[RWP] Latency, is shorter always better?

[Keith Hinton]

Did you see my toy computer reference Chris buddy? LOL

On 7/22/14, Chris Belle <cb1963 at sbcglobal.net> wrote:
> Not supprising with audacity since it does not support asio and you have
> to manually set the offset to get tracks to play back in time.
>
>
> On 7/22/2014 10:37 AM, Jim Snowbarger 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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-- 
Sincerely,

Keith

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