• Amp Modeller ... lounge
  Some of us (Guitar Players) will never give up our tubes but you can get a good overview of the several of the alternatives in ASAT's lounge.
  
• Get your Line6 patches at vettaville.nl
  The people-at-Bose have created several patches for the special PS-1 output mode of the Line6 gear. This output mode was designed for the Bose System.
  

• Orient the microphone so that it is not pointing directly at your Cylindrical Radiator™ loudspeaker. It’s best when your system is not directly behind you- give greater than 5 feet distance between you and your system.
  
• Try using a “close-miking” technique to increase the gain at the microphone and to reduce unwanted stage sound.
  
• Use the HIGH tone control knob for the microphone channel carefully. Feedback could occur when this is set too high.
  
• Upgrade your power stand to newer software that offer a high gain preset. see: Software Upgrading (This covers firmware update and Version 2.0 Presets)
  
• If using a vocal effects processor, make sure that it is not contributing to the feedback problem.
  
• If stand -mounting a directional microphone, tilt the microphone up ten or twenty degrees off the horizon so that it is less sensitive to direct sound from the speakers.
  
• Try a different microphone such as a directional mic. Hypercardioid microphones work well with the system.

• All players should be playing / singing through the Bose System closest to them
  
• Every doubling of distance between the microphone and the singer reduces gain-before-feedback by a factor of 4. The difference between 1/2 inch and an inch can be significant.
  
• Wherever possible "acoustic" instruments should use pickups instead of microphones.
  This is important because many players find it a struggle to keep a consistant and close distance between the microphone and their instrument. Also, an instrument can be a source of feedback as it resonates with the amplified sound.
  
• Another (low prority) design guide-line could be to keep open mics as far apart as possible. Neighboring systems with open mics can mututally decrease gain-before-feedback
  

quote:

Originally posted by ST:
To be prepared for people who just want to walk in and play the old way... Have a conventional mixer wired up with mains out running to your two systems (channels 3/4).

quote:

Originally posted by ST:
Terminology
Like many others, this community is developing its own lexicon. Look for a few terms and quick references here.

Terminology

quote:

Originally posted by Cliff-at-Bose: in Out of Phase Overheads
These are not "overheads" but 57's placed below the cymbals and above the top heads of a small kit. The mics are wired out of phase, basically 2 and 3 reversed on an XLR. There's lots written about this on our forum. But the basic idea is that the pair will cancel onstage bass but pick up local instruments on the kit, in an effort to reduce rebroadcasting onstage bass from the drum mics. This way we get a stronger signal than typical overheads and cleaner bass onstage. To "mix" these, just move the mics closer to the instrument you want to emphasize. Typical setup is on either side of the snare facing the snare. Snare is everywhere and there is no need to put a mic directly on it. It's actually pretty easy to set up and adjust, moving the mics around for a good kit mix. With 3 mics and only 2 inputs on the PS1, we get a really nice amplified sound for our shows. I listened to the output of the pair recently (multitrack recordings of a recent show) and the cancelling thing really works great. Give it a try.

quote:

Originally posted by Hilmar-at-Bose:
Here's the nerds view
Wiring the the two mics out of phase creates essentially a "dipole". Everything that is in the middle (i.e. equal distance) between the two microphones will get equally but out-of-phase so it cancels when the two microphone signals are summed together. In essence it creates a "blind spot" for the microphones for whatever is right in the middle plane. For the drums, that's mainly the kick (as Larry pointed out) and also the drummer (when he/she is hemming and hawing, squeaking with the chair, yelling about or in general having a grand old time).
Another nice trick is to place the L1 that gets the mic signal somewhere the middle plane of the microphones. This drastically reduces potential for feedback and unwanted regeneration.
Sound sources that are significantly closer to any one of the microphones are not much affected by the whole procedure.

Hilmar

---

Unofficial Users' Guide	Need a sketch?		Bose L1® Wiki
Unofficial Performers' Guide	How long have you been gigging?		My Bose Blog
Search the wiki

quote:

Originally posted by Hilmar-at-Bose:
Oh dear, I just stumbled across this thread. We are currently working full throttle to get the A1 on the store shelves and the last few weeks in a product development cycle are always a little hectic.

I’m certainly happy to take on all questions and comments so far, but I may have to do it in multiple installs. Feel free to ping me if I’m slacking off (or if I bore everyone out of their mind).

Let’s start with some technical stuff (philosophy will be in the next installment)

1) Crossover
If there is no B1 and nothing connected to the Bass Line Out. The L1 sees frequencies from 110Hz up. Feeding it anything lower, doesn’t make sense, since it couldn’t produce any acoustic output and if would rip the drivers to shreds.

In any other case the L1 sees signals only from 180Hz up. There is no other variation in frequency or gain for the L1 no matter what else happens

2) Bass Line Out and B1 behavior
This is based on the design goal that “You should always sound the same; no matter how much Bass stuff is attached” I can try to explain my view of why this is a good design goal (of which you may disagree) but let’s look at the actual behavior first.

Without Bass Line out
1xB1: 40Hz-180Hz, B1 specific EQ, some nominal gain that we call 0dB
2xB1: 40Hz-180Hz, B1 specific EQ, -6dB as compared to nominal

With Bass Line Out
0xB1: 40-180 Hz, flat, roughly the same gain as 2 B1
1xB1: 40Hz-180Hz, B1 specific EQ, -6dB as compared to nominal
2xB1: 40Hz-180Hz, B1 specific EQ, -12dB as compared to nominal

What this complex behavior does is the following. No matter if you attach 1, 2, or 4 B1s, you will get pretty much the same balance between all combined B1s and the L1s. It’s a little off for 3, 5, 6, 7 & 8 B1s, but still reasonably close.

3) Frequency content of an acoustic guitar
Oldghm, you did some really interesting experiments there. However, you have to be really careful when using an RTA. You can feed these things a pure sine wave at 80 Hz and by turning it up make the 63 Hz and even the 40Hz LED light up. They will be lower than the 80 Hz LED, but still come on. That does NOT mean, that the sine wave contains any other frequency than 80 Hz (it certainly doesn’t). It only means that the RTA has a pretty limited frequency resolution. The 63 Hz LED will respond best to 63 Hz signal but it’s in no way “blind” to 80 Hz signal.
Thus being said, the actual frequency content is not easy to determine. All sounds that have a pitch are certainly constraint to 80 Hz and up (in standard tuning) and there isn’t actually too much energy at the fundamental. However, the “non-pitched” sounds like a hard string attack or whacking the top with your hand can very well have lower frequencies. Unfortunately, I don’t have any hard data on that, but we will measure that at some point.

4) Equal loudness curves
Here is the bunch
http://hyperphysics.phy-astr.gsu.edu/hbase/sound/eqloud.html
These curves tell us two things:
First, the same physical sound energy produces different perceived loudness depending on frequency. You can turn that around into “The physical sound energy required to produce the same perceived loudness varies with frequency”.
Second, this frequency dependency is a function of overall level.

The first statement is not particularly bothersome. Your auditory system is well calibrated to that. A voice sounds normal because it sounds like what you are used to, not because it has “constant sound energy” or “constant perceived loudness” with frequency.

The second statement is much more trouble. It basically says that if you amplify an acoustic source (even if you do it perfectly), the perceived spectral balance will change. This is a well known effect, and most of our home entertainment systems have actually and “automatic loudness compensation” that changes the system voicing with overall level. We actually contemplated adding this to the Personalized Amplification System™ but after some soul searching we thought it would be too intrusive on the musician.
The main corrections are at very low levels, and in most practical live music settings, the effect is pretty minor.
As a rule-of-thumb guideline, turn the bass up a notch as you turn the volume down.

5) L1 versus B1 fall-off with distance
As many have observed, only the L1 qualifies as a Cylindrical Radiator™ loudspeaker, the B1 certainly doesn’t as it looks a lot more like “cubical” radiator. Only a cylindrical source will display 3 dB per distance-doubling falloff. The B1 is a conventional speaker and falls off with 6 dB per distance-doubling. Does that mean that the spectrum gets unbalanced with distance, i.e. not enough bass as we move away from the source?

Not really, and here is why: The observation of so-and-so dB per distance doubling is only true in “free field”, i.e. in some imaginary space that doesn’t have any reflective surfaces. Such a thing doesn’t exist. Most places where you play generate lots and lots of reflections. At any point in the room, the sound field consists of two components: 1) The sound that comes directly from the source aptly called “direct sound” and 2) all the sound that comes bouncing back from the walls, called “reverberant field”.

The level of the reverberant field tends to be roughly the same everywhere in the room. When you are close to the source, the “direct sound” dominates. As you move away from the source, the direct sound drops in level and at some point called “critical distance” the direct sound has the same level as the reverberant field. From this point on the reverberant field dominates and the sound level remains pretty much constant no matter how much further you move away.

The level of the “reverberant field”, the “critical distance”, and the “reverb time” are all close room acoustical cousins and basically determined by the geometry and amount of absorption of the room. In nearly all rooms, there is more absorption at high frequencies and less absorption at low frequencies. Less absorption makes the reverb time longer, the reverberant field level higher, and the critical distance shorter.

Another factor that influences the critical distance is the directivity of the sound source. Let’s make a thought experiment: Imagine a sound source that radiates “normally” to the front but nothing to the back. The direct sound level doesn’t change, but the reverberant level drops by 3 dB since the total energy radiated into the room has dropped by half. That means the critical distance has increased. Of course, that’s only true if you stand in front of the source. In the rear, there is no direct sound any more and the critical distance has become zero. We see that the directionality of the sound source increases the critical distance within it’s beam, cone, pie slice (or whatever shape it radiates), but decreases the critical distance outside.

Now what has all that to do with our initial L1/B1 problem? As it turns out the L1 has a much higher critical distance than the B1. That has two reasons: First, there is more absorption at higher frequencies. Second, the L1 is highly directional: it doesn’t radiate up or down.

Taking all this together we see roughly the following picture: The L1 has a fairly large critical distance, i.e. it’s mostly direct sound and that falls off with ca. 3 dB per distance doubling. The B1 has a short critical distance (not directional, low room absorption) so the behavior becomes quickly a mix of reverberant field and direct sound which tends to also look fairly similar to a 3 dB per distance-doubling fall-off over a good stretch.

To add insult to injury, this is all grossly simplified. In actual rooms, the reverberant field is never really constant, room modes get in the way, the L1 behaves not quite cylindrical in the lower mids, bass levels increase in the vicinity of wall, etc.

If it’s any consolation, we have actually measured the fall off versus distance for our combined system (L1 & B1) in a couple of different rooms and the 3dB per distance- doubling describes the measured data remarkably well.

If anyone is still awake after this lengthy lecture (apologies), I’ll try to tackle the “philosophical” stuff next.

Hilmar

quote:

Originally posted by Kyle-at-Bose:
Hi Greg/Jerry. Some random things to consider when upgrading your system by the download-burn CD method.

• Burn the downloaded file as a "audio" CD, not a "data" CD
  
• Make sure that your burn software has all processing disabled. Some software applies normalization. Check the options to make sure that this is OFF.
  
• When you burn the CD, I actually prefer to burn 3 tracks of the same file in a row. So, in other words, tracks 1, 2, and 3 would be the same file.
  
• On your DVD player you MUST TURN OFF ALL DSP PROCESSING like dynamic range compression, 3D effects, speaker size selection
  
• If you are using a portable DVD player with a volume control, make sure that the volume control is at MAX.
  
• On your DVD player, try BITSTREAM MODE first. If that doesn't work, we've seen PCM MODE work also.
  
• When flashing the PS1 using the DVD player, it's sometimes good to start with the DVD player at STOP. Turn the PS1 off, then back on. Next, press play on the DVD player to start the upgrade process. Be patient, it sometimes takes 10 seconds or so to start the blinking.

I hope these suggestions help!

---

Unofficial Users' Guide	Need a sketch?		Bose L1® Wiki
Unofficial Performers' Guide	How long have you been gigging?		My Bose Blog
Search the wiki

quote:

The system will only recognize the remote if it's connected when the system is powered up. On the other hand, if you set and then disconnect the remote after the PS1 is powered up, the system will retain those settings until the power is turned off.