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Output Impedance , Negative Feedback

چهارشنبه 20 ژوئن 2018
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هرچقدر فیدبک منفی کمتر باشه امپدانس خروجي بيشتر ميشه و صدا بهتره.

Ypsilon Electronics Hyperion monoblock power amplifier its output impedance was relatively high for a solid-state design, at 0.35 ohm

Ypsilon Aelius monoblock power amplifier The output impedance was high for a solid-state design, at 0.44 ohm at 20Hz and 1kHz, 0.47 ohm at 20kHz.

AkitikA GT-102 power amplifier 0.2 ohm at 20Hz, 0.11 ohm at 1kHz, and 0.13 ohm at 20kHz

Constellation Inspiration Stereo 1.0 power amplifier The output impedance was low, varying from 0.11 ohm at 20Hz and 1kHz to 0.12 ohm at 20kHz.

Bel Canto Design Black ACI 600 integrated amplifier The output impedance at the speaker terminals was 0.2 ohm at all audio frequencies

Pass Laboratories XA25 power amplifier The output impedance (including 6′ of speaker cable) was low, at 0.08 ohm at 20Hz and 1kHz, rising very slightly to 0.1 ohm at 20kHz.

PS Audio Stellar M700 monoblock power amplifier The output impedance (including 6′ of speaker cable) was low, at 0.1 ohm at 20Hz and 1kHz.

Pass Laboratories XA60.8 monoblock power amplifier The XA60.8’s output impedance, including 6′ of speaker cable, was 0.07 ohm at 20Hz and 1kHz, rising to 0.1 ohm at 20kHz.

Dan D’Agostino Progression Mono monoblock power amplifier The output impedance was high for a solid-state design, at 0.44 ohm at 20Hz and 1kHz, rising slightly to 0.49 ohm at 20kHz.

Siltech SAGA power amplifier output impedance was higher than is usual for a solid-state design, at 0.23 ohm

Krell OLD KST-100 power amplifier output impedance was close at 0.15 ohms, rising to 0.2 ohms by 20kHz.

Emotiva XPA Gen3 two-channel power amplifier The output impedance, including the series resistance of the speaker cables I used, was very low at 20Hz and 1kHz, at 0.09 ohm, and rose only slightly at 20kHz, to 0.125 ohm.

Mark Levinson No.536 monoblock power amplifier The output impedance was low, at 0.1 ohm at 20Hz and 1kHz, rising slightly to 0.13 ohm at 20kHz.

Audionet Max monoblock power amplifier The output impedance was very low, at 0.045 ohm (including the speaker cable) at 20Hz and 1kHz, rising slightly to 0.07 ohm at 20kHz.

AVM Ovation MA8.2 monoblock power amplifier The output impedance (including 6′ of cable) was very low, at 0.08 ohm at 20Hz and 1kHz, rising very slightly to 0.1 ohm at 20kHz

Boulder Amplifiers 2150 monoblock power amplifier Measurements The output impedance, including 6′ of speaker cable, was extremely low, at 0.02 ohm at low and middle frequencies; this rose only slightly, to 0.026 ohm, at the top of the audioband

Bel Canto e.One Ref600M power amplifier output impedance, including 6′ of cable, was low, at 0.1 ohm from 20Hz to 20kHz.

First Watt J2 power amplifier The output impedance was relatively high for a solid-state design, at 0.5 ohm from 20Hz to 20kHz.

PS Audio BHK Signature 300 monoblock The output impedance (including the series resistance of 10′ of speaker cable) was on the high side for a solid-state amplifier, at 0.16 ohm at all audio frequencies

Ayre Acoustics MX-R Twenty monoblock power amplifier the output impedance was higher than usual for a solid-state design: 0.26 ohm at low and middle frequencies, rising to 0.28 ohm at 20kHz

Simaudio Moon Evolution 860A power amplifier The output impedance, including 6′ of speaker cable, was very low, at 0.09 ohm at low and middle frequencies, rising to 0.11 ohm at the top of the audioband.

Bricasti Design M28 monoblock power amplifier The output impedance, including 6′ of speaker cable, was very low, at 0.05 ohm at low and middle frequencies, rising to 0.06 ohm at the top of the audioband.

Constellation Audio Performance Centaur Mono monoblock power amplifier The output impedance was low for a bridged design, at 0.07 ohm including the cable at low and middle frequencies, rising slightly to just under 0.1 ohm at the top of the audioband.

Symphonic Line Kraft 400 monoblock power amplifier The output impedance was 0.04 ohms or less at low and mid frequencies, and under 0.07 ohms at 20kHz

Mark Levinson No.53 Reference monoblock power amplifier The output impedance at low frequencies was very low, at 0.06 ohm, including 6′ of speaker cable. This rose to 0.25 ohm at the top of the audioband

darTZeel NHB-458 monoblock amplifier The output impedance was high for a solid-state design, at 0.3 ohm (including 6′ of speaker cable) at all frequencies.

Lamm M1.2 Reference monoblock power amplifier The output impedance at the “1–6 ohms” setting was 0.15 ohm at low and middle frequencies, rising to 0.18 ohm at 20kHz.

MBL Reference 9011 monoblock amplifier The output impedance was very low, at 0.05 ohm at 1kHz and below, rising to 0.1 ohm at 20kHz.

Goldmund Mimesis 8 power amplifier output impedance: assessing this by measuring the voltage drop at a 1W level when an open-circuit load was replaced by an 8 ohm resistor gave a figure of 0.04 ohms at 20Hz and 1kHz. Repeating the measurement using a 4 ohm load, thus demanding twice as much current, gave a figure twice this, at 0.08 ohms. Though this is still inconsequential, it puzzles me, as does the fact that at 20kHz, the discrepancy was smaller, at 0.12 ohms (8 ohm load) vs 0.15 ohms (4 ohm load).

Soulution 710 power amplifier The output impedance, including the series resistance of 6′ of speaker cable, was very low, at 0.06 ohm at 20Hz and 1kHz, rising to 0.075 ohm at 20kHz.

Luxman B-1000F monoblock power amplifier The output impedance was 0.125 ohm at 20Hz and 1kHz, rising slightly to 0.15 ohm at 20kHz.

Electrocompaniet AW400 monoblock power amplifier The output impedance is specified as being below 0.01 ohm, which is very low. However, including 6′ of speaker cable, I measured 0.135 ohm at low and midrange frequencies, rising very slightly to 0.17 ohm at the top of the audioband.

Bryston 7B SST2 monoblock power amplifier The Bryston’s output impedance was low for a balanced design, at 0.11 ohm at low and middle frequencies, rising slightly to 0.17 ohm at 20kHz.

Aesthetix Atlas power amplifier The output impedance was relatively high for a design with a solid-state output stage, at 0.34 ohm at low and middle frequencies, rising very slightly to 0.36 ohm at 20kHz.

EAR/Yoshino M100A monoblock power amplifier Transformer Coupled It was lowest from the 4 ohm output, at 0.55 ohm across most of the audioband—equivalent to a damping factor of just under 8—rising to 1.05 ohms at 20kHz. The 8 ohm figures were 1.1 ohms, rising to 2.3 ohms at 20kHz, while from the 16 ohm transformer I measured 1.9 ohms at 20Hz and 1kHz, and a very high 4 ohms at 20kHz.

Adcom GFA-565 monoblock power amplifier output impedance was uniformly low—ranging from 0.02 ohms from the bass to the upper midrange and increasing slightly to 0.03 ohms at 20kHz.

Musical Fidelity 750K Supercharger monoblock power amplifier The output impedance was a little higher than usual for a solid-state design, at 0.11 ohm at 20Hz and 1kHz, rising to 0.22 ohm at 20kHz

Electrocompaniet Nemo monoblock power amplifier The output impedance measured higher than specified, at between 0.17 and 0.23 ohms

Accuphase M-2000 monoblock power amplifier The output impedance was a extremely low 0.035 ohm at 1kHz, increasing to a measured maximum of 0.038 ohm at 20Hz into an 8 ohm load, and 0.063 ohm at 20kHz into 4 ohms.

Halcro dm88 Reference monoblock power amplifier The dm88’s output impedance was very low across almost all the audioband, at 0.1 ohm including 6′ of multistrand speaker cable. It rose slightly at 20kHz, to 0.14 ohm

McIntosh MC501 monoblock power amplifier output impedance, I was surprised to find it to be lowest from the 8 ohm tap, at 0.08 ohm, and highest from the 2 ohm tap, at 0.13 ohm

Edge NL-12 power amplifier output impedance was 0.1 ohm over most of the audioband, rising to 0.3 ohm at 20kHz

Pass Labs X1000 monoblock power amplifier The output impedance measured between 0.18 and 0.20 ohms (the higher values were at 20kHz)

47 Laboratory 4706 Gaincard power amplifier I measured around 0.15 ohm at 20Hz and 1kHz, this dropping to 0.08 ohm at 20kHz

Hovland Radia power amplifier The output impedance was a low 0.2 ohm across most of the band, this rising inconsequentially to 0.26 ohm at 20kHz.

Krell Full Power Balanced 350mc monoblock amplifier the output impedance was 0.18 ohms at 1kHz and 20kHz, 0.15 ohms at 20Hz

Musical Fidelity Nu-Vista 300 power amplifier the output impedance was 0.08 ohms/0.04 ohms, left/right, respectively, across most of the audio band.

Linn Klimax 500 Solo monoblock power amplifier The output impedance measured a minuscule 0.043 ohms at 1kHz, increasing to 0.071 ohms at 20kHz, both assessed by measuring the voltage rise when an 8 ohm load was open-circuited. Using 4 ohm loads, this increased by a maximum of 0.01 ohm at 20kHz, to 0.081 ohm.

Mark Levinson No.33H monoblock power amplifier output impedance was extremely low at 0.02 ohms at 20Hz and 1kHz, rising slightly to 0.026 ohms at 20kHz.

Ayre AX-7 integrated amplifier Its output impedance was moderately high for a solid-state design, at 0.4 ohm across the audioband.

ASR Emitter II Exclusive integrated amplifier The ASR’s output impedance was very low: <0.08 ohm across the audioband (including the series impedance of 6′ of multistrand speaker cable).

Mark Levinson No.383 integrated amplifier The power-amplifier output impedance was very low at approximately 0.055 ohms across most of the audioband, this rising to a still negligible 0.065 ohms at 20kHz

Luxman L-509X integrated amplifier The output impedance at the speaker terminals was a very low 0.075 ohm at low and middle frequencies, rising to 0.11 ohm at the top of the audioband.

NAD C370 integrated amplifier output impedance across most of the audioband a moderate 0.16 ohm, this rising slightly to 0.19 ohm at 20kHz.

Ayre Acoustics AX-5 integrated amplifier The AX-5’s output impedance was a little high for a solid-state design, at 0.27 ohm across the audioband, this due to the circuit’s lack of loop negative feedback.

Roksan Kandy K2 integrated amplifier  output impedance was a very low 0.09 ohm at low and middle frequencies, rising slightly to 0.15 ohm at 20kHz due to the usual series Zobel inductor.

Sugden A21ai The amplifier’s output impedance from the speaker jacks was a low 0.12 ohm at 1kHz, rising to 0.39 ohm at 20Hz and 0.25 ohm at 20kHz.

Krell FBI integrated amplifier The output impedance (including 6′ of speaker cable) was very low, at 0.08 ohm at low and midrange frequencies.

darTZeel CTH-8550 integrated amplifier The CTH-8550’s output impedance was quite high for a solid-state design, at 0.4 ohms in the bass, rising to 0.5 ohms at 20kHz.

Devialet D-Premier D/A integrated amplifier I measured 0.04 ohm at all audio frequencies, but thus includes the impedance of the 6′ of speaker cable I used for the test. But the fact that the D-Premier has an extraordinarily low output impedance

Naim Audio NAIT 5si integrated amplifier The output impedance ranged from 0.34 ohm at 20Hz and 1kHz to 0.36 ohm at 20kHz—a very slight increase.

Creek Evolution 100A integrated amplifier The output impedance at the speaker terminals was less than 0.1 ohm, including 6′ of speaker cable.

GamuT Di150 LE integrated amplifier The Di150’s output impedance was very low, at 0.09 ohm (including 6′ of speaker cable) at low and middle frequencies, and rose only slightly, to 0.11 ohm, at 20kHz.

Cyrus 6vs integrated amplifier impedance of 0.08 ohm at low and midrange frequencies, rising very slightly to 0.1 ohm at 20kHz.

Arcam Alpha 10 integrated amplifier The output impedance is comfortably low—under 0.06 ohms up to 1kHz, rising to a maximum of 0.16 ohms at 20kHz

Chord CPM 3300 integrated amplifier the source impedance was a very low 0.06 ohm, rising slightly to 0.08 ohm at 20kHz.



Jadis JA200 Mk.II monoblock power amplifier The output impedance (including 6′ of speaker cable) was very low for a transformer-coupled tube amplifier, at 0.18 ohm at 20Hz and 1kHz, and 0.14 ohm at 20kHz.

PrimaLuna ProLogue Premium power amplifier The output impedance from both output taps was significantly higher than that of the ProLogue Premium integrated amplifier. From the 8 ohm tap, I measured 9.2 ohms at 20Hz, 8.75 ohms at 1kHz, and 8.45 ohms at 20kHz; from the 4 ohm tap, the respective impedances were 4.7, 4.5, and 4.35 ohms.

Thöress 300B monoblock power amplifier The output impedance was relatively low for a single-ended design, at 1.9 ohms at 20Hz and 1kHz, rising to 2.4 ohms at 20kHz

Air Tight ATM-1S power amplifier the Air Tight’s output impedance was high, ranging from 3.6 ohms at 20Hz to 3.3 ohms at 20kHz. This suggests that the single output-transformer tap is optimized for a 4 ohm load

VTL Siegfried Series II Reference monoblock power amplifier he lowest impedance was in triode/MDF mode, at 0.88 ohm in the midband and 0.9 ohm at the extremes of the audioband. The output impedance rose a little with each reduction of feedback, reaching 1.36 ohms at 1kHz and 1.4 ohms at the frequency extremes in the LDF condition.

Sophia Electric 91-01 300B monoblock power amplifier output impedance was also high, at 5.6 ohms at 20Hz, 3.7 ohms at 1kHz, and 8–9 ohms at 20kHz

Lamm Industries ML3 Signature monoblock power amplifier Even without any negative feedback, the 4 ohm tap’s output impedance was moderately low for a single-ended design, at 1.55 ohms at low and middle frequencies, rising to 1.9 ohms at the top of the audioband. Without NFB, the output impedance was much higher from the other taps, the 8 ohm tap measuring 2.9–3.9 ohms

Lamm ML2.2 monoblock power amplifier The output impedance also depended on the transformer tap used. From the 4 ohm tap, the impedance was a low 0.38 ohm at low and middle frequencies, rising to 0.5 ohm at 20kHz. As expected it was higher from the 8 ohm tap, at 0.7 ohm at 20Hz and 1kHz, and 0.8 ohm at 20kHz, and higher still from the 16 ohm tap: 1.35 ohms at all audio frequencies.

Octave Audio RE 290 power amplifier The output impedance was fairly high, measuring 2.2 ohms at 20Hz and 1kHz, rising slightly to 2.4 ohms at 20kHz.

Jadis SE300B monoblock amplifier The output impedance was 2.5 ohms at 1kHz, 2.7 ohms at 20kHz, and 0.76 ohms at 20Hz.

Jadis Defy-7 Mk.II power amplifier with solid-state amplifiers it is usually sufficiently low to be considered negligible: 0.1 ohm or under. With a number of tube amplifiers it can be rather higher. For example, the “1000”—a low-feedback triode amplifier made by Audio Innovations in the UK—has an output resistance of several ohms. This is by no means an isolated example. The Defy-7’s output or source resistance was quite constant over the audio band, ranging from 0.45 ohms at 20kHz, to 0.4 ohms, midband, and 0.45 ohms at 20kHz.

Allnic Audio A-5000 DHT monoblock power amplifier output impedance was impressively low for a single-ended tube design even from its 16 ohm tap, where it measured 1.6 ohms at 20Hz, 1.35 ohms at 1kHz, and 1.5 ohms at 20kHz. As expected, the respective impedances were lower from the 8 ohm tap: 1, 0.8, and 0.85 ohm.

Conrad-Johnson LP125M monoblock power amplifier The source impedance from the single pair of output terminals was a high 2 ohms at low and middle frequencies, rising slightly to 2.2 ohms at 20kHz

VTL MB-450 Series III Signature monoblock power The output impedance also depended on the amount of negative feedback and the operating mode. The lowest impedance was in triode mode with the maximum amount of feedback, where it measured 0.87 ohm in the midrange, rising very slightly to 0.89 ohm in the low bass and 0.9 ohm at the top of the audioband. With the lowest amount of feedback, these figures rose to 1.45, 1.5, and 1.53 ohms, respectively. Operating the MB-450 in tetrode mode gave a slightly higher source impedance, ranging from 0.95 ohm at 1kHz with the maximum feedback, to 1.95 ohms at 1kHz with the lowest feedback.

Balanced Audio Technology VK-55SE power amplifier the output impedance from the VK-55SE’s High and Med taps was high, at 2.7 ohms in the midrange, rising to almost 3 ohms at high and low frequencies

AudioValve Baldur 70 monoblock power amplifier The Baldur’s output impedance, from both the 8 and 4 ohm transformer taps, was lower than usual for a tubed amplifier, at 0.66 and 0.42 ohm, respectively.

Yamamoto A-08 power amplifier The right channel measured 3.2 ohms at 20Hz, dropping to 2.85 ohms at 1kHz and 2.16 ohms at 20kHz. The corresponding figures for the right channel were a little lower, at 3 ohms, 2.73 ohms, and 2.05 ohms, respectively, but these are still high in absolute terms.

Bel Canto SET 80 monoblock power amplifier The measured output resistance was 1.7 ohms, corresponding to a low damping factor of 4

Cary Audio Design CAD-805 monoblock power amplifier output impedance is 0.74 ohm

Cary CAD-1610-SE monoblock power amplifier impedance from the 8 ohm output was high at 3.4 ohms across most of the band, rising to 4 ohms at 20kHz.

Graaf GM 200 OTL power amplifier The output impedance was 0.79 ohms at 1kHz, 0.7 ohms at 20Hz, and 0.83 ohms at 20kHz

Audio Research VTM200 monoblock power amplifier output impedance also varied according to which tap was used: a reasonably low 0.35 ohm from the 4 ohm tap, rising to 0.6 ohm from the 8 ohm tap and 1 ohm from the 16 ohm tap

Sonic Frontiers Power 3 monoblock power amplifier output impedance a maximum of 0.17 ohm at 20Hz and 1kHz, increasing to a maximum of 0.22 ohm at 20kHz

Wavac SH-833 monoblock power amplifier The SH-833’s output impedance was on the high side, at 5 ohms from the 8 ohm transformer tap and 2.6 ohms from the 4 ohm tap. These figures were maintained across most of the audioband, with a rise at 20kHz from the 4 ohm tap to 3.6 ohms. Unexpectedly, the 20kHz source impedance from the 8 ohm tap dropped from 5 to 4.3 ohms.

EAR 890 power amplifier the EAR 890’s output impedance was around 1.5 ohms from its 16 ohm tap and 0.5 ohm from the 8 ohm tap

Wavelength Audio Cardinal XS monoblock amplifier output impedance was a very high 3.4 ohms at 1kHz, increasing to 5.9 ohms at 20Hz and 5.3 ohms at 20kHz.

Antique Sound Lab Explorer 805 DT monoblock power amplifier output impedances were high, at 7.2 ohms (16 ohm tap), 3.65 ohms (8 ohm tap), and 2 ohms (4 ohm tap)

Audiopax Model Eighty Eight monoblock power amplifier output impedance also varied with the control settings. It was lowest with them at full clockwise, but was still very high in absolute terms at 3.5 ohms across most of the audioband. Unusually, the source impedance dropped at low frequencies, reaching 1.8 ohms at 20Hz.

Air Tight ATM-211 tube monoblock power amplifier output impedance measured a high 3.7 ohms at 1kHz. Unusually, this decreased slightly to 3.5 ohms at 20Hz and 20kHz.

Hovland Sapphire power amplifier The output impedance from the 16 ohm tap was a very high 3.55 ohms across most of the audioband, rising to 4.3 ohms at 20kHz.

Wavelength Audio Gemini monoblock power amplifier The low-frequency source impedance with the 2A3 was 3.7 ohms (8 ohm tap) or 1.7 ohms (4 ohm tap); the midrange impedance was 3.1 and 1.5 ohms, respectively; while at 20kHz, the impedance rose to 4.6 and 1.8 ohms, respectively. The respective figures with the 45 tube were 3.2 and 1.5 ohms, 5.5 and 2.6 ohms, and 6.8 and 2.9 ohms.

McIntosh Labs MC2000 power amplifier The output impedance measured a maximum of 0.4 ohm—quite high in comparison with solid-state amplifiers but admirably low for a tube amp.

Nagra VPA monoblock power amplifier output impedance was quite high, varying between 2.23 and 2.26 ohms depending on frequency and load impedance (measurements taken at the 8 ohm output).

Manley Laboratories Stingray integrated amplifier The output impedance, however, was also high, ranging from a maximum of 3.87 ohms at 20Hz to a minimum of 2.83 ohms at 20kHz—values guaranteed to affect the amplifier’s frequency response through real loudspeakers.

EAR V20 integrated amplifier The output impedance from the 8 ohm speaker terminals measured a minimum of 0.77 ohms and a maximum of 1.14 ohms, the latter at 20kHz. The corresponding values from the 4 ohm taps were 0.47 ohms and 0.62 ohms—respectably low values for a tube amplifier. The output impedance at the V20’s tape-monitor jacks was 1k ohm with a 25 ohm source impedance, and 1.56k ohms with a 600 ohm output impedance, indicating an unbuffered tape output.

Audio Note Jinro integrated amplifier The output impedance from the single transformer tap was commendably low for a single-ended-triode design, at 2.5 ohms

Leben CS300 integrated amplifier The output impedance at 20Hz was similar from all three outputs at 3–3.5 ohms. It did differ at higher frequencies: at 1kHz, the impedance was 2.65 ohms (8 ohm tap), 2.25 ohms (6 ohm tap), and 1.8 ohms (4 ohm tap); at 20kHz, it was 2.4, 1.6, and 1.1 ohms, respectively.

Quad II Classic Integrated amplifier The output impedance was on the low side for a transformer-coupled tube amplifier, at 0.28 ohm at 20Hz and 1kHz, dropping to 0.2 ohm at 20kHz.

Woo Audio WA5 integrated amplifier The output impedance from the speaker terminals was also high, at 3.2 ohms across the audioband

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شنبه 2 ژوئن 2018
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یه برند خوب

On The Road: A Visit To BorderPatrol

طراح بلندگوي 2way شركت ليوينگ وويس تو كل مراحل طراحي و ساخت اين بلندگو از پوش پول 300b شركت kondo audionote استفاده كرده و چون قيمت kondo ربطي به قيمت اين بلندگو نداره طبيعتا بايد براي Kondo دنبال جايگزيني مي گشتم كه به اين آمپلي فاير BorderPatrol P20 رسيدم.

Each PSU contains three independent tube rectified, choke input filter high voltage supplies to independently feed the 300B’s, the input/driver tubes and the negative bias supply, together with filament supplies for the 300B’s and small signal tubes. The use of choke input filtering is a critical difference and is unique to BorderPatrol power amplifiers. Other brands use tube rectification and choke smoothing but do so in the cheaper, smaller and easier to implement capacitor input filter configuration. A choke input filter design has far superior voltage regulation (stiffness) and noise rejection. The PSU’s stiffness explains why the BorderPatrol amplifiers have bass performance and dynamics unlike any other 300B amplifiers.



Why design a 300B amplifier in push-pull rather than single-ended-triode (SET) mode?

“A push-pull 300B design will typically have 20Wpc, which allows it to work with a wider range of loudspeakers,” Dews said. “Push-pull amps typically have tighter, punchier bass than SETs, can drive much more difficult speaker loads, and play a wider selection of music. A stiff power supply, like the ones used with BorderPatrol SETs, addresses the SET dynamic and bass issues significantly, but a similarly executed push-pull like the P21 will still be superior in those areas.”

Note Dews’s description of the typical sound of 300B SET amplifiers: “[They] are often charming and beguiling but have low power, are limited dynamically, and have poor bass performance. They need to be partnered with loudspeakers that have very high sensitivity, and a high and uniform impedance characteristic. Complex music with heavy bass is best avoided. SETs predominantly generate even-order harmonic distortion, which gives them the characteristic open, airy, romantic sound. It’s not accurate, but it can be nice. Push-pull amplifiers predominantly generate odd[-order] harmonic distortion, which leads to a sharper, less romantic sound, but one that is still engaging.”


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آقاي محمدپور تشكر

دوشنبه 21 می 2018
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بسيار بسيار تشكر از آقاي محمدپور واقعا خيلي شرمنده لطف ايشون شدم ، انشاالله جبران كنم .

فروشگاه آكوستيك به مديريت جناب آقاي محمد پور يكي از فعال ترين فروشگاه هاي تهران در زمينه ارائه سيستم هاي صوتي مي باشند.

ميشه براي شنيدن صدا و ورود به دنياي هاي فاي از اينجا شروع كرد ، آقاي محمدپور بسيار جدي و فعال پيگير پاسخ دادن به نياز مشتريان هستند در ضمن در زمينه هاي فاي مطالعه هم دارند.
آدرس فروشگاه :
آدرس اينستاگرام:

خيابان جمهوري تقاطع حافظ پاساژ بزرگمهر طبقه دوم واحد ٧٦
تلفن ٦٦٧٣٤٩٣٦ ٦٦٧١٧٣٦٢
همراه ٠٩١٢١٥٧٨٧٠٩

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ASR Emitter I Exclusive Battery Review

دوشنبه 21 می 2018
/ / /

مرسی از آقای محمدپور برای امکان تست امیتر



من دو برند ترانزيستوري رو اگر دوستان خاطرشون باشه معرفي كردم كه بياد ايران يكي همين ASR Emitter بود و ديگري Vitus Audio و معتقد بودم براي زير ١٠٠ وات از ويتوس و براي بالاي ١٠٠ وات ميشه از ASR براي بلندگوهاي داخل ايران كه حساسيت كمتر از ٩٥ دي بي دارند بهره گرفت.

آمپلي فاير اميتر يه تنظيمات و امكاناتي داره كه براي استفاده بهتر توضيح ميدم.

از سال ٢٠١٦ به بعد تغييرات عمده اي در اميتر يك و دو رخ داده كه به اعتقاد طراح اين برند صدا خيلي بهتر شده.

اميتر سه تا طبقه مدار پشت سرهم داره كه همشون از FET يا Mosfet استفاده شده و طبق نوشته منوال فيدبك داخلي نسبتا بالايي داره براي كنترل ديستورشن.

مدلهاي ٢٠١٦ به بعد اين دستگاه امكان تنظيم درايو بلندگو رو (از طريق تنظيم فيدبك منفي) با ٤ سوييچ براي انتخاب چهار حالت به شما ميده. در حالت ديفالت فقط يك سوييچ فعال هست و شما ميتونيد هر چهار سوييچ رو فعال كنيد براي افزايش كنترل روي بلندگو و يا ميتونيد هر چهار سوييچ رو غير فعال كنيد براي بلندگوهايي با افيشنسي بالاتر.

نامه اي كه طراح اميتر توضيح داد :


So more compensation so more stable is the amp.  

Compensation switch 

The comp switch has two positions :

Off is to the front, ON is  to the back- is also marked on the board. 

The compensation switch put some additionally parts in the feedback loop

at the main Main unit

but that may make the sound a little bit slower slower and less detailed.

The stability at complex loads is getting better.

اميتر يك از دو ترانس ٧٠٠ ولت امپر استفاده ميكنه و براي درايو بلندگو تا امپدانس ١/٥ اهم هم توانايي لازم رو داره. اميتر يه ورودي Dir داره كه ورودي غير بالانس هست و پيشنهاد ميكنه بهترين كيفيت صدا از همين ورودي غير بالانس RCA درمياد و پيشنهاد ميكنه بهتره از كابل كوتاه RCA غير بالانس استفاده كنيد.

اميتر بانك خازني بسيار بسيار بزرگي داره و وقتي موزيك در حال پخش هست اگر دستگاه رو از حالت روشن ببريد روي استندباي متوجه ميشيد كه تا يك دقيقه بعد هم همچنان موزيك پخش ميشه كه دليلش همين بانك خازني بزرگ هست و به همين دليل پيشنهاد من اينه قبل هر تغييري حتما سيگنال رو از سورس قطع كنيد.

به دليل وجود همين بانك خازني بزرگ و ورودي Dir ملاحضات ديگري هم براي روشن خاموش كردن توصيه ميشه :

براي روشن كردن اول از همه مطمئن شويد ASR روي وضعيت خاموش هست و ابتدا كابل نقره اي psu رو وصل كنيد بعد كابل نقره اي باتري رو وصل كنيد
بعد كابل برق psu رو بزنيد و بعد از يك دقيقه كابل برق باتري رو وصل كنيد.
دستگاه رو به مد Standby ببريد و بعد از يك دقيقه از حالت استندباي به وضعيت روشن ١ يا ٢ تغيير بديد.

اميتر بعد از روشن شدن حداقل ٢٠٠ ساعت نياز به اب بندي داره كه از نظر من با توجه به كيفيت خوب قطعات و حجم بانك خازني بعيده زير ٦٠٠ ساعت كامل اب بندي بشه.

براي خاموش كردن اميتر ابتدا دستگاه رو به مد استندباي ميبريم و بعد از گذشت يك دقيقه به وضعيت خاموش سوييچ ميكنيم. بعد از مدت يك دقيقه كابل برق باتري رو ميكشيم و بعد كابل برق psu رو جدا ميكنيم . بعد از گذشت چند دقيقه اتصالات كابل نقره اي باتري و بعد psu رو جدا كنيد.

اميتر پولاريته درست اتصال هر دو كابل برق به باتري و psu رو (سمت چپ از روبرو) در دفترچه توضيح داده.

پولاريته درست ASR Emitter :


با تنظيم پشت psu بصورت سوييچ صفر و يك ميتونيد از ايجاد گراند لوپ در سيستم جلوگيري كنيد به اين شكل كه فرض كنيد DAC شما و اميتر هردو با كابل برق به گراند پريز متصل هستند. اگر DAC شما طوري طراحي شده كه گراند آن توسط كابل RCA به اميتر متصل ميشه ميتونيد با صفر قرار دادن وضعيت از اين اتصال جلوگيري كنيد و اگر DAC شما گراند رو به كابل RCA وصل نميكنه ميتونيد از وضعيت ١ استفاده كنيد.
طبق پيشنهاد دفترچه هر دو حالت رو ميتونيد براي رسيدن به كمترين نويز امتحان كنيد.

وقتي كليد پشت psu در حالت صفر هست گراند يونيت ASR به گراند بدنه psu و گراند بدنه باتري وصل نيست.

اميتر فاز سيگنال رو برعكس نميكنه و اگر سيستم درست گراند شده باشه نبايد هيچ نويزي در بلندگو بشنويد.

عمر متوسط باتري اميتر ٦ تا ٧ سال است من شدیدا توصیه میکنم بعد ٧ سال عوضش کنید.

بهتره باكس psu از خود ASR و ديگر كامپوننت ها دور باشه بدليل ميدان مغناطيسي قوي و حداقل فاصله ٢٠ سانت در دفترچه ذكر شده است.

هرگونه تغيير در سيستم مانند جداكردن كابلها و يا اتصال كابلها بايد سيگنال موسيقي را قطع كرده و اميتر را در حالت خاموش نگه داريد.

امپدانس ورودي اميتر قابل تنظيم بوده و اگر DAC شما يه Phono شما امكان جريان دهي بالايي دارد ميتوانيد امپدانس ورودي را در پايين ترين حالت قرار دهيد و اگر DAC شما امكان جرياندهي بالايي ندارد بهتر است با انجام تنظيمات امپدانس ورودي را افزايش دهيد.

مدار اميتر بصورت پوش پول Class A/B بوده و در بيشتر مواقع در مد Class A Moderate كار ميكند و گراند لوپ داخلي ندارد.

تمامي كابلها و اتصالات داخلي اميتر نقره خالص هستند.

ترجيحا اميتر را با يك خروجي بلندگو سفارش دهيد تا سر راه خروجي رله قرار نگيرد.

طراح اميتر معتقد است اين آمپلي فاير هم براي بلندگوهاي بد درايو و هم براي هورن و بلندگوهاي حساسيت بالا مناسب است.

بيشتر مطالب بالا برگرفته از دفترچه راهنماي اميتر بود و در ادامه ممكنه يه ريويو از صداي اميتر بنويسم .

مقايسه صدا بين اميتر با باتري و بدون باتري در شرايطي كه پيور پاور نداشته باشيد ١٠٠٪؜ با باتري وضعيت بهتر هست.

ادامه دارد …

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Speaker Placement

دوشنبه 30 آوریل 2018
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خیلی وقتمو گرفت و بهتر دیدم خلاصه هرچی در مورد جابجایی بلندگو تو سایت رومی و کتاب جیم اسمیت دیدم رو بنویسم:

اين وقت و هزينه هايي كه از سمت من انجام ميشه و با حاشيه هم همراه هست اميدوارم حداقل شما مخاطب عزيز بهره لازم از اين نوشته هارو ببريد.

اخر هر جمله يه ستاره * گذاشتم كه تفكيك مطالب انجام شده باشه.

It is my strong conviction that an acoustic system (and practically horn) should sit in the middle of a room. *

We are always careful to place the speakers equal distance from the listening position. *

when the loudspeakers are closeting to the DPoLS then the narrowing speakers or toeing then out create a wider soundstage AND at the same time the center image movies FORWARD. *

The inspiration suggests that time alignment is the key for DPoLS setting. *

ou set up left loudspeaker (left for you) where I need to be. I always suggest starting from left loudspeaker as it cares the fist violins, sopranos of choruses and generally in western culture is associated with HF lead. *

the loudspeaker should not be source of sound in your room. *

in fact any speaker should be kept at least 3’-5’ from the walls to let “imaging to breathe”. *

When you walk into a good room it clearly heard as a different acoustic event. It is very pleasant to be and very pleasant to talk in a good room. I am always aware of its acoustics in good room. *

The inspiration suggests that time alignment is the key for DPoLS setting *

1) Alignment of the Channels of the same loudspeaker
2) Alignment of BOTH loudspeaker in the room
3) First major reflections of the room for both loudspeakers. *

Sure the DPoLS are practically not known techniques and it is great to “handle” it but there is something else in there. The DPoLS sure is very powerful and superbly influential but they are the last stork of brash, the final kink in the art of playback positioning. However, the majority of the installations out there do not even approach to the point of failure at the DPoLS level as they fail at much lover level – the macro-positioning. *

if the Imbedded Macro-Positioning is well-thought and properly implemented then whatever the common audio-knowledge suggest about speakers positioning become … irrelevant. *

Acoustic system is not only loudspeakers and not the loudspeaker-room interaction but rather the sound of the room itself. The loudspeakers juts trigger the room… if the loudspeakers in-phase with room *

3) If your objectives are the “ADVANCED AUDIO AND EVOLVED MUSIC REPRODUCTION TECHNIQUES” THEN practically all Commercial loudspeakers are compromised. It is not necessary because they are bad but because any Commercial off the shelf loudspeaker does not utilizes the “Imbedded Macro-Positioning” and therefore fundamentally underperforms. (Unless the specially built rooms that it very different subjects). I am very much NOT against the Commercial loudspeakers – they just should be designed differently for the “evolved music reproduction”. *

4) There are no such a things as bad sounding rooms, at least at the Macro-Positioning level. The rooms sound bad because the Imbedded techniques were not used properly or were not used at all. *

5) Acoustic treatment as it exists in today Hi-Fi (unless we are talking about VERY large diffusers and very large resonators… that never used) are not applicable at sub approximately 700Hz. The carp that the Morons use for bass and upper bass control is juts horrible as it shrink reverberation time at higher frequencies. Unless we do for VERY high expense and large custom made solutions the sub 700Hz are not intentionally controllable. The irony is that with the Imbedded Macro-Positioning … it is good they are controllable because it becomes the… benefit. *

Yes, a cheep but good consumer amplifier with a inexpensive old JBL monitor, properly Macro-Positioned, will literally destroy a performance of $250.000.00 high-end installation of the installation is … against the interests of the room. Do I have to pump you up more? *

So, what to do next? Let forgot whatever crap you have heard about better and worst rooms for Sound, unless you build a listening dedicated rooms from scratch, that, as a concept, has it’s own problems. Let forgot a point about the topologies of loudspeaker. Also, let forgot the idiotic idea that the industry have implanted in you – “you have to find best position in your room for your given loudspeaker”. *

one of the major task of the Imbedded Macro-Positioning to place your loudspeaker in-wide-phase with the Room Polarity. This is the absolutely mandatory. Without your acoustic system (or at least the fundamental channels of your loudspeakers) operating in-wide-phase with your room to get an “interesting sound” is practically impossible. *

Ok, how to start. First of all we have to understand that out subject of attention will be a region between approximately 80Hz and 500Hz or something that shape the fundamentals of the “melody range” within your playback. Search my site I have written about the prominence of the upperbass a lot ….

Most of the mid and small size rooms (very large room is a separate subject) have two types of the room modes: narrow modes and wide modes. Your primary task at this point is to find where those wide modes in your room will be. As you understand when I talk about the room modes I mean the modes in the Melody Range. The narrow modes are 1/4 octave wide and you should discard them. You need to search for picks that would be wider then ½ octaves. I have seen some rooms that have plus 5-6dB at 130Hz and 1.5 octaves wide. The smaller room the more room modes might be and the more “problem” the room most likely will have. Do not worry about it – search for the wildest bandwidth and for the highest amplitude of modes in your room and those “problem” will not be problems if you use them properly.

How to search it? It is deepens of many parameters. Your listening expertise and topology of your loudspeakers (the Melody Range channels) are not the last among all variables. Generally a simple RTA with 1/6 octave and higher will do. You do not need any good quality microphone as you do not care about absolute number but rather about the relative values. Running pink noise with very fast averaging and resetting itself after, I would say, 15 seconds is a good tool. You need to connect one channel of your Melody Range and to let your friend to tango with the speaker (or with a Melody Range substitute this will be even better) across your room. Do not forget moving your microphone and well if it is necessary. You should sit at your desirable listing spot with microphone, looking at your RTA and to listen the sound. The combination of what you hear and what the RTA shows is be a good tool for you to get a sense of directions. Change at least a half of dozen listening positions until you feel that you found the SECOND UGLIEST LOCATION IN YOU ROOM. What does it mend the “second ugliest”. The first the most ugliest location for your Melody Range will have a maximum amplitude of the narrow modes (Do not approach to the room’s walls closer then 3 feet). Keep looking you need to look for the WIDEST BANDWIDTH, EVEN AT SLIGHTLY LOVER AMPLITUDE.

If you are ability lucks of any sensibility and hearing then you might use you loudspeaker as a microphone. Fill your room with band-pass noise (Melody Range) and look with a meter or scope how much signal you your speaker pick up at their output. It is very erroneous way but it juts a handicapped way for the deaf beginners. Do not forget that in this case you will need to conduct a number of discrete measurements at different very narrow band-pass frequencies in order to determine how wide bandwidth of the room gain. Do not forget also that the position of room noise-filing speaker should be at the position where you will be listening from.

Anyhow, some people might propose to use some kind of modeling software but I personally have no positive experience with software and I considering then superfluous toys. Perhaps it is juts me. You might use whatever means are available in your disposal to found the “boomiest”, across wide bandwidth, spot in your room. *

Two important comments. This relatively-wide bandwidth “agilest” spot in your room will not be a spot but rather a quite large space. It is very difficult to make a generalization about it but generally this quite “large bloomy space” will be a space equal to, I would say, 1/8-1/12 of the room dimensions. I call this large bloomy space as ACOUSTICAL EROGENOUS ZONE (AEZ) of your listening room. Now you need to found the AEZ’s dimensions. Mark the found location of your AEZ and move your speakers a few feet in all direction measuring loosing the room gain within the Melody Range. Eventually you will discover an approximate relatively large region in your room that might be considered as AEZ. Now, you need to find the AEZ’s polarity. The AEZ’s polarity is the AEZ’s side where the room gain in the Melody Range would be at maximum. The AEZ might be right-polarized or left-polarized.

So, what to do next? Now you need…. to place both of your loudspeakers in the AEZ, trying to position the right channel in the most polarized region of your AEZ. Regardless speaker’s topology is used your loudspeaker’s Melody Range they, the loudspeakers, must be INSIDE of the AEZ. It is not only because you have extra feed db gain in there – this it beneficial but very secondary. The key is that now your artificial room transducer (speaker) is in-phase with nativity of your listening room and then do not work against each other.

If your Melody Range channel (or live instrument as well) are in AEZ then they are capable of wonderful thing by “TURNING THE ENTIRE ROOM ON” – you can’t not accomplish it with better speakers or better amplifiers. Let dive slightly in the dangers territory. Driving your room by loudspeakers from outside AEZ location is like pleasing different parts of female body – it gives positive and effective atomic (individual, isolated) effects to her gratification. However, but performing necessary actions on women G-spot is capable to create for some women not atomic but a FULL-BODY REACTION. The Melody Range lodging a room from AEZ is very much hits the G-spot of your listening room and allows you get a VERY different and very evolved result of the room loading that is unimaginable if you drive your room from outside of AEZ. I can assure you that if you are not a Moron™ and each time you heard any more or less interesting sound from any playbacks then the loudspeakers, in one way or other and in most of cases completely accidentally, were near the AEZ.

The very next actions will be removing all filters, impedance normalizes, resonators from your upperbass driver (and perhaps your lower midrange driver) and let your upperbass installed in the enclosure-topology of your choose to play full range. In the next my post I will continue describing the next steps… *

BTW, I forgot to mention one very important thing that might be very easily overlooked: the Imbedded Macro-Positioning Reality lives in three dimensions. It is not about the right, left, closer and further but also up and down. In fact the vertical positioning usually has much less tolerance then tolerance in other dimensions and I have seen situation when moving a speaker latterly 5 inches up of down quite hugely affected “room loading”. So, do not overlook the vertical positioning… *

Currently the speakers are 12.5 feet apart and I put my listing chair 10 feet apart. The Macondo is pointed just behind the shoulders – my regular configuration. *

it would be a good idea to separate LF section of two channels –Upper LF and Lower LF. Then the lower LF would go to the corner of the room where the bass is the best *

Bill, first reflection of your horn might be from the back wall but when you are talking about reflection you imply HF that propagates according to the rules of transverse waves. The LF waver act more like pressure (longitudinal) wave and they do not have definitive first reflection in your case. The first reflection is more applicable to the HF content of your horn. *

in bad position : 1) There is no proper tone
2) Horizontal imaging is less refined then I would like it to be.
3) Sound is not wet enough.
4) At high volumes room can’t dissipate HF
5) Sound does not have uniformed density and more reminds a dug and raked backyard.
6) Strenuous and laborious presentation.
7) Playback does not sound with the room but in the room. *

The room is larger and has consequentially longer reverberation time at HF. It is absolutely not treated in a typical sense of this word and has naked walls as now. The room does not sound “bright” but I think the HF reflections inject the HF noise into MF, producing HF mist that dilute the tone. I still do not want to treat the wall and ceiling with explicit treatment. *

If your channels, particularly above 200Hz, are time-misaligned then your playback sounds like shit. *

In fact, after the measurements, the bass in this room is not useful and I unfortunately have to recognize that I will not be able to use Melquiades SET to handle bass. *

It looks like everything suggests that I would need to get rid in my new room my LF line arrays and go for smaller dedicated ULF sections, made to fit to the specific of the given room. Here was what I predicted years back that it will be the next manifestation of my playback. (Amir: it means You may need design new bass channel in a room to have good response below 100hz) *

I might to use digital correction of open-end analog notching but only on lower bass channel. (Amir: Romy may use both Class D amplifier and Digital EQ for below 100hz) *

LF the digital EQ will flatten the response for a single listening location. It will do it with no problem. Will it result sonic improvement? This is more complicated question. Let pretend that EQ does not destroy sound by DSP in case it is digital and does not spin phase. Still, flat response is not the objective. I have seen in some rooms very un-flat response that did not affect listening. Some peaks and some dives in the response are fine – it all depends how wide, where, how they related to the rest of the room response and how they masked out. No one advocate running digital IQ to flat bass (it is not complex) but to use digital IQ discreetly, fixing the major problems I think would worth to explore. BTW, what you do is not much different – only you use resonating limp panels. I consider your way of doing the thing is preferable but you endure a lot of acoustic treatment in your room. If it is a dedicated or demo room then it is fine but I do not have an objective to have DEMO room, I am looking for to make it living room.

Frankly I think that my way to do the ROOM is more preferable: do not fine with room and do not treat the room but rather to design the accustom system around the specific room behavior. I admit that so far I do not have in my room success but I just started. I think the final result might use some of your limp methods but the EQ ONLY for LF doe not sound too absurdish as well. My problem not is not with EQ but with absent of ULF channel with which I would model some lower bass behavior. Again, the jury is out but I still would not put the treated room in the epicenter of attention. *

OK, I decided to stop today to do what I do as it does not feel like me anymore. For a month as I live in here and have the playback up my playback is a source of great pain and I do not like it. I am accustomed that I turn the thing on and USE it. I did many experiments and had weeks or month playback in bad shape but I know what is going on and I had no problem with it. Now, in new house I for some reasons constantly fight with playback and practical do not USE it as I would like to. The audio experiments I fine, I like them but today I convinced myself the unit I have a dictated midbass channel I will not get the sound I would like to get. So, what people do after they have a strike of revelation? Right, they begin to re-read my site.
So, I look at my comments in the Audio For Dummies section: do not pursue full-range without being ready, do not go for lower bass without being ready and I asked myself – what I am doing? *

The ASC tube traps do show it as phenomenal HF consumption tool but I absolutely do not want them to deal with my sub 1000Hz *

Tube traps are very powerful HF consumer and the need to be use VERY cautiously, it easy to over dry my room and it need some attention what and how to use the tube traps. *

Thinking about tube traps more I conceded that they are no properly designed. *

ASC Tube Traps , They kind if insert a very fine file of oil on surface of bass water, eating up all micro dymick of bass. *

This is very unpleasant feeling. I think the problem that tube traps has is that they are trying to do for bass and I think this is a mistake. The tube traps are super good HF consumers and this is where they need to be. *

The perfect location of my listening chair had found, this is a huge move forward to me. Now I need lock everything on this location and bring the Macondo to the proper calibrated level with respect to this listening location. *

There are however some constrains. For instance I would like to keep Macondo islands as far as possible from back wall. I would like to have the Macondo islands to be spread as wide as possible but without deformation of center image. I would like the chair to be equidistant between crossover altered midbass and the rest MF channels. There are many other constrains and wishes. Sure putting up room treatment and by other means the critical listening position might be adjusted and it will be adjusted but the main skeleton will be pretty much remain. *

For those who both chair and speaker I would advise to find one stationary objects. For me in most of the case it is the back wall of the speaker when I know how far my speaker needs to be from there. This would give to you the base line for the speaker. Everything comes from there. You know your base line, you know your spread of the R/L channels, so you know more or less your listening distance. In many way it is back and forth ceremony…. *

It is well known that the requirements to acoustic environment for life music and for audio installations are very different. For life music we need much more what we would call in audio “live” acoustic setting, letting the instruments to breathe. For audio we need much more controlled environment, with much- much shorter reverberation time. What we in audio consider “too live” room would be “super dead” even for a string trio and in the environment those musicians would consider live enough for them audio out would hardly be able to operate as it will be too life for us. *

However, not a lot of people know that listening room compress dynamic. The compression is less visible and not as much “in your face” as the imaging. The irony is that the settings for best imaging most frequency (in context of conventional box loudspeakers) directly contradict the best settings for the least dynamic compression. *

The DpoLS is the ONLY one setting where there is no conflict between the dynamic and imaging. *

Interesting that the search for the least dynamic compression eventually do leads to the DPoLS, and whan the loudspeakers are in the DPoLS position then all aspect of “imaging” get resolved at orders of magnitude more interesting level then the people practicing the “imaging positioning” even could imagine. *

One more tip. The “least dynamic compression positioning” mostly managed by upper bass and bass channels while in the “imaging positioning” the MF and HF channels play more dominating roles. Therefore, if you use a typical single box loudspeaker then you most likely have quite few tools to manage the situation as the different channels would most likely demand the different optimum positions in your room (unless you are incredibly lucky!!!). Still, even is you do have a separation of the enclosures between the channels then it might be quite complex to take care of the LF’s “dynamic compression” the MF’s “imaging”, the phase consistency and many other this… at the same time and within the very same installation…. *

Would a properly, DPoLS-based, playback installation enrich a listener ability “to be lost in music”? Unquestionably would. How a listener without knowing the expressive methods of audio might bring up his/her system up to the DpoLS level? To do it requires a ceremony of connected sensations, actions and motivations… and this I try to make someone to think about. All that I was saying was that by perusing the “imaging minded” ceremony it is hardly possible to produce decisions leading to the actions that might produces any DPoLS-fruitful results. Contrary to this the training to recognize the dynamic compression of listening rooms take the DpoLS searching skills through the roof. *

The DpoLS is not a “newfound musical significance” but the only existing ways to get the “real audio sound”. *

The DPOLS is an art of the speakers setting. *

At the level #1 I concern about the sonic performance of the playback. Imaging, soundstage, separations, presentations, tonal balances, stereo tricks, sizes, deferent type of dynamics, relationship with room, dynamic imaging with volume fluctuation and the rest of the typical audio routine. At this level the calculators, measuring tapes and high resolution RTA really help. At this level it is possible to get a very good hi-fi sound. The precision of the DPLOS proximity I would say around 4”-15”

At the level #2 I look for the relationship between the audible and sensible sensation and the relationship between the tonal pressure and acoustic dB pressure. The DPLOS proximity I at this level around 1.5”-4”. At this level a new listing awareness is born (including the organic “phantom” sensations that you very accurately described) and the each characteristics of the level #1 get magnified and improved.

At the level #3 I look for the preservation of all that was accomplished of the level #2 but in addition I look for the amplitude of produced intentions. If music calls for thinking, sorrow, joy, melancholy or pomposity then it should be very extreme thinking, sorrow, joy, melancholy or pomposity. At this level the quality of the composition or performance become prominent and better performances should yield higher listing amplitude. At this level the playback system should AMPLIFY not sound but the musicality of a performance. Interestingly that when a system is made up to operate properly at the level #3 then some qualities of the level #1 and level #2 do over the roof. For instance, a listening perception get a possibility to accommodate itself to any aspect of Sound (for instance any single instrument, or any single phrase) and to abstract the selected “item” out of everything. At the level #3 the precision of the DPLOS proximity (if everything else was made correctly) would be less than 1”.

At the level #4 all bets are off and the precision of the DPLOS proximity is around 1/16”… and most like at the different location then it was the previous levels. 🙂 At this level the dynamics, imaging, soundstage, separations, presentations, tonal balances, and the rest things from level #1 return back not now, they have totally different meaning as they become connected to the physical experiences of a listener. When a listening awareness operates at the level of RECOMPOSING or RE-PERFORMING then the synchronization between the cerebral processes of a listener with the “heard sound reproduction” becomes an important expressive tool. There are many other things that are going on at this level and how the DPLOS proximity might affect it. I try do not share them publicly in order do not feed the reviewers and other industry dirt from stealing the evaluation points and then, using them for their primitive objectives while having no comprehending what it all might means (there was a few occurrences).

I also generally keep these thoughts to myself, particularly about the level #3 and level #4, because there are not a lot of people who might understand it (and are some other reasons…) *

جيم اسميت ميگه كف اتاق حتما حتما چوب يا پاركت باشه چون صدا خيلي موزيكال تره. حتما چوب ساليد و محكم بچسبه به كف. ميگه هيچ دستگاه بهتري نميتونه چنين تاثير مثبتي روي توناليته بزاره.

تو گوشه هاي ٩٠ درجه كه كنج هست ميگه بيس ترپ و اكوستيك نگذاريد و كتابخانه مثلثي استفاده كنيد.

اگر محل نشستن به ديوار پشت نزديكه حتما بايد اولين رفلكت بلندگو كه از طريق ديوار پشت سر شنونده به گوش تابيده ميشه كاملا با مواد جاذب گرفته بشه.

جيم ميگه براي اتاق هاي داخل خانه كه بدليل وجود فرش و مبل به اندازه كافي جذب صدا دارند ميشه از درختچه و گياه در اولين رفلكت ها استفاده كرد.
جيم با استفاده زياد از مواد جاذب موافق نيست.

مهمترين قسمتهايي كه در اتاق بايد كنترل شوند

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Comparison by contrast

جمعه 27 آوریل 2018
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من بيشتر از قبل به مساله تست از طريق تمايز خواهم پرداخت چون از نظر من مساله مهمي هست.

همين الان شما ميتونيد يه تست جالب انجام بديد ، پولاريتي سيگنال رو عوض كنيد ببينيد چقدر با حالت قبلي تفاوت داره و اين كار رو هم ميتونيد با كامپيوتر انجام بديد و هم راه ساده تر اينه مثبت منفي كابل بلندگو رو برعكس بزنيد تا پولاريتي عوض بشه.

هر چقدر اختلاف بين دو حالت بيشتر باشه يعني دستگاه شما با متد Comparison by contrast وضعيت بهتري داره.

تو حالت پولاريتي درست صدا خيلي ريلكس تر حالت نفس زدن بهتر بيس بسيار نرم تر و شفاف تر و راحت تر ميدرنج حسي تر و جذاب تر و هاي فركانس چسبيده به ميد داره و كل سه ناحيه فركانسي صدا يك صداست (مقاله قبلي رو ببينيد در مورد اكستندد ميدرنج) و حالت جدا جدا نداره اما تو حالت پولاريته اشتباه صدا انگار ايستاده و سفت و فلت ميشه و حالت نرم و ريلكس رو از دست ميده و ممكنه شنونده مبتدي اين حالت خشك و ايستاده رو به حساب دقيق تر شدن بزاره كه اشتباهست و مجموعا صدا مجزا شنيده ميشه و ما در جريان و فلو صدا حالت ملايم و به هم پيوسته نداريم و فركانسهاي پايين و بالا جدا از ميدرنج شنيده ميشن.

حالا حرف من اينه اگر تو سيستم شما با تغيير پولاريته تفاوت بشدت احساس نشه يعني دستگاه شما نميتونه اطلاعات صدا رو خوب انتقال بده و يا بلندگوي شما تو نقطه مناسبي جاگذاري نشده .

از من اينو بشنويد هرجا براي شنيدن يه سيستم رفتيد هر تستي خواستيد اول انجام بديد اما اخرش اين تستي كه ميگم رو حتما انجام بديد اونم اينكه يه سي دي با ركورد AAD مثل همين دايانا كرال ببريد و سي دي كپي (و يا ام پي ٣) همون البوم رو هم ببريد و ببينيد اون سيستم فرق بين كپي و اصل رو با چه ميزان تمايز نشون ميده.

اگر سيستم نتونه تمايز بين كپي با اصل رو با وضوح بالايي نشون بده يعني اون سيستم اطلاعات رو فيلتر ميكنه و نمره پاييني مي گيره.

اگر حتي امكان كپي كردن سي دي (يا تهيه ام پي تري) هم نبود از صاحب دستگاه بخواهيد پولاريتي رو براي شما تغيير بده تا صدا رو تو هر دو حالت بشنويد . سيستمي كه با تغيير پولاريته خيلي تفاوتي رو نشون نده اون سيستم سيستم شفافي نيست. از اين ساده تر نميتونم توضيح بدم واقعا روش بسيار راحتي رو به شما پيشنهاد دادم .

با اين متد شما يك پله جلوتر خواهيد بود به نسبت تست با روش A و B .

من ميدونم شنونده هاي حرفه اي با هر دو روش راحت تشخيص ميدن كدوم بهتره اما شنونده هاي معمولي بهتره روش دوم رو حتما در كنار روش اول انجام بدهند.

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Audio is Expensive but …

چهارشنبه 25 آوریل 2018
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آيا ميشه راه حلي پيدا كرد كه ما بدون صرف هزينه بسيار به اون صدايي كه دوست داريم برسيم؟

سوال سختيه والبته يكم نياز به توضيح داره تا از حالت كلي دربياد

ما برندهاي گران زيادي داريم و البته بيشتر اونها به نسبت قيمت ارزش زيادي ندارند اما برندهايي هستند كه از نظر من در عين گران بودن ارزشمندند.

براي فرار از پرداخت هزينه هاي يك دستگاه بسيار گران يك راه حل DIY هست و راه حل ديگه پيدا كردن برندهايي كه با قيمت پايين تر صداي فوق العاده اي ميدهند.

راه حل DIY تا يه سطحي جواب ميده مثلا شما يه صداي معمولي خوب بخواهيد براي رقابت با دستگاه هايي تو رنج قيمت هاي زير ١٠ هزار دلار اما من اصلا اعتقادي به اينكه با DIY بتونيم در سطوح خيلي بالا High End صداي فوق العاده اي بگيريم ندارم.

نمونه موفق حتي در سطوح بالا داريم مثل بلندگوي رومي (هزينه ساخت اون بلندگو زير ٤٠ هزار دلار نيست) اما اگر مساله احتمالات رو درنظر بگيريم حقيقت اينه اين احتمال در اين مورد خيلي كمه و رفتن به سطوح بالاتر قيمتي هم ريسك داره هم نتيجه با احتمال كمي خوب خواهد بود و من اصلا فكر نميكنم با DIY ميشه سراغ high End رفت و حداقل خودم چنين ريسكي نميكنم.

توجه داشته باشيد DIY بسيار بسيار وقت و انرژي ميبره و سعي و خطاهاي بسياري داره كه مجموعا براي من راه حل جذابي نيست .

راه حل دوم اينه ببينيم تو دنيا چه برندهايي با قيمت هاي كمتر صداي بهتري ميدهند.

در مقايسه با راه حل DIY من با راه حل دوم بسيار موافقم

براي دوستاني كه هزينه هاي هاي فاي براشون توجيه نداره و همچنان علاقه مند به داشتن دستگاه هستند من پيشنهادم اينه دو مساله رو درنظر بگيرند :
اول اينكه بايد اين برندهايي كه با قيمت كم صداي بسيار خوبي ميدهند رو پيدا كنند.
دوم اينكه وارد فضاي سعي و خطا نشوند و قبل از خريد به اندازه كافي وقت و انرژي قرار بدهند تا به شكل درستي انتخاب كنند چون هزينه سعي و خطا بسيار بسيار زياد هست.
بهترين راه هم استفاده از تجربيات كساني هست كه بهشون اعتماد داريم و دوري كردن از مجلات و نظرات ضعيف اينترنتي.
سوم اينكه بايد وقت و انرژي بيشتري براي تيون و جاي بلندگو بگذارند.

اما آيا راه حل بالا براي كساني كه high end يعني اون اخرين مرحله رو ميخوان ممكنه،
منظورم اينه اگر مثلا شما از صداي هورن ليوينگ وويس با كوندو (مجموعا ٢ ميليون دلار) لذت برديد آيا راهي براي ندادن اون هزينه خواهيد داشت ؟
منم به اين سوال فكر ميكنم و نتيجه اي تاحالا نگرفتم و فكر ميكنم راهي وجود نداره .

متاسفانه اگر CEC اگر Kondo Audio Note اگر هورن Living Voice اگر گرام مايكروسيكي بهترين هستند تجربه به من ميگه راهي براي جايگزيني اينها با برندهاي ارزانتر وجود نداره .

اما خبر اميدواركننده براي خودم و براي شما اينه با يه ست معمولي خوب اگر جاي بلندگو تو نزديكاي DPOLS پيدا بشه و سورس صدا انالوگ باشه و برق پيورپاور باشه ما به اون لذت شنيداري بسيار زياد ميرسيم بدون اينكه دو ميليون دلار بديم.

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Bass/Mid/High vs Extended Midrange

سه‌شنبه 24 آوریل 2018
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I got today an email with description of a playback sound that made me to think that the author linguisticly formed and nailed down one aspect of sound that I very long looking myself and even practicing but never was able to put it into the words. I think he did it very nice and very thoughts provoking. Here are his words, reprinted without permission:

“The first thing that impressed me was the bass, or really the lack of bass, you don’t hear bass you don’t hear highs there is only midrange, … I was expecting to listen to a bunch of concepts like the air between the instruments or the midbass slam or the transparency of the highs, there is none, there is only seamless extended midrange, but lets call it plainly, there are only instruments playing, and a bunch of them! “

A single sentence but it very loaded and it encompass years and year of observation about sound. What the sentence introduces is very tangible characteristic of sound of a playback – the bandwidth of Midrange. I sound oxymoronic – how a Midrange might have a different bandwidth? What we call Midrange is defined but there is a lot of “buts”.

The key in the sentence above the describer “Seamless extended midrange” which implies that a playback must be of course full-range but the wider bandwidth is perceived as Midrange the better sound is. I mean if the system has extended bass and extended HF but they sound as Midrange then something in the system is done properly. The irony is that live sound has no HF and no LF – it has ONLY Midrange but the bandwidth of this Midrange and the amount of information in this Midrange is enormous. The reproduced sound very frequently gets broken down to LF, Midrange and HF that is in a way a surrogate of sound reproduction. So, the rule is:

The wider Midrange bandwidth is without trained listener experience deficiency in upper and lower octaves the more pregnant reproduced Sound is.

I think I need to start to review from this perspective the performance of the orchestral conductors and musicants…

Rgs, Romy the caT

یه دنیا حرف اینجاست ، برای همینه یه 2way خوب که درست Place شده برای من ۱۰۰۰ بار بهتر از 3way هست وقتی نشه از 3way صدای درستی بگیریم.

این متن خیلی با ارزش هست.

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فركانس هايي كه تو بلندگو با اكوستيك بيشتر درگيرند

سه‌شنبه 24 آوریل 2018
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از ٣٠ تا ١٠٠ هرتز كه خود اتاق ميتونه نقاط رزونانس داشته باشه و شايد بقدري كار با اتاق سخت بشه كه مجبور بشيد زير صد هرتز رو با اكولايزر ديجيتال (فقط تو همين محدوده فركانس و نه بالاتر) تغيير بديد يا تغييراتي تو خود اكوستيك بوجود بياريد.
اين قسمت فركانس بشدت تابع رفتار اتاق هست و اگر كسي خيلي بره رو بيس اكستريم طبيعتا كارش مشكل ميشه. تو اين قسمت جابجايي بلندگو يكم كمك ميكنه ولي راه حل صد درصد نيست.

از ١٠٠ هرتز تا ٥٠٠ هرتز كه از نظر جيم اسميت و رومي مهمترين قسمت براي حس خوب موزيك همين محدوده فركانس هست و با جابجايي بلندگو ميشه به نقطه اي رسيد كه ما در اون فركانسها بيشترين گين (مثلا ٥ تا ٦ دي بي بيشتر) رو تو بازه فركانسي حتي ١/٥ اكتاو داشته باشيم.
رومي معتقده با قراردادن بلندگو در چنين جايي در اتاق شما در محدوده ماكرو قرار مي گيريد و ميتونيد تو اون منطقه دنبال نقطه بسيار بسيار اپتيمم بگرديد.

از ٥٠٠ هرتز تا ٥ كيلوهرتز و بالاتر هم بر اساس جاي بلندگو و اكوستيك كردن اتاق بصورت استفاده زياد از ديفيوزرها و در حد كمي هم استفاده از جاذب اكوستيكي ميشه به نتايج خوبي رسيد

بنابراين ما با سه بخش سروكار داريم كه هركدوم شرايط خودش رو دارند

رومي معتقده شما اصلا از بلندگو نبايد صدا بگيريد به اين معني كه بلندگو فقط يه تريگر هست كه بايد تو جاي خوبي قرار بگيره تا اتاق رو تحريك كنه و شما بايد صداي اتاق رو بشنويد . در اينحالت ما همه جاي اتاق صداي خوبي ميشنويم و بلندگو محو ميشه تو اتاق.

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USB to SPDIF Converter Legato Art

دوشنبه 23 آوریل 2018
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اين دستگاه ٨٠٠ دلار هست و يو اس بي ميگيره و spdif تحويل ميده فقط و فقط خروجي 16bit فركانس 44.1khz داره.

با ترانس ديتاي خروجي رو ايزوله ميكنه و امپدانس دقيقي داره تا اون فركانسها درست منتقل بشه و در ضمن جيتر بسيار كمي داره.

گزينه جالبي هست و مورد تاييد و پيشنهاد Gordon Rankin . اسم مهندس اين دستگاه pat هست . قيمتش يك سوم تا نصف قيمت Berkeley Alpha USB هست.

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تست دستگاه با روش Contrast

چهارشنبه 28 مارس 2018
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The old method: comparison by reference

We should begin by examining the method in current favour: The usual procedure is to use one or more favoured recordings and, playing slices of them on two different systems (or the same system alternating two components, which amounts to the same thing); and then deciding which system (or component) you like better, or which one more closely matches your belief about some internalized reference, or which one “tells you more” about the music on the recording. It won’t work! … not event if you use a dozen recordings of presumed pedigree … not even if you compare the stage size frequency range, transient response, tonal correctness, instrument placement, clarity of test, etc. – not even if you compare your memory of your emotional response with one system to that of another – it makes little difference. The practical result will be the same: What you will learn is which system (or component) more closely matches your prejudice about the way a given recording ought to sound. And since neither the recordings nor the components we use are accurate to begin with, then this method cannot tell us which system is more accurate! It is methodological treason to evaluate something for accuracy against a reference with tools which are inaccurate – not least of which is our memory of acoustical data. Therefore, it is very likely to the point of certainty that a positive response to a system using this method is the result of a pleasing complementarity between recording, playback system, experience, memory, and expectation; all of which is very unlikely to be duplicated due to the extraordinarily wide variation which exists in recording method and manufacture. (Ask yourself, when you come across a component or system which plays many of your “reference” recordings well, if it also plays all your recordings well. The answer is probably “no;” and the explanation we usually offer puts the blame on the other recordings, not the playback system. And, no, we’re not going to argue that all recordings are good; but that all recordings are much better than you have let yourself believe).

Recognising that many will consider these statements as audiophile heresy; we urge you to keep in mind our mutual objective: to prevent boredom and frustration, and to keep our interest in upgrading our playback system enjoyable and on track. To this end it becomes necessary that we lay aside our need to have verified in our methodology beliefs about the way our recordings and playback systems ought to sound. As we shall see, marriage to such beliefs practically guarantees us passage to AUDIO HELL. It is our contention that, while nothing in the recording or playback chain is accurate, accuracy is the only worthwhile objective; for when playback is as accurate as possible, the chances for maximum recovery of the recorded program is greatest; and when we have as much of that recording to hand – or to ear – then we have the greatest chance for an intimate experience with the recorded performance. It only remains to describe a methodology which improves that likelihood. (This follows shortly).

Listeners claiming an inside track by virtue of having attended the recording session are really responding to other, perhaps unconscious, clues when they report significant similarities between recording session and playback. As previously asserted, no-one can possibly know in any meaningful way what is on the master tape or the resulting software, even if they auditioned the playback through the engineer’s “reference” monitoring system. Anyone who thinks that there exists some “reference” playback system that sounds just like the live event simply isn’t paying attention; or at best doesn’t understand how magic works. After all, if it weren’t for the power of suggestion, hi-fi would have been denounced decades ago as a fraud. Remember those experiments put on by various hi-fi promoters in the fifties in which most of the audience “thought” they were listening to a live performance until the drawing of the curtain revealed the Wizard up to his usual tricks. The truth is the audience “thought” no such thing; they merely went along for the ride without giving what they were hearing any critical thought at all. It is the nature of our psychology to believe what we see and to “hear” what we expect to hear. Only cynics and paranoids point out fallibility when everyone else is having a good time.

Another relevant misunderstanding involves the correct function of “monitoring equipment”. The purpose of such equipment is to get an idea of how whatever is being recorded will play back on a known system and then to make adjustments in recording procedure. It should never be understood by either the recording producer or the buyer that the monitoring system is either definitive or accurate, even though the engineer makes all sorts of placement and equipment decisions based on what their monitoring playback reveals. They have to use something, after all; and the best recording companies go to great lengths to make use of monitoring equipment that tells them as much as possible about what they are doing. But no matter what monitoring components are used, they can never be the last word on the subject; and it is entirely possible to achieve more realistic results with a totally different playback system, for example, a more accurate one. Notice “more accurate,” not “accurate.” It bears repeating that there is no such thing as an accurate system, nor an accurate component, nor an accurate recording. Yet as axiomatic as any audiophile believes these assertions to be, they are instantly forgotten the moment we begin a critical audition.

The proposed method: Comparison by contrast.

When auditioning only two playback systems using the usual method, we will have at least a 50% chance of choosing the one which is more accurate. However, evaluations of single components willy-nilly test the entire playback chain; therefore efforts to choose the more accurate component are compounded by the likelihood that we will be equally uncertain as to the accuracy of each of the system’s associated components if for no other reason than that they were chosen by a method which only guarantees prejudice. How can we have any confidence that having chosen one component by such a method that its presence in the system won’t mislead us when evaluating other components in the playback chain, present or future?

The way to sort out which system or component is more accurate is to invert the test. Instead of comparing a handful of recordings – presumed to be definitive – on two different systems to determine which one coincides with our present feeling about the way that music ought to sound, play a larger number of recordings of vastly different styles and recording technique on two different systems to hear which system reveals more differences between the recordings. This is a procedure which anyone with ears can make use of, but requires letting go of some of our favoured practices and prejudices.

In more detail, it would go something like this: Line up about two dozen recordings of different kinds of music – pop vocal, orchestral, jazz, chamber music, folk, rock, opera, piano – music you like, but recordings of which you are unfamiliar. (It is very important to avoid your favourite “test” recordings, presuming that they will tell you what you need to know about some performance parameter or other, because doing so will likely only serve to confirm or deny an expectation based on prior “performances” you have heard on other systems or components. More later.) First with one system and then the other, play through complete numbers from all of these in one sitting. (The two systems may be entirely different or have only one variable such as cables, amplifier, or speaker).

The more accurate system is the one which reproduces more differences – more contrast between the various program sources.

To suggest a simplified example, imagine a 1940’s wind-up phonograph playing recordings of Al Jolson singing “Swanee” and The Philadelphia Orchestra playing Beethoven. The playback from these recordings will sound more alike than LP versions of these very recordings played back through a reasonably good modern audio system. Correct? What we’re after is a playback system which maximizes those differences. Some orchestral recordings, for example, will present stages beyond the confines of the speaker borders, others tend to gather between the speakers; some will seem to articulate instruments in space; others present them in a mass as if perceived from a balcony; some will present the winds recessed deep into the orchestra; others up front; some will overwhelm us with a bass drum of tremendous power; others barely distinguish between the character of timpani and bass drum. In respect to our critical evaluation process, it is of absolutely no consequence that these differences may have resulted from performing style or recording methodology and manufacture, or that they may have completely misrepresented the actual live event. Therefore, when comparing two speaker systems, it would be a mistake to assume that the one which always presents a gigantic stage well beyond the confines of the speakers, for example, is more accurate. You might like – even prefer – what the system does to staging, but the other speaker, because it is realizing differences between recordings, is very likely more accurate; and in respect to all the other variables from recording to recording, may turn out to be more revealing of the performance.

Some pop vocal recordings present us with resonant voices, others dry; some as part of the instrumental texture, others envelope us leaving the accompanying instruments and vocals well in the background; some are nasal, some gravelly, some metallic, others warm. The “Comparison by Reference” method would have us respond positively to that playback system, together with the associated “reference” recording, that achieves a pre-conceived notion of how the vocal is presented and how it sounds in relation to the instruments in regard to such parameters as relative size, shape, level, weight, definition, et al. Over time, we find ourselves preferring a particular presentation of pop vocal (or orchestral balance, or rock thwack, or jazz intimacy, or piano percussiveness – you name it) and infer a correctness when approximated by certain recordings. We then compound our mistake by raising these recordings to reference status (pace Prof. Johnson), and then seek this “correct” presentation from every system we later evaluate; and if it isn’t there, we are likely to dismiss that system as incorrect. The problem is that since neither recording nor playback system was accurate to begin with, the expectation that later systems should comply is dangerous. In fact, if their presentations are consistently similar, then they must be inaccurate by definition simply because either by default or intention no two recordings are exactly similar. And while there are other important criteria which any satisfactory audio component or system must satisfy – absence of fatigue being one of the most essential – very little is not subsumed by the new method of comparison offered here.

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