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The Xonar Essence STX Q/A, tweaking, impressions thread

Discussion in 'Computer Audio' started by telix, Apr 28, 2009.
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  1. spacedskunk
    I'm reading through this massive thread now and trying to pick out an answer about op amps on the Essence ST.
    I bought 2x LME49720NA's replacing the stock and left the other stock, and it was a mistake in reality since I've got AKG Q701's and even though I'm finding the massive soundstage and crisp clear mids/highs the loss of bass has ruined it somewhat.
    What op amp set would you guys recommend for these headphones?
  2. PurpleAngel Contributor
    I like my AD797BRs, $60 for 3 (6 op-amps mounted on three adapters).
    The LME49990 seem well liked, $60 for 3 (6 op-amp chips mounted).
  3. WiR3D
    PurpleAngel's recommendations are good as always. and the AD797 has particular synergy with my AKG K242HD, clarity is astounding, but they are unstable, and may pack up eventually. Small price to pay. Also if you want a smoother presentation, then THS4032. But the AD797 is hard to beat.
    I actually trust a SW resampler more then the driver one of the Xonar, a good SW resampler that is. The one built into the Bass library that musicbee uses is good, no idea about foobar - if someone can recommend a good one I will update the wiki. 
    upsample to 48kHz. Its probably a problem with the drivers knowing asus. And you can change the settings for ASIO, whilst its not in use (aka no playback)
  4. gtiboy
    I use 3 x LME49720HA in my STX and it sounds great, much better separation of instruments and more clarity.
    How would the STX sound if I was to install LME49710HA in the buffer while still using the 2 x LME49720HA in the IV?
  5. stv014
    Here is a short comparison of the hardware 44.1 kHz sample rate, and resampling to 48 kHz in foobar2000 with the PPHS resampler, and foo_dsp_resampler_0.8.0.zip from here. All the tests were performed under the following conditions:
    - loopback from the 1/4" headphone output jack to the line input
    - WASAPI output, 24 bit resolution
    - sound card configuration: 64-300 Ω gain setting, -6.5 dB volume for tests 1 and 2, -4.5 dB volume for tests 3 to 6, "sharp roll-off" DAC filter, 64x oversampling
    - foobar2000 volume: 0 dB for tests 1 and 2, -2.02 dB for tests 3 to 6
    For the first set of tests, I used a 7-bit (127 samples) MLS at 25% of the 0 dBFS level. This is a periodic waveform with a fundamental frequency of 347.24 Hz, and all harmonics have the same amplitude. So, the spectrum of the output of a mathematically "ideal" converter would have peaks at equal magnitude at integer multiples of 347.24 Hz up to 22050 Hz, and nothing else. However, a slower roll-off has the advantage of less ringing. Basically, the filter used has to make a trade-off between imaging/aliasing, rolled off high frequency response, and ringing. The SoX resampler also allows the choice between a linear or a minimum phase filter; the latter avoids pre-ringing, but has a non-flat phase response (the group delay increases at the highest frequencies). The hardware DAC filter, the PPHS resampler, and the default setting of the SoX converter are all linear phase. Cirrus Logic DACs (such as the CS4398 on the Xonar D1 and DX) tend to use a minimum phase filter. Differences in the filtering are not that important, though, converters that have a flat response and no significant aliasing in the audio frequency range are likely to sound the same in practice.
    The results, from left to right and top to bottom:
    1: 44.1 kHz sample rate, no conversion
    2: PPHS resampler, 44.1 -> 48 kHz conversion, "ultra mode" is disabled
    3: PPHS resampler, "ultra mode" is enabled
    4: SoX resampler, 44.1 -> 48 kHz conversion, default settings ("normal" quality, 95% passband, "allow aliasing" disabled, 50% phase response)
    5: SoX resampler, "best" quality
    6: SoX resampler, "best" quality, "allow aliasing" is enabled
    1, 2:
    fr_no_resample.png     fr_pphs_normal.png
    3, 4:
    fr_pphs_ultra.png     fr_sox_normal.png
    5, 6:
    fr_sox_best.png     fr_sox_best_a.png
    Only the PPHS converter in normal mode has noticeable aliasing, but it is still at a fairly low and likely not audible level. The PPHS "ultra" mode is rolled off very steeply, and it therefore rings the most. The increased noise floor in the hardware 44.1 kHz mode is quite apparent.
    A second set of tests used two high frequency sine waves mixed in a way that the peak amplitude of the reconstructed analog signal is about +2 dBFS, to check for clipping in the DAC or the sample rate converters. The numbering of the graphs is the same as above. I did not include the last three pictures, as they do not look significantly different from the third one.
    Without resampling, there is no major problem, although note that there would be clipping on the line output at the maximum volume. The second graph is the only case when I run the sample rate conversion in foobar2000 at 0 dB volume, and the output is indeed clipped. Decreasing the foobar2000 volume by 2 dB, and increasing the hardware volume by the same amount fixed the clipping. The effect might not be audible on music, but the workaround does not really involve a trade-off either, as the noise floor on the headphone output is not increased noticeably.
    1, 2:
    cl_no_resample.png     cl_pphs_normal.png
    WiR3D likes this.
  6. WiR3D
    Bravo stv014!!! Again a wonderful job, as always, and above my head somewhat. So the conclusion is? 
    Actually wait, the cut off should be 22050Hz? So just based on where they end, graphs 3 and 4 look best. But my question is, in graph 3 is the HW output set to 48kHz or 44.1kHz (thus leaving the drivers to do the resampling) which would be quite a surprise. 
    Also in graph one, Is the HW output set to 48kHz or 44.1kHz (basically using HW resampling, which would be quite horrendous, or is it demonstrating the 44.1kHz@24bit bug?)
    Now I'm even more curious to see how the Bass library resampler of MusicBee performs.
  7. stv014
    The hardware is running at 44.1 kHz only in the case of graph 1, and 48 kHz for all others, which resample the 44.1 kHz input file using the foobar2000 "PPHS" and "SoX" sample rate converters.
  8. WiR3D
    so then the best one is 
    3: PPHS resampler, 44.1kHz->48kHz "ultra mode" is enabled
    followed by, surprisingly.
    4: SoX resampler, 44.1 -> 48 kHz conversion, default settings ("normal" quality, 95% passband, "allow aliasing" disabled, 50% phase response)
  9. stv014
    The filtering in the SoX resampler is configurable, and can be made similar to the PPHS "ultra" mode if you increase the passband setting. However, the default should be fine. I recommend reading the documentation of the "rate" effect in the SoX manual for more information, and the HydrogenAudio threads I linked earlier.
  10. spacedskunk
    Thanks for your advice on this, I have to admit I'm struggling to find any of these in the UK. On eBay all I'm finding is this;
    Thing is I can't solder myself do I'll have to get them ready to plug in. :frowning2:
  11. WiR3D
    The reason I ask is I use MusicBee not Foobar. I'm just trying to find the bottom line for the noobs that pop in here and ask for settings (i.e. I want to update the wiki)
  12. PurpleAngel Contributor
  13. geoxile
    Quick question, did the STX get discontinued? Can't find it at any big vendors.
  14. DamageInc77
  15. geoxile
    Oops, thought those were just all marketplace vendors. But it looks like Tigerdirect/CC has it
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