[RWP] Latency, is shorter always better?

Tue Jul 22 11:48:59 EDT 2014

We expect a little baby toy  computer to become a huge, furry kitty
cat with paws waving around and a tail that swishes and a meow to
escape the voicebox of the CPU. LOL just kidding.
Couldn't resist.
Fun thread for sure, I'm liking reading all about this stuff.
I didn't know digital audio was so latency-crazy.
But apparently it is.
As for 30 Gigahurts?
Why not 1 billion gigahurt processors? LOL!
Could you imagine how high your electrical bill would be?
one billion gigahurts with a quadrillion cores.
Bet you couldn't even run something like that. LOL.
Not realistically.

On 7/22/14, Jim Snowbarger <Snowman at snowmanradio.com> wrote:
> Tape decks with a moveable playback head?  Very nice.  I didn't know about
> that.  But, that would have been great for adjusting the rate of
> regenerative feedback to match the tempo of a song.
> Another way to do that was to vary the speed, except that it had frequency
> response implications as well.
>
> An interesting experiment with DAWs, record a series of clicks into track 1.
>
> Arm track 2, and route the playback of track 1 into the input for track 2.
> Do it externally, rather than inside the computer, so you get the benefit of
>
> the entire chain of processing.
>
> Now, play tracks 1 and 2 back together.  Listen for the time delay between
> the clicks on the two tracks.  Ideally, they will be coincident.
> On reaper here, it isn't quite perfect, but it is pretty darned good.
> Last time I tried Audacity, they arrived in different time zones.
>
> ----- Original Message -----
> From: "Chris Belle" <cb1963 at sbcglobal.net>
> To: "Reapers Without Peepers" <rwp at reaaccess.com>
> Sent: Tuesday, July 22, 2014 12:46 AM
> Subject: Re: [RWP] Latency, is shorter always better?
>
>
>> One more thought  about this latency thing.
>>
>> And us old farts who used to play with tape decks will remember this.
>>
>> How about those 3 head decks where you could listen to play-back while
>> recording?
>>
>> YOU could hear the input while it was going down, and listen to your
>> playback head, and some of those were moveable.
>>
>> So you could change the latency between when something got recorded and
>> played back by moving the head closer or further away from the recording
>> head.
>>
>> That was on the commercial machines.
>>
>> I never had one of those, but i did have a very nice 3 head cassette
>> deck.
>>
>> This is, in fact, somewhat similar to what hapens in your daw,
>> even if your not recording but just listening to the signal comming back
>> from your daw once it goes through the internal processing, and any
>> plug-ins you might have,
>> and they add their own latency you can bet, and most modern daws
>> compensate for that under the hood.
>>
>> Sonar had automatic plug in delay compensation
>> way before many daws, including protools ever had it.
>>
>> Yes, you go back and listen to first episodes of the home recording show
>> on protools 9 and 10,
>> and you can hear them talking about lining up tracks manually after the
>> fact, to make the audio come out right, after going through all the
>> processing plugs.
>>
>> Boy howdy, now, isn't that a real pain in the posterior?
>>
>> INteligently keeping up with when in the time line a recording starts, and
>>
>> how to play it precisely in the right way to account for plug-ins
>> latencies, and then play it properly again when you take plugs of is not
>> an easy task, but your daw does that all for you under the hood, if it's
>> worth a squat.
>>
>> People sometimes get in real trouble even with this automatic
>> stuff going on by not routing their monitoring right,
>> because there are certain ways of routing and recording which makes it
>> impossible for your daw to implement delay compensation properly.
>>
>> So this is why I tend to like to not do plugs until after I've laid my
>> audio.
>>
>> Not always possible, you can't lay that heavy rock guitar track easily
>> only hearing plink, plunk, twang,
>> but you can believe your daw is doing the latency shuffle dance when you
>> have many tracks playing and you are laying guitar amp simms which has
>> latency going both ways, because remember, you are going audio in, and
>> audio back out,
>> and this is why with mastering plugs which cause a lot of latency,
>> especially multi-band compressors with look ahead and back in my early
>> early days of learning this stuff it used to drive me nuts, why are my
>> midi tracks being delayed so much when I press a note but they play just
>> fine on playback?
>>
>> Well, it's that delay compensation working for you.
>>
>> Imagine having to figure out how much delay you had and fixing all that
>> manually?
>>
>> You can get interesting things happening when using reverb in projects by
>>
>> turning off delay compensation,
>> you get a built in pre-delay, which is a setting on high quality reverb
>> units, the reverb doesn't start right away, and
>> this helps make room in the mix when you don't wan the verb in the way,
>> and it kicks in after the initial atack of your audio.
>>
>> Or do we remember real world latency,
>> and the days when distructive editing was the only kind you did, if you
>> wanted to process an equalizer, or chorus fx, you hit the button, and then
>>
>> go have a sandwich and waited for your 486 to process that track, and
>> you'd come back 10 minutes later and maybe have a wet and dry track.
>> and you could do interesting things with that by time delaying the wet
>> track 'grin'.
>>
>> When I do drum replacement by generating midi tracks from transient points
>>
>> of an audio drum track and then feeding it to a audio bus with samples, I
>>
>> have to time align the new track to match the old one,
>> at least in the old days we had to do more of that before delay
>> compensation was automatic.
>> in most daws.
>>
>> Still, most daws will only do this in a certain range, see above, where I
>>
>> mention mastering plugs,
>> linear phase equalizers are also notorious for introducing way too much
>> delay to use them in real time.
>>
>> So are transient  processors, shapers.
>>
>> Maybe when we get processors running at 30 gigahertz
>> we'll be able to do that stuff in real time, and did I hear silly people
>> want to make a daw out of an ipad?
>>
>> Right now in 2014, an ipad will just barely run a guitar simm with low
>> enough latency
>> \to be playable.
>>
>> Well, what do you expect from a little baby toy computer?
>>
>>
>> On 7/21/2014 9:33 PM, Jim Snowbarger wrote:
>>> Now and then, I feel like a slight departure from topic..  And, this is
>>> one of them.  So, stand bye with your delete key ready as I carry on.
>>>
>>> This probably belongs over on MidiMag.  But, I don't feel like joining
>>> just so I can post this once in a blue mooner.
>>>
>>> One of the great things that digital audio processing has brought to us
>>> is so-called latency.  You might just call it delay.  but, in the 21st
>>> century, we like to use clever names.  It makes us feel smarter.  So,
>>> let's co-opt the term latency, which had a totally different meaning
>>> before the techno-gods got hold of it. And, let's now define latency as
>>> the act of being late.  But, however you slice it, it comes down to
>>> delay.
>>>
>>> Digital devices impose delay mostly because data consumers, like sound
>>> cards, or recording devices, have learned to be defensive, knowing full
>>> good and well that data providers, such as input sound cards, or other
>>> streaming devices, can not be counted on to keep up a steady stream of
>>> data.  Internet congestion, or scheduling congestion inside your own
>>> machine, can temporarily block the normal flow of things.  Sound playback
>>>
>>> requires a rock-solid comsumption rate of the data. The sampels need to
>>> keep flowing. You might not get that next buffer load of data in time.
>>> so, it pays to keep a backlog.  The more backlog, the safer you are.
>>> But, if the backlog is too great, you get, latency, that annoying delay.
>>>
>>> I recently picked up one of those fine Computers Chris is always talking
>>>
>>> about from StudioCat.com.  That is one very fine box. And, now that I
>>> also own Chris's Delta 1010, I was enjoying fine-tuning my latency down
>>> to acceptable levels,  not carefully measured, but clearly less than 10
>>> milliseconds.
>>>
>>> Most of the recording work I do involves a microphone and headphones.  I
>>>
>>> am quite typically listening to my own voice as I speak.  If you have
>>> listened to the Snowman Radio Broadcasts, you know the kind of
>>> multi-track microphone work I'm guilty of.
>>> When living on machines where such short delays were not possible, my
>>> habit was to listen to my own foice direct out of the mixer, and not
>>> going through Reaper.  So, I kept the reaper monitor off. What was
>>> annoying about that is that, if I panned my various character voices in
>>> the stereo  mix, then, my direct microphone sound would not be panned the
>>>
>>> same as the character voice track I was recording into.  So, when it
>>> played back, it came from elsewhere, and was more than a little bit
>>> confusing.
>>>
>>> But, with delay this short, I find that I switch off the direct sound,
>>> and now can monitor the signal coming back from reaper with the monitor
>>> turned on.  So, I'm now listening to a delayed version of my voice, and
>>> it is panned to the same place where that character voice sits, which
>>> helps me keep track of who I am supposed to be right now.  And, I can
>>> more easily tell now whetehr a track is armed, and even if one is armed
>>> that should not be. It's nice to be able to work like that, just
>>> listening to reaper's output.
>>>
>>> But, here is the cool thing.  The exact amount of latency you provide
>>> affects the quality of what you hear in your headphones.
>>>
>>> No matter how good your phones, the sound that you hear when you are
>>> listening to yourself speaking live into a microphone, is actually the
>>> composite of at least two signal paths, and maybe more.  Yes, there is
>>> the direct signal coming through Reaper. Then, there is bone
>>> conductivity, the sound of your own voice coming through the structure of
>>>
>>> your head, which will very somewhat with density.  If you don't get any
>>> of that, you might wonder about that density stuff.
>>> And maybe even, there is leakage around the ear muffs.  In all, it is a
>>> complex sond that actually reaches your ears.  And, the phase
>>> relationship between all of the various contributors will affect the
>>> frequency response of the final signal that you hear.
>>>
>>> In the old days, we knew about the affect that phase would have on such
>>> things.  Having your head phones out of phase with your microphone left
>>> you feeling empty headed, due to the phase cancellation that took place.
>>> But, since delay was in the nanoseconds, we didn't get to know so much
>>> about the effect that delay would have, despite our compulsive
>>> preoccupation with tape delay.
>>>
>>> Phase is mostly a frequency independent phenomenon.  Yes, we know that
>>> some systems, especially mechanical transducers, or even cheap
>>> equalizers, which will have a reactive component to their impedance,
>>> introduce a variable amound of phase shift, depending on frequency.  But,
>>>
>>> usually those effects are at the far ends of their usable range.
>>> In general, especially in mixer land, where things are nice and linear,
>>> and where impedances are strictly non-reactive, if you put something 180
>>>
>>> degrees out of phase, you will get perfect cancellation, all across the
>>> frequency band.
>>>
>>> Enter the digital age, and the new innovation, latency.
>>> The relationship between signal phase, and a delay is frequency. For
>>> example, a delay of 4 milliseconds is one full cycle of a 250 Hertz tone.
>>>
>>> But, it is only half a cycle of a 125 hertz tone.   It is all still a 4
>>> millisecond delay.  But, the phase impact depends on the frequency.
>>> Combining the pre and post delays of these two tones with that 4ms delay
>>>
>>> will have completely different effects. The 125 hertz tone would be
>>> nulled out.  The 250 hertz tone would actually see a 6 db increase.
>>>
>>>
>>> The result is that, if you put a delay in front of your headphone mix,
>>> you will cause what is referred to as a comb filter effect on the
>>> perceived headphone signal.  It is a filter that has a frequency response
>>>
>>> curve that looks like a rola coaster, with hills and valleys.  If you
>>> were listening to an audio tone sweep, one that you would actually need
>>> to sing, in this case, in order to get that bone conductivity thing
>>> happening as well, As you move steadily up in frequency, the sound would
>>>
>>> be much stronger at some frequencies, and much weaker at others.  As the
>>>
>>> tone rises, you would hear rising and falling of the net response.  And,
>>>
>>> changing the amounbt of delay slides that comb up and down the audio
>>> spectrum.
>>> Depending on several things, the frequency range of your voice, response
>>>
>>> of your headphones, your ears, the density of your grey matter, your
>>> preferences, and on and on, you might have preferences about the optimal
>>>
>>> position of that comb.  What frequencies do you like to accentuate?  And,
>>>
>>> which to attenuate.
>>>
>>> The cool thing is that, by fine-tuning your headphone latency, you can
>>> position that comb how you like, and can optimize your headphone
>>> experience. The latency needs to be short enough to not give you a delay
>>>
>>> echo effect. But, beyond that, the shortest possible latency may not give
>>>
>>> you the headphone experience you like.  Instead, relax it a little, and
>>> see what enriching tones come your way.
>>> Silly you.  And you always thought shorter was better.  And now you
>>> know.
>>> TROTS.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
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>>>
>>
>>
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-- 
Sincerely,

Keith

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