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RayW

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  1. LD, I've been using a Promise DS4600 for some time now. I bought it diskless and added 4 Hitachi enterprise-grade 1 TB drives that were formerly on the CASH list. It is set up as RAID 5. The manual gives transfer rates as up to 480 Mbps (60 MByte/s) for USB 2.0 and up to 800 Mbps for Firewire 800. My Weiss DAC2 is connected to the Firewire port on my iMac. I prefer not to put anything else on the Firewire ports, so the DS4600 is connected via USB. This is certainly fast enough for audio. A 10 min track of 44.1 kbps, 24 bit stereo is about 160 MByte. I'm using Pure Music in memory player mode, in which it loads the entire track into RAM at once. For a 10 minute track, the DS4600 activity lights illuminate for about 3 seconds, which is consistent with the claimed transfer rates. There were some initial problems with the DS4600 that seem to have been cured by a firmware update. (I'll elaborate if anyone asks.) There is some barely audible fan noise and disk seek noise, but other than that it is doing its job. Ray
  2. Assuming your recording is stereo and the vocals are in the center, you may be able to get the effect you want using the Channel Mixer effect. With this, you can subtract a fraction of R from L and a fraction of L from R. So whatever is the same in L and R gets reduced. Vary the fraction as you listen to get it right. I've used this on Church recordings as well, which were recorded from a mix in which the vocal mics fed L & R equally, and a stereo pair of mics more distant from the source provide ambiance. Sometimes the vocals are too strong, and this technique makes it better. Ray
  3. Miska, Thanks for your comments. The question remains as to why some 88.2 and 96 kHz recordings have spectra that fall off rapidly around 22 kHz. I hate to think that they have been upsampled from 44.1 kHz. If they were recorded in DSD and then converted, I wonder whether or not some DSD-to-PCM conversion software includes low-pass filtering. I took a look at a 2.8 Mbps DSD live recording I made on a Korg MR-1000 and then converted to 96/24 PCM using Korg's AudioGate software. The spectrum bottomed out at about 25 kHz and then increased at higher frequencies. So this software does not low-pass filter. Ray
  4. I took a look at the spectra of 6 of my purchases from HD Tracks, specifically to determine what exists above 22 kHz. (I used Adobe Audition to do this.) I'm disappointed to report that only 1 of the 6 seems to have musical information there: See attached Phil_Orch_Strauss_Hindemith.JPG. Two recordings in my sample look like CHSA_5048.JPG. Here there is rising quantizing noise above about 22500 Hz that covers whatever musical information may be there. This I believe is characteristic of the DSD recording process. The remaining three recordings had spectra like the one shown in MDG_906_1363_6.JPG. There is a sharp rolloff above about 22 kHz, as with 44100 CDs. But these have rates of 88200 and 96000! More confusing still is that the liner notes indicate the MDG recording was issued as SACD. I'm guessing that these were recorded in DSD, and since DSD quantizing noise dominates above about 22500 Hz, it was filtered out in the conversion to PCM. Does anyone out there know if this is the case? What worries me about 22500 Hz low pass filters is that some of them have pre-ringing. This is why Meridian introduced apodization: to get rid of the pre-ringing. I can't hear tones anywhere near 20 kHz, but I can hear the difference apodization makes on some early CDs. Ray
  5. I've been using one for a few months now with my iMac and am generally satisfied. I bought it diskless and installed four of the Hitachi 1 TB Enterprise grade drives from the C.A.S.H. List. It is configured as RAID 5. Setup went smoothly. The drive trays are rather flimsy plastic, but it doesn't really need to be rugged, I guess. I initially connected it to the FireWire 800 port on the iMac. The FireWire 400 port connects to a Weiss DAC2. There were a few occasions on which DAC would stop working. (There is only one FireWire controller servicing both ports.) I switched the DS4600 to USB and have had no such problems since. It has a variable speed fan. Most of the time I can't hear it, but it does have an annoying habit of reving up the fan for about 1 second on occasion. This is only noticeable if no music is playing. Disk seeks are also audible if no music is playing. The DS4600 comes with a host application which allows checking health and status of the drives, which I like. On the other hand, this application has crashed a few times! Ray
  6. Barry has explained well the word lengthening that occurs with digital signal processing. I just want to add one comment: As computer audiophiles, we do not need to convert the result back to 16 bits! I've been doing some mild parametric EQ using Adobe Audition. I import audio from CD and temporarily save as 32-bit floating point. All processing is done in floating point. My final step is to convert the floating point to 24-bit integer (using dither). The 24-bit files go into my music library on disk and are played through a 24-bit DAC (Weiss DAC-2). I believe the 16-bit CD remains the weakest link, that is, what quantization noise I've added is negligible compared to the quantization noise added in producing the CD. Ray
  7. Here are a couple of alternatives to the DataTale Direct Attached Storage (DAS) unit. Like the DataTale, they hold 4 drives and have multiple RAID modes and interfaces. Anyone have experience with or comments about these? Promise Technology DS4600: About $370 diskless or $800 with 4x1TB drives. Pros: They publish a disk drive compatibility list. Enterprise-class drives with TLER (Time-Limited Error Recovery) are recommended. This unit is sold by the Apple Store, so should be qualified for Macs. LaCie 4TB 4big. Can be bought for $657 (B&H Photo Video) with drives. Pros: Claims to have given much design attention to making the fan quiet. Also sold by the Apple Store, so should be qualified for Macs. Cons: Not sold without drives. They choose the drive model. You don't know what it is until you get your hands on one. If a drive needs replacement, you are supposed to contact LaCie. DataTale 4-bay: About $300 diskless. Cons: No drive compatibility list. This is a concern because of the TLER issue discussed in another CA thread. Ray
  8. Chris, I'm thinking about getting a RAID 5 with Firewire, so I checked this out. They offer it without disk drives or with WD green drives. I just read about the issues with green drives and RAID in the QNAP TS-459 thread, so I asked OyenDigital for their opinion. Here's what they said: My question: > WD recommends against using desktop drives in a RAID because of deep recovery cycle - See WD Answer ID 1397. Yet you sell the DataTale 4-Bay with WD desktop drives. What is your position on this issue? -------------------------- Hello Ray, Regarding WD's hard drive compatibility issue, we are aware of this note for a period of time. We have also consulted with WD to ask the TLER in more detail, but WD did not address this issue to us per our request. We find that only WD raises the issue of different version of HDDs when applied to RAID systems. We have not encountered any problem when using WD desktop drives HDDs in the DataTale RAID series. However, since WD has raised this issue, you can certainly purchase the RAID enclosure and install any drives that you wish, such as Enterprise drives or another manufacturer's drives. Please let us know if you have further questions. Regards, Steve / Customer Support Oyen Digital LLC 4668 Bald Eagle Ave Saint Paul MN 55110 Phone: 866-768-0659 Email: [email protected] ----------------------------------------- So I'm still not sure what to do. Ray
  9. Here's what works for me: I do Parametric EQ for room correction. (I won't go into how I got the parameters here, nor argue with the purists who don't believe in room correction. That would be beyond the scope of this thread.) The EQ is not done in real time, but rather as part of the process of transferring audio from CD to hard drive. I use Adobe Audition on a PC, although other Digital Audio Workstation software could be used. The steps are: 1) Read in CD. 2) Convert 16-bit integer to 32-bit floating point. 3) Do EQ processing in 32-bit floating point, so additional quantization distortion is negligible. 4) Convert floating point to 24-bit integer and save as .wav files. 5) Transfer 44.1 kHz, 24-bit .wav files to iTunes on MAC and play through Weiss DAC2. There is actually another step between Steps 3 and 4: 3a) Check for peaks above full scale that would clip in Step 4. Note maximum over CD and attenuate all tracks by that amount while still in floating point. The weakest link in the whole chain from recording to playback remains the 16-bit quantization used in producing the CD. (I am also experimenting with apodization filtering, as championed by Meridian, as part of this process. I've found that it does improve soundstage depth on some older classical CDs, i.e., from the 1980s & 1990s. Oops, I'm off topic again, so I'll quit.) Ray W. -------- iMac -> Weiss DAC2 -> McIntosh C46 -> McIntosh Mc402 -> Dynaudio Evidence Temptations
  10. RayW

    imac???

    Chris, Thanks for setting me straight on that one. My iMac is in a cool basement and the CPU is almost idle when playing music, so perhaps my fan runs so slowly that I can't hear it. RayW
  11. RayW

    imac???

    Also, the iMac is fanless and thus silent. I'm happy with mine. I'm using it with a Weiss DAC2 via Firewire. RayW
  12. A Bit Perfect Test: I started to write this up a couple of weeks ago, but dropped it because I got the sense that bit perfection of iTunes was a closed issue. But after reading the latest postings in this thread, perhaps it is still open. So, for whatever it's worth, here's what I did: I recently got a Weiss DAC2 and an iMac and wanted to know whether iTunes was passing bit perfect audio to the DAC2 via FireWire. I was happy to verify that it does! Here's how I know: I also have a dual boot Windows XP / Linux PC with a Lynx Two audio card that has an AES/EBU input. The DAC2 has an AES/EBU output, so it was easy to make a direct digital recording of the DAC2 output while playing a 96/24 WAV file (one of the HD Tracks free samples). The Lynx Two card has only Windows drivers. The card is controlled from the Lynx Mixer application. I set this up so there were no other inputs being mixed in and Sample Rate Conversion was off. The recording was done with Adobe Audition 1.5. To record 24-bit audio, one must record in 32-bit floating point and then save the file as packed 24-bit WAV, with dither off. Next, to compare the source and output files, given that they have different lead-in lengths because of manually clicking Record on the PC and Play on the Mac, they first must be aligned. I'm an old C programmer, so I wrote some code under Linux to do this (named afcmp.c). It does the following: 1) Finds the first run of 8 sample pairs in one of the files with energy above a threshold (so we're in the music and not the lead-in). 2) Looks for the same 8 sample pairs in the other file. 3) If found, compares all samples from this point to the end of the shorter file and reports the number unequal and the total number of samples compared. To verify that afcmp itself was working correctly, I used Adobe Audition to change the values of a few individual samples. afcmp correctly reported the number changed. afcmp uses the audiofile library found in some Linux distributions and available free from http://www.68k.org/~michael/audiofile/ It ought to port to the Mac, but I haven't tried this. I will be glad to share the source code afcmp.c if anyone wants it (and if Chris can provide a way to post it). Some more of the ever-important details: 24 inch iMac with both Firewire 400 and 800. Firewire 400 used with DAC2. Mac OS-X 10.5.7 iTunes 8.2 Desired sampling rate set with AudioMidi before starting iTunes. In AudioMidi, default output set to DAC2 and Mac system sounds output set to built-in audio output. I also checked playback with the Finder's Preview function and with Soundtrack Pro 2 (part of Apple's Logic Studio). In the latter, Monitor Volume was verified to be at the default 0 dB. Both of these are bit perfect as well at 96/24. RayW
  13. Audio_ELF: Here is a good article on the subject: http://www.stereophile.com/features/396bits/index.html The main issue with S/PDIF & its professional equivalent AES/EBU is that the master clock is at the source (sound card) and the receiving DAC must recover and sync to this clock. Doing this with low jitter is difficult, as the above article describes. The authors address the question of whether the interface is fundamentally flawed. Recent high end DACs, e.g., Weiss and Berkeley Audio, do a very good job of overcoming this difficulty. See for example some of the white papers and manuals on the Weiss web site. My own choice of the Weiss DAC2 was strongly influenced by its FireWire mode, in which the master clock is in the DAC and the computer is slaved to it. But it also sounds great with S/PDIF input from a Meridian G08 CD player. RayW
  14. Ras, I now have about 12 parametric EQ sections per channel. I got to this point after several iterations of choosing parameters to flatten the response, taking care not to apply too much boost at any frequency, and then listening. I fixed the biggest deviations from flatness first, and then went on to the smaller ones. All corrections were below 3 kHz. It's not necessarily the case that more bands are better. As for my remark about not spending $, I was thinking one thing and wrote another. I've spent lots of $ on this hobby and probably will continue to do so. What I meant to say was that the various room correction devices available use different algorithms and I didn't want to commit to one manufacturer's approach based on reading the reviews. I wanted to experiment first. Its not just the destination but the journey that's enjoyable. Thanks for the links. From a quick look, I see in DRC-FIR that the author has put a great deal of effort into developing his system and has provided detailed documentation. I'll enjoy studying it and trying the software. The room EQ I have now sounds significantly better to me than without EQ, but I'm sure it can be improved. Also, there's the obvious issue of compatibility with hi-res. Some room correction hardware is designed to go between preamp and power amp and so consists of A/D conversion, DSP, and D/A conversion. Clearly, you wouldn't want to run the output of your Berkeley Alpha DAC through one of these! I doubt if there are any room correction devices with digital I/O that will run at 176.4 kHz. Software room correction seems like the only viable alternative for hi-res playback at this time. It doesn't have to run in real time; you can pre-process your music files. It may be heresy to some of you, but I have run some of the Reference Recordings HRx material through my Adobe Audition-based parametric EQ. The computations were done at the source sampling rate of 176.4 kHz and 32-bit floating point. What came out was still hi-res, but with a flatter response at my ears and so more realism. RayW
  15. VincentH, Room correction is what got me interested in computer audio (in addition to hi-res). Before getting into how to do it, I would like to offer my opinion that electronic room correction can make a significant improvement, even with high-end speakers and electronics and some physical room correction (in my case, absorber panels made of fiberglass house insulation to reduce reflections from the side walls). I wanted to experiment with room correction without investing thousands of $ in hardware. I read various papers published by the Audio Engineering Society and in Stereophile and The Absolute Sound and settled on the following approach: Measure the frequency response at the listening position and also at several other random positions. Do this separately for left and right speakers, and don't go higher than about 3 kHz. If there are significant peaks at the listening position, attenuate them. If there are dips at the listening position, only boost them if the dip appears at all the other mic positions. To do this, I used the classic parametric equalizer approach. Each section of a parametric equalizer has 3 parameters: center frequency, Q (i.e., bandwidth), and gain/attenuation. Rather than buy a hardware parametric equalizer, I used Adobe Audition digital audio workstation software that I already used in my amateur recording endeavours. Audition has a parametric equalizer in its effects menu. After several iterations, I found a group of settings that sounded good to me. At first, I would rip a CD to the computer, equalize it and burn it to another CD. This is not a good way to do it because when equalizing, the sample word length increases beyond 16 bits, so one has to round it down to 16 bits again to burn another CD, adding more quantization noise. The preferred approach is to write the equalized audio to hard drive at 24 bits and play that through the DAC. RayW
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