Sample Delay bug

Does has anyone else noticed that Sample Delay doesn't remember its setting when a project is closed ? Or is this just a problem with my system ?
1) Insert a Sample Delay plug-in on any track and set the delay value to anything other than zero
2) Save your project, close it, re-open it
3) Check the delay value--it's reset to zero
Happens every time I close the project. Bummer.
James
[email protected]

To get rid of the playing through the channel strip thing, control-click (or right click) - it may be a long click, I don't remember - on the speaker/play icon in the sample editor and you have the option to listen through the pre-listen channel.

Similar Messages

Avoiding phase cancellation with sample delay plug in

Hi
I had soem horns in a mix sounded flat so i used the sample delay plug in to slightly delay the right side then they sounded great , really big and sat in the mix but when witching to mono they nearly dissapear.
Is there any trix to avoid this phase cancellation when swithching to mono?

Data Stream Studio wrote:
1.-I think if you reverse polarity on 1 of the channels of your mix instead of just reversing polarity of one side of the problematic horns, you may well solve one phase issue and create a whole lot of new ones.
2.-This is a good idea, maybe there's a setting that sound as fat as the one you're using but will not phase cancel in mono.
3.-I'm not sure what audio file you'd move... could you explain?
If it's a Stereo file, split it into two mono tracks, and move on of the two up/down the timeline by a few samples, until you get the least phase combing effect.
Maybe the OP needs to go back and re-mix that song, and find the culprit horn out of the whole section... if he can...

I have Photoshop Elements 6 and am dealing with the GIF frame delay bug. Must I upgrade?

I am dealing with the Photoshop Elements 6 GIF animation frame delay bug. I understand there has not been a fix for some time so I want to know if one has been developed. If not what do I have to do to address this. Can I get an upgrade? What steps do I have to take and who do I deal with at Adobe for this?

The gif delay problem persisted for many versions of PSE. You can buy PSE 12, but it's silly to buy it from adobe since their upgrade price is more than the street price for the full version at big box stores and online resellers.
If your question is will adobe give you a free upgrade because of this bug, then the answer is no.

Sample Delay - samples into millisecond delay time?

Hi guys
Hopefully a simple question and I'd love some info.
I'm mixing in 5.1 and want to simply put up a general 25 millisecond delay in the left rear channel and 35 millisecond in the right rear channel. Some room calculations for what I recorded require 25 - not 20 and 35 not 30 milliseconds...... the standard stereo Delay won't allow increments of 5 miliseconds.
I'm therefore thinking of using the SAMPLE DELAY ........ whilst I have a 24 bit 48 khz recording setting ....... is it possible that within the maximum 10000 samples I can get up to 35 milliseconds ....?
How do we make the calculation work ..... the manual says 10 centimetres mic distance is 13 samples at 44k
is it possible to get up to 35 milliseconds using this?
Many thanks
Dick

My dear Christian ....... that is just the quickest most wonderful solution to question I've ever had ..... thank you for that PERFECT answer ....... I'm a happy man!
All the very best
Dick

Sample Delay VI

Currently i'm using Labview 7.1 embedded together with speedy-33.
i had my audio input to speedy-33 from my PC sound card and i wish to delay the time of my signal by let say 1sec.
I had my left signal connect to the sample delay vi then from sample delay vi to audio output.
Audio in (only one channel,left)--> sample delay --> audio output
i should heard a delay of 1sec when a audio signal i played but there is no delay.
i use two waveform graph to observe the signal before and after the sample delay vi and indeed there is a delay as the waveform are different.

The file attach has a input sin wave connected to a sample delay vi and then to a analog output.
Waveform graph were ploted before and after the delay for comparsion.
Thanks for the help.
Attachments:
Sin sample.vi ‏36 KB

"fine/sample delay", mixing down live drums, esp for JustinC!

I'm curious about the "fine/sample delay" and "phase inversion" techniques briefly mentioned by JustinC in an older post regarding recording live drums. I understand using phase inversion on snare mics that are pointing up/down, but I'm not 100% sure about the fine delay adjustment. Is this actually shifting the audio to line up waveforms from mics that are different distances from the same source? I've done this with an upright bass before (F-hole mic and another a few feet away) and it seemed to get a much fatter, clearer sound. I really appreciate any tips you might have on how to pinpoint and solve the problems.

hey over-man,
I was prowling tonight and ran across your post. I don't know your reference specifically but I understand your question.
Sample delay regarding a multimic'ed source:
This is based on phase cancellation and reinforcement. It is initiated by everything in a multimic'ed environment. The times, distances, microphone choice, effects choice-it's everywhere.
The best approach for me (typically) is to get the most appropriate sound earlier on unless you can expect variables ahead of time and factor them in right away. We'll work with an example of simply getting a natural drum sound that will line up with existing tracks (bass, gtr, vox...).
1) You can set up the sound/mics as close as possible to fit the sound you're after (we'll breeze over these points not specific to your question).
2) Before destroying the sound with a handful of plugins adjust delays. Some assumptions can be made:
A: Overheads/Room Mics will typically be the reference for the reason that they are what you have least control over because they represent more sources (Drums in this case).
B: These Mics will often have the largest delay from the start for the reason that they are farthest from the source.
C: These Mics should be adjusted less than others, I would EQ them quite minimally. Solve any problems as early in the signal chain as possible and use only what you know you'll need. Do not apply FX you don't need and position mics to better suit your desired sound.
So your OHs/Rooms are as good as they can sound naturally and sit with the track as well as possible, you can now start adjusting delays/intrtoducing other elements.
The order can vary here but I tend to start with the tracks/mics which will be more prominent. I will use snare in this example.
Some useful tools before we resume: Logic's sample delay, Expert Sleepers' Latency Fixer, because Logic's Sample Delay will not predelay. Found here:
http://www.collective.co.uk/expertsleepers/latencyfixer.html
I would normally just load both of these up, I use full PDC and apply these AUs to each track-remember to turn off Software Monitoring!
Create an instance of Latency Fixer, set it to 2048 samples. You have now added 2048 samples of predelay to your track.
Create an instance of Sample Delay, set it to 2048 samples. You have now adjusted your track to play back at the original record position.
Copy both instances to each adjacent track.
So we now have Latency Fixer followed by Sample Delay on each track for the kit (not necessary for the OHs unless you want to shift everything a little, but I will often do this from the arrange by ms).
At this point you will not need to adjust Latency Fixer's settings and simply use sample delay to offset delay in either direction.
Bring the snare track up to a reasonable level (the closer you are the better). Adjust Sample Delay in either direction and you will notice the sound changes from phase cancellation/reinforcement. Certain frequencies will stand out more at different settings. It normally won't take much, we don't want it to sound like a delay. Someplace within +/- 300 samples is often best for a natural sound. It's a good idea to have a phase inverter (Gain PlugIn) handy for 2 reasons: It may sound better inverted. You can invert the phase to pronounce things you DON'T want to hear and then flip it around the right way after setting the delay to pronounce them. Volume has a huge bearing on interaction here, a good delay setting may sound bad after reducing the level 2dB so Sample Delay setting, volume and phase should all be updated as soon as possible.
By now you have noticed how adjusting the delay affects the sound, inverting the phase is an exaggerated example of this principle. Inverting the phase of two identical tracks cancels them entirely, applying fine delay to one will introduce some of the frequencies and leave some cancelled. You can hear it scan through which frequencies by adjusting the delay. With multiple sources, different mics and different distances these delays already exist. Delaying them so they match (based on distance) is not always the best sounding but try to start somplace close. After setting the delay and Volume as close as possible you should be able to get away with much less surgical types of PluggIng and ultimately have more natural sounding tracks. Will often comprise 75% of the sound (if you want natural tracks).
If there are dynamic effects you know you'll want to apply there are special considerations.
Many of Logic's "lookahead" paramaters delay the signal.
Compression/Gating, etc. (on a per track basis) will alter the volume significantly enough that if you know you want a track compressed... ahead of time you should probably set these as you would then adjust the delay.
I tend to use predelayed tracks as keys for the audible tracks, this allows you to get around the aforementioned lookahead problems and will provide you with more control over the "shape" or envelope.
Concerning phase inversion based on which head you mic: It is a generalized solution to mic setup, though not a hard and fast rule which applies in every case.
Delaying in this manner is aligning waveforms. Based on frequency you are causing some frequencies to stand out more or less since we are working with complex waveforms. Very small deviation from original position will affect higher frequencies most and as you get further away from the original postion it begins to affect mids and eventually low frequencies. Of course this is because the frequency/wavelength. Align and misalign the waveforms for your desired sound. It does require you listen very closely when setting it up but I really think the results are worth the effort. Hopefully you'll find this helpful, it's easier to hear the effect than to describe it, hopefully it is apparent once you hear it.
Finally, I don't have any Platinums. If it doesn't work better for you, don't believe it needs to. Hopefully others will chime in here so you can get some second opinions on the approach and/or general drum mixing techniques.
Cheers, J

Sample Delay

Hi,
with using ProTools there is a nice function that allows you to adjust the sample delay due to adding several plug-ins to a track.
this way you can compensate the physical micro-dealy created by the
poor/stressed DSP by processing that amount of data at the same time..
now, where can i find the same fuction here in LE?
how is ti called?
thanx

You're talking about plug-in delay compensation. It's built right into Logic (Express and Pro). You can switch its behaviour to act on All tracks, Audio Tracks and Instruments, or None in the Audio preferences (I'm not in front of Logic at the moment...can't remember the exact path).
You'll want to leave it on "Audio Tracks and Instruments" when you're tracking, and "All" when mixing. Other than that, it's automatic. No adjustments required.

Do you agree this is a real delay bug in Flash?

Hi,
Over the last year I was asking on forums about audio delay
problems in my swf's. However I did not receive any answer that
could help me. Maybe text only was not enough to explain my problem
so now I have made a very simple demo fla and attached it to a post
(see:
http://board.flashkit.com/board/showthread.php?t=741867
Problem: There are circumstances where the audio clock of a
swf is stopped for some time and so unacceptable audio delays
occur.
Open errorDemo.fla. Please read the comment in the second
frame where the few lines of code are explained. Then do "test
movie" (ctrl-enter). Now you will hear the audio clock. Now with
the mouse: move the movie player window on your screen. You will
hear the audio clock is paused. Worse is when pressing on the title
bar (on the bar with "errorDemo.swf") with the left mouse button.
Now the clock stops even longer (about 250 ms in my case). The same
happens when another windows program is started.
There seems to be NO relation with CPU usage and it happened
in the same way on several PC's an PPC's and with several Flash
player versions and different framerates. My question is: Is it a
bug or is there somehow a solution that the audio clock of the
movie keeps running when the screen is used?
Of course there is a possibility I am doing something wrong
or maybe my expectations for Flash are to high. Please tell me!
Thanks for your reply, Paul

Thanks, dzedward,
I posted a reply with quite some effort in it, but it is
dissapeared after pressing the reply button. what I as explaining
was that I use an attached sound and so streaming is not an issue.
Please look at the few lines of code I use in my demo (please use
the link above for the fla with the WAV):
audioClock=new Sound(createEmptyMovieClip("sound1",1));
audioClock.attachSound("theSync");
audioClock.setVolume(90);
stop();
//When playing a swf (in most cases) the audio clock is on
every 46.4399093 ms.
//A onSoundComplete function only starts ON a clock puls!
That explains "gaps"
//when people complain working with audio loops in Flash.
//The audio sample used in this test has a 200 ms duration.
It can't be shorter.
//So it is started 25 ms from the end. When the sample is at
the end, the flash player
//pauses for 46.4399093 minus 25 ms, and starts the audio
sample from the same point
//again ( that is at the next clock puls). That's fine for
hearing audio clock and
//this test:
audioClock.start(0.175,1);
audioClock.onSoundComplete = function() {
audioClock.start(0.175,1);

Triggered dig. input, 1.st sample delay?

I find the "Start & Stop Trig Input.vi" useful, and want to use it in an application where I sample the state of five photo detectors. When a vehicle passes a detector, I want to time stamp the sample. I collect the samples in an array, and search the array after the acquisition ends. Sample index devided with sample rate is then the time. The problem is that I get a varying time offset from the start trigger (sample 0) to my time stamps. If I wire a bit directly to the trigger pin, I get a time difference from about 10ms to several hundred ms between the trigger and that bit. My DAQ is a DIO-32HS.
Gunnar

Hi Gunnar,
Let me make sure I understand you correctly. You are trying to perform pattern input at a specific rate and you want to be able to timestamp your inputs. You notice that when you apply a rising edge signal to both, the trigger and a bit, there is an acquisition difference of 10ms-100ms. What type of trigger are you using? Is it a pattern match? What is your sampling rate?
Are you using the control "pts after stop trigger"? Does your delay correspond to this number of points divided by your frequency? What I would try using is a simple "Buffered Pattern Input-Trig.vi" from the shipping examples with LabVIEW and connect your trigger to one of the bit lines. I've tested it on my PCI-6534 and I see my signal as the first data point in my arra
y. Anyway, let me know what specifics you are dealing with and I might be able to help you out. Have a good day.
Ron
Applications Engineer
National Instruments

LP8 Sample Editor Bug

Selecting a section of audio (say in the middle) and turning on cycle should play the selection from the start and end points selected. Instead it continues past the end point without cycling, playing the entire wave to the end and then cycles back to the very start of the audio and not as selected.
As above with cycle turned off, again it starts correct but continues past the end points.
This also intermittently happened in LP7.
Reported to Apple.

Hi Peter,
When you ask are you using the local cycle in the Sample Editor, yes?
I take it you are referring to the speaker play icon (spacebar) and cycle between view and volume?
Then the answer is yes
Can send you a small movie to demonstrate.
Thanks for trying it out your end?
I just tried it again with the same results ;-(
Logic user since Atari

Vendor id has changed to the name of the sample! Bug??

I started filling in the information on the itmsp file. Then I closed it and reopened it.
where before the Vendor ID was a number, now it has the file name of the Sample?? I haven't finished the itmsp file so I don't know what will happen when i try and upload but I'm guessing the program won't like it!
anyone had a similar problem or know hot to fix this??

Hello Bett_n, I have the exact same problem. Did you solve it in the meantime? Any tips?

Sample editing bug in 10.1

For some reason the transient editing mode/slicing in LPX 10.1 is not working correctly.
Have a look at this video and you'll see that the slices i make are not reflective of the exs24 instrument created.
VIDEO:
http://cl.ly/0e1d0A042m2H
Any input would be great. Thanks

Hey Benji,
I've been working on a project this am.... uisng the cionvert transients to Sampler feature and no such issues here..
Manually created or moved transients are accurately reflected in the EXS instrument created no matter what i have tried to duplicate your issue.
Same settings as yourself...
Except.... you have a plugin (Elastique) on your Audio track.... remove the plugin and see if that affects things

Color Sampler Tool bug?

Am I nuts here, or is this something with my particular machine? I'm running Photoshop CS4 (11.0.1) and I'm seeing the following:
Create a new image.
Add a folder.
Add a mask to that folder (I've done this with a raster mask, not a vector)
Create an adjustment layer inside that folder (I've tried this with curves, hue/sat, and/or channel mixer).
Duplicate the folder, and invert the mask on the copy.
Highlight an adjustment layer (either one).
Put out a color sampler inside the area masked by the current folder/adjustment layer. This becomes sampler #1. This shows the current values (correctly).
Put out a color sampler OUTSIDE the area masked by the current folder/adjustment layer. This becomes sampler #2. This shows no values (correctly).
Highlight the adjustment layer in the second masked folder.
Both samplers now show no value (incorrect).
Delete sampler #1. The new #1 now shows the current values (correct).
Recreate the other sampler (now at #2). Sampler #2 does not show color values (correct).
Highlight the first adjustment layer again.
Both samplers now show no value (incorrect).
It appears that *all* color samplers are either using the masked area of sampler #1, or combining the masked area of all lower numbered samplers, to determine if they should calculate values. The workaround is to remove all samplers every time you switch masked folders (which can get a bit insane if you have a lot of masked folders). This is not the way it worked in CS3.
Can anyone else duplicate this issue? Or is this unique to my machine? I'm running a fresh install (less than a week old) on Mac OS X 10.5.8, and I've updated to PS 11.0.1.
Message was edited by: achbed
Added version information

Simply put: Color Samplers will report no information with a Color Adjustment layer selected if it is contained in a masked Group and if the #1 Color Sampler is in the masked region.
The cursor sampler WILL work however. Unfortunately, I often need the mouse elsewhere while making adjustments.
This makes working in masked groups quite bothersome so any help would be appreciated.

Sample delay from software to hardware

Hi,
I am interested to find out the time it takes for a sample generated in LabView until it hits "the air" (channel output) of one of your DAC cards? I couldn't find any data on this in the different product specifications nor in the forums.
Thank you very much,
Thomas Schmmid

Hello tschmid. The time that a value takes to appear at the analog output of a DAQ board can be divided into 2 distinct parts. The first part is the time it takes to get from the software to the buffer on the DAQ board. The second part is the time it takes to progress through the buffer on the DAQ board to the output. The first part is nondeterministic since there is no way to benchmark this process, but should be very quick since all the software does is send the output value to the board. The second part can be found in the user's manual of a specific DAQ device under "Maximum update rate" for analog output channels. For example, assuming one channel is being used, the PCI-6251 has a maximum update rate of 2.8 MS/second. Thus, if you have a clear buffer, it will take this second part 1/2.8M seconds to execute. If the buffer contains 99 values before your desired output, the second part will take 1*100/2.8M seconds since 100 values have to be written to the output in order to see that value that was just written to the board from software. There is a way to ensure that your onboard memory doesn't slow down this process too much. There is a property node in DAQmx called AO.DataXferReqCond which allows you to chose one of the three following options: "On Board Memory Empty," "On Board Memory Half Full or Less," or "On Board Memory Less than Full." Choosing the "On Board Memory Empty" option will allow a more responsive output assuring that the current value written from software will be seen on the board's output quickly, but it will limit the achievable update rate. However, if you are concerned with seeing one value appear on the board in the shortest amount of time, "On Board Memory Empty" is your best option. I hope this helps and if you have any additional questions, I would be happy to address them. Have a great day!
Brian F
Applications Engineer
National Instruments

8.0.2 horrendous bug still exists!!!!!!!!!!!!!

The effects and sample caching bug is horrendous in my opinion. It's been there for years. It wastes huge amounts of my time. While there are some things you can do to minimize these glitches, there are NO 100% reliable work arounds. If you work in Logic and want a professional result you absolutely must listen to the finished bounce very carefully for glitches. These glitches come in many forms and will often times only be noticed after you receive the finished product from the mastering house.
The main culprits are:
1. Reverb and delay tails. If you have reverb or delay on a track and you stop playback and move your position, the reverb or delay will begin where it left off.
2. Samples. For example I often replace kicks and snares with samples. When I go to start a bounce random samples will play at the beginning of the track.
3. Clicks from compressor and EQ plug ins. These are less obvious, but generally mastering brings them out loud and clear after it's too late.
Keep in mind that these glitches don't generally happen until a split second before a region starts. That means it is very common for them to happen in the middle of the song. Starting and stopping playback a few times does NOT completely clear the cache.
It's a total craps shoot and the most embarrassing, time wasting aspect of Logic.
I really wish I could trust Logic to give me a bounce that is exactly "as expected."
I just sent a bug report... I encourage you all to do the same. They are obviously working on stuff and I think it's our responsibility to make it a priority for them.
http://www.apple.com/feedback/logicpro.html

Hi Rohan,
I meant to begin my last post "with all due respect". I read you all the time, you are always well informed, plus you came up with the 8.02 key command fix/discovery..
Rohan Stevenson1 wrote:
hi pancentre,
no mate that's not correct re logic flushing the buffer. that is actually my point - it can't, not the way the audio engine is designed - god, but my memory is sketchy on this - i remember getting the ins and outs of this a long time ago...
Right, I understand exactly what you're saying, this is indeed the way Logic works... my point was it's an archaic system that was devised at a time when computers had 1/1000th the processing power we have now. There's no need for such great efficiency, plugins can be kept partially active, this is how 9 out of 10 DAWs flush the buffers, the plugin is always active to some extent.
one of the things that makes logic efficient is the way it handles plugs - it doesn't process them the whole time - only when there is sound to process.
Agreed, see above. I think this is related to a couple of Logic's other quirks.
Efficiency in one area is causing less efficiency in other areas, the overload messages. What happens when a song is playing and it comes upon three regions with a couple of space designers and an instance sculpture? Instead of these plugs being partially active and using a little processor, they all become active at once and often cause a spike... ie; system overload or other kind of glitch.
it is the 3rd party plugs that have to flush its own audio buffer - its not something logic can effect without changing the way it works to become much less efficient.
but if you were to call it a 'logic bug' then i would find it hard to disagree with you:
Well, I guess it's hard to say but it seems like Apple came up with a workaround for Logic's behaviour and then asked everyone else to do the same. Until Apple version 7.xx Logic's plugins used to do the reverb tails and all other manner of little tics & glitches.
There's still a couple of Logic's plugs that aren't all the way compliant.
pancenter-

Sample Delay bug

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