My profession is actually in audio and video processing. The main thing was comparing the Decco65 DAC output with Oppo DAC output feeding Decco65 amp for must 1:1 comparison on Monitor Audio GX300 speakers. I'm testing just CD but would expect the sound to be close as the both us Sabre DAC. Some possibility for the difference may be 105 is decoding 44.1 and not upsampling to 176.4/192. The difference in sound is brush on cymbals is delicate and distinct on Decco65 DAC while on 105 is harsh and not individually separable.
Yes, this is interesting.
I suggested listening to high-resolution sources so that you can confirm that the 105 is really working properly. This unit is certainly at its best with high-resolution and modern sources.
Also, I still encourage you to compare to headphone, as this output is different (double DAC lines) and does not use the output coupling capacitors. This would give you a sense of how the unit will perform without the output coupling capacitors, and with some simple modifications.
As far as the CD playback goes, You could try several things. First of all you can rip these To a .FLAC file and then play it from a USB thumb drive. This will improve the quality of 44 kHz material. Coris has also taken this one step further and converted the files to 192 kHz, 24-bit and reduced the level slightly (you can find the details on one of his posts). He says this improves the result. And this may allow you to control specifically how the up sampling is being done. Let us know if you find a good sounding solution. It is known that the 105 works best as a DAC.
Oppo has not said what type of digital filtering they are using for the CD playback. The Sabre DAC is converting to very high frequency using a 54 MHz clock. Probably the defaults are being used.
I suggested listening to high-resolution sources so that you can confirm that the 105 is really working properly. This unit is certainly at its best with high-resolution and modern sources.
Also, I still encourage you to compare to headphone, as this output is different (double DAC lines) and does not use the output coupling capacitors. This would give you a sense of how the unit will perform without the output coupling capacitors, and with some simple modifications.
As far as the CD playback goes, You could try several things. First of all you can rip these To a .FLAC file and then play it from a USB thumb drive. This will improve the quality of 44 kHz material. Coris has also taken this one step further and converted the files to 192 kHz, 24-bit and reduced the level slightly (you can find the details on one of his posts). He says this improves the result. And this may allow you to control specifically how the up sampling is being done. Let us know if you find a good sounding solution. It is known that the 105 works best as a DAC.
Oppo has not said what type of digital filtering they are using for the CD playback. The Sabre DAC is converting to very high frequency using a 54 MHz clock. Probably the defaults are being used.
In may opinion, the headphone output is not very high in quality. On quite low impedance headphone (Sennheiser IE80), the noise level is enough high. Also the final amp chip for headphone it is a good quality one, but is a very common consumer device...
Honestly, I`m not very impressed by this output.
Honestly, I`m not very impressed by this output.
First, a little correction here: When about up sampling from 44,1, it should be used 176,4 Khz. For original 48Khz files, is to be used 192Khz.
This ensure the best results. Of course it works all kind of up sampling frequencies, but the best results is the right multiple for the base sampling frequency...
Up sampling the original recorded material it brings really important improvements in quality (only when the playback is done on a high enough quality device). I will suggest to everybody to try this by digital file processing, but not using the devices/players up sampling settings.
Here is the procedure (best is using of a 64bit computer/OS):
- Rip the original CD material (44,1 Khz), bit perfect using accordingly software. I use EAC. If you have compressed files originated from CDs (mp3 or another), you may apply this procedure too, with very interesting results...
- Open the extracted files (one by one) in Adobe Audition. This software is highly recommended for processing. You will see, that even you have extracted from a very good recording, there are quite big differences in the amplitude level in the CD files. There is recommended to adjust the dB level to -1 dB for all your files. There is a function in Audition called Normalize. Do not use that. Just write the right dB figures in the right place to get a general level correction to -1dB for the whole file. Then use "Repair the DC offset" as a rule. Not all the files it may have DC problems, but is more safe to apply for every file this treatment. Then up sample accordingly. Here are some settings to be used... So, get in to these settings and choose what you may think are the best ones.
Save the up sampled file as RF64 (sample type - 176,4 Khz/32bit). So, the 32 bit file is saved as wav 64 bit. The saved files it take huge place on your hard drive. So you may have enough place and a quite fast machine, to succeed comfortable in this task. After converting, you may of course, delete these wav files.
- Use the last version of Foobar. Open the wav 64 bit file (RF64) in Foobar. Use the Conversion option, and setup for FLAC 176,4Khz/24 bit. Only last version of Foobar can convert 64bit wav to FLAC 24 bit.
So, play the file as you wish... On a modified Oppo...
Or on your high end device. You will for sure notice the improvements.One may think this procedure is complicated. Is not! Once one get used with the steps, all goes quite fast. Is not to be used batch function, as each file it may be exterminated and corrected accordingly for the right level.
I do myself this every time I get a CD (I prefer to buy actually to ensure a good recording quality for 44,1.). This method is to be applied too for mp3 files. Open it in Audition as wav and process as described. You will be very surprised to experience the sound... Is very interesting that even though quite heavy compression, the original recorded informations, it still be inside these files, when the extraction of these informations is made using a better digital resolution, as this procedure do. Why is like this, is to be discussed in another thread...
At least this post here it may be a little bit out of topic, but anyway...
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I was just curious: Why did Oppo use four of the DACs from one of the two ESS Sabre Reference DAC chips for the left and right channels of the Headphone amp output?
Why didn't they use four DACs for the stereo XLR balanced output, two for the stereo RCA unbalanced output, and two for the Headphone output?
* The other DAC chip (8-channel) is for the multichannel analog (RCA) output (7.1).
Why didn't they use four DACs for the stereo XLR balanced output, two for the stereo RCA unbalanced output, and two for the Headphone output?
* The other DAC chip (8-channel) is for the multichannel analog (RCA) output (7.1).
Pleased to see someone else do these experiments and get the same results ... I've posted several time about doing this on a very straightfoward PC setup, and just using the mainboard DAC for playback - I've applied the technique to heavily compressed MP3 and seen a remarkable subjective 'rescuing' of the content. Just used Audacity, and Nero for playback - foobar is a no-no for me, it degrades quality on my machine.I do myself this every time I get a CD (I prefer to buy actually to ensure a good recording quality for 44,1.). This method is to be applied too for mp3 files. Open it in Audition as wav and process as described. You will be very surprised to experience the sound... Is very interesting that even though quite heavy compression, the original recorded informations, it still be inside these files, when the extraction of these informations is made using a better digital resolution, as this procedure do. Why is like this, is to be discussed in another thread...
At least this post here it may be a little bit out of topic, but anyway...
When you start a thread on the matter, I'll be happy to join in ...
I should have said four channels out of the Sabre DAC towards the stereo analog XLR output, etc.
Now, because you only use one stereo analog connection (XLR or RCA), then you probably have two channels from the DAC going to each Left and Right jack outputs. ...Then it don't matter if it is the balanced or unbalanced, it is configured as four channels out of the DAC going either way?
...Or two for the XLR and two for the RCA? I just don't remember.
The output from the headphone amp is low for certain pair of cans?
What should they do with the next BDP-115D?
In BDP95 the DAC channels it were distributed in a better way, but this model it have not headphone out. Oppo have justified/explained why they chosen this way. It make enough sense their explanation, but I still think it was better to chose another design, with more focus on the balanced/unbalanced outputs. Why not just use a dual mono configuration for these stages? One more ESS9018 chip it will not increase very much the production costs for the player, when is about a production of many thousands of devices.
Even though the headphone out in 105 model it use two of DAC stereo channels, I`m not impressed at all of the performances on this output. It should be used another type final analogue processing circuits to make well benefit of these two/four staked channels....
Pleased to see someone else do these experiments and get the same results ... I've posted several time about doing this on a very straightfoward PC setup, and just using the mainboard DAC for playback - I've applied the technique to heavily compressed MP3 and seen a remarkable subjective 'rescuing' of the content. Just used Audacity, and Nero for playback - foobar is a no-no for me, it degrades quality on my machine.
When you start a thread on the matter, I'll be happy to join in ...
Pleased for me too, to find out that similar method is used by someone else...
I did not read it about your findings yet. It looks like we came independently to the similar procedures. It is not very strange also that it may happen...
I agree that it can be fortunate opening of a thread for this subject. Everybody can do this also...
As I know per today Foobar is the only one player/software which it have implemented the (at last) new FLAC improved version, with wav 64 bit conversion option, and another improvements.
Else I use Foobar for playback and I`m quite satisfied with. One can configure enough deep the audio stage settings using this player. It works very well for me. But we may discuss this in the right thread if this it will be started....
BTW headphone out of 105...
It is interesting to try coupling this output to a amplifier, and appreciate its quality. I did not tried yet, but I did used this way with good results with another device.
I think to try this one day on my BDP105...
One should only solder a 300 (or 600 or another right value) ohm resistor on the output of headphone channels (as load), and connect it further to an amplifier input. Start from zero volume level on both player and amplifier, and carefully go up and find the optimal volume level.
It is interesting to try coupling this output to a amplifier, and appreciate its quality. I did not tried yet, but I did used this way with good results with another device.
I think to try this one day on my BDP105...
One should only solder a 300 (or 600 or another right value) ohm resistor on the output of headphone channels (as load), and connect it further to an amplifier input. Start from zero volume level on both player and amplifier, and carefully go up and find the optimal volume level.
Pleased for me too, to find out that similar method is used by someone else...
I did not read it about your findings yet. It looks like we came independently to the similar procedures. It is not very strange also that it may happen...
I agree that it can be fortunate opening of a thread for this subject. Everybody can do this also...
As I know per today Foobar is the only one player/software which it have implemented the (at last) new FLAC improved version, with wav 64 bit conversion option, and another improvements.
Else I use Foobar for playback and I`m quite satisfied with. One can configure enough deep the audio stage settings using this player. It works very well for me. But we may discuss this in the right thread if this it will be started....
I've tried what you suggested, with Audition and RF64 high bit-rate, high sampling rate conversion. To me at least, the upsampled file produced loss of focus and mushy bass. There was some spatial expansion, but the loss of focus and sense of "uncertainty" were quite unpleasant to me. However, I can see how this Audition' processing and upconversion can sound interesting... Conversion to flac did not make much difference, but I preferred the Flac fronted encoder for Win 7 64 bit compared to Foobar...
I used ProTools before and just installed Adobe Audition to give it a go. I must say that I was impressed by Audition (!!) Very fast, intuitive, with great options, in particular cacheing and switch between two or more file versions. Excellent software.
Nick
I will suggest to those interested in software processing of audio files, to join the new thread started on this subject:
http://www.diyaudio.com/forums/pc-based/251727-software-processing-audio-files.html#post3830307
So, Extreme_Boky thanks for your feedback. I will comment/answer in the new thread...
http://www.diyaudio.com/forums/pc-based/251727-software-processing-audio-files.html#post3830307
So, Extreme_Boky thanks for your feedback. I will comment/answer in the new thread...
Minimum phase filtering using SOX
Okay, I found out how to do the kind of processing I was looking for.
So starting with a 16-bit Redbook CD, I'm using foobar with the SOX plug-in to do the up sampling. I have been up sampling to 88 kHz and 176 kHz at 24 bits FLAC files. In particular, I wanted to compare normal up sampling to minimum phase filters.
The 176 kHz files are huge, and I presume they should be saved for special circumstances. They sound good, but to start with I mostly listened to 88 kHz files. In both cases up sampling in advance seems to be an improvement.
However, the minimum phase upsampling seems better, and seems more musical, and with fewer "digital" artifacts. Of course, we are not achieving SACD quality, but more of the magic seems to be there when using the minimum phase filters.
In particular I set SOX to bandwidth of 90% and minimum phase, with anti-aliasing, and best quality. Apparently, there is an updated plug-in that will allow you to go to 87.5% bandwidth which may be even better.
Eric
Okay, I found out how to do the kind of processing I was looking for.
So starting with a 16-bit Redbook CD, I'm using foobar with the SOX plug-in to do the up sampling. I have been up sampling to 88 kHz and 176 kHz at 24 bits FLAC files. In particular, I wanted to compare normal up sampling to minimum phase filters.
The 176 kHz files are huge, and I presume they should be saved for special circumstances. They sound good, but to start with I mostly listened to 88 kHz files. In both cases up sampling in advance seems to be an improvement.
However, the minimum phase upsampling seems better, and seems more musical, and with fewer "digital" artifacts. Of course, we are not achieving SACD quality, but more of the magic seems to be there when using the minimum phase filters.
In particular I set SOX to bandwidth of 90% and minimum phase, with anti-aliasing, and best quality. Apparently, there is an updated plug-in that will allow you to go to 87.5% bandwidth which may be even better.
Eric
There are plots of the impulse response of these filters here:
Transporter Round Two [SOX] (was Transporter for £999...) - The Hitchhikers Guide To Meridian
In the second half of the page.
Transporter Round Two [SOX] (was Transporter for £999...) - The Hitchhikers Guide To Meridian
In the second half of the page.
Yes, I understand this, but I decided to post here for several reasons.
First of all, I think that this method is particularly relevant to the OPPO, as that is the DAC that I'm using.
Since I am doing some of the digital processing in advance on a PC, the method is directly associated with the OPPO. Finally, I think of this method as a modification to improve the 16-bit playback of the OPPO.
As far as physical modifications go, last week I also added several damping sheets to the top of the disc player mechanism, and the underside of the top of the chassis. I also added some small pieces to the vertical struts and to the bottom of the chassis as well. I also grounded the top of the disc player mechanism. This all had a beneficial effect, especially during disc playback.
Eric