Marantz PMD660 portable digital recorder review
Paul Rubin
Walkman. I got it a few weeks ago and immediately recorded Consonance
with it. I haven't had time to do much with it since, and there's
some functions I haven't tried, but I'll review it now anyway. My
rating for it is 3 stars out of 5. I'll add that I'm a computer geek
and not a recording whiz, so that colors the review.
Other units I considered are the Edirol R-1 (a direct competitor) and
some units intended mainly as playback units (iRiver H320, Neuros
Audio Computer) that have recording functions. I downloaded the R-1
instruction manual and found it had lots of stupid misfeatures and
figured it was a first generation product, and the playback units
seemed to mostly treat recording as an afterthought and would have
needed an external mic preamp and maybe external level meters (too
much junk plugged together) so I chose the PMD660. The PMD660 is a
follow-on to Marantz's earlier PMD690, PMD690 II, and PMD670, so I
figured they'd have gotten the bugs out by now. But I think it's
still a long way from being a mature product.
My goal was to have a recorder that I could use for a whole
3-day-weekend event completely on battery power, i.e. 30 hours of
recording not nonstop, but using an affordable and portable quantity
of batteries and media. I calculated that with an external battery
pack (NiMH D cells) and with 2 GB of CF media (about $130 at today's
prices), I could record that much.
Prior to now, I've been recording these things with a camcorder, and
before that, with analog cassette recorders, and have generally been
happy with the results. So I'm not after ultra-audiophile quality
sound. The main thing I wanted was a way to make 30 hours of
recordings without a huge pile of tapes and batteries, and this does
it wonderfully.
Unfortunately I didn't have a chance to get an external mic working
before Consonance (see the XLR connector issue below). I have a Sony
ECM-MS907 single point stereo mic which is your basic Minidisc-style
cheap external mic, but only recently got the adapter to let it work
with the PMD660. So I recorded most of the con in stereo with the
internal mics, except for some parts that I did in mono with a
borrowed Shure SM58.
I'll see about posting some mp3 samples later so people can judge the
sound quality for themselves. I'm mostly going to comment on the
hardware and interface aspects and leave the audio to the audiophiles.
Good things about the PMD660:
- Records to standard media (compact flash) in standard filesystem
format (FAT32) instead of using a proprietary system like the NJB3.
Supposedly supports the USB mass storage interface so you can plug
it into a computer and drag/drop files, but I haven't tried this
(see below).
- Uses standard or at least published recording formats (MP3 and WAV)
instead of proprietary ATRAC like Minidisc recorders.
- Uses standard AA batteries instead of proprietary lithium batteries
that in many cases aren't even removable from the unit.
- Has XLR mic inputs with phantom power, so you can use professional
microphones. These connectors are much more solid and reliable than
the typical 1/8" stereo miniplug.
- Battery runtime (at least using the built-in mics, no phantom
power etc.) is somewhat longer than the advertised 4 hours. I
used both Redcell alkalines and some 1600 mAH NiMH cells from my
digicam and got 5+ hours either way. Redcells aren't the greatest
alkalines, and NiMH is available with up to 2500 mAH now, so with
good cells, runtime should be even longer. But see below for
how they could have improved on this.
Not-so-good things that they really should fix:
- Construction quality isn't the greatest. It feels like decent
quality consumer electronics (soft plastic case, etc.), not
professional equipment designed to be banged around. It's advertised
as being rugged and professional, so it's below expectations in this
regard, but I'm ok with it since I'm a fairly light-duty user.
- Mic preamps are sensitive to electrical noise. You get hash in
the recordings if you use it next to a computer, at least with the
built-in mics. It really should have better shielding. This is a
digital gizmo and people are going to use it in conjunction with
computers or digital mixers, so this is a significant design flaw.
- User interface is clumsy. Lots of menus to scroll through, lousy
tactile feedback from the buttons. My old recorder is a Sony
TC-D5M analog cassette recorder that was built like a tank in the
1980's and is absolutely natural to use. The PMD660 is a great
leap backwards by comparison.
- The MP3 recording bit rate is limited to 128kbps (stereo) or
64kbps (mono). This is stupid. The earlier PMD670 supported a
range of MP3 bit rates and there was no good reason to remove that
choice. The 128k stereo recordings are listenable but the 64k
mono sounds like crap. This is predictable since like most encoders,
the PMD uses joint stereo format, i.e. it records (L+R, L-R) instead
of (L,R). L+R is basically the mono signal and L-R is the difference
between channels. Since the channels tend to have similar content,
joint stereo encoding usually gives more of the available bits to the
L+R signal (let's say 96kb L+R and 32 kb L-R). That means for mono
recordings to sound as good as 128kb stereo, they need to be encoded
at 96kbps or faster.
This is a significant bug because the 64kb recordings really do
have noticable warbling and stuff. MP3 was just not designed for
such low rates. But the recorder is partially targeted at
journalists and lots of them will want to record mono. I did some
music recordings in mono because I could at the time only get my
hands on one microphone. There's such a thing as an XLR Y-cable
that lets you plug a single mic into two inputs, but I don't know
what that does to the impedance match, or what happens if you
short the two channels of phantom power together like that.
- The line-level input uses a 1/8" stereo minijack, which is fairly
typical. But there's no way to use that jack as a mic input. That
also seems crazy. This recorder is in the same general category as
a minidisc recorder or small DAT, and those all use 1/8" mics and
most of the mics made for such devices have 1/8" plugs. It's good
to be able to use XLR mics but it's nuts to not have a switch that
lets you also use 1/8" plug mics.
I thought adapters would be easy to find too, but they turn out to
be quite hard to find. They didn't exist in any electronics or
music store that I checked. Of course you can buy loose
connectors and solder it all together. I was too lazy for that
and finally did find an online source of an adapter cable ($25!).
It's the Sound Professionals SP-XLRM-MINI-2 (soundprofessionals.com)
if you need one. Since the thing appears to be handmade, if you
add up the costs of parts and labor, the price isn't too bad. But
the thing just shouldn't be needed.
- You can't use the USB cable unless the AC adapter is plugged in.
That's also a drag. This thing is sold as a field recorder and
you can't upload files from it on battery power? This isn't too
big a deal, since the media capacity in MP3 format is so large,
and you can always transfer files by removing the CF card and
plugging it into the laptop with a PC card adapter (which is what
I did at home), but it's kind of silly. Anyway, I haven't tried
the USB cable at all yet. I don't know if that limits the
"virtual editing" feature which I also haven't tried.
- The docs don't mention if the USB port supports USB2 high speed so
I assume it doesn't. That means transfers are probably slow.
But this is just an unconfirmed suspicion for now.
- There's a pushbutton that lets you make "edit marks" while you
record. Labels in the LCD menus give the impression that the edit
marks actually make subaudible sounds in the recording, in the
style of dictation recorders--ugh! But instead, the unit leaves a
"pmd660.edl" file on the CF card where "edl" presumably means
"edit list". This is a binary file in an undocumented format
presumably understood by some proprietary crapware on the CD that
came in the box, that I haven't opened. But the file has fairly
obvious visual structure when examined with a hex editor, so I have
reasonable chances of reverse engineering it (I haven't made any
real effort at this yet).
- The battery gauge is the typical semi-useless "full-ok-empty" icon
with just 3 visible states. There's a menu selection to tell the
recorder whether you're using NiMH or alkaline batteries and I
guess this sets the threshholds. Either way, the icon goes to the
"empty" state when you still have several hours of juice left, and
after that there's no way to know just how much is left. This
recorder is intended for sophisticated users so instead of the
"idiot light", they should just display the battery voltage, like
5.37 volts or whatever.
- Uses a somewhat hard to find (i.e. Radio Shack didn't have it)
connector for the external power supply. I'd planned to make an
external battery pack but need to find that connector first. On
the other hand, I've ordered a bunch of 2500 mAH NiMH cells so I
might just use those instead, swapping them out every 5-6 hours
whether it's needed or not.
Conclusion:
This is a reasonable first cut at what a portable recorder should be,
with some serious imperfections. I have some buyer's remorse. It
came with a 30 day return period from a friendly retailer and I've
been thinking of sending it back (I have a few more days) but I
probably won't, because I'm lazy and because I don't know what I'd get
instead, and I figure if I want something else, I can sell this on
ebay without taking too big a bath. It's way more civilized than a
DAT recorder or Minidisc because it allows transferring gigabytes of
recordings to a computer (17 hours/GB in stereo MP3 format, 1.5
hours/GB in 44/16 WAV) in just a few minutes in direct digital form.
But, I think much better units will come along, and could be made
right now without having to wait for any further technology advances.
So this is a reasonable unit if you need something right away. I'd
hope the industry would get its act together before long, but I'm
amazed that it's taken as long as it already has to make the PMD660,
so I wouldn't hold my breath expecting too much very soon.
Mike Rivers
In article <7xbr8wl5...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> My goal was to have a recorder that I could use for a whole
> 3-day-weekend event completely on battery power, i.e. 30 hours of
> recording not nonstop, but using an affordable and portable quantity
> of batteries and media. I calculated that with an external battery
> pack (NiMH D cells) and with 2 GB of CF media (about $130 at today's
> prices), I could record that much.
Me, too. But I figure that 10 hours of 44.1 kHz 16-bit stereo is about
6.6 GB. What compromises are you making to that in order to fit it on
to a 2 GB flash card? Or are you figuring on recording on the card
during the day and then dumping it to your computer overnight (you
mean you actually sleep during a Con?) while you recharge the
batteries? Seeing what you wrote about "good things" I guess you
didn't do that.
> - Has XLR mic inputs with phantom power, so you can use professional
> microphones. These connectors are much more solid and reliable than
> the typical 1/8" stereo miniplug.
I like it. But the preamps (the part between the mic connector and the
A/D converter) has to be of usable quality. Marantz has a couple of
portable CD recorders with disappointingly poor noise performance -
better than a cassette but far from "CD quality" and the only thing I
can attribute this to is a noisy analog input stage. They can't be
using converters that bad in a contemporary design.
> - Battery runtime (at least using the built-in mics, no phantom
> power etc.) is somewhat longer than the advertised 4 hours.
I usually can find a place to plug in my Jukebox power supply, but
I've run it (playing) for a round trip flight across country on a
charge. 4 hours on a set of D cells isn't anything to crow about.
Nagras do way better than that (though admittedly use more cells).
> - Construction quality isn't the greatest.
This has been a characteristic of Marantz portable recorders ever
since they came out with the cassette recorder that became the cheap
standard for field recording and sound gathering. It goes with the
territory and it's a tradeoff for the "affordable" price. Bottom line
is that it costs X dollars to record for Y years. If Y is sufficiently
small, you can throw away the recorder at the end of Y years and the
cost is the cost of the recorder. But for large Y, you may need to buy
several recorders, which increases the value of X.
> - Mic preamps are sensitive to electrical noise. You get hash in
> the recordings if you use it next to a computer, at least with the
> built-in mics.
I've heard this complaint before, though not necessarily related to
the internal mics. It would be good if in future experiments (if it
sticks with you long enough) you could determine if this is related to
the internal mics or the overall shielding of the unit.
> - User interface is clumsy. Lots of menus to scroll through, lousy
> tactile feedback from the buttons.
Are there presets that you can set for turn-on-and-go recording? When
looking at the PMD-670 (I think) at a trade show, I asked about this
sort of thing and the guy demonstrating it showed me that it could be
set up for defaults of recording mode, sample rate, whether the
limiter was engaged, input source, etc. They wanted to make it so that
a news reporter could turn it on and record without having to fuss
with all that stuff every time.
> - The MP3 recording bit rate is limited to 128kbps (stereo) or
> 64kbps (mono). This is stupid.
Stupid to limit it, but probably practical when you consider the cost
(or maximum size) of the flash memory card. For most applications,
it's more important for them to be able to advertise "10 hours
recording on a single card" than "near CD quality," which it probably
wouldn't be anyway, for other reasons.
> the PMD uses joint stereo format, i.e. it records (L+R, L-R) instead
> of (L,R). L+R is basically the mono signal and L-R is the difference
> between channels. Since the channels tend to have similar content,
> joint stereo encoding usually gives more of the available bits to the
> L+R signal (let's say 96kb L+R and 32 kb L-R). That means for mono
> recordings to sound as good as 128kb stereo, they need to be encoded
> at 96kbps or faster.
That's interesting. I haven't heard of this technique, but then I
never use MP3 enconding other than for things I'm only going to listen
to once. I didn't realize that the encoding scheme worked on two
channels independently rather than on a stereo stream.
> I did some
> music recordings in mono because I could at the time only get my
> hands on one microphone. There's such a thing as an XLR Y-cable
> that lets you plug a single mic into two inputs, but I don't know
> what that does to the impedance match, or what happens if you
> short the two channels of phantom power together like that.
Not to worry, but I'm surprised that it doesn't have a switch to send
the single input to both channels. The cassette models do. Perhaps
they considered this unnecessary since there's a common mono MP3
format. If you were recording in WAV format, however, and planned to
burn an audio CD from the recording, if you were recording in mono,
you'd probably want to do that. The Y-cable trick is often used by
journalists to let them set different input levels on the two tracks.
That way they don't have to depend on the limiter if something gets
too loud - it's probably safe on the lower gain track. Sometimes handy
for festival music recording, too.
> - The line-level input uses a 1/8" stereo minijack, which is fairly
> typical. But there's no way to use that jack as a mic input. That
> also seems crazy.
Not to someone who wants to use real microphones. What seems crazy to
me (and I assume this is the case) is that other than swithing in a
pad, there's no way to use the XLR inputs as line inputs.
> This recorder is in the same general category as
> a minidisc recorder or small DAT, and those all use 1/8" mics and
> most of the mics made for such devices have 1/8" plugs.
My portable DAT recorders has XLR inputs. The only thing I dislike
about my Jukebox 3 is that it has only a mini phone jack. But then
I've never owned a typical "Minidisk recorder microphone." If you want
to play with the pros, get a more "pro" mic. You'd be surprised at how
your recordings will improve, and it's not just because of the XLR
connector.
> finally did find an online source of an adapter cable ($25!).
> It's the Sound Professionals SP-XLRM-MINI-2 (soundprofessionals.com)
I can see that you might find it distasteful to have to spend $25 to
hook up your $75 mic. But you realize that you're doing an upgrade
here by bringing that recorder into your life, and that you can't
always just upgrade one piece of a system without upgrading a couple
more things along the way. $25 is cheap to "upgrade" your mic so you
can use it with the new recorder. However, given the MP3 limitations
and the media capacity constraints when recording in WAV format, it
may not be worth completing the upgrade with a $500-ballpark stereo
mic such as the Rode NT-4 or somewhat less expensive but not trivial
Audio Technica AT-825.
> - You can't use the USB cable unless the AC adapter is plugged in.
> That's also a drag. This thing is sold as a field recorder and
> you can't upload files from it on battery power?
That is a little silly. I guess they want to be sure that you don't
lose power mid-transfer. It might not make a lot of difference going
out, but if it crashes while transferring a file into the recorder
(like you might want to listen to some CDs on your trip to the Con) it
could hose the formatting on the memory card and destroy or destroy
access to whatever else is on there. The drain on the batteries in my
digital camera seems to go up when I plug in its USB cable. I solved
that problem with a $10 memory card reader for my computer that I plug
into the computer's USB port. Plug in the memory card rather than the
camera (or in your case the recorder) and this problem is solved.
> - The docs don't mention if the USB port supports USB2 high speed so
> I assume it doesn't.
Generally two-channel devices are USB 1.1.
> That means transfers are probably slow.
> But this is just an unconfirmed suspicion for now.
I use USB to transfer recordings from my Jukebox to my laptop
computer. For practicality (since I then put them on to CD) I start a
new file every hour, and it takes about 20 minutes to transfer an
hour's worth of 16/44 WAV file (a bit over 600 MB) over USB 1.1. They
tell me that Firewire (which the Jukebox also accommodates) is a lot
faster but I've found that it about halves the transfer time from USB,
that's all.
> - There's a pushbutton that lets you make "edit marks" while you
> record. Labels in the LCD menus give the impression that the edit
> marks actually make subaudible sounds in the recording, in the
> style of dictation recorders--ugh! But instead, the unit leaves a
> "pmd660.edl" file on the CF card where "edl" presumably means
> "edit list". This is a binary file in an undocumented format
> presumably understood by some proprietary crapware on the CD that
> came in the box, that I haven't opened.
No, it's something that's understood by the recorder so that when
playing back recordings on the recorder, you can locate your marked
points. A utility that reads this and opens a text editor so that you
can annotate your recording would indeed be a cool thing, but I'll bet
they didn't think about it. The crapware CD may contain USB drivers
for computers that don't recognize the USB mass storage device. Or it
may have an audio editing program.
> This is a reasonable first cut at what a portable recorder should be,
> with some serious imperfections. I have some buyer's remorse. It
> came with a 30 day return period from a friendly retailer and I've
> been thinking of sending it back (I have a few more days) but I
> probably won't, because I'm lazy and because I don't know what I'd get
> instead
I know the feeling. I ususlly end up taking things like that back and
waiting to see what else comes along. But if you decide to keep it, do
think about getting a mic worthy of the recorder insted of using the
one you have now. Get a mic that's good enough to use with your NEXT
recorder so you won't have to upgrade again too soon.
--
I'm really Mike Rivers (mri...@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Aaron Davies
> Paul Rubin <http://phr...@NOSPAM.invalid>
>
> > That means transfers are probably slow. But this is just an unconfirmed
> > suspicion for now.
>
> I use USB to transfer recordings from my Jukebox to my laptop computer.
> For practicality (since I then put them on to CD) I start a new file every
> hour, and it takes about 20 minutes to transfer an hour's worth of 16/44
> WAV file (a bit over 600 MB) over USB 1.1. They tell me that Firewire
> (which the Jukebox also accommodates) is a lot faster but I've found that
> it about halves the transfer time from USB, that's all.
Firewire is capable of speeds of up to 400Mbps (USB 1.1 is 12Mbps),
which could in theory transfer your CD's worth of music in 12 seconds,
but I imagine the internal storage format of your Jukebox is the
limiting factor here. For that matter, most *hard drives* aren't capable
of receiving data at 400Mbps--that's what RAID 1 is for.
--
Aaron Davies
Opinions expressed are solely those of a random number generator.
"I don't know if it's real or not but it is a myth."
-Jami JoAnne of alt.folklore.urban, showing her grasp on reality.
John in Detroit
recording) and I'll post my comments My HI-MD is an MZ-NHF800
Media at 3 bucks a pop from J&R Music New York is 1 gig, holds about 1.5
hours of PCM (Wav) recording, MDs use a 20 bit depth if memory serves so
it is better than CD PCM recordings may then be transfered to computer
and converted to WAV with no loss, This is about as good as it gets
folks, this is CD or better quality. Note, there are better recording
methods but if your goal is a CD.. Why spend the money.
No XLR inputs (Box is not big enough) but.. IT has a line in 1/8 inch
jack and I have an external professional grade mixer and IT has XLR in
Battery life (It uses one AA either alkaline or Rechargeable) is
advertised at about 1 disk per battery, Unit will test the battery when
you press RECord and tell you "Not enough battery"
Expierence was 2 to 3 disks per battery (Alkaline)
If you record in ATRAC mode you can put up to 35 hours on one disk,
however the quality is what they call "Voice Quality" At If you record
in Hi-MD mode at standard quality it's as good or better than my MZ-R55.
HOWEVER though you can transfer the recording to the computer (Using
SOnic Stage software) You can not convert it to wav.
I have, however, played the disc into the recorder and used Total
Recorder to capture the digital stream on it's way to the digital-analog
converters and saved the resulting file to disk.. That works.. And T.R.
Professional version will auto-split recordings so you can have all
sorts of fun (Not all programs like 5 gig-a-byte files, in fact FAT-32
does not support files that large)
And I like MD Disks much better than flash ram,,, Though I do admit it
would be nice to have a couple of gigs in a flash box so I could do
faster transfers (Takes an hour to transfer an hour's recording into
this laptop.. Faster on the big box in the basement)
Paul Rubin wrote:
> This is a solid state digital recorder about the size of a large
> Walkman. I got it a few weeks ago and immediately recorded Consonance
> with it. I haven't had time to do much with it since, and there's
> some functions I haven't tried, but I'll review it now anyway. My
> rating for it is 3 stars out of 5. I'll add that I'm a computer geek
> and not a recording whiz, so that colors the review.
--
John F Davis, in Delightful Detroit. WA8YXM(at)arrl(dot)net
"Nothing adds excitement like something that is none of your business"
Diabetic? http://community.compuserve.com/diabetes
Rich Brown
> The Y-cable trick is often used by
> journalists to let them set different input levels on the two tracks.
> That way they don't have to depend on the limiter if something gets
> too loud - it's probably safe on the lower gain track. Sometimes handy
> for festival music recording, too.
That trick won't work if the recorder is internally using L+R and L-R as
its two recorded tracks.
Paul Rubin, you mention the Neuros Audio Computer as one of the contenders
when you started shopping. I've been intrigued by that model. Can you
explain why it didn't make the cut?
__
There's naught but harm and blight and pain in the pitch of the Redmond
man
Rich Brown -- rab -- http://freemars.org/filk/
Paul Rubin
mri...@d-and-d.com (Mike Rivers) writes:
> > I calculated that with an external battery
> > pack (NiMH D cells) and with 2 GB of CF media (about $130 at today's
> > prices), I could record that much [30 hours, not nonstop].
>
> Me, too. But I figure that 10 hours of 44.1 kHz 16-bit stereo is about
> 6.6 GB. What compromises are you making to that in order to fit it on
> to a 2 GB flash card?
Heh, 128kbit/sec mp3, should fit 34 hours in 2GB. I figure it should
sound at least as good as the analog cassette recordings I used to make,
that always satisfied me. Yes, it's definitely a compromise compared to WAV.
> Or are you figuring on recording on the card during the day and then
> dumping it to your computer overnight
Well, if I want to dump to a computer, it shouldn't take overnight. My
slow 1GB CF card uploads through a plain PCMCIA adapter in about 11
minutes. A fast CF card and a Cardbus adapter should get it down to
maybe 3 minutes per gigabyte. There's exhaustive speed tests for
different CF card combinations at www.robgalbraith.com (pro digicam
site).
I prefer not to bring a computer to a con, but if I want to record WAV
and upload to disc, I might get something like one of these:
http://store.yahoo.com/tonyh/apshstcdwi40.html
http://store.yahoo.com/tonyh/xprovp.html
http://store.yahoo.com/tonyh/trusb20postd1.html
They are small hard drive enclosures with CF slots and battery packs, so
you can upload your CF card to disc in the field. Digital photographers
like to use these things. One of the ones above has a built-in mp3
player too.
> (you mean you actually sleep during a Con?) while you recharge the
> batteries? Seeing what you wrote about "good things" I guess you didn't
> do that.
I do sleep at cons, but usually only when I'm falling over from
exhaustion, which means if I have batteries to recharge, I won't remember
to do it before crashing, so I have dead batteries the next day. That's
happened to me several times. That's why my buying criterion was being
able to bring enough fully charged batteries to record 30 hours with no
on-site charging needed.
Nowadays there are several AA NiMH systems on the market that recharge in
30 minutes or even 15 minutes, which make recharging during a session
feasible. So I can consider that but would rather keep things
streamlined and avoid the need for it.
> I like it. But the preamps (the part between the mic connector and the
> A/D converter) has to be of usable quality. Marantz has a couple of
> portable CD recorders with disappointingly poor noise performance -
> better than a cassette but far from "CD quality" and the only thing I
> can attribute this to is a noisy analog input stage. They can't be
> using converters that bad in a contemporary design.
Yeah, the noise specs aren't so great, it's kind of sad. I'd think
they could use better stuff without increasing costs too much.
> I usually can find a place to plug in my Jukebox power supply, but
> I've run it (playing) for a round trip flight across country on a
> charge. 4 hours on a set of D cells isn't anything to crow about.
> Nagras do way better than that (though admittedly use more cells).
4 hours on D's is terrible, you should be able to do much better.
Have you tried rechargeable D's? They should have much less voltage
sag than alkalines. Here's a cheap source--I can't make promises
about the quality. I may order some anyway and report on them later:
http://batteryspace.com/index.asp?PageAction=VIEWPROD&ProdID=709&HS=1
> > - Construction quality isn't the greatest.
>
> This has been a characteristic of Marantz portable recorders ever
> since they came out with the cassette recorder that became the cheap
> standard for field recording and sound gathering. It goes with the
> territory and it's a tradeoff for the "affordable" price.
Oh well. I think they built this model to be able to sell it in the $200
range, so there's a huge markup right now that they're getting because
there's so little competition. In terms of hardware per dollar it
doesn't seem like a very good value right now. I paid an early adopter
premium but I expect prices will drop just like with any other computer
stuff.
> > - Mic preamps are sensitive to electrical noise.
>
> I've heard this complaint before, though not necessarily related to
> the internal mics. It would be good if in future experiments (if it
> sticks with you long enough) you could determine if this is related to
> the internal mics or the overall shielding of the unit.
Thanks, I will try to do this.
> > - User interface is clumsy. Lots of menus to scroll through, lousy
> > tactile feedback from the buttons.
>
> Are there presets that you can set for turn-on-and-go recording? When
> looking at the PMD-670 (I think) at a trade show, I asked about this
> sort of thing and the guy demonstrating it showed me that it could be
> set up for defaults of recording mode, sample rate, whether the
> limiter was engaged, input source, etc. They wanted to make it so that
> a news reporter could turn it on and record without having to fuss
> with all that stuff every time.
Yes, there's 3 presets and the unit does keep the settings between
battery stations. Switching between the presets involves a little
bit of button twiddling. It's not too bad as long as you remember
what settings the presets correspond to. Otherwise you have to scroll
through the menus to see how everything is set.
> > - The MP3 recording bit rate is limited to 128kbps (stereo) or
> > 64kbps (mono). This is stupid.
>
> Stupid to limit it, but probably practical when you consider the cost
> (or maximum size) of the flash memory card. For most applications,
> it's more important for them to be able to advertise "10 hours
> recording on a single card" than "near CD quality," which it probably
> wouldn't be anyway, for other reasons.
Nah, they could still advertise "10 hours recording [at 128 kbps]" and
still be able to record at higher bit rates. The PMD670, the R-1, and
just about every other mp3 recorder except the Ripflash lets you choose
the bit rate. I had been hot to buy the R-1 despite its bugs, and was
intending to record at 256kb/s or 320kb/s, but the PMD660 was announced
just as the R-1 started shipping, and it seemed like a nicer unit, so I
held out for it. By all indications the PMD660 was a shrunken down
PMD670 with most of the same features. It just amazes me that they
crippled the mp3 encoder like that.
> > I did some music recordings in mono because I could at the time
> > only get my hands on one microphone. There's such a thing as an
> > XLR Y-cable that lets you plug a single mic into two inputs,
> Not to worry, but I'm surprised that it doesn't have a switch to send
> the single input to both channels. The cassette models do. Perhaps
> they considered this unnecessary since there's a common mono MP3
> format. If you were recording in WAV format, however, and planned to
> burn an audio CD from the recording, if you were recording in mono,
> you'd probably want to do that.
There's a mono/stereo setting in one of the menus, instead of a switch.
It uses one of the mic inputs to make a mono mp3 or wav file. There's no
need to burn 2x the CF space recording the same mono signal on two
channels. You can always convert the mono file to a stereo file on your
computer afterwards, if you need a stereo file.
> The Y-cable trick is often used by journalists to let them set
> different input levels on the two tracks. That way they don't have to
> depend on the limiter if something gets too loud - it's probably safe
> on the lower gain track. Sometimes handy for festival music recording, too.
Very interesting idea! I had meant to include a little rant about ALC
mentioning the possibility of recording multiple channels at once, maybe
with separate mic elements. I hadn't thought of just using the two
stereo channels at different levels. As someone else mentioned, it might
confuse the mp3 encoder, but it should work if recording WAV.
> > - The line-level input uses a 1/8" stereo minijack, which is fairly
> > typical. But there's no way to use that jack as a mic input. That
> > also seems crazy.
>
> Not to someone who wants to use real microphones.
I think this machine is aimed at a market that wants to use both kinds
of microphones.
> What seems crazy to me (and I assume this is the case) is that other
> than swithing in a pad, there's no way to use the XLR inputs as line inputs.
Correct, as far as I can tell. And I presume that switching in a pad
still means putting the signal through an unnecessary preamp stage, adding
further noise and other bad stuff.
> > This recorder is in the same general category as
> > a minidisc recorder or small DAT, and those all use 1/8" mics and
> > most of the mics made for such devices have 1/8" plugs.
>
> My portable DAT recorders has XLR inputs.
Yeah, I was thinking of the Walkman-style portable DAT, PCM-M100 or whatever.
> The only thing I dislike about my Jukebox 3 is that it has only a mini
> phone jack. But then I've never owned a typical "Minidisk recorder
> microphone."
You've mentioned that and it was one of the things that influenced me to
buy this model (thanks!).
> If you want to play with the pros, get a more "pro" mic. You'd be
> surprised at how your recordings will improve, and it's not just
> because of the XLR connector.
I might do that eventually, though I'm probably not up for another big
expenditure on this stuff real real soon. I'm actually trying to resist
attempting high quality sound, since besides the expense, it mean I'll
have to start paying attention to mic placement (i.e. run around putting
the mic in front of performers) and recording level adjustments. What I
really want is something with either good ALC or very low noise and
24-bit A/D's, so I can set it up once, press a button, and record a whole
evening without having to touch the machine again, even if that means
taking some sonic compromises.
I'm already on Plan B using the Sony mic since I'd hoped to rely entirely
on the internal mics. But they just don't sound so good (the R-1's have
gotten favorable comments by comparison), and of course whenever I move
or adjust the unit, the mics pick that up.
> However, given the MP3 limitations and the media capacity constraints
> when recording in WAV format, it may not be worth completing the
> upgrade with a $500-ballpark stereo mic such as the Rode NT-4 or
> somewhat less expensive but not trivial Audio Technica AT-825.
Hmm, the AT is $350 and the Rode is $450 at bhphoto.com. Is the Rode
significantly better than the AT? I think I saw the Rode at Guitar World
and it looked nice. I didn't get a chance to actually try it.
I've also been playing with the idea of getting a couple of PZM mics
and tossing them out on the carpet in the middle of the room. Is that
another reasonable approach? Radio Shack used to make some PZM's that
were fairly cheap and that got good reviews.
Finally, someone on rec.audio.pro mentioned the Crown SASS-P a while
back, a pretty cumbersome contraption that's something like two PZM mics
in a binaural-like setup inside a huge enclosure. I've seen those go on
ebay in the $500 range. I haven't seriously considered bidding on one
but it's an interesting idea to fool around with.
> > That means transfers are probably slow.
> > But this is just an unconfirmed suspicion for now.
>
> I use USB to transfer recordings from my Jukebox to my laptop
> computer. For practicality (since I then put them on to CD) I start a
> new file every hour, and it takes about 20 minutes to transfer an
> hour's worth of 16/44 WAV file (a bit over 600 MB) over USB 1.1. They
> tell me that Firewire (which the Jukebox also accommodates) is a lot
> faster but I've found that it about halves the transfer time from USB,
> that's all.
Yucch, I think it's best to use a card reader or pc card adapter instead
of USB in this case. I just have to figure out the PMD660's virtual
editing stuff, which does seem like a useful feature. It lets you
(if I understood it right when I flipped through the manual briefly)
play back recordings on the unit and insert edit points during playback,
and then the recorder uploads the big file as a bunch of separate small
files.
> > - There's a pushbutton that lets you make "edit marks" ...
>
> No, it's something that's understood by the recorder so that when
> playing back recordings on the recorder, you can locate your marked
> points. A utility that reads this and opens a text editor so that you
> can annotate your recording would indeed be a cool thing, but I'll bet
> they didn't think about it.
Hmm, I had the impression from the manual that the CD has some software
that understands that file. If I figure out the format and write a
utility, I'll post something about it. I figure I'd want to just
it to split the recording into separate files at the edit points.
> I know the feeling. I ususlly end up taking things like that back and
> waiting to see what else comes along. But if you decide to keep it, do
> think about getting a mic worthy of the recorder insted of using the
> one you have now. Get a mic that's good enough to use with your NEXT
> recorder so you won't have to upgrade again too soon.
Thanks, that sounds like good advice. I keep thinking about making my
own recorder. An acquaintance of mine made a playback-only gizmo in an
Altoids tin, which is pretty neat:
http://www.ladyada.net/make/minty/index.html
a recorder would be more complicated but would still be all off-the-shelf
parts and software, so it's not a completely crazy notion.
> [Rich Brown]
> Paul Rubin, you mention the Neuros Audio Computer as one of the
> contenders when you started shopping. I've been intrigued by that
> model. Can you explain why it didn't make the cut?
Well, it seems like a nice playback unit, but recording was sort of an
afterthought. It has no mic preamps (line input only) so I'd need an
outboard one or an amplified mic. It has no level meters (or at least no
satisfactory ones) so soundprofessionals.com advises buying an outboard
one. The internal rechargeable battery is non-removable so you have no
choice other than using an external pack if you want to record more than
a few hours. I simply bought a 40-pack of cheap alkalines to run the
PMD660 for Consonance since I didn't have time to put a good rechargeable
solution together. That's not possible with the Neuros. So you end up
with a big pile of junk plugged together. And I saw a number of user
forum posts saying that there were sometimes skips in the recordings.
I was very happy with my old Sony recorder and the Marantz looked made in
a similar spirit, at least in the picture. Nice big tape-recorder like
buttons to do recording with, and not a lot of useless stuff like
recording FM broadcasts and matching them up with databases.
As mentioned though, if I send back the PMD660, I don't know what I'd get
instead, and the Neuros would still be a possibility.
> [John in Detroit]
> I recently switched from MD to HI-MD for con-recording (and other
> recording) and I'll post my comments My HI-MD is an MZ-NHF800
>
> Media at 3 bucks a pop from J&R Music New York is 1 gig, holds about
> 1.5 hours of PCM (Wav) recording, MDs use a 20 bit depth if memory
> serves so it is better than CD PCM recordings may then be transfered
> to computer and converted to WAV with no loss,
Hmm, that's better than I'd heard. I thought you could only transfer
recordings out the analog port in real time. Also, 1 gig PCM is just 1.5
hours per disc, right? I wanted to record for 6 hours minimum nonstop.
256kb/sec ATRAC could easily do that, but then there's the question
of how to upload and convert.
I keep hearing Sony is going to face the music (no pun intended) and
put MP3 recording capabilities into Minidisc. MD will get much more
interesting then, but I've stayed away from ATRAC.
> This is about as good as it gets folks, this is CD or better
> quality. Note, there are better recording methods but if your goal is a
> CD.. Why spend the money.
My goal is files on a computer, preferably in FLAC or Vorbis format,
but for now I'm ok settling for mp3.
> Battery life (It uses one AA either alkaline or Rechargeable) is
> advertised at about 1 disk per battery, Unit will test the battery
> when you press RECord and tell you "Not enough battery"
>
> Expierence was 2 to 3 disks per battery (Alkaline)
That's also better than I expected (that's recording in WAV mode? Does
ATRAC recording use more battery power?)
> If you record in ATRAC mode you can put up to 35 hours on one disk,
> however the quality is what they call "Voice Quality" At If you
> record in Hi-MD mode at standard quality it's as good or better than
> my MZ-R55. HOWEVER though you can transfer the recording to the
> computer (Using SOnic Stage software) You can not convert it to wav.
Ugh, that sounds intolerable. I just refuse to use anything that can't
at least convert to wav.
> I have, however, played the disc into the recorder and used Total
> Recorder to capture the digital stream on it's way to the
> digital-analog converters and saved the resulting file to disk..
Yeah, but that means 1 hour to transfer an hour of audio, plus you have
to mess with special hookups and software. That's more of what I wanted
to avoid with the PMD660. Just plug in the CF card and boom, you can
transfer an hour of mp3 in about 10 seconds, or an hour of WAV in a
couple of minutes. No software needed.
> That works.. And T.R. Professional version will auto-split recordings
> so you can have all sorts of fun (Not all programs like 5 gig-a-byte
> files, in fact FAT-32 does not support files that large)
True, FAT32 has a 4GB file size limit which is about 6 hours. That's
not an obstacle for conventions (just press the button to go to a new
file now and then) but I'd like to use the line input to digitize my
analog cassette collection. I bought a Sony MTL-10 cassette changer
(autoreverse deck with 10 tape autoloader magazine, radio stations used
to use these) so that would be 15 hours for 90 minute tapes. I'll have
to do smaller batches if I use WAV. Or else I'll just get a decent
A/D converter for my computer (the Griffin iMic turns out to be very
noise sensitive).
> And I like MD Disks much better than flash ram,,, Though I do admit it
> would be nice to have a couple of gigs in a flash box so I could do
> faster transfers (Takes an hour to transfer an hour's recording into
> this laptop.. Faster on the big box in the basement)
Flash works pretty well if you don't mind the cost. Right now a 4GB
card is about $300 and a 6GB Microdrive is $250 (the PMD660 does support
microdrives). I just have a 1GB and a 512MB right now (I had a second
512MB but sold it) and that's enough to do a whole con. I'll probably
buy a 2GB card sometime before the next con. The prices keep dropping
so I'll wait til the last minute.
Mike Rivers
In article <7xd5tb3...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> > Or are you figuring on recording on the card during the day and then
> > dumping it to your computer overnight
>
> Well, if I want to dump to a computer, it shouldn't take overnight.
Maybe not, but overnight might be the best way to do it. When I'm done
with a day at a festival, I sure don't want to stay up even another
fifteen minutes babysitting a computer if I don't have to.
> slow 1GB CF card uploads through a plain PCMCIA adapter in about 11
> minutes. A fast CF card and a Cardbus adapter should get it down to
> maybe 3 minutes per gigabyte. There's exhaustive speed tests for
> different CF card combinations at www.robgalbraith.com (pro digicam
> site).
My problem with this is that you have to do it at all. I'd rather take
home a pocket full of flash cards and put them on the shelf like I
used to do with tapes, but they're too expensive for that, at least at
this time. And whether I wanted to recycle the memory cards or keep
them on the shelf, I'd still feel an obligation to fairly promptly
transfer them to a medium that has a better chance of being read 50
years from now. I'm not talking about the bits falling out of the
memory card in 50 years, I'm talking about the availability of a slot
to put the memory card in in order to read it. The digital camera that
I bought a bit over a year ago uses xD ("the latest type") memory
cards. How many of those do you see for sale now? Fortunately I was
able to pick up a couple at blowouts, and find a USB reader for them.
Once burned, twice shy I say.
> They are small hard drive enclosures with CF slots and battery packs, so
> you can upload your CF card to disc in the field.
Those are pretty cool, but it's one more thing to buy, maintain, and
replace when it fails.
> I do sleep at cons, but usually only when I'm falling over from
> exhaustion, which means if I have batteries to recharge, I won't remember
> to do it before crashing, so I have dead batteries the next day.
I know the feeling, which is why I don't want to have to dump memory
every day, too. (except mine)
> Nowadays there are several AA NiMH systems on the market that recharge in
> 30 minutes or even 15 minutes, which make recharging during a session
> feasible. So I can consider that but would rather keep things
> streamlined and avoid the need for it.
There are also power outlets in most civilized places. If I'm going to
be recording in one place for half an hour I'll try to find one.
> Yeah, the noise specs aren't so great, it's kind of sad. I'd think
> they could use better stuff without increasing costs too much.
It could be a result of using low power parts (they're cheap because
they're designed for cell phones) and insufficient shielding.
> 4 hours on D's is terrible, you should be able to do much better.
> Have you tried rechargeable D's?
Didn't you say 4 hours on D cells for the Marantz, or did I misread or
misremember that?
> Oh well. I think they built this model to be able to sell it in the $200
> range, so there's a huge markup right now that they're getting because
> there's so little competition. In terms of hardware per dollar it
> doesn't seem like a very good value right now. I paid an early adopter
> premium but I expect prices will drop just like with any other computer
> stuff.
The Marantz recorders don't have a history of prices dropping.
Understand that you're using something that's far from a high demand
item no matter how many people you see in your circle who are
recording live events. The market is in cheap players that are
mediocre (if at all) recorders. I can't imagine this model ever
selling for $200.
> Yes, there's 3 presets and the unit does keep the settings between
> battery stations. Switching between the presets involves a little
> bit of button twiddling. It's not too bad as long as you remember
> what settings the presets correspond to. Otherwise you have to scroll
> through the menus to see how everything is set.
What are the nature of the presets? Is three enough to cover your
bases for most of the recording that you do? Is one enough cover most
of your situations? And does it come back to the last-used preset when
you power it up, or do you always have to check which preset is active
and change it if it isn't the one you want?
> I had been hot to buy the R-1 despite its bugs, and was
> intending to record at 256kb/s or 320kb/s, but the PMD660 was announced
> just as the R-1 started shipping, and it seemed like a nicer unit, so I
> held out for it.
I'm holding out for the Edirol R-4. I don't anticipate the need for
four channels, but I like the size (big enough so it doesn't fall off
the table), the mic inputs, and the internal hard drive. I haven't
investigated battery life yet.
> There's a mono/stereo setting in one of the menus, instead of a switch.
> It uses one of the mic inputs to make a mono mp3 or wav file. There's no
> need to burn 2x the CF space recording the same mono signal on two
> channels.
There is, if you can record the two channels at different levels to
allow for varying source level. But no point to this if, as you
explained with the L+R/L-R recording, it doesn't afford any protection
against overloads. You should experiment with this.
> Very interesting idea! I had meant to include a little rant about ALC
> mentioning the possibility of recording multiple channels at once, maybe
> with separate mic elements.
ALC!!!! ARGH!!! Hopefully you can turn it off.
> > What seems crazy to me (and I assume this is the case) is that other
> > than swithing in a pad, there's no way to use the XLR inputs as line inputs.
>
> Correct, as far as I can tell. And I presume that switching in a pad
> still means putting the signal through an unnecessary preamp stage, adding
> further noise and other bad stuff.
Although in principle you're correct, well designed mic preamps
shouldn't add extra noise and other bad stuff. Many mixers that we all
use send their line inputs through the mic preamp stage. It's not all
that bad, unless the preamps are really bad. But from what you say,
they might be, in which case it would be advantageous to bypass them
entirely.
> I might do that eventually, though I'm probably not up for another big
> expenditure on this stuff real real soon. I'm actually trying to resist
> attempting high quality sound, since besides the expense, it mean I'll
> have to start paying attention to mic placement (i.e. run around putting
> the mic in front of performers) and recording level adjustments.
To be honest, I have made recordings of informal jam sessions and
concerts and I never listen to them other than perhaps once for
reference - to get the words to a song, or learn a tune. They're never
really enjoyable listening. If you want to make a recording that
you'll enjoy hearing again and again, you have to pay attention to all
of those details, and sometimes that's impossible (or at best requires
that you be obnoxious and interruptive to the audience some times).
Better that those things are pre-arranged, even if it means taking a
mix from the PA console (if there is one).
> What I
> really want is something with either good ALC or very low noise and
> 24-bit A/D's, so I can set it up once, press a button, and record a whole
> evening without having to touch the machine again, even if that means
> taking some sonic compromises.
It depends a lot on the situation. ALC isn't the answer, but setting
the level intelligently and looking at it now and then to see if it's
creeping upward (it usually does early on) or downward (as the night
wears on, people drift away, and energy levels are lower). But levels
don't vary more than about 20 dB or so. After all, if you can't hear
with your ears, there's not much point to being there. And that's not
too much dynamic range to accommodate with decent modern gear.
> > upgrade with a $500-ballpark stereo mic such as the Rode NT-4 or
> > somewhat less expensive but not trivial Audio Technica AT-825.
> Hmm, the AT is $350 and the Rode is $450 at bhphoto.com. Is the Rode
> significantly better than the AT?
Honestly, I don't know. The A-T is smaller and looks less obtrusive
when set up, which may be better for your situation. And it doesn't
suck.
> I've also been playing with the idea of getting a couple of PZM mics
> and tossing them out on the carpet in the middle of the room. Is that
> another reasonable approach? Radio Shack used to make some PZM's that
> were fairly cheap and that got good reviews.
The Radio Shack PZMs aren't what they used to be. I don't think this
would work very well.
> Finally, someone on rec.audio.pro mentioned the Crown SASS-P a while
> back, a pretty cumbersome contraption that's something like two PZM mics
> in a binaural-like setup inside a huge enclosure. I've seen those go on
> ebay in the $500 range. I haven't seriously considered bidding on one
> but it's an interesting idea to fool around with.
That's very different than a couple of PZMs on the floor and it does
work fairly well, but it's big and ugly, hard to hide, and it comes in
a large attache case.
> I just have to figure out the PMD660's virtual
> editing stuff, which does seem like a useful feature. It lets you
> (if I understood it right when I flipped through the manual briefly)
> play back recordings on the unit and insert edit points during playback,
> and then the recorder uploads the big file as a bunch of separate small
> files.
File structure of digital recorders is something that's far from
standard. You'll just have to sort that out. I doubt that it's well
documented from the manufacturer, but user support forums pop up for
things like this and you might find someone who's already done the
work. Maybe it'll be you.
> Thanks, that sounds like good advice. I keep thinking about making my
> own recorder. An acquaintance of mine made a playback-only gizmo in an
> Altoids tin, which is pretty neat:
>
> http://www.ladyada.net/make/minty/index.html
That's not exactly a casual project but it proves that there are
people out there who are capable of this sort of thing.
> > I recently switched from MD to HI-MD for con-recording (and other
> > recording) and I'll post my comments My HI-MD is an MZ-NHF800
> Hmm, that's better than I'd heard. I thought you could only transfer
> recordings out the analog port in real time.
I've read enough about the High-MD to understand that you can use
special (comes with it) software to transfer audio as files from the
recorder to a computer. It converts to a PC-usable format (WAV or MP3)
on the fly. But the rub is that after making the transfer (I think you
get two shots, so you can make a backup copy) it deletes the original
recording on the disk. So it's not always practical to store the disk
in an archive and work with the version that you've transferred to the
computer. Of course you can make unlimited real-time transfers as long
as the MD player works.
Mike Rivers
> > The Y-cable trick is often used by
> > journalists to let them set different input levels on the two tracks.
> That trick won't work if the recorder is internally using L+R and L-R as
> its two recorded tracks.
This recording method is a new one on me. What's the logic behind
this? L+R for a reasonably balanced stereo input is going to be 3 to 6
dB higher than each channel individually. Why lose 6 dB of gain (or
gain structure) in the process? This is all low voltage, low power
stuff which requires a low internal signal level in order to achieve
usable headroom. Sounds like a design compromise to me, but what does
it make better?
Paul Rubin
> This recording method is a new one on me. What's the logic behind
> this? L+R for a reasonably balanced stereo input is going to be 3 to 6
> dB higher than each channel individually. Why lose 6 dB of gain (or
> gain structure) in the process? This is all low voltage, low power
> stuff which requires a low internal signal level in order to achieve
> usable headroom. Sounds like a design compromise to me, but what does
> it make better?
I would expect that L+R is computed digitally by adding the two A/D
outputs together, so there's no noise issue.
Paul Rubin
> > Well, if I want to dump to a computer, it shouldn't take overnight.
>
> Maybe not, but overnight might be the best way to do it. When I'm done
> with a day at a festival, I sure don't want to stay up even another
> fifteen minutes babysitting a computer if I don't have to.
True, I understand that very well ;-). Of course with a second CF
card, it's possible to dump one of them to a computer while recording
to the other one, if you don't mind messing with a computer or
disk-storage-gizmo during a recording session.
> My problem with this is that you have to do it at all. I'd rather take
> home a pocket full of flash cards and put them on the shelf like I
> used to do with tapes, but they're too expensive for that, at least at
> this time. And whether I wanted to recycle the memory cards or keep
> them on the shelf, I'd still feel an obligation to fairly promptly
> transfer them to a medium that has a better chance of being read 50
> years from now.
Sure, I'm the same way, and in fact part of what got me into this
digital recording stuff was the experience of having lost a collection
of old cassette tapes that I made years ago at such events. The
digital recordings I make now get loaded to my computer and backed up
like my computer files. In a good system (I'm aiming for this but
don't really have it yet) that means there's frequent backups, stored
at a site far away from where the computer is, etc. As it is now, at
least once I've uploaded the CF to the computer, I try to keep it so
that the files always exist on two separate storage devices (i.e. I
don't erase the CF card until I've copied the files from disk to CD-R
or something).
> I'm not talking about the bits falling out of the memory card in 50
> years, I'm talking about the availability of a slot to put the
> memory card in in order to read it. The digital camera that I bought
> a bit over a year ago uses xD ("the latest type") memory cards. How
> many of those do you see for sale now? Fortunately I was able to
> pick up a couple at blowouts, and find a USB reader for them. Once
> burned, twice shy I say.
Eh? xD is still in production and easy to find:
http://www.zipzoomfly.com/jsp/ProductList.jsp?ThirdCategoryCode=012815
and they're still designing it into new cameras. Generally though,
multi-vendor standard media that have controllers in the cards instead
of in the camera (that means CF and SD) will tend to stay around
longer than controllerless media (SmartMedia, xD) or proprietary media
(Memory Stick).
> There are also power outlets in most civilized places. If I'm going to
> be recording in one place for half an hour I'll try to find one.
Yeah, I could do that most of the time, but I decided I wanted to
avoid creating such a burden on the room. There's people walking in
and out all the time and more cords for them to trip over isn't so
good; it means I'd have to find particular spots to sit in, etc.
Also, there are other people who need those seats and outlets more
than I do (e.g. performers with electric keyboard instruments). I'm
not trying to be "stealth" but I do want to be "low impact". I even
wanted to avoid using an external microphone (more cables) but I'm
going to have to concede on that one.
> > 4 hours on D's is terrible, you should be able to do much better.
> > Have you tried rechargeable D's?
>
> Didn't you say 4 hours on D cells for the Marantz, or did I misread or
> misremember that?
Sorry, I misread you, I thought you were saying 4 hours on D's for
your NJB3. The Marantz is spec'd for 4 hours on four AA's and that
spec appears quite conservative (at least if I don't use phantom
power). With high capacity NiMH AA rechargeables it should be well
over 6 hours, maybe more like 8 hours. At that point maybe I don't
need an external D pack so much, so that makes things easier. But a D
pack should last 20+ hours.
I now think I might try making an external 4AA or 4AAA pack instead.
The recorder supposedly automatically switches to internal power if
the external power gets yanked. And it can operate on external power
with no batteries in the unit. So my idea is to plug the small
external pack in when the internal batteries start getting low. Then
with the pack plugged in, I can change the internal batteries without
interrupting the recording. Then I can unplug the external pack again
and put it back in my pocket. The external pack would be tiny, since
it would only be used for very brief intervals while changing the
internal batteries every 6 hours or so. And it would be out of the
way (i.e. in my pocket) almost all the time. I still haven't had a
chance to go to Fry's and look for the power connector but I will try
to do it soon.
> The Marantz recorders don't have a history of prices dropping.
> Understand that you're using something that's far from a high demand
> item no matter how many people you see in your circle who are
> recording live events. The market is in cheap players that are
> mediocre (if at all) recorders. I can't imagine this model ever
> selling for $200.
I think it's inevitable, whether it's this model's price dropping or a
new, less expensive model replacing it. The PMD690 is already
discontinued and the current PMD670/671 are less expensive than the
PMD690 was and they do more. I think there just has to start being
low cost decent recording devices; look how many people bought the
WM-D3 and are buying Minidisc recorders now. As flash memory keeps
getting cheaper, we're sure to see PMD660-like recorders (ok, maybe
without XLR jacks) at Minidisc-like prices.
> > Yes, there's 3 presets and the unit does keep the settings
>
> What are the nature of the presets? Is three enough to cover your
> bases for most of the recording that you do? Is one enough cover most
> of your situations? And does it come back to the last-used preset when
> you power it up, or do you always have to check which preset is active
> and change it if it isn't the one you want?
The unit's configuration includes a dozen or so settings chosen from
menus. That includes stuff like the sample rate and recording format,
the battery type, the input selection, etc. Each preset holds a
complete set of choices. At Consonance I did find myself changing
stuff around, but as I get more settled in with the machine I expect I
won't have to change it as much. Yes, it does go back to the
last-used preset on power-up, even after you change batteries, but
there's the still the matter of remembering which preset that is (or
twiddling buttons and menus to check).
> I'm holding out for the Edirol R-4. I don't anticipate the need for
> four channels, but I like the size (big enough so it doesn't fall off
> the table), the mic inputs, and the internal hard drive. I haven't
> investigated battery life yet.
I'll be interested to know what you think of it. Having 4 channels
would also be interesting. Yes I think they should make a hard drive
version of the PMD660 (microdrive doesn't count). They put hd's in
devices like iPods which are a lot smaller.
> > Very interesting idea! I had meant to include a little rant about ALC
> > mentioning the possibility of recording multiple channels at once, maybe
> > with separate mic elements.
>
> ALC!!!! ARGH!!! Hopefully you can turn it off.
My ALC rant was intended to be about how ALC provokes "ARGH!!" only
because the existing implementations are done badly, and they should
just get with the program and do it right. Almost all cameras are
autoexposure and autofocus now, almost all cars are available with
automatic transmissions, etc., but to make a good recording I still
have to sit around looking at level meters and twiddling knobs like a
dork. Something just seems broken about that. I've actually
entertained the notion of getting a high class preamp with a 24-bit
A/D (Core Sound 2496) just to be able to set the gain high enough to
record the quiet stuff and rely on the 24 bit headroom to not have to
worry about overload. Then I'd record everything at the same level
and adjust with a computer afterwards ("two-pass ALC"). Of course
using a 4 channel recorder using your trick of high levels on one pair
of channels and low levels on the other pair, might work even better.
Anyway, yes, you can turn off the Marantz's ALC, and in fact it's off
by default and you need to flip through numerous menu choices to turn
it on.
> Although in principle you're correct, well designed mic preamps
> shouldn't add extra noise and other bad stuff. Many mixers that we all
> use send their line inputs through the mic preamp stage. It's not all
> that bad, unless the preamps are really bad.
Hmm, interesting. There is a 20 dB pad in the unit that you can turn
on and off through yet another menu choice, if that helps.
> To be honest, I have made recordings of informal jam sessions and
> concerts and I never listen to them other than perhaps once for
> reference - to get the words to a song, or learn a tune. They're never
> really enjoyable listening. If you want to make a recording that
> you'll enjoy hearing again and again, you have to pay attention to all
> of those details,
Hehe, that's because you know what you're doing and so your standards
of good recordings are much higher than mine and I don't mind listening
to dreck ;). Think of the 78 rpm records that people used to listen
to all the time because that's all they had, and how awful they sound
by today's standards.
And of course too, I like the recordings for their personal
significance, like if you go to Paris and take a cliche digicam
snapshot of your friend standing in front of the Eiffel Tower, your
snapshot will still be more meaningful to you than some professionally
done Eiffel Tower photo that's far better both artistically and
technically.
But you're right that I don't tend to listen to these informal
recordings over and over the way I'd listen to something really well
done. My tendency has been after an event to find some memorable
parts of the recordings and listen to those several times, and archive
the rest, and every now and then go back to some old recording and
listen to it again to remember it better. I mostly wouldn't play them
in the car for a commute or anything like that. I still get
satisfaction from making them and listening to them, so I keep doing it.
> Better that those things are pre-arranged, even if it means taking a
> mix from the PA console (if there is one).
For the most part there's no PA console in the sessions that interest
me the most (open circles).
> > What I really want is something with either good ALC or very low
> > noise and 24-bit A/D's, so I can set it up once, press a button,
> > and record a whole evening without having to touch the machine
> > again, even if that means taking some sonic compromises.
>
> It depends a lot on the situation. ALC isn't the answer, but setting
> the level intelligently and looking at it now and then to see if it's
> creeping upward (it usually does early on) or downward (as the night
> wears on, people drift away, and energy levels are lower). But levels
> don't vary more than about 20 dB or so.
It's much worse than that at these sessions, and I think there's more
than 20 dB of difference in levels. That method would make sense at a
normal concert, where there's a stage with performers on it so the mic
is always aimed at the same place. In these sessions, typically
there's a bunch of people in a large room. Somebody (maybe the person
3 feet away from me) plays a song, perhaps a song about dead parrots.
Then someone 20 feet away on the other side of the room happens to
know another song about dead parrots, so she jumps in after the first
one finishes and plays her song, and it goes around like that at
random. And the 3-feet-away person might be a loud performer while
the 20-feet-away one might be rather quiet (shy person, etc.), maybe
to the point of having to strain to hear her even in the room, but her
lyrics are sheer genius and so they need to be intelligible in the
recording.
I've made lots of ALC recordings of these things with an Aiwa Walkman
cassette recorder and more recently with a camcorder and the results
have always been listenable. I've done somewhat worse with the ALC in
the Marantz, but I need to experiment with it some more.
> > Hmm, the AT is $350 and the Rode is $450 at bhphoto.com. Is the Rode
> > significantly better than the AT?
>
> Honestly, I don't know. The A-T is smaller and looks less obtrusive
> when set up, which may be better for your situation. And it doesn't suck.
Hmmm again. I see by the specs that the Rode is around 9 dB more
sensitive than the A-T, so maybe that's helpful for recording quiet
material with the Marantz's noisy preamp. Does that make sense?
> The Radio Shack PZMs aren't what they used to be. I don't think this
> [putting them in the middle of the room] would work very well.
Hmmm #3. How about with better PZM's?
> > Finally, someone on rec.audio.pro mentioned the Crown SASS-P...
> That's very different than a couple of PZMs on the floor and it does
> work fairly well, but it's big and ugly, hard to hide, and it comes in
> a large attache case.
Yeah, I'm not concerned about hiding it, but just lugging it around is
more hassle than I really want to deal with.
> I've read enough about the High-MD to understand that you can use
> special (comes with it) software to transfer audio as files from the
> recorder to a computer. It converts to a PC-usable format (WAV or MP3)
> on the fly. But the rub is that after making the transfer (I think you
> get two shots, so you can make a backup copy) it deletes the original
> recording on the disk. So it's not always practical to store the disk
> in an archive and work with the version that you've transferred to the
> computer.
Arggh, that's crazy, I'd never buy anything like that. Maybe they'll
couth up someday. Thanks.
Btw, I've started webifying the review:
http://www.nightsong.com/phr/pmd660.html
I plan to add some more stuff to it when I get a chance.
Mike Rivers
In article <7x64z28...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> I would expect that L+R is computed digitally by adding the two A/D
> outputs together, so there's no noise issue.
If you add two full scale values, you'll get an error. So to be sure
that you don't, you have to either leave some headroom when recording
or attenuate the signals before adding.
However, I really have no idea just what this L+R/L-R recording method
really is, how it works, why it works, or what's behind it all. At
this point, it sounds kind of screwy to me and I have a feeling that
it's not as simple as all that.
Paul Rubin
> > I would expect that L+R is computed digitally by adding the two A/D
> > outputs together, so there's no noise issue.
>
> If you add two full scale values, you'll get an error. So to be sure
> that you don't, you have to either leave some headroom when recording
> or attenuate the signals before adding.
Nah, you can just add and put the sum in an internal register that's
wider than the input signal. For example, the Motorola 56000 series
DSP's operates on 24-bit input words but has a 56-bit accumulator,
hence the "56" in the part number. So you can add a 24x24 bit product
into the acculator 256 times before worrying about overflow.
Of course I don't know how the Marantz goes about it.
> However, I really have no idea just what this L+R/L-R recording method
> really is, how it works, why it works, or what's behind it all. At
> this point, it sounds kind of screwy to me and I have a feeling that
> it's not as simple as all that.
MP3 is quite complicated and the bits that come out of the encoder
have no resemblance at all to the input signal until they're decoded.
There's an enormous amount of processing between the input and the
output. But I thought the basic L+R/L-R scheme was familiar in analog
recording as mid/side (M/S) recording, e.g. my Sony MS-907 mic is set
up that way, hence the "MS" in it. Also, stereo FM radio works the
same way.
Joint Stereo shouldn't be confused with "intensity stereo", which is
where the mp3 encoder takes the lowest and highest input frequencies
and converts them to mono to save more bits, based on the idea that
human hearing is not very directional at those frequencies. I just
did a Google search on "joint stereo" and it looks like there's quite
a bit of controversy and confusion over this.
Kurt Albershardt
> In article <pan.2005.04.04...@freemars.org> rab...@freemars.org writes:
>
>
>>> The Y-cable trick is often used by
>>> journalists to let them set different input levels on the two tracks.
>>
>>
>> That trick won't work if the recorder is internally using L+R and L-R
>> as its two recorded tracks.
>
>
> This recording method is a new one on me. What's the logic behind this?
MP3 Joint Stereo uses sum & difference encoding a la FM multiplex.
Wonder if this is what he's referring to?
Rich Brown
> MP3 Joint Stereo uses sum & difference encoding a la FM multiplex. Wonder
> if this is what he's referring to?
Exactly.
--
John in Detroit
It is the format that FM Stereo is broadcast in, The primary chan is L+R
so that a cheep mono-receiver hears everyghing. The sub channel is L-R
(It has been years, Julian would may need to refresh me here but I think
the sub chan carrier is on 19Khz) When decoded in phase and then
properly added you take the main plus sub carrier and you have L+R+L-R
in short 2*L and then you subtract you get L+R-L+R or 2*R and that is
what gets fed downstream in the stereo receiver.
Why one would do this in recording... I haven't a clue.
--
Mike Rivers
In article <7x1x9q8...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> Of course with a second CF
> card, it's possible to dump one of them to a computer while recording
> to the other one, if you don't mind messing with a computer or
> disk-storage-gizmo during a recording session.
If it's a gig and you gotta do it, you gotta do it, but it's one more
thing to multitask. But if you're going to the show for your enjoyment
and you want to bring back a recording, you don't want to burden
yourself with all the technology. The only way you'll enjoy the show
then is listening to your recordings when you get home - and they
probably suck anyway.
> part of what got me into this
> digital recording stuff was the experience of having lost a collection
> of old cassette tapes that I made years ago at such events.
How do you lose a collection of cassettes? Lost in a move? A flooded
basement?
> digital recordings I make now get loaded to my computer and backed up
> like my computer files. In a good system (I'm aiming for this but
> don't really have it yet) that means there's frequent backups, stored
> at a site far away from where the computer is, etc.
Just how valuable is this stuff, anyway? I don't think I'd cry if my
tape collection got burned to a crisp. I enjoyed the shows, and most
of those tapes I haven't listened to in years. It's nice to know that
I have them, but it's hard to get too sentimental.
> Eh? xD is still in production and easy to find:
>
> http://www.zipzoomfly.com/jsp/ProductList.jsp?ThirdCategoryCode=012815
Zipzoomfly? No, I want to stop at the drug store on a Saturday
afternoon to pick up some "film" for my camera. I don't want ot have
to buy it on line. It's practically not in stores any more.
> There's people walking in
> and out all the time and more cords for them to trip over isn't so
> good; it means I'd have to find particular spots to sit in, etc.
> Also, there are other people who need those seats and outlets more
> than I do (e.g. performers with electric keyboard instruments).
So stick an outlet strip in your backpack. And sit where it makes
sense.
> Sorry, I misread you, I thought you were saying 4 hours on D's for
> your NJB3. The Marantz is spec'd for 4 hours on four AA's and that
> spec appears quite conservative (at least if I don't use phantom
> power).
Well, at least that's a little lighter than 4 D cells. The Jukebox
uses a rechargeable battery. I've never run it down so I don't know
how long it lasts, but I've recorded for more than six hours on a
charge and played for over ten. And you can plug in a second one for
double the capacity.
> > The Marantz recorders don't have a history of prices dropping.
> I think it's inevitable, whether it's this model's price dropping or a
> new, less expensive model replacing it.
I suspect the latter, and it might be a few bucks less than yours, but
with more features.
> The unit's configuration includes a dozen or so settings chosen from
> menus. That includes stuff like the sample rate and recording format,
> the battery type, the input selection, etc. Each preset holds a
> complete set of choices. At Consonance I did find myself changing
> stuff around, but as I get more settled in with the machine I expect I
> won't have to change it as much.
You settle on a sample rate, mic inputs, batteries, and so on, and you
can probably use that setting nearly all the time.
> My ALC rant was intended to be about how ALC provokes "ARGH!!" only
> because the existing implementations are done badly, and they should
> just get with the program and do it right.
But nobody does it right, and it's really a physical impossibility.
Just say "no" to ALC. You don't have to continually twiddle with the
knobs, though. That's a common mistake. The volume of music goes up
and down, so your recording should, too. Some people sing louder than
others so you set the level to accommodate the source and for the most
part you can leave it alone for the duration of the set. ALC will
bring up the level of the background noise when he's singing or
talking quietly. You would, too, if you turned up the recording
volume. That's why you don't do it.
> Hehe, that's because you know what you're doing and so your standards
> of good recordings are much higher than mine and I don't mind listening
> to dreck ;). Think of the 78 rpm records that people used to listen
> to all the time because that's all they had, and how awful they sound
> by today's standards.
I love listening to 78 RPM records. They might have background noise
but the music is so much better than a lot of the jam sessions that
are fun when you're part of them.
> It's much worse than that at these sessions, and I think there's more
> than 20 dB of difference in levels. That method would make sense at a
> normal concert, where there's a stage with performers on it so the mic
> is always aimed at the same place. In these sessions, typically
> there's a bunch of people in a large room. Somebody (maybe the person
> 3 feet away from me) plays a song, perhaps a song about dead parrots.
> Then someone 20 feet away on the other side of the room happens to
> know another song about dead parrots, so she jumps in after the first
> one finishes and plays her song, and it goes around like that at
> random.
There's really no way you can record a session like that very well if
you're sitting on the circle. An omni mic suspended from above can
work, but you don't want to do that. It spoils the vibe. This is an
instance where a PZM on the floor near the center of the circle might
actually work out.
> Hmmm again. I see by the specs that the Rode is around 9 dB more
> sensitive than the A-T, so maybe that's helpful for recording quiet
> material with the Marantz's noisy preamp. Does that make sense?
Definitely. As long as the specs aren't lying to you. You need to be
careful about that. It's often hard to compare spec sheets because
there are different ways of measuring the same things.
> Hmmm #3. How about with better PZM's?
Better is always better. You can't go wrong with the genuine article
(Crown) who used to make the PZMs for Radio Shack.
Mike Rivers
> MP3 Joint Stereo uses sum & difference encoding a la FM multiplex.
> Wonder if this is what he's referring to?
Sounds like it, but why? Is that to give the MP3 encoder a good mono
signal to work with, for some reason?
Mike Rivers
In article <7xsm266...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> MP3 is quite complicated and the bits that come out of the encoder
> have no resemblance at all to the input signal until they're decoded.
> There's an enormous amount of processing between the input and the
> output. But I thought the basic L+R/L-R scheme was familiar in analog
> recording as mid/side (M/S) recording, e.g. my Sony MS-907 mic is set
> up that way, hence the "MS" in it. Also, stereo FM radio works the
> same way.
Yes, I understand the principle, but since I've only recorded using
M-S with analog systems where the principles of headroom are a little
different than digital (and where I have complete control of levels)
I've never pushed it to the limit.
> Joint Stereo shouldn't be confused with "intensity stereo", which is
> where the mp3 encoder takes the lowest and highest input frequencies
> and converts them to mono to save more bits, based on the idea that
> human hearing is not very directional at those frequencies.
It's not all that directional at low frequencies, but you have to get
pretty high before you lose a sense of directionality because of
reflections. "Joint stereo" sounds like one of those newfangled things
that I'd probably best avoid by avoiding the things (such as MP3
encoding) that require or promote it.
Mark
Mike Rivers wrote:
> In article <3be99tF...@individual.net> ku...@nv.net writes:
>
> > MP3 Joint Stereo uses sum & difference encoding a la FM multiplex.
> > Wonder if this is what he's referring to?
>
> Sounds like it, but why? Is that to give the MP3 encoder a good mono
> signal to work with, for some reason?
>
>
MP3 encoder uses to make the best use of the available bits bandwidth.
Since most programs material has a large mono content, the encoder
allocates more bits to the sum signal and less bits to the difference
signal instead of having to allocate 1/2 the bits to the left and 1/2
the bits to the right. Joint stereo is supposed to give better overall
results i.e. less compression artifacts because there are more bits
available for the predominantly L+R signal.
Once it comes out of the decoder, joint stereo is (ideally) the same as
the individual L and R that went in.
Mark
John in Detroit
to know why
Thanks... Please continue this thread, I doubt I'll contribute as it's
well beyond me but I'm learning from it so please continue
--
Paul Rubin
> Mike and others...... I knew I prefered MD to MP3, but ... I'm
> starting to know why
ATRAC3 (like AAC, like Vorbis) is a more modern and generally better
compression scheme than MP3, but like MP3, it almost certainly uses
joint stereo or something similar. Even FLAC uses it and FLAC is
lossless, so clearly joint stereo doesn't necessarily do anything bad
to the sound. There is, however, the question of exactly when to use
it (MP3 turns it on and off on a frame by frame basis) and how much
bandwidth to assign to the mid channel and how much to the side
channel. In other words, it adds some more parameters that the
encoder designer has to get right, and getting them wrong can mess up
the sound. So, the criticisms of it (where valid) have to mainly come
from instances where it wasn't done with enough discernment.
Paul Rubin
> It's not all that directional at low frequencies, but you have to get
> pretty high before you lose a sense of directionality because of
> reflections. "Joint stereo" sounds like one of those newfangled things
> that I'd probably best avoid by avoiding the things (such as MP3
> encoding) that require or promote it.
Yeah, I'd rather avoid MP3 too, but recording 30 hours with no
compression would require a hard drive and its attendant noise and
power consumption. I was lured by the siren call of being able to
record 30 hours on fairly small quantities of batteries and also of
having a noiseless machine (one that could make good recordings on
internal mics since there's no motor noise). The Edirol R-1's
internal mics have gotten favorable comments but the PMD660's are
probably not as good. The internal mic plan I think turns out not
to be workable, so I'm already resigned to plan B.
Meanwhile, if of any interest, Marantz's new 24/96 CF recorder (the
PMD671, follow-on to the PMD670) is $799 at soundprofessionals.com.
It's about 2x the 660's size and appears much better than the 660 in
just about every way. I missed the end of my 30-day PMD660 return
period (there's always ebay) but otherwise I'd be tempted to pay the
extra $300 and upgrade. The PMD660 is clearly a stopgap machine but a
PMD671 will stay useful for a long time, sort of like the PMD430
cassette deck. The reason I'm not jumping at the PMD671 is that with
a machine that big and expensive, I'd want it to have a hard drive.
Joe Kesselman
> However, I really have no idea just what this L+R/L-R recording method
> really is, how it works, why it works, or what's behind it all.
Basically: Rather than two cardioid microphone cartridges pointing
across each other to record your stereo image, use one cardioid pointed
straight ahead and a bidirectional at right angles to it. The "center"
mike is hearing the center of the performance area at the best point in
its pick-up pattern, just like any mono cardiod mike. The "side" mike's
figure-eight pattern doesn't hear what's in the center at all, but hears
increasingly strongly to either side -- with a phase difference (which
is how that center null is achieved).
Adding those together... Center can be thought of as L+R (strongest for
a source in the center; weaker as you go off-axis); side, because of the
phase difference, can be thought of as L-R (or, if you invert the phase,
R-L). By putting these through a summing amplifier, you can reconstruct
R and L to get a stereo image.
The advantage is that by controlling how much of the side signal you mix
back in, you can effectively adjust the pickup pattern of that image
after you get back to the studio, widening or narrowing its area of
greatest sensitivity. To do that with an R/L coincident pair, you have
to adjust the physical placement of the mikes at the time you record...
and if you don't like what you got, and you can't do another take,
tough; you're stuck with it.
I'm not sure what the implications are for bit rate, but at this point I
think we're getting rather far offtopic for rec.music.filk and should
take it either to the audio newsgroup or the production mailing list or
both.
Kurt Albershardt
> Mike Rivers wrote:
>
>> However, I really have no idea just what this L+R/L-R recording method
>> really is, how it works, why it works, or what's behind it all.
>
>
> Basically: Rather than two cardioid microphone cartridges pointing
> across each other to record your stereo image, use one cardioid pointed
> straight ahead and a bidirectional at right angles to it. The "center"
> mike is hearing the center of the performance area at the best point in
> its pick-up pattern, just like any mono cardiod mike. The "side" mike's
> figure-eight pattern doesn't hear what's in the center at all, but hears
> increasingly strongly to either side -- with a phase difference (which
> is how that center null is achieved).
Believe it or not, Mike understands how M-S works. He was just
unfamiliar with how it related to raw data stored on disk in this
particular case.
David G. Bell
mri...@d-and-d.com "Mike Rivers" wrote:
> In article <7x64z28...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
>
> > I would expect that L+R is computed digitally by adding the two A/D
> > outputs together, so there's no noise issue.
>
> If you add two full scale values, you'll get an error. So to be sure
> that you don't, you have to either leave some headroom when recording
> or attenuate the signals before adding.
>
> However, I really have no idea just what this L+R/L-R recording method
> really is, how it works, why it works, or what's behind it all. At
> this point, it sounds kind of screwy to me and I have a feeling that
> it's not as simple as all that.
For one thing, it's much the same as the signal format used in FM
broadcasting. L+R is the mono signal, while L-R is on a sub-carrier
using lower bandwidth.
L = ((L+R)+(L-R))/2
R = ((L+R)-(L-R))/2
That decode is pretty easy in either analogue or digital. You could do
it with Op-Amps and standard circuits, no trouble. (Doing it well could
be harder, but it's a standard broadcast radio decode, so there'll be
dedicated hardware available.)
The advantage is that while the L+R does need the full bandwidth, you
can cut back the L-R bandwidth a lot. Which is good for both bitrates
and RF spectrum use.
--
David G. Bell -- SF Fan, Filker, and Punslinger.
"I am Number Two," said Penfold. "You are Number Six."
Julian Adamaitis
"Paul Rubin" <http://phr...@NOSPAM.invalid> wrote in message
news:7xu0mk4...@ruckus.brouhaha.com...
> John in Detroit <Bla...@sbcglobal.net> writes:
>> Mike and others...... I knew I prefered MD to MP3, but ... I'm
>> starting to know why
>
> ATRAC3 (like AAC, like Vorbis) is a more modern and generally better
> compression scheme than MP3, but like MP3, it almost certainly uses
> joint stereo or something similar.
My understanding is ATRAC Stereo is 2 ch L & R mono, but LP modes use sum
difference for the reasons mentioned with regard to the Marantz.
Julian
David G. Bell
ku...@nv.net "Kurt Albershardt" wrote:
Part of the compression used for MP3 -- the raw data doesn't get stored.
But Joe's post has explained something new to me. We're coming at this
with different knowledge-bases.
Mike Rivers
> Mike and others...... I knew I prefered MD to MP3, but ... I'm starting
> to know why
> >>MP3 is quite complicated and the bits that come out of the encoder
> >>have no resemblance at all to the input signal until they're decoded.
> >>There's an enormous amount of processing between the input and the
> >>output. But I thought the basic L+R/L-R scheme was familiar in analog
> >>recording as mid/side (M/S) recording
After a couple of days of this L+R/L-R recording nonsense, my memory
has been jogged back to my first real MP3 listening experience. I had
just received my Jukebox 3, and before going on a trip, loaded it up
with several CDs, using its supplied file transfer program and
built-in MP3 encoder. One was The Waybacks. I was listening to this on
headphones while on the plane, and noticed than in several places, the
lead acoustic guitar seemed to jump from side to side in the midst of
a solo. These shifts were almost musical - that it, they occurred at
the end of a musical phrase, not at random - but it didn't seem to
make any artistic sense.
I mentioned it on rec.audio.pro as a mixing curiosity, and Bruce
Kaphan, the engineer of that recording, responded that this wasn't at
all how he mixed it, that solos were anchored solidly in place. I went
back and listened to the CD, and sure enough, they were stable. I
figured that this weirdness must just be an anomoly of the MP3
encoding.
If things got combined to mono as part of the data compression
process, and then got separated incorrectly, that could certainly
explain the change in apparrent pan position. It could be happening
all the time and it was just obvious enough to be jarring on a clean
and pure recording of acoustic instruments.
Oh, by the way, is everyone interested in this reading rec.audio.pro?
I'd like to stop cross-posting to rec.music.filk now that the topic has
left "what should I get?" and shifted to "what's wrong with what I got."
John in Detroit
Rec.Music.Filk But do you filk as well?
(I normally see you in alt.audio.minidisc by the way)
Folks here in Filk, Julian knows audio, rather well. I can not comment
on how well Julian knows music but Julian knowns audio.
--
Arny Krueger
> Mike and others......
> I knew I prefered MD to MP3, but ... I'm starting to know why
Except that you probably don't know the reasons why well enough,
unless you think that no choices are better than choices that others
make for you.
Furthermore, MP3 isn't the only widely-used alternative to MD -
there's always AC-3 and AAC.
Arny Krueger
> John in Detroit <Bla...@sbcglobal.net> writes:
>> Mike and others...... I knew I prefered MD to MP3, but ... I'm
>> starting to know why
>
> ATRAC3 (like AAC, like Vorbis) is a more modern and generally better
> compression scheme than MP3, but like MP3, it almost certainly uses
> joint stereo or something similar.
Agreed.
> Even FLAC uses it and FLAC is lossless, so clearly joint stereo
doesn't necessarily do anything bad
> to the sound.
Agreed. In the context of lossless encoding, joint stereo only happens
if the signal is truely mono.
> There is, however, the question of exactly when to use
> it (MP3 turns it on and off on a frame by frame basis) and how much
> bandwidth to assign to the mid channel and how much to the side
> channel. In other words, it adds some more parameters that the
> encoder designer has to get right, and getting them wrong can mess
up
> the sound.
Point being that joint stereo is a reasonble extension of the idea of
perceptual coding.
>So, the criticisms of it (where valid) have to mainly come
> from instances where it wasn't done with enough discernment.
There have been so many implementaions of MP3 by so many people that
MP3 software implenting every possible flaw has been released at least
once. That doesn't mean that the best modern implementations have
every possible flaw.
Mike Rivers
>
> Mike Rivers wrote:
> > However, I really have no idea just what this L+R/L-R recording method
> > really is, how it works, why it works, or what's behind it all.
>
> Basically: Rather than two cardioid microphone cartridges pointing
> across each other to record your stereo image, use one cardioid pointed
> straight ahead and a bidirectional at right angles to it.
Right - I know all about that. I want to know what the benefit of
converting left and right inputs to sum and difference before dropping
out data that's subjectively insignificant is. And why it's still
insignificant when converted back to left and right. But I probably
wouldn't understand the answer. Maybe there's an article in the AES
Journal that I haven't got around to reading yet.
Mike Rivers
In article <7xoecs4...@ruckus.brouhaha.com> //phr...@NOSPAM.invalid writes:
> I was lured by the siren call of being able to
> record 30 hours on fairly small quantities of batteries and also of
> having a noiseless machine
Life is full of tradeoffs. That's yours.
> Meanwhile, if of any interest, Marantz's new 24/96 CF recorder (the
> PMD671, follow-on to the PMD670) is $799 at soundprofessionals.com.
> I missed the end of my 30-day PMD660 return
> period (there's always ebay) but otherwise I'd be tempted to pay the
> extra $300 and upgrade.
Most places don't watch the clock when you're willing to give them
more money. Call them and ask about getting full credit for your 660
toward a 671.
> The PMD660 is clearly a stopgap machine
It's beginning to sound like that's the case, or maybe a machine
that's designed to appeal to those who want "professional" features
but don't want to pay the pro price. Lotsa stuff like that around.
Scott Dorsey
>
>In article <d2vj4c$cps$1...@domitilla.aioe.org> keshlam...@comcast.net writes:
>
>>
>> Mike Rivers wrote:
>> > However, I really have no idea just what this L+R/L-R recording method
>> > really is, how it works, why it works, or what's behind it all.
>>
>> Basically: Rather than two cardioid microphone cartridges pointing
>> across each other to record your stereo image, use one cardioid pointed
>> straight ahead and a bidirectional at right angles to it.
>
>Right - I know all about that. I want to know what the benefit of
>converting left and right inputs to sum and difference before dropping
>out data that's subjectively insignificant is. And why it's still
>insignificant when converted back to left and right. But I probably
>wouldn't understand the answer. Maybe there's an article in the AES
>Journal that I haven't got around to reading yet.
Well, the argument in favor of M-S recording is that the center of the
stereo imagine is on-axis on the center mike, and totally absent in the
side mike, so the response is pretty good. Whereas, with an X-Y pair,
the center of the stereo image is off-axis on both mikes and therefore
has poorer response.
That is, the M-S method is basically a way of getting around some of the
unfortunate pattern errors with real microphones.
Now, as far as the ATRAC lossy compression goes, it's designed to work
on stereo signals, and if you give it an M-S signal it's going to do
some goofy things to the imaging. ATRAC looks at sum and difference
signals seperately and treats them individually... if you put an M-S
pair into it, it will effectively be generating virtual right and left
channels and processing them differently as part of the compression
algorithm.
The lossy compression systems that are most common are intended for use
with stereo signals that have high correlation between channels and tend
to fall apart with anything else.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Julian Adamaitis
"John in Detroit" <Bla...@sbcglobal.net> wrote in message
news:15Q4e.19615$DW....@newssvr17.news.prodigy.com...
> Julian... I know you are an audio pro and this is cross posted to
> Rec.Music.Filk But do you filk as well?
Don't even know what filk is. I just don't remember to check and see if a
message I reply to has been cross posted EVERY time I reply. Especially
after getting flamed for intentionally cross posting when I wanted to reach
Arny on this group regarding a thread on alt.minidisc, I feel a little
embarrassed.
Julian
Mark
Mike Rivers wrote:
> In article <d2vj4c$cps$1...@domitilla.aioe.org>
keshlam...@comcast.net writes:
>
> >
> > Mike Rivers wrote:
> > > However, I really have no idea just what this L+R/L-R recording
method
> > > really is, how it works, why it works, or what's behind it all.
> >
> > Basically: Rather than two cardioid microphone cartridges pointing
> > across each other to record your stereo image, use one cardioid
pointed
> > straight ahead and a bidirectional at right angles to it.
>
> Right - I know all about that. I want to know what the benefit of
> converting left and right inputs to sum and difference before
dropping
> out data that's subjectively insignificant is. And why it's still
> insignificant when converted back to left and right. But I probably
> wouldn't understand the answer. Maybe there's an article in the AES
> Journal that I haven't got around to reading yet.
>
>
>
best use of the avaialbe bits, i.e. to give the best possible quality
for a given bit rate by making use of the fact that most audio contain
more l+r then l-r.
please see:
http://harmsy.freeuk.com/mostync/
Mark
Mike Rivers
> Now, as far as the ATRAC lossy compression goes, it's designed to work
> on stereo signals, and if you give it an M-S signal it's going to do
> some goofy things to the imaging.
> The lossy compression systems that are most common are intended for use
> with stereo signals that have high correlation between channels and tend
> to fall apart with anything else.
So is this "L+R/L-R recording" that the brother who has the Marantz
660 talks about pure nonsense? Or do they do it anyway, for some
reason that they think is better than preserving the stereo imaging?
(wonder what that could be)
Kurt Albershardt
> "John in Detroit" <Bla...@sbcglobal.net> wrote in message
> news:15Q4e.19615$DW....@newssvr17.news.prodigy.com...
>
>> Julian... I know you are an audio pro and this is cross posted to
>> Rec.Music.Filk But do you filk as well?
>
>
> Don't even know what filk is.
Neither did I <http://home.earthlink.net/~kayshapero/filkdef.htm>
Mike Rivers
> joint stereo (in this context) is a way for MP3 encoders to make the
> best use of the avaialbe bits, i.e. to give the best possible quality
> for a given bit rate by making use of the fact that most audio contain
> more l+r then l-r.
>
> please see:
>
> http://harmsy.freeuk.com/mostync/
This world seems to have developed a language of their own, or rather,
their own usage for terms commonly understood in pro audio circles.
This makes it rather difficult to understand, but I was able to follow
the gist of the article. He's (and apparently this technology) uses
terms that I know of as stereo microphone placement techniques to
describe encoding systems.
The article states "The technique of manipulating stereo audio signals
in Mid/Side format was devised solely for the purpose of enhancing
audio compression." Maybe someone should tell the developers of the
revered and muchly antiqued Fairchild 670 mastering compressor that.
And here we go, using the term "compress" to mean two different
things.
Another quote: " Well, the M/S Joint Stereo technique takes advantage
of the fact that for most recorded music, there is comparatively
little difference between the audio signals for the Left and Right
channels." Yeah, I guess that most pop music is mostly mono. They
don't want to take a chance that the driver of the boom-SUV will miss
anything that goes on over on the passenger's side, and vice versa.
I'm starting to get the impression that the end point here isn't to
restore the original stereo spread and image, but rather to make the
stereo (as much of it as there is) into mono, then bugger it so that
there's some stereo spread - which doesn't necessarily have to be the
same as what went in to the encoder.
All in all, I think it's a pretty decent article as far as it goes,
and I thank you for the link. But the more I know about what goes on
inside the box, the more I confirm my belief that it's not that
important to have 20,000 tunes stored on the little fob hanging around
my neck. Particularly when I'll be listening to them on headphones.
Bob Cain
Mike Rivers wrote:
> Another quote: " Well, the M/S Joint Stereo technique takes advantage
> of the fact that for most recorded music, there is comparatively
> little difference between the audio signals for the Left and Right
> channels." Yeah, I guess that most pop music is mostly mono. They
> don't want to take a chance that the driver of the boom-SUV will miss
> anything that goes on over on the passenger's side, and vice versa.
> I'm starting to get the impression that the end point here isn't to
> restore the original stereo spread and image, but rather to make the
> stereo (as much of it as there is) into mono, then bugger it so that
> there's some stereo spread - which doesn't necessarily have to be the
> same as what went in to the encoder.
Mike, it's just that for a given bit budget and with
signifigantly correlated channels you perceive less
degradation if the loss is biased toward the L-S. After
recombination to L and R you don't really hear stereo image
degradation for psychoacoustic Vodou reasons but rather you
just hear an increase in the overall quality. I don't think
there is a whole lot of theory to explain why this is so.
It just is.
Bob
--
"Things should be described as simply as possible, but no
simpler."
A. Einstein
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com> wrote
> Right - I know all about that. I want to know what the benefit of
> converting left and right inputs to sum and difference before dropping
> out data that's subjectively insignificant is. And why it's still
> insignificant when converted back to left and right. But I probably
> wouldn't understand the answer. Maybe there's an article in the AES
> Journal that I haven't got around to reading yet.
First of all, you don't loose any data by going L+R / L-R. When combined
together properly you get exactly what you started out with: L & R.
This is the way stereo FM radio always works. Trust me, there is NO
disadvantage.
I'm not an expert on this, but in layman's terms: What you get is better
quality for the same amount of data because L and R are usually basically
the same amount of data. L+R is also basically the same amount of data.
(L+R means MONO.) Since most audio mixes are mostly mono with somewhat less
stereo information, the L-R channel usually takes up a whole less amount of
data. If you record L+R and L-R you can use only what is needed for the L-R
and allocate the bulk of your bandwidth to the L+R.
I don't think it has anything to do with insignificant data as you
mistakenly say above. It just has to do with the fact there is a smaller
amount of data needed to describe L-R, so it is more efficient.
Julian
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com> wrote
> Yeah, I guess that most pop music is mostly mono. They
> don't want to take a chance that the driver of the boom-SUV will miss
> anything that goes on over on the passenger's side, and vice versa.
Not at all. It has to do with in the 60's they used to separate things
really wide. John would be in the left speaker only and Paul only in the
right. The drums in one speaker and the guitar in the other. Those were
weird mixes and don't sound anything like reality. Engineers figured out
reality is mostly mono, so they started mixing mostly mono to make things
sound more realistic. There is an art to making believable stereo mixes.
They are mostly mono, but certain things, like reverbs, delays etc., ARE
spaced as far apart as possible. Its just that what sounds the most real is
usually about 80% mono. REALLY!
> I'm starting to get the impression that the end point here isn't to
> restore the original stereo spread and image, but rather to make the
> stereo (as much of it as there is) into mono, then bugger it so that
> there's some stereo spread - which doesn't necessarily have to be the
> same as what went in to the encoder.
No, you misunderstand. L+R combined with L-R is EXACTLY the same as L and
R. Nothing is buggered. Nothing is lost, nothing is added. Now encoding
is another story and stuff IS lost and added, but none of the lost and added
is due to the L+R / L-R part.
In FM radio they do it so there is an easy mono signal for places where the
reception is too weak for stereo. In MPG they do it because it improves the
quality.
Julian
David G. Bell
mri...@d-and-d.com "Mike Rivers" wrote:
> In article <d2vj4c$cps$1...@domitilla.aioe.org> keshlam...@comcast.net writes:
>
> >
> > Mike Rivers wrote:
> > > However, I really have no idea just what this L+R/L-R recording method
> > > really is, how it works, why it works, or what's behind it all.
> >
> > Basically: Rather than two cardioid microphone cartridges pointing
> > across each other to record your stereo image, use one cardioid pointed
> > straight ahead and a bidirectional at right angles to it.
>
> Right - I know all about that. I want to know what the benefit of
> converting left and right inputs to sum and difference before dropping
> out data that's subjectively insignificant is. And why it's still
> insignificant when converted back to left and right. But I probably
> wouldn't understand the answer. Maybe there's an article in the AES
> Journal that I haven't got around to reading yet.
Isn't most lossless stereo a bit of a fake anyway?
Not the live recording with two mics, but the studio work -- as I recall
it, the individual tracks are essentially mono, mixed together, and that
mixing to create the stereo image only uses volume differences, not the
phase differences that would be there in a true binaural recording. And
that makes the difference signal a lot simpler.
Does that make any sort of sense, or am I badly out of date on what
happens in a studio?
Julian Adamaitis
""David G. Bell"" <db...@zhochaka.demon.co.uk> wrote
> Isn't most lossless stereo a bit of a fake anyway?
>
> Not the live recording with two mics, but the studio work -- as I recall
> it, the individual tracks are essentially mono, mixed together, and that
> mixing to create the stereo image only uses volume differences, not the
> phase differences that would be there in a true binaural recording. And
> that makes the difference signal a lot simpler.
>
> Does that make any sort of sense, or am I badly out of date on what
> happens in a studio?
I'm badly out of date in a studio myself. I last mixed an album a couple
years ago and its been 5 or 6 years before that since I did it regularly.
But, yes, it is a fake, but an elaborate one Good engineers who use good
mics, good mic placement, stereo mics, delays, eq effects, and reverbs can
create amazing images. Yes they're fakes, but if you close your eyes its
like being there. I've done a lot of live recording too, which actually is
done being there and the fake studio stuff can be every bit as rich.
Julian
Mike Rivers
In article <1159ouf...@corp.supernews.com> nospamJ...@Access4Less.net writes:
> No, you misunderstand. L+R combined with L-R is EXACTLY the same as L and
> R. Nothing is buggered. Nothing is lost, nothing is added. Now encoding
> is another story and stuff IS lost and added, but none of the lost and added
> is due to the L+R / L-R part.
I understand all of that. I've used the mic technique for years. What
I don't understand is why they do this when encoding MP3. Someone
started to give an explanation that mono encodes more efficiently, and
that's what L+R is. I'll accept that for now.
--
I'm really Mike Rivers - (mri...@d-and-d.com)
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com> wrote
> I understand all of that. I've used the mic technique for years. What
> I don't understand is why they do this when encoding MP3. Someone
> started to give an explanation that mono encodes more efficiently, and
> that's what L+R is. I'll accept that for now.
Hi Mike,
I'll try one last time in over simplified terms. IF 80 % (roughly) of the
information is MONO, THEN only 20% is L-R. It takes a lot less data to
encode the small amount of 20% L-R information than either the full
bandwidth L or R or Mono channels.
Julian
Mark
the L+R carries most of the information so you have one "major"
channel to encode and one minor (the L-R) channel instead of 2 major
channels.
Mark
Julian Adamaitis
"Mark" <mako...@yahoo.com> wrote in message
news:1112924815....@l41g2000cwc.googlegroups.com...
Mark
Julian Adamaitis wrote:
> nicely said
>
>
thank you
Mark
Mike Rivers
In article <115boeg...@corp.supernews.com> nospamJ...@Access4Less.net writes:
> I'll try one last time in over simplified terms. IF 80 % (roughly) of the
> information is MONO, THEN only 20% is L-R. It takes a lot less data to
> encode the small amount of 20% L-R information than either the full
> bandwidth L or R or Mono channels.
I understand that. Oversimplification of a very complex process
doesn't work for me.
If there's a useful oversimplification, it's the assumption that 80%
of the information is mono. How is this deduced? Surely you don't have
80% complete duplication in the two channels. Maybe you have 80%
duplication if you allow a 20% (or some other figure) fudge factor,
saying that the two channels are 'close enough' 80% of the time. Is
that how it works?
This may be valid for a pop recording, but on an at least somewhat
professional recorder like the Marantz (which, sadly, seems to have
some quite un-pro features) you'd think they'd want to do better. I
doubt that true two-mic stereo recordings have near 80% mono content.
But then I've never thought about it, and I don't really know how to
go about thinking about it. But I've seem plenty of Lissajous patterns
and few of them look like tight ovals.
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com>
> I understand that. Oversimplification of a very complex process
> doesn't work for me.
Try Mike's simplified version in this thread! I think maybe even you can
accept it! It is absolutely true but doesn't get caught in concepts that
require theory.
> If there's a useful oversimplification, it's the assumption that 80%
> of the information is mono. How is this deduced? Surely you don't have
> 80% complete duplication in the two channels. Maybe you have 80%
> duplication if you allow a 20% (or some other figure) fudge factor,
> saying that the two channels are 'close enough' 80% of the time. Is
> that how it works?
No I'd GUESS its typically 80%, but it can vary drastically with program
material. Like I said earlier the old Beatles stuff where everything was
panned hard left and right was most certainly much less than 80% mono. I
thought I made it clear the exact number was not necessary to understand the
concept of L-R encoding. What is important is that IF *L-R is less than
L+R* you save data. Even of you have 60/40% the techniques still works
Why did I say 80%? I WAS GUESSING a purely ballpark number based on albums
I've personally mixed and albums I've watched excellent engineers mix. I
pan most instruments dead center or 10 /11 o'clock left or 1/2 o'clock
right. Few things I pan 3:00 / 9:00 like stereo drum overheads, stereo
pianos etc., and those things usually have much common information so there
still isn't that much difference between left and right channels. The ONE
and ONLY thing that do I always pan hard left and right is reverb, which in
total volume is the quietest part of the entire mix.
Based on that I came up with a totally seat of the pants number 80% mono.
The technique works if it is only 60% mono. You are welcome to come up with
a more precise number if that's what your after. My only point is it sounds
better the more difference there is. Your criticism of my non-technical
explanation is inappropriate as I was responding to a guy who just wasn't
getting even after reading 3 explanations and being MORE precise would have
probably confused him even MORE. You are welcome to come up with you own
explanation that is both technically precise and simple to understand to non
technical people. I look forward to reading it if you do so!
> This may be valid for a pop recording, but on an at least somewhat
> professional recorder like the Marantz (which, sadly, seems to have
> some quite un-pro features) you'd think they'd want to do better. I
> doubt that true two-mic stereo recordings have near 80% mono content.
> But then I've never thought about it, and I don't really know how to
> go about thinking about it. But I've seem plenty of Lissajous patterns
> and few of them look like tight ovals.
The small amount of stuff I mix that IS panned very hard is VERY much out of
phase That's the whole point of spreading it out as far as possible! It's
still a minority of total decibels however. It still takes significantly
less data to describe the out of phase material than the in phases material.
Julian
Mark
by the way...
I think I read here:
http://harmsy.freeuk.com/mostync/
that the MP3 encoder automatically switches back to L and R encoding if
it sees that the L-R signal is too complex and the L+R and L-R encoding
stratagy would fail to give better results.
So you get the best of both worlds, if there is a benefit to the L+R
L-R encoding, it uses it, if not, it doesn't.
Mark
Julian Adamaitis
"Mark" <mako...@yahoo.com> wrote
by the way...
I think I read here:
http://harmsy.freeuk.com/mosty要c/
that the MP3 encoder automatically switches back to L and R encoding if
it sees that the L-R signal is too complex and the L+R and L-R encoding
stratagy would fail to give better results.
So you get the best of both worlds, if there is a benefit to the L+R
L-R encoding, it uses it, if not, it doesn't.
Mark
That makes sense. If the mono signal is more than difference, it would
actually take more data to do it sum difference. Seeing as popel can create
anything from mono to mostly stereo, a system that didn't know what to do in
that case would be of limited use.
Julian
Carey Carlan
> If there's a useful oversimplification, it's the assumption that 80%
> of the information is mono. How is this deduced? Surely you don't have
> 80% complete duplication in the two channels. Maybe you have 80%
> duplication if you allow a 20% (or some other figure) fudge factor,
> saying that the two channels are 'close enough' 80% of the time. Is
> that how it works?
It's a reasonable assumption. Typically the most important source is
centered. That applies to classical and modern content alike. If you
encode the "mid" with high res and the "side" with low res, the most you
can lose is the stereo image.
Kurt Albershardt
Phase anomalies between the two recorded tracks were quite common in the
analog days, especially after the signal had passed through a couple
generations and at least one pass through a routing switcher. M-S
encoding drastically reduced the sonic impact of those anomalies (while
preserving mono compatibility.)
Mike Rivers
In article <115d8eo...@corp.supernews.com> nospamJ...@Access4Less.net writes:
> The small amount of stuff I mix that IS panned very hard is VERY much out of
> phase That's the whole point of spreading it out as far as possible! It's
> still a minority of total decibels however. It still takes significantly
> less data to describe the out of phase material than the in phases material.
I hope I'm misunderstand you here, but I hope you aren't purposely
making recordings that I would consider unlistenable just for the sake
of making the compression algorithm work better. Say it aint' so, Joe.
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com> wrote
> I hope I'm misunderstand you here, but I hope you aren't purposely
> making recordings that I would consider unlistenable just for the sake
> of making the compression algorithm work better. Say it aint' so, Joe.
You are misunderstanding me. My goal is to make natural sounding
recordings. Having Ringo sing out of one speaker and Paul out of the other
is not making a natural sounding recording IMO! Mike you seem very
knowledgeable on a lot of subjects. I don't understand why you think
"listenable" recordings means stuff panned hard???? Simply not true, my
good man!
Julian
Mike Rivers
In article <115mkon...@corp.supernews.com> nospamJ...@Access4Less.net writes:
> Mike you seem very
> knowledgeable on a lot of subjects. I don't understand why you think
> "listenable" recordings means stuff panned hard???? Simply not true, my
> good man!
I don't. But what I read from your post to which I commented, I
thought you said you intentionally recorded with wide separation.
Forget it. I saw a Marantz 660 over the weekend anyway.
--
I'm really Mike Rivers (mri...@d-and-d.com)
Eric
the same problem I do:
With no pad on, levels are normal. Unit is overdriven with moderate
inputs, faster than I would like, but this isn't the problem. When I
switch on the 20db pad, the level decreases by 38db instead of 20db. I
checked this with a repeatable sound source (test tone through computer
speakers) and got consistent results with different mics and phantom
power on/off.
I believe "20db pad" means the peaks should be 20db lower, right?
Denon/Marantz has taken a while to answer this question, so I'm posing
it to the RAP community.
It makes this unit nearly unusable because 0db pad distorts amazingly
fast, and 20db pad (actually 38db pad) requires turning up the gain
until the thing is pretty noisy. That, and the fact that I can't see
the LED meters in bright sunlight makes it difficult to make field
recordings of a loud, outdoor activity (drum & bugle corps).
Eric
ela...@3x.net <--- change x to z to reply
Julian Adamaitis
"Mike Rivers" <mri...@d-and-d.com> wrote
> I don't. But what I read from your post to which I commented, I
> thought you said you intentionally recorded with wide separation.
No, I said the exact opposite.
Julian
Mike Rivers
In article <1113361704.4cade86c7b197c5e55469b302c428e3a@teranews> ela...@3z.net writes:
> With no pad on, levels are normal. Unit is overdriven with moderate
> inputs, faster than I would like, but this isn't the problem. When I
> switch on the 20db pad, the level decreases by 38db instead of 20db. I
> checked this with a repeatable sound source (test tone through computer
> speakers) and got consistent results with different mics and phantom
> power on/off.
>
> I believe "20db pad" means the peaks should be 20db lower, right?
While it might be the peaks that count, it's difficult to measure
attenuation using peak values. This is where a good old fashioned VU
(or VU-style) meter is useful. I've only fondled one of these
recorders and not had an opportunity to put it on the bench, but it's
easy enough to make a 20 dB pad that I can't imaging that they got
that wrong - unless it's not really a pad. That would be bad.
> It makes this unit nearly unusable because 0db pad distorts amazingly
> fast, and 20db pad (actually 38db pad) requires turning up the gain
> until the thing is pretty noisy. That, and the fact that I can't see
> the LED meters in bright sunlight makes it difficult to make field
> recordings of a loud, outdoor activity (drum & bugle corps).
Some people have it in mind that the recording level is too low if the
meters don't hit the peak much of the time. Perhaps you should try
recording with the pad in and just turn up your playback volume a bit.
Is the noise you get when you turn up the gain real electronic noise,
or are you talking about background noise? Of course that will come up
(along with what you really want to record) when you turn up the gain.
One of the questions I asked of the person whos PMD660 I was looing at
over the weekend was whether the meters were easy to read. He said
they were, so I guess we have a difference of opinion there. He mostly
records acoustic instruments in jam sessions, so overload is no
problem for his application. In fact he was pretty happy using the
internal microphones (which he was using when I saw him with the
recorder).
Eric
pad operates, and this is the first recorder I have with a pad setting
so I hope it's not user error. With the pad engaged the levels are
microscopic, so it appears something is wrong.
In my first recording, I could not use the no-pad because I get
distortion even with moderate volumes. Changing the gain knob merely
sets what level the flattops occur! With the pad in, I turn the gain
knob up to 75% or more and the levels are still peaking at -30 or so. I
did turn up playback volume (or normalized) but this made my recording
sound like it was done oceanside!
Although far from pro, I've done enough recordings to know something
funky is going on. It is making my Sharp minidisc look pretty good.
Maybe another PMD660 owner encountered this problem since I noticed it
within the first few minutes of use.
For the meters, I can probably build a hood or wear the recorder around
my neck. :)
Eric
In article <znr1113391513k@trad>, mri...@d-and-d.com says...
>
> In article <1113361704.4cade86c7b197c5e55469b302c428e3a@teranews> ela...@3z.net writes:
>
> While it might be the peaks that count, it's difficult to measure
> attenuation using peak values. This is where a good old fashioned VU
> (or VU-style) meter is useful. I've only fondled one of these
> recorders and not had an opportunity to put it on the bench, but it's
> easy enough to make a 20 dB pad that I can't imaging that they got
> that wrong - unless it's not really a pad. That would be bad.
>
Mike Rivers
In article <1113521327.94035ea97bb99a42b737db710f5360f8@teranews> ela...@3z.net writes:
> In my first recording, I could not use the no-pad because I get
> distortion even with moderate volumes. Changing the gain knob merely
> sets what level the flattops occur!
That means that the mic is clipping the front end of the preamp, ahead
of the pad (bad) and ahead of the gain/record level control.
> With the pad in, I turn the gain
> knob up to 75% or more and the levels are still peaking at -30 or so.
Maybe it's broken, maybe it's a bug. Can you generate a steady tone
that you can feed to the mic input and actually see what the pad is
doing?
Eric
more than 20db of stuff. With the pad in, I turn up the gain to 75%
with a decent source, peaks are still at -30 or so. The pad was simply
too much and this made no sense.
In article <znr1113526669k@trad>, mri...@d-and-d.com says...
>
> Maybe it's broken, maybe it's a bug. Can you generate a steady tone
> that you can feed to the mic input and actually see what the pad is
> doing?
Yes, it was broken or something like this. After speaking with the
people at Denon/Marantz (and sending them audio files) they finally
wrote a return authorization and I mailed it back. They returned it
VERY QUICKLY without explanation, except that the work required four 22K
resistors.
Everything works great, gain staging is great, levels are decent, no
distortion. This is a pretty good little recorder except for a few
personal preferences. It's a mystery why my unit supposedly had
incorrect or missing resistors in it. I bet there's more in the same
condition, waiting to be discovered.
Eric
ela...@3x.net <---- change x to z to reply
Mike Rivers
In article <1115082804.7ccbcd5f1ca70449e04189b568c3f32b@teranews> ela...@3x.net writes:
> Recall that I have a Marantz PMD660 where the 20db pad seemed to remove
> more than 20db of stuff. With the pad in, I turn up the gain to 75%
> with a decent source, peaks are still at -30 or so. The pad was simply
> too much and this made no sense.
> After speaking with the
> people at Denon/Marantz (and sending them audio files) they finally
> wrote a return authorization and I mailed it back. They returned it
> VERY QUICKLY without explanation, except that the work required four 22K
> resistors.
>
> Everything works great, gain staging is great, levels are decent, no
> distortion. This is a pretty good little recorder except for a few
> personal preferences. It's a mystery why my unit supposedly had
> incorrect or missing resistors in it. I bet there's more in the same
> condition, waiting to be discovered.
Often they make mistakes and things have to be patched up. If they
catch it in time, you'll find components or jumpers tacked in to a
brand new unit. If they don't, then they have to fix it later. Of
course it's cheaper to do it right from the beginning, but they know
that a certain number of users will never discover, or will never be
affected by a built-in problem. Good thing you figured out that
something had to be wrong and that there was a solution.
I recall in the early days of the Mackie CR1604 mixer there was an
incorrect value resistor in early production runs and the two meters
didn't read the same for an identical input signal.