[Luxey]

Do they have true thru mode, of source monitorong, from mic to line out?
Do they all have thru mode of any kind, appart from headphones out for monitoring?
Do they have separate line outs, or headphone out mimics as one/ is switchable?

[Mike Rivers[_2_]]

On 3/22/2013 5:55 AM, Luxey wrote:
Do they have true thru mode, of source monitorong, from mic to line out?
Do they all have thru mode of any kind, appart from headphones out for monitoring?
Do they have separate line outs, or headphone out mimics as one/ is switchable?

Some do and some don't (to some or all). By "true thru" do you mean that
it doesn't go through anything digital? That might be hard to find and
would require a measurement to determine it. This isn't something that
you'll find on anyone's spec sheet, in any language.

What do you really need, and what units are you considering? There's a
lot of stuff out there.




--
For a good time, call http://mikeriversaudio.wordpress.com

[Neil Gould]

Luxey wrote:
Do they have true thru mode, of source monitorong, from mic to line
out?
Do they all have thru mode of any kind, appart from headphones out
for monitoring? Do they have separate line outs, or headphone out
mimics as one/ is switchable?

As Mike suggests, they are not all the same, even from a single
manufacturer. It is a good idea to narrow the range of handhelds that you're
considering in some way. Price? Feature set? How they'll be used?

--
best regards,

Neil

[jason]

On Fri, 22 Mar 2013 02:55:43 -0700 (PDT) "Luxey" wrote
in article

Do they have true thru mode, of source monitorong, from mic to line out?
Do they all have thru mode of any kind, appart from headphones out for monitoring?
Do they have separate line outs, or headphone out mimics as one/ is switchable?

The H2n and H4n have a single headphone/line-out jack. The recording
source is there but there is no "3rd head mode," as on some recorders,
that plays back what was just stashed on the memory card - very handy for
peace of mind.

[Adrian Tuddenham[_2_]]

Luxey wrote:

Do they have true thru mode, of source monitorong, from mic to line out?
Do they all have thru mode of any kind, appart from headphones out for
monitoring?
Do they have separate line outs, or headphone out mimics as one/ is
switchable?

Tascam DR-05 has only a headphone output, which also serves for
monitoring when recording.

I don't know whether the monitoring signal is direct or A/D = D/A, but
I suspect the latter (although no lag is noticeable in the headphones
when recording a live source).

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

[Neil Gould]

Adrian Tuddenham wrote:
Luxey wrote:

Do they have true thru mode, of source monitorong, from mic to line
out? Do they all have thru mode of any kind, appart from headphones
out for monitoring?
Do they have separate line outs, or headphone out mimics as one/ is
switchable?

Tascam DR-05 has only a headphone output, which also serves for
monitoring when recording.

I don't know whether the monitoring signal is direct or A/D = D/A,
but I suspect the latter (although no lag is noticeable in the
headphones when recording a live source).

The lack of noticeable delay suggests that the monitoring signal is pre A/D.
--
best regards,

Neil

[Mike Rivers[_2_]]

On 3/23/2013 9:41 AM, Neil Gould wrote:

The lack of noticeable delay suggests that the monitoring signal is pre A/D.

It depends on what "noticeable" means. While I wouldn't expect this in a
handheld recorder, I've reviewed a couple of computer audio interfaces
that include a hardware-based DSP mixer for monitoring with mic/line
input to monitor output delay of less than 0.5 ms at 44.1 kHz. That's
going through an A/D and D/A but I'd hardly call it noticeable.

As far as separate headphone and line outputs, taking a quick inventory
around here, the only one I have that offers that is the Korg MR-1000.
It's not handheld, but it's battery powered and eminently portable. I
use it when I have to look more professional on a remote than to use a
handheld recorder, which works just as well.

As I recall, the Sony PCM-D10, while not having separate headphone and
line output jacks, has a switch in a menu that turns off the power stage
of the headphone amplifier if you're using the jack as a line output.
It's a battery saving measure and it practically doubles the battery
life. Not the same thing that the OP was looking for, I suspect, but at
least they're thinking.



--
For a good time, call http://mikeriversaudio.wordpress.com

[Neil Gould]

Mike Rivers wrote:
On 3/23/2013 9:41 AM, Neil Gould wrote:

The lack of noticeable delay suggests that the monitoring signal is
pre A/D.

It depends on what "noticeable" means.

Agreed. I mean it as undetectable as a phase shift or echo effect. How do
you mean it?

--
best regards,

Neil

[Adrian Tuddenham[_2_]]

Neil Gould wrote:

Mike Rivers wrote:
On 3/23/2013 9:41 AM, Neil Gould wrote:

The lack of noticeable delay suggests that the monitoring signal is
pre A/D.

It depends on what "noticeable" means.

Agreed. I mean it as undetectable as a phase shift or echo effect. How do
you mean it?

What I meant was: when listening with headphones and a 'live' source
simultaneously at similar volume levels (e.g. with the headphones partly
off the ears), there was no noticeable echo or phasing effect.

In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

[Don Pearce[_3_]]

On Sat, 23 Mar 2013 17:23:50 +0000 (UTC), Jeff Henig
wrote:

Adrian Tuddenham wrote:


In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


Okay, here's where I get educated.

I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

Can you point me to a good tutorial that explains how you perform these
measurements? Because you just piqued my curiosity.

Just to be sure, I think Adrian meant 0.714 milliseconds, not
microseconds.

d

[Mike Rivers[_2_]]

On 3/23/2013 11:23 AM, Neil Gould wrote:
The lack of noticeable delay suggests that the monitoring signal is
pre A/D.
I mean it as undetectable as a phase shift or echo effect. How do
you mean it?

I don't know how you hear phase shift, but you might mean what my
biggest annoyance with small amounts of delay (1-3 ms) is, and that's
comb filtering that occurs when your voice coming up your throat mixes
at your eardrum with the sllightly delayed copy coming from the D/A
converter.

Delays of 30 ms or more are pretty obvious. Delays of 5-10 ms are
usually tolerable and often not noticeable, depending on what you're doing.


--
For a good time, call http://mikeriversaudio.wordpress.com

[Mike Rivers[_2_]]

On 3/23/2013 1:23 PM, Jeff Henig wrote:
I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

I use a click generator (a real piece of hardware) and a dual trace
scope. I send the click to one chanel of the scope and to the device I'm
testing. I send the output of the device under test to the other channel
of the scope. Let 'er click and measure the time difference between the
click on the two traces.

If you're measuring the delay (latency) of a stereo (or multichannel)
computer audio interface, you can do it all on a computer by feeding the
click to one input, patching the monitor output back into the other
channel, start the click, record both channels, and then read the time
difference between them.

I'm pretty sure I described this method in an article about latency that
I have on my web site.


--
For a good time, call http://mikeriversaudio.wordpress.com

[Neil Gould]

Mike Rivers wrote:
On 3/23/2013 11:23 AM, Neil Gould wrote:
The lack of noticeable delay suggests that the monitoring signal is
pre A/D.
I mean it as undetectable as a phase shift or echo effect. How do
you mean it?

I don't know how you hear phase shift, but you might mean what my
biggest annoyance with small amounts of delay (1-3 ms) is, and that's
comb filtering that occurs when your voice coming up your throat mixes
at your eardrum with the sllightly delayed copy coming from the D/A
converter.

Yep, that's how I would describe a detectable phase shift effect.

Delays of 30 ms or more are pretty obvious. Delays of 5-10 ms are
usually tolerable and often not noticeable, depending on what you're
doing.

Well, I can tolerate a lot of things that I can detect if it doesn't hamper
what I need to do in some way.

--
best regards,

Neil

[Adrian Tuddenham[_2_]]

Don Pearce wrote:

On Sat, 23 Mar 2013 17:23:50 +0000 (UTC), Jeff Henig
wrote:

Adrian Tuddenham wrote:


In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


Okay, here's where I get educated.

I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

Can you point me to a good tutorial that explains how you perform these
measurements? Because you just piqued my curiosity.

Just to be sure, I think Adrian meant 0.714 milliseconds, not
microseconds.

Sorry, slip of the typing finger. I actually meant to type 714
microseconds, which is the same thing as 0.714 milliseconds.


--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

[Adrian Tuddenham[_2_]]

Jeff Henig wrote:

Adrian Tuddenham wrote:


In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


Okay, here's where I get educated.

I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

Can you point me to a good tutorial that explains how you perform these
measurements? Because you just piqued my curiosity.

I used the twin-beam oscilloscope method (without a byte of software in
sight). I don't know for any tutorials, but you should be able to work
out a methood from any basic analogue electronics textbook.

As a brief summary:

The two inputs of the 'scope were connected to the Tascam, one to the
input and one to the output. A sinewave signal generator was connected
to the input and the recorder was set to Record (actually in Pause mode,
but that should not matter). The two traces were accurately centred
(vertically) and the sensitivity controls were adjusted until the two
traces were of similar height.

At a lowish frequency (e.g. 100 c/s) the traces were compared and, if
necessary, one could be inverted to make them the same polarity. The
frequency was then increased until they were exactly out of phase. it
was further increased until they came back into phase. Those two
frequencies ought to be related in the ratio 1:2, if they aren't, it
indicates that the starting frequency was too high.

The frequency at which they come back in phase (in cycles per second) is
the reciprocal of the delay time (in seconds).


I have set up the test again and photographed it so you can see what
happens.

www.poppyrecords.co.uk/other/twinbeam/twin.htm

The RH input terminals of the 'scope are connected to the signal
generator (pale cyan coloured instrument perched on top of the 'scope).
The test leads from the output of the signal generator are clipped to
the jackstrip underneath the 'scope, from where they go to the input
mini-jack of the Tascam DR-05 which is situated between the two mics.
The output mini-jack on the side of the Tascam is connected to the LH
'scope input.

The waveforms are shown for four different frequencies. The input
waveform is displayed slightly larger than the output waveform, so as to
distinguish between them. The input is synchronised so as to always
start at the zero axis crossing on the left of the screen. The time
displacement of the output signal is constant, but as the frequency
rises you will see that it intercepts the input at different points on
the waveform as the frequency rises and the cycles shorten.


--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

[Scott Dorsey]

Adrian Tuddenham wrote:

The two inputs of the 'scope were connected to the Tascam, one to the
input and one to the output. A sinewave signal generator was connected
to the input and the recorder was set to Record (actually in Pause mode,
but that should not matter). The two traces were accurately centred
(vertically) and the sensitivity controls were adjusted until the two
traces were of similar height.

At a lowish frequency (e.g. 100 c/s) the traces were compared and, if
necessary, one could be inverted to make them the same polarity. The
frequency was then increased until they were exactly out of phase. it
was further increased until they came back into phase. Those two
frequencies ought to be related in the ratio 1:2, if they aren't, it
indicates that the starting frequency was too high.

The frequency at which they come back in phase (in cycles per second) is
the reciprocal of the delay time (in seconds).

See, I don't like this method. What I like to do is to use lissajous
method, with one signal going to the X input of the scope and the other
signal going to the Y input. If they are both in phase, you get a nice
diagonal line... if they are 90' out of phase you get a circle... and you
can make some rough judgements about values in-between.

If you don't have a scope, you can do it with a VU meter, just summing the
two inputs together.... adjust the frequency and the meter will dip lowest
at the frequency where there is best cancellation. I don't really recommend
this method but I have done it before. It's not all that accurate but it
is sometimes accurate enough.

There are always three ways to do anything: the right way, the wrong way,
and the Army way.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Adrian Tuddenham[_2_]]

Scott Dorsey wrote:

Adrian Tuddenham wrote:

The two inputs of the 'scope were connected to the Tascam, one to the
input and one to the output. A sinewave signal generator was connected
to the input and the recorder was set to Record (actually in Pause mode,
but that should not matter). The two traces were accurately centred
(vertically) and the sensitivity controls were adjusted until the two
traces were of similar height.

At a lowish frequency (e.g. 100 c/s) the traces were compared and, if
necessary, one could be inverted to make them the same polarity. The
frequency was then increased until they were exactly out of phase. it
was further increased until they came back into phase. Those two
frequencies ought to be related in the ratio 1:2, if they aren't, it
indicates that the starting frequency was too high.

The frequency at which they come back in phase (in cycles per second) is
the reciprocal of the delay time (in seconds).

See, I don't like this method. What I like to do is to use lissajous
method, with one signal going to the X input of the scope and the other
signal going to the Y input. If they are both in phase, you get a nice
diagonal line... if they are 90' out of phase you get a circle... and you
can make some rough judgements about values in-between.

With either method, the limit on the accuracy will be the thickness of
the 'scope trace but there could be additional error in the twin-beam
method if the operator was a bit careless and the two traces weren't
accurately centered on zero. The lissajous method has the advantage of
only requiring a single beam 'scope but the twin-beam method requires
less explanation if you are demonstrating first principles to a
beginner.

If you don't have a scope, you can do it with a VU meter, just summing the
two inputs together.... adjust the frequency and the meter will dip lowest
at the frequency where there is best cancellation. I don't really recommend
this method but I have done it before. It's not all that accurate but it
is sometimes accurate enough.

If there is more than about 6dB difference in amplitude between the two
signals, the dip would be difficult to find accurately - but if you have
sufficiently fine gain control to make the two signals exactly equal,
that would be the most accurate method of all.

Nulling input against output is also a good way to show up other faults
such as distortion and noise-behind-signal (...as long as you are sure
that the noise and distortion aren't occuring in the attenuating side
chain and are only in the device under test). :-)


--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

[Edi Zubovic]

On Sat, 23 Mar 2013 17:23:50 +0000 (UTC), Jeff Henig
wrote:

Adrian Tuddenham wrote:


In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


Okay, here's where I get educated.

I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

Can you point me to a good tutorial that explains how you perform these
measurements? Because you just piqued my curiosity.

--Well, here are two primers of oscilloscope measurements by
Tektronix. The first one is older, I have it in paperback for decades
and it's really well done and easy to understand.

http://njarc.org/books/XYZs%20of%20U...ng_a_Scope.pdf

The second one is recent:--
http://aries.ucsd.edu/najmabadi/CLAS.../XYZ-Scope.pdf


Edi Zubovic, Crikvenica, Croatia

PS. As to azimuth errors, provided that they are constant across the
tape, ie. that the tape machine is mechanically flawless but
misaligned in respect of the currently reproduced tape, I'm no more
doing any reproduce head adjustments. All can be very neatly done by
software, sliding the samples until both of the channels match. Just
perfect.

[Scott Dorsey]

Edi Zubovic edi.zubovic[rem wrote:

PS. As to azimuth errors, provided that they are constant across the
tape, ie. that the tape machine is mechanically flawless but
misaligned in respect of the currently reproduced tape, I'm no more
doing any reproduce head adjustments. All can be very neatly done by
software, sliding the samples until both of the channels match. Just
perfect.

This takes care of the interchannel delay, but it doesn't fix the comb
filtering problems within each channel.
--scott



--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Edi Zubovic]

On 24 Mar 2013 07:57:37 -0400, (Scott Dorsey) wrote:

Edi Zubovic edi.zubovic[rem wrote:

PS. As to azimuth errors, provided that they are constant across the
tape, ie. that the tape machine is mechanically flawless but
misaligned in respect of the currently reproduced tape, I'm no more
doing any reproduce head adjustments. All can be very neatly done by
software, sliding the samples until both of the channels match. Just
perfect.

This takes care of the interchannel delay, but it doesn't fix the comb
filtering problems within each channel.
--scott


-- True, as the gap isn't 100 pct. perpendicular to the recorded track
when there's missalignment. It is angled. I'll have to reproduce the
problem to see whether comb filtering occurs in a significant amount
if there is a 1-4 samples difference between the channels.

Edi Zubovic, Crikvenica, Croatia

[Les Cargill[_4_]]

Jeff Henig wrote:
Adrian Tuddenham wrote:


In the interests of science (and for the avoidance of arguments) I have
now measured the delay and found it to be 0.714 microseconds. It gives
180 degree phase shift between input and output at 700 c/s and 360
degrees at 1.4 Kc/s. This is equivalent to 8.5 inches physical
displacement between microphone and earphones.


Okay, here's where I get educated.

I'm assuming that you use some sort of software to measure all of this. In
a quick Google search, I saw something about a two-trace oscilloscope with
two probes.

Can you point me to a good tutorial that explains how you perform these
measurements? Because you just piqued my curiosity.



This can be done by visually examining the various waveforms. Count
the number of samples of offset, then translate that into time.

You can also run a deconvolution between two tracks. The result
will have a first, largish peak - it's the basic delay between
the two.

Whether that's easier than using a scope is something
open to debate.

--
Les Cargill

[Luxey]

Thank you all,

Itention is to buy the thing for it's pourpose, but also use it as stereo pair for multitrack recording.

I know, I can always slide in DAW, but I'd prefer not to.

Thanks again for good advice. Now, please continue arguing I enjoy it.

Poseban pozdrav Ediu, odavno ga nisam video na RAPu.

[Arny Krueger[_5_]]

"Scott Dorsey" wrote in message
...
Adrian Tuddenham wrote:

The two inputs of the 'scope were connected to the Tascam, one to the
input and one to the output. A sinewave signal generator was connected
to the input and the recorder was set to Record (actually in Pause mode,
but that should not matter). The two traces were accurately centred
(vertically) and the sensitivity controls were adjusted until the two
traces were of similar height.

At a lowish frequency (e.g. 100 c/s) the traces were compared and, if
necessary, one could be inverted to make them the same polarity. The
frequency was then increased until they were exactly out of phase. it
was further increased until they came back into phase. Those two
frequencies ought to be related in the ratio 1:2, if they aren't, it
indicates that the starting frequency was too high.

The frequency at which they come back in phase (in cycles per second) is
the reciprocal of the delay time (in seconds).

See, I don't like this method. What I like to do is to use lissajous
method, with one signal going to the X input of the scope and the other
signal going to the Y input. If they are both in phase, you get a nice
diagonal line... if they are 90' out of phase you get a circle... and you
can make some rough judgements about values in-between.

If you don't have a scope, you can do it with a VU meter, just summing the
two inputs together.... adjust the frequency and the meter will dip lowest
at the frequency where there is best cancellation. I don't really
recommend
this method but I have done it before. It's not all that accurate but it
is sometimes accurate enough.

There are always three ways to do anything: the right way, the wrong way,
and the Army way.

Obvious flaw in both of the methods described above is that they depend on
phase shift that repeats these effects every 360 degrees. The methodology
suggested in an earlier post does not have this flaw. It can also be
implmented using a computer with an audio recording interface which is a
commonly-available resource in this day and age.

On balance, the phase-based methods are usually correct, unless you are
measuring the latency due to longer delays such as the delays between the
recording head and monitoring head on an analog tape machine. Use of a lower
frequency tone can remove much of this ambiguity.

On analog tape machines there is often a signficiant change on timbre and
color (at least a slight one) between direct source monitoring and off-media
monitoring. With good modern digital equipment, that is far less likely.

[Luxey]

Any experience with Olympus LS models?

[Scott Dorsey]

Edi Zubovic edi.zubovic[rem wrote:
On 24 Mar 2013 07:57:37 -0400, (Scott Dorsey) wrote:
Edi Zubovic edi.zubovic[rem wrote:

PS. As to azimuth errors, provided that they are constant across the
tape, ie. that the tape machine is mechanically flawless but
misaligned in respect of the currently reproduced tape, I'm no more
doing any reproduce head adjustments. All can be very neatly done by
software, sliding the samples until both of the channels match. Just
perfect.

This takes care of the interchannel delay, but it doesn't fix the comb
filtering problems within each channel.

-- True, as the gap isn't 100 pct. perpendicular to the recorded track
when there's missalignment. It is angled. I'll have to reproduce the
problem to see whether comb filtering occurs in a significant amount
if there is a 1-4 samples difference between the channels.

You should be able to estimate it with simple trigonometry, given the gap
width. My guess is that at the 1-4 sample range it's not a huge deal with
a half-track head, but do the math and see. It will be much worse with a
full-track head or with 1/2" 2-track....
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Mike Rivers[_2_]]

On 3/25/2013 8:42 AM, Luxey wrote:
Any experience with Olympus LS models?

Nobody buys those. They're way too popular.

I've seen good reviews of them, but whenever I fondle one at a trade
show thinking about asking for one to review, I always get the feeling
that there's already something else out there that does the same job
just as well. And the truth is that they all work about as well as each
other. The difference is in the kind of details that you aren't likely
to find in a review - details that are important to you but not to the
reviewer.

Why don't you buy one, play with it for a while, and if it doesn't meet
your needs, return it and try something else.



--
For a good time, call http://mikeriversaudio.wordpress.com