ponedeljek, 6. september 2010

Yamaha LS-9 / 32 ch.

Since releasing the PM1D digital console eight years ago, Yamaha's design engineers have become quite skilled at migrating advanced digital mixing technology to lower price points. On the heels of the PM1D came the hugely successful PM5D. More recently, Yamaha introduced the M7CL and the subject of this review, the LS9. Available in 16- or 32-channel configurations, the LS9 packs serious DSP power into a compact desk that can be easily transported by a single person.

QUICK AND INTUITIVE

My LS9-32 arrived literally hours before a series of local gigs with Wreckords Records artist Closenuf — just enough time to unpack it and give it a spin. Out of the box, the LS9 has I/O only on balanced XLRs: 32 inputs and 16 omni outputs. The input XLRs are self-explanatory. The output XLRs can be configured to serve as aux, group, matrix outs and main outs. At first sight, this was intimidating — I had visions of digging through menus just to learn how to assign the L/R master bus to two of these outs — but when I noticed that outs 15 and 16 were also labeled “L” and “R,” I opted to leave the manual in the box and go for the snoop approach: Turn it on and see what happens.

Within approximately 10 minutes of powering up, I was able to successfully do all of the following: connect the L/R master outs to a power amp, a Lab.Gruppen FP+10000; assign mix 1 to omni output 1 and mix 2 to omni output 2 for discrete monitor mixes; and assign rack 1 (a 31-band graphic EQ) to mix 1 and rack 2 (another 31-band graphic EQ) to mix 2 for monitor EQ. Within the same 10 minutes, I called up a great-sounding stereo reverb patch from the onboard effects library for use on send 13; recalled a mono delay from the library for use on send 14; edited that delay, named and stored it; tested all the inputs using a CD player; stored and named the console scene; and then left for the gig. The ability to do all this without cracking the manual is a tribute to Yamaha's operating system and the LS9's intuitive layout.

LEAVE THE RACK at HOME

Connecting a TRS line-level source to the LS9 requires either a direct box (in the case of the CD player, two Countryman Type 85s) or TRS-to-XLR adapters. For those in need of TRS I/O, the LS9's rear panel has two mini-YGDAI expansion slots (the LS9-16 has one), which accept cards for additional analog I/O, as well as ADAT, TDIF or AES/EBU digital I/O, and CobraNet. I was initially concerned that I'd have a cable problem at the gig, but then I remembered that I wouldn't need to bring my processing rack. The LS9's onboard DSP provides compression, gating, 4-band EQ and highpass filter on every channel, plus a total of eight patchable effect “racks” that are user-assignable.

Each channel also features a 100mm motorized fader, 7-segment LED meter, and buttons for channel on, SEL and cue. The LS9-32 provides control over a maximum of 64 inputs on two 32-channel layers (32 inputs via the XLR ins and 32 more via expansion cards). Four dedicated buttons let you choose between the two input layers — a “master” layer and a custom layer — where you can mix and match any combination of input, output and matrix channels or mix masters. For example, if you want the star channel, two aux masters and a matrix master on one layer, you can have that.

NO TRS? NO PROBLEM!

Setup at the gig was a breeze. All I needed for patching was four XLR cables: two for the L/R bus and one each for the two monitor mixes. The band had a Shure wireless system, a bunch of wired mics and Dis for keys and bass, so the fact that I had no TRS I/O was no problem. Gain for the head amp is digitally controlled across a range that encompasses line- through mic level, negating the need for a mic/line switch. When you store a console scene, the gain is also stored. Occasionally, I could hear “stepping” while adjusting head-amp gain, which is not unheard of in digitally controlled analog preamps.

During the first show, I had no problem getting around the LS9. While mixing, I was able to quickly find the Preferences page where I could set the desk to auto-select, so that by touching a channel's fader or button I could automatically select it for editing using the dedicated channel controls. These controls are found next to the color LCD and include head-amp gain, pan, selected send, dynamics 1 (threshold) and dynamics 2 (threshold), and a set of EQ controls. You can also view the desk's parameters using the increment/decrement buttons, cursor and data wheel, all of which were easy to read, even in a club's dim lighting.

To the left of the LCD are a series of lit push-buttons divided into two groups. One group provides access to mixes 1 through 16, while the other set is for global controls such as scene memory, monitor, setup, channel job, recorder, meter and racks 1 through 4 or 5 through 8. Thanks to these hot buttons, navigating the LS9 is very rapid. Once I had the mix up during the second show, I was able to name channels (kick, snare, etc.) and assign icons to each channel — the band really got a kick out of that. Ringing out the monitor wedges using the 31-band graphic EQs was facilitated by switching the EQ to Fader Assign mode, in which the faders function like the sliders on a graphic EQ — very slick and way more intuitive than using scrolling cursors and rotaries. Directly underneath the LCD are two extremely important controls labeled Cue Clear and Home. Home is invaluable: Any time I was into a deep menu, all I had to do was hit Home and the selected channel reappeared on the LCD. Cue Clear does exactly as it name implies, and is a control that every console needs.

Pressing any Mix button once tells the LS9 which send you want to address using the Selected Send rotary in the selected channel section. (This procedure isn't as convoluted as it sounds.) If you press the Mix button a second time, then the desk enters Sends on Faders mode, in which the faders control send level. While this is happening, the LCD blinks “Sends on Fader” to remind you that the faders are changing aux levels, not level to the L/R bus. Because I had two monitor mixes running, I could toggle between the two and clearly see what faders were routed to which mix and then hit Home to jump back to the house mix. Monitor engineers will love this feature.

PLENTY OF DSP

As I expected, the LS9-32 is loaded with DSP, but unlike some of its competitors, the channels on the secondary layer (channels 33 to 64) are not crippled; you get exactly the same EQ, routing and dynamics capabilities on all 64 channels. Channel EQ is capable of subtle changes or serious “surgical” manipulation. When you are not on the Home screen, moving any of the selected channel knobs causes a pop-up window to appear that shows the value of that parameter. I found that when adjusting EQ, I could summon a shortcut to the EQ display that let me view the curve I was carving: Quickly press one of the EQ controls and then press Enter. (This works the same way for the threshold controls and the dynamics screen.) This works much faster during a show than using a cursor to navigate the screen to gain access to a particular processing section.

The dynamics capabilities should accommodate any application you might require. Dynamics 1 offers a choice of gate, ducking, compressor or expander, while dynamics 2 offers compressor, compander (hard), compander (soft) or de-esser. Compression can be set to run the range from subtle to completely squashing the audio signal. In one situation, when I mixed a dance-music diva with serious pipes, I used the EQ to pull out a bit of 3.5 kHz, as well as heavy compression to tame her peaks, which resulted in a smooth, musical vocal sound.

EXTRA, EXTRA

One of the LS9-32's bonuses is a built-in USB recording/playback device. The USB port accepts a standard memory stick (up to 2 GB) and allows you to record the desk's output to 96, 128 or 196kbps MP3 files. You can record audio from any bus on the console and route the player's output either to a physical output or internally to two channels. The USB recorder can also play MP3s that are stored on the stick. MP3s may be linked to scenes so that when a scene is recalled, a specific MP3 plays automatically. (I admit that I had to read the manual to learn how to do this.) This feature will be a boon in theater applications.

Other features include assignable talkback, 12 user-defined function keys (I used two to bypass my main stereo effects) and the ability to link channels in nonstandard pairing. At one show, drum overheads were patched to channels 8 and 9. The LS9 can pair those channels, whereas many digital desks can pair either 7/8 or 9/10. Along with the 31-band graphic EQ, the LS9's Flex15GEQ provides two linkable channels, where up to 15 of the traditional 31 graphic bands can be active, which is useful when mixing IEMs or stereo wedges.

A CLEAR WINNER

One of the LS9's most important features is its sound — excellent. The audio paths are clean and have enough headroom to handle hot inputs, despite the absence of a dedicated mic/line switch, and the mix buses are as quiet as the day is long. In addition to the effects mentioned, you get chorus, flange, echo, tremolo and pitch shift of the Yamaha SPX nature that we all know and love. After using the LS9-32 for several weeks, it became apparent that this console is intuitive enough for less-experienced engineers and deep enough to perform some very serious routing assignments, including discrete L/C/R panning. Equally important, the operating system is rock-solid.

For many local gigs, I wrapped the LS9-32 in a blanket and transported it in the hatch of my Volkswagen Golf. It fit perfectly, and I was able to carry it solo. The power supply is built-in, as well as everything else you need for the “front end” of a very impressive P.A. system.

Basix about sound engineering

Live Public Address (Live Live P.A.) systems come in many different shapes and sizes and can often confuse the newbie into not knowing even just the basics. This article is aimed at giving a basic overview of non-specific equipment configurations in an attempt to de-mystify some of the typical errors a newbie can make with Live P.A. systems.

The function of a Live P.A.

Two types of Live P.A.

Reinforcement

  • This is for speech or music which would sound good in a small room without artificial assistance.
  • In the case of a classical guitar which is a very quiet instrument, this natural sound is only good for an audience of around 200 - 300 depending on room size.
  • For an audience of 500 - 600 it is possible to reinforce the sound so that everyone can hear clearly and to most people it will still sound natural.

Amplification

This is where the original sound is insignificant in comparison with the amount of sound coming from the Live P.A.

The aims of public address:

1.) To provide adequate volume ( not necessarily loud ).

2.) To provide adequate clarity.

A Thought on Acoustics

A discussion on sound reinforcement is impossible without a mention of acoustics.

The free field

If the venue is an outdoor event then the engineer need not concern himself/herself a great deal with acoustics as this is the ideal situation.

Sound in the open air travels away from the source and keeps going until it's energy is used up ( inverse square law ). There are no walls for the sound to bounce off and return to interfere with the next wavefront.

Indoor Live P.A.

Sound behaves in much the same way as any other wave, it bounces off walls (reflects), and bends around them (diffracts), and cannot pass directly through materials. Therefore speaker placement becomes important as does speaker coverage.

Consider Figure 1.0.

The sound waves being ommited from the cabs have a coverage of 120 degrees therefore it can be seen that there will be obvious 'blind spots' in the coverage.

Figure 1.0 Shows a small venue and a typical coverage angle of a driver.

It should be pointed out that the directionality of sound waves is somewhat frequency dependent and that the above diagram shows a potential problem for high frequencies.

High frequencies have a smaller wavelength than low frequencies hence they are very directional ( λ = v/ƒ ), λ = wavelength, v = velocity, ƒ = frequency .

Objects placed in the Live path of HF block them, whereas LF tend to bend (diffract) around them. Therefore a subject positioned behind the wall in Figure 1.0. will hear an attenuation in HF hence a dull sound.

Golden rule number 1.

Always ensure nothing is in the line of sight of a HF driver otherwise you may encounter loss of HF.

Reflection and phase cancellation (comb filtering )

Consider Figure 1.1.

Live Live P.A.

Figure 1.1 Shows the Live path of a sound wave, the venue is assumed to have reflective surfaces and after a period of time the wave can be seen to have travelled around the room bouncing off the surfaces and crossing over other sound waves.

This situation can create what is know as 'comb filtering'. As you can imagine the sound takes time to travel around the room (340 metres/sec) and if the reflected wave (having being time delayed) coincides with another wave whose polarity is the inverse or a fraction of, then cancellation will occur.

Conversely, if the combination of merging waves have the same polarity then addition will take place.

The name comb filtering is adopted because looking at the frequency response of the product, the shape of the teeth on a comb can be seen. Showing areas of addition and subtraction.

Live Live P.A.

Figure 1.2 The peaks and troughs of comb filtering can be seen in this frequency response plot.

Golden rule number two.

Ensure the minimum amount of reflection by pointing the speakers in a suitable direction.

The main thing is to keep comb filtering to a minimum this is sometimes easier said than done as most venues have reflective surfaces. There will always be pockets of 'bad sound' and pockets of 'good sound'.

If you wonder round a venue and listen to the mix you will find these spots, it is your job as an engineer to keep the 'bad spots' to a minimum through speaker positioning, coverage and equalisation.

In professional venues architectural acousticians get paid lots of money to design environments which produce 'good spots' throughout the venue by such methods as absorption paneling.

Standing waves

Standing waves are a result of sound being reflected back and forth between two parallel surfaces.

As the first wave reflects it meets a newly arriving wave and the result can be that a stationary wave is produced which resonates at a frequency dependant on the transmitted waves and the distance between the Live parallel surfaces.

The wavelength of the transmitted waves in relation to the distances between the Live parallel surfaces is important for consideration then.

If this distance equals the wavelength or a ratio of it then a standing wave could be be made to oscillate.

Example

The wavelength of a 20 Hz wave is 17 metres, if this wave was transmitted between two Live parallel surfaces whose distance was 17 metres an oscillation could occur.

Standing waves can be a problem in venues where the dimensions of the venue coincide with Live particular wavelengths.

At LF, standing waves can 'creep up' on the engineer as they gather energy and appear to 'feedback', which could of course occur if the standing wave was picked up by the microphones on stage and amplified.

Careful use of room equalisation and speaker positioning can combat standing waves to a degree.

What's wrong with a lot of Live P.A.'s

  • Low efficiency speaker systems
  • cure - ensure you have efficient speakers.
  • Not enough amplifier power
  • cure - ensure you have plenty of amp power
  • Poor frequency response
  • cure - ensure all components in the chain have a 'flat' response
  • Miss half your audience
  • cure - ensure you have enough speakers that are angled to cover everyone
  • Room reverberation swamps the sound
  • cure - choose speakers with suitable directional and dispersion qualities, thus avoiding reflective surfaces

Basic systems for two different sized rooms

Example 1

A small sized room having the dimensions of around 30 by 30 by 10 feet.

Live Live P.A.

Live Live P.A.

Figure 1.3

The system in block diagram form.

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Figure 1.4

This would be a suitable setup giving adequate coverage.

The power amp would be rated at around 150/200 watts per channel and the speakers would be full range.

Example two

A medium sized room having the dimensions of around 50 by 40 by 15 feet.

Live Live P.A.

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Figure 1.5

Live Live P.A.

Figure 1.6

This system is known as a two-way system, and for this room a total power of around 1KHZ would be adequate. The audio spectrum is split in two at around 250 Hz. Thus two power amplifiers are nessessary. Percentage wise the Low end would have around 65%. Leaving a further 35% for the mid and top end.

Large Live P.A.

Large Live P.A. systems can often be anything from two-way to five way and can contain massive amounts of drive untis for each seperate bandwidth.

A large Live P.A. system would have two engineers. One for the front of house one for the monitirs mix.

Often delayed loudspeakers are needed.

In a large open air concert say, a person standing 340 meters from the stage will not hear the emitted sound wave until one second has elapsed if he/she is standing 640 meters then two seconds will have elapsed before the sound can be heard.

This is due to the speed with wich sound travels through the air. i.e. 340m/s.

By the time the sound has travelled this distance it has suffered great losses. Therefore, further speakers will be needed for the audience to the rear of a concert.

The sound from these drivers need to delayed in order for the sound emitted from the main drivers to be of the same phase.

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Figure 1.7 The basic configuration of delayed loudspeakers.

The delayed signal could come from groups, from the main mix or aux's etc.

Further points to note:

A rock Live P.A. should be as intelligable as a West end musical.

It should cover the audience evenly allthough this is not always possible.

The system should be visibly in tune with the type of work and surroundings.