MD: Why MD?
I got started with MD to make compilations of LPs to play in my car. But lately, in addition to music, I've been recording interviews with family members for a genealogy project. One of my friends at work borrowed my recorder and mics for a weekend to record some of his relatives and was hooked. He claimed that the result sounded as good as NPR (National Public Radio network) and was so impressed he purchased an MD recorder the next day! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Hi speed dubs from computer...
There's no techincal reason that 2x should not be possible, because 96 KHz S/PDIF sound cards have been avaialble for some time. Also, most existing MD transports are capable of up to 5x. It just hasn't been implemented by any existing MD model. I'm sure that once people got a taste of 2x, 4x would not be far behind... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: WHY?
I have a Sony MD MZ R 70, and today I have discovered my remote control is not working anymore... The remote is just a resistor ladder. If you've got an ohmmeter, connect it to 2 of the pins on the "edge connector" at the end of the remote cable (I forget which 2, but the other 2 are not connected on the R70 remote). When the meter is connected to the proper pins, it will indicate various resistances when you press keys; the value depends on which key is pressed. Sounds like your problem is a open circuit. This test will allow you to determine if its the remote or the recorder. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: computer to MD - digital vs analog
Lately, there has been some discussion and speculation on MD-L about analog vs digital for computer to MD transfers. Below I've enumerated the technical advantages and disadvantages. The advantages of analog are: 1) it *always* works 2) easy level adjustments The disadvantages of analog are: 1) many computers/soundcards have poor internal signal grounding, not to mention a *lot* of high-frequency hash (which leads to noise) and/or undersized coupling capacitors (which produces phase shift and poor frequency response). 2) possible ground problems between the computer and the MD which causes mains leakage and resultant hum. 3) unreliable connections due to the low contact pressure of the mini-phone plugs. 4) an extra D/A and A/D conversion. Digital transfer has the advantages (when it works): 1) No added noise, distortion, phase errors or poor frequency response. 2) No ground problems with optical. Even if coaxial is used, ground problems will not cause noise, as with analog. Digital transfer has the disadvantages: 1) Possible incompatibility problems. For example, the S/PDIF from the DIO2448 is not accepted properly by the MDS-PC2. 2) Bad connections or ground problems may cause intermittent dropouts or pops. 3) Due to lack of "flow control" on S/PDIF, high CPU activity may cause dropouts or pops. For me, the advantages of digital make it worth the hassle. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Lawyers, Discs and Money
I've been getting a lot of e-mails from people who don't understand why they can't simply copy the MP3s to MD faster than real time, so I've put together a rather lengthy explanation. http://home.sprintmail.com/~dalegreer/Mac-MD/lawyers.html An interesting account of the history of SCMS, etc. I disagree, however, that SCMS limits one to 1X recording. For example, Sony has a combo CD/MD that *is* SCMS compliant and transfers to MD at 4X speed. Current S/PDIF computer hardware can produce a 96 KHz S/PDIF, so it would theoretically be possible to construct a SMCS compliant MD machine that would accept this high speed S/PDIF to record at a little more than 2X. (Since MD transports already record in burst mode at 5X speed, it would require redesign of the chip with the S/PDIF interface, but no other major redesign.) Right now, we're seeing a wave of USB to S/PDIF hardware. It ultimately be a more cost effective and easier-to-use solution to *emulate* this hardware in the MD recorder itself. One could also emulate the a USB to serial to Slink (over the same physical USB cable). Doing it this way would retain maximum compatibility with existing hardware and software, but still offer almost all of the computer transfer advantages of MP3 players, and it would still be SCMS compliant! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: RM-X69RF question...
Does anyone know where I can get a service manual for the RM-X69RF? I'd like to add an extra IR sensor as I plan to put the display module behind a drop-down door. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Just got a MZ-R70; love it! Does RM-MZR55 remote work with it?
I have an MZ-EP11 (with LCD remote) and an MZ-R70 (remote with no LCD). The EP11's LCD remote plug fits the R70 and the controls work, but the LCD doesn't. On the 4-pin remote plug, 2 of the pins are for the controls and 2 for the LCD. My guess is that the LCD pins are not connected in the R70. I was somewhat disappointed to discover this, because I would have swapped remotes otherwise. I don't know for sure, but I believe the remote LCD pins *are* connected in the R90. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Speakers
If price were no object, I'd have Martin Logans! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Digital recordings and track marks...
But there is no code or signal in S/PDIF that means "new track starts now." I don't know how the CD subcode information is translated for S/PDIF transmission, but the subcode "P" channel on the disc is just that, a "new track starts now" signal. It is the CD's version of the "wide spiral groove" between tracks on an LP. I've don'e know if the "P" channel makes it into the S/PDIF data stream, but if it does, then the track start signal is available. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: forward from CasetteHouse... cheap
But you have to factor in the shipping and service charge. That will raise the price somewhat. I ordered 100 Hi-Space 74 MDs from Casette House a few months ago. I don't remember the exact details, but I remember shipping and handling as being quite reasonable. They weren't inflating their price that way. So far, I've used the following batches of MDs and all have worked equally well: 5 TDK 74 min 40 JVC "Crystal Gold" 74 min 50 HHB 74 min 100 HiSpace 74 min - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: CDDA
Not wanting to get into a flame war with Rat, but I stand by my previous explanation of CDDA. For those interested, there's some information online: http://www.howstuffworks.com/cd1.htm Also, Ken Pohlman's book, "Principles of Digital Audio" has in-depth explanations of many formats, including CDDA and MD. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: CD-DA
OK, I finally found a page with a good technical expanantion of CD-DA ripping and the problems involved: http://www.cdpage.com/Audio_Compact_Disc/daextraction.html Also some interesting general info on CD, CDROM, MD and DVD here: http://www.sonydadc.com/pr_handbook.asp I've got K Pohlman's book somewhere around here. I'll look and see if he gives the definitive answer about CD-DA track format. (He probably does.) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Sony MD mono workarounds...
I assume they are being combined before going through the MD's ADC I beleive the ADC outputs a normal stereo signal. The channels are combined in the digital domain *after* the ADC, at least on the JE520. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: CD-DA
Rather, the CDDA is stored with an RF encoded carrier wave on the disc. In other words, it's stored in analog. Different transports will come up with slightly different sets of CDDA after reading it. CDDA is *not* analog! It is as digital as CDROM, your computer hard drive, etc. CDs use a sprial track (similar to an LP) that was originally designed to be played continuously, not random accessed. Unlike computer disks, which have a header on each sector containing the sector number so the drive can be 100% certain which sector it just read, CDDA contains no such identifying information in each frame. As long as you play CDDA continuously (like an audio CD player does) there's no problem. Since the frames are in order on the disc, the data in the resulting audio stream will be in order. The problem comes in when one is extracting the digital audio information to a WAV file. In this case, the player does not read continuously but instead reads the data in "chunks". After each "chunk" it has to seek back to where it left off before reading the next "chunk". Since there is no header to unambiguously identify the frame, it seeks back to *approximately* the same place, reading some overlapping data. It then compares the data and discards the overlapping data. Some CDROM drives are better at this than others. A few are very bad, causing the program to give up, but a few are perfect. Most are somewhere in the middle. But as long as the program can identify the overlap, the resulting WAV file will be a perfect copy of the audio data. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony MD mono workarounds...
When recording mono from analog, it appears that it would be best to mix to mono before the signal reaches the MD, then connect only one input channel of the MD, leaving the other unconnected. To get the level correct, the digital record level would have to be set at +6 dB (higher is OK; lower digital record level coupled with higher analog signal level might cause ADC overload, which is *not* indicated by the "over" indicators). Recording mono from digital would require a similar proceedure, done with an audio editor on the PC: mix to mono and place the resulting mono mix on one channel of a stereo WAV file and silence on the other channel, then boost the digital record level 6 dB at the MD. I see so easy way of making a digital mono recording directly from a CD player an optical connection. Maybe one way of fixing it would be to feed the signal though an S/PDIF receiver, invert the LRCLK signal, then connect to an S/PDIF transmitter. This would reverse the left and right channels, but it might also introduce a 1-sample shift. Trouble is, it might be in the same direction as the MD, resulting in a total shift of 2 samples. That said, I think the effect of the 1-sample-error is fairly minimal in many cases due to the steep slope of the high freqency rolloff, when viewed in terms of the "octave" properties of hearing perception. Also, most listeners are much more tolerant of a very stable phase error such as this, than an unstable error such as that caused by tape edge damage on analog master tapes (for example, Neil Young's "Cinnamon Girl"). Meantime, I'll be eagerly awaiting word on which Sony decks have this problem, and if (or when) Sony has it fixed for good! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: minidisc on CNNfn
It's nice to hear when MD vs MP3 is treated objectively in the media. pros: cheaper media longer recording time higher quality recording wider choice of players more editing/mix capabilities I definitely agree with the "pros". (I think higher recording quality is based on the 128K MP3's that are widely available in the 'net; MP3 gives fairly high quality at higher bit rates.) cons: double compression The damage has been done by the 128K MP3 compression. I think that ATRAC 4.5 or "R" compression makes an inaudible difference after that. larger than mp3 players True, but the R70 is really darn small. needs high-end recorder (???) The R70 *is* a recorder. It even has digital in. The real con is that your computer has to have digital out to use it. Also, it only does 1X transfer. Sony *could* change that, though, to 2X if you used 88.2K on something like a DIO2496. Or maybe even 4X with USB. Another point that would eliminate a potential advantage of MP3 is if there was software to automatically transfer the titles from the MP3 files as they were being recorded. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: DIO2448
I have had a DIO2448 for about 6 months. I'm writing this because DIO2448
information is currently on the Minidisc Community page. The DIO2448 has 6
connectors on the attached bracket: TOSlink in, TOSlink out, S/PDIF coax in,
S/PDIF coax out, and 2 RCA connectors for analog line level output.
The good features of the DIO2448 are:
1) High quality, low distortion, low noise analog output, much better than
many sound cards or motherboard audio.
2) TOSlink and coax digital inputs and outputs.
3) Native support for both 44.1KHz (MD) and 48KHz unlike some 48KHz only
outputs.
4) RCA analog output connectors; much more reliable than mini-phone jacks
due to the higher contact pressure.
5) Up to 24-bits supported. Machines with ATRAC 4.5 generate 20-bit audio,
ATRAC "R" generate 24-bit audio. But resolution greater that 16-bits is not
well supported by most PC software.
6) Low cost. I got mine for $120.
The problems with the DIO2448 are:
1) Digital output not accepted by Sony MDS-PC2.
2) No digital input switching. The TOSlink, coax and internal TTL header
are all ORed together, so you can only connect a signal to one at a time.
Routing features, may be useful, may be a problem, depeding on your
application:
1) Analog outputs can be switched to monitor the digital output or the
digital input.
2) Digital outputs can be switched to the DAC or to the digital inputs
("loop" mode).
3) No analog input, so you have to use the ADC in the minidisc.
Loop mode does *not* retime or modify the digital audio signal in any way.
Several people have suggested that I connect the S/PDIF header on my CDROM
to the header on the DIO2448 in an attempt to fix the "DIN unlock" problem
that occurs between tracks. Not only does the *not* change anything, but it
effectively disables use of the TOSlink or coax inputs!
I've been using the DIO2448 for input (from the MDS-PC2) only. I added a
TOSlink transmitter to my motherboard (16-bit only) and am using that to
transfer from the PC to the MD.
-
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MD: wish list for Sony et al
I currently have an Sony MDS-PC2 MD recorder and a Sony CRX-120E USB CD recorder. This wish list covers both MD and CDR, since one of the things I would like to see is better software integration. 1) For the MD: USB - USB - USB for both audio and control! This would solve a couple current problems: a) Sony MDS-PC2 does not accept S/PDIF from Midiman DIO2448. b) Eliminate the forest of cables between MDS-PC2 and computer. c) Eliminate the need for a sound card with digital I/O. d) Eliminate the CAV-50C and a need for an RS232 on the computer. 2) For the CDR: USB drivers that are compatible with other mass storage. It currently has *major* compatibility problems with my Iomega Zip 250 USB; I can't have the drivers for both installed simultaneously. Microsoft is the root of this problem, since they don't provide much support for USB mass storage under Windows 98, so each manufacturer hacked their own (incompatible) support into Windows. The goal of USB is to provide a positive experience for the user: easy installation, hot swapping, etc. This definitely fails miserably! 3) Integrated software that handles both MD and CDR. One could start by merging MD Editor 2 and CD Extreme into a single program. Some neat features: a) Easy drag and drop track-by-track copying from MD to CD and CD to MD, all in the digital domain, including titles! b) One button "copy entire disk"; CD to MD and MD to CD. c) Support for copying to/from MP3 and/or WAV (CD Extreme has this, MD Editor 2 does not). One thing that is absent from my wish list is MD Data support via USB. I wouldn't complain if they implemented it, but I don't really need it. I am very comfortable using CDR for data archiving. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: cheaper optical cables
I bought some Recoton TOSlink cables at Best Buy a few months ago. They are very flexable and work great. The only thing that might be a problem is that the TOSlink connectors are rather large; they just barely fit the side-by-side jacks on the DIO2448... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: monoaulizing algorithms...
I will try to check my MDS-PC2 for the 1-sample-out-of-phase error, providing I can figure out how to get Cool Edit to generate waveforms one sample out-of-phase. (I'll have to use a frequncy, say 15 KHz, sufficiently below the filter cutoff to be unaffected and rotate the two channels to some phase difference of less than 180 degrees that produces exactly 1 sample delay.) I will check using an optical connection from the computer. Doing this in the analog domain might pose some interesting problems. I don't know if I can get sample accurate phase from the the analog output of the computer. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: demor-- er, monauralizing algorithms
L+R/2 is almost universally used. FM broadcast is trasmitted this way for comptibility with monaural receivers. The "mono" switch on preamplifiers does this. However, this simple, almost universally used transformation does not provide a perfect mono version of a stereo source. The reason is that stereo channels add by *power* acoustically, whereas the electrical sum L+R/2 adds by *voltage*. Therefore, when recordings mixed with prominent material in the center of the sound image, such as vocals, bass, etc. are electrically mixed to mono (L+R/2), the sounds in the center of the stereo will be 3dB higher in the resultant mono mix compared with those at the left or right edge (known in the recording industry as the "3dB center buildup"). How to solve this? sqrt(L^2+R^2) is one method. This method would be particluarly good for a DSP implementation. Another method is to rotate the phase of the channels with an "all-pass" filter so that they are 90 degrees apart and add in quadrature. This method is better for an analog implementation. I'd love to see a deck that implemented the first solution! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Napster, RIAA
I understand that the RIAA has done some good in protecting artist's and recording company's rights to profit from their creations, but I really have gotten tired of hearing them complain about how much money they're losing. Did they ever stop to compute how much the music consumer has lost? When I replace an LP or cassette with a CD, do I get a special deal for the "upgrade". No such luck. They tax all MD blanks, assuming that I will use them *only* for copying copyrighted material, of which I do some, but I also make much use of my MD to make voice recordings of my relatives for a genealogical project. These are the ways that I have lost money to the RIAA. Although I've never had a reason to defeat SCMS yet, but I don't feel to bad about owning a CO3 (mainly for conversion from/to AES/EBU) or a digital sound card, knowing that I have the capability to circumvent their barrier if I even need to. I don't feel bad about owning a Spressa CDRW, that does not require audio CDRs to which the RIAA's tax has been added. (Most of the CD's I write are data, anyway.) To be honest, I've downloaded some great tracks from the "starving artists" section of MP3.com because they have a lot of stuff which is not available at record stores. Its somewhat of a production to download, decode and record to MD or write to CD. I've never downloaded from Napster, because the vast majority of what is available there is also available at local record stores (or CDNOW, for that matter). I find it easier and achieve better results (than with the highly variable encoding quality of many MP3s) if I just buy the CD and copy off the tracks I like. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: whether MD tracks are files
MD shares many of the characteristics of a computer file system. It has a freelist. It has a directory. One could even consider the track titles to be sort of a "file name" although it doesn't mind duplicate titles, since access is by track number. Data chunks can be stored out-of-order. MD allows good editing features. Contrast this with an audio CD. It's directory is somewhat primative. There is no freelist, hence the CDR mis-feature of adding/erasing only at the end. CDR has generally poor editing features. Tracks must be stored sequentially. Track titles are not covered by the basic standard, but are handled by the "CD text" extension, which most equipment and CDs don't support. MD is a lot closer to a traditional computer disk file system than CD. Cassette is easily a poorer file system than CD or MD. I can still remember the early days of microcomputers, loading programs from cassettes and/or paper tape. How far we've come! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Mini Disc to CD-R Transfer
I have the hardware one would need hooked to my computer (Sony MDS-PC2 with CAV50C, Maudio DIO 2448 (although 2496 is recomened) Sony Spressa CDX-120E USB CDRW). The problem is that no unified software exists. The MDS-PC2 is controlled by the"Minidisc Editor 2" program, while the Spressa CD recorder is controlled by the "CD Extreme" program. The user interface of each program is very different from the other and there is no easy way to get them to "talk" to each other. What I have done to date is to manually play each track from the MD using the "record" function of CoolEdit to create a WAV file for each track. Then drag the WAV files into CD Extreme. CD Extreme *does* allow you to title the tracks on the CD, but again, there's no easy way. I have to use the Windows clipboard to copy and paste each track title individually. (MD Editor 2 allows me to save the entire track list to a CSV file, but CD Extreme does not allow a "title" file to be imported.) As you can see, this whole process requires a *lot* of manual intervention. If Sony is reading this list, here is my wish list: 1) Better software integration. Merge CD Extreme and MD Editor 2 into a single program that controls both CDRW and MD. 2) Better MD - computer hardware interface. Right now I have a sound card with digital I/O and the CAV50C (supplied with the MDS-PC2) which uses up a serial port on the PC. Not to mention all the cables between the 'PC2 and the PC. All of this could be replaced by a USB jack on the MD, just like the one on the Spressa CD recorder! I know it would be a *big* job, but if someone has the time and would like to write better PC software, that would solve much of the problem. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Revolutionary?
Oh no, I _like_ the clamshell mechanism! I have an 'R70 (clamshell) and an 'EP11 (slot-in). The slot-in mechanism feels a *lot* more solid. I always feel as if I have to be super-gentle with the R70. I wish Sony was offering portable recorders with slot-in. Especially recorders, because the magnet head assembly makes them even more fragile than play-only... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Is 128kbps MP3 bad? MD has loadsa life yet!
I've found that mp3s of songs with vocals sometimes do not transfer over to MiniDisc well (analog), especially 128k files ... at 128, some of the higher stuff is muffled, or "combed" or really sharp. I've noticed some minor problems with 128k MP3, sometimes sort of a "hard" sounding high end, but 99% of the problems with MP3s are with the source material (azimuth error on analog source tape, encoding from cheap computer sound card, etc). Some of the stuff I've picked up from MP3.com is fairly well encoded and very enjoyable to listen to. Mind you, *no* 128k file has ever sounded as good as a well made CD or even an MD, if care is taken when encoding, they can sound quite good. For transfer to MD, I've used an optical connection from the computer. I use CoolEdit2000 (Fraunhofer licensee) to decode the MP3 and play it back. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: mini vs mp3 recorder
I think that solid state memory media *is* the wave of the future. However, I do see some drawbacks: 1) Right now, the cost of solid state is very high compared to MD: $100 vs $2.50 for the same amount of storage. 2) Right now standalone MD recorders are a mature technology, which means they're very available at a resonable price. Standalone MP3 recorders (as opposed to *players*), are not very common. 3) MD media is a much better choice for archival recording that FLASH memory. FLASH uses floating gate technology. Eventually the charge will leak off of the floating gates, causing it to "forget", whereas the recording on MD is more permanent (unless the disk is heated to "record" temperature). - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony MD and CDRW
I just purchased a Sony CRX100E/X2 CD recorder, which has a USB interface. I also have a MDS-PC2 MD drive connected to my computer. So far my wish list includes two general items: 1) The CRX100E/X2's USB hardware interface is just great! I'd like to see something like that when Sony gets around to making an MDS-PC3. It's much less complicated and less expensive than the cumbersome hardware interface on the MDS-PC2. 2) I'd like to see some integration of the software. Right now there are two separate programs: "CD Extreme" for the CDRW and "MD Editor 2" for the MD. I'd like to see a consistant, single program that could operate both types of device. I'd be interested in any opinions concerning the future of MD and CD in multimedia computing. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MS stereo microphone
Anyone have any experience with Sony's ECM-MS907? It's a very reasonalbly priced mid-side microphone. I'd be interested to know if they use a specially manufactured figure-of-8 electret capsule to generate the "S" signal; I'm not aware of general availability figure-of-8 electret capsules... Any of the microphone gurus that read md-l know where to get figure-of-8 electret capsules? http://www.sel.sony.com/SEL/consumer/ss5/home/accessories/microphonesfordigi tal/ecm-ms907_specs.shtml - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony MD changer in car on non Sony head unit
there is an undocumented +12 volt trigger output on the RM-X69RF controller I am using this wire successfully in my installation. I have a classic 1965 Ford Fairlane in which has modern electronics in the original radio case. Inside the radio, I have a 12VDC DPDT relay controlled by the RMX69RF's "trigger" output that switches the audio. Works great! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: amps with digital in/out
Thought I would add this to the discussion: The amps with digital I/O that are available now are analog amplifiers with built-in D/A convertors. Some of the high-end models use DSP for level, tone/equalization, reverb, etc, but all eventually convert the signal to analog and pass it through an analog power amp. However, there are several companies that are close to changing that! They use a DSP to convert the incoming signal to bi-level or tri-level pulse-width modulated. The PWM signal switches high voltage (100V), high current MOSFETs. The output of the MOSFETs goes through a passive low-pass filter and then directly to the speakers. A neat approach, because the signal remains in the digital domain until the last moment... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
re: MD: compression (correcting a couple errors)
1. On the VHS SP vs EP discussion: EP *always* results in inferior quality compared to SP because the helical tracks *overlap* in EP mode (they do not in SP), causing a reduced S/N ratio. 2. On the cellular discussion: Although normal cellular uses analog FM audio transmission (non-compressed), the carrier almost always converts it to digital and use heavy compression between the tower site and the switch. I think that a formal compression system implies the capability of expansion. Sampling at a lower rate throws away information with no hope of recovery... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: CD/MP3/MD story
It sounds like a radio and nothing like my cd's" this coming from a 15 = year old female who IMO is not an audiophile at all As one who spent many years as an engineer at an US FM radio station, I can say that radio stations work hard to sound bad! FM radio inherently sounds very much like a CD, but the typical station feeds the audio through several thousand dollars worth of processing equipment to screw it up before it gets to the transmitter. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: analog quality (was Sony Minidisk Recorder with USB)
I find that, even when transfering 128K MP3 to MD, it is best to use digital I/O so further degradation does not occur. Analog sound cards often have a much poorer S/N ratio than the original MP3 file, and they often have frequency response or phase shift errors at low frequencies due to undersized coupling capacitors. The major analog problems are almost never in the codec chips, but rather are in the supporting circuitry (power supply, coupling capacitors, circuit topology and ground layout). Since MP3 intrinsically introduces a different set of problems, the analog problems get added to the already far-from-perfect MP3 source... I haven't studied the MP3 specs enough to know if 16 vs 20 vs 24 bit digital I/O makes a difference. But I do know that, given the analog quality of typical sound cards, digital I/O is definitly better transfering MP3 to MD. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Salvaging a recording made with levels set too high?
The result you achieve will depend on how much effort you put into it. If its a *really* important recording, you can use Cool Edit (or your favorite audio editor program) to painstakingly find every annoying pop produced by the clipping and delete the offending samples. You have to zoom *way* in, because often you need to delete only a couple samples at each point, but you have to do this literally hundreds of times. I spent about a week worth of free time once restoring a 12 minute song that had quite a bit of clipping because it was an unbelievably great live performance. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Mains cable as Speaker cable
The largest improvement, especially at low frequencies, come from low resistance. Since the speaker itself is only a few ohms, series resistance added by small gauge cable and normal connectors will quickly destroy the high damping factor of modern solid-state amplifiers. The PA rig I built several years ago used 12 AWG rubber covered mains cable with 277V twist-lock power connectors. (I chose 277V since that power voltage is used only for industrial lighting and is never present elsewhere so the speakers would not get plugged into the power mains by mistake, although the woofers would have withstood this, very loudly! :-) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: dataplay
It's a neat idea. The small size would be attractive for integration with palm computing. But, if the Wired article is right, Dataplay is write-once. Sounds like a problem to me. One of the things I like about minidisc is the ability to re-write if I make a mistake. Although it sounds like Dataplay has addressed most all of the other problems of CDR, its apparent lack of rewritability is a major flaw, so far as I can see. The only flaw of expanding minidisc into other applications, for instance palm computing, is that its size is almost to large. The MDs size is probably due to the fact that in the early 90s when the format was designed, the present size is about as good as they could do while keeping the cost reasonable. With some effort though, it might be possible to cram a slot-in MD recorder into a Palm III sized palmtop computer... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MP3 to Optical Out
I am using a DIO2448 (well, sort of) optically connected to a Sony MDS-PC2. If you have a motherboard sound system, you will have to use the control panel to select the DIO2448 as the wave playback device. After that, the problem I experienced is that the MDS-PC2 does not lock properly to DIO2448's signal. According to M Audio tech support, there is some quirk with Sony's S/PDIF implementation... However, the DIO2448 *does* accept the S/PDIF *from* the MDS-PC2. Therefore, I am only able to use the DIO2448 for transfering from MD to HD. I added a TOSlink transmitter to my motherboard's codec for transfering from HD to MD. When you get things working properly, the MD will be able to record everything that comes out of the WAV audio. I usually use Cool Edit for playback from the HD, but have even recorded to MD successfully from Real Player. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: coax vs toslink...
I think the purists prefer coax to toslink because the slight timing ambiguity instroduced by the multi-mode fiber can't pass their pico-second jitter specs. However, jitter is only important when driving a DAC, and every "good" DAC has a sophisticated timing recovery circuit in the receiver. If everything works as it should there is NO difference in the signal delivered to the DAC. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MZ-R70
I just received an MZ-R70 from Minidisco. The plan is to use this portable recorder to record interviews with family members for a genealogy project. The unit is very small, compared to my MZ-EP11, and I find that its clamshell mechanism feels a bit more delicate than the slot-in mechanism of the MZ-EP11. As was noted, it makes a quite audible noise at intervals, which would prohibit using a "T" mic mounted on the jack. I'm planning on using a couple "tic-clip" mics with long cables, so this shouldn't be a big problem. (The MZ-EP11 is comparatively rather quiet.) The MZ-R70's front-panel buttons are much easier to operate than those on the MZ-EP11. Interesting to note: the MZ-R70 has 2 headphone jacks, one next to the line-in and mic jacks and another as part of connector for the remote. The jacks are color coded: black for headphones, white for line/digital in and red for mic in. One nice touch on the MZ-R70 LCD display is a 10-segment level meter (with 1-second peak hold) which operates during both record and playback. As usual with portable Sony recorders, the MZ-R70 requires one to press the annoying "END SEARCH" button when recording. All told, this recorder looks like an good deal for a rather low price. Now, does anyone know a good source of Mogami 2697 cable to make microphone cords? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: portable recorder recomendation?
Hi, I'm looking for a portable MD recorder for my sister to use for recording interviews with my relatives for a family genealogy project. She's not gadget-phobic, but I'd like to keep operation as simple as possible so she can put her thought into the interview, not the recorder. I think the following features would be nice: 1) NO END SEARCH. Since I'll be "cleaning up" the raw recordings she makes, I'd rather that she doesn't record over anything. This probably rules out Sony... 2) Manual record level from the microphones, which I'll set before send the recorder to her. MD has *plenty* of dynamic range, so I can get away with setting it plenty low to avoid overload. 3) Long record time. Don't know whether she'll prefer alkalines or recharagables, but this is much more important than small size, so a unit that has multiple cells or even an external pack (if the whole thing could fit into some sort of bag, if it could be operated without removing it from the bag) would be OK. My plan is to use 2 clip-on mics, one for her and one for the interviewee. I would set up the recorder to record her on one channel and the other person on the other. That way I'd have a good opportunity to balance things. I'd do the clean-up with Cool Edit 2000, then transfer the resulting mono back to MD. Any recomendations? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
RE: MD: CD-rom to soundcard?
do you have to digitally connect your cd-rom drive to your soundcard to make use of the digital out on the soundcard? Here's my experience with a Samsung 40X CDROM and an M Audio DIO2448. The Samsung does have an S/PDIF (TTL level) header, but the problem is that it does not produce a valid signal unless the drive is playing an audio CD. This causes glitches when the drive starts playing or seeks to a new track. The sound card *does not* remove these glitches. The only way to dub from the CDROM via S/PDIF to the MD is to play the last few seconds of the previous track and start recording at the beginning of the desired track. There is reportadly a real-time CD ripping plug-in for winamp. CD ripping is *not* a trivial task, since the CD data format and CDROM drives were not designed to do it reliably. The performance you achieve will be highly dependent on the drive model and CPU speed. A standard CD audio drive with digital output is still by far the best way... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDX-65RF
Hey, the display looks exactly like the one that comes with the RMX-69RF! I've got 3 regular MDX-65s in the trunk (with a XA-C30 switcher) that are controlled by an RMX-69RF. With this arrangement, I could expand up to 7 changers (any combination of MD or CD) by cascading up to 2 more XA-C30s. What I'd really like to add, though, is a Unilink digital radio to receive the satellite broadcasts that will begin in North America this year. Don't know if they planned this far ahead when they made the RXX-69RF... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Why USB?
I'm glad the MD - USB thread has proven to be a lively topic of discussion! You'll get no argument from me that USB is not technically the "best" interface. However, it might turn out to be the very popular for these reasons: 1) USB is available on the motherboard of almost every new computer these days and there are a wide variety of USB peripherals. It is multi-drop (unlike serial RS232) and *designed* from the start to be two-way (unlike the printer port). It is cheap (unlike SCSI). 2) There are many advantages over current methods of digital transfer (TOSlink/SPDIF/ControlA1). Although TOSlink/SPDIF can theoretically hand high speed transfer, there is no readily available hardware for faster than 96K (2x). TOSlink/SPDIF is uni-directional, so you need 2 cables to go both directions. TOSlink/SPDIF has no "flow control" or retransmission; one has to be careful of buffer overruns/underruns with Windows. TOSlink/SPDIF does not handle deck control; you must use a separate ControlA1 (or other more primative) interface. Beside handling all these functions in one cable, USB would also supply the modest power requirements of a portable MD recorder! So it would essentially replace 4 cables to the MD with one cable, certainly an important point when considering the compact size of a portable MD recorder. 3) It would be a simple, incremental improvement from the manufacturing point-of-view. This is important, because no manufacturer wants to incure the possibility of out-of-control costs associated with a more open-ended design project. Adding USB capability to a portable MD recorder would be a fairly "contained" design effort, constrained to the single chip that implemented the USB interface. No transport modifications would be required to implement high-speed transfer, since the MD transport in most machines can already run at 5X speed. I know USB is not the ultimate in terms of performance, but it could be the most cost-effective. Judging from what I've seen on MD-L, cost is a *big* incentive. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: JB940
The 5 hour mode that everyone has generated such interest is indeed a neat feature, but the feature that is most inseresting to me is the ability to change the level of already-recorded tracks. I'm guessing that this feature uses an entry in the UTOC for each track that specifies how much the to shift the level *during playback* of that track, without actually changing the audio on the disc. This idea was mentioned once a while back by someone on the md-l. Is anyone still interested in this level-shift feature? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony JB940 info...
http://www.video-direct.com/news/preview3-7/mdsjb940.html Sony MegaStorage® CD Changer Control CD Text and Custom FileT Transfer from CD Via Control A-1 II Coaxial and Optical Digital Inputs and Outputs Long Play Mode - Captures up to 5 hours, 20 minutes of stereo music on a single 80-minute MiniDisc Scale Factor Edit - Enables you to adjust the volume of MiniDisc tracks even after they're recorded. Equalizes tracks recorded from different sources Disc and Track Titling Through PC Keyboard Playback Pitch Control ATRAC Type "R" I wonder how they implemented "long play mode", by using a low sample rate? I also wonder about scale factor edit. Does it actually rewrite the entire audio track or does it make a change in the UTOC which then gets applied to the track during playback. I'll be on the edge of my seat awaiting the answers! Hope someone gets one soon so they can give us a report... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Sony USB 2nd try?
Here's an interesting thought. If someone *does* buy the Sony USB adapter, could they pry it open and let me know the numbers on the chips. Might be that the codec chip has an S/PDIF output pin (fairly likely with modern codecs). Its less likely to have and S/PDIF input pin. If Sony has used their own chips, documentation might be hard to come by. But they used a NEC microprocessor in the CAV-50C interface, so maybe their latest trend is to move away from designing their own chips for everything. One other comment: S/PDIF I/O is, in general, less of a design hassle than analog I/O. The only reason it is currently more expensive is economy-of-scale. If manufacturers roll out S/PDIF sound cards in the quantity they do with analog sound cards, the price will go *way* down, becuase S/PDIF is actually *simpler* from the manufacturing standpoint. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Sony's new Internet Audio Recording Interface
Is S/PDIF hard to generate? Computer audio chips tend to send/receive it right with pins on the main sound chip and CD/MD equipment also tends to have it integrated at the chip level (this is the most economical way to do it). So my guess is that it's no big deal, if Sony has designed a special chip. Alternatively, Crystal Semiconductor makes generic S/PDIF transmitter receiver chips. Make it run at a special 4X or 8x S/PDIF rate, so that a portable MD player could compete in downloading convenience with an MP3 player. 4X is no big deal if Sony designs their own chips (both in the interface and the MD). So far as I know, the highest speed that the Crystal chips run at is 96K (there's not a big market for higher speed yet), which would at least handle the 88.2K rate for 2X transfer. I think the fastest USB transfer rate would limit things to 4X or so... I'm hoping that the USB - MD interface is so successful that Sony starts putting it into new recorders at the chip level. That would allow them to produce USB-enabled MD recorders for little more than the price of the USB connector! If they did it at the chip level, maybe they could (in the future) revive MD data, so you could make the MD recorder do double duty. (Although MD data disks are expensive right now, its only because the market is extremely small compared to the audio market. If there was a large base of MD data equipment; they price would come down. As a matter of fact, since there's no "SCMS tax" on data disks, it could go even lower!) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Ken Pohlmann Principles of Digital Audio 4th ed...
"Priciples of Digital Audio - 4th", considered by many to be the "bible" of digital audio theory, is now available. My copy, which was pre-ordered at BN.com, arrived yesterday. The latest edition is up to 736 pages, and contains an entire chapter on MD, as well as chapters for CD, DAT, DVD, PC audio and digital broadcasting. I've not had a chance for detailed reading yet, but it seems to bring together a *lot* of basic theory together in one place. Maybe not enough to actually build the devices it talks about, but certainly plenty of information to conduct an intellegent conversation with those who do build the stuff. All in all, a great reference book! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Micro keyboard for titling
There is your answer ... MD connected to a Palm Pilot I have a Palm III and various MD equipment. At this point, the *only* common connection is IR, specifically, the Palm with send/receive IR and my JE520 and MDS-PC2 will receive it. The Palm will send/receive RS232 and, with the addition of the CAV-50C RS232/Slink interface, so will the MDS-PC2. But I think it would be better to concentrate on the portables, rather than these home-type decks. I think the best solution for *existing* portable MD would be a cable that connected the RS232 hotsync on the Palm to the 4-pin remote connector on the MD. Two of the 4 pins are control, which is normally implemented by a resistor ladder and pushbuttons, but I've seen PC interfaces that use CMOS analog switches. The other two are data from the MD to the LCD (or the label printer). Looks like an opportunity to use a PIC single-chip microprocessor to make the connection. I'm sure that it would be possible to write a Palm program similar to the PC program Sony supplies with the MDs-PC2. Might make me think of replacing my MZ-EP11 player with a portable recorder! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: crossfade between tracks (was: My Diamond Rio Experience)
could shock memory be replaced with the ability to eliminate silence and have a good transition between songs? Like those CD changers that take 5 to 10 seconds to change disks, can you set them to record some in memory so by the time the disk has to change, you could have gapless playing? Sony megadisk CD changers will do crossfade transitions if you have two changers. The Sony MDX65 car changer does disc change without interruption (but not crossfade) using the shock memory. I'd like to see a crossfade feature that works regardless of whether the tracks are on the same disc, different disc in the same changer or different disc in a different changer. Given a sufficient amount of RAM, it doesn't look like it would be a big deal... In an ideal implementation, it would be user adjustable, say from a uniform 5 second gap between tracks to a 5 second overlap. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MD USB summary...
First, a quick thanks to all who have taken part so far in this thread. Much of the discussion has centered aroung the obvious fact that an MD with a USB port starts to look similar to an MP3 player to the user. This is indeed true, but my original reason for proposing USB for MD was to eliminate the many problems of S/PDIF / TOSlink / Control-A1. So far as I know, the software for making automatic MD compilations from CD/WAV/MP3/RA is limited to non-existant. I feel this is mostly because there are too many competing "standards" for deck control and, to a lesser extent, because S/PDIF is somewhat problematic due to poor adherence to the EIAJ standards and due to its lack of flow control. Already, I use RealAudio and MP3 as a substitute for the radio. This gives me access to a much wider variety of programming than is available over-the-air. It would be great to have software that allowed me to do unattended "timer" recordings of streaming audio feeds. This would allow me to listen to some of what's available on the Internet when I'm away from the computer, for example driving in the car. It's my feeling that USB on MD is a small step in the direction of unification and standardization that would advance the development of hardware and software which would allow people to use MD in ways that we haven't even though of yet. A totally different solution on the horizon at this point is dedicated Internet audio appliances (http://www.kerbango.com http://www.penguinradio.com), but we'll definitely be at the vendor's mercy when it comes support (or lack thereof) for interfacing with MD. In summary, I believe that USB on MD would be an simple, inexpensive and reliable method of connecting an MD to a PC that would greatly expand the usefulness of the MD audio medium. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Stop press: The Cassette is back!
Wow! A $200 cassette! Do you want fries with that? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: CDROM digital output
I've received more than one suggestion that I run the CDROM's digital output through my sound card and somehow the digital output from the sound card will be valid when the CDROM is not playing. I have a DIO2448, which does have a header for the CDROM's digital signal. Running it though the sound card makes no difference. To get the CDROMs signal to the TOSlink output on the card, the card must be in "pass through" mode, where it is routing one of its digital inputs to the digital outputs. In this mode, the output is not valid unless the input is valid. The basic problem that seems to be confusing everyone is that the S/PDIF signal is a *synchronous* signal where the clock and data are encoded onto the same signal. (The technique they use is called Manchester encoding.) The clock must be extracted at the receiving device using a phase-locked-loop. This has two problems: 1) The PLL has a finite lock time, so you end up missing the first several samples before lock occurs. Remember, this is *synchrounous*, unlike normal RS232, which is asynchronous acquires sync on every byte. 2) The extracted clock may be unstable, depending on the data. Certain data patterns are worse than others. Stability is improved by increasing the lock time. Good receivers use a two speed lock algorithm: high speed when acquiring lock and slow speed to improve stability, once lock is acquired. There is no magic that the soundcard can do to acquire lock that the MD cannot do. The damage caused by interrupting the S/PDIF stream is irreversable. The bottom line is: it takes a finite amount of time to re-acquire lock, regardless of whether its a sound card, an MD, a DAC or whatever... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: USB - MD
I just thought of one more advantage if the MD had USB from transfering digital audio the PC: flow control. The data flow on the USB can be throttled by the sender or the receiver. S/PDIF, OTOH, *must* continue uninterrupted. If you get a buffer overrun/underrun, too bad, it'll put a skip or a dropout in the audio... For S/PDIF to work, you have to have plenty of excess CPU speed. I also find it helps to close all unnecessary programs. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MP3 - MD
With my new computer and MDS-PC2, I've been making compilations of MP3s on MD for the past couple weeks. While the "playlist" method using winamp or Real jukebox will produce satifactory results, I've been using Cool Edit 2000 to play back songs one-at-a-time. This has allowed me to catch and correct problems with the MP3 files, generally level-related, although the occasional file has had a DC offset problem. One problem I have noted with Cool Edit is that it "remembers" the value of the last sample that was played, so if you play a bit of the file as a test, then stop playback, it will produce a pop when it is restarted at the beginning unless silence is played first. I've been using a TOSlink connection from the PC to the MD, with excellent results. My impression is that, while MP3 encoding produces more audible artifacts than ATRAC, one can achieve very satisfactory results from well encoded MP3 (128Kb) files. The most frequent problem with MP3 files seems to be clipping from excessive "overs" resulting from too-high levels. Second on the problem list is aliasing, probably caused by doing A/D conversion on a computer sound card with a poor anti-alias filter. I've heard some really good sounding (and also some really bad sounding) MP3 files, so my guess is that a lot of the flaws often attributed to poor MP3 sound is really the result of poor encoding practices. In summary, I feel that the MP3/MD marriage will bring me a lot of musical enjoyment by allowing me to hear a lot of great music that I would otherwise not be exposed to. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDS-PC2 bugs
Here's a problem I've noticed periodically with the MD Editor 2 program that is bundled with the MDS-PC2. Sometimes when adding tracks to an MD that already has a few tracks, some of the titles on the pre-existing tracks get screwed up *on the MD edit window* but not on the UTOC. I think this is because only writes new titles to the MDS-PC2s RAM when you make changes. So ejecting and re-inserting the MD fixes the display. Boy I caused my heart to skip a beat the first time! But I've learned that it can safely be ignored... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony MD-Cam - 650MB - 4.5 hr audio!
Video Direct is now selling the Sony MD Camera with the new 650MB MD drive. Among other things, it will record 4.5 hours of audio! http://www.video-direct.com/sony/mddiscam/mddiscam.html - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MD - PC connection
In my previous post (about the Sony commercial), I made the observation that transfering between PC and MD is extremely problematic at this point (1X transfer via S/PDIF or analog, titling via Slink, IR, etc). This response seemed somewhat interesting: better use a USB or FireWire Adding a USB connector to an MD deck is a great idea, especially one such as the MDS-PC2, which is made *specifically* for connection to a PC. I believe the high-speed mode on USB is 12Mbit/s, which is enough to transfer audio at a little better than 4X! Given that the most MD current transport mechanisms are designed to run at about 5X when filling/emptying the buffer, this looks like an easy change for the manufacturers to implement by simply designing a data coder/decoder chip with USB pins on it. Basicallly, the rest of the deck design would remain unchanged. 2X and 4X CD - MD decks (that use standard MD transport mechanisms) are already a fairly common... The USB connector is physically small, so it would work great on a portable MD recorder and given the modest power requirement of modern MD portables, the USB could probably power the MD as well as transfering audio, titling and remote control. Wow, this would certainly cut down on the forest of wires I've got with my MDS-PC2! FireWire is a possiblility, but its not near commonplace as the now ubiquitous USB connectors on new computers, and while its extremely high transfer rate would be nice, it would require extensive design changes, such as higher-speed MD transports, etc, to exploit its high speed. Maybe down the road... Man, I'm excited just thinking about the implications of USB! Now this would make MD a *clear* winner over the MP3 players! Sony, let me know if this type of thing is in the works; I'm ready to buy it now... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony MD commercial
I've seen the Sony MD commercial on MTV serveral time now selling MD as a way to record music from the Internet. While I think it's great that we're finally seeing MD ads, I wonder if this is really the right approach. I am a big supporter of MD, specifically Sony (I have a JE520, an MZ-EP11, an MDS-PC2 and 3 MDX-65s), but it seems to me that its *more* difficult to record MP3s to MD than it is to, for example, a Rio. It takes longer, since the MD has to be recorded at 1X; the titles don't transfer automatically, etc. I'm afraid these commercials might backfire if MD newbies are disappointed when they find out that MD is actually not the best format for this specific application. Sony *could* have addressed some of these issues: 1) There are digital soundcards that will support 88.2KHz, so 2X transfers would be possible today if it was supported by the firmware in the MD recorder; the hardware already mostly supports it! Not much PC support for higher than 96KHz yet, but that may change with DVD-computer integration; I think AC-3 uses 192 or 384K, so theoretically 4X and 8X would be possible as newer soundcards become available. 2) Some PC software to make things easier. Drag-and-drop hard disk to MD. Maybe somehow integrate a CD ripper into it, so that you could just do drag-and-drop beween the CDROM, MD, or HD. 3) Right now, the digital audio goes via TOSlink and the control goes via control-A1 (which portable recorders don't have). Maybe they could figure out a clever way of embedding deck control into the user bits of the TOSlink. That would certainly make it a *lot* simpler to set up... Any thoughts about this? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Digital out ...
Adding digital out *may* be as simple as adding a TOSlink transmitter *if* the data decoder chip has an S/PDIF output pin. Most modern ones do. If not, it will involve adding a digital audio transmitter chip, like the CS8401, to take the serial data/bit clock/frame sync signals from the output of the data decoder and encode them into S/PDIF. I've done it both ways in home CD players, where there's tons of room to add the extra parts, but I wouldn't want to try it in the cramped quarters inside a portable MD unit... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: S/PDIF channel status user bits
Just wondering if anyone knows of a program to display the contents of the channel status and user bits from an S/PDIF data stream? I've got an M Audio DIO2448 soundcard (CMI8738 chip). One alternative I may investigate is connecting the "C" (channel status) and "U" (user bits) pins from a CS8414 receiver chip to the 'scope to get a rough idea of what's there. Also, does anyone have access to the EIAJ standards that describe S/PDIF? Some information about the channel status bits is available in Crystal Semiconductor's now (in)famous AN22... Does anyone know how much of this is implemented by average consumer equipment? Are there any extensions? Is there any infomation about the user bits? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: SLINK for interfacing with MD
Sony makes an RS232 - control-A1 convertor, the CAV-50C. Its a very small box with an attached 1 foot cable terminated in a DE-9F, which plugs into the serial port. It requires no external power; it is powered by DTR from the serial port. It comes with the MDS-PC2. Maybe you could order it separately directly from Sony. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDS-PC2 review (long)
This review is in two parts: (1) the MDS-PC2 deck and (2) the PC Interface. Physically, the front panel of the MDS-PC2 is slightly larger than a half-height 5.25 floppy drive. Along the left side is (from top to bottom) the Power button (standby/operate), the IR sensor for the remote, a mini-phone jack for headphones and the Input selector (analog/optical 1/optical2). Buttons on the right side are (top to bottom) are Eject, Play/Pause, Back, Forward, Stop and Record. The front panel Back and Forward buttons operate like those on a portable. That is, pressing quickly causes it to skip to the next/previous track (the | and | functions), while holding the button down gives foward and rewind (the and funtions). However, the buttons on the supplied remote are like a normal home deck. That is, there are four separate buttons: | |. Also on the front panel is a 1-line flourecent display, similar to the displays on normal Sony home MD decks. The case is surprisingly deep. On the rear panel is a non-detachable mains cord, two mini phone jacks wired in parallel for the Control A1, a TOSlink optical output, 2 TOSlink optical inputs and 4 RCA jacks for the analog line inputs and line outputs. The deck itself is electrically very similar to the JE520 in my stereo system. It uses ATRAC 4.5 and produces 20-bit digital output. The front panel controls are very limited (there's no menu button, for instance), but all of the controls that I have come to expect on a home minidisc deck are available on the supplied IR remote. Operating the MDS-PC2 from the remote is almost identical to operating the JE520. Most features I tried operated flawlessly, however the A-B erase mode (which I use to trim off the excess recording at the start of the song) seemed to crash when I tried to move the "B" edit point forward a frame with the | button. While not essential for playing discs, the remote is needed when recording if it is necessary to access the level meters (with the display button on the remote) or adjust the record level. The computer interface is a small box labeled "Vision Touch CAV-50C". It has an attached 1 foot cable with DE-9F for connection to the computer's serial port and a single mini-phone jack for connection to the Control A1 cable from the deck. It contains an NEC 78082GA26 single chip CPU, a Maxim 3221 RS232 interface and various other parts. It requires no external power; it is powered by the DTR line from the computer. The interface only works with a normal PC-type serial interface; it will not work with high-speed PCI serial cards. The installation instructions suggest using analog audio connections to the computer. At first I thought this strange, but given the experiences I've had the past few weeks, I can see that they do this because they can be sure it will work. I have been unsuccessful with connecting the S/PDIF output from my Samsung SC140 CDROM because it, like most other CDROM drives, only outputs valid S/PDIF data when playing, not during stop, pause or seek. This is unusable with Sony's drag-and-drop software. The digital inputs on the MDS-PC2 seem also seem to have a problem accepting 16-bit S/PDIF from my M Audio DIO2448 sound card, although the DIO2448's S/PDIF input does accept the MDS-PC2's output (at least 16-bits of it; Cool Edit 2000 is prone to skipping (probably needs something faster than my 433 Celeron) if I use it's 32-bit mode, which is necessary to completely capture to 20-bit output of the MDS-PC2). The MDS-PC2 *does* accept the 16-bit S/PDIF from the CS4299 codec on the computer's motherboard. Sound confusing? That's probably why Sony recommends analog! The PC software comes on a CDROM along with a label making program and a user operation manual in PDF format (the only written instructions that come with the software are the installation instructions). Installation is very simple, just insert the disk and click on the install buttons. Operation is fairly intuitive. Upon starting the MD program, you are presented with a "remote control" bar that has title readout, running time, and play, stop, next track previous track buttons. There's also a unit select drop down box on the left that allows you to select the device you're controlling: the MD deck, the CDROM drive (PC CD) or an external CD player (via Control A1). For more complex functions, there's a drop-down button on the right that brings up an "edit" menu bar with, from which you can select the MD edit window, an edit window for an external CD player, an edit window for the CDROM player, an edit window that has the external CD and the MD for drag-and-drop copying from CD to MD and an edit window with the CDROM and MD for drag-and-drop. Some may find mildly annoying the fact that *all* windows in the program are fixed size and cannot be resized, but usually it's not a problem. Hands down, the best feature of the MD edit window is the track titling feature. When a minidisc is inserted,
MD: CDROM - stable S/PDIF
A question and an observation... Can anyone recomend (or point me to) which CDROMs have an S/PDIF output that is stable during pause, seek, etc. I think that stable S/PDIF is necessary regardless of whether the CDROM is connected to an MD recorder or a sound card. The S/PDIF receiver chip (the most popular are Crystal CS8412 CS8414) has a PLL to extract the clock signal, which requires a finite amount of lock time. It *does not* lock on the first frame! That said, I have to give poor marks to the S/PDIF inputs on the Sony MDS-PC2. They are unable to lock to the S/PDIF from an M Audio DIO2448 and they MDS-PC2 requires longer than it should to lock to to CDROM's S/PDIF after it glitches. This may have to do with the fact that it tries to lay down (multiple) track marks when the S/PDIF lock is shaky. (M Audio tech support did finally get back to me, stating that Sony products were known to have problems sometimes with accepting S/PDIF from non-Sony.) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Samsung CDROM to Sony MDS-PC2 problems...
Well, I took time this evening to connect a TOSlink transmitter to the S/PDIF output of the Samsung CDROM drive so I could try the drag-and-drop CD to MD feature in the Sony MDS-PC2's PC software. The problem I encountered is that the Samsung CDROM does not output S/PDIF unless the drive is playing. During stop and pause on the CDROM drive, the MDS-PC2 gives "DIN unlock", and it glitches briefly when seeking from track to track. This behaviour of the Samsung drive (not continuing to output valid S/PDIF at all times) makes the S/PDIF output somewhat useless, because no matter what it is connected to, some time will be required to acquire lock, thus you'll always miss the beginning of playback. I have heard some mention that this is a common problem with the S/PDIF output of CDROM drives. One way that the drag-and drop software could partially get around this problem is to play the last few seconds of the preceeding track after each seek (with the MD in record-pause) to allow the MD recorder a chance to lock up. Then it would start the MD recording at the instant playback reached the desired track. This would work for eveything but the first track. Right now, I see no way to cleanly get the beginning of the first track. Doing this would not, so far as I can see, create any new problems. (Sony, are you listenning?) I should note that the Sony setup instructions intend for you to do the dub using *analog*, possibly for this very reason. I don't think I'd want to do the dub through a typical computer sound system, the way they suggest, because of the big-time audio degradation (compared to a digital dub). Does anyone here have experience with using the digital audio output of a CDROM drive? If so, what have you found? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: CDROM to MDS-PC2 (pre-roll)
I've been thinking a bit more about the idea of having the CDROM drive play the last few seconds of the preceding track to allow the MD recorder a chance to lock up after a seek. The industry (especially video) name for this is "pre-roll". Professional video tape editing equipment *always* prerolls the tapes, so that they are running stable, at speed, locked to each other, then it switches to record mode at the edit point. I used to have a pair of JVC home VCRs that did this to produce clean, frame accurate edits. (In the case of the MD recorder, it could be placed into record-pause during pre-roll, since it will start recording instantly due to its RAM buffer.) Sony could make pre-roll an option (or even an adjustable parameter) in the PC software. This would certainly fix the problems caused by the S/PDIF drop-outs during seek, except for the track 1 or the CD. I know that the Red Book standard mandates that the first 2 seconds of the CD are reserved for the TOC and such, so in fact, it would be possible to pre-roll track 1 *if* the CDROM drive would allow it. The non-audio TOC data would make a really loud noise, but the MD wouldn't start recording until the start of the first track. My personal preference, even if they couldn't figure out a way to pre-roll track 1, would be to pre-roll after seeks to all other tracks. The way it is right now, it glitches after a seek regardless of the track number. This way, it would at least work properly for all tracks except track 1. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: bit-for-bit copy via S/PDIF (was Linux S/PDIF soundcard)
This is a more complex than it sounds... There are a couple factors that can affect the bit accuracy when going, for example, from a WAV file to MD via (or vice vesa) via SPDIF. 1) Dropped samples due to system activity. My experience is that a bunch get dropped at the same time, causing a "skip". I usually try to close all programs other than the record/playback program. 2) Hardware/software resolution. From what I see on the 'scope, the MDS-PC2 always puts out 20 bit resolution. The motherboard S/PDIF always outputs 16 bit. The DIO 2448 can output/input16 or 24 bits. Most all software will do 16 bits. Cool Edit 2000 has a 32 bit mode that can be used with the DIO to get 24 bit resolution, but it's fairly processor intensive and prone to skipping. Maybe this will change when my memory upgrade comes... 3) Digital Signal Processing in the soundcard. I know that this is present in high-end cards like the ZA2 (which disables it unless you ask for it). Don't know whether the 3D processing some cards do is applied to the digital bitstream or the analog. My guess is that it only affects the analog, because it would require a lot of DSP horsepower. Of all these, I would say that (2) is hardest to detect, unless of course, you monitor the bitsream with a 'scope. There are also a *lot* of other factors that affect the sound quality, if you do *any* manipulation (editing) while the sound file is on the computer. For further reading, I direct you to http://www.digido.com (which I'm not in any way affiliated with; I just happen to think he really knows his stuff). - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: bit-for-bit (cont'd)
Of course, this biggest reason, as has been mentioned, is that the ATRAC in the MD prevents the MD from playing back a bit-accurate copy of what was recorded. But I'm assuming that modern ATRAC algorithms are fairly good. I don't think one can make that assumption on a wholesale basis, for example, about a sound card that truncates a 20-bit signal to 16-bits. (The proper way to shorten word length is to apply dither *before* truncation.) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: DIO2448 - MDS-PC2 problems...
After a *long* evening of work, I accomplished my first *digital* transfer from Cool Edit to the MDS-PC2. I have so far been unsuccessful at transfering from the output of the DIO2448 to the MDS-PC2; I get intermittent "DIN Unlock" when there's any audio. (It seems to lock to silence OK, but how useful is that?) I found that the codec on the computer's motherboard, a CS4299, had an SPDIF output on pin 48. A TOSLink transmitter, a little microsoldering to the pin on this tiny chip (about 1/4 the size of a postage stamp) and I was in business! I must note that I have hooked a Midiman CO3 format converter to both the DIO2448 and the motherboard output. Both give a continuous "source valid" indication. The CO3 has a CS8414 (it's the 96KHz version of the CS8412) digital audio receiver, so I connected a scope to the data output (pin 26) triggered from frame sync (pin 11). (This suggestion comes from Bob Katz http://www.digido.com/bitscope.html ) I got a very stable display, regardless of whether I was looking at the DIO2448 or the motherboard digital output. Could it be that the DIO's clock rate is off to far for the MDS-PC2? Or does it's output have some screwiness in the subcode bits that is tripping up the 'PC2? Anyway, now that I have some evidence that the 'PC2 is functioning properly, I'll go back to M Audio customer support. I'd be interested in knowing if anyone else has experience with either the DIO2448 or the MDS-PC2... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDS-PC2 first impressions...
=== The original message was multipart MIME=== === All non-text parts (attachments) have been removed === I purchased an MDS-PC2 a couple months ago. Up until today, I have been = using it in stand-alone mode (no PC), because its software would not run = on my NT machine. It has proven to be a fine unit, even without a PC, = over the past couple months. Today I got a second PC (HP Brio BA200) running Windows 98 with an M = Audio DIO 2448 digital audio card. I installed the Sony MD Editor 2 = program (that came with the MDS-PC2) and the DIO driver. An interface = is supplied with the MDS-PC2 allowing its Control A1 jack to connect to = a COM port on the PC. A single 1-meter TOSLink cable is supplied with = the MDS-PC2 (no cables are supplied with the DIO 2448), which was not = long enough for my installation. Anyway, 2 are needed, so I purchased 2 = Recoton 6-foot cables. (Note: the TOSLink connectors were almost to = large to fit the closely spaced connectors on the DIO.) After setup, I started up the MD editor and it asked me to select the = COM port (more about that in a bit). I then pressed play and it worked! = All in all, a fairly painless installation. There are 2 unresolved issues at this point. One is the COM port. The = BA200 only has 1 com port on the motherboard (COM1). Since I have other = devices (a Palm Pilot, etc), I ordered a 2-port add-on card. The BA200 = has no ISA slots, only PCI, and ports on the PCI serial card (Boca = IOP652) were not recognized by the Sony program, so I have to use COM1. = The second issue is that, although there was a header on the DIO for TTL = S/PDIF from the CDROM (and also onthe Samsung 40X CDROM drive that came = installed in the Brio), there was no cable supplied with the card, so = I'll probably have to make one. (I haven't installed my CD ripper - = CDWorx yet. A assume I'll be using it to go from CD to WAV.) The DIO = does have an internal header for the analog audio from the CDROM. Maybe = I'll just move the analog cable from the Brio motherboard audio to the = DIO, but it seems such a waste to use analog when all I need is a = digital cable. I only played around briefly with the Sony MD Editor program, but it's = operation looks to be fairly intuitive. Sony supplies a fairly = comprehesive manual in PDF format, which has answered all my questions = so far. I've been busy installing other software on the machine, = including Adobe Acrobat (so I could read the manual) and Cool Edit 2000. = I had to phone Syntrillium, as my previous registration was only good = for Cool Edit 96, but for an update fee of $29.95 (charged to my credit = card) they emailed the new registration, which I received in 5 minutes. One last recomendation I can give is to invest in some really good = speakers; most "computer" speakers are not very good. I've been using a = pair of Proton 301's for many years (biamplified 2" cone tweeter 5 = watts, 5" sealed woofer 20 watts, no sub). The response is = astonishingly flat between 100-2 Hz with none of the ringing = characteristic of some dome tweeters. The 301's use a 6dB shelving = low-end boost to get good response to 100 Hz, so it drops of rapidly = below 100 Hz. I'll write a bit more after I've had a chance to use the program for a = while.=20 === MIME part removed : text/html; === - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Questions re: Audio Outputs on Denon Deck
1. What is meant by "balanced" vs. "unbalanced" The unbalanced RCA connector is S/PDIF digital. The center pin is "hot" and the outside of the connector is ground. The signal is connected between "hot" and ground. The XLR connector is AES/EBU digital. The signal is between pin 2 and 3 (neither is grounded) and pin 1 is ground. 2. If I want to use the RCA plug digitial output The miniplug input is optical, so you need something to convert the RCA coaxial electrical information into optical information for use with fibre optic cable. One such device which does this is the Midiman CO2. Midiman also makes a model CO3, which has RCA (coax) and TOSlink S/PDIF as well as XLR AES/EBU. It has the added advantage that it allows one to tweak the SCMS bits, when appropriate. (This is often necessary when converting AES/EBU to S/PDIF, since AES/EBU does not define SCMS.) - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Intertrack noise...
Intertrack noise on CDs or MDs can be caused by 1) DC offset on the original *digitally transferred* recording. This causes a "pop" regardless of where the edit point is placed. The best way to fix this is to use the "Remove DC offset" checkbox in the "normalize" function of Cool Edit. 2) ADC noise in a computer sound card for *analog* transfers. Cheap soundcards may have poor anti-alias filters, so the noise might be caused by alias components. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: digital soundcard...
I just found a distributor called "Doctor Audio" http://www.doctoraudio.comselling Midiman DIO 2448 for $119! Looks like it might be the perfect way to hook MD to the computer. Anyone have any experience with this card or with Doctor Audio? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDS-PC2 has arrived!
I received the Sony MDS-PC2 from Video Direct. I'll have a detailed review after I use it a bit. This information is for computer interface hackers... Included with the MDS-PC2 is a Sony CAV-50C RS232 - Control A1 interface. I opened the interface to find the following major components: NEC 78082GA26 microprocessor with 5 MHz xtal Maxim MAX3221 RS232 transceiver Appears to be powered by the DTR and one other line, connected through steering diodes and a 3 terminal regulator. The CAV-50C says "VisionTouch" on its label, which leads me to believe that it is also used in Sony's Home Theater control system... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Sony XA-C30 with 3 MDX-65 changers...
For the past week I have been using a car system with 3 MDX-65 changers and an XA-C30 source selector box. The XA-C30 is mounted in the trunk along with the changers. Previously, my single MDX-65 was powered through the Unilink cable, but I had to run an unswitched battery line back to the trunk to power the XA-C30. The XA-C30 also requires a power ground connection which I fastened under a convenient screw in the trunk. Sony seems (wisely) to have used the shield in the audio cables for the ground that powers the analog circuitry to prevent noise from being introduced by the power ground. I bought 3 1-meter (Sony RC-61) Unilink cables to connect the XA-C30 to the MDX-65 units. Upon auto-detecting the XA-C30, a single digit (identifying the changer) appeared to the right of the "MD" on the display. In normal mode, the system will play through all tracks on all discs in MD1 in order, followed by MD2, then MD3 and it will then go back to MD1. The disc "+" and "-" keys function much as expected: MD2 disc 1 follows MD1 disc 6. However, it does *not* use the RAM buffers to provide uninterrupted play when switching changers as it does when switching discs in the same changer. This proved to be a major disappointment for shuffle mode 3 (where it picks random tracks from all available changers), causing about 10 seconds of silence while the display says "LOAD" between *every* track. The advantage of using multiple MD players is that one can have much more music "on-line" at a given time; great for long trips! However, in my opinion, the 10 second interruptions make shuffle mode 3 rather annoying; so having multiple changers is *not* that useful if you plan to use shuffle mode. Luckily, most of my discs are compilations that I recorded, so lack of shuffle is somewhat tolerable, just disappointing. Perhaps someone at Sony watches this list and will improve the software to fix shuffle mode 3 in future releases. My guess is that it would involve buying a new head unit (BTW I'm using the unilink control part of the RF adapter with the audio feeding directly into "aux" inputs on my radio) and maybe a new switching unit... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: digital clipping indicators...
There has been some discussion of digital clipping and how the "over" indicator behaves. While this depends on the actual implementation, here are the important issues: On record, the "over" indicator should indicate when the maximum sample value has been exceeded. Since it is impossible to record a value greater than the maximum sample value, there should never be any true "over" during playback. However, some equipment infers that 3 contiguous samples of the maximum value are almost always caused by an "over" during record. That is, it infers that the middle sample of the 3 must have been "over". That stated, it should be pointed out that consumer equipment is often not this rigorous when it comes to level indication. However, I should also point out that occasional "overs" are almost always inaudible. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDS-PC2 ordered!
I just ordered an MDS-PC2 from Video Direct, in Florida (800) 368-5020 for $289 + $15 shipping. They inform me that they expect to receive a batch of them from Sony in 10-15 days, so I should receive it mid to late October. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: In car changer, shuffle mode
I've got a Sony MDX-65. It starts the disc change while the end of the current track is playing so that there is no pause in the audio. And at normal level... I'll be installing a 3-unit Unilink switcher and 2 more MDX-65s shortly, so I can let you know about the multi-changer shuffle mode, if there is any interest. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: : Re: Digital Recording: why is volume low?
When digital dubbing, it's important not to record the signal any higher than the "0" mark or you'll get clipping and distortion (when making a digital recording--analog things like cassette decks let you go over the "0" a tad before you start to hear distortion). Whilst I agree with this statement totally, in principle, it does seem that somehow the Sony engineers have allowed for accidental excursions over the 0dB mark. "0" indicates the largest number that can be expressed, and the "over" indicator means that you've exceeded that number, so clipping *will* occur. In all but the most stringent listening environments, occasional clipped peaks are almost impossible to detect. Most would have a difficult time hearing even moderate clipping on a portable unit with "fold-up" headphones or ear buds. But be aware that clipping *does* occur every time the "over" indicator lights, so one should consider whether the resulting distortion will be masked by noise and distortion in the source material or playback environment. Solutions to the problem of not-enough-headphone-level include outboard amplifiers, changing to a unit that has higher output voltage (usually one with more available battery voltage) and changing to more sensitive headphones. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: Subject: RE: What is a plug in power mic?
1. Could using a regular dynamic mic really damage my 702? Before I saw the warning in the manual, I tried a regular mic and it seemed to work OK (but occasionally there was a lot of static -- could the power supplied by the 702 have built up somewhere and then discharged, causing the static?) A dynamic microphone has a small coil of wire attached to the diaphram. This coil moves through a strong magnetic field to generate the output voltage. This coil will obviously conduct DC current, and will present a much lower resistance DC path than the FET, so the 702 will supply much higher-than-normal current, which will certainly shorten the battery life even if it doesn't damage things. The static probably is the result of small changes in resistance during slight movement of the plug which normally would not cause a problem, but because of the high current, they create enough voltage change to hear. Bottom line: pay attention to the warning! If you want to use a dynamic mic, place a capacitor, say 47 uF/16 V in series with the hot lead (+ to the 702, - to the mic) to block the DC. 2. Is it OK to use an electret mic that takes a battery? I'd think that with the battery in, it would "look" like a regular dynamic mic to the unit, so would present the same danger that the manual warns against. I'd also think that with the battery out, there would be an open circuit so the FET wouldn't be powered. Should work OK, because most mics with the battery in the plug have a series capacitor to keep the DC out of the recorder input. The best way to use such a mic however, is just to cut off the plug (containing the battery) and replace it with a normal (non-battery) plug that fits the recorder. Since the recorder already has the circuitry for powering the mic, you'll eliminate the duplicated circuitry in the battery plug. BTW-- The power circuitry for electret mics generally consists of a series capacitor (47 uF/16 V, + to the mic, - to the input stage of the recorder) to block the DC. A small resistor 1.5K to 2.7K is connected from a DC voltage source to the mic to provide power. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Subject: RE: What is a plug in power mic?
The discussion of electret microphones here is mostly correct in that they are a type of condensor microphone and that they require an excitation voltage, usually between 1.5 and 10 VDC. All capacitor mirophone capsules are inherently very high impedence devices, so high that they cannot drive the connecting cable. Therefore, a preamplifier (in the early days a vacuum tube, but more recently a FET) is placed in the microphone case to drive the cable. A normal condensor microphone requires a fairly high excitation voltage, so often a small switching power supply is also placed in the microphone case to convert the "phantom" power supplied via the cable to high voltage for capsule excitation. Electret microphones have permanently charged capsules, made by heating a special dialectric material while it is in a strong electrostaic field, so there is no need for high voltage excitation (and the required switching supply). The excitation voltage for an electret microphone powers the FET amplifier. An electret, in practice, is a very simple and elegant microphone... It is made up of just the capsule and a FET transistor. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: Analog input on JE520
I have done more experimentation regarding the analog input levels on the JE520. I have come to the conclusion that the "over" indicators do not accurately report ADC clipping if the input gain is set to less than 0.0. In fact, in most cases, no amount of input signal will light the "over" indicators at negative gain settings, however, severe clipping may result in the ADC. Positive input gain settings result in some loss of ADC resolution; +12.0 dB (the maximum allowed) causes a loss of 2 bits. Therefore, optimum results are obtained by setting the input gain to 0.0 and controlling the level externally. That said, there should be little reason to use a negative gain setting under normal circumstances, since the gain between the analog inputs and the ADC is fixed requiring a *very* hot input to overload the ADC. Therefore, the most common situations will require a positive gain adjustment causing slight loss-of-resolution (almost inaudible). When recording from vinyl, I am using a PS Audio IV preamp which has a line output amp (after the level control) allowing it to put out a hot enough signal to overload the JE520's ADC. I obtain the best results by setting the JE520's input gain to 0.0 and using the level control on the preamp. However, switching off the line amp and boosting the level on the JE520 produces acceptable results. The moral of this is to beware of ADC clipping if the JE520's input gain is set below 0.0, because the over indicators will be unreliable in this case! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: JE520 analog input level...
Perhaps I should have been more clear. The underlying issue I was getting at is: what is the resolution of the A/D converter on the JE520 and how is that affected by adjusting the record level *after* the signal has been converted to digital? Let's say you increase it by 12 dB. I think that means you end up with two less bits of resolution that if you'd have increased the level by 12 dB *before* the A/D. My observation to date is that this effect is fairly subtle on most program material. Let's say you decrease it by 12 dB. Does the A/D have enough headroom to sample a signal that's 12 dB hotter than normal? My observation is that the A/D *does* clip in this situation; the loss of dynamic range is clearly audible on just about any material. My point is this: it seems that to get optimum performance out of the A/D, it seems best to put a potentiometer at the analog input of the deck. BTW -- If you use a pot with a conductive plastic element, this will minimize any degradation. In my opinion, such a setup would be far better than operating the A/D at a non-optimum level. Digital record level adjustment seems to be the wave of the future; it's less expensive for the manufacturer since the hardware to do it is already there. It's a firmware issue. I'm not so sure that the end user, who just wants to turn a knob to adjust the level, can understand the tremendous potential for negative effects without first having an understanding of the technical aspects of digital audio. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: JE520 analog input levels...
Here are some questions and observations regarding analog recording on the JE520. Does anyone know what tradeoffs are involved in using the analog input level adjustment on the setup menu? I don't think there's any way on the JE520 to adjust the input level to the A/D converter, so it would seem that one looses resolution when a low-level signal is applied and boosted with the adjustment in the setup menu. OTOH, the converter may be overloaded when a high-level signal is reduced with the setup menu level adjustment. My experience to date is that it is best to adjust the input level with an external pot so that the setup level adjustment can be left at 0.0. Using the setup menu to boost the level doesn't seem to leave many sonic artifacts, but I haven't tried "difficult" material (like low level solo piano). Using the setup menu to cut the level seems to be the kiss of death, because I think that the A/D gets overloaded. Given that one usually likes to maximize the level on the disc to make the best use of the available disc (is this indeed true with ATRAC encoding?), does it make sense to ever adjust the record level setup to less than 0? Does anyone know exactly where in the digital signal path the record level setting is applied? A flow chart, showing the order of processing steps along with word length (# bits) and sample rate (1x, 2x, etc) would be very helpful... Unless I hear a reason not to do so, I'll continue to leave the setup level adjustment at 0.0 and make the adjustment with an external pot. - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: RM-X69RF
How is the RM-X69RF controller supplied? Is it supplied with the RF block (which feeds the FM car radio) built into the controller circuitry or as a separate block, apart from the controller box, which just receives power from the controller box and the signal from the changer? It is in one small metal box 5" x 4" x 1 1/4" with the following connectors on one of the 5" faces: L audio in (RCA), R audio in (RCA), wired remote in 1/8 phone, Unilink, LCD display. On the opposite face: power (4 pin Molex), antenna (Motorola jack on a cable), RF to receiver (Motorola plug on a cable). The LCD module is 4 1/2" x 1 1/4" x 3/4", at the end of a 5 foot cable that can be detached at the main unit. It consists mostly of a large backlit display, a single "power" button that initiates (or suspends) playback, an IR detector for the wireless remote and a "paper clip" CPU reset button. To use it without the FM modulator, simply do not hook up the antenna and RF cables. Also, how do some car radios open up the line-input when the auxiliary source is started? Does this involve any special wiring to cause this trigger and can this controller provide that special trigger arrangement for these radios? Various radios work differently, but the RM-X69RF has an undocumented "amp turn-on" lead (the red/blue wire from the power Molex connector) that goes to +12V when playback is initiated. I guess you *could* just use it with an amp (no radio) if you put a volume control between the changer and the amp. Good luck! - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
Re: MD: RM-X69RF...
The RM-X69RF can indeed be used as "just a controller" by connecting the RCA outputs of the MDX-65 to another device. Here's the catch: due to the way the MDX-65 gets its analog ground, you have to at least connect the shield of one of the RCA's on the MDX-65 to the shield of the RX-X69RF RCA (either channel will work and the center pin need not be connected). This requirement MAY already be taken care of since the radio and the RM-X69RF have the same ground; it was not on the powered speaker. My guess is that this so that the MDX-65 analog circuitry gets a "clean" ground, rather than use the "dirty" power ground in the Unilink cable. This probably helps eliminate electrical noise from the various motors in the MDX-65. I've got an XA-C30 3-input Unilink switcher on the way from Sony (had to order it direct since no dealers seemed to know what it was). If anyone's interested, I can report how this affects the situation when multiple changers are hooked up to the RM-X69RF in controller-only mode. My guess is that they still require the analog ground (separate from the Unilink cable)... - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
MD: MDX-65 problems...
I'm having problems playing discs in an MDX-65 changer. I have it connected to a RM-X69RF controller. Right now its sitting on the workbench running from a 12VDC bench supply. The audio output is connected to a powered speaker. When it is running, the display indicates normal operation. I can see the track numbers and time, etc. The problem is that I get no audio output. The discs in the player were recorded in mixed mono/stereo (mostly mono) and play back fine on my JE520 and MZ-EP11. I've checked the obvious like the digital/analog switch in the analog position. This may be a clue: when I look at the TOSLink connector, there is no visible light (even with the switch in digital). I've had this problem with 2 units now (the first I exchanged as defective). Can anyone offer me any suggestions or let me know their experiences with the MDX-65? - To stop getting this list send a message containing just the word "unsubscribe" to [EMAIL PROTECTED]
