Bunpei,
where can I actually download some 32-bit audio files (FLAC?) for testing?
I'm using a kingrex UC192 USB->IIS converter and it works flawlessly with my buffalo II DAC. I have however not found any 32/192 (highest supported) material to test.
Nic
where can I actually download some 32-bit audio files (FLAC?) for testing?
I'm using a kingrex UC192 USB->IIS converter and it works flawlessly with my buffalo II DAC. I have however not found any 32/192 (highest supported) material to test.
Nic
First, I should confess that I'm not an expert of hires audio files and my experience is limited.
I have never found any downloadable 192 kHz/32 bit PCM files. I think you need to synthesize 32 bit files using any resampling or audio editing programs.
When I wanted to test 32 bit audio files on SDTrans192, I asked a Norwegian big name availability of those files. His answer in his private e-mail was;
"No 32 bit recording in a real world".
I understood his message as "Even in the professional world of studio recording, 32 bit audio files are merely intermediate products during editing or effecting processes. In this situation, the term "32 bit" mainly means "floating point" rather than "unsigned integer" that ES9018 can play.
Therefore, I just made a new file using a conversion program with dither. I detected no difference from the original.
As for 352.8 kHz PCM, only a single source was once downloadable on this internet site.
Complimentary High Resolution Downloads Courtesy of First Impression Music (24/88.2, 24/176.4, 32/352.8) | Computer Audiophile
However, the file is not available anymore.
yes the extra bits are generally used for channel mixing and multiple channels in recording. some audio interfaces will have 32 or even >64bit processing for effects and EQ etc, but the actual audio file will still be 24 bit. as Bunpei says, floating point 32 bit process, not 32bit audio files. you will not be able to tell any difference even if you can manage it. sabre mostly uses the extra bits for multiple channels, volume control and filters. it will play 32 bit audio, but nobody uses it because there really isnt any point, as anything you find that is this format will have just been upsampled/dithered
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I'm downloading a concert from Linn Records and I guess I'm pretty happy that it is not 32-bit...... The 24/192 studio master version is 4516 MB and it going to take me about two days to download it.....
I guess I will need new hard-disks and internet provider if hi-rez audio becomes a hobby.
With the UC192 connected to BII via IIS the files plays without hiccups and the sound is indeed extraordinary.
I guess I will need new hard-disks and internet provider if hi-rez audio becomes a hobby.
With the UC192 connected to BII via IIS the files plays without hiccups and the sound is indeed extraordinary.
Hi
I am about to start a build with a buffalo II. I am canabilising the transport from a Cambridge Audio CD500SE. I have bought a Tentlabs X03.2 clock, and I am planning to install it in the system, and feed the DAC direct from the XO3.2 via spdif.
Given that the ESS9018 chip is going to reclock anyway, is there any point in adding the Tentlabs clock? If there is not going to be a significant gain in SQ, then I may as well leave it out.
http://www.tentlabs.com/Components/cdupgrade/xo2xo3/index.html
I am about to start a build with a buffalo II. I am canabilising the transport from a Cambridge Audio CD500SE. I have bought a Tentlabs X03.2 clock, and I am planning to install it in the system, and feed the DAC direct from the XO3.2 via spdif.
Given that the ESS9018 chip is going to reclock anyway, is there any point in adding the Tentlabs clock? If there is not going to be a significant gain in SQ, then I may as well leave it out.
http://www.tentlabs.com/Components/cdupgrade/xo2xo3/index.html
I don't know of Buffalo's but you can use the sabre in synchronous manner , probably loose some DNR that way but you retain the bit-perfect signal, sounds like your system is a traditional chain with volume at the preamp, so it might worth a try.
Do you have a scope? You can compare the S/PDIF signal with and without the Tentlabs clock and see if it is an improvement.
I'd just get a better transport (even WLAN), an ASRC is mucho happier with high res , and upconversion is possible with good results on computer.
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Dublin, I would strongly suspect that the Tent clock would be worthwhile. My experience (and that of plenty of other people too) is that the Sabre DAC is not immune to jitter, ASRC or not.
Thanks. I will continue with my plans and use the clock. Just needed some reassurance.
Brian: I will try and borrow a scope to compare.
Brian: I will try and borrow a scope to compare.
2 questions
Question 1. Do I have a problem? My BII/Legato plus 4 Opus boards are in the new chassis and sounding wonderful. All of the boards are wired with homeruns (both power and digital). However, because of Brian's comment on buffering I thought I'd peek at the signal integrity into the DAC that is farthest from the I2S source (4.5-5 inches). For this purpose I thought the L-R clock would be fairly predictable (96kHz at the time). The result does not appear 100% clean, because the rising and falling shoulders are not uniformly sharp. I should add that I don't know how clean is the signal entering the chassis. Sound is good, of course. Worry or don't worry?
Question 2. The DAC board from which I steal I2S uses a DB25 cable from a PCi interface card in the computer. I re-boxed the basic DAC circuit board to make room for all the added Twisted Pear boards. I had to use an extension cable (about 9") to carry the main signal into the base DAC board's integral 25 pin connector. I had a ribbon cable on hand so I used it. With the signal routed through the ribbon cable extension - no lock! With the DB25 cable direct into the base DAC board, immediate lock giving the results shown on the scope picture. What am I missing and any suggested solutions for an in-chassis extension?
Thanks in advance!
Frank
Question 1. Do I have a problem? My BII/Legato plus 4 Opus boards are in the new chassis and sounding wonderful. All of the boards are wired with homeruns (both power and digital). However, because of Brian's comment on buffering I thought I'd peek at the signal integrity into the DAC that is farthest from the I2S source (4.5-5 inches). For this purpose I thought the L-R clock would be fairly predictable (96kHz at the time). The result does not appear 100% clean, because the rising and falling shoulders are not uniformly sharp. I should add that I don't know how clean is the signal entering the chassis. Sound is good, of course. Worry or don't worry?
Question 2. The DAC board from which I steal I2S uses a DB25 cable from a PCi interface card in the computer. I re-boxed the basic DAC circuit board to make room for all the added Twisted Pear boards. I had to use an extension cable (about 9") to carry the main signal into the base DAC board's integral 25 pin connector. I had a ribbon cable on hand so I used it. With the signal routed through the ribbon cable extension - no lock! With the DB25 cable direct into the base DAC board, immediate lock giving the results shown on the scope picture. What am I missing and any suggested solutions for an in-chassis extension?
Thanks in advance!
Frank
Attachments
Concerning the LRCK image, let me add that the larger 'deformations' in the waveform don't change when the scope is following the signal on internal triggering. The only thing that changes is the shape at the very corner of each plateau. Again, your thoughts will be much appreciated... 🙂
Answers to my nonEE Qs
It turns out that my 20MHz scope is too slow to look meaningfully at digital inputs other than something as slow as the LRCK. [No complaints because the scope was free... 🙂] My conclusion is that the small amount of variation I see there is not an issue...
...and even thought the ribbon cable checked out with a multimeter, I suspect that the connections by the perforating 'teeth' were simply insufficient for the likes of PCM. I soldered a nice shielded cable in place and all problems solved.
It turns out that my 20MHz scope is too slow to look meaningfully at digital inputs other than something as slow as the LRCK. [No complaints because the scope was free... 🙂] My conclusion is that the small amount of variation I see there is not an issue...
...and even thought the ribbon cable checked out with a multimeter, I suspect that the connections by the perforating 'teeth' were simply insufficient for the likes of PCM. I soldered a nice shielded cable in place and all problems solved.
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Multi-Channel Asynchronous USB to I2S Interface
I’ve just started a thread on a new design for 8-channel Asynchronous USB to I2S interface. Please have a look: http://www.diyaudio.com/forums/digi...i-channel-asynchronous-usb-i2s-interface.html
Any feedback is highly appreciated.
exa
I’ve just started a thread on a new design for 8-channel Asynchronous USB to I2S interface. Please have a look: http://www.diyaudio.com/forums/digi...i-channel-asynchronous-usb-i2s-interface.html
Any feedback is highly appreciated.
exa
dear all Oppo fans: I have a question regarding the I/V stage of ESS9018... Has anyone tried THS4032 for I/V and LME45720 for output Ops?
Output Level
Hello diyers, I have a question for you regarding the output stage of this ess dac and of any dac in general.
I see that most dacs output 2Vrms unbalanced and double the voltage for balanced operation. Having my amplifier an output sensivity of 4dbu=1.227Vrms I would like to have a dac with a balanced output matching this value at 0dbfs, or less.
I recently had a talk with a designer who told me that decreasing the gain of the output stage of the dac would compromise the SNR of the signal. His advice would be not going lower than 1Vrms unbalanced/2Vrms balanced.
I don't get why this should be the case since in my opinion the gain of the output stage should increase the both signal AND SNR, and outputting a lower signal should be always better than outputting a higher one and then attenuating it (in any way).
Hope you can enlight me on this?
Thank you
Hello diyers, I have a question for you regarding the output stage of this ess dac and of any dac in general.
I see that most dacs output 2Vrms unbalanced and double the voltage for balanced operation. Having my amplifier an output sensivity of 4dbu=1.227Vrms I would like to have a dac with a balanced output matching this value at 0dbfs, or less.
I recently had a talk with a designer who told me that decreasing the gain of the output stage of the dac would compromise the SNR of the signal. His advice would be not going lower than 1Vrms unbalanced/2Vrms balanced.
I don't get why this should be the case since in my opinion the gain of the output stage should increase the both signal AND SNR, and outputting a lower signal should be always better than outputting a higher one and then attenuating it (in any way).
Hope you can enlight me on this?
Thank you