Mani, congratulations with Pacific.

You will not have problems to drive with balanced Pacific your single-ended amp. Juts buy a good quality XLR to RCA adapter or order XLR to RCA cable and you will be set. You do not even need to care about the pins layout as it might be configurable by the unit itself. You will lose 3dB but Pacific has plenty output in its D/A section, I do not remember exactly but at 0dB digital it will swing near 7V or something like this. I do not use digital gain/attention with it.

 I do not think the output impedance of Pacific DA would be a problem.  It is directed-coupled and it has 20R – it will drive whatever you can imagine. The only thing that you need to be cautious is Pacific’s input impedance on analog side. It is 13K balanced and even less unbalanced.   You would need a relatively strong source to drive it to get of of your recordings a full LF bloom.

Thanks Romy, I'll try connecting with an XLR to RCA cable. I've actually ordered a Henry Matchbox HD also because I can use it elsewhere if it doesn't help in this particular case. In any event, I'm hoping to share some thoughts on the Model 2's sound as a DAC early next week, once I've had a few days to listen to it.

Keith, I've contacted the people who sold the Model 2 to me, and they may have more that they are happy to sell. They started with 18 (yes 18!) units and have off-loaded quite a few already. The company is a recording/rental studio in the US called DMT. Feel free to contact Vicki (vicki@dmtrentals.com) to see where they stand. But if you're looking to import into the UK, get prepared for a hefty import duty bill!

Mani.

coops wrote:

I did hear a rumour that the Berkeley boys were going to make a new 'model III' but Mike Ritter quashed that particular rumour…

I did not hear this rumor but if I did I would not give to it much credit. With all credentials and capacity of the Berkeley Boys I do not think that there is a market today for another “expensive” mastering processor. The Pacific’s David told me that it cost $15K for them to manufacture Model Two, it was very expensive, it is very expensive and it is why no one leases them from DMT.  The contemporary audio professionals feel more comfortable with mass-market single-chip devises that have 48 channels aboard.  I would not blame them: looking what quality sound in films and CDs they produce and looking that no one care I see why they need anything better. I think that Mecca of digital design was in 90s; then people were trying to punch the things… Perhaps the elements base is better today but it is not only the supply but also demands…

Anyhow, the subject about the Pacific Microsonics Model III is very interesting as it would be fun to conceptualize what I would like to have improved in Pacific Model One and Two.

I would like Pacific to recognize the incoming stream and adjust own sampling rate and resolution accordingly. If Pacific need some time to load own operation system for given mode then I do not mind for a first minute it did down/up conversion. This functionally is absolutly not needed for pro use and particularly mastering.

I would like Pacific is able to use two software-switchable digital inputs in case it used in single wire mode.

I would like Pacific to have one more exhausting fan on its back which would assure a long “hot” operation.

I would like Pacific to have an option for turning the whole display out.

I would like Pacific DAC to have a more extreme bass.

I would like Pacific A/D to have an option to defeat the DC offset correction.

I would like Pacific to have USB option. Not just the USB jack and a marketing booklet but a properly implemented USB with an option to run the DAW’s reader right from the Pacific master clock.

I would like Pacific to have an onboard network card.

Since Pacific is a damn huge mastodon I do not see why Pacific shall not have a slot for insertion of a contemporary 20G-80G flash card or a hard drive to write a master file right here locally right from the clock that drives A/D with no external interfaces.

A visitor of my site just sent me a link today:

http://www.ultraaudio.com/features/20090201.htm

“Ritter told me that they may consider development of a Model Three processor in a year or so.”

An interesting article all together, still too much in the “Soundstage” frame of mind...

For a living, I teach presentation skills to executives and senior managers ( www.youtube.com/watch?v=DJetXp2r3pc ) . I always begin by considering the listeners and what they are looking for. I'll ask people what turns them off when they are in an audience listening to a speaker, and having asked this to literally thousands of people all over the world, the resounding no. one answer is, “A monotonous voice”.

manisandher wrote:

For a living, I teach presentation skills to executives and senior managers ( www.youtube.com/watch?v=DJetXp2r3pc ) . I always begin by considering the listeners and what they are looking for. I'll ask people what turns them off when they are in an audience listening to a speaker, and having asked this to literally thousands of people all over the world, the resounding no. one answer is, “A monotonous voice”. So, what's the difference between a monotonous speaker and one who uses intonation, varied volume, varied tempo, pausing and good enunciation? They can both deliver exactly the same content/information... but the former will bore you and the latter will engage your heart and mind... by emphasizing key messages through totally congruent behaviour, he/she will help you understand the ebb and flow of his/her ‘argument’. (Great story-tellers have this art down to a T.)   For me, an audio system has a similar effect. An audio system either bores me (pretty much irrespective of the quality of the music itself) or engages my heart and mind by helping me to ‘understand’ the conductor’s/musicians’ interpretation of the music.   This weekend I compared my trusty Esoteric D70 DAC to my newly acquired Pacific Microsonics Model Two. The former simply bored me. The latter engaged me... in a way that I haven’t felt in a long time.

Mani, your enthusiasm about Pacific’s DAC is more in my view is an indication of what the Esoteric D70 DAC is. Do not get me wrong: Pacific is very good DAC but you needed to understand that it was built as a mastering processor as DAC is just for a controlling functionality. The major bloom in this machine went for a Rolls-Royce level DAC and a truly fantastic D/D converter.  What you try to record with it you will see how far it is from anything you might use. Particularly pay attention what Pacific does with imaging of analog source… The boring intonation is a properly of most of the AD/DA but also all of them flatten space.

Pacific is a fun machine and it is hard do not like. The opportunely that we had with DMT were ridicules and in a way laughable – it will hardly happen again. BTW, I have a guy who asked me to get Pacific for him. I spoke with DMT and they told that they all gone and that they then have 3 offers of the first refusal. Probably they just needed immediate cash as they could easily sell PM for twice or more of money they asked. If I did not have Model One then I would probably pay the twice price…

I did today my first recording with Model 2. The configuration-wise it was very cool. The Pacific now in single-wire mode while it does A/B to 88/24 and the clock also outputs 88K. The single-wire mode is the reason why I moved from the Model One to Model Tow.

Celebrating the American D-day (the opening thyme notes of the symphony mean in Morse code letter V) today WHRB broadcasted the live–to-tape performance of Beethoven: Symphony No. 5 in C, Op. 67 by Bavarian Radio Symphony Orchestra lead by Mariss Jansons. The concert took place in January 2004  it was prepay good, but I generally a sucker for live play of German’s “second tier” orchestras. I recorded the broadcast and played it back. I initially thought to have both Pacifics: Model 2 and Model 1 record the same feed ( Lynx card’s mixer allow to do it on the sane DAW) but the Model 1 was not hooked up what the broadcast was about to start. So, I recorded only with Model 2. Listening it I have to say that I did not detect any difference compared to what I have accustomed with Model 1. It was the very same sound and I did not recognize any auditable difference….

Mani, interest in using Pacific as the only all-time on DAC revised my interest in pacific D/A converter section. I consider it as a great 16bit DAC and very good 24Bit DAC. The reason I did not elevate in my appraisal Pacific DAC to great status with high resolution format because Lavry Gold in my observation has “harder” the very lowest end.

A few days back I deseeded to review my outlook about it. The key in this was to be able to connect Pacific to my preamp with identical quality as Lavry Gold has. Lavry used PAD Vintage Dominus cable; among those few who know what it is it will not be more questions. The Pacific needs 2M cable run, and I have no extra 2M Dominus. The way how Lavry is connected I can’t replace the cable. So, It took for me for a while to find for Pacific a 2M cable that more or less would be compatible to Lavry’s Dominus. At last I did and was able to listen the Pacific DAC at 24Bit in more or less as optimum mode as I have Lavry setup.

What can I say” it is not as bad as I remember. Well, it never was “bad” but this time the advantage of Lavry in the very lowest bass is much-much-mach less prominent then I remember I experienced last year. In fact it is a near negligible level and might be observed ONLY during a direct switching. There were some changes in operation of Pacific since last year – any of them might be a reason for the change on sound: Pacific now is powered from PS2000, Model Two vs. Model One, new cable between Pacific and the load.

In concussion it looks like Pacific might be a very good DAC to keep it as the ONLY DAC if the auto-detection of the rate is not a subject. The only mystery that I feel at this point is the HDCD detection. Mani reported that the BSO files he downloaded were HDCD encoded. The Audio Mirror guy suggested that my complain about the BSO files’ screwed imaging shall not be affective if I engage HDCD decoding. So I did but Pacific does not recognize the HDCD code in the BSO files. I played with operation mode and tried the different detection resolution but I see no HDCD code in there. Interesting….

Mani, Pacific is freaky and it detects HDCD code only at the resolution that it was encoded.  If it was coded at 16 bit then it will not read the HDCD at 20 Bit. There is an option in there that set the HDCD detection death… I did not play a lot with HDCD. What I got my Model One I experimented with playing and recoding in 16Bit and HDCD. It was exceptionally good for 16Bit. However going for 20Bit and 2X I recognized no benefits of HDCD at all and I never used it, or cared about it. I do not particularly buy in Sound Mirror’s idea of running HDCD at 24/88…

I think the reason why I do not read the HDCD code because I do not use any of Pacific DD modes. When I reset Pacific in DD operation then sure it recognizes the HDCD but to let it to run properly I need ether to disconnect the digital output cables from my Lynx card or to remap the Lynx mixer. In my scenario Lynx’s has two active recording channels (Pacific and Lavry 122) and two active output channels (Pacific and Lavry 924), both of the outputs run from the same Playback channel. So, what I use Pacific in DD mode then Pacific and Lynx run digital circular reference and this mode does not make any sense to me, unless I do specific DD task, that I practically never do. 

So, my standard operation mode is Romy88 that is basically AD2X at 24 bit. In the mode I can AD my source to 88/24 and monitor it live with ether Pacific or with my externals Lavry as my second playback channel on the Lynx mapped to the first playback channel. An additional benefit is that if I have no analog signal and I would like to run Pacific as a pure DA then I flip the meter (second button) from Output to Digital-IN (it set to follow meter) and then I read my digital stream. I stays with this mode as it allows me do not change anything when I wish to use Pacific ether as standalone DAC, or as a standalone ADC, or as a combined ADC + DAC. What is unfortunate is that in AD2X mode Pacific apparently does not read HDCD when I source the Digital-IN. it kind of make senses…

Anyhow, I detected today another very interesting discovery: the Model Two runs substantially cooler then Model One. I ran today Model Two all day long. Yearly morning I was interested to record the live play of Elgar’s Piano Quintet from Rockport Chamber Music Festival and Perahia’s live play of Mozart’s Concerto No. 21 with Bavarian Radio from Munich. The mid of the day it was. Both were very good. At the mod of the day the Chicago Symphony with Alan Gilbert, David Robertson and Paavo Jarvi where trying to play American music. It was live-to tape, I was recording but I did not listen it yet. I very am interested about the Bernsteins’ First Symphony with David Robertson. It was the same guy who played Orff tow year back with SF and it was stunning! At evening it was Lyric Opera of Chicago live-to-tape Massenet’s “Manon”. So, the Pasic run from 7AM to midnight, I just did not care to shut it down. There were two things that I observe.

First: the Model Two was just slightly warm after so many hours – the Model One would be hotter. Second, the Model Two looks like outputs less noise into the line and impacts less my tuners with digital noise. Both are very good signs…

Romy the Cat wrote:

So, what I use Pacific in DD mode then Pacific and Lynx run digital circular reference and this mode does not make any sense to me, unless I do specific DD task, that I practically never do.  The Cat

manisandher wrote:

What I'm having more difficulty with is playing 176.4/192KHz sample rate files with the Model Two in Master mode. The Model Two can only output a clock at 1x or 2x 44.1KHz or 48KHz. My AFI1 cannot double the clock frequency to match the sample rate.

Yes, it was exactly what the Model One did – it outputted one 1X clock. That sucks. You can double the rate with dual-wire: the 88K with Model One and 176K with Model Two but to double a reference clock is a totally another problem and I would not go it. However, I do not play anything at 176K….

manisandher wrote:

So, at these sample rates, I cannot use the Model Two in Master mode, which is the only mode I can live with.

Mani, this is a controversial subject and in my view it is not so straight forward. When you record you defiantly would like to slave the recording devise to A/D. You might do it not only in Master mode, but you might slave the sampling marks right from you AES cable, like many other DACs do.  The idea of a separate interface for clock is divisive, Mr. Lavry argued this subject many times, claiming that the clock slaving interfaces have more jitter and more damage than better internal clocks. You can read here and in many other his white papers.

http://recforums.prosoundweb.com/index.php/t/14324/0/

I do not have my own position on the subject but what I feed my computer player to DACs of the Lavry Gold or Pacific levels I very much sure that their own clock are way more superior and do not need to be slaved. It is not to mention that that they relock data anyhow. To slave the sources from the upstream devise? I do not know if I like it. Did you try to listen and did detect sonic benefits from slaving your PC interface from you DAC?

Romy the Cat wrote:

Did you try to listen and did detect sonic benefits from slaving your PC interface from you DAC? The Cat

1. MASTER (strong, strong preference - I will not listen to any other mode)
   
2. REFWCLK
3. DIG_IN

manisandher wrote:

Oh yes, I've tried this for sure. There is a difference between the three modes; DIG_IN, REFWCLK and MASTER. The first two seem to add something in the HF that I can imagine many people judging as being 'extra resolution' and therefore better. It's not unpleasant, but I find it distracting.

Interesting. I always thought that following component must be slaved to the leading component not wise-versa, unless an external separate master clock used for everything. Since I use for play Lavry Gold that does not offer clock output (I use Pacific only to record) I never bother to consider to slave my DAC, or to slave my Lynx cared from my DAC. Perhaps I need to re-listen my files with Lynx slaved from Pacific in Master mode as you do. When I did my listing of Lavry vs. Pacific I always was running Lynx own clock and Pacific own clock (I thought it would be too absurd to let Lynx to slave Pacific as I feel Pacific clock shall be much better, presumably).  Also, do not forgers the Model One with which I made all of my Lavry vs. Pacific experiment has only 1X clock output, so what I was running 88K files I had reference only for one wire. I spoke many times with Lynx and with Pacific people about the mastering and slaving at “half-speed” and they assured me that it is an OK mode to operate. I might be so but I did not feel comfortable as when I run high resolution sampling rate monitor on both lines of the 88K in dual-wire mode then I clearly saw that the rate changes in each wire with slightly different tempo. That was the ONLY reasons why I bought the Model Two – to run 88K with 88K clock during A/D. Still, 99.99999% of my files are my own FM recording, the source that is HF challenged to begin with, and it has nothing above 16K. So, to have “added something in the HF” might be something that I would not even recognize. Anyhow, following you advise I will excrement more with slaving Lynx from Pacific during D/A.

Still, there is a catch in all those arguments. It is absolutely unknown how you slaving interface is implemented and it is very much that the specific implantation of YOUR or MINE or SOMEBODY ELSE slaving interface would be something that would descried “quality”. I for instance have no idea how my Lynx card behave what it slaved from external clock when it does not record. I also do not know if my WaveLab playing software uses in this mode the Lynx’s clock or the Pacific’s clock. Theoretically the WaveLab shall read the Lynx’s clock and Lynx shall be synchronized from Pacific. Considering that WaveLab read data first I wonder how much in this come from pure imagination. I wonder which devises runs a “delay” of data until the WaveLab get own sampled from Pacific…

manisandher wrote:

Thanks for the link. It's interesting that Dan Lavry doesn't see the point in using an external clock unit, believing the DAC's internal clock to be the best. Even when you have many units that you need to sync together, Dan seems to feel that daisy-chaining them is the best way.

Yes, Dan is a proponent of idea that digital interfaces that case just clock data bring more jitter and more problems than “properly implemented” internal clocks. He might be right or might be wrong, most likely he is driven by patronage for his products but he, with his experience on the subject, is the voice that very much shall not be discarded. In fact there is a lot of rational in this view of his. I personally think that there is no right or wrong in this subject but it all depends the level at which the given topological solution is implemented. I am very much not the person to judge how “properly” slaving or re-cloaking mechanisms were made. Unquestionably Dan Lavry, Michael Pflaumer, Michael Ritter, Daniel Weiss, Ed Meitner  and few others  are and I let them to argue the point. In case of Lavry I was very much “sold” with his  very sexy idea of “properly made multibit”. You might read about it in here:

http://www.lavryengineering.com/white_papers/DA924m.pdf

… starting from the page #9 where he described the theory of operation.

Also here is another relative reading:

http://www.lavryengineering.com/white_papers/jitter.pdf

I do not know if Lavry strikes the idea of D/A slaving just to promote his ideas that he used in his DA924 or it is how he truly feel. It is difficult to say when we talk about manufacturers as the always have an agenda…

manisandher wrote:

In any event, it looks like I'm stuck with using modes 2. or 3. above for 176.4/192KHz sample rates. This is certainly not a disaster... until many more downloads that I'm interested in are recorded and offered exclusively at these rates (probably not very likely).

Well, I am not 176.4/192KHz – minded person and I also have very low expectation for industry will be furnishing the right quality 176 files. (I do not care about 48X sampling rate at all). When I talk about “right quality” I mean that audio people are fools and when they see 176KHz sampling rate then they automatically assume some king of “quality”. The industry takes now and will take in future a full advantage of that simplicity. In my world the definition of quality is not sampling rate but the absence of the intermediate idiots who “master” the files. I would rather prefer to have a raw (never was resaved!!!) file at 44KHz then an edited file at 176KHz. Digital can’t be edited. Any activation of after-conversion DSP is installs kills the file – but it looks the Morons who are and will facilitate us with music do not “get” it…

manisandher wrote:

But another reason why I'd like to use the Model Two in MASTER mode at these rates is that I think it uses a non-oversampling filter. When talking about the Model Two's ADC capability, Ritter wrote: ""If it's going to be a 176.4 or 192kHz DVD-Audio release, then we will not decimate that signal; we use a proprietary filter [non-oversampled] optimized to that sample rate. If it's going to be 88.2/96 kHz, we use 2:1 decimation, and once again we use a filter optimized to that frequency. But in both high-resolution settings, the Nyquist frequency is high enough that we don't use the 'dynamic decimation' process that becomes necessary when we go down to 44.1 or 48 kHz."

Yes, you said about it before and it is a cool to know. I did not know that Model Two shuts down oversampling at 4X. Frankly I personally fees that it shall shuts down any post conversion filtration 176kHz. As Ritter said the sampling rate is so far from auditable range that why to use any filtration at all. You see, Pacific DAC is multibit. (reportedly, as it does not measured as “pure”  SAR multibit). The “regular” Delta-Sigma of the dynamic Delta-Sigma (they call if Delta-Sigma multibit) after conversion has a wide-band nose doe to DC. Not with SAR multibits. Those babies have completely reconstructed signal and noise only at sampling rate minus 20kHz or double 20kHz, I do not remember exactly. So, if our DAC runs at 176kHz then it will out noise at ~140kHz… who cares…. It would be very noise to run this signal out with no filtration of any kind or to put in there a first order L filter. My initial sentiment was that Pacific does something like this. I thought about it what I saw that Pacific provide the XLR-to-XLR adapter with chokes in them as some of the high feedback SS equipment will not be able to handle high intensity 140kHz noise and will oscillate. I think that the same trick shall be used at 88kHz, but this is another story… Anyhow, Ritter said that they use another “proprietary filter”, probably lower order but I would like do not see in there any filters at all…

One more thing…

Romy the Cat wrote:

When I talk about “right quality” I mean that audio people are fools and when they see 176KHz sampling rate then they automatically assume some king of “quality”. The industry takes now and will take in future a full advantage of that simplicity. In my world the definition of quality is not sampling rate but the absence of the intermediate idiots who “master” the files. I would rather prefer to have a raw (never was resaved!!!) file at 44KHz then an edited file at 176KHz. Digital can’t be edited. Any activation of after-conversion DSP is installs kills the file – but it looks the Morons who are and will facilitate us with music do not “get” it…

I would like to stress this point, the 34923847290345 times. When I send a question about the AudioMirrors about the HDCD encoding their files the studio manager replied that was not pleased with my attitude toward to the industry people who “edit” the files. That borough in my memory an event that took place probably 7 years ago or so. A guy contacted me. He was surprised with my disrespect to the hard working editing studios put in the CD. I do not want to give up his name but I would say that he is one of the top notch sound engineers with good industry reputation.  He proposed to me to demonstrate of what is possible in the things are done properly. He proposed me to send him a 16/44 files, he will do his “magic” and send it back to me. So, we did. At that time (it was 7 years ago) he used some crazy machines that people used at that time for film’s CGI and FMV. He explained to me how different it was and why he run all his prosing at 32bit and 704KHz. I hardly understood what he was taking. Anyhow, in a few weeks I got the file from him. Playing my original files and his “edited” files was very clear why I feel that those idiots shell be not allowed to touch sound and if they do then somebody shall cut their hands off.

Romy the Cat wrote:

Perhaps I need to re-listen my files with Lynx slaved from Pacific in Master mode as you do... Anyhow, following you advise I will excrement more with slaving Lynx from Pacific during D/A.

Romy the Cat wrote:

It is absolutely unknown how you slaving interface is implemented and it is very much that the specific implantation of YOUR or MINE or SOMEBODY ELSE slaving interface would be something that would descried “quality”.

Romy the Cat wrote:

He proposed me to send him a 16/44 files, he will do his “magic” and send it back to me. So, we did.

manisandher wrote:

Yes, please try this - I would be very interested in your experience. (Does your Lynx have built-in 75 Ohm termination?) .

Yes, Lynx has 75R clocks in and out.

http://www.lynxstudio.com/pop/product_file.asp?i=26

I do not use them and read he Pacific A/D clock from the AES line…

manisandher wrote:

Romy, would it be feasible to post these files so that the rest of us could have a listen to them also? I'd be interested especially in knowing what he felt the improvements were, and then judging for myself.

Romy the Cat wrote:

Yes, Lynx has 75R clocks in and out.

Romy the Cat wrote:

... all that you need to do it record a file. Then copy the file. Then open one of the files with any file editor you have, make any action that engage DSP, like reduce volume for .5dB on one of the channels. Save the file, then reversed he volume back and save the file again. You will have one “row” file and another edited file. You will be able to see how sound change – the resaved file. Besides some pure sonic degradation, will have some sense of energizing indifference…. Then try to activate any DSP plug-ins that those “mastering fools use” – it will be a nightmare…