylines, two AES67 and AVB-compatible ports, two Ethernet ports, one option slot, and redundant power supplies.
The system is 1.5 RU high, and has a shallow mounting depth and low-noise design. It’s fully compatible with all of Riedel’s current and legacy intercom panels, including the company’s new RSP-2318 Smartpanel.
Riedel’s TNG-200 turns the Tango platform into an efficient intercom system that users can tailor according to their needs, including choice of matrix size starting with 40 x 80. The asymmetric 40 x 80 matrix size is another Riedel development, allowing for standard premium-quality stereo audio connection to Riedel panels.
New Pulse software enables configuration of the Tango TNG-200 platform. The software allows users to access, set up, and control any aspect and function of the platform and its installed applications, including Riedel’s intercom application. Programming is easy, thanks to convenient drag-and-drop functionality and 3D views.
“We have always considered the networked approach to signal distribution to be the most desirable option for today’s broadcasters,” says Thomas Riedel, CEO of Riedel Communications. “We have been on the forefront of adoption of standards for several years now, and we are proud to introduce at IBC2014 the world’s first networked and expandable open hardware platform that can also run a powerful intercom application. As an exceptionally flexible platform, Tango not only adapts to users’ specific requirements, but also accommodates current and future standards in broadcast, theater, and live event environments.”
Thursday, September 11, 2014
In The Studio: My Top 10 Microphone Mistakes
Do you ever revel in someone else’s mistakes? Ever learn something from them?
Yeah, me too.
Here’s a list of some of my “best” mistakes I’ve made when it comes to using mics.
’Tis both enjoyable (and educational):
1. I almost blew up a $1,600 ribbon mic because I plugged it into a preamp with phantom power already on. (Turns out that’s a bit of a myth, but I freaked for a while.)
2. I stepped up to the mic in front of 400-plus people to sing, and…I had forgotten to turn on the wireless mic.
3. I sang an entire take of a vocal into the back of a condenser mic without realizing it.
4. I bought one of those headword condenser mics (a.k.a., the “Garth Brooks mic”) to do podcasts and webinars. Turns out it sounded like garbage, and I looked like a dork.
5. Recently tracked drums and didn’t realize I overloaded the overhead mics at the preamp. (Sounded cool in the end, but embarrassing that I didn’t realize it was happening at the time.)
6. Tracked lead vocals for an album through a (Shure) SM7B from roughly 1-foot away. It sounded okay, but had way too much sibilance. I was too far from the mic.
7. Spent a day tracking acoustic guitar (with two mics), only to realize afterwards that I had the mics too close and was recording a very boomy-sounding guitar.
8. After technical difficulties setting up a headphone mix, I tracked a female vocalist without really checking to see if I liked the vocal tone. Turns out I didn’t like it that much.
9. Close-miked a lead vocal once on a condenser with a hypercardioid pattern. Ended up sounding really weird due to the exaggerated proximity effect.
10. Recorded acoustic guitar for an EP right next to a window, while it was raining (during the great 2010 Nashville flood). The sound of rain is all over the EP.
While we’re on the topic of micropohones, be sure to check out Joe’s recent free webinar: No Frills Guide to Choosing and Using Microphones in Your Home Studio.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner. Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
Full Compass Hosting Yamaha NUAGE Demo, Nuendo Master Class In Madison, WI
David Lees and Marcel Mauceri of Yamaha will be on hand to demonstrate NUAGE and discuss the latest tips and tricks of Nuendo recording software
A NUAGE advanced production system demo and Nuendo Master Class will be presented by Yamaha Commercial Audio and hosted by Full Compass Systems on Tuesday, September 23, at 9770 Silicon Prairie Parkway, Madison, WI.
The event is free of charge. Product tours will begin promptly at 6 pm, 7 pm and 8 pm.
A joint collaboration between Yamaha and Steinberg, NUAGE offers workflow-efficient hardware and Nuendo 6 software operating together in harmony. It enhances productivity and flexibility while delivering very high audio quality in an innovative design.
David Lees and Marcel Mauceri of Yamaha Commercial Audio along with Full Compass staff will be on hand to demonstrate NUAGE and discuss the latest news, tips and tricks of Nuendo recording software.
Full Compass Systems
Yamaha Commercial Audio
Fairlight Launches Compact QUANTUM.Live Table-Top Digital Console
Comes with faders accommodating 144 signal paths over 12 layers and delivers fast tactile access and command over two monitor systems
Fairlight has launched QUANTUM.Live Table-Top (TT), the newest addition to its live console family that includes the EVO.Live digital mixing system.
Based on Fairlight’s audio processing and control surface hardware, the range of consoles can switch between live and post production at the touch of a button, with models available from 12 to 60 faders, and further, as stand-alone chassis, in-surface modules and table-top configurations.
The entry level QUANTUM.Live TT is the smallest console in Fairlight’s Live line-up. It comes with faders accommodating 144 signal paths over 12 layers, and delivers fast tactile access and full command over two monitor systems. A second TT frame can be added, increasing the system to 24 faders.
At the core of all Fairlight Live consoles is a powerful audio processing engine designed with FPGA (Field Programmable Gate Array) technology. The audio processor combines Fairlight’s proprietary Crystal Core engine, a redundant power supply, all interfaces for control screens, GPIOs, storage, local control room I/O, MADI connections for remote I/O, and other system hardware.
A QUANTUM.Live Table-Top base configuration with 12 faders, a center section, audio processing engine with 48 channels, 32 buses and a full complement of local audio I/O, starts at U.S. $40,000 (€30,000).
The Visual Microphone: Passive Recovery Of Sound From Video
There’s some interesting research going on regarding the ability to capture sound via video of common items in a room, such as a bag of potato chips or a house plant. The project is a joint effort between MIT, Microsoft and Adobe.
As the brief on YouTube notes: “When sound hits an object, it causes small vibrations of the object’s surface. We show how, using only high-speed video of the object, we can extract those minute vibrations and partially recover the sound that produced them, allowing us to turn everyday objects—a glass of water, a potted plant, a box of tissues, or a bag of chips—into visual microphones.
“We recover sounds from high-speed footage of a variety of objects with different properties, and use both real and simulated data to examine some of the factors that affect our ability to visually recover sound. We evaluate the quality of recovered sounds using intelligibility and SNR metrics and provide input and recovered audio samples for direct comparison.
“We also explore how to leverage the rolling shutter in regular consumer cameras to recover audio from standard frame-rate videos, and use the spatial resolution of our method to visualize how sound-related vibrations vary over an object’s surface, which we can use to recover the vibration modes of an object.”
The video below explains more, and there’s additional information as well as a white paper available here.
Avid Announces New VENUE | S3L-X Compact Live Mixing System
Builds on S3L system in making it easier to meet the increasing scope, size, complexity, and diversity of modern live sound requirements
Avid has announced the new VENUE | S3L-X compact live mixing system, building on the Avid | S3L system in making it easier to meet the increasing scope, size, complexity, and diversity of modern live sound requirements.
VENUE | S3L-X enables engineers to mix and record live shows with power and efficiency, and create new material or mix down live recordings for commercial purposes. When used together, S3L-X and Pro Tools software—both powered by the Avid MediaCentral Platform—deliver a streamlined, economical, and tightly integrated approach to mixing and recording live productions.
“Live sound professionals face tight budgets, pressure to get more value from their assets, and a multitude of technology choices,” says Chris Gahagan, senior vice president of products and technology at Avid. “When we first introduced the S3L, our customers immediately embraced it at major festivals, clubs, and tours for its sheer sound quality, power, and mixing creativity. Now with the Avid VENUE | S3L-X we’re enabling them to use one system for both live sound performance and mixing down final assets to monetize.”
The new system delivers expanded networking, control, and processing. Users can reduce complexity and cost by sharing the same I/O across multiple S3L-X Systems, with full automatic gain compensation. With support for 64-bit AAX DSP plug-ins, plus the open EUCON and Ethernet AVB network protocols, it ensures compatibility with a variety of Avid and third-party products now and in the future.
S3L-X utilizes new VENUE 4.5 software:
• Share the same I/O across two or more Avid S3L-X Systems, with complete auto gain compensation
• Mix DAW sessions using Avid S3 as a stand-alone mixing surface and 4 x 6 audio interface
• Keep pace with the latest sound processing with 64-bit AAX DSP plug-in support
• Scale the modular system to accommodate any size performance, from 16–64 mic pres
• Get reliable Ethernet AVB connectivity without the cable bulk
• Achieve maximum performance with 2x more RAM compared to Avid S3L
• Record directly to Pro Tools (or other DAW) through a simple laptop Ethernet connection
• Get reliability for the road with the rugged, reinforced engine design
• Monitor through the high-output, low-noise headphone amp
The new Avid VENUE | S3L-X will be available early Q4 2014 through Avid resellers worldwide, while the VENUE 4.5 software upgrade will be available in late Q3 2014 to current S3L System customers.
Wednesday, September 10, 2014
There’s Still Time To Enter DPA d:screet Mic Competition – How Do You Wear Yours?
Up to 100 participants have opportunity to win a d:screet Necklace mic
There’s still time to enter the DPA Microphones “How Do You Wear Yours” competition, offering up to 100 participants with the opportunity to win a d:screet Necklace microphone, which was introduced earlier this year.
The competition, running on DPA’s Facebook page and website, asks contestants to come up with an ingenious idea for how and where to use the d:screet Necklace mic. All entries must be submitted by September 28, 2014.
Five microphones will be awarded to the five most “liked” submissions on Facebook, while the remaining 95 will be given to the people who, in DPA’s opinion, come up with the best, most creative and most original ideas. An example of one of the ideas can be found at this YouTube video, created by DPA Microphones, here.
Incorporating a d:screet 4061 omnidirectional miniature capsule in a soft rubber necklace, the d:screet Necklace mic, as the name implies, simply attaches to a necklace catch, useful for situations where mounting and consistent audio output are the primary requirements.
“We’re extremely excited about this product and want to share our enthusiasm with the rest of the world,” says DPA Microphones CEO Christian Poulsen. “These new microphones clearly meet a unique production need because they can be mounted and removed quickly several times by untrained talent without a sound expert nearby. We are well aware that they are perfect for reality show settings as they have already been used for the Danish adaptation of Big Brother. But, we think there are plenty of other situations where their simplicity, audio quality and size make them ideal. We hope that people entering our competition will use their imagination to come up with some really inventive ideas to surprise and amuse us.”
The competition also has a second phase for those who win one of the initial 100 d:screet Necklace mics. Once winners receives their microphone, they can put their ideas in motion, documenting them and posting pictures or videos to the company’s Facebook page. Two grand prize winners will receive a collection of DPA Microphones of their choosing, up to a value of $1,300 USD (list price).
Entry to this competition is via the DPA website or Facebook page. Again, all entries must be submitted by September 28, 2014. The 100 initial winners will be announced on October 1, 2014. Contestants can enter multiple ideas for an opportunity to win multiple necklace microphones, but only one microphone will be awarded per idea.
Entry to phase two of the competition will open after October 1, with the two ultimate winners announced on November 10, 2014. The first grand prize will go to the most liked picture or video on Facebook, while the second will go to the application idea that DPA perceives to be the best and most well documented.
To enter and for more information go to the company website (here) and/or Facebook page (here).
JoeCo Launches New Flagship BlackBox BBR1MP Recorder
24 channels, 24 integrated high-quality mic pres and 24-bit/96 kHz performance in 1RU
JoeCo has unveiled the new flagship in its BlackBox line, the BlackBox BBR1MP recorder, a 24-channel, stand-alone unit delivering all standard BlackBox functionality while also offering 24 in-house developed microphone preamps and operating at up to 24-bit/96 kHz.
Housed in a 1RU package, the BBR1MP is equipped with a range of connection options, including individually switchable mic/line inputs, balanced outputs, video sync, timecode and word clock inputs.
User installable Dante and MADI interface cards are also available as options, adding 24 channels of Dante or MADI i/o to the BBR1MP unit.
Microphones can either be connected to the unit via tails from the rear D-Sub connectors, or via an optional 2RU breakout panel with XLR connectors. System components are available individually or as part of a bundle.
Audio is recorded direct to external USB2/3 drive in broadcast WAV format for instant ingest into post production. Alongside the BBR1MP’s multi-channel recording capabilities, provision is included for creating a simultaneous stereo mixdown. Full support for iXML data is also provided.
The BBR1MP will run off of a 12-volt power source. Unused channels and features can be disabled in order to preserve battery life on location.
The unit is fully controllable via JoeCoRemote for iPad. This includes the facility to set up individual mic pre channel parameters—level, mic/line, phantom power, hi-pass filter, soft limiter and phase reverse. Levels are shown on high-resolution meters and the monitor mix can be adjusted using expandable channel strips with graphic faders, pan, solo, mute and other parameter controls.
“We’re delighted to be releasing our new flagship BBR1MP Recorder,” says JoeCo’s Joe Bull. “Customers have been seeking a more compact complete solution to their multi-track audio recording needs and the BBR1MP will fill this gap in the market admirably. Squeezing this many high-quality mic preamps into such a compact solution has been a challenge but we are thrilled with the results.”
Novation Announces New Audiohub 2x4 Audio Interface/USB Hub
USB hub powers up to three USB devices simultaneously, meaning a single power supply can power an entire setup
Novation has introduced Audiohub 2x4, a combined audio interface and USB hub providing a low-latency stereo input, up to four quality audio outputs (four RCA or two balanced), and a headphone output, all housed in a durable aluminum case
The new Auidohub 2x4 is designed to deliver 96 kHz, 24-bit performance, incorporating “Focusrite sound quality.”
In live applications, connect main loudspeakers and cue sub via high quality balanced and unbalanced outputs, with totally independent level control. All volume and status controls are on the top of the unit to afford easy access, even in the darkest environments.
An integrated USB hub powers up to three USB devices simultaneously, meaning a single power supply can power an entire setup, and connect everything to a computer (or an iPad with a Camera Connection Kit).
Audiohub 2x4 overview:
• Two input/four output audio interface and USB hub
• “Loud” outputs for live performance and DJ applications
• Integrated USB 2.0 hub with three ports
• Bus powered and class compliant
• Ableton Live Lite and 1 GB Loopasters samples included
• Connects to iPad via Camera Connection Kit
Studer Unveils New Vista V Digital Console
Based on same Quad Star technology as Vista X but in a more compact footprint for live productions, smaller studios, and OB trucks
The new Harman’s Studer Vista V is a 52-fader digital console based on the same Quad Star technology as its predecessor the Vista X, but in a more compact footprint for live productions, smaller studios, and OB trucks.
The Vista V includes a built-in meter bridge, high-quality motorized faders, and a built-in Dynamic Automation system with DAW remote control. It’s also fully surround-sound capable, with versatile panning and monitoring functionality.
At the heart of the Vista V is the Infinity Core, which uses CPU-based processors to deliver 800-plus audio channels with high sonic quality, and more than 5,000 inputs and outputs.
The use of CPU-based processors presents possibilities for scaling up to even larger channel counts, and for running third-party algorithms.
Being able to program in high-level languages like C++ speeds up the time of implementing new features, which is not possible when using DSPs and FPGA processing technology.
Infinity Core provides 12 high-capacity A-Link ports (1,536 channels per port) for D23m I/O integration or direct connection into routing systems from Artel, Evertz, Riedel and more. The Vista V also offers easy integration into most AoIP networks used in broadcast like Dante and Livewire, and AES67.
Quad Star technology uses four processors to achieve deep levels of redundancy in the control surface, while CPU-based DSP makes it viable to provide two completely independent DSP cores running in parallel with instant changeover, without a single sample of audio dropout.
The Vista V also comes with VistaMix, Studer’s proprietary automated microphone mixing algorithm, based on gain sharing ideal for simplifying the mixing task at multi-contributor events like game shows, debates etc.. VistaMix removes the need for an operator to manually adjust all the faders all the time, leaving the microphones of talking participants open, while closing the microphones of silent participants in order to reduce spill and background noise. As VistaMix runs directly on the core, no external boxes are needed.
The proprietary Vistonics interface builds Vista V’s rotary controls and buttons directly into the flat screen displays, providing visual feedback exactly where the engineer operates. Each audio function is always associated with the same color, e.g., red for EQ and filters, green for dynamics, and so on.
FaderGlow combines with assignable channel naming to further reduce stress by illuminating each fader in the color relating to the relevant Vistonics function, creating an instant overview of console status. Studer’s new Spill Zone feature enables users to line up a group’s contributing channels with the press of just one button. FaderGlow then identifies their affiliation assigning the appropriate illumination.
The Vista V also has a built-in, high-class loudness meter. The large-scale bargraph meter monitors every channel from mono to surround, plus a history display of up to the last 50 seconds of audio to capture any annoying clicks and overloads that are hard to find in a multi-mic live show.
Built into the Vista V is the BSS DPR-901 dynamic EQ plug-In, assignable to any channel as desired and running directly on the Infinity Core. In addition, up to six Lexicon PCM96 Surround reverb processors can be connected with their parameters stored in the mixer’s snapshots and accessible directly from the Vistonics interface.
The Vista V also allows users to connect up to four Soundcraft Realtime Racks, providing access to Universal Audio’s (UAD) plug-In library assignable on up to 64 insert channels where their parameters are stored and managed directly in the mixer’s CUE-List automation.
Tuesday, September 09, 2014
In The Studio: Microphone Leakage—How It Sounds & How To Control It
Suppose you’re recording a jazz session, close-miking a drum kit and a piano at the same time (Figure 1, below).
When soloing the drum mics, you hear a close, clear sound. But when you mix in the piano mic, that nice, tight drum sound degrades into a distant, muddy sound.
The problem is happening because the drum sound leaked into the piano mic, which picked up a distant drum sound from across the room. It’s as if the piano mic has become a “room mic” for the drum kit.
To keep the recorded sound tight and/or upfront, it’s important to reduce leakage—to increase the isolation between microphones. There are several methods to do it, but how well do they work? Let’s find out with help from several audio samples. The results may be enlightening.
Problems Created By Leakage
Leakage (also called “bleed” or “spill”) is the overlap of an instrument’s sound into another instrument’s microphone. It’s unwanted sound from instruments other than the one at which the mic is aimed. For example, the piano mic also “hears” the drums; the acoustic guitar mic also hears the vocal, and so on.
Figure 1: Example of leakage. The piano mic picks up sound leakage from the drums, which changes the close drum sound to distant.
This makes it difficult to control the mix. Turn up the piano mic, the drums come up as well, and they start to sound “off-mike.” Or, turn up the guitar mic, the vocal also comes up, and it sounds strange due to comb filtering – the phase interference between two mics picking up the same source at different distances.
Another problem caused by leakage is off-axis coloration. While a microphone may have a flat response on-axis (in front), it may have a non-flat response to sounds arriving from behind it. The result can be an ugly, filtered sound when the leakage mics are turned up.
In addition, leakage can create “ghost tracks.” Suppose you have a cymbal track with some electric-guitar leakage on it. Solo the cymbal track, and you hear the guitar solo faintly in the background. Punch in a correction on the guitar-solo track and play back the mix, and you might hear some of the previous solo. That’s a ghost track.
Fortunately, there are many ways to keep leakage under control:
• Mike each instrument closely. That way the sound level at each mic is high. Then you can turn down the mixer gain of each mic, which reduces leakage at the same time. (Don’t mike too close because the tone becomes unnatural). Also, close miking with directional mics creates a bass boost called the proximity effect. When you roll off the lows to compensate, that reduces low-frequency leakage as well.
• Overdub each instrument one at a time. No leakage at all. Total isolation of tracks.
• Record loud instruments first, then quiet instruments. Leakage becomes a problem when you record loud instruments and quiet instruments at the same time. Since the quiet instruments’ mics require a lot of gain, they can pick up serious leakage from their loud neighbors. So you might record the rhythm section first (drums, bass, electric guitars and keyboards), then overdub the acoustic guitar, solos, vocals, etc.
• Record direct. For example, record an acoustic guitar from its pickup during tracking, then overdub the guitar with a mic. Record an electric guitar from its pickup during tracking, then play the guitar track through a guitar-amp plug-in during mixdown. Or record a guitar-amp emulator such as the Line 6 POD. Record a bass direct, and monitor its sound with headphones—don’t use a bass amp.
• Filter out frequencies above and below the range of each instrument. Use a high-pass filter on all tracks (except the kick and bass) to reduce low-frequency leakage.
• Use directional mics (cardioid, etc.) instead of omni mics. To increase isolation, aim the “dead” rear of cardioid mics at nearby instruments that you don’t want to pick up. An omnidirectional mic lacks this discrimination. To compensate, you have to mike closer with an omni. Clip-on omni mics can have very good isolation because they’re so close to their instruments.
• Record in a large, fairly dead studio. In such a room, leakage reflected from the walls is weak.
• Put portable walls (gobos) between instruments or use an isolation booth. A massive gobo has transmission loss: it reduces the energy of sound passing through it, which softens any leakage. Low frequencies still diffract around the top of the gobo. The taller the gobo, the lower the frequency at which it attenuates sound. Under each gobo, avoid openings because they leak sound easily. You can use a piano lid as a gobo: aim the open end away from the drums, or close the lid and cover it with moving blankets.
• Use noise gates on drum tracks. That way, leakage occurs only for a short time when a drum is struck.
Sometimes a little leakage is a good thing. It can add a sense of “air” or ambience, giving a live feel to recordings of a big band, for instance.
We can hear the effects of leakage in a controlled experiment. I set up a drum kit in my studio and miked it overhead with a cardioid, large-diaphragm condenser mic at chin height. I’ll call that the “drum mic.” I also placed a Shure SM57 several feet away, aiming away from the kit, as if it were picking up another instrument. I’ll call that the “leakage mic.”
Figure 2. below, shows the overhead drum mic, plus the leakage mic at 5 feet from the drum mic. In another test I placed the leakage mic 10 feet from the drum mic. The drum mic and leakage mic were recorded on separate tracks and mixed in various ways to create the audio samples you’ll be hearing.
Depending on studio acoustics and mic choice, your results may vary.
The Sound Of Leakage
Here’s the drum mic alone: Listen
Now, the leakage mic alone, 5 feet from the drum mic: Listen
You can hear how the leakage mic sounds muddy and distant compared to the close-up drum mic. That’s because it’s picking up a lot of early reflections from the studio surfaces.
Let’s mix the drum mic and leakage mic that is located 5 feet away: Listen
Figure 2: Leakage test setup.
The mix sounds more distant than the drum mic alone. Play the drum mic alone to hear the difference.
Effect Of Mic Separation
What happens if the leakage mic is 10 feet away instead of 5 feet? Let’s compare.
Here’s the drum mic alone: Listen
Here’s the drum mic mixed with the leakage mic 10 feet away: Listen
Moving the leakage mic farther from the drums did not make a big difference (in this studio at least). That means we can place performers fairly close together and still get a good recording. Sometimes it provides a tighter sound than if the players are far apart, because the leakage has a shorter delay.
Effect Of Close Miking
Now let’s listen to the effect of moving the leakage mic closer to its instrument. If we reduce the miking distance by half, that increases isolation by 6 dB because we can turn down the mic’s signal by 6 dB.
Here’s the drum mic alone: Listen
Here’s the drum mic mixed with the leakage mic 5 feet away: Listen
And here’s the drum mic mixed with the leakage mic turned down 6 dB: Listen
As you can hear, moving the leakage mic closer to its source (and turning down its level) makes the drum sound tighter or drier. In other words, we can make the drums sound tighter by miking everything else more closely.
Effect Of Signal Alignment
Note that the leakage mic’s signal is delayed compared the drum mic’s signal due to the travel time of sound through the air from source to mic.
As an experiment, I aligned the leakage-mic signal with the drum-mic signal. That is, I slid the leakage mic’s track earlier in time so that the drum mic and leakage mic were heard exactly together. Let’s compare:
Here’s the drum mic mixed with the leakage mic without aligning their signals: Listen
Now here’s the drum mic mixed with the leakage mic with their signals aligned: Listen
It didn’t make much of an improvement, if any. Although the two tracks are aligned in time, the leakage mic is still picking up wall sound reflections which make the perceived sound distant.
Signal alignment can work well on multiple drum tracks because the signals are relatively dry.
For example, suppose you’re using two overhead mics on the kit. Place them equidistant from the snare to produce a more coherent, centered image of the snare drum. Or slide the cymbal tracks earlier in time so their snare leakage aligns with the snare-track hits.
Figure 3: Test setup with a gobo in place.
Effect Of A Gobo
Next I set up a 4-foot high, 8-foot wide padded plywood gobo in front of the drum kit (Figure 3). Although low frequencies can diffract around the top of a gobo, the mids and highs are attenuated, reducing leakage.
Here’s the drum mic alone: Listen
Here’s the drum mic mixed with the leakage mic, and with the gobo in place: Listen
The gobo does a good job of isolating the drums, at least at mid-to-high frequencies. Figure 4 shows the frequency response (sound attenuation vs. frequency) of a 4-foot high gobo.
The Sound Of Leakage On Cymbals
I also recorded a cymbal using the drum mic and the leakage mic.
Figure 4: Attenuation vs. frequency of a 4-foot high gobo (approximate).
Here’s the cymbal picked up by just the drum mic: Listen
Here’s the leakage mic 5 feet away: Listen
Here’s the drum mic mixed with the leakage mic at 5 feet: Listen
Here’s the drum mic mixed with the leakage mic at 10 feet: Listen
As you can hear, the leakage mic 5 feet away colors the tone of the cymbal, but the leakage mic 10 feet away has less coloration. That’s because the highs diminish with distance in a room.
I hope that these audio examples have demonstrated the effects of leakage, and demonstrated the effects of various methods to control it. Good luck in your quest to tame leakage.
Bruce Bartlett is a recording engineer, audio journalist and microphone designer (www.bartlettaudio.com). His latest books are Practical Recording Techniques 6th Edition and Recording Music on Location 2nd Edition.
Microsoft Production Studios Rolls Out Audinate Dante’s Audio Over IP Network
Networking approach has transformed the audio possibilities in HD and 4K production operations
The largest full-service HD and 4K production facility in the Pacific Northwest, Microsoft Production Studios includes three sound stages, three audio rooms and numerous editing suites among its 65,000 square feet located on the Microsoft headquarters campus.
The impressive facility is well-equipped to accommodate multiple in-house projects while providing rental space to outside productions. Thousands of productions and events are created in the studios each year.
Audio is an integral part of the overall workflow at Microsoft Production Studios. In mid-2013, facility’s engineering team began to investigate how an audio over IP networking solution could transform the possibilities in their production operations.
After consulting with Audinate’s team in Portland, the MPS team recognized they could deliver multi-channel across their existing Cisco network switch fabric, while providing greater flexibility with the Dante networking solution.
“The transition to a Dante audio over IP environment just keeps generating steam, and I don’t see any end in sight,” says John L. Ball, systems engineer, Microsoft Production Studios. “It’s turned into a situation where it’s about who can think of the next cool idea on how Dante can transform and simplify our production while maintaining tight synchronization, and ultra-low latency to reduce the time and complexity for setting up the gear and sessions.
“We’re working toward a total networked environment for audio, communications and eventually video. Dante is core to making this easy by making every step a significant timesaver in setup, integration and network programming.”
Dante was initially installed to support in-store audio experience at Microsoft retail stores. Ball uses the software based Dante Virtual Soundcard to stream audio to a Symetrix Edge router, enabling playout of unique audio channels from PCs across multiple retail zones.
MPS then upgraded its RTS ADAM intercom systems, which are integrated with Dante. This has simplified the ease of use of the intercom communications across the greater Microsoft campus, ensuring production teams can reliably move audio between the studio and any remote location.
Microsoft expanded the system with a multi-vendor interoperable Dante solution including RTS Intercom, Shure ULX-D microphones and Yamaha CL5 mixing consoles. The system was then extended by integrating Focusrite RedNet 6 MADI bridge units to an existing Grass Valley Miranda audio/video router.
The newest network expansions incorporate Skype feeds and support multi-lingual feeds. Ball recently purchased Dante-enabled Studio Technologies Model 215 Announcer’s Consoles to control and distribute live programs in multiple languages simultaneously. His first project with the Model 215 units supported an Xbox Live broadcast from a gaming event in Germany, monitoring feeds over the Dante network and streaming the program in English, German, Italian, French and two Spanish dialects.
“The cool thing about this application was using the actual video edit rooms as voiceover rooms,” he says. “We just connected the Studio Technologies gear and the Shure microphones right to the Dante network in each edit room. We used to have to run analog cables and connections all over the floor between multiple rooms to support this kind of event before. But now, using Dante for this configuration, it takes less than 30 minutes and a network connection to be off and running. We save at least a day and a half of work in setup time alone.”
The MPS team continues to discover new creative use cases to take advantage of Dante’s integration into PCs. With the Dante Virtual Soundcard, they broadcast on-air interviews via Microsoft’s product called SkypeTX: Microsoft Production Studio’s network two-way communications with Skype users, and translate those feeds into the broadcast and production world. By replacing analog audio I/O systems with Dante Virtual Soundcards, this has reduced infrastructure and complexity for both live and taped productions across broadcast and corporate applications.
“The original SkypeTX box required two cables in and out to bring it into the intercom environment, and then we had to transport it to audio control and convert it to digital just to get the broadcast signal out,” Ball explains. “To enable talk-back, we’d have to convert it back to analog and basically reverse the entire workflow. With Dante, I can connect the audio and intercom in all IP digital, with no analog degradation. We simply capture the audio from SkypeTX, put that on the Dante network and feed our audio rooms for broadcast and production purposes. Dante is enabling a more modern means of communication with exceptional ease of use.”
Ball is now in the design phase of complete Dante network upgrades for all audio control rooms in the facility. Moving forward, he ultimately expects to upgrade their network infrastructure to a 10Gig pipe to enable the existing virtual LANs, or VLANs, to essentially to extend the Dante network and its production capabilities far beyond Microsoft’s Redmond campus.