Posts by 2004Billy

    i can’t imagine a situation where a 1x12 isn’t loud enough. If its too loud for a 1x1w then your hearing is getting seriously damaged even with ear protection. For 30 years I used a Dual Rectifier with either one or 2 Mesa EVM12L Thiele cabs. One was plenty loud enough but two aloud a better spread for big stages. I’ve always hated 412 cabs anyway but a 1x12 can be devastatingly loud.

    :)The Thiele EVM12L with 110 dB/W in the mid range is the loudest 112 cabinet around, so I am not surprised. It's small but I don't like the weight (20kg).

    Finally, measured, in idle, 220v, the headrush consumes around 3.8A of current, and while playin and mid volume goes to 4.3A, that means the headrush uses around 750W+ of nominal real power, so there you have it, 2000W looks like marketing, at most this thing would be a 1200W peak speaker, but it indeed sounds like a louder one. Still, no luck for your pedalboard :/ wont make it with 2A

    I think this measurement is flawed. Not at all likely that a class D amplifier in idle draws 750W.

    I bought my stage directly from them. And also my remote. I rather prefer Kemper to put the money into development versus giving a portion to Thomann. I had a very positive customer service experience: fast turnaround on first stage, the 2nd stage appears stable. They also repaired (free of charge) the LEDS on my good old toaster.

    My stage was fine. I used it every week for a few months. No clock setup error.

    Now I return after 10 days abroad. During two weeks the stage sat unplugged.

    Today at rehearsal I had the clock error. Now at home it happens again.

    When connected by USB to rig manager, the error appears but in the system menu both time and date are correct.

    When not connected to rig manager, time and date are back to 2010/01/01 00:00:00.

    Running version 15295 on stage #2.

    Just go easy on the tweeter of the Headrush. Never let the clip light come on (if I have to believe the TS315 web posts).

    I got interested in the HW details so went out for a web search and dug up a lot of numbers, but not for Alto.

    I can't find any rating for the Alto drivers. It's easier for the other contenders which I both own:

    The DXR10 has a YD659A00 compression driver which appears to be a rebranded CDX1-1445 (1.4" Celestion, PETP membrane)

    The CLR neo 2 has a T5486A compression driver which is another name for a CDX1-1731 (1.75" Celestion, PETP membrane)

    Both are very similar in response (as in very smooth frequency curve), but the 1.75" handles double the power and is s couple dB more efficient, so you get about 6dB more out of it.

    The DXR v2 may have upgraded to the 1731 (don't know, just guessing)


    I also checked on the FRFR power myth - I think I can explain "2000W" of the headrush!


    To do that: first how it works for the DXR10.

    The DXR schematics can be found on the web. I see +/-55V rails. The woofer is differential driven by bridged mono. So it can actually put out 110V over 8 Ohm, for a power of V*V/R = 1500W. This is the type of funny nrs that manufacturers (but not Yamaha) talk about.

    In reality it could make an RMS sine of 110/sqrt(2) = 78V, for a power of 760W. That is the undistorted sinusoidal peak power that could be driven to the speaker. If the actual resistance of the woofer is 6.4 ohms, you'd end up at 950W (which is claimed).

    With a sensitivity of 97 dB/W/m for the YD655B00 woofer of the DXR10 (it falls off at the low end but that is compensated partly by +6dB because of bass-reflex), you need around 100W RMS to achieve 120 dB RMS. It's a vented magnet, it probably can handle that for a long time. With 100 dB sensitivity, you'd need 1000W peak to achieve 130 dB, so the 131 dB peak number is tailored to the 1100W claim.

    At 82.41 Hz (low E of guitar) we're limited by speaker excursion, not power. With 4 mm x-max, the speaker of the DXR10 can deliver about 107 dB (+ 6 dB for the bass reflex which effectively doubles the membrane area), so about 113 dB. I don't know the actual x-max of the DXR10 woofer. If it's 3 mm, the number goes down by 3 dB so then it's 110 dB. You need about 10W to achieve that...

    Only at higher frequencies you reach the 120 dB. In comparison, a 15" with 4 mm x-max puts out 115 dB (+ 6 dB) at 82.41 Hz.

    The compression driver of the DXR is also driven bridged by a 2x75W chip, powered between 0 and 55V. This means it can put a total of 55V over the compression driver - plenty for the 20W continuous rating of the driver. I suppose the DSP protects it.

    So 950W + 150W = 1100W, but the RMS power at full blast will be of the order of 100W, maybe 200W.

    On the DXR I checked the 400V capacitor of the multi voltage switch power supply. It's 680 uF. Given that I found a 120 W multi-voltage switch PSU with 2x68 uF, I believe the power supply can deliver about 600W to the amplifiers. I can't check the transformer rating, so I don't know if that link is a strong as the rest, but I could be believe it. It's all well laid out.

    So there you have it: the Yamaha actually isn't that far off of the specs, and that's why it's loud.


    Many FRFR's use a +/-40V rail and drive the woofer bridged like Yamaha does, but then with the TDA8954 or 8953 chip (for example in the ZLX 12P). This amplifier chip is tiny, costs only a few euros and puts out 320W peak sine power at 0.5% distortion over 8 Ohm. That means a bit less headroom than the DXR.


    If you use 4 Ohm speakers (Alto, Headrush do), the funny calculation could be made to fit with +/-40V rail: drive 80V over 4 Ohms = 20 Amps, so 1600W to the woofer. 40V over 4 ohms = 10 amps, so 400W to the tweeter. Together: 2 kW. I doubt that the switched power supply that supplies the amps is rated anywhere near 2kW. Reality remains bound by what the speakers can handle, so ~100W RMS, and peak 2 to 3 times that.


    I have owned both units for years and never felt the need to figure out what's inside. I just play. Anyway, now we know.

    From : link


    General Electric showed in a study in the 1960s that the human ear used higher ordered (and in particular, odd ordered) harmonics in order to sense sound pressure. This is actually quite easy to demonstrate with what is today rather simple test equipment, and according to the Radiotron Designer's Handbook published by RCA, we have apparently known since the 1930s that we are more sensitive to higher ordered distortion products (this might be the 'Inconvenient Truth' of audio, as this particular fact is largely ignored by the majority of the audio industry).
    The ear has a masking principle wherein louder sounds (distortion) mask the presence of quieter sounds (detail) and also because the ear/brain system converts all forms of distortion into tonality (in this case, brighter and/or harsher). The odd bit is that the ear/brain system also has a tipping point where tonality caused by distortion can be favored over actual frequency response errors. So you can turn down the treble in a bright system, but if the brightness is the result of distortion rather than frequency response, it will still sound bright!


    So this is why a low distortion studio monitor will not necessarily sound loud, but will sound very detailed.
    And a tube amp which generates more odd harmonics than the Kemper profile you're comparing it to may be perceived as having more sound pressure, even when a voltmeter or scope would show the same reading as the Kemper output.

    I own the toaster for years. Love it.
    I also wanted a floor board solution. Bought an Helix LT. Indeed without IR's sound was dead. So I had to go find the right IR's to make it work. Even then I could not bond with it. I returned it after a week.
    BTW I couldn't bond with the DXR10 either, but Atomic CLR was instant love.
    Some things just work right away for me and some don't.

    [quote='kneelie',index.php?page=Thread&postID=82671#post82671]I would love 192k/24bit, 176.4k/24bit, or 96k/24bit. out, but what is the internal sample rate because a raw output of that would be best to limit the number of needless conversions.


    found this:
    9. Can you talk about what’s under the hood (processors, speeds, sampling rate, A/D conversion etc)?


    The main DSP is a Freescale DSP (formerly Motorola) running at an equivalent of 400 MHz speed. The code consists of tens of thousands of lines of pure assembler code. The global sampling rate is 44.1 kHz, while the internal sampling rate is partially much higher. The algorithm for the tube simulation runs on more than 700 kHz sampling rate (!).


    source: http://www.guitar-muse.com/kemper-profiling-amp-2949-2949


    Looking at the Freescale site, something like the Freescale Symphony series DSP56720 looks about right: dedicated audio DSP, dual core 2x200 Mhz = 400 Mhz equivalent, supports SDRAM, SRAM, DRAM, EPROM, Flash, SPDIF built in, 10 yr longevity program.
    I can imagine that you want to keep the signal chain at 1 samplerate, so in first instance that would also be what goes to SPDIF.
    On the other hand, the 56720 has a Asynchronous Sample Rate Converter (ASRC) that possibly may be used to convert 44.1 to 48 kHz at the backend for SPDIF.
    According to Freescale:
    The ASRC is hard-coded and implemented as a co-processor, requiring minimal CPU or DSP controller intervention.
    - allows multiple audio data rates in a system
    - 10 channels
    - supports input and output sample rates from 32 kHz to 192 kHz
    - supports three asynchronous output clock domains simultaneously
    So my 2 cents: if this is the DSP then hardware supporting sample rate conversions appears to be available