[linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
28 Oct
2005
28 Oct
'05
9:22 a.m.
Hi all,
I would like to route a microphone through a sound card and back to
powerful amplified speakers.
As we know in analog PA gear you have the microphone feedback problem
(usually it comes in form
of high pitched whistle sounds).
But if I route a mic from into the soundcard and out to the speakers,
there will be a small delay
due to the audio card buffers. Even if it's only a few msecs it makes
the problem much worse
than in the case of analogue gear because the feedback sound will come
in chunks that's one audio card buffer at time.
For example if I use 64 samples per buffer which gives me acceptable
latency for a live singer, the feedback noise
could possibly generate a much lower pitched signal/interference (I
assume something like 44100/64 Hz) which is I think
not easy to filter out compared to the high pitched feedback (is in the
latter case sufficient to cut some high frequencies using an EQ ?).
How can one solve the feedback problem in case of mic to speaker delays
of let's say 5-10msecs ?
Is an echo canceller algorithm needed ? If yes does this compromise the
quality of the input signal (the singer).
I think this is a very interesting topic and it would be cool if
knowledgable people could come up with topics and ideas.
(for example people that are good at DSP, room correction etc, like Fons
A. etc).
PS: I know that cards like DELTA 1010 and others (RME) can do zero
latency monitoring (hardware pass-thru) but
I'd prefer a software based routing since you can apply effects and
stuff before forwarding the output.
thanks for infos and toughts
Benno
http://www.linuxsampler.org
28 Oct
28 Oct
4:07 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
On Friday 28 October 2005 12:26, Benno Senoner wrote:
Hi all,
I would like to route a microphone through a sound card and back to
powerful amplified speakers.
As we know in analog PA gear you have the microphone feedback problem
(usually it comes in form
of high pitched whistle sounds).
But if I route a mic from into the soundcard and out to the speakers,
there will be a small delay
due to the audio card buffers. Even if it's only a few msecs it makes
the problem much worse
than in the case of analogue gear because the feedback sound will come
in chunks that's one audio card buffer at time.
For example if I use 64 samples per buffer which gives me acceptable
latency for a live singer, the feedback noise
could possibly generate a much lower pitched signal/interference (I
assume something like 44100/64 Hz) which is I think
not easy to filter out compared to the high pitched feedback (is in the
latter case sufficient to cut some high frequencies using an EQ ?).
How can one solve the feedback problem in case of mic to speaker delays
of let's say 5-10msecs ?
Is an echo canceller algorithm needed ? If yes does this compromise the
quality of the input signal (the singer).
Actually applying a little delay is fairly standard in conventional PA as
well.
Consider that 10ms is roughly 10 feet of acoustic delay, and that you usually
want the direct sound to arrive at the audience a little (10 - 30ms) before
the sound from the PA as this really helps localisation.
Thus even on a small stage you may be delaying the FOH by 40ms or so to help
localise the singer.
Monitors are a little different obviously, but even there, it just is not a
problem at these levels.
As for feedback, you notch it out with a graphic (or better a parametric),
the same as always.
In fact it turns out that a constant delay makes the feedback slower to build,
and may alter the exact frequency slightly (by adding a constant phase shift
at any given frequency), but does not radically change the overall system
response in the way you think.
Just about every modern PA system has at least one stage of AD-DA conversion
(Think things like the digital crossovers so common in modern systems) which
probably introduces 5ms or so right there.
Regards, Dan.
29 Oct
29 Oct
1:16 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
Benno!
There are some missing parameters in your description, but I will
assumme your problem arises out of the monitor-mix rather than the
house-mix.
Suggestion: Invert the phase of the offending microphone. This should
kill dead feadback in the frequency-band currently annoying you. The
downside is that it will also emphasize other frequency-bands that did
not pose a problem before. The theory is that those resonanses will be
minor relative to your current major problem.
mvh // Jens M Andreasen
On Fri, 2005-10-28 at 13:26 +0200, Benno Senoner wrote:
Hi all,
I would like to route a microphone through a sound card and back to
powerful amplified speakers.
As we know in analog PA gear you have the microphone feedback problem
(usually it comes in form
of high pitched whistle sounds).
But if I route a mic from into the soundcard and out to the speakers,
there will be a small delay
due to the audio card buffers. Even if it's only a few msecs it makes
the problem much worse
than in the case of analogue gear because the feedback sound will come
in chunks that's one audio card buffer at time.
For example if I use 64 samples per buffer which gives me acceptable
latency for a live singer, the feedback noise
could possibly generate a much lower pitched signal/interference (I
assume something like 44100/64 Hz) which is I think
not easy to filter out compared to the high pitched feedback (is in the
latter case sufficient to cut some high frequencies using an EQ ?).
How can one solve the feedback problem in case of mic to speaker delays
of let's say 5-10msecs ?
Is an echo canceller algorithm needed ? If yes does this compromise the
quality of the input signal (the singer).
I think this is a very interesting topic and it would be cool if
knowledgable people could come up with topics and ideas.
(for example people that are good at DSP, room correction etc, like Fons
A. etc).
PS: I know that cards like DELTA 1010 and others (RME) can do zero
latency monitoring (hardware pass-thru) but
I'd prefer a software based routing since you can apply effects and
stuff before forwarding the output.
thanks for infos and toughts
Benno
http://www.linuxsampler.org
--
2:17 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
On Sat, Oct 29, 2005 at 05:16:53PM +0200, Jens M Andreasen wrote:
Suggestion: Invert the phase of the offending
microphone. This should
kill dead feadback in the frequency-band currently annoying you. The
downside is that it will also emphasize other frequency-bands that did
not pose a problem before. The theory is that those resonanses will be
minor relative to your current major problem.
(Somehow I missed Benno's original post)
Good suggestion.
Benno Senoner wrote:
> For example if I use 64 samples per buffer which gives me acceptable
> latency for a live singer, the feedback noise
> could possibly generate a much lower pitched signal/interference (I
> assume something like 44100/64 Hz) which is I think
> not easy to filter out compared to the high pitched feedback (is in the
> latter case sufficient to cut some high frequencies using an EQ ?).
There should be no difference: the processing delay has just the same
effect as any delay, for example the one introduced by speaker to mic
distance. It's in a different place in the loop, but loops being what
they are, that doesn't matter.
One thing I learned is that effect processors can make the situation
much worse. So if your problem is with the monitor mix as Jens assumed,
I'd suggest to leave out the effects there. And in that case you could
equally well use zero delay monitoring either from you card or directly
from the mixing desk.
--
FA
3:43 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
Benno Senoner wrote:
Hi all,
I would like to route a microphone through a sound card and back to
powerful amplified speakers.
As we know in analog PA gear you have the microphone feedback problem
(usually it comes in form
of high pitched whistle sounds).
This problem is common as you know, and it has been solved a long time
ago. The solution is to do a slight frequency shift to the audio.
Small frequency shifts are not noticeable to the listener, and that it
what is used at all those loud pop concerts.
So, use ardour and do a simple real time sound capture, add a frequency
shifter plugin, and then play it out all in real time.
Another application of feedback suppression is satellites where the
forward uplink frequency is slightly different from the forward downlink
frequency.
James
7:20 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
On Sat, Oct 29, 2005 at 06:43:05PM +0100, James Courtier-Dutton wrote:
Benno Senoner wrote:
Frequency shifting will give a few dB extra, not more. It's based on the
fact that a typical room response will have many very narrow peaks only
a few Hz or less apart, and shifting the frequencies will smooth out the
response as the sound circulates around the loop.
It's less useful for stage monitoring for two reasons: first, there is
usually a dominant direct path from the monitor to the artist and the mic
(otherwise the monitor isn't very useful), and the room response has less
impact on the loop gain, and second, the combination of the direct sound
and the shifted one can be very confusing for the artist.
As we know in analog PA gear you have the
microphone feedback problem
(usually it comes in form of high pitched whistle sounds).
This problem is common as you know, and it has been solved a long time
ago. The solution is to do a slight frequency shift to the audio.
Small frequency shifts are not noticeable to the listener, and that it
what is used at all those loud pop concerts. Another application of feedback suppression is
satellites where the
forward uplink frequency is slightly different from the forward downlink
frequency.
Slightly different in this case usually means at least several tens of
MHz, and often up and downlink will be in different microwave bands.
There'is only one application where up and downlink frequencies are
linked to each other, and in that case they have a constant ratio, not
difference. This makes the downlink frequency coherent to an on-ground
reference, and thus allows for Doppler measurements.
--
FA
8:41 p.m.
New subject: [linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?
On Saturday 29 October 2005 22:25, fons adriaensen wrote:
Frequency shifting will give a few dB extra, not more.
It's based on the
fact that a typical room response will have many very narrow peaks only
a few Hz or less apart, and shifting the frequencies will smooth out the
response as the sound circulates around the loop.
Actually it also causes the phase shift around the loop to be variable and
thus tends to remove the required phasing conditions in fairly short order.
Very helpful for conference work where the suits just will not speak into the
damn mic.
It's less useful for stage monitoring for two
reasons: first, there is
usually a dominant direct path from the monitor to the artist and the mic
(otherwise the monitor isn't very useful), and the room response has less
impact on the loop gain, and second, the combination of the direct sound
and the shifted one can be very confusing for the artist.
The second is the real kicker here, the difference between the direct sound
(mostly bone conduction), and the frequency shifted one can make it almost
impossible for a vocalist to work.
However monitors can almost always be tamed by correct placement and mic
choice, for example, when using hypercardoid vocal mics, two monitors placed
30 degrees off axis will be FAR better then one, look at the mic polar
diagram for why!
Ohh yea, putting the guitar cab on a flightcase so it is at ear level can
really help reduce overall stage volume! I want to find out where they teach
guitar players that they have ears in their knees, then nuke from orbit!
Regards, Dan.
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linux-audio-dev@lists.linuxaudio.org
6 comments
5 participants
participants (5)
-
Benno Senoner -
Dan Mills -
fons adriaensen -
James Courtier-Dutton -
Jens M Andreasen