[RWP] Latency, is shorter always better?

[Chris Belle]

Yes 'grin'.

On 7/22/2014 8:58 PM, Keith Hinton wrote:
> 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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>>>>>
>>>>
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