Why, yes, I have! Right up close too. That's the one I have, actually. It sounds really good with my DAC board. Just have to remember in Windows to set the output level down a bit, maybe to 92, in Windows sound control panel for the USB board feeding the SRC. That way there is always some headroom to prevent clipping during AK4137 SRC processing and also in the ES9038xxx anti-jitter ASRC processing.
If you wanted to improve your PC based audio, it is time to try this (instead of Windows):
AP-Linux-V4 available | AudioPhile Linux
AP-Linux-V4 available | AudioPhile Linux
Avcc evolved in the last 10 years
Here goes a good summary of evolution of AVCC by glt (he is a famous author of Hifiduino and lots of other Sabre based audio projects).
B-II | H i F i D U I N O
Here goes a good summary of evolution of AVCC by glt (he is a famous author of Hifiduino and lots of other Sabre based audio projects).
B-II | H i F i D U I N O
Hi Janos,
thanks for this interesting link. Seems TPA have also played with different opamps and circuits on the latest developement of AVCC PSU boards(trident) - a pitty they are so expensive and no circuits available online - but ok, it is theire knowledge they want to make money with...
cdsgames went a different way with their Katana which soon will be released, adding a super cap and a ladder of paralleled film (?) caps with smaller values the closer you get to the DAC chip AVCC input.. dont know about the output LF Impedance behaviour and PSSR of this solution compared to an opamp reg. but seems they also investigated a lot here.
thanks for this interesting link. Seems TPA have also played with different opamps and circuits on the latest developement of AVCC PSU boards(trident) - a pitty they are so expensive and no circuits available online - but ok, it is theire knowledge they want to make money with...
cdsgames went a different way with their Katana which soon will be released, adding a super cap and a ladder of paralleled film (?) caps with smaller values the closer you get to the DAC chip AVCC input.. dont know about the output LF Impedance behaviour and PSSR of this solution compared to an opamp reg. but seems they also investigated a lot here.
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The AVCC solution I used for ES9038Q2M has its own dedicated reference voltage regulator, and is not taken from the digital 3.3v power. It is also located on the other side of the ground plane from the digital circuitry, keeping it well shielded from their local electric fields. It is also located as close a physically possible to the AVCC pins on the DAC chip. That makes it pretty similar to what was described for some of the later AVCC iterations for Buffalo DACs. Also, one important thing that is different between Buffalo and the DAC board in this thread is that Q2M requires much less AVCC current than a PRO chip with 4 channels in parallel for each stereo output channel. Since this DAC needs much less AVCC current, an LME49720 based AVCC supply can work fine here.
Regarding a version of Linux for audio, it wasn't immediately clear to me why it would be used instead of Windows. We know how to use Windows for 'bit perfect' audio, but it does take some attention to it since it is not guaranteed by default due to resampling sometimes performed by Windows sound engine. So long as it is used correctly, it works reliably, and it along with MacOS are still the two main standards for professional audio use. For instance, I don't think Pro Tools is available for Linux, nor are their drivers available for much of the associated specialized hardware.
Sure. I went to the AP-Linux website to look. It just wasn't immediately clear to me what would make it an improvement over Windows or MacOS? I did notice where it said 'bit perfect' a few times, but since other OS choices can do that I was looking for some statement of benefits or something that would help to understand why it would be an improvement. Maybe I just missed it?
It is claimed that the OS is stripped down to a minimum so that CPU and RAM are not busy with other processes that are not relevant for sound reproduction. It is often stated that a high and irregular CPU and RAM load can have negative impact on SQ. Maybe related to noise modulation that is caught by other components if not well isolated.
I think in a well isolated setup with a clean and well isolated PSU for the Audio interface with its own clock circuit this should not make a difference for 2 bit perfect systems.
Anyway, as nothing is perfect, it should be no disadvantage to keep the systems simple.
I think in a well isolated setup with a clean and well isolated PSU for the Audio interface with its own clock circuit this should not make a difference for 2 bit perfect systems.
Anyway, as nothing is perfect, it should be no disadvantage to keep the systems simple.
For and older and or slower computer Linux might very well make good sense. Particularly one optimized for home entertainment or something of that nature.
Around here and with some people I help out for fun sometimes, we record, mix, and sometimes even master records for small time groups. With a lot of tracks and plugins using Pro Tools or some similar DAW program it is possible to bring a fairly powerful computer down to its knees, so to speak.
Computers can certainly run low on resources and that is when there can start being resulting problems with sound. Usually, simple stuttering or choppy playback is a common symptom if a computer is unable to keep up with high priority threads in real time, such as for audio playback buffers. If you look in task manager and see that memory and or cpu are getting above 80% utilization then that is getting up there close where problems may start occurring, although it may not get to being an actual problem up to closer to 90%.
On the other hand, at least in my experience, if a computer is not getting up close to 80% for things like cpu and or memory, or maybe the disk, it should be able to keep up with higher priority stuff like audio and the video display threads.
Maybe to put it another way, on occasions troubleshooting some computer audio problems, can't seem to recall any problem caused by the computer getting loaded down with processing that didn't also show corresponding indications in task manager or process monitor.
Around here and with some people I help out for fun sometimes, we record, mix, and sometimes even master records for small time groups. With a lot of tracks and plugins using Pro Tools or some similar DAW program it is possible to bring a fairly powerful computer down to its knees, so to speak.
Computers can certainly run low on resources and that is when there can start being resulting problems with sound. Usually, simple stuttering or choppy playback is a common symptom if a computer is unable to keep up with high priority threads in real time, such as for audio playback buffers. If you look in task manager and see that memory and or cpu are getting above 80% utilization then that is getting up there close where problems may start occurring, although it may not get to being an actual problem up to closer to 90%.
On the other hand, at least in my experience, if a computer is not getting up close to 80% for things like cpu and or memory, or maybe the disk, it should be able to keep up with higher priority stuff like audio and the video display threads.
Maybe to put it another way, on occasions troubleshooting some computer audio problems, can't seem to recall any problem caused by the computer getting loaded down with processing that didn't also show corresponding indications in task manager or process monitor.
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I've been looking at buffers to use as an output stage and there are a few types so I need soem guidance.
This one uses a pair of NE5534's:
NE5534 DC Buffer HIFI Stereo OP-AMP Preamplifier Preamp Board support OPA627 | eBay
This one uses four NE5532's:
Balanced to Unbalanced XLR to RCA Stereo Preamp Board ultra low distortion 699962939918 | eBay
Then there are two different transistor based units:
JFET input Cascoded Buffer Board Preamplifier assembled ! 699957464661 | eBay
Diamond Buffer Board Stereo Broadband Strong Driving Capability Assembled ! 699918298748 | eBay
This one uses a pair of NE5534's:
NE5534 DC Buffer HIFI Stereo OP-AMP Preamplifier Preamp Board support OPA627 | eBay
This one uses four NE5532's:
Balanced to Unbalanced XLR to RCA Stereo Preamp Board ultra low distortion 699962939918 | eBay
Then there are two different transistor based units:
JFET input Cascoded Buffer Board Preamplifier assembled ! 699957464661 | eBay
Diamond Buffer Board Stereo Broadband Strong Driving Capability Assembled ! 699918298748 | eBay
Ok, so AP is maybe not for U. To make a long story short ;-) all I2S (USB etc) users of 9038Q2M board owners are welcome to try AP-linux and compare to other OS! Why, because 9038Q2M is good enough to show us the differences between AP-linux and MS. What we need, just a hard drive. I have AP installed on an old 2.5" SATA HD. That is all.
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I have to mention, that the link I posted is more than 4 years old, so there are some improvements still done (like TPA's new circuit) you have mentioned. Also, John Walton published a paper about measurements of some power supply designs. Based on their measurements, there are a few of them with an output impedance in the 10x µOhm range, may matches well to power the AVCC:
Linear Audio | your tech audio resource
Linear Audio | your tech audio resource
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jonos_904, I sure hope they are not different, that is, if configured so they shouldn't be different. If they are different and Windows sounds worse, as you probably already know most modern OS's including MacOS and maybe non-audio-specific Linux can and do use a default sample rate for all audio sent to a sound card. That means they will perform real time Sample Rate Conversion (SRC) to convert any non-default sample rates to be played to the default sample rate. They particularly do it if the sound card device is set to be the system default device. The reason they do it is so that multiple sound streams from different applications can play at once.
The most reliable method to prevent problems is to use an ASIO driver for a music sound card and make sure it is not set as the default sound device. Foobar can then be configured to use the ASIO sound device.
Otherwise, it may be necessary to set the sound device default sample rate to the sample rate of the music file you want to play. On the one hand, it is a bit of pain, no question, and especially if only one sound card in the computer. On the other hand it means that it is still possible to run all the applications that may be needed for other things besides playing audio. Although, understood that there are a lot of Linux apps that these days that could meet people's main needs.
Anyway, I would agree the 'feature' of most modern operating systems we are trying to overcome, the ability to play audio from multiple apps at once is a problem for be audio SQ. It turns out Windows does have a sound driver model that allows an app to override that default behavior but almost no device drivers has been written that support it. As of yet, anyway.
It could well be that the easiest solution for some people would be to dedicate an old computer to playing audio and to install an audio player specific OS to prevent problems from occurring in the first place. Then there would be no need to worry about it more after that.
The most reliable method to prevent problems is to use an ASIO driver for a music sound card and make sure it is not set as the default sound device. Foobar can then be configured to use the ASIO sound device.
Otherwise, it may be necessary to set the sound device default sample rate to the sample rate of the music file you want to play. On the one hand, it is a bit of pain, no question, and especially if only one sound card in the computer. On the other hand it means that it is still possible to run all the applications that may be needed for other things besides playing audio. Although, understood that there are a lot of Linux apps that these days that could meet people's main needs.
Anyway, I would agree the 'feature' of most modern operating systems we are trying to overcome, the ability to play audio from multiple apps at once is a problem for be audio SQ. It turns out Windows does have a sound driver model that allows an app to override that default behavior but almost no device drivers has been written that support it. As of yet, anyway.
It could well be that the easiest solution for some people would be to dedicate an old computer to playing audio and to install an audio player specific OS to prevent problems from occurring in the first place. Then there would be no need to worry about it more after that.
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Could be. But obviously almost any length of wire connecting something like that to AVCC pins will degrade performance. One would have to be very careful indeed to prevent that from happening. Most of the pics I have seen of people trying different AVCC supplies have interconnecting wires with resistance that appears to be ignored. If someone were very careful to make sure that the resistance to the AVCC pins was small enough then it might be more of a fair comparison between different regulators.
Also, most people don't do very scientific listening comparisons, which actually is not an easy thing to do well. People like Jakob2, DPH, etc., have pointed out some of the issues that can cause errors in listening comparisons and make them unreliable. On that, I think I would have to mostly agree with those folks.
Bottom line for me is that the ESS circuit with this board if laid out about the way I did it is probably fine. At least the wire lengths are minimal, it is well shielded from digital noise sources on top of the board, and return impedance through ground to the dac chip is about as low as it can be. Therefore, I'm not convinced another approach would be audibly better or has ever properly been shown to be audibly better.
A more interesting set of problems might be how to do some of the things like AVCC and I/V stages for ES9038PRO, where output related currents are much higher and thus necessitate development of new circuits.
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@Ian, You can use LME497xx family opamps for buffers just fine (including LME49860). Just be sure to do as ESS recommends and use good quality SMD thin film resistors, C0G caps for any caps in the signal path (unless you must use big DC blocking electrolytics, then make sure they have very low impedance even down at 20Hz). Also use 10uf tantalum caps in parallel with X7R .1uf ceramic caps to bypass each IC at each of it's power pins to a proper ground plane. Do not use cheap ebay boards for this unless you want worse sound quality than this DAC chip is capable of providing. It is hard to get down to -120dB distortion and easy to add distortion back in if sufficient care is not taken.
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I don't think I'm capable of designing my own buffer then fabricating it without having a PCB to work with.
There is this LME49720HA + LME49600 based PCB available:
LME49720HA + LME49600 Headphone Amp Pre-amp Audio Signal Amplifier Buffer Board 699934972165 | eBay
Then there is this, the Walt Jung Super Buffer, which requires one per channel:
Super buffer w/ low noise regulator Walt Jung | eBay