> It certainly preserves *something*
>better but what should that be called?
One thing conserved is the condition of the natural world
to which we have four billion years of adaptation.
Thanks for a fascinating discussion,
Chris Hornbeck
6x9=42
Reply by Bob Cain●April 5, 20052005-04-05
Andor wrote:
> Really, I don't know why we argue. We both agree that scientific
> listening tests should be conducted. However, until such results are
> published, I try to refrain from general statements as the above.
We are mostly in agreement. My statements were with respect
to my own testing and my own impressions and interpretation.
Sorry if that was not clear.
Bob
--
"Things should be described as simply as possible, but no
simpler."
A. Einstein
Reply by Andor●April 4, 20052005-04-04
Bob Cain wrote:
...
> That possibility is exactly why I suggested DBT. Why are
> you being antagonistic?
Bob,
I'm just not happy in the way the linear vs. minimum phase issue is
discussed. This is why I like to post a reference to that article of
Michael Gerzon, because he is one of the very few that starts from
audible effects (the colouredness of high Q mimimum phase filters) and
tries to connect them to certain technical features.
I find general statements like "bass equalisation with linear phase
filters sounds bad because of pre-ringing" are not warranted by the
current state of knowledge about the human auditory system. Neither are
statements like "linear phase equalisation sounds better because it
delays all frequency components by an equal amount independent of
magnitude response".
Really, I don't know why we argue. We both agree that scientific
listening tests should be conducted. However, until such results are
published, I try to refrain from general statements as the above.
Regards,
Andor
Reply by Jerry Avins●April 3, 20052005-04-03
Bob Cain wrote:
>
>
> Jerry Avins wrote:
>
>> Oh? The delays at all frequencies are matched. What do you mean by
>> timing?
>
>
> Energy begins to emerge from a drum before it is hit.
>
>
> Bob
But the dominant rise is exactly delayed. The price one pays for
avoiding preringing is time ambiguity of the dominant rise. A large bell
and a triangle struck simultaneously might sound as if that had been
struck sequentially.
It's east to know what to do when the goal is fidelity. When the aim is
distorting something, you have to pick your poison.
Jerry
--
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������
Reply by Bob Cain●April 2, 20052005-04-02
Fred Marshall wrote:
> It certainly preserves *something*
> better but what should that be called?
Exactly what I've come to wonder.
> Minimum phase filtering is preferable in audio systems or systems where
> latency and pretransients are important because minimum phase filters
> minimize pretransients (at the cost of changing the phase relationships of
> individual spectral components). The latter is apparently (much?) less
> noticeable than the former. Somehow I recall someone saying that the ear is
> not phase sensitive.
But it is certainly TOA sensitive which belies that to some
extent.
> So, while a heard waveform might have amplitude or
> envelope differences because of phase distortion, does the ear perceive it?
> I'm sure there are pathological cases where it does. However, for general
> sounds, is this a major factor? The suggestion was that it isn't. So,
> maybe that's all there is to it.
As with all things, it depends. Upon what is a question not
well enough explored.
Bob
--
"Things should be described as simply as possible, but no
simpler."
A. Einstein
Reply by Bob Cain●April 2, 20052005-04-02
Jerry Avins wrote:
> Oh? The delays at all frequencies are matched. What do you mean by timing?
Energy begins to emerge from a drum before it is hit.
Bob
--
"Things should be described as simply as possible, but no
simpler."
A. Einstein
Reply by glen herrmannsfeldt●April 2, 20052005-04-02
Fred Marshall wrote:
(big snip)
> Minimum phase filtering is preferable in audio systems or systems where
> latency and pretransients are important because minimum phase filters
> minimize pretransients (at the cost of changing the phase relationships of
> individual spectral components). The latter is apparently (much?) less
> noticeable than the former. Somehow I recall someone saying that the ear is
> not phase sensitive. So, while a heard waveform might have amplitude or
> envelope differences because of phase distortion, does the ear perceive it?
> I'm sure there are pathological cases where it does. However, for general
> sounds, is this a major factor? The suggestion was that it isn't. So,
> maybe that's all there is to it.
I would say not very phase sensitive. My understanding is that the
nerve impulses for low frequencies are generated on a certain part of
the cycle, and for high frequencies only on some cycles.
I believe that the timing information (to do stereo imaging) is done
separately from sound (frequency, etc.) information. I don't know how
much effect phase has on each, but it might be very different.
> It's probably important to say something about "minimum phase". Minimum
> phase means minimum delay at each frequency doesn't it? Whereas, linear
> phase means flat delay and we know that the delay is longer than the minimum
> at each (or most) frequency(ies). So, when there is a wideband transient we
> want all of the frequency components to be delayed the least in order
> minimize the delay of the transient. To one degree or another, the
> transient will be distorted in amplitude but what does that mean?
> If the transient includes a long precursor or pretransient then that could
> be viewed as an amplitude distortion also couldn't it? It's just of a
> different type.
I am not so sure about this, but it sounds fine to me.
-- glen
Reply by Jerry Avins●April 2, 20052005-04-02
Fred Marshall wrote:
> "Jerry Avins" <jya@ieee.org> wrote in message
> news:2sqdnd60nNzVR9PfRVn-rg@rcn.net...
>
>>Bob,
>>
>>I want to (mostly) amplify what you wrote.
>>
>>Bob Cain wrote:
>>
>>>
>>>Andor wrote:
>>>
>>>
>>>>Bob Cain wrote:
>>>>
>>>>
>>>>>Philosophy? I don't understand your relegation of the
>>>>>question to philosophy when there are clear physical
>>>>>implications about when within a sampled sequence things are
>>>>>heard after filtering.
>>>>
>>>>
>>>>
>>>>It is philosphizing (musing) like Aristoteles practiced it: think hard
>>>>about a physical problem and then come up with a result based solely on
>>>>"logical" deduction.
>>>
>>>
>>>You're wrong. First I saw it on the audio waveform display of some
>>>binaural recordings I'd done and equalized with linear phase filters to
>>>attempt to preserve timing info as I thought linear phase would do. What
>>>I saw prompted me to do some listening tests of the same filters linear
>>>and minimum phase to see if the difference audible. I eventually did do
>>>randomizing of blocks within a filtered piece, some filtered one way and
>>>some the other, and although the effect is subtle I was able to reliably
>>>tell which was which. This is not an acceptable protocol for publication
>>>but it satisfied me that there was an identifiable effect. I found that
>>>I preferred the minimum phase because it sounded crisper without added
>>>high frequency content. It was empirical observation and
>>>experimentation, not at all what you are describing.
>>
>>I hear it too. A mathematically-minded colleague built himself a
>>linear-phase preamp -- bass, center and treble controls -- as well as
>>equalizer for his magnetic phono cartridge. He was disappointed in the
>>sound and invited several of us to listen and share ideas. His thought was
>>some flaw in the ADC-DAC chain which used the same modules (yes, modules,
>>not chips!) that I used for instrumentation.
>>
>>I was able to quickly point out that the records were made with
>>minimum-phase preemphasis and roll-off, and that restoring the amplitude
>>with a linear-phase equalizer actually distorted the phase. Running the
>>phono cartridge into a an analog pre-amp and that into a line input
>>quickly showed the correctness of that idea, but extreme positions of the
>>tone controls still sounded unnatural, Boost was worse, presumably because
>>the offending effect was louder. We couldn't at the time assign a reason
>>except to say vaguely. "Analog seems to sound better." Eventually, someone
>>suggested redoing the tone-control code with IIRs, and the "analog is
>>better" consensus went away.
>>
>>
>>>I no longer think that the effect of linear phase is that of preserving
>>>timing relationships such as we have long thought it was.
>>
>>Oh? The delays at all frequencies are matched. What do you mean by timing?
>>
>
>
>
> Jerry,
>
> I posted a response but in another slot in this thread. I think the real
> issue here is that we're using mixed metaphors. We talk about phase and we
> talk about delay in the same breath as we talk about pretransients, etc.
> So, here is a try at unifying the discussion:
>
> The usual definition of phase only applies *after* the transient effects are
> gone. That doesn't mean that we can't measure phase after a relatively
> short time, but it does mean that we can't measure it in too short a time.
>
> Linear phase *does* preseve timing relationships - but only after the
> transient. I don't know, might it be said that minimum phase preserves
> *transient timing relationships* better? It certainly preserves *something*
> better but what should that be called?
>
> Maybe one good model would be a simple delay. Here everything is preserved,
> transient, amplitude and phase. And, a simple delay is a linear phase
> device of the simplest type. A very high fidelity recording fits into this
> category - with the delay being arbitrary and very long indeed! And, except
> that the delay is arbitrary, we can't measure phase relationships until some
> time after the playback is initiated - particularly at the lowest
> frequencies.
> We can contrast this with a linear phase lowpass filter where there is a
> nonzero pretransient.
>
> I learned something in this discussion because I had not understood how
> minimum phase filters affect the psychoacoustics - as compared to linear
> phase filters. Much the same as the example given. So, to apply what I've
> learned here, this is what might be said(?):
>
> Minimum phase filtering is preferable in audio systems or systems where
> latency and pretransients are important because minimum phase filters
> minimize pretransients (at the cost of changing the phase relationships of
> individual spectral components). The latter is apparently (much?) less
> noticeable than the former. Somehow I recall someone saying that the ear is
> not phase sensitive. So, while a heard waveform might have amplitude or
> envelope differences because of phase distortion, does the ear perceive it?
> I'm sure there are pathological cases where it does. However, for general
> sounds, is this a major factor? The suggestion was that it isn't. So,
> maybe that's all there is to it.
>
> It's probably important to say something about "minimum phase". Minimum
> phase means minimum delay at each frequency doesn't it? Whereas, linear
> phase means flat delay and we know that the delay is longer than the minimum
> at each (or most) frequency(ies). So, when there is a wideband transient we
> want all of the frequency components to be delayed the least in order
> minimize the delay of the transient. To one degree or another, the
> transient will be distorted in amplitude but what does that mean?
> If the transient includes a long precursor or pretransient then that could
> be viewed as an amplitude distortion also couldn't it? It's just of a
> different type.
Fred,
Thanks? for starting me thinking. (The question mark indicates that
maybe we're thinking too much.) When listening is the criterion, we
usually want the best fidelity we can get -- low distortion and flat
response both in frequency and phase. There are exceptions. Some may
want to boost the bass to emphasize the oom-pah-pah quality that they
like. Others, guitarists, say, may want to generate distortion. Let's
ignore the use of filters and nonlinearities to generate special
effects. The criterion of "naturalness" doesn't apply to those cases
anyhow. So why use filters at all? Basically, as equalizers. When the
original frequency distortion is linear phase (absorption of highs in a
dead room?), use a minimum-phase equalizer. When the original frequency
distortion is minimum phase (most physical networks; phono preemphasis),
use a minimum-phase equalizer. I doubt that one can get better sound
than by those rules in those simple cases.
Crossover filters aren't as simple as that. Their results depend
strongly on speaker placement, so they can't be treated in isolation.
Crossover filters are rarely sharp enough to generate much preverb (OK,
so it's not my coinage) and are rarely set so low that preverb persists
for long. Nevertheless, I don't think one does much better in general
than Linkwitz-Reilly, and that happens to be minimum phase.
Whatever, I think that is needs to be decided by listening case by case,
that it often doesn't matter, and that advertised use of linear phase is
very often hype for the ignorant.
Jerry
--
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������
Reply by Fred Marshall●April 2, 20052005-04-02
"Jerry Avins" <jya@ieee.org> wrote in message
news:2sqdnd60nNzVR9PfRVn-rg@rcn.net...
> Bob,
>
> I want to (mostly) amplify what you wrote.
>
> Bob Cain wrote:
>>
>>
>> Andor wrote:
>>
>>> Bob Cain wrote:
>>>
>>>> Philosophy? I don't understand your relegation of the
>>>> question to philosophy when there are clear physical
>>>> implications about when within a sampled sequence things are
>>>> heard after filtering.
>>>
>>>
>>>
>>> It is philosphizing (musing) like Aristoteles practiced it: think hard
>>> about a physical problem and then come up with a result based solely on
>>> "logical" deduction.
>>
>>
>> You're wrong. First I saw it on the audio waveform display of some
>> binaural recordings I'd done and equalized with linear phase filters to
>> attempt to preserve timing info as I thought linear phase would do. What
>> I saw prompted me to do some listening tests of the same filters linear
>> and minimum phase to see if the difference audible. I eventually did do
>> randomizing of blocks within a filtered piece, some filtered one way and
>> some the other, and although the effect is subtle I was able to reliably
>> tell which was which. This is not an acceptable protocol for publication
>> but it satisfied me that there was an identifiable effect. I found that
>> I preferred the minimum phase because it sounded crisper without added
>> high frequency content. It was empirical observation and
>> experimentation, not at all what you are describing.
>
> I hear it too. A mathematically-minded colleague built himself a
> linear-phase preamp -- bass, center and treble controls -- as well as
> equalizer for his magnetic phono cartridge. He was disappointed in the
> sound and invited several of us to listen and share ideas. His thought was
> some flaw in the ADC-DAC chain which used the same modules (yes, modules,
> not chips!) that I used for instrumentation.
>
> I was able to quickly point out that the records were made with
> minimum-phase preemphasis and roll-off, and that restoring the amplitude
> with a linear-phase equalizer actually distorted the phase. Running the
> phono cartridge into a an analog pre-amp and that into a line input
> quickly showed the correctness of that idea, but extreme positions of the
> tone controls still sounded unnatural, Boost was worse, presumably because
> the offending effect was louder. We couldn't at the time assign a reason
> except to say vaguely. "Analog seems to sound better." Eventually, someone
> suggested redoing the tone-control code with IIRs, and the "analog is
> better" consensus went away.
>
>> I no longer think that the effect of linear phase is that of preserving
>> timing relationships such as we have long thought it was.
>
> Oh? The delays at all frequencies are matched. What do you mean by timing?
>
Jerry,
I posted a response but in another slot in this thread. I think the real
issue here is that we're using mixed metaphors. We talk about phase and we
talk about delay in the same breath as we talk about pretransients, etc.
So, here is a try at unifying the discussion:
The usual definition of phase only applies *after* the transient effects are
gone. That doesn't mean that we can't measure phase after a relatively
short time, but it does mean that we can't measure it in too short a time.
Linear phase *does* preseve timing relationships - but only after the
transient. I don't know, might it be said that minimum phase preserves
*transient timing relationships* better? It certainly preserves *something*
better but what should that be called?
Maybe one good model would be a simple delay. Here everything is preserved,
transient, amplitude and phase. And, a simple delay is a linear phase
device of the simplest type. A very high fidelity recording fits into this
category - with the delay being arbitrary and very long indeed! And, except
that the delay is arbitrary, we can't measure phase relationships until some
time after the playback is initiated - particularly at the lowest
frequencies.
We can contrast this with a linear phase lowpass filter where there is a
nonzero pretransient.
I learned something in this discussion because I had not understood how
minimum phase filters affect the psychoacoustics - as compared to linear
phase filters. Much the same as the example given. So, to apply what I've
learned here, this is what might be said(?):
Minimum phase filtering is preferable in audio systems or systems where
latency and pretransients are important because minimum phase filters
minimize pretransients (at the cost of changing the phase relationships of
individual spectral components). The latter is apparently (much?) less
noticeable than the former. Somehow I recall someone saying that the ear is
not phase sensitive. So, while a heard waveform might have amplitude or
envelope differences because of phase distortion, does the ear perceive it?
I'm sure there are pathological cases where it does. However, for general
sounds, is this a major factor? The suggestion was that it isn't. So,
maybe that's all there is to it.
It's probably important to say something about "minimum phase". Minimum
phase means minimum delay at each frequency doesn't it? Whereas, linear
phase means flat delay and we know that the delay is longer than the minimum
at each (or most) frequency(ies). So, when there is a wideband transient we
want all of the frequency components to be delayed the least in order
minimize the delay of the transient. To one degree or another, the
transient will be distorted in amplitude but what does that mean?
If the transient includes a long precursor or pretransient then that could
be viewed as an amplitude distortion also couldn't it? It's just of a
different type.
Fred
Reply by Jerry Avins●April 2, 20052005-04-02
Bob,
I want to (mostly) amplify what you wrote.
Bob Cain wrote:
>
>
> Andor wrote:
>
>> Bob Cain wrote:
>>
>>> Philosophy? I don't understand your relegation of the
>>> question to philosophy when there are clear physical
>>> implications about when within a sampled sequence things are
>>> heard after filtering.
>>
>>
>>
>> It is philosphizing (musing) like Aristoteles practiced it: think hard
>> about a physical problem and then come up with a result based solely on
>> "logical" deduction.
>
>
> You're wrong. First I saw it on the audio waveform display of some
> binaural recordings I'd done and equalized with linear phase filters to
> attempt to preserve timing info as I thought linear phase would do.
> What I saw prompted me to do some listening tests of the same filters
> linear and minimum phase to see if the difference audible. I eventually
> did do randomizing of blocks within a filtered piece, some filtered one
> way and some the other, and although the effect is subtle I was able to
> reliably tell which was which. This is not an acceptable protocol for
> publication but it satisfied me that there was an identifiable effect.
> I found that I preferred the minimum phase because it sounded crisper
> without added high frequency content. It was empirical observation and
> experimentation, not at all what you are describing.
I hear it too. A mathematically-minded colleague built himself a
linear-phase preamp -- bass, center and treble controls -- as well as
equalizer for his magnetic phono cartridge. He was disappointed in the
sound and invited several of us to listen and share ideas. His thought
was some flaw in the ADC-DAC chain which used the same modules (yes,
modules, not chips!) that I used for instrumentation.
I was able to quickly point out that the records were made with
minimum-phase preemphasis and roll-off, and that restoring the amplitude
with a linear-phase equalizer actually distorted the phase. Running the
phono cartridge into a an analog pre-amp and that into a line input
quickly showed the correctness of that idea, but extreme positions of
the tone controls still sounded unnatural, Boost was worse, presumably
because the offending effect was louder. We couldn't at the time assign
a reason except to say vaguely. "Analog seems to sound better."
Eventually, someone suggested redoing the tone-control code with IIRs,
and the "analog is better" consensus went away.
> I no longer think that the effect of linear phase is that of preserving
> timing relationships such as we have long thought it was.
Oh? The delays at all frequencies are matched. What do you mean by timing?
...
Jerry
--
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������