Buy asus sound card. Sound card ASUS Xonar DX

The Asus Essence One is the top product in the line of external audiophile sound cards. This is the first external consumer card, in which as many as 11 replaceable operational amplifiers can be changed for tuning the sound path. The card has professional XLR connectors for connecting home active studio monitors (more and more often purchased for PCs instead of standard Hi-Fi kits). Essence One can also be useful for musicians, since it has support for ASIO 2.0 and is capable of operating up to 192 kHz 24-bit with low latency via USB. Another interesting feature, not often found among external sound cards, Asus Essence One can work in the mode of an external DAC without connecting to a computer, positioning itself as an "external DAC". The analog path uses two PCM1795 chips, each enabled in mono mode and providing one balanced channel for record-breaking performance. In addition, there is a built-in upsampler operating at 352 and 384 kHz, outperforming competitors using only 192 kHz.

Asus is quite active in the audiophile market. At first the company actively pushed Creative among sound cards (Asus Xonar D2 release), and after the remaining “professional” cards like E-MU1212m, ESI [email protected] and some audiophile products like the Audiotrak HD2 with the Xonar Essence ST and Xonar Essence STX cards. New products, for example Onkyo SE-300, can compete with ST and STX, but only as another promising product, without any advantage over Asus, since the filling is generally similar, and the affordability for the buyer is much worse than that of Asus.

At the moment, the niche of internal cards is occupied rather tightly, but there is still a gap among external sound cards. The most striking example among external devices not from the world of professional cards is ESI Dr. DAC Prime. Prime replaced the Audiotrak Dr. DAC 2 and was not a particularly successful upgrade. On the one hand, the SRC function was added to the device, and on the other, the ability to adjust parameters using replaceable op-amps was reduced, replacing the headphone amplifier path with a soldered microcircuit. The device has risen in price, but something revolutionary has not appeared in it. The only thing that pleased me was the availability in our market. Asus Essence One is a logical continuation of Dr. DAC 2, where you can see the development of this ideology. According to this ideology, we have in front of us only a DAC, with the ability to reproduce a signal from digital inputs without connecting to a PC, and when connected via USB from a PC, work up to 96 kHz and 24 bit versus standard 48 kHz and 16 bit for the nearest external sound cards. So is the Asus Essence One today: while most cards and DACs only support 96 kHz 24-bit over USB 2.0, the Asus Essence One supports 192 kHz 24-bit. Instead of five replaceable op amps in Dr. DAC 2, 11 op amps can be changed in Essence One. Like Prime, Essence One has an SRC function (moreover, it works at higher frequencies and proportional to the main one). Added audiophile attributes - an internal power supply unit on a toroidal transformer, instead of an external pulse power supply unit. The dimensions and weight have grown, but is it really worth grieving if Dr. DAC 2 and Prime still had external power and, accordingly, could be mobile devices, but not portable?

The best part is the price, which, of course, is not ultra-low, psychologically expected at $ 300, but also not transcendental under $ 1000, but comparable to Prime at $ 500. The fact that this is a mass product gives a certain hope that we are facing another "killer" of famous products with price tags like phone numbers. On the other hand, the desire to get a low price tag forces the manufacturer to focus only on those parameters and features that will provide massive sales. And here you can remember STX with 23 dB, which in fact turned out to be only 113 dB for the 44.1 kHz mode, where the 123 dB record turned out to be only in the 96 kHz mode. Let's see what we got with the Asus Essence One.

Specifications

The summary table on official sources is somewhat chaotic (for example, the signal amplitude parameter is the same for Vpp and rms, although the ratio between them is Vrms = 0.3535Vpp), therefore, the brief characteristics are below, combined from the official data and our measured ones.

Appearance

The front panel houses an eye-catching power button with LED backlighting. The Upsampling button is responsible for upsampling. The next button switches the input sources: coaxial, optical or USB input. The Mute button mutes the sound. There are two volume controls: one for line outputs, the other separately for headphone output. The levels are individually adjustable.

The rear panel contains analog line outputs, balanced XLR and unbalanced RCA connectors. Digital outputs include coaxial, optical and USB inputs. Near the power connector (full-size pin with ground) there is a switch that allows the device to work with both 220 V and 115 V. When you turn it on for the first time, it is worth checking the set value, because if the voltage is set to 115 V, then the fuse will burn out inside, and until you replace it yourself, the device will not work.

Inside

Inside you can see a tight, neat installation. It was not possible to see any points on the basis of which one can suspect possible problems. For frequencies that are multiples of 44 and 48 kHz, two separate oscillators, 45.16 MHz and 49.15 MHz, are used.

Unlike conventional W-shaped transformers, toroidal ones do not spread interference around themselves and are allowed in devices with tight mounting. By the way, initially toroidal transformers were developed for the military industry just to be able to use tight wiring in compact devices. In conventional technology, W-shaped transformers prevailed, since it was permissible to locate the boards at some distance from the transformer.

The second advantage of such a power supply against external or internal impulse power supplies is that there is no propagation of pickups over a common network. It quite often happens that expensive Hi-End equipment is very sensitive to the presence of pulsed power supplies and, accordingly, does not sound so soulful, and external filters do not always save. Hence the main complaints about computers from which the sound is "bad".

The DSP ADI ADSP - 21261 is responsible for the main logic. It is likely that the SRC function is implemented just through it.

C-Media CMI 6631 is responsible for USB operation, and thanks to it, the Asus Xonar One has support for 192 kHz over USB.

SPDIF-receiver, implemented on AKM AK4113, is responsible for transferring digital streams from coaxial and optical inputs.

The block diagram shows all the op-amps that can be changed, there are 11. Next to the DSP of the ADI ADSP-21261 is an EN25F40 flash memory chip, which is similar to an op-amp, but it is not an op-amp. Accordingly, it is highly discouraged to change it to an OS ...

To convert from digital to analog, two PCM1795 microcircuits are used, each of which operates on one of the channels in PCM1795 mono mode.

Main technical characteristics of PCM1795:

  • work with data: 32 bit
  • dynamic range and S / N: 123 dB, (mono: 126 dB)
  • KGI + Noise: 0.0005%
  • Stop Band Attenuation: 98 dB

PCM1795 belongs to Burr-Brown's top line of DACs, where its main advantages are work with 32-bit data (does not require conversion from 32-bit to 24-bit) and low distortion. The noise level of the PCM1795 is not the highest for the DAC line, but its characteristics are not inferior to the top representatives from Crystal Semiconductor, AKM and Wolfson.

By default, NE5532 is located on the section of filters (1) and adders (2), these are widespread op amps that are more preferable than 4580, nevertheless, this is not the top, for the feeling that everything is right in the card, you should put OPA2132 or LM4562 there , which showed on the example of Dr. Dac2 one of the best results in objective and subjective tests. You can try other op-amps, including the rather expensive discrete op-amps from Bursone Audio, but you just need to be prepared for serious costs.

In sections 3A and 3B, the LM4562 op-amps are located, which are responsible for the output gain, controlled by the volume controls. The volume control is carried out through the low-noise regulators of the Japanese company ALPS.

The amplifier path consists of two replaceable LME49720NA and two non-replaceable LME49600TS.

By replacing various op-amps, you can choose a more successful sound for yourself, which is better coordinated with the rest of the path.

In work

During the first start-up, you will most likely need to update the firmware, there are detailed instructions for this. After installing the driver, an icon appears in the tray, when you click on it, an information window is called up. There are no settings or other controls. During the work, no problems were found.

During playback from Direct Sound and WASAPI, only the playback frequency is displayed, and during playback from ASIO, the Bit Perfect indicator lights up.

ASIO

Since the card does not physically have an input, it was not possible to determine the real minimum latency (in the test via RMAA). The latency level depends on the buffer size, and based on its value, you can calculate the latency, however, in the series of Xonar internal cards the latency was actually higher, which was not critical for consumer cards. Low latency in ASIO is essential for musicians to play virtual instruments live.

One cannot select the buffer value in samples, but only in ms, where the minimum value is 10 ms, which is similar to 440 samples for 44 kHz. Usually a buffer of 512 samples is considered normal, but already the extreme value for working live with instruments. Therefore, on the one hand, the minimum number is not very critical, but it could be much lower, as in professional products, in the form of 2 ms or even lower. But here it is worth noting that it comes about a USB device, where 10-20ms is the value at which the card works without crashing in the form of clicks.

When playing projects from under Ableton Live at frequencies from 44 kHz to 192 kHz with the selected minimum delay of 10 ms, there were no failures, it was clear that the bottleneck here was the laptop's CPU.

Listening

Two tests were carried out: using speakers and headphones.

With the help of acoustic systems, an assessment of the sound of the DAC path of the device was obtained, and with the help of headphones, an additional path of the built-in headphone amplifier in Asus Essence One was obtained.

Asus Essence One vs Asus Essence STX

A high-quality Technics SE-A5 amplifier, referred to as Hi-End models, was used in the path, with a measured distortion level of no more than 0.001% at average power. Acoustic systems based on Eton8-800 / 37HEX + Vifa XT25 were used as monitors. The same woofer is used in the ADAM S2.5A, and a similar tweeter in the BlueSky SAT 6.5 MK II satellites. For those who focus on quality = price, the cost of similar passive monitors on similar components will be approximately $ 2000-2500 per pair. The cost of the amplifier to adjust for inflation is $ 1500. The speaker impedance is 6 ohms. Additionally, Microlab Pure 1 was involved (cost at the time of active sales - $ 700).

Listening was carried out in the blind test mode. Both sources were brought to a single level with a spread of no more than a hundredth of a decibel. A separate switch on the wafer switch was responsible for switching sources.

When the test is carried out by the sighted method, and, moreover, with unregulated levels, then a large error in the result is introduced both by different psychoacoustic perception of sound at different volumes, and self-hypnosis (often from the type of price tag or preferences from brands).

The main thing that I would like to note is a different construction of the scene. In One, the scene is larger, if the distribution of sources in STX is a kind of diverging triangle, then the separation of instruments in direct comparison is not lined up in straight rows, but in an arc. If you don't listen too much, then at first it is noticeable that the location along the edges of the stereo panorama is the same (for the first row), and the center is far inland. What is more correct and what is better? It was possible to give an answer to this question by making a comparison with a "more reference" source, but in the Hi-End and upper Hi-Fi categories, and even with a lot of new products, this is difficult, since it is not difficult to challenge someone's standard ...

Therefore, the listener should choose what suits him best - an accented and approximate center (usually vocals) in the case of the STX, or a more distant center with an emphasis on the background and a circular panorama in the case of One. C One sounds less aggressive, more relaxing. Distortion, detail and reverb are all at a similar level.

Asus Essence One and headphones

To assess the potential, headphones were listened to as the usual top-class from the $ 300-500 category, and the highest category, from $ 1000. Taking into account the filling of the card and its large-scale production, it is quite fair to expect high quality from Asus One, comparable to more expensive piece-made devices from the Hi-End world. And what else to assess the quality, if not headphones from a similar Hi-End category?

For two models of headphones, a separate headphone amplifier, studio SPL Auditor, was used as an opponent, where the sound rating was set in a blind test.

Asus Essence One and Denon AH-D2000

The power reserve is more than sufficient, no problems were felt in the sound. The pressure in the low-frequency range is completely preserved, the bass articulation is good. In principle, the D2000 can be classified as a headphone with a sensitivity above average for its class (full-size headphones with low impedance), so the assessment with them is how well the amplifier works at low output power with a low impedance load. In this case, the weak point of the amplifiers is the possible audible background, which in the case of the D2000 was not. If not striving for the compression "warm" sound of tube amplifiers, then the built-in amplifier in One is quite self-sufficient.

Asus Essence One and HiFiMan HE-500

There were no problems with an isodynamic load close to a capacitor one. In addition, although the HE-500 is low-impedance, its sensitivity is low for its class (full-size headphones with low impedance). There were no problems with enough power, the One amplifier coped well with the "pumping". The development of reverberations is at a more than good level. For those who are not very versed in the types of headphones, isodynamic headphones are closest to electrostatic ones. Of the inexpensive models, there are the Fostex RP line, which are inferior in the quantity and quality of low frequencies due to the use of membranes of a smaller area.

Asus Essence One and Sennheiser HD 800

The HD-800 is a high impedance headphone. High impedance headphones are usually low sensitivity headphones. On the one hand, high impedance headphones are a relatively light load for an amplifier, and most often with such a load, the amplifier has minimal distortion. Basically, the problem lies in the maximum voltage that the amplifier is capable of delivering without clipping. There were no problems with the HD-800, no distortions arose at a volume higher than a comfortable one, the listener's ears were overloaded much earlier. Thus, the emphasis in the advertising brochure on a good headroom for high-impedance headphones is correct. But is the amplifier good enough for the HD-800 in terms of quality? Here, to unleash further potential, a separate amplifier, but already close to the cost of the HD-800 itself, or a “gustatory” amplifier, like a tube amplifier with a compression shade, will not hurt.

Asus Essence One and Audez'e LCD-2

Audez'e LCD-2, like the HE-500, is also isodynamic. Good bass control, there is a headroom. The feeling that the amplifier is a bottleneck did not arise, the only thing, because of the characteristics of the middle of the LCD-2, for them a tube amplifier might have been more justified. In direct comparison with the LCD-3, the older model, the middle of the LCD-2 is slightly sharper, and a "softer" amplifier could mask it. However, some loss of bass control can be expected when using a tube amp. As with the HD-800, it feels like a separate amplifier won't hurt to further unleash the potential.

Asus Essence One and Audez'e LCD-3

LCD-3 is the flagship of Audez'e. The use of Lotus technology has improved the transmission of the midrange. In general, the result is the same as with the LCD-2, but without the feeling that with another amplifier in the tube family, the bundle would be better. Good bass control, no obvious distortion. Adequate volume headroom. Considering the cost of the LCD-3, the owner will probably look towards a separate amplifier, but at the same time they will feel quite normal with the one available in the One.

E-MU1616m vs Asus One vs SPL Auditor

Everything is learned in comparison, and the studio amplifier SPL Auditor from the upper price category (over $ 1000) was chosen as a reference close to the standard of the headphone amplifiers. The best representatives were chosen as headphones: from high impedance - HD-800, from low impedance - LCD-3.

The double-blind test showed that the difference between the devices is easily distinguishable. The E-MU1616m and One were surprisingly close to each other, where the main difference was the One's tighter sound. The Auditor is more different, showing a more natural presentation of the sound, expressed in a better panorama. At some point, a suspicion crept in that the Auditor had a scene formation algorithm similar to those regulated in Phonitor, but technical tests showed a complete absence of any interference with the signal.

In terms of volume, the One surpasses 1616m with both low impedance and high impedance headphones. The One had slightly less distortion with the HD-800, and parity with the low impedance ones.

Measurements

When measured with the E-MU1616m, it was possible to obtain about 117.9 dB with level equalization. Without equalizing the levels, the signal level of One is lower, and therefore the value of 113-115 dB appears in the standard measurement reports.

This is practically the limit of the E-MU1616m's capabilities, and indicates that the device is really close or provides the declared 120 dB when measured on more accurate equipment in the form of Audio Precision.

The distortion of the device is very low, in the order of 0.0004%. When using One with USB, you can see a skirt around the pitch, which is indicative of low frequency jitter. Due to the low noise floor, it looks more intimidating than it actually is.

Upsampling

One of the features of Essence One is the use of upsampling. Usually, upsampling is done by a separate block in the DAC chip and for 44 and 48 kHz frequencies it is eight times (for 88 and 96 kHz - four times and for 174 and 192 kHz - two times). The main task of the block is to generate intermediate coordinates for constructing a sound wave, and if without this block the wave is "steps", then after upsampling the frequency increases and the steps become smaller. And with small steps - and a smoother wave itself.

The DAC block is far from always perfect, and therefore an intermediate solution is to use an external microcircuit. The SRC microcircuit receives a stream of 44 or 48 kHz, and from it already 172 or 192 kHz to the DAC itself (which, in turn, often already doubles the frequency with its block).

Since the PCM1795 is capable of receiving 352 and 384 kHz, Asus put a resampler capable of delivering just such frequencies, and emphasized this advantage against those devices that only upsample up to 192 kHz. The second plus of Asus emphasizes that frequencies that are multiples of 44.1 increase in the same multiples of 352 kHz (while Dr. Dac Prime 44.1 does not multiply converts to 192 kHz). A multiple counting often threatens with greater distortions than a multiple counting.

Since modern DACs no longer build a wave with steps, but they are significantly smoothed out, the quality of upsampling can be assessed only indirectly, according to the frequency response graph, the level of distortion and the external meander waveform through an oscilloscope.

When the upsampler is activated, one can observe an early blockage in the high-frequency region, which apparently is the price to pay for the fight against aliasing. This blockage is clearly displayed on the meander in the form of a fall, the shape of which was recorded using an oscilloscope.

You can observe how the wave front is slightly smoothed and has a larger angle along the deviation from the vertical, and the number of decay periods is reduced. In these devices, the frequency response, the form of the meander and the spectrum of distortions are interrelated parameters, and the improvement of one of them sometimes does not reflect the rest in the best way.

Distortion spectrum with upsampling looks worse than without it, but this is a price to pay for a different kind of impulse response.

If we consider multiple harmonics, then upsampling demonstrates the best results, the second and third harmonics are comparable in level, and the fourth and seventh went beyond the noise floor from −135 and 134 dB, respectively, to a level below −145 dB, and the fifth from −123 dB to −130 dB.

The result is overshadowed by the presence of multiple harmonics with a step of about 600-700 Hz above the fundamental tone. There is already a reason to think about what is better - reduced multiple harmonics or additional non-multiple ones. However, the resulting distortion rate in both cases is 0.0004%.

RightMark Audio Analyzer Test Report

Overall results

According to the data obtained, the internal resistance is 14 ohms and is constant in frequency. This means that if the headphones have no peaks on the impedance graph and the line is a straight line, then there will be no change in the frequency response of the headphones. In general, the output impedance is low and the effect of headphone impedance will be small.

To assess the distortion, it was performed using ARTA STEPS on 10 different resistive loads: 16, 23, 30, 56, 97, 202, 237, 609 and 1076 Ohm. For each load, the test was carried out at a frequency of 1 kHz, with an estimate of over 100 levels from 0 dBFS to -40 dBFS with the volume knob set to maximum. At the output, paired values ​​of the output voltage level and the level of harmonic distortion were recorded.

Based on the graphs obtained, we can conclude that at the lowest load values ​​of 16 and 23 Ohm, the amplifier goes into distortion at maximum volume (where the line goes up sharply). The analysis of the values ​​showed that this is due to the current level limitation of 0.15 A.

Based on the analysis of distortions, graphs of the maximum output voltage were obtained. The green graph indicates the measured values ​​subject to the current limitation, and the yellow one is theoretical if there were no limitations.

The maximum no-load level was 6.6 V (rms), under load this level decreases according to the internal output impedance of 14 ohms (yellow graph) and additional current limiting (green graph, where there is a discrepancy with yellow). For 16 Ohm, the maximum level is 2.5 V, for 30 Ohm - 4.2 V, for 100 Ohm - already 5.8 V, for 300 Ohm - 6.3 V. Usually, the output level for most cards is at the level of 2 V, less often 5 V, thus the card has really good volume headroom and most high impedance headphones are not a problem for the Essense One.

conclusions

Asus has released an interesting product, combining great design, good sound and audiophile attributes (internal power supply, replaceable op amps). At its cost, the product is competitive and can be safely recommended as part of high-quality paths, where the most optimal use would be with active monitors. Two separate controls allow you to independently adjust the volume of your monitors and headphones. The amplifier is well made, it can swing tight headphones without any problems. For headphones under $ 500, the amplifier is self-sufficient. From a technical point of view, there are some shortcomings, but they are not critical for this price category. The ability to replace a large number of op amps allows you to fine-tune the character of the sound of the device, and the upsampler allows you to get a different color of the sound.

Frequency response (in the range 40 Hz - 15 kHz), dB

Introduction Nowadays, when a sound adapter is built into every motherboard, not many manufacturers dare to release discrete sound solutions. Many companies left the market altogether, others stopped development and froze work on drivers, so the appearance of any new device today arouses a keen and genuine interest in the computer-music community. And if a new brand is introduced to the market, it is tantamount to a sensation. The fact is that a sound card must seriously surpass the built-in sound of motherboards in something, otherwise it simply won't make sense to buy it. Considering Creative's almost complete monopoly on gaming technologies, almost the only argument in favor of a sound card is the quality of music playback, more precisely, even the price-quality ratio, because the use of expensive components and careful study of the printed circuit board will unjustifiably increase the price of the product, and cheap components will not give proper quality. The price will also include considerable costs for the development of drivers and regular fixes of errors found in them. Having weighed all the pros and cons, most manufacturers preferred not to deal with such an inconvenient product as non-professional sound cards anymore, but a holy place, as you know, is never empty. The young and daring company Auzentech loudly announced itself in 2006 with the release of several interesting products based on C-Media controllers, and with X-Fi Prelude 7.1 it became known literally all over the world. But in the past 2007, an even more serious player appeared on the sound card market - the world famous manufacturer computer equipment ASUSTeK Computer Inc. began selling an ambitious audio solution for PCI and PCI-Express peripheral buses.

Formally, ASUS cannot be called a novice in the field of sound for personal computers, since in the last century it already released a sound card for the PCI bus - ASUS 3DexPlorer AXP-201. Even earlier, you can recall an audio-video combine for a specific PCI / ISA slot, which was soldered on some motherboards of the company, for example, ASUS T2P4. About ten years have passed since then, and the company is making a second attempt to conquer the sound card market. Taking into account the previous experience, ASUS relied on the extensive capabilities of the product, coupled with the highest technical characteristics, and also chose a strategic partner who helped to quickly and competently complete the task. For the Taiwanese company C-Media Electronics, a partnership with such a large manufacturer as ASUS is no doubt a good springboard for further growth, so the cooperation can be considered mutually beneficial.

In January 2006, at an exhibition in Las Vegas, C-Media presented many interesting new products, two of which are directly related to the hero of today's review. This is a sound controller for the PCI C-Media Oxygen HD bus, which is not inferior in its capabilities to the very common VIA Envy 24HT, and the C-Media Hydrogen digital audio processing software package, which provides support for DirectSound3D, licensed Dolby Laboratories technologies, and many other functions, including which we will return to in the course of the review of the ASUS sound card. The combination of these two novelties - oxygen and hydrogen - gave birth to many sound cards of hitherto little-known firms: Bluegears, Sondigo, HT Omega, the aforementioned Auzentech and even Razer. ASUSTeK could not stand aside either.

Let's take a look at the characteristics of the C-Media Oxygen HD versus the VIA Envy 24HT.


Both chips feature programmable output channel remapping and digital input monitoring. One of the five two-channel I²S outputs of the Envy24HT, assigned to the digital output, has a built-in transmitter, but still allows connection of an additional device with an I²S bus. Oxygen HD is deprived of such an opportunity, but it also has a built-in S / PDIF receiver, the signal from which can be sent directly to the digital output, which allows using the sound card as an adapter from an optical cable to a coaxial cable and vice versa. The Envy24HT only allows simultaneous recording of two stereo sources (usually a signal from an A / D converter and S / PDIF), while Oxygen HD allows simultaneous recording of three sources (one of which is eight-channel), and the sources can be extremely flexibly selected from four I²S input pairs. built-in digital input and two AC'97 codecs at once.


Functional diagram C-Media Oxygen HD


Judging by the description, this chip has no noticeable weaknesses and can become the basis for a professional sound card, but today we will consider a product aimed at home use in a media center or a gaming computer.

Appearance


The bundle of the ASUS sound card is quite rich. In addition to the installation instructions, it includes four analog cables "3.5 mm → 2xRCA" 1.8 meters long, a thin one and a half meter optical cable, an additional bracket with a MIDI interface and a bunch of CDs: installation CD, application software, demo Dolby Laboratories disc.


The card itself is strikingly different from other members of its family in that it is covered with a blued aluminum shell, thanks to which it looks and feels like a very serious device. Due to the round cutout in the "armor", it looks more like a video card than a sound device.


However, the panel with connectors on ASUS Xonar D2 is absolutely typical, one-to-one repeating that of Auzen X-Fi Prelude 7.1: six 3.5 mm connectors (microphone input, line input, four stereo outputs) and two RCA connectors (digital input and digital output).


Reminiscent of Auzentech and combo digital ports that allow the connection of both coaxial and optical cables through special adapters plugged into the RCA jack.


However, even here, ASUS engineers found a way to distinguish themselves - inside all six connectors, multi-colored LEDs are installed, vaguely resembling the color coding of connections. prescribed by instruction PC 99, page 60.


Such an elegant solution, in addition to purely aesthetic functions, facilitates the process of connecting wires to the sound card. You can determine the required connector by the disappearance of one of the color spots on the wall behind the computer or even on your own hand. By the way, the bundled cables also have an unobvious plus - a small diameter of the plastic part of the connectors, which allows you to easily connect a multichannel speaker system to the card.


The PCB also contains the CD IN, AUX IN and MIDI I / O jacks. You can connect low-quality sound sources to the first two, for example, a TV tuner - the signal from them goes to the recording mixer, but it can also be sent to the outputs of the sound card by activating the corresponding Monitoring button in Xonar Audio Center. An additional mini-DIN rail is connected to MIDI I / O, and large DIN connectors MIDI In and MIDI Out are "obtained" using a Y-shaped adapter.

Device

Something about the filling hidden under the protective casing can be found in the manufacturer's description.

Sound processor ASUS AV200 High-Definition Sound Processor (max. 192 kHz / 24 bit)
24-bit Burr-Brown PCM1796 DAC * 4 (123 dB SNR, max. 192 kHz / 24 bit)
24-bit ADC Cirrus Logic CS5381 * 1 (120 dB SNR, max. 192 kHz / 24 bit)

The rest opens up if you dismantle the casing: original rectangles made of film capacitors (which we have never seen on any large-scale sound card) in DAC filters, solid-state electrolytic capacitors, a scattering of miniature electromagnetic relays (there are even more of them than in Creative X- Fi Elite Pro), on the right edge of the board, as well as a myriad of op amps.



Since the digital-to-analog converters used on the card (like most high-end microcircuits of this type) have balanced current outputs, a current-to-voltage conversion is required before the low-pass filter, and this increases the number of required operational amplifiers by three times.


On three of the four output channels, cheap two-channel amplifiers, model 4580 manufactured by Texas Instruments, are used for the converter and filter, and at the front output, the much more expensive NJM2114 (in the stage of I / U conversion) and LM4562 (in the high-pass filter) are used. In addition, the front output uses two RC4580s as buffers to connect the headphones directly to the sound card. A similar circuit design is used at the headphone output of the M-Audio Revolution 5.1 and Audiotrak Prodigy HD2 sound cards.


The analog-to-digital conversion filters are made on the NJM5532, but the RC4580 is again used as an inverter for the balanced inputs of the DAC. Such a variety of operational amplifiers may indicate a thorough work on optimizing the quality and cost of the sound card, let's see how all these microcircuits differ.


All four models are recommended by manufacturers for use in audio devices, and three of them have fairly similar characteristics. They are united by a very good immunity to supply voltage and ground drops, a low level of harmonics in the audio range. However, the LM4562 stands out from this group in terms of speed and distortion, in connection with which the logic of the card designers raises certain questions.


You can read about the advantages of the LM4562 over other operational amplifiers, including the NJM2114 using the Creative X-Fi Elite Pro as an example. in our previous article... Operational amplifiers for I / U conversion are used in inverting connection, which means that LM4562 would work well and in this cascade. Since it has a much wider gain bandwidth and less distortion, its combination with the NJM2114 looks odd. In addition, the output buffer on the RC4580 improves performance with headphones, but when operating at high impedance loads, it practically negates all the advantages of the LM4562. Moreover, the film capacitors in the feedback loop of the filtering units are adjacent to the ceramic capacitors in the feedback of the I / U conversion stage. One gets the feeling that the individual blocks of the analog part were designed by different engineers who did not consult with each other ...

Let's take a closer look at the converters used in ASUS Xonar D2. Burr-Brown PCM1796 belongs to the Advanced Segment DAC class and, according to the manufacturer, combines excellent dynamics with low sensitivity to jitter. The dynamic range of the converter reaches 123 dB, harmonic distortion in normal conditions is at the level of 0.0005% for sampling frequencies of 44.4, 48 and 96 kHz, but slightly increases at 192 kHz. The graphs below are obtained on a typical DAC switching circuit. The I / U conversion and filtering stages of the typical circuit use the NE5534 op-amp, which does not have outstanding distortion characteristics, so even better results can be expected in other implementations.




However, in the range of digital-to-analog converters manufactured by Texas Instruments there is an even better model PCM1792A, which provides a dynamic range of 127 dB and a distortion level of less than 0.0004% for sampling frequencies of 44 and 48 kHz.


It is noteworthy that this model is fully compatible with PCM1796 in terms of contacts and command system. If it were not for the four times higher price, it could have been used in ASUS Xonar D2 without modifying the printed circuit board. Frankly, I don't see much point in making all four outputs of a sound card of the same quality, and ASUS engineers clearly adhere to the same opinion, since they use cheap operational amplifiers on all outputs except the front one. Adhering to the principle of reasonable sufficiency, PCM1791A (dynamic range 113 dB, distortion 0.001%) could be used on the rear channels, which not only costs 30% less than PCM1796, but also does not need an I / U conversion stage, which will save six operational amplifiers and even more trim parts. There is also an even more radical option that allows you to create a low-profile version of the card - the six-channel PCM1602A (dynamic range 105 dB, distortion 0.002%), which provides three stereo outputs at once at a price less than one PCM1796. By the way, the recently released low-profile Xonar DX card is built exactly on this ideology: a high-quality two-channel Cirrus Logic CS4398 DAC (as on the Creative X-Fi Elite Pro) for front output and a six-channel Cirrus Logic CS4362A (dynamic range 114 dB, distortion 0.001%) for the three remaining exits.

In contrast, the ASUS Xonar D2 A / D Converter is based on the best Cirrus Logic model - the CS5381, which provides a signal-to-noise ratio of 120 dB and distortion at the level of 0.0003% at all sampling frequencies, including 192 kHz.


This is possibly the highest quality mass-produced analog-to-digital converter. Theoretically, with its help it is possible to carry out the most accurate measurements of signals in the frequency range of about 50 kHz (above the signal suppression with a digital filter begins even at a sampling frequency of 192 kHz), but there is one "but" - the analog part must correspond to the same the highest level... The manufacturer himself recommends implementing a low-pass filter on ultra-low-noise operational amplifiers LT1128 manufactured by Linear Technology, which have a minimum offset voltage and excellent distortion (about -130 dB), but ASUS engineers used not so high-quality NJM5532.

Driver capabilities

The ASUS Xonar D2 driver differs from the base driver for C-Media Oxygen HD in its support for OpenAL, SVN and DS3D GX functions, as well as a different control panel styled as a pocket media player with the characteristic name Xonar D2 Audio Center. In the latest drivers, the panel is completely Russified, which may be important for many Russian buyers.

After installing the driver, the panel reveals a minimum of controls - a few buttons and a volume control, and the rest of the space is occupied by a huge information display with an indication of the current settings and a primitive spectrum analyzer.



He is primitive because the accuracy of his testimony does not stand up to criticism. The round volume control looks nice, but it is not very convenient to use - it is not possible to turn it by any movement of the mouse, and it does not react at all to the rotation of the mouse wheel.


You can view the versions of software components and some other information in the window called by the button with the letter "i".


The "SVN" button near the volume control activates the volume normalization mode - a function that has proved to be very useful when watching movies with quiet speech, but loud special effects. In games, it is better not to turn on the automatic volume change, because it blurs the sharpness of shots and explosions, seriously impairing the auditory experience.


The group of five buttons in the lower right corner of the audio center includes the DS3D GX function and one of three sound processing options - games, movies, music. HF, on the other hand, turns off all processing; it was in this mode that I measured and evaluated the sound in music. The rest of the controls are hidden under the "cover" of the information display sliding upwards.



The selection of the card's reference frequency, the selection of the number of speakers, the selection of the digital output data format, and the settings of the Dolby licensed technologies are found here. For example, for headphones, any of them includes Dolby Headphone, and for stereo speakers, Dolby Pro Logic IIx and Dolby Virtual Speaker in one of two modes of operation.




Many other options are displayed by clicking the buttons with the corresponding names. Why hide all this wealth under a very slowly sliding cover? Thank God that you need to wait for it to move only once, and then the selected position is remembered in the registry.


As you can see, the available set of settings depends on the selected speaker configuration. For some reason, the two speakers are missing DPL IIx and DTS settings, with Pro Logic turning on automatically when Dolby Virtual Speaker is activated. With headphones, the Virtual Speaker Shifter turns on only after Dolby Headphone is activated.

In the basic drivers, the C-Media Virtual Speaker Shifter operates independently of the Dolby Headphone, allowing you to activate the technologies for converting stereo sound into multichannel, from which we can conclude that the Virtual Speaker Shifter should create a virtual 7.1 system and then mix the sound into the required number of channels. In the latest drivers, nothing of the kind is observed, which means that you can suspect a banal error in the driver or its control panel.


The position of the speaker on the plan of the virtual room determines the volume of the corresponding playback channel, and if you move, for example, the right speaker to the left, then both channels will be mixed into the left "ear". This setting can be viewed as a simplified alternative to Creative THX Console with significantly less functionality. The THX Console allows you to set the distance in meters and direction in degrees for each speaker of the speaker system, based on which the volume of the corresponding channel and the signal delay are adjusted. The ASUS implementation is not tied to any units of measurement at all - apparently, the user needs to select the position of the speakers by ear. Then why is there no test signal for this?

For comparison, the basic C-Media drivers at least display the relative loudness in decibels, allow you to set delays for the center and rear channels, and also offer three test signals to evaluate the result. By the way, if you install the ASUS driver manually, and not through the Setup.exe executable file, you will be able to see the original C-Media control panel.




ASUS digital output settings also differ from the basic driver, but this time for the better. The output format and clock frequency are separately set, which is valid for the entire card (including DAC / ADC) and is observed when the digital output is disabled. The C-Media control panel allows you to select the base frequency only when S / PDIF is enabled.


S / PDIF on ASUS Xonar D2 can transmit uncompressed stereo with a sampling rate of 44.1 to 192 kHz, 5.1 sound in Dolby Digital Live or DTS Interactive formats, as well as broadcast a signal from a digital input without processing. The latter option allows you to use the card as an adapter from a coaxial cable to an optical one and vice versa. When you enable DDL or DTS encoding, the card is switched to 5.1 mode and the analog outputs are disabled. At the same time, the volume control continues to work and it is possible to turn on automatic SVN control.


Dolby Pro Logic IIx in this case is responsible for stereo decomposition into 5 speakers, although with headphones or a two-channel speaker system, this technology can also perform the opposite function, converting multi-channel sound to stereo. DPL IIx supports seven-channel speaker systems and has three modes of operation, of which only “Music” and “Movie” are available in ASUS Xonar, and for some reason there is no “Game” mode. In “Music” mode, you can adjust the balance of the mid frequencies (vocals) between the center and front two speakers, as well as the depth of the sound field. The "Panorama" option present in the original C-Media control panel is also lost somewhere.


Alternative technology DTS Neo: PC can be used to decompose stereo into a multichannel configuration, which has completely similar operating mode settings.


But full, how long can you study one section of the settings? Although he is the main one, he is far from the only one. Next in order is the "Mixer" section.

As with many other sound cards, the playback mixer is separate from the recording mixer, which forces you to press an extra button when you need to change the recording parameters. In the original C-Media control panel, playback and recording settings are on the same page, which is much more convenient. But, most importantly, the Xonar D2 Audio Center does not have a single level indicator (the so-called peak-meter), although a pair of indicators is present even in the C-Media panel. A model to follow here is the Audiotrak Prodigy 7.1 control panel, where each volume control is equipped with a signal level bar.


In addition to the usual Wave, MIDI and CD volume controls, playback settings allow you to separately set the volume of each of the analog outputs, and by default they are set far from the maximum volume.


The recording mixer allows you to choose one of a variety of signal sources for recording: the familiar "SPDIF In", "Line In", "CD In", "Aux" and "Mic", less often "Wave" and "Mix", as well as a unique Alt. If “Wave” is a digital loop-back, allowing you to record the reproduced signal with bit-precision, then “Alt” is the bridge between the line-out and the line-in of the card. This recording source can be useful when playing protected content for which, according to the terms of the license agreement, all digital recording sources must be disabled. I didn’t have a chance to check this thesis in practice, however, for “Alt” you can find other useful applications, for example, measurements in RightMark Audio Analyzer are recommended by the manufacturer through it.

"Mix", as you might guess from the name, combines signals from all sources. A laudable feature, but this feature would be doubly useful if the user was given a choice of which sources to mix and which to ignore. It is also somewhat inconvenient that the nominal recording level is obtained at the maximum position of the level control. To increase the volume of a quiet signal, you will have to use some kind of sound editor. Because of this, in particular, I could not carry out full measurements of the card with headphones connected: the volume with a low-impedance load dropped by more than 10 dB and RightMark could no longer cope with level normalization, and using a Creative sound card for recording led to inadequate results.

Enough about the shortcomings, there is a recording mixer in the ASUS Xonar D2 and a very convenient thing - monitoring of all inputs. Buttons with the image of an eye allow you to start playing a signal from the corresponding input for auditory control, and any number of these buttons can be pressed simultaneously. Now, if the "input selector" worked according to this principle ...

However, I got carried away again. Let's take a look at what possibilities are hidden on the other pages of the impromptu menu.


On the tab called "Effect" we see the lurid settings for the ambient effect (reverb) and the equalizer. The logic of the creators, who rendered four variants of acoustic environments on the buttons, and left the rest in the drop-down list, the selected value of which will be activated when the fifth button is pressed, gives me a feeling of slight bewilderment. In addition, the lack of adjustment of the reverberation intensity is saddening. Let's say I would like more echo in games, but have to be content with what they give.

Equalizer is a separate song. Not only is it just tiny, and it is almost impossible to set the necessary parameters with its help, it also works in a strange way. After trying to slightly muffle the 4 kHz band, I did not notice any changes in the sound, and moving the slider to the very bottom, I realized that 4 kHz had been cut out completely. The funniest thing was that it was possible to bring this frequency back to life only by resetting the settings with the "Default" button. Saving custom EQ settings is also not intuitive ♠. To do this, you need to type the name of your setting in the lower entry field and press the button with the plus sign, and the saved setting is deleted by the minus button.

In my humble opinion, the equalizer was worthy of a separate menu tab, especially given the lack of tone controls familiar to most users.


Three settings are available on the tab named "Karaoke": the tempo of the music, the suppression of the voice in the song, and the echo effect for the microphone. The meaning of these settings is clear to everyone who has ever sang "karaoke", so let's move on to the next, very interesting menu item called "FlexBass".


Here, in full accordance with the name, the bass distribution between the speakers is flexibly configured. The crossover slider determines the limit below which frequencies are sent to the subwoofer and cut from those channels that are set to small size. Large speakers do not cut off the low frequencies.


The crossover works very well, providing a symmetrical roll-off of 36 dB per octave and without introducing any distortion into the sound.


The two bottom buttons, "AEC" and "VocalFX", appeared in the drivers quite recently.


The "AEC" mode, that is, "Acoustic Echo Cancellation", is intended for video conferencing and other important negotiations over the Internet, therefore it turns off all processing effects and tries to suppress sounds entering the microphone from the speakers. A quick test of this mode showed high suppression efficiency.

In my opinion, it would be more convenient to activate AEC with a button in the lower right corner of the Xonar D2 Audio Center (where preset sound processing options are grouped), since the entire setting is essentially represented by one “On / Off” checkbox.

The last tab, formerly called "Magic Voice", contains several additional functions for processing the signal from the microphone. Among them are changing the timbre of the voice (male, female, "cartoon" and the voice of a monster), the imposition of one of four options for the environmental effect, as well as a special effect for games.


VoiceEX applies the reverb to the player's voice as specified by the game for the player's current location. This feature was first introduced by Creative Labs as part of EAX 5, and now ASUS can proudly declare that its product supports the most modern technologies competitor. Whether this is really so, we will find out a little later, when we test the card in games, but for now we will consider other parts of the driver hidden from the eyes. For example, ASIO support.

ASIO is a special interface for transferring data to a sound card with a specified delay and is used in the vast majority of audio processing programs. ASUS Xonar D2 provides full ASIO 2.0 support with 16- or 24-bit fidelity and frequencies of 44.1, 48, 96 and 192 kHz. In addition, the driver supports such a wonderful feature as ASIO multi-host, thanks to which several programs can simultaneously work with ASIO. There is no FSM on the card, therefore a signal with a sampling frequency different from the base one is recalculated by a high-quality software oversampling algorithm. For comparison, Creative X-Fi in this case begins to methodically switch the frequency of the generator, accompanying this process with incessant clicking of the relay.

The ASIO implementation in ASUS Xonar could be called exemplary if it provided parallel recording of more than one source, as well as several additional playback channels, of which Creative X-Fi, for example, has as many as 18. Let me remind you that the C-Media Oxygen HD audio controller used on the ASUS Xonar D2 allows simultaneous recording of eight streams. It would also be nice to be able to bring up the ASIO setup window from somewhere in the Xonar D2 Audio Center.

Oh yes, I completely forgot to talk about another original function of this sound card. After installing the driver, you will find two sound devices in the system, ASUS Xonar D2 Audio and ASUS Xonar D2 Converter. The latter device is designed for quickly applying effects to music recordings. If you use an mp3 player and like the spatial effect created by Dolby Headphone or Virtual Speaker technologies, or just want to adjust the sound of the recording with an equalizer, using the bundled ASUS Portable Music Processor you can transcode your favorite songs into mp3 or WMA formats with any available the driver with a special effect.

On this note, I propose to finish the study of the potential capabilities of the card and move on to more relevant field tests.

Listening to music

ASUS Xonar D2, as a very high-end sound card, was compared with the best non-professional models to date - Creative X-Fi Elite Pro and Auzen X-Fi Prelude 7.1. To assess the sound quality of sound cards, we used Grado SR 325i headphones together with a C.E.C. amplifier. HD53R Ver. 8.0, connected to a sound card with a Monster Standard Interlink 200 cable, records of various genres saved from CDs in wave format, as well as the players foobar2000 0.9.5 and WinAMP 2.95. The sound cards were set to the same volume - standard 2 V RMS in order to get the maximum dynamic range. The replay gain was disabled in the foobar settings, the sound was output via DirectSound in 32-bit format. WinAMP was used in conjunction with the plugin ASIO output (dll version) 0.67 SSE2.

The reason for such a detailed description of versions and settings is trivial. Even when writing a review of Auzen X-Fi Prelude 7.1, I noticed that not every player or sound output plug-in provides the correct transmission of reverberations and sound localization in space, and these two options are relatively reliable. For such high-quality sound cards, choosing the right software player is of no small importance. For example, the ASIO audio output plugin for Foobar version 0.9 does not sound correct, and this is easily confirmed by measurements.



Intermodulation distortion graph when outputting sound via ASIO different
players. Digital loop, 44.1 kHz 16 bit


The first experience of listening to music on ASUS Xonar D2 with the supplied drivers left not a very favorable impression. Its generally very clean and detailed sound was devoid of depth, saturation of overtones. Using additional epithets, I will call this sound faded, faded. In addition, the soundstage turned out to be wide, but absolutely flat, without separation, which completely discouraged listening pleasure. Fortunately, the programmers found a bug, and things went much better with the current drivers - so don't hesitate to update them.

The card instantly captivates with its deep and dynamic sound. I was very impressed by the confident transmission of the smallest details against a loud background, which is especially noticeable in the richness of the sound of violins, including in a large orchestra. The voices of other musical instruments are also quite natural, the mids and highs are clean and detailed, the bass is crisp and rich. But, with all these advantages, there is not always enough "air" in the sound, after-sound and natural reverberations are quickly lost in the general mass, which is why, with many instruments sounding simultaneously, there is a feeling of lack of detail. This is clearly audible in comparison to the Auzen X-Fi Prelude. How delicate she is about room acoustics! Spaciousness, realistic spaciousness even on bass, excellent detail and micro-dynamics in the upper register, giving high natural timbres, make Prelude the best choice for jazz music and other "live" recordings. Xonar differs from Prelude in harder high frequencies, but noticeably better macrodynamics, which unambiguously settled the dispute in favor of Xonar on the Judas Priest recordings.

The sound of Creative X-Fi Elite Pro initially strikes with energy and accentuated detail, but elastic, well-developed bass coexists with caustic high frequencies, and the middle ones are inexpressive and frankly colored - after ASUS Xonar D2, it seems that most instruments play a semitone higher. Mixing colors from several instruments lead to a deterioration in spatial resolution, while reflections of high-frequency sounds from the studio walls (reverberation) lose their isolation and form something difficult to recognize with the original sound. Something like that, namely harshness, aggressiveness, unpleasant to hearing, I have already met on the good old Audiotrak Prodigy 7.1, which still remained in my computer thanks to one accidental opening.

While studying the documentation for the Wolfson WM8770 digital-to-analog converter, I noticed how much the characteristics of the digital filter differ at different sampling rates, and tried software oversampling (SSRC) of records at 192 kHz, and then at 176.4 kHz - a multiple of standard for music 44.1 kHz. The result pleased me very much: the sharpness disappeared, the scene became much more voluminous, with such a sound the desire to buy an even more expensive sound card disappeared. Much later, when I decided to replace the operational amplifiers, an interesting detail emerged - with models of operational amplifiers faster than the stock NJM4580, oversampling does not give any improvement! The Creative X-Fi Elite Pro's front output uses NJM2114 op amps, which are close enough to those of the NJM4580, so I decided to try software oversampling again.

Creative sound cards of the X-Fi family do not support higher sampling rates in the Audio Creation mode, and therefore we had to limit ourselves to 96 kHz, but this was enough to significantly improve the situation. Coloring becomes much less and high-frequency "plaque" disappears from the scene almost completely, and the mids are much better worked out. First, I used oversampling in the ASIO audio output plugin for WinAMP, then I simply set the desired clock frequency in the Creative mixer settings, using the hardware capabilities of the card for recalculation - the result was the same. Switching the sound card to the "Entertainment" mode, where stereo recordings are always recalculated to 192 kHz, with the equalizer, Crystalizer, CMSS and SVE turned off, I got an even cleaner and more juicy sound at the cost of very little loss of spatial clarity. Thus, Creative X-Fi owners who do not want to mess with soldering chips on their sound card can be advised to stay in “Entertainment” mode while listening to music. The hardware conversion to a different sampling rate in X-Fi is implemented very well, and the sound as a result turns out better than with honest reproduction of bit-for-bit at a clock frequency of 44.1 kHz.

However, in a previous review, I noted that the Auzen X-Fi Prelude 7.1 exhibits a different behavior, namely a drop in clarity without noticeable improvements in other characteristics.

But what about Xonar, does oversampling give it anything, since both NJM2114 and RC4580 are involved in the front output? To begin with, I just clicked the frequency selection setting in the ASUS Audio Center, but I could not hear any obvious differences in the sound - somewhere there was no difference at all, on vocals it sometimes seemed that the sound became dirtier. Then the same WinAMP was launched with the plug-in for audio output via ASIO, and doubts immediately disappeared: recalculation to 192 kHz with "Ultra" quality gives room, the lack of which I noted above, makes the high frequencies softer and more detailed, and also adds a little more dynamics. The differences are less noticeable than on the Creative X-Fi Elite Pro, but still enough to take into account.


The signal with 16 bit precision and 44.1 kHz sampling rate is transmitted without distortion through any interface available to the card. The results for ASIO, slightly exceeding the theoretical capabilities of the format, are a consequence of an error in the playback procedure of Rightmark Audio Analyzer, since when playing through WinAMP with an ASIO plug-in, a complete match with the sample signal is obtained.


At first glance, all interfaces transmitted the test signal of 24-bit precision without loss, however, a detailed study of the noise shelf in the presence of a tone with a high level showed that the Kernel Streaming interface turned out to be the closest to the original, although DirectSound with MME practically does not differ from it ... But ASIO this time added a small amount of distortion, which is not present when playing a signal using WinAMP.


Even the prehistoric Wave Out in the latest versions of ASUS Xonar drivers copes with 32-bit precision signal transmission, and DirectSound even managed to slightly exceed the theoretical dynamic range limit. However, the measured level of noise and interpenetration of channels unambiguously indicates that some rounding is performed in the driver beyond the 24-bit signal accuracy. Perhaps this is a consequence of the conversion of the signal to floating point format for the volume controls and the oversampling algorithm.

The ASIO result is completely similar to that obtained with 24-bit precision, from which one can make an assumption about the nature of the error in working with ASIO in Rightmark Audio Analyzer. The fact is that C-Media allows two variants of ASIO, with 16- and 24-bit signal representation accuracy, while the implementations of all other manufacturers adhere to 16 and 32-bit values, which are more convenient for the central processor. Apparently, this development was not foreseen by the developers of the test package.

Interestingly, in this case, Kernel Streaming turned out to be somewhat worse than the exemplary interface. The picture below shows the spectrum of the recorded signal in the dynamic range test, which most clearly illustrates the behavior of the driver.



Dynamic range test spectrogram


Now let's move on to measuring the quality of the analog part of sound cards. The first step is to determine which card has a better line input, for which it is necessary to measure the line outputs through its own line input and the input of another card.


The line inputs of ASUS Xonar and X-Fi Elite Pro turned out to be almost identical in quality, with a minimal advantage of ASUS Xonar D2 in terms of noise, if we do not take into account the fact that for Creative X-Fi it was necessary to select a PCI slot with the least power noise (it turned out to be the bottom slot of the motherboard), and ASUS Xonar D2 performed equally well in any slot.

Since connecting one card to another leads to poor measurement results, further research will be carried out through the own line input of each sound card. First, let's examine the parameters of the front and rear channels of the two cards. Let me remind you that they differ only in the models of the operational amplifiers.


Judging by the data obtained, the front output of the ASUS Xonar D2 differs from the front output of the Creative X-Fi Elite Pro only in the coefficient of nonlinear distortion. The situation is absolutely the same with the rear outputs. Of course, according to meager numbers, ASUS Xonar D2 turns out to be a winner, but in the Auzen X-Fi Prelude review I already wrote that the best number does not necessarily give the best sound, so let's start comparing the levels of individual harmonics.



Xonar D2 Out 1 (Front) Distortion Spectrum



Xonar D2 Out 2 (Rear) Distortion Spectrum



X-Fi Elite Pro Out 1 (Front) Distortion Spectrum



X-Fi Elite Pro Out 2 (Rear) Distortion Spectrum


The front output of the ASUS Xonar D2 showed a significant difference between the channels. In the left channel, the level of harmonics is generally lower by about 10 dB, but in the right channel, the fifth harmonic is completely absent. I can’t judge the reasons for this behavior, I can only say that the rear output did not demonstrate anything of the kind, but lost a lot even to the worst of the two front output channels. The linear output of the Creative X-Fi Elite Pro lost just as much in terms of the third and fifth harmonics, and its seventh harmonic is more than 10 dB higher. As a result, ASUS Xonar D2 wins the competition of “main” outputs by a large margin, but the rear channels of Creative's sound card look much more convincing.

Now let's check how the measured characteristics of two cards change depending on the sampling frequency of the signal.



Xonar D2



X-Fi Elite Pro


Creative X-Fi does not allow recording a signal with a frequency of 192 kHz, therefore measurements were not carried out in this mode, however, it is so noticeable that its behavior is noticeably different from the ASUS Xonar D2 - the best results in noise and dynamic range were obtained at a frequency of 96 kHz. In addition, ASUS is significantly inferior in these parameters at 44 kHz. Is this a problem with signal playback or recording? We'll have to take measurements again through the line inputs of another card.


Obviously, the decrease in the dynamic range at a sampling rate of 44 kHz in ASUS Xonar D2 is not connected with the peculiarities of the analog-to-digital converter, but entirely lies on the conscience of the playback path. Is this the reason for the noticeable improvement in the sound of music when using software oversampling? As we found out earlier, the driver completely correctly transmits a digital signal with an accuracy of 24 bits, which means that the dog is buried somewhere deeper, perhaps in the interface with the DAC or in the clock generator. I would like to believe that ASUS, together with C-Media employees, will be able to sort out this problem in the card's operation, as they have already eliminated the intermodulation distortions observed when using earlier versions of drivers for Xonar D2.

Taking this opportunity, I conducted a similar study at a sampling rate of 96 kHz.


Comparing the obtained noise figures with the previous results, you come to the conclusion that the Creative X-Fi Elite Pro analog-to-digital converter works best at a frequency of 96 kHz, while a similar functional unit ASUS Xonar D2, on the contrary, provides the greatest dynamic range at a frequency of 44.1 kHz. If it were not for the annoying problem with playback at this frequency and the smaller distortion spread between the left and right channels of the front output, the measurement results of the ASUS sound card could be even more impressive.

Those who wish to see the full test results in the RMAA can download the archive with them (7.7 MB).

Conclusion

After a close acquaintance with ASUS Xonar D2, we can admit that this is one of the most versatile sound cards on the market. The highest quality technical performance allows it to show the best measurement results among mass products, it sounds great in music, provides excellent surround sound in games and has the most versatile driver I have ever seen. The number of different sound effects and useful add-ons, such as automatic encoding of multichannel audio into Dolby Digital and DTS formats, will satisfy even the most sophisticated user. Of course, one can complain about the lack of a driver for Linux, however, on the other hand, if the sound card is used in a multimedia computer running Windows Media Center Edition, the ASUS Xonar D2 driver will install a special version of the control panel optimized for display on a TV.

Support for MIDI devices, excellent ASIO 2.0 implementation and superior sampling quality of analog inputs make the card suitable for professional use in sound recording. For those who want to try their hand at making music, the card comes with a set of professional applications with somewhat limited functionality. The only drawback in this regard is the impossibility of recording several sources at the same time, although the heart of the card - the C-Media Oxygen HD controller - quite allows it.

On the positive side, you can also refer to the licensed Power DVD 7 and a solid set of accessories for connecting the card to an amplifier or receiver. If you are going to use only headphones, then you do not need to worry about an additional amplifier, since the card copes well even with a low-impedance load. Although for those who have no reason to connect headphones directly to a sound card, an additional stage on low-quality op amps is more likely to spoil the depletion, since it inevitably degrades the sound quality, but does not rid the card of sensitivity to interconnect cables. To ASUS 'credit, the bundled cables are of sufficient length and quality; it makes little sense to look for something else.

In order not to retell the content of the entire review, I will not summarize the advantages and disadvantages of ASUS Xonar D2, especially since the main complaints relate to the Xonar D2 Audio Center control panel. The versatility of the card makes it difficult to unambiguously determine the target audience, but the main strong point of the card can be called - the highest quality of all analog outputs. This makes it particularly attractive for creating a home media center with very high quality multichannel acoustics. Household DVD-players with components of comparable quality will cost an order of magnitude more than ASUS asks for its product.

And yet, for complete audiophile happiness, I would like more - the Deluxe version of the card, with an even better Burr Brown PCM1792 digital-to-analog converter and the highest quality operational amplifiers at the front output.

Other materials on this topic


X-Fi for music lovers: sound card Auzen X-Fi Prelude 7.1
Grado SR325i and Sennheiser HD 600: high-end headphones
Creative SoundBlaster X-Fi: October Revolution of Audio Cards

Even after several years, the ASUS Xonar DX sound card remains a very attractive purchase. ASUS products in the field of computer audio generally became popular and in demand very quickly, since the manufacturer really offers a worthwhile implementation for a reasonable price.

Of course, the Xonar DX is in many ways a stripped-down option, but for the average user, the card can be a real boon. Nevertheless, the solution is based on fairly serious digital-to-analog converters Cirrus Logic CS4398 and CS4362. They support 192 kHz 24-bit operation and provide a signal-to-noise ratio of 120 and 114 dB, respectively. All this gives the manufacturer reason to assert that in some parameters ASUS Xonar DX is tenfold better than integrated audio. There is a bit of slyness in this, but this fact does not diminish the dignity of the interface.

Therefore, our Xonar DX review will try to figure out if this sound card is really as good as the mainstream built-in HD Audio.

Contents of delivery

The completeness of the card is not particularly outstanding. In the box you can find:

  • replaceable socket for compact enclosures with two screws;
  • adapter for additional power supply;
  • optical cable adapter;
  • signal cable 3.5 mm - 2 RCA;
  • instructions and a disk with the driver.

As you can see, there is not even a complete set of cords for connecting a multichannel system. For example, the bundling of Creative cards of the corresponding level is often much more serious.

Card construction

As the main sound processor ASUS uses its proprietary chip, marked as AV100. Its real developer is the C-Media company, and in its performance the microcircuit was called OxygenHD CMI8787. By the way, the differences between ASUS AV100 and ASUS AV200, which are installed in more expensive cards, are very conditional, and they boil down only to algorithms for working with software. ASUS Xonar DX has a PCI Express x1 interface, and a PEX812 controller from PLX Tehnology is used for communication.

A Cirrus Logic CS5361 microcircuit works as an analog-to-digital converter. Cirrus Logic CS4398 192 kHz 24 bit with 120 dB SNR is used to reproduce the sound of the front channels. For all others, Cirrus Logic CS4362 114 dB SNR is applied. These microcircuits are quite good, one might say, top-end. You can even find them in sound cards that are many times more expensive.

The organization of the board itself is quite thoughtful in our opinion. A low-profile design is used, due to which ASUS Xonar DX can be installed not only in regular desktops, but also in smaller cases. For example, good sound will obviously not hurt in HTPCs based on mini-ITX boards.

Outputs and inputs are designed in the form of 3.5 mm gold-colored connectors. The optical output is combined with the analog one, so a special adapter is included in the delivery set for using it. I only disliked the close proximity, which is why high-quality cables with thick plugs simply rest against each other. But again, this is a low profile tribute.

By the way, the board has a standard plug for connecting the front connectors on the front of the computer. It is very comfortable. By the way, for the ASUS Xonar DX to work, you need to supply additional power to the board, for which a standard connector is used. This is installed in card readers and other small peripherals. In this case, we do not like its implementation, since during operation, contact may be lost, and the card simply stops working. I would like to see a more secure connector with a latch.

ASUS Xonar DX software

The set with the sound card comes with quite convenient software, which does not differ in excessive saturation of settings, but makes it possible to quickly use all the functions. The first tab of the Xonar DX Audio Center settings allows you to select the operating mode of the audio interface. First of all, the speaker configuration and sampling rate are set. Xonar DX can output both simple stereo and signal decomposition up to 7.1 mode.

The mixer allows you to adjust the volume of the individual inputs. Naturally, this can be done for both playback and recording.

The effects tab allows you to use various ready-made presets to change the character of the sound or make such settings yourself. Here we will just say that any sound card has something similar, but few prefer to use it.

Naturally, there are also various additional chips. For example, the function of suppressing acoustic communication, various voice presets for voice communication, and so on.

Testing and subjective opinion

To test the audio card, we used common software, having completed all the preliminary settings according to the manufacturer's method. I must say that the program used was first proposed by a group of developers closely associated with the famous site iXBT.com. To date, it has undergone many revisions, and, in fact, is the de facto standard in amateur measurements of the parameters of the audio path. This is largely due to the ease of use, since all you really need to get the results is the software and the connecting cable.

And the presentation of the data obtained is organized in the best way. After running for a couple of minutes, the program displays a simple plate with integral estimates plan "excellent", "good" or "bad", by which you can immediately judge the quality of the device. For a more detailed study, you can see the graphs with the spectra. As a result: in terms of the sum of such parameters as availability, measurement accuracy, ease of use, RMAA is the best option today. For example, ASUS uses it as an official measurement tool. The results for different modes of operation can be seen below.

As you can see, the ASUS Xonar DX sound card has very impressive characteristics. In 24-bit 48 kHz mode, we managed to achieve a noise level of -111 dB, a dynamic range of 111 dB, and harmonic distortion - a negligible 0.0007 percent. In general, the result is really excellent. Only the interpenetration of the channels turned out to be not very good, but this can be attributed to the nuances of a particular instance or the subtleties of the computer used. In general, there are more than enough such numbers to ensure an excellent sound level.

To make our results more understandable, let's look at them in comparison with the built-in sound using the example of the Realtek ALC889 controller. The first parameter - the unevenness of the amplitude-frequency characteristic - speaks for itself. It indicates how much decibel the signal (sound pressure) differs from the target level at different frequencies. While this parameter is extremely important for acoustics, for modern cards it is purely nominal. For example, for ASUS Xonar DX we have +0.01, -0.07 dB. In fact, it is a straight line. Hearing this difference is simply unrealistic. Embedded audio provides the same result.

The noise level characterizes the quality of the entire path, showing how large the intrinsic noise is when there is no signal at all. In our case, we have -111 dB, the best controllers on motherboards give about -90 dB. This is already a very tangible difference, showing the superiority of high-quality cards. In fact, the dynamic range also depends on this parameter, which, in simple terms, indicates the difference between the minimum and maximum signal levels that the device can reproduce. Naturally, the higher it is, the better. In our case - about 110 dB, for a good built-in audio, for example, the same ALC889 - a maximum of 90 dB.

Distortion, both harmonic and intermodulation, characterizes the nonlinearity of the path. Simply put, the percentage of "garbage" in the original signal. The available parameter at the level of 0.0007% lies in the field of theoretical research, and cannot be heard by the ear.

In fact, all these numbers are beyond the sensitivity for the human ear, but surprisingly, there is a subjective difference. To confirm this, we performed a subjective comparison test using KRK V8 v2 near-field monitors. They took the ESI audio interface as opponents [email protected] and embedded audio on Realtek ALC889. When listening to various material, the differences between the Xonar DX and [email protected] we did not notice, at least not explicit. But the superiority over the ALC889 is felt. This is clearly seen in the detail of the midrange.

Conclusion

As a result, we can say that in today's reality for ASUS Xonar DX is a good choice for a multimedia computer. The card is affordable, good in games, and provides a sound level suitable even for very serious acoustics. Naturally, you can fully use the headphones.

Do not forget about the additional advantages, for example, the low-profile design, due to which the solution is perfect for HTPC.

Pros:

  • affordable price;
  • high class converters;
  • low profile design.

Minuses:

  • close arrangement of connectors;
  • combined input for microphone and line input;
  • flimsy power contact.

Price

You can buy ASUS Xonar DX for about 2,500 rubles, and it really has few comparable worthy competitors. The closest option is probably the Creative SoundBlaster X-Fi Titanium. This card is no less versatile. It is suitable for gamers, and provides a decent sound level for high-quality audio lovers.



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