I have read the thread Coris started with its excellent mods but unless I missed something they all appear to be for the analog output side of the STX card. I recently purchased an STX to replace a very old PCI sound card for the purpose of moving LP music to digital. The problem I am having (other than very poor Asus support) is that even with every slider I can find set at 100% and my RIAA preamp adjusted almost to the clipping point, the signal level in Audacity very low. Are there any known mods to improve the line-in signal level with this card? Thanks.
The only Effect/Amplify I can find is in Audacity and as near as I can tell it can only be applied after the recording in a manner similar to Effect/Normalize. I was looking for a mod or solution that would provide normal signal levels direct from the STX . With my previous sound card I could adjust the levels to much less than 100% and didn't have to worry about over driving the preamp to just to have any signal.
.
.
I had a large post typed out and I lost it all by mistake >.<
Either way the input to the Xonar follows the data sheet for its ADC fairly well. The first pair of opamps that you reach are a unity buffer followed by an inverter. This creates a balanced feed. This then goes into another pair of opamps that condition the signal prior to entering the ADC.
The first opamps aren't really suitable for modding. The second pair are but I would do so with caution. The Xonars PCB is quite fragile and the surface mount parts are quite small and tightly packed. Not to mention there are plastic SMD caps here too. These do not take kindly to an accidental poke from a soldering iron, often outright failing if you do so.
Either way I have attached an image that shows you what is what.
I strongly recommend you use a preamp before the xonar instead, or mod your phono stage to output more voltage perhaps.
Either way the input to the Xonar follows the data sheet for its ADC fairly well. The first pair of opamps that you reach are a unity buffer followed by an inverter. This creates a balanced feed. This then goes into another pair of opamps that condition the signal prior to entering the ADC.
The first opamps aren't really suitable for modding. The second pair are but I would do so with caution. The Xonars PCB is quite fragile and the surface mount parts are quite small and tightly packed. Not to mention there are plastic SMD caps here too. These do not take kindly to an accidental poke from a soldering iron, often outright failing if you do so.
Either way I have attached an image that shows you what is what.
I strongly recommend you use a preamp before the xonar instead, or mod your phono stage to output more voltage perhaps.
Attachments
I am sorry you had to do it twice. I know from personal experience how frustrating it can be. More than once I have taken so much time on a post for one reason or another that when I reached the point to submit the post I was asked to log in again which resulted in the post being dumped. In any case, I certainly appreciate the time you have put in on replying. Thank you.
I have a Xytronic digital soldering station that I have had for years and I have used it many times on radios and other electronic equipment. The caveat however is that most of the equipment I have worked on had very little in the way of SMD components and I never had to try soldering or desoldering them. After looking your jpg, I think it is safe to say this is definitely not a good first time project to develop those skillsl. In addition although the hands are still steady, the eyes are not what they used to be. All in all, I think the smart move is to take a pass on the mod.
As for trying to increase the signal levels prior to them reaching the card, I seem to recall in another post either here or on the Asus forum that the STX is very unforgiving to strong input signal levels and that you can without too much effort over drive it to the point of failure. Based on this perhaps that prudent thing to do is not modify the preamp or add another stage of amplification.
I know you have already spent considerable time so I don’t expect you to add anything further but perhaps someone else will. It would appear to me that if the volume is low in Audacity that might mean that the digital information that is being placed on the bus by the card is not using all the bits that could be used to reflect changes in volume and that in turn would mean less dynamic range? Is that really the case or is the Xonar Audio Center and/or driver software not performing correctly and actually reducing the original volume after the data leaves the ADC. If isn’t in the software, would amplification after the fact in Audacity really help or would you only increase all volume levels an equal amount with dynamic range remaining the same?
I guess the smart thing is to find out the maximum safe signal level at the line-in input and then look at the output of the preamp with a scope to determine normal peak levels and see if there is any room for improvement..
The bottom line is that this has been a very disappointing purchase and experience. I would have thought that if you purchased a reasonably good sound card that you could just install it and have it work properly.
The inputs to the Xonar wont fry unless you feed them with a signal that is far past the clipping point. All that happens is that the first opamp in the input chain gets its inputs fried, replacing the opamp solves the problem. The opamps used cost very little but if you want to replace them you need to be careful while doing so.
The input to the Xonar is configured so that an analogue signal of ~2Vrms = the maximum amplitude on the digital scale (0dBfs) from the ADC. This is the normal standard voltage level that CD players, DVD players etc will output when reproducing a full scale digital signal. If you exceed this value the inputs clip but you have to overdrive them quite a lot for them to blow up.
Now the control centre is very basic, all it does is digitally attenuate the signal coming out of the ADC. This means that if the analogue signal level reaching the ADC is clipping the ADCs inputs, that turning the inputs down in the control centre wont do anything to stop the inputs clipping. All you do is turn down the clipped signal digitally.
All sound cards tend to work in this kind of way, if any change in levels is needed it is all done digitally, this is because we don't exactly have potentiometers wired up to our sound cards such that we can alter signal levels in the analogue domain. This means that if the Xonar did offer some gain on the inputs it would most likely mean that it would be done digitally anyway, after the signal had been digitised. The headphone output on the Xonar for example can be configured to work at different output levels depending on the sensitivity of your headphones, this is done in the digital domain. The PCM1792 DAC has a built in volume control and all the Xonar does is turn the output down via the DAC if you select 64ohm sensitive headphones.
The input to the Xonar is configured so that an analogue signal of ~2Vrms = the maximum amplitude on the digital scale (0dBfs) from the ADC. This is the normal standard voltage level that CD players, DVD players etc will output when reproducing a full scale digital signal. If you exceed this value the inputs clip but you have to overdrive them quite a lot for them to blow up.
Now the control centre is very basic, all it does is digitally attenuate the signal coming out of the ADC. This means that if the analogue signal level reaching the ADC is clipping the ADCs inputs, that turning the inputs down in the control centre wont do anything to stop the inputs clipping. All you do is turn down the clipped signal digitally.
All sound cards tend to work in this kind of way, if any change in levels is needed it is all done digitally, this is because we don't exactly have potentiometers wired up to our sound cards such that we can alter signal levels in the analogue domain. This means that if the Xonar did offer some gain on the inputs it would most likely mean that it would be done digitally anyway, after the signal had been digitised. The headphone output on the Xonar for example can be configured to work at different output levels depending on the sensitivity of your headphones, this is done in the digital domain. The PCM1792 DAC has a built in volume control and all the Xonar does is turn the output down via the DAC if you select 64ohm sensitive headphones.
Last edited:
ADC chips in soundcards often feature analog input attenuator (IPGA), hooked by I2C/IntelHDA/some 3wire to the soundcard chip, controlled thus by the driver. I have never seen an attenuator before the input opamp, but the input attenuator in the actual ADC chip is analog. For details see e.g. http://www.akm.com/datasheets/ek4620a.pdf , DAC/ADC used in Infrasonic Quartet.
Interesting, it's the first chip I've seen with something like that. I can tell you that that isn't the case for the Xonar though ;/ What'd be perfect is if they used something like the CS3318 volume control chip. This isn't cheap but it has the performance and is flexible. If used correctly one chip could easily adjust the gain in the analogue domain for both the input and the output sections simultaneously.
I can't imagine any consumer grade sound card ever using something like that however, unless it'd fit right into the marketing of the card as being the reason to buy this card over something else.
IPGAs integrated into the ADC chip are not that uncommon. E.g. many cards use AKM ADCs, and only AK4528 used in Juli/Delta has no IPGA - git.alsa-project.org Git - alsa-kmirror.git/blob - i2c/other/ak4xxx-adda.c . Others (ak4524 - Terratec, Hoontech, WM8776 -
Maya44, Prodigy Hifi, WM8770 - Terratec Phase, STAC 9460 - Prodigy192, Waveterminal192, AK5365 - Revolution5.1 etc.) have either full IPGA or at least mic input volume control. For details check linux alsa source code git.alsa-project.org Git - alsa-kmirror.git/tree - pci/ .
Are you sure? I do not have the card, but it appers to have CMI9780 before the CS5381 ADC Asus Xonar Xense Premium Gaming Audio Set Review Page 2 - Closer Look: Sound Card - Overclockers Club . git.alsa-project.org Git - alsa-kmirror.git/blob - pci/oxygen/xonar_pcm179x.c While the actual ADC has no input gain control, the AC97 mixer CMI9780 seems to provide mixing and input gain control capability git.alsa-project.org Git - alsa-kmirror.git/blob - pci/oxygen/cm9780.h git.alsa-project.org Git - alsa-kmirror.git/blob - pci/oxygen/oxygen_mixer.c
I guess I should rephrase what I said previously. I've only ever paid attention to the top line DAC/ADCs from semi manufactures and these don't tend to have that functionality. It isn't at all a surprise to find that some of the multi channel or 'all in one chips' have variable analogue gain. Now that I know some can do that it would make sense if all ADCs would come with some variable gain on the input but perhaps that would deteriorate the performance somehow. I do know that most digitally controlled pots tend to have rather lacklustre distortion performance. The CS3318 is one of the only products out that bucks the trend.
Yeah the Asus switches the input between two different sets of circuitry with a relay. One set for the microphone, which goes through the 9780 and then the other set which simply comprises of a few opamps and then the CS5381 for the line in. I also tested this out before posting by feeding a signal that was slightly too hot into the Xonar so that I had clipping on the spectrum analyser, which I then turned down using the software and the spectrum remained clipped, only then attenuated by X dB.
Interesting discussion, but I think it is time for me to bow out. 5th element, thank you very much for taking the time to explain a few things to me in your last post in response to mine. After your comments about the Line-In input not being as sensitive as I was suggesting, I went back and found the post. It is here
http://vip.asus.com/forum/view.aspx?id=20090519034729846&board_id=21&model=Xonar+Essence+STX&page=1&SLanguage=en-us
or here
http://tinyurl.com/6wp6r33
After reading it a second time I noted in his second post further down the page that he back pedaled a bit so apparently you are right.
It is too bad you and phofman were not on the design team for the STX. It might have turned out to be a much better card for those looking for a good line-in input. On the other hand the bean counters would probably have still won in the end. Such is life. Thanks again.
P.S. I finally heard from Asus and their only suggestion was to RMA the card if I thought there was a problem. They either don't get it, or they are choosing not to get it.
http://vip.asus.com/forum/view.aspx?id=20090519034729846&board_id=21&model=Xonar+Essence+STX&page=1&SLanguage=en-us
or here
http://tinyurl.com/6wp6r33
After reading it a second time I noted in his second post further down the page that he back pedaled a bit so apparently you are right.
It is too bad you and phofman were not on the design team for the STX. It might have turned out to be a much better card for those looking for a good line-in input. On the other hand the bean counters would probably have still won in the end. Such is life. Thanks again.
P.S. I finally heard from Asus and their only suggestion was to RMA the card if I thought there was a problem. They either don't get it, or they are choosing not to get it.
There's no doubt that the line input on the Xonar is very high quality, I use it to perform measurements and Stereophile's own measurements of the card only back up how vice free it appears to be. Sure you can do better but only if you really want to spend high.
The fact that the ASUS does not provide any gain on the input is really one of minor consequences and is certainly not any fault of the original designer. Having the ADC reach 0dBfs when fed with a 2Vrms signal is the standard and usually it is better to follow standards, rather then to ignore them.
Now as to your problem, what kind of level issues are we talking about? If you record something with Audacity for example, how much gain can you apply without clipping the signal?
Aside from that though record players don't tend to deliver particularly high signal to noise ratios. The ASUS, from my use reaches around the 120dB mark which is nearly as good as it can get, the input stage is very quiet. If you were to make a recording from a record that has a signal to noise ratio of 70dB, then you'd be free to apply up to around 50dB of gain in audacity before the cards performance starts to contribute.
If the peak level on the record reaches -30dB on the digital scale, the noise of the record would lie around -100dB. If you applied 30dB of analogue gain before the card, so that the peaks hit 0dB, the noise would hit -70dB. In both situations the card would accurately capture the signal applied to it. If you use analogue gain before the card, or digital gain after it, the end result would be the same, a noise floor of -70dB as dictated by the vinyl medium with the audio reaching peaks of 0dBfs.
The fact that the ASUS does not provide any gain on the input is really one of minor consequences and is certainly not any fault of the original designer. Having the ADC reach 0dBfs when fed with a 2Vrms signal is the standard and usually it is better to follow standards, rather then to ignore them.
Now as to your problem, what kind of level issues are we talking about? If you record something with Audacity for example, how much gain can you apply without clipping the signal?
Aside from that though record players don't tend to deliver particularly high signal to noise ratios. The ASUS, from my use reaches around the 120dB mark which is nearly as good as it can get, the input stage is very quiet. If you were to make a recording from a record that has a signal to noise ratio of 70dB, then you'd be free to apply up to around 50dB of gain in audacity before the cards performance starts to contribute.
If the peak level on the record reaches -30dB on the digital scale, the noise of the record would lie around -100dB. If you applied 30dB of analogue gain before the card, so that the peaks hit 0dB, the noise would hit -70dB. In both situations the card would accurately capture the signal applied to it. If you use analogue gain before the card, or digital gain after it, the end result would be the same, a noise floor of -70dB as dictated by the vinyl medium with the audio reaching peaks of 0dBfs.
I finally found linux amixer list of STX controls https://launchpadlibrarian.net/95171160/Card0.Amixer.values.txt These controls show real hardware capabilities as the guy who wrote linux STX driver has access to Asus documentation for the card. There is no line input gain control listed, only the mic gain. The side input aux connector offers gain control but that is just an auxiliary feature.
First of all my apologies for not responding to your post sooner.
What you have said makes sense and I also did some follow up of my own. I found the paperwork that came with the preamp and it indicated that the preamp has a maximum output of 1.4 Vrms. In theory that is 70% of the maximum input level of the card. I wasn’t seeing values that high in Audacity but my guess is that I had the preamp adjusted down from the maximum value to make sure I was avoiding clipping in the preamp. At this point I stand corrected on the cards performance and my choices are another preamp with more output or amplify it after the recording session in Audacity. Based on what you said the advantage would be to do it in Audacity. It saves the expense of another preamp and I will never have to worry about clipping occurring during the actual translation from analog to digital. Thanks again for taking the time to respond.
You're welcome.
I think it is also perhaps worth pointing out that analogue and vinyl are slightly different from digital when it comes to absolute signal levels. Digital is extremely easy as the maximum possible equals data encoded at 0dBfs on the digital medium. As a result of this it is also easy to define guidelines on what a 0dBfs signal should represent in the analogue domain. In this case 2Vrms is the standard.
With a record this isn't quite the same, I am sure there is a size that could be cut into a groove that would represent something of a maximum, but the trouble here is that some styluses might be okay with a certain groove size, whereas others might jump the groove etc. Then coils come in all sorts of different types and sensitivities, making it difficult to define an absolute maximum. In this regard you could monitor the output of the preamp and find that with 10 records the signal doesn't clip and set the gain accordingly, however on playing back an eleventh record you find that it does clip.
It makes sense to err slightly on the side of caution, by keeping the pre amps gain lower, so as to have more headroom available should you encounter a record with higher then average peak levels.
I think it is also perhaps worth pointing out that analogue and vinyl are slightly different from digital when it comes to absolute signal levels. Digital is extremely easy as the maximum possible equals data encoded at 0dBfs on the digital medium. As a result of this it is also easy to define guidelines on what a 0dBfs signal should represent in the analogue domain. In this case 2Vrms is the standard.
With a record this isn't quite the same, I am sure there is a size that could be cut into a groove that would represent something of a maximum, but the trouble here is that some styluses might be okay with a certain groove size, whereas others might jump the groove etc. Then coils come in all sorts of different types and sensitivities, making it difficult to define an absolute maximum. In this regard you could monitor the output of the preamp and find that with 10 records the signal doesn't clip and set the gain accordingly, however on playing back an eleventh record you find that it does clip.
It makes sense to err slightly on the side of caution, by keeping the pre amps gain lower, so as to have more headroom available should you encounter a record with higher then average peak levels.
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.