[RWP] Latency, is shorter always better?

[Chris Belle]

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.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> RWP mailing list
>>> RWP at reaaccess.com
>>> http://reaaccess.com/mailman/listinfo/rwp_reaaccess.com
>>>
>>
>>
>> _______________________________________________
>> RWP mailing list
>> RWP at reaaccess.com
>> http://reaaccess.com/mailman/listinfo/rwp_reaaccess.com
>>
>
>
>
> _______________________________________________
> RWP mailing list
> RWP at reaaccess.com
> http://reaaccess.com/mailman/listinfo/rwp_reaaccess.com
>