I did the measurements this time in a fast meaner... I know one may elaborate a plan about these... It is quite a task/work, and take time... Well, maybe one day...
I know well what you mean about the hot air flow, and is true. To enlarge the holes in the chassis it may be a good solution, but this have to be done by design... Here are not only the holes in the chassis the problem, but the holes in the main PCB which is the biggest problem. You know, the holes in the chassis are almost covered by the main PCB. The holes in the PCB do not match very well with the holes in the chassis... The holes in the main board do not match actually very well with the heat sinks. They used a pitch in the PCB design for the heat sinks ribs, but in production they used another kind of heat sinks with different ribs pitch... So, quite poor match in this respect. The holes in PCB have to be in between every of the ribs of the heat sinks...
Else, the holes in PCB can not be too many and too big, because it have impact for its mechanical properties. There are also here about traces to be routed... Quite complicated. I do not think the Oppo designers will want to struggle too much with this aspect.
The ventilation and the air flow in this hot device it can be better if one give more air access in the back of the enclosure/tube. As I have explained earlier, the holes under the PCB it helps to initiate a air flow when the temperature increase slowly (power on), but when the heat is very well concentrated in the front of the enclosure, and goes up through the upper window, then more fresh air can come in from ( a little bit far away) larger opening in the back end of the enclosure. I think in this way the ventilation and air movement is more efficient. Best solution in this case it should be a back panel as a grill. It is possible to be designed/produced such to cover both tasks: mechanical support for the back side connectors and pass through for more air. But until a such solution may be applied, it works well to open the back end as I did (pictured in a previous post).
I know about the noise, when using a forced ventilation. I`m not an adept of using fans in such audio devices. But I think a fan with a slowest possible speed, it will not create an audible noise... Noctua is a very good and quiet fan. The only problem is its dimensions. Too big/thick for this enclosure... Well, I just ordered some parts, and I will experiment... I think it may work well. Because the heat developed, there are already the necessary conditions to have a air flow inside. It should be only increased the speed of that air flow, by using a fan at the lowest possible current/speed to make it spinning... A little bit more pressure inside it may contribute to a really cooler device. I will come back with infos, when I will have it...
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I got today the ordered fan, to experiment a forced ventilation for HA-1. And done it!
The result: a dramatic decreasing in temperature level for the whole device.
As I predicted, only a very little increasing of the air flow (more than the natural one) inside the enclosure, it lower the device temperature with about 20 dg. C.
I had measured before 56 - 58 dg. C on the upper grill, and a 50dg for the whole enclosure. I have now (with the back panel mounted 5 mm far from enclosure), the hottest part (class A power regulators) at 39-40dg, while the rest of the enclosure measure 33 dg. The class A heat sinks measured outside the upper grill, shows 34 dg. The heat sinks of the power regulators for class A amp are outside of the main air stream generated by the fan.
I may say that after the electronic improvements for sound quality, using of a fan inside the enclosure is the next big improvement for functionality of HA-1.
I have to renounce now at the thermal isolations of the caps near the heat sinks. Not necessary at all, at this low level overall temperature. BTW, the room temperature is almost the same as before...
But want about the noise of this fan? Well, in a total quiet room, I can not hear anything at 20 cm far from the upper grill. At 5 cm distance, it is audible a very low noise made by the air turbulences... Quite remarkable!
Some more detail about the fan: I was looking for the thinnest one (80 mm), and without ball bearings. So I have found one on Ebay, for few $. I have determinate the lowest possible power to make it start up and spinning. Using this fan at this very low power, it make no any audible noise at 20 - 30 cm far from it.
The mounting of the fan as is to be seen in the picture is not final. I used silicone dampers, and I may fine tune a little bit more the mounting of it in place. But mainly it will look like in the picture, and it will be placed in the same place.
Quite satisfied about this successful modification...
The result: a dramatic decreasing in temperature level for the whole device.
As I predicted, only a very little increasing of the air flow (more than the natural one) inside the enclosure, it lower the device temperature with about 20 dg. C.
I had measured before 56 - 58 dg. C on the upper grill, and a 50dg for the whole enclosure. I have now (with the back panel mounted 5 mm far from enclosure), the hottest part (class A power regulators) at 39-40dg, while the rest of the enclosure measure 33 dg. The class A heat sinks measured outside the upper grill, shows 34 dg. The heat sinks of the power regulators for class A amp are outside of the main air stream generated by the fan.
I may say that after the electronic improvements for sound quality, using of a fan inside the enclosure is the next big improvement for functionality of HA-1.
I have to renounce now at the thermal isolations of the caps near the heat sinks. Not necessary at all, at this low level overall temperature. BTW, the room temperature is almost the same as before...
But want about the noise of this fan? Well, in a total quiet room, I can not hear anything at 20 cm far from the upper grill. At 5 cm distance, it is audible a very low noise made by the air turbulences... Quite remarkable!
Some more detail about the fan: I was looking for the thinnest one (80 mm), and without ball bearings. So I have found one on Ebay, for few $. I have determinate the lowest possible power to make it start up and spinning. Using this fan at this very low power, it make no any audible noise at 20 - 30 cm far from it.
The mounting of the fan as is to be seen in the picture is not final. I used silicone dampers, and I may fine tune a little bit more the mounting of it in place. But mainly it will look like in the picture, and it will be placed in the same place.
Quite satisfied about this successful modification...
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Coris, here's an idea for you! Despite its original intention is to protect vehicles air intakes from snow, its properties can be quite nice for your mods.
Water and Dust Proof Fabric Air Filter Prefilter | eBay
A little expensive, I know, but It does a really nice job keeping the components free of dust and the chassis holes would be turned into splashproof holes!
Hope it someway helps you!
Water and Dust Proof Fabric Air Filter Prefilter | eBay
A little expensive, I know, but It does a really nice job keeping the components free of dust and the chassis holes would be turned into splashproof holes!
Hope it someway helps you!
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Thanks eduardokbb for your contribution.
This filter it may work for air intakes, where is place enough to apply it. Unfortunately not the case of this quite tight enclosure, and its enough dispersed air intakes...
While in car field there is about very high air flows for the air intakes, here is not the same.
There is an slightly air flow by natural ventilation (original design), based on temperature differences. This natural air flow it can also transport dust inside the enclosure. Using of this fan at its limit speed/spinning it add a very little speed to the natural air flow inside this device. I`m not very sure now what is the impact of this slightly air flow increasing in getting inside the dust. I will see what happen, after a while.
I really think that an eventual dust accumulation it may not be as big problem as the high heat dissipation/generation inside of this class A amplifier, it is for the components inside, its stable in time parameters, in many hours continuous normal daily use of such device.
Forcing the ventilation only a little bit more over the natural way, it have a dramatic impact for the developed temperature inside. If dust it may accumulate inside, then this is quite easy to be removed (blow it out) from time to time.
After more than 6 hours continuous powered on, I had the follow measured (stabilized) temps: Class A heatsinks - 45dg.C, Class A power reg. - 40dg.C, the whole enclosure - 35dg.C. More reasonable than over 50 dg.C as it was before...
This filter it may work for air intakes, where is place enough to apply it. Unfortunately not the case of this quite tight enclosure, and its enough dispersed air intakes...
While in car field there is about very high air flows for the air intakes, here is not the same.
There is an slightly air flow by natural ventilation (original design), based on temperature differences. This natural air flow it can also transport dust inside the enclosure. Using of this fan at its limit speed/spinning it add a very little speed to the natural air flow inside this device. I`m not very sure now what is the impact of this slightly air flow increasing in getting inside the dust. I will see what happen, after a while.
I really think that an eventual dust accumulation it may not be as big problem as the high heat dissipation/generation inside of this class A amplifier, it is for the components inside, its stable in time parameters, in many hours continuous normal daily use of such device.
Forcing the ventilation only a little bit more over the natural way, it have a dramatic impact for the developed temperature inside. If dust it may accumulate inside, then this is quite easy to be removed (blow it out) from time to time.
After more than 6 hours continuous powered on, I had the follow measured (stabilized) temps: Class A heatsinks - 45dg.C, Class A power reg. - 40dg.C, the whole enclosure - 35dg.C. More reasonable than over 50 dg.C as it was before...
If the fan use less than 100mA (my case), then it can be powered from Power ON remote control system of the player (if one do not use this option for the task it was designed). There is there the output with 12v/100mA. The small transformer beside the toroid do the job to provide power to the remote control system.
It will be a mod in this area. I`m waiting now for the produced PCBs.
BTW, I will suggest using of teflon isolated wires to connect the fan to the power, as these wires can come in contact with hot heat sinks, inside the enclosure. Else it have to be routed very carefully.
The power for class A amp is regulated by two regulators (biggest heat sinks) beside the toroid. The toroid transformer provide power for the whole device in a quite sophisticated (controlled and precise) power system.
It will be a mod in this area. I`m waiting now for the produced PCBs.
BTW, I will suggest using of teflon isolated wires to connect the fan to the power, as these wires can come in contact with hot heat sinks, inside the enclosure. Else it have to be routed very carefully.
The power for class A amp is regulated by two regulators (biggest heat sinks) beside the toroid. The toroid transformer provide power for the whole device in a quite sophisticated (controlled and precise) power system.
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Replaced the original 100Mhz oscillator with a SAW 108Mhz one.
Please note that the new oscillator is mounted in a temporary meaner, for experimenting purposes. The final mounting it have to avoid such connection wires.
The result is very difficult to describe. Hard to find the right words for such improvement...
The improvement is general, in many directions. There is a dramatic increasing in fidelity.
The precision of the sounds/instruments in the space of soundstage is unbelievable. Nothing as diffuse sounds. Just every source it have a well defined place in sounstage. Well, in good quality recordings... The sounds, instruments, voices act in the room, just in front and around listener. Even the smallest detail is to be heard. I was never aware about many details I can hear now in very known of my recordings (FLAC encoded - 176,4Khz). Localisation of the instruments and the sound sources is just amazing.
I had as source my computer, but using my Note2 to play out through USB the same FLAC files, the sound quality is even higher. The only explanation I find so far is the less noisy power of the phone, maybe a lower activity of the operating system, while outputting these files, or just a better software application. However the highest sound quality is when using my phone to send out files to the HA-1 USB.
Such music playback/reproduction have to be experienced to know what really is all about.
I have also replaced the original oscillators for asynchronous USB relocking stage, with NDK oscillators. No change in SQ more than what I got by replacing the DAC oscillator (with the previous listed modifications in place).
Please note that the new oscillator is mounted in a temporary meaner, for experimenting purposes. The final mounting it have to avoid such connection wires.
The result is very difficult to describe. Hard to find the right words for such improvement...
The improvement is general, in many directions. There is a dramatic increasing in fidelity.
The precision of the sounds/instruments in the space of soundstage is unbelievable. Nothing as diffuse sounds. Just every source it have a well defined place in sounstage. Well, in good quality recordings... The sounds, instruments, voices act in the room, just in front and around listener. Even the smallest detail is to be heard. I was never aware about many details I can hear now in very known of my recordings (FLAC encoded - 176,4Khz). Localisation of the instruments and the sound sources is just amazing.
I had as source my computer, but using my Note2 to play out through USB the same FLAC files, the sound quality is even higher. The only explanation I find so far is the less noisy power of the phone, maybe a lower activity of the operating system, while outputting these files, or just a better software application. However the highest sound quality is when using my phone to send out files to the HA-1 USB.
Such music playback/reproduction have to be experienced to know what really is all about.
I have also replaced the original oscillators for asynchronous USB relocking stage, with NDK oscillators. No change in SQ more than what I got by replacing the DAC oscillator (with the previous listed modifications in place).
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Class A Buffer OP on a rear OP XLRs ?
Hi Coris,
To modify the XLR PreAmp OP feed from current OP Amps feed to Class A Buffer OP - making it with a extra wires bypass, would You take the balanced signal before OP relays near BAL. HP Neutrik Jack or after the relays ? will You take the GND for Rear XLRs from the GND - HP buffers use or is it better leave the Rear XLRs GND as is now ? Current rear Traces to XLR pins also need to be cut, maybe just before XLR pins and soldered the new bypass wires directly on the rear XLR pins, what do You think would be the best option for this mod ?
Thank You for the answer,
Andreas
Hi Coris,
To modify the XLR PreAmp OP feed from current OP Amps feed to Class A Buffer OP - making it with a extra wires bypass, would You take the balanced signal before OP relays near BAL. HP Neutrik Jack or after the relays ? will You take the GND for Rear XLRs from the GND - HP buffers use or is it better leave the Rear XLRs GND as is now ? Current rear Traces to XLR pins also need to be cut, maybe just before XLR pins and soldered the new bypass wires directly on the rear XLR pins, what do You think would be the best option for this mod ?
Thank You for the answer,
Andreas
Hi Andreas
I`m not very sure I understood well your approach.
Mainly the improvements of a device are based quite much on analysing, appreciating and in fact, experimenting in a way the original design(ers) did not, or never thought about...
So, if you may appreciate that your idea it may lead to improvements, then just do it so how you think. If success, then (if you want) you may share the results, and promote your success. If not, then one may try something else... or buy a new device.
I support the concept of improving devices, but I`m quite sceptic about making wider changes, with wires all over, hanging components, and so on. Such may not leads to good results, or it may not be appreciated as an improvement...
One may not ignore that simple fact that designing a product from scratch is a huge task, it need lot of resources, and are many people involved in such. Mainly, all these efforts have a positive result, in a quite good designed device. In the same time, is very normal that in a designing process, the involved ones can not cover all the existing aspects, about the functionality or parameters of that device. One or another thing may be lost on the way, or the designers have to follow some financial border, and something it may be excluded from the design by financial reasons.
I think right here is the role of the modder/improver. To put back what the designers have ignored or excluded (for vary reasons).
To conclude, I think a designed product it may not suffer for large modifications, which it may actually degrade its parameters, than improve something. A reasonable balance in an improvement approach it may be kept.
But one can just feel free for experiment his ideas, take the risks and appreciate the final results... And share it here if will like so...
I`m not very sure I understood well your approach.
Mainly the improvements of a device are based quite much on analysing, appreciating and in fact, experimenting in a way the original design(ers) did not, or never thought about...
So, if you may appreciate that your idea it may lead to improvements, then just do it so how you think. If success, then (if you want) you may share the results, and promote your success. If not, then one may try something else... or buy a new device.
I support the concept of improving devices, but I`m quite sceptic about making wider changes, with wires all over, hanging components, and so on. Such may not leads to good results, or it may not be appreciated as an improvement...
One may not ignore that simple fact that designing a product from scratch is a huge task, it need lot of resources, and are many people involved in such. Mainly, all these efforts have a positive result, in a quite good designed device. In the same time, is very normal that in a designing process, the involved ones can not cover all the existing aspects, about the functionality or parameters of that device. One or another thing may be lost on the way, or the designers have to follow some financial border, and something it may be excluded from the design by financial reasons.
I think right here is the role of the modder/improver. To put back what the designers have ignored or excluded (for vary reasons).
To conclude, I think a designed product it may not suffer for large modifications, which it may actually degrade its parameters, than improve something. A reasonable balance in an improvement approach it may be kept.
But one can just feel free for experiment his ideas, take the risks and appreciate the final results... And share it here if will like so...
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You're correct - I wasn't clear enough,
My ask was mainly to use the class A buffer also as a Pre-Out, instead of current Pre-Out made with OP Amps, so the question was where to take off balanced signal from class A buffers immediately after Buffers, or after relais which are also already Headphones Balanced pins connected to, then route this signals to the rear output existing XLR connectors. Using a dedicated shielded balanced cables internally to do this and preserve hum free integrity, so visually no external adapters needed with existing bal. Headphone socket on a front side.
To summarize - Feed the Class A Balanced signals internally to the back side of the enclosure and for this task to reuse the already existing Line OP XLR Connectors.As You already have this unit and dissembled so it is easier to look where would be more appropriate to take the signals then just me guesing from various photos of dissembled units on the web. I plan to purchase this unit to try the sound of it as all the reviewers praise it and especially for me it is interesting its direct support to DSD 128x and higher DSD 256x couse all my digital library files I have are all DSD 128x and this would be possible a good choice to give it a try, But I do use only balanced connections, not interested in Phones much, but mostly in the Class A LineOut Balanced feed instead .
Thank You for the answer,
Cheers,
Andreas
Well, more clear now...
There is a big clue to use the class A amp for headphones, as line out, to amplify it further with an power amp. I use my HA-1 this way (with the best results), as I`m not interested in using headphones, and I do not own actually some quality ones.
In my opinion the signal it may be tapped after relays, as these have to have their function to establish the connection when it were designed to do it. I think is more safe in all the cases doing so.
The simplest way is to buy or make some adaptors for the existing connections of the HA-1. In my case I do not like to have cables in front of the device. I have chosen to route on the back panel the output I needed.
The only available place is the AES/EBU connector (not so used actually). There is quite easy to replace it or adapt it to a similar standard connector, but with the necessary pins.
One should use shielded cables of course inside the enclosure, because is quite long distance, from front to the back end.
I did not used yet the balanced headphone out, as my amp it still be a unbalanced one. Of course is a better quality when using the balanced out.
Even though I use the unbalanced out (from headphone out), but with balanced source, the quality of the sound is amazing this way (well some mods in between... ).
So yes, you can do it so. HA-1 is a good investment. You can be satisfied with it as it is, but if you want more, then is quite easy to get more from this device.
(For those who may found not so easy to get more from the device, may contact me...)
There is a big clue to use the class A amp for headphones, as line out, to amplify it further with an power amp. I use my HA-1 this way (with the best results), as I`m not interested in using headphones, and I do not own actually some quality ones.
In my opinion the signal it may be tapped after relays, as these have to have their function to establish the connection when it were designed to do it. I think is more safe in all the cases doing so.
The simplest way is to buy or make some adaptors for the existing connections of the HA-1. In my case I do not like to have cables in front of the device. I have chosen to route on the back panel the output I needed.
The only available place is the AES/EBU connector (not so used actually). There is quite easy to replace it or adapt it to a similar standard connector, but with the necessary pins.
One should use shielded cables of course inside the enclosure, because is quite long distance, from front to the back end.
I did not used yet the balanced headphone out, as my amp it still be a unbalanced one. Of course is a better quality when using the balanced out.
Even though I use the unbalanced out (from headphone out), but with balanced source, the quality of the sound is amazing this way (well some mods in between...
).So yes, you can do it so. HA-1 is a good investment. You can be satisfied with it as it is, but if you want more, then is quite easy to get more from this device.
(For those who may found not so easy to get more from the device, may contact me...
)
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Well, more clear now...
There is a big clue to use the class A amp for headphones, as line out, to amplify it further with an power amp. I use my HA-1 this way (with the best results), as I`m not interested in using headphones, and I do not own actually some quality ones.
...
Hi Coris,
Thank You for the answer,
So after the relays is the best signal take off point, great, I plan to use already existed PreOut XLR connectors and cut the traces from the orig. signal ones and connect tthere the Class A OP buffer Balanced Signals.
This would be the only mod for now.
Did You try any DSD files to play with HA-1 and how You enable the direct DSD playback - ASIO is the driver from OPPO but how to not resample to PCM in between ?, what player do You use to support direct DSD playback without any resampling on the fly via processor on the PC ?
How to avoid this? Can You share Your impressions about HiRes DSD sound played back with HA-1?
Do You use PC or MAC for playback HiRes files?
Thank You for the answers,
Cheers,
Andreas
I have experience with DSD playback with DSD64 Files encoded within DoP.. It plays them flawlessly. I suspect the FIR filtering used within the ESS9018 is using default sloaps ......but based on my experience with other ESS9018 DSD decodes --- this default is the best anyways. I do wish they exposed some choices like all the others do --- I find it strange. I will soon be trying native DFF/DFS streams once I get the latest plugin working, but not sure if it works with Squeezeplay which I use with this unit.
Hello jt25741,
Great to hear You use DSD files, Do You use the HA-1 for DA conversion and playback ?
I also use only DSD 128x DSD - DFF files for digital playback and for me is the only acceptable digital format sound wise I can listen to. You mention some new plugins, are they for the Squeezeplay player or for some other - Foobar come to mind. How do You encode the files with DoP encoder ? Are the options in OPPO driver soft to enable DSD Playback in OS (Win or MAC) or the player auto recognize the files by themself?
Cheers,
Andreas
You know, I may say that I was not able to play directly DSD files through PC USB interface (ASIO Oppo driver). I have tried Foobar, and AudioGate which can (supposedly) playback such files. AudioGate it show actually that it send out DSD, but I can only see on my HA that is received PCM (352,8Khz sampling)... Foobar can play out if set it on PCM (352,8), while HA show PCM file playback. When set it on DSD output, Foobar it playback the file, but now sound out of HA...
Quite confusing, and the result is zero when using DSD over USB in this way.
I had full success to play DSD native, using my phone and the dedicated player app. It works just fine to output DSD recognized by HA with not problem, and great sound. The only problem is that using the phone to play such files is very inconvenient. DSD files are big to be stored on a phone memory, and the whole approach very uncomfortable.
So, I`m also very interested in informations about a more convenient way to play DSD files directly from PC, through a player which can in fact do it...
BTW, the older version of AudioGate can convert PCM to DSD files (but they ask for a tweet every time to do this...).
To be honest, the sound quality difference I can notice between playing back PCM files (high sampling) and DSD ones (64, 128) is very minimal.
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What I have been doing is using the modified DoP version (packed) of the foo_input_sacd plugin for foobar located here:
https://docs.google.com/file/d/0B4jZ4NDGECzEczBjYVByNUdjdDg/edit
Setting up the plugin in the configuration menu of foobar, you now have a selection to choose for DSD2PCM mode: DSD over PCM
This creates 176.4Khz FLAC files upon Foobar conversion of the PCM files. These FLAC files are packed DSD64 or DSD128 if the native sample rate is doubled to 352.8Khz). I have not tried DSD128 as my playback devices cannot handle this native bitrate -- plus I have no content at DSD128.
There is a applet for the Squeezebox Touch called EDO that turns on USB output and allows for up to 192Khz streaming support. This is important for this to work.
But for the HA-1 I do not use a Squeezebox Touch, but rather a small Linux computer that runs squeezeplay. Someone has taken this EDO plugin and recompiled it to work with Squeezeplay, and I use this with the HA-1 to enable these higher bitrates. It works fine with the same DoP files I pre-prepared in Foobar as above.
Here is the Squeezeplay EDO plugin I use for the Joggler (Linux PC):
***
wget http://www.jogglerwiki.com/forum/download/file.php?id=329 -O EnhancedDigitalOutput.tar.gz
sudo tar -xvf EnhancedDigitalOutput.tar.gz -C /opt/squeezeplay/share/jive/applets/ --overwrite
***
Note the above is not needed if you are using the "real" Squeezebox Touch and simply install the EDO applet from the main menu.
In Logitech Media Server version 7.9 there is a new LMS side plugin called DSDPlayer (also Written by Triode). This seems quite amazing, and does on-the-fly conversion of DFF/DFS files to DoP allowing one to not have to go through the step I did above in Foobar. I have not tried it yet, but I know it works from other's posts.
As to how it all sounds......of course this is where we all differ. For me DSD64 on the ESS is just OK. I feel it needs to be upsampled with cutoff filter pushed way further up to the ultrasonic for DSD to sound better. ESS9018 does not do this native, but rather follows scarletbook convention. For me the most amazing thing the ESS9018 does is decode redbook CDs better than any other DAC chip. On the HA-1 as on others, to my ears -- there is less as difference between high-res content and a great CD than others. For best quality in my library PCM 24/192 is king through HA and other ESS9018DACs. DSD64 sounds good, but not great. Again to my ears what is best with DSD64 is microdynamics, rising edge cleanness and fluidity. HBR PCM sounds better in all other regards assuming a quality mastering.
As mentioned earlier I have not yet heard any DSD128 or DSD256 content....but rumor has it that these are much better.
HTH
DSD 128x
Hello Coris, jt25741,
Thank You guys for real detailed infos about real life use of the HA-1. It is piece of a good quality audio playback system but it lacks a real dedicated PLAYER to support all its very high praised Play Back qualities. If it will be equipped with a dedicated software player for Win & Mac than it will be the best from the best.
Thank You both for the detailed infos about software You used and how You both manged to make it best of use to sound nirvana.
Coris, it is interesting that You use phone for best DSD playback and it surprised me that tis is at least possible to do with a phone at all, may i ask You what phone do You use with such capabilities ?. How You install ASIO driver into phone OS ?
Is it some NOKIA with Win Phone soft running on IT ? Amazing that You can do DSD native playback with Phone, but not so easy with PC where it should be much easier, but it isn't.
OK OPPO made its own ASIO driver to support the hardware but its purpose is mainly to support latency for PCM Playback, is it adapted to support also direct DSD playback ? How it supposedly skip or bypass Windows mixer and other unnecessary softw. garbage isn't clear or written in HA-1 pdf user manual.
So quite a lot of guessing i think.
There is another game player on the scene the Orig KORG Headphone Amp. with USB interface but equipped with Orig. KORG DSD and all the rest PCM DA converter, DA is the same as in KORG DSD MR2000S, with theirs drivers to max native support DSD Playback, but a bad side of a good news is that all the analogue stages are IC based, so no discrete, no Class A - less involving sound - only promises to audio nirvana on the fare horizon.
Here the OPPO HA-1 really excel but if ..... it will be packed also with dedicated Player software, than this would be really a full packed product to fulfill even all the highest demand for PC based Audio playback needs.
Well I use mostly Analog LP based playback for my pleasure, Classic only - 3500 LPs + and some 350 pcs -1 inch Master Tapes for My Studer C37 Tube based Tpe playback and half of my tapes are first Master Copies. I don't use CDs, DVD audio or other Disc based playback stuff, mostly because of such inferior sound abilities and elementary missed all the emotions from music! while from the LPs and tapes emotions were 100% present.
I also have borrowed MR2000S for DSD files playback which I get from my very good friend who has 8 of them in his studio. he mostly use four of them so he packed one with his own life classical recordings he made (DSDIFF 128x) and give - lend it to me.
This is the only digital playback which I currently own and it is the closest to Analog Stuff I use, so I was thinking that it is time now to go with HA-1 + PC supported DSD Playback would be fine and the managing the playback files would be a breeze comparing to MR2000S 80GB internal HDD and 6MB/s USB transfer rate the files on to its HDD, and not mention the naming the files with its rotary encoder is a real sickness.
I was order the HA-1 now waiting on it, when it'll be in my hands I will try all the suggested playback options and I'll compare the same files playing them via HA-1 and MR2000S side by side and try to found out the differences in resolution and fine macro details from each and see if thee's quantitative differences between the two DSD Playback equipment.
I can only HOPE there would be enough USERS requests to OPPO Digital to make a dedicated PLAYER software to full support the HA-1 hardware capabilities ...... till then only wishes remains....
With my best regards
Andreas
Hello Coris, jt25741,
Thank You guys for real detailed infos about real life use of the HA-1. It is piece of a good quality audio playback system but it lacks a real dedicated PLAYER to support all its very high praised Play Back qualities. If it will be equipped with a dedicated software player for Win & Mac than it will be the best from the best.
Thank You both for the detailed infos about software You used and how You both manged to make it best of use to sound nirvana.
Coris, it is interesting that You use phone for best DSD playback and it surprised me that tis is at least possible to do with a phone at all, may i ask You what phone do You use with such capabilities ?. How You install ASIO driver into phone OS ?
Is it some NOKIA with Win Phone soft running on IT ? Amazing that You can do DSD native playback with Phone, but not so easy with PC where it should be much easier, but it isn't.
OK OPPO made its own ASIO driver to support the hardware but its purpose is mainly to support latency for PCM Playback, is it adapted to support also direct DSD playback ? How it supposedly skip or bypass Windows mixer and other unnecessary softw. garbage isn't clear or written in HA-1 pdf user manual.
So quite a lot of guessing i think.
There is another game player on the scene the Orig KORG Headphone Amp. with USB interface but equipped with Orig. KORG DSD and all the rest PCM DA converter, DA is the same as in KORG DSD MR2000S, with theirs drivers to max native support DSD Playback, but a bad side of a good news is that all the analogue stages are IC based, so no discrete, no Class A - less involving sound - only promises to audio nirvana on the fare horizon.
Here the OPPO HA-1 really excel but if ..... it will be packed also with dedicated Player software, than this would be really a full packed product to fulfill even all the highest demand for PC based Audio playback needs.
Well I use mostly Analog LP based playback for my pleasure, Classic only - 3500 LPs + and some 350 pcs -1 inch Master Tapes for My Studer C37 Tube based Tpe playback and half of my tapes are first Master Copies. I don't use CDs, DVD audio or other Disc based playback stuff, mostly because of such inferior sound abilities and elementary missed all the emotions from music! while from the LPs and tapes emotions were 100% present.
I also have borrowed MR2000S for DSD files playback which I get from my very good friend who has 8 of them in his studio. he mostly use four of them so he packed one with his own life classical recordings he made (DSDIFF 128x) and give - lend it to me.
This is the only digital playback which I currently own and it is the closest to Analog Stuff I use, so I was thinking that it is time now to go with HA-1 + PC supported DSD Playback would be fine and the managing the playback files would be a breeze comparing to MR2000S 80GB internal HDD and 6MB/s USB transfer rate the files on to its HDD, and not mention the naming the files with its rotary encoder is a real sickness.
I was order the HA-1 now waiting on it, when it'll be in my hands I will try all the suggested playback options and I'll compare the same files playing them via HA-1 and MR2000S side by side and try to found out the differences in resolution and fine macro details from each and see if thee's quantitative differences between the two DSD Playback equipment.
I can only HOPE there would be enough USERS requests to OPPO Digital to make a dedicated PLAYER software to full support the HA-1 hardware capabilities ...... till then only wishes remains....
With my best regards
Andreas
Yes, indeed is surprising that an android phone can use an dedicated application to play native DSD, while a PC can not do it, but through complicated procedures.
I have an (old now) Note 2, but I think is not the phone itself which is important, but the very good application, which use its own software processor (bypassing android OS), and the USB interface to make this playback possible. I have notice (before testing DSD capabilities) that the same files played from my phone it sounds much better than played from PC (as pointed in a previous post). It may be very possible that the way the application process the data, leads to such results.
I just start to think to a possible hardware which can run this app on android OS, used exclusively for playback of digital files. Using the phone for such task is not comfortable, at least for me. As the app can play over a network, it still yet to try this way too, using the storage for files on a PC. Still experimenting about...
There is not about installing ASIO on a phone, but just this app (USB Audio Player Pro), and that`s it. The app runs on its own platform, but use the storage on the device (or a network the phone can communicate with) and output the results on USB interface. As I understood out of their description.
Fully agree that Oppo should think about a dedicated software as player, and maybe a little bit more, to control and fine tune (customize) the DAC inside HA-1 and/or some another of the firmware functions. Integrate this way the HA with a computer, it may result a very high end and versatile hifi digital system.
Will be interesting to hear about your impressions/opinions about HA-1 as it is out of the box.
HI Andreas,
Looking forward to how the HA-1 sounds with your DSD128 content. DSD playback is just a feature of the ESS DAC that came along for the ride. I do not think that OPPO did much to support it other than ensure the USB port was enabled with sufficient XMOS based async control to enable it. But it supports sample rates that are even higher than my other high-end DAC with DSD because of this pairing. As Coris says, I look forward to your impressions out of the box (as yet my unmodified HA-1 still is very enjoyable).
Looking forward to how the HA-1 sounds with your DSD128 content. DSD playback is just a feature of the ESS DAC that came along for the ride. I do not think that OPPO did much to support it other than ensure the USB port was enabled with sufficient XMOS based async control to enable it. But it supports sample rates that are even higher than my other high-end DAC with DSD because of this pairing. As Coris says, I look forward to your impressions out of the box (as yet my unmodified HA-1 still is very enjoyable).
A reason to keep in place the HA-1 designers concept of AC coupling caps (especially when about inputs DC isolation):
As I pointed out previously, HA-1 it have a quite sophisticated power (monitoring) system. Part of this system is the protection at detecting DC levels on inputs.
If square waves. or strong limited signals, goes in through AC coupling caps, it may be detected as DC levels, and the device protection it may be triggered (power off).
Eliminating the AC coupling caps on inputs, may lead to problems... It may work with a very well controlled DC level (zero offset) on the sources outputs.
Safest is to keep the concept of AC coupling on HA-1 inputs...
As I pointed out previously, HA-1 it have a quite sophisticated power (monitoring) system. Part of this system is the protection at detecting DC levels on inputs.
If square waves. or strong limited signals, goes in through AC coupling caps, it may be detected as DC levels, and the device protection it may be triggered (power off).
Eliminating the AC coupling caps on inputs, may lead to problems... It may work with a very well controlled DC level (zero offset) on the sources outputs.
Safest is to keep the concept of AC coupling on HA-1 inputs...
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