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| [#479] Benchmark DAC2 Fans Club http://hometheaterhifi.com/reviews/dac/benchmark-dac2-dx-review-updated-polished-gem-past/ "Conclusions THE BENCHMARK DAC2 DX had some of the lowest distortion of any DAC we have ever tested." |
batmanamesIA 正式會員209.xxx.xxx.64 |
2016-02-04 23:33 | |
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| [#478] Benchmark DAC2 Fans Club http://audioxpress.com/article/Fresh-from-the-Bench-Benchmark-DAC2-DX-Stereo-D-A-Converter.html Fresh from the Bench: Benchmark DAC2 DX Stereo D/A Converter July 22, 2015by Gary Galo The DAC2 series are the latest digital-to-analog converters from Benchmark Media Systems, the American audio manufacturer based in Syracuse, NY. There are three current models, each incorporating the same high-performance digital and analog circuitry, but offering different features and I/O configurations. The DAC2 HGC and the DAC2 L were designed for users who also have analog sources—both include two sets of line-level analog inputs, and have features useful in home theater installations. The HGC version has a headphone amplifier. The L version does not. Operating Buttons The model reviewed here is the DAC2 DX, which has been tailored for users who don’t have analog sources or who use a separate analog preamp. The DAC2 DX has six digital inputs—two S/PDIF coaxial, two TOSlink optical, one USB, and is the only DAC in the series that also has an AES/EBU input. The DAC2 DX has two output buses—Main and Aux. The Main output bus feeds one left/right set of RCA outputs and the XLR balanced outputs; the Aux bus feeds the other RCA outputs. Each bus can be separately configured as a variable output controlled by the front-panel volume control, or as a fixed, calibrated output. The Mute and Dim buttons on the front panel are normally used to completely mute all outputs, or reduce the output volume levels by 20 dB. But, if the Mute button is held for 10 seconds, the Main output bus goes into the fixed calibrated mode, bypassing the front-panel volume control. The Dim button functions the same way for the Aux output bus—illumination of the M and A LEDs indicate that those buses are now in the Calibrated mode. This is the preferred mode if the DAC2 DX is used with a separate preamp; the need to hold the Mute and Dim buttons for 10 seconds ensures that the Calibrated mode can’t be selected accidentally. The Power button on the front panel toggles the DAC2 DX on and off, and two additional buttons cycle the unit through the various digital inputs. LEDs corresponding to the six digital inputs indicate which option has been selected. A steady light indicates a proper digital lock, and a flashing light indicates an improper digital input or the absence of one. Four LEDs indicate the input sampling frequency—they are labeled 44, 48, 2X, and 4X and accommodate all four standard pulse code modulation (PCM) sampling rates from the 44.1-kHz CD standard through 192 kHz. Two additional LEDs indicate the incoming word length, either 16 or 24 bits. The DAC2 DX also accepts Direct-Stream Digital (DSD) data, indicated by illumination of both the 2X and the 4X LEDs. Two headphone jacks are also located on the front panel. In addition to the digital input and analog output connectors, the rear panel also has a 3.5-mm 12-V trigger I/O jack. This jack can either be used to power up the DAC2 DX from an external 12-V trigger, or it can supply a 12-V trigger to other pieces of equipment when the power is turned on with the front-panel button. Other Benchmark products, including the AHB2 power amplifier, use this same system, so they can be turned on using a trigger voltage supplied by the DAC2 DX, or vice versa. Photo 1: The DAC2 DX’s front view has an LED display that indicates sampling frequency, word length, selected digital input, and whether the Main and Aux outputs are in the Calibrated mode. The volume knobs control the output level in the digital domain, using the ES9018 DAC’s internal 32-bit volume control. A silver front panel is also available. New Design The DAC2 series has replaced the older DAC1 products, and the changes are not mere functional upgrades. The DAC2 is an entirely new design reflecting significant advances in digital hardware and Benchmark’s formidable design expertise. The new features included in the DAC2 converters include: - Native DSD Conversion - Asynchronous USB supporting V. 1.1 and V. 2.0 - Volume-control bypass - S/PDIF digital pass-through - Word-length display - Sample-rate display - Bi-directional 12-V trigger - More flexible I/O connectivity The USB input supports the 1.1 standard of 96 kHz/24 bits without any driver. The Multi-Mode Asynchronous 2.0 standard supports PCM at up to 192 kHz/24-bit, and DSD following the DoP 1.1 standard. A driver is required for this mode on Windows computers, and is downloadable from the Benchmark website; no driver is needed for Mac OS X. DoP 1.1 is a method for transmitting DSD data streams over an S/PDIF interface, and has been adopted by a number of manufacturers of digital audio servers. The S/PDIF and AES/EBU inputs also accept DSD over DoP 1.1, along with PCM up to 192 kHz/24-bit. The TOSlink optical input supports only PCM, up to 96 kHz/24-bit. The DAC1 incorporated a superb jitter-suppression system that Benchmark called UltraLock. This has been upgraded to UltraLock2 in the DAC2 series. All DAC2 converters support the Red Book de-emphasis standard, which is important to those of us who have early CDs with pre-emphasis. Photo 2: The DAC2 DX’s rear panel shows the balanced and unbalanced analog output, the six digital inputs, 12-V trigger I/O, and the IEC power connector. The DX version is the only DAC2 that includes an AES/EBU digital input. Hardware Details In the DAC1 USB all the digital inputs were routed through AKM Semiconductor’s AKM4114 Digital Audio Interface, and from there to Analog Devices’ AD1896 High-Performance Asynchronous Sample Rate Converter via the I2S bus. In the DAC2 converters these two chips are replaced with a single Texas Instruments/Burr-Brown SRC4392 High-End Combo Sample Rate Converter. Quoting from the manufacturer’s description: “The SRC4392 is a highly-integrated CMOS device designed for use in professional and broadcast digital audio systems. The SRC4392 combines a high-performance, two-channel, asynchronous sample rate converter (SRC) with a digital audio interface receiver (DIR) and transmitter (DIT), two audio serial ports, and flexible distribution logic for interconnection of the function block data and clocks. The DIR and DIT are compatible with the AES3, S/PDIF, IEC 60958, and EIAJ CP-1201 interface.” This chip supports up to eight digital inputs and represents the state-of-the-art in sample rate conversion. It also forms the basis for Benchmark’s UltraLock 2 jitter suppression, which delivers performance superior to two-stage PLL (phase-locked loop) systems. The DAC1 resampled all digital inputs to 110 kHz, but the DAC2 series upsamples all incoming data to 210.9375 kHz. John Siau, Benchmark’s Director of Engineering, explained: “This upsampling shifts the ESS interpolation filters out of the audio band, and we replace them with improved filters. The upsampling also provides near-perfect rejection of jitter on all of the digital inputs… jitter-induced sidebands never exceed -140 dBFS. This means that jitter-induced distortion is virtually non-existent and if any exists, it is well below audibility.” The heart of the DAC2 DX is ESS Technology’s ES9018 Sabre32 D/A converter chip, which continues to gain favor with digital converter manufacturers (this chip is also used in the OPPO BDP-95 and BDP-105 universal players). The ES9018 is an eight-channel, 32-bit device. Various manufacturers of converters and stand-alone players have their own approaches to making use of the multiple channels. Benchmark parallels three sections each for the left and right Main outputs. The remaining two channels are used for the left and right Aux outputs. Since the HGC and the L DACs don’t have separate Main and Aux buses, four sections of the ES9018 are paralleled for each channel in those versions. I asked Siau if having only three ES9018 sections in parallel reduced the performance of the DX version compared to the HGL and L version. He commented: “In theory, every doubling of the channel count improves the SNR by 3 dB. Summing two channels to one produces a maximum improvement of 3 dB. Summing four channels to one produces a maximum improvement of 6 dB. Summing three channels to one produces a maximum improvement of 4.8 dB. In theory the difference between summing four channels and summing three channels should be 1.2 dB. In reality the difference is much less than this because the noise of the I-V converters and output stages must be factored into the equation. We boosted the output level of the internal signal path by 2 dB to recover all of what was lost by summing three channels instead of four. For this reason, the SNR of the DAC2 DX is virtually identical to that of the DAC2 HGC.” Benchmark also uses the ES9018’s internal 32-bit volume control for output level adjustment. In the DAC2 DX a single potentiometer controls the output level entirely in the digital domain. Photo 3: The inside view of the DAC2 DX shows the main PC board. The shielded, switching-mode power supply is in the upper left. The ES9018 DAC chip is toward the bottom, right of center, and the LME49600 headphone amp buffers are just below the vertical section of the ribbon cable. No ES9018 Shortcuts Siau commented on the shortcuts that some manufacturer’s take when implementing the ES9018 DAC chip: “There are many ways to reduce the cost of using an ES9018 converter. The ES9018 support components can be significantly reduced relative to what can be seen in the DAC2. These cost-saving measures usually degrade the performance of the ES9018. Benchmark’s focus is always on achieving maximum performance, so we have not implemented any of these common cost-saving measures. Instead, we have gone the other way and have added significant refinements to the circuits suggested by ESS. Some of these enhancements are focused on improving the voltage reference input to the ES9018. We actively filter and buffer the reference voltage input, and control it with a precision regulator. This reduces distortion and noise, and provides a stable and predictable output level. Two LME49860 devices and a precision voltage reference are used for this task. This reference circuit also includes 16 additional passive components. Our entire reference circuit could have been replaced by a pair of resistors, but our goal was performance.” He added, “Many competing products omit the current-to-voltage converters, the differential amplifiers, the analog low-pass filters, and the voltage reference in order to reduce costs. Replacing the LME49860 op amps can also generate significant cost savings. For these reasons, there are huge variations in the performance of ES9018 equipped DACs. The ES9018 is the best D/A converter IC available, and Benchmark has worked hard to extract every bit of performance from the component.” USB I/O is performed by an SMSC USB3318 High-Speed USB transceiver. An XMOS programmable microcontroller controls the asynchronous USB functions (the XMOS chips are also used for asynchronous USB input functions by OPPO in its BDP-105 player, and by PS Audio in its NuWave, PerfectWave, and DirectStream DACs). Most of the digital signal processing is performed by a Xylinx Spartan 6-series field-programmable gate array (FPGA) chip. With any programmable chip, each manufacturer will normally tailor the programming to meet its own needs and performance specifications. Benchmark’s Multi-Mode Asynchronous Advanced USB Audio has been designed to eliminate jitter-induced distortion. As the owner’s manual explains: “The DAC2 has a low-jitter master clock which controls the transfer of audio data from the computer to the USB sub-system. The computer asynchronously transfers audio data to a buffer in the DAC2. The contents of the buffer are then asynchronously transferred to the D/A conversion sub-system. This second asynchronous transfer eliminates any traces of jitter that accumulate as the data is transferred between the USB and conversion subsystems. No traces of jitter-induced distortion are measurable to our measurement limits (better than -140 dBFS).” Benchmark claims that its USB implementation offers the convenience of computer playback without the usual compromises. Intersample Overs One issue that has been getting some attention lately is intersample clipping, also called “intersample overs.” When audio is sampled, it’s possible for the A/D conversion process to miss peaks that occur between samples. Those peaks are often more than 3 dB higher than 0 dBFS (dBFS means deciBels Full Scale, which is the maximum allowable digital recording level before clipping). When the audio is reconstructed in the D/A conversion process, the missing peaks result in clipping in the reconstructed waveform. Siau explained how Benchmark addresses this often-overlooked problem: “We scale the input signal down by 3.5 dB before applying any filtering or sample rate conversion processes. This prevents clipping in the SRC, and the ES9018. Normally, intersample overs cause overflow problems in fixed-point SRC devices and this normally gives them an undesirable sonic signature. Our technique makes the SRC process transparent.” “We also feed the ESS DAC chip at –3.5 dB. The ES9018 will clip in response to intersample overs. Our technique entirely prevents this clipping. The only tradeoff is that our SNR is 3.5 dB lower than it would be without the extra DSP headroom. The DAC2 is capable of a SNR of 129.5 dB, but we ‘throw away’ 3.5 dB to avoid intersample clipping (yielding the published SNR of 126 dB, A-weighted). Sonically, this is a very good tradeoff. Intersample overs can reach 3.01 dB over 0 dBFS, so our 3.5 dB of headroom has us covered. It is important to understand that the clipping caused by intersample DSP overloads usually causes non-harmonic distortion that splatters across the entire audio spectrum. Avoiding digital signal processing overloads due to intersample overs is a really big deal.” Because of Benchmark’s low-noise voltage reference and I/V conversion circuits mentioned earlier, the DAC2 is much quieter than most other products using the ES9018, even with the 3.5 dB reduction in S/N required to prevent intersample overs. Benchmark’s website has two short articles by Siau that I highly recommend: “Audio That Goes to 11” and “Why Audio Goes to 11.” There’s also a link to a very amusing video that you should watch. The clip is from the 1984 film This is Spinal Tap, in which rock guitarist Nigel Tufnel (played by Christopher Guest) explains to director Marty DiBergi (played by Rob Reiner) how all of the controls on his “very special” Marshall amplifier are calibrated from 0 to 11, rather than the normal 0 to 10. The ability to “go to eleven” allows the amp to give a little “extra push” if you need it. (Even the tone controls “go to eleven;” after all, you would never want less of something—only more!). This is all very tongue-in-cheek, of course, but if 10 represents normal digital zero, intersample overs will create clipping in the conversion process. Benchmark’s technique enables the DAC2 to “go to eleven” without distortion. Siau also commented on the effect the sampling rate has on intersample overs: “Low sample rate recordings (CD and 48 kHz audio for video) are prone to having problems with intersample overs. In contrast, high sample rates rarely have intersample over problems because there is very little audio energy at the frequencies where the worst-case intersample overs occur. The worst case occurs at Fs/4. For a CD this is 11.025 kHz. There is often enough short-term energy near 11 kHz to create issues. At a 96 kHz sample rate, the worst case intersample over occurs at an audio frequency of 24 kHz (where there is much less audio energy). For this reason, high sample rates tend to significantly reduce the intersample over problem” Siau added, “One might suggest that upsampling 44.1 to 96 kHz (or higher) would solve the problem. Actually the reverse is true. Intersample overs are exposed to clipping while upsampling. When upsampling filters clip, the resulting distortion products splatter across the entire audio band. Given this information, one might suggest that 44.1 kHz should never be upsampled. Unfortunately, all modern D/A converters upsample. Modern oversampled 1-bit and multi-bit converters have brought many significant improvements to digital audio, but they have created an intersample clipping problem. Upsampling DSP processors can be designed so that they don’t clip. One solution is to use floating point math. Another solution is to reduce the signal level by at least 3.01 dB prior to processing. We have chosen the second option. The second option is the only option when an SRC or an upsampler has fixed-point PCM inputs and outputs. Our SRC and the ES9018 are both limited to fixed-point operation… The sin(x)/x reconstruction filters in the D/A will reconstruct the intersample peaks provided they fall within the Nyquist limit, and are not clipped by an upsampling DSP process.” In addition to the other sonic advantages, this is a good reason to make master recordings at high sampling rates, even if the finished product will be a Red Book CD. Analog Circuitry The analog circuitry in the DAC1 used TI NE5532 op-amps as I/V (current-to-voltage) converters, and Texas Instruments/National LM4562 op-amps elsewhere. The DAC2 uses Texas Instruments/National LME49860 chips throughout. I counted a total of 17 of these dual op-amps in the DAC2 DX. This high-performance audio IC features a slew rate of 20 V/µSec, gain bandwidth product of 55 MHz, very low voltage noise of 2.7 nV/√Hz, total harmonic distortion (THD) of 0.00003%, and will operate on up to ±22 V supply rails. The ES9018 has differential current outputs, and Benchmark feeds these to precision differential I/V converters to minimize common-mode distortion. The outputs of the differential amplifiers are AC-coupled with a corner frequency of 0.016 Hz. Siau mentioned that he prefers AC coupling to DC servos, and says that the keys to success with this approach are as follows: - Set the corner frequency very low (under 0.1 Hz) - Select good capacitors (Benchmark usually uses at least two different types of capacitors in parallel) - Manage leakage currents carefully Signal-routing relays with gold-plated contacts route the signals to the appropriate output buffers and/or the headphone amplifiers. Benchmark notes that the balanced and unbalanced outputs of the DAC2 have very similar performance. The headphone outputs are driven by a pair of TI LME49600 high-current audio buffers. Judging by the specifications and the simplified schematic diagram in the datasheet, this chip is similar to the Texas Instruments/Burr-Brown BUF634. It’s an open-loop, current-feedback device with a slew rate of 2,000 V/µSec, output current of 250 mA, bandwidth adjustable from 110 to 180 MHz, and is designed to be used with an op-amp inside the feedback loop. The LME49600 has been designed especially for high-performance audio applications with a THD of 0.00003%, and maximum DC supply voltages of ±20 V (the older BUF634 is limited to ±18 V rails and the THD is not specified for that device). The high output current of the LME49600 buffers enables headphones to be directly driven—no series output resistor is used, yielding a 0-ohm output impedance. Most headphone amplifiers have some resistance in series with the outputs, which is usually at least 10 Ω and often higher with circuits based on unbuffered op-amps. Since headphones don’t have flat impedances curves, the series output resistor alters the frequency response and degrades the damping factor. Benchmark’s design is the best possible solution for headphone amplification. As with all new Benchmark products, the DAC2 DX has a switching-mode power supply. Benchmark has found that the elimination of the power transformer’s hum field, combined with placing the switching frequency well above the audio band, results in lower noise than a linear supply. Benchmark also uses choke filtering at the output of the switching supply—there are eight chokes, four at 15 µH and four at 2.2 µH, used in conjunction with first-stage regulators. They also use generous local regulation—at least one dedicated, low-noise voltage regulator is used for each critical subsystem. Internal Jumpers The DAC2 cover is held in place with eight small Phillips machine screws and is easily removable, allowing access to several user-configurable jumpers on the PC board. To accommodate headphones with varying sensitivities, the headphone amp gain can be set to three levels: 0 dB, –10 dB, or –20 dB. The factory default is 0 dB. As shipped, the main output bus will mute when headphones are plugged into the left-hand jack. Jumpers enable the user to apply this to the right-hand jack, both, or neither. There’s no industry standard for gain and input sensitivity on power amplifiers. Ideally, a normal listening level should be achieved with the preamp’s volume control near the 12:00 position, but many power amps are designed with gain set too high to allow this. The XLR outputs on the DAC2 are equipped with low-impedance passive pads to reduce the output level when the unit is used with high-gain power amps. As shipped, the pads are disabled, which sets the output level at 0 dB. Jumpers enable the pads, allowing attenuation of –10 dB or –20 dB. At 0 dB output, the XLR jacks have an output impedance of 60 Ω (30 Ω for each leg). Setting the pad to –10 dB or –20 dB increases the output impedance to 425 Ω or 135 Ω, respectively. But Benchmark notes that the dynamic range is not affected. This will affect the maximum cable length that the DAC2 can drive without degrading high-frequency performance, but even with the worst-case 10 dB pad, cables up to 96’ can be used with only a 0.1 dB loss at 20 kHz. Two additional jumpers enable the Digital Pass-Through mode. This converts RCA connector D5 from a digital input to a buffered digital output. With the Digital Pass-Through mode enabled, the data from any selected digital input—whether USB, S/PDIF, AES-EBU, or TOSlink—is passed to connector D5. This can be especially handy for digital format conversion, or for connecting a second DAC2 as a slave. Benchmark notes that the Digital Pass-Through is bit-transparent—the data is actively buffered but is not rewritten or |
batmanamesIA 正式會員 184.xxx.xxx.50 |
2015-12-01 07:25 |
| [#477] Benchmark DAC2 Fans Club 向各師兄求救！ 買咗DAC2 D，從網上download driver : http://benchmarkmedia.com/pages/dac2-drivers 搞嚟搞去都駁唔通無聲，又唔識搞返去usb 1.1，搞到現在都未瞓，投哂降。 有無邊位師兄可以指點迷津？！ 唔該哂。 |
iloveu 正式會員 203.xxx.xxx.95 |
2015-06-15 00:46 |
| [#476] Benchmark DAC2 Fans Club C130 ching Yes. You can use p9_26 as the DSDOE of amanero. I have further update on this (but not related to Benmark DAC2) : http://www.hiendy.com/hififorum/forum.php?mod=viewthread&tid=79555&extra=page%3D1%26filter%3Dtypeid%26typeid%3D11&page=3 最後修改時間: 2015-03-12 13:00:32 |
steveha 正式會員 1.xxx.xxx.108 |
2015-03-12 12:47 |
| [#475] Benchmark DAC2 Fans Club 為何會2粒fuse 同時通電的? |
2011 正式會員 123.xxx.xxx.26 |
2015-02-25 20:19 |
| [#474] Benchmark DAC2 Fans Club 請教師兄 dac2,fuse 是否500ma,系咪即是o.5a ,長度是否20mm 謝謝 |
2011 正式會員 123.xxx.xxx.26 |
2015-02-25 19:17 |
| [#473] Benchmark DAC2 Fans Club Quoted from user manual: "HOME THEATER BYPASS The DAC2 HGC can be integrated into a home theater system. Typically the DAC2 HGC drives the left and right power amplifiers, and an analog feed from the home theater system is connected to either of the analog inputs on the DAC2 HGC. A programmable Home Theater Bypass function sets the pass-through gain to unity when the surround system is active. The DAC2 HGC includes an indicator light that is illuminated when bypass is active." |
jimmyihv 正式會員 223.xxx.xxx.95 |
2015-02-25 11:01 |
| [#472] Benchmark DAC2 Fans Club 師兄、請問后面有乜野制？有乜用？thx 最後修改時間: 2015-02-25 00:28:45 |
2011 正式會員 123.xxx.xxx.26 |
2015-02-25 00:15 |
| [#471] Benchmark DAC2 Fans Club 後面有個制 |
jimmyihv 正式會員 223.xxx.xxx.95 |
2015-02-24 23:08 |
| [#470] Benchmark DAC2 Fans Club 各位師兄: 恭喜發財 , 申請入會, 遲來的春天, |
2011 正式會員 123.xxx.xxx.26 |
2015-02-24 22:53 |
| [#469] Benchmark DAC2 Fans Club 即係冇得bypass |
lcw1010 正式會員 123.xxx.xxx.109 |
2015-02-16 12:51 |
| [#468] Benchmark DAC2 Fans Club turn the volume control knob to max level |
batmanamesIA 正式會員 209.xxx.xxx.91 |
2015-02-16 09:04 |
| [#467] Benchmark DAC2 Fans Club 請問各位ching dac2 怎樣校機才可以fix 電平作bypass去preamp? |
lcw1010 正式會員 182.xxx.xxx.105 |
2015-02-16 08:05 |
| [#466] Benchmark DAC2 Fans Club Steveha ching, 好勁! 想請教bbb行botic點simulate amanero dsdoe? 可否直接出p9_26? |
cl30 正式會員 66.xxx.xxx.228 |
2015-01-26 15:16 |
| [#465] Benchmark DAC2 Fans Club 唔好笑我啦!呢度好多師兄燒得仲勁,我呢啲初哥好皮毛咋! 最後修改時間: 2014-10-26 18:15:58 |
A5X 正式會員 202.xxx.xxx.14 |
2014-10-26 18:15 |
| [#464] Benchmark DAC2 Fans Club A5X兄,你燒得好勁!!1 |
jason4400 正式會員 61.xxx.xxx.211 |
2014-10-26 17:01 |
| [#463] Benchmark DAC2 Fans Club 各位師兄,想請問邊條電源線(筆直約5K左右)for Benchmark dac2 會比較理想,因小弟近日想揾條美式電源線俾部Benchmark用,煩請各師兄教路。謝謝各位! 最後修改時間: 2014-10-26 16:34:07 |
A5X 正式會員 202.xxx.xxx.21 |
2014-10-26 16:31 |
| [#462] Benchmark DAC2 Fans Club good information |
batmanamesIA 正式會員 209.xxx.xxx.91 |
2014-10-10 07:57 |
| [#461] Benchmark DAC2 Fans Club Now, I understood how to simulate DSDOE of Amanero Combo384 USB. He He. The same should apply to LKS Dac.  |
steveha 正式會員 1.xxx.xxx.114 |
2014-10-09 19:17 |
| [#460] Benchmark DAC2 Fans Club I2S 最高級玩法 - I have my BBB hacked into a Poppulse Twin 9018 DAC. The BBB simply replaced its original Amanero Combo384 USB module. All PCM, DXD, DSD64 and DSD128 are supported.  最後修改時間: 2014-10-09 19:12:27 |
steveha 正式會員 1.xxx.xxx.114 |
2014-10-09 19:10 |
