I'm not able to hear the difference between 192kHz vs 96kHz, and I don't think I can even tell between 44.1kHz lol. But I do have some vinyl needledrops sampled at 192kHz and couldn't notice any problems with them.
Waaay OT here, but... if there is no audible difference, then why spend the money? Engineering analysis of 192Khz states that sample rate (or rather, how we can physically implement it) can only harm the sound. So you are left with a choice to spend $$ to get little to no benefit (and maybe get something detrimental) to the perceived sound of your system.
@RocketScientist. I got ambitious today... headphone jack problem solved with some resist-tweezers and a crap ton more force than I wanted to use (at least I know it won't be coming apart any time soon). Making the holes a bit bigger is the right answer, but at least you know that the jacks can be persuaded.
The difference is on the studio side, it is easier to master a 24/192 recording, A good sound engineer like Hoffman can have RBCD sound as good as hirez but average studios do better with 24/192 since this is the native recording from thier ADC's, doesn't need as much processing(downsampling/filtering/etc) on their end. Thats all hirez is about, making things easier on the studios (mainly the audiophilish music types.) And of course charging more for less expertise.
If you have a few 24/192 recordings just use a vst to downsample and dither to 24/96, there is no audible difference unless you build a dedicated 24/192 DAC (and sacrifice RBCD quality.)
@STM, I assume you mean the holes in the PCB not the holes in the front panel?
Sampling Rates (OT) - This is off topic and it's been fairly covered elsewhere--especially on HydrogenAudio. I have to side with Benchmark, Lavry, etc. 24/96, 24/192 and DSD are very useful for recording and mastering work. And 24 bits at whatever the native sampling rate is (44 Khz for 99% of digital music) is useful for playback when you're using an "upstream" digital volume control such as a PC with a USB DAC.
Meyer and Moran demonstrated when you use all 16 bits, even 16/44 with decent hardware is sufficiently transparent that nobody could detect any differences compared to hi-res DSD from SACD in over 500 listening trials. Those with nearly superhuman listening abilities, using ideal hardware and very rare source material, might hear a slight difference under such unrealistic conditions. But that's hardly an endorsement for hi-res audio playback for the other 99.99% of listening conditions.
So while I'm a fan of 24 bit audio I believe music is best played back at its native sampling rate. If you need jitter reduction, a high quality properly implemented hardware Asynchronous Sampling Rate Converter (ASRC) is a proven way to achieve it. But you can't just link a few chips together on a PC board using an ASRC and expect to have low jitter. John Atkinson at Stereophile, and others, have demonstrated many so called "low jitter" designs are anything but.
I really believe high-end DAC design is out of reach of most DIYers as they have no way to properly measure the result. Even many popular "commercial" DIY designs, such as those from Twisted Pear and AMB, are never properly measured. So while one can debate the merits of various hi-res audio formats, I think there are bigger problems with many diyDACs and even some commercial ones.
I have an eBay headphone DAC on my bench from a relatively well known "brand" that uses all the right designer ICs, looks great, and ticks all the right boxes. It turns out the DAC chip is supposed to be configured by a microcontroller but in this design is hardwired for 24/96 operation all the time. So using the DAC via USB at 16/44, the digital filtering in the DAC isn't even trying to remove all the out-of-band garbage from 22 Khz to 48 Khz. The result are aliasing distortion products that get "mirrored" down into the audio band and are likely audible. This DAC measures great on RMAA but on the dScope it's obvious in 5 seconds it has a serious problem.
Sampling Rates (OT) - This is off topic and it's been fairly covered elsewhere--especially on HydrogenAudio. I have to side with Benchmark, Lavry, etc. 24/96, 24/192 and DSD are very useful for recording and mastering work. And 24 bits at whatever the native sampling rate is (44 Khz for 99% of digital music) is useful for playback when you're using an "upstream" digital volume control such as a PC with a USB DAC.
Meyer and Moran demonstrated when you use all 16 bits, even 16/44 with decent hardware is sufficiently transparent that nobody could detect any differences compared to hi-res DSD from SACD in over 500 listening trials. Those with nearly superhuman listening abilities, using ideal hardware and very rare source material, might hear a slight difference under such unrealistic conditions. But that's hardly an endorsement for hi-res audio playback for the other 99.99% of listening conditions.
So while I'm a fan of 24 bit audio I believe music is best played back at its native sampling rate. If you need jitter reduction, a high quality properly implemented hardware Asynchronous Sampling Rate Converter (ASRC) is a proven way to achieve it. But you can't just link a few chips together on a PC board using an ASRC and expect to have low jitter. John Atkinson at Stereophile, and others, have demonstrated many so called "low jitter" designs are anything but.
I really believe high-end DAC design is out of reach of most DIYers as they have no way to properly measure the result. Even many popular "commercial" DIY designs, such as those from Twisted Pear and AMB, are never properly measured. So while one can debate the merits of various hi-res audio formats, I think there are bigger problems with many diyDACs and even some commercial ones.
I have an eBay headphone DAC on my bench from a relatively well known "brand" that uses all the right designer ICs, looks great, and ticks all the right boxes. It turns out the DAC chip is supposed to be configured by a microcontroller but in this design is hardwired for 24/96 operation all the time. So using the DAC via USB at 16/44, the digital filtering in the DAC isn't even trying to remove all the out-of-band garbage from 22 Khz to 48 Khz. The result are aliasing distortion products that get "mirrored" down into the audio band and are likely audible. This DAC measures great on RMAA but on the dScope it's obvious in 5 seconds it has a serious problem.
I think it's mostly to attract more of the audiophile crowd. Anyways I downloaded the needle drops for free from out of print LPs, so it wasn't up to me to chose the sampling rate. But If I ever come to the choice, I'll definitely favor 96kHz as the max possible, provided that there isn't any microcontroller problems that RocketScientist pointed out.
@stockpickerhk, I'm glad the O2 is popular in HKI suspect this might be partly a language translation issue. If one battery dies first, or one battery become disconnected during use, the O2 can output 7+ volts of DC into the headphones. This will almost certainly damage many headphones. This is one of the biggest problems, along with lack of output current, of most dual battery Cmoy-type amplifiers.
But even if you use the O2 only on AC power you still need the power circuitry. For example, when you turn on the AC power, the positive and negative regulators do not turn on at the same time (due to differences in the regulator ICs). This means for a short time the O2 tries to run with only one power supply, this causes a large transient into the headphones.
In both cases above the circuit is protecting the headphones. As a side benefit, the circuit also helps protect the batteries from being discharged too low and being damaged (by cell reversal). So it's true the circuit also help protect the batteries. The circuit does both things.
is it technically incorrect to say O2 got a "headphone protection circuit" ? since a typical "headphone protection circuit" should have relays sitting between the output of an amplifier and speaker.
@stockpickerhk, there can be many types of headphone protection circuits besides a relay at the output, but you're correct that's what most people think of when you mention "protection". Even relay circuits may not be fast enough to protect some headphones against some conditions (especially circuit failures). It's very challenging to design a circuit that can respond quickly enough to a DC failure but not trip on high level deep bass signals in music. Doug Self spends several pages on the topic in one of his books.
In addition to the power control circuit, the current limiting built into the NJM4556 is also a form of headphone protection. It progressively reduces the maximum power output of the O2 into loads below 50 ohms. That can help prevent damage to more sensitive low impedance headphones if the gain is set too high and/or someone accidentally turns the volume up too high.
@Fcrawly, I have listed most of the resources I know about in the O2 Resources section. Most in the USA are buying from one of those sources, but the PC board group buy is now closed. I suspect once people start building/receiving their O2 amps, assuming they like them, we'll see more options for purchasing in North America and elsewhere. I'll be updating the O2 Resources section as new options become available.
In addition to the power control circuit, the current limiting built into the NJM4556 is also a form of headphone protection. It progressively reduces the maximum power output of the O2 into loads below 50 ohms. That can help prevent damage to more sensitive low impedance headphones if the gain is set too high and/or someone accidentally turns the volume up too high.
@Fcrawly, I have listed most of the resources I know about in the O2 Resources section. Most in the USA are buying from one of those sources, but the PC board group buy is now closed. I suspect once people start building/receiving their O2 amps, assuming they like them, we'll see more options for purchasing in North America and elsewhere. I'll be updating the O2 Resources section as new options become available.
I have finally read the Circuit Description section of O2 Details documents. Thanks for explaining the design in details!
There was on thing that I don't understand: how is 1 Ohm output resistance supposed to protect from short circuit (I mean 1 Ohm is very low and allows for 2A current with 2V output voltage), and how can an amplifier with 1 Ohm resistor have output resistance of 0.5 Ohm? Did I get the phrase about 0.5 Ohm wrong?
There was on thing that I don't understand: how is 1 Ohm output resistance supposed to protect from short circuit (I mean 1 Ohm is very low and allows for 2A current with 2V output voltage), and how can an amplifier with 1 Ohm resistor have output resistance of 0.5 Ohm? Did I get the phrase about 0.5 Ohm wrong?
@Alexium, you're welcome. Any resistance in the output, even 1 ohm, helps limit peak current levels which can increase the odds the device will survive a short circuit. The NJM4556 has internal active current limiting at around 100 mA peak per op amp section which is the main protection.
There are two amplifier sections in parallel. Each has a 1 ohm output impedance. 1 ohm in parallel with 1 ohm is 0.5 ohms.
There are two amplifier sections in parallel. Each has a 1 ohm output impedance. 1 ohm in parallel with 1 ohm is 0.5 ohms.
FYI: At least one user has already reported a bad MOSFET (Q1) which was likely damaged by ESD (static). It's really important to follow the tips about handling and soldering the MOSFETs in the Circuit Board Construction section.
This points to one more advantage of the desktop board: No MOSFETs.
This points to one more advantage of the desktop board: No MOSFETs.
No, please read:
Circuit Board Construction
Circuit Description
Cautions
The power circuit is necessary to reduce power on/off transients even with AC-only power. I've tried to make that as clear as I can. It's even on the schematic.
The majority of the radiation is probably taking place from the lead from the wallwart to the O2 itself, it's probably the most effective radiating antenna in the assembly, most of the other conductors carrying those currents are pretty small, and inside the box anyway. The pickup would then be primarily on the leads to the input. The fact that the problem is reduced by grounding the input to the box tends to support this, the RF probably stays on the outside of the box when this is done. You could take a few turns of the lead from the wallwart through a ferrite close to where it enters the box, which will reduce the RF getting onto the cable and radiating. A passthru ferrite on the PCB at the input might help too.
Thanks Counter Culture. I agree the power "cord" is part of the problem. But this isn't an RF problem. It's 60 hz electro magnetic radiation with a smaller 120 hz component. A ferrite will do nothing at power line frequencies.
The good news is grounding the enclosure, and getting rid of the cap on the AC input, has effectively eliminated the problem.
The good news is grounding the enclosure, and getting rid of the cap on the AC input, has effectively eliminated the problem.
What is radio if not 'electro-magnetic radiation'?
Of course you do get induced hum, but this is is usually where the conductors are in close proximity and the field strength obeys an inverse cubic law, as opposed to inverse square, and you wouldn't fix that by grounding to the enclosure.
120Hz doesn't propagate much, but then it's a switching waveform, so the frequencies involved are related to the risetime, not just the pulse repetition frequency.
'Course I haven't seen what the effect is at the audio output.
It's a comparatively trivial current.
You have to wonder why the problem doesn't show up in 100W amplifiers driving speakers with much bigger PSU currents despite the full-wave rectification, but mostly they have the tx next to the diodes and caps, not much of a radiating antenna.
Still, if you've fixed the problem, who cares what the cause was...
Of course you do get induced hum, but this is is usually where the conductors are in close proximity and the field strength obeys an inverse cubic law, as opposed to inverse square, and you wouldn't fix that by grounding to the enclosure.
120Hz doesn't propagate much, but then it's a switching waveform, so the frequencies involved are related to the risetime, not just the pulse repetition frequency.
'Course I haven't seen what the effect is at the audio output.
It's a comparatively trivial current.
You have to wonder why the problem doesn't show up in 100W amplifiers driving speakers with much bigger PSU currents despite the full-wave rectification, but mostly they have the tx next to the diodes and caps, not much of a radiating antenna.
Still, if you've fixed the problem, who cares what the cause was...
@counter culture, I guess it's semantics, but most would consider 60 hz an audio frequency, not a radio frequency. The current waveform is a small portion of a distorted sine wave.
The caps in the O2 are without any charging current for something like 90% of the time. So all the energy is condensed into an unusually small portion of the AC waveform. The caps fall much further from cycle to cycle than in a full wave amp which makes the peak currents much higher.
If you go way back in this thread agdr did a simulation of the diode currents in the O2 power supply. They're surprisingly high. I've brought other amplifiers near the O2's power supply and they too pick up the EMI.
Your point about big power amps is valid but it can still be a problem there. Doug Self talks about what he calls "inductive distortion" and I've seen it in amps with poor layouts, etc. But it only shows up when they're driven hard not when they're idle with nothing playing.
The caps in the O2 are without any charging current for something like 90% of the time. So all the energy is condensed into an unusually small portion of the AC waveform. The caps fall much further from cycle to cycle than in a full wave amp which makes the peak currents much higher.
If you go way back in this thread agdr did a simulation of the diode currents in the O2 power supply. They're surprisingly high. I've brought other amplifiers near the O2's power supply and they too pick up the EMI.
Your point about big power amps is valid but it can still be a problem there. Doug Self talks about what he calls "inductive distortion" and I've seen it in amps with poor layouts, etc. But it only shows up when they're driven hard not when they're idle with nothing playing.
My point precisely. Do you imagine I post without reading the thread?
It's not the absolute peak current, but the rate of rise of absolute magnitude of current that determines the level of emission and it's harmonic content. The pulses occur 120 times per second, but they contain harmonic energy at greater than 120Hz, this is why they propagate sufficiently to affect surrounding equipment. As I said; 120Hz does not propagate that well, or perhaps it's more useful to point out that a quarter-wave whip would be 625 kilometers high.
The power cord is the most effective antenna in which these currents travel, if you can keep them out of the cord you may be able to prevent them affecting devices in the surroundings as well as the amplifier itself, which would be no bad thing, we all have a responsibility to minimize radio noise in the environment.
I don't know that you can attenuate the content sufficiently with a ferrite, it may, as you say, be too low in frequency to be absorbed by the lowest frequency material available, but, on the other hand, whatever the frequencies, they ARE propagating, which tends to suggest that they fall in the range susceptible to absorbtion by ferrites, they have an audible effect on nearby equipment and this makes them EMI, by definition radio noise.
It's not the absolute peak current, but the rate of rise of absolute magnitude of current that determines the level of emission and it's harmonic content. The pulses occur 120 times per second, but they contain harmonic energy at greater than 120Hz, this is why they propagate sufficiently to affect surrounding equipment. As I said; 120Hz does not propagate that well, or perhaps it's more useful to point out that a quarter-wave whip would be 625 kilometers high.
The power cord is the most effective antenna in which these currents travel, if you can keep them out of the cord you may be able to prevent them affecting devices in the surroundings as well as the amplifier itself, which would be no bad thing, we all have a responsibility to minimize radio noise in the environment.
I don't know that you can attenuate the content sufficiently with a ferrite, it may, as you say, be too low in frequency to be absorbed by the lowest frequency material available, but, on the other hand, whatever the frequencies, they ARE propagating, which tends to suggest that they fall in the range susceptible to absorbtion by ferrites, they have an audible effect on nearby equipment and this makes them EMI, by definition radio noise.