Are you asking for live speech to be "delayed" up to 10 minutes?

That will be some delay.

There is nothing in existance that is designed just to do that in hardware form. You could do it with a DAW, time compression/expansion + resampling/interpolating will have artifacts it the room is live, there is ambient noise, or if there are multiple voices.

You should investigate MATLAB.

You could use a highend delay, then feed the output into an old analog delay, like those used in the 70's and 80's for guitar. But it will change the pitch of the vocal as well. What you really need is a real-time, RAM based time compression/expansion, like that used in broadcasting.

delay?

Thanks for the comment. The application is for interpreters... they would get a live feed.. it goes into the box, gets buffered and they can leak it out the other end into their headphones.. problems they generally have are that the speaker is too rapid in delivery and or the material is technical that requires mind meld decipering at a rapid pace.. stressful and drives them berserk.. thinking a 440 tone is blasted simultaneously onto either a right or left channel.. and then scanned and leveled to 440 (maybe an autotune function) thus.. the speech while speeded up.. is pitched down to avoid chipmunk itis... think its feasible but want to know if a device like this exists.. I think its a computer (every box is these days).. thanks,
Mak

You will not find a 10 minute delay in any Digital Audio Workstation plug-in.

If you look around at the various translating services that networks employ, you can find out what they are using. 99.9% are using Orbans, Eventides, etc. They are standards in broadcasting, where delays are used to edit out profanity and translating.

"The BD600 offers an unprecedented 80 seconds of the highest quality revenue and license-protecting delay - twice as much as other delays. You have the flexibility to customize the delay buffer length from four seconds to the full 80 seconds. Dump can be customized in 1/2 second increments up to 10 seconds. Each time the DUMP button is pressed, only one segment is deleted. For example, if a unit is set to the full 80 seconds of delay buffer, and is set up to provide 8-second dumps, you will have ten separate dump segments. Pushing the DUMP button the first time cuts out one 8-second segment. You still have 72 seconds of delay protection. If this is followed by more trash-talk, push the DUMP button again and you still have over a minute of delay protection. The "sneeze" button momentarily "edits" audio entering the delay, allowing the host to sneeze, cough, or make a short comment without being heard on air - and without dead air. Your station cannot afford to work without a net. With the BD600, you will never worry about running out of delay memory.

THE NEW STANDARD FOR FEATURES

If 80 seconds of delay protection were not enough, the BD600 has a secondary backup called PANIC. This plays a .WAV file stored on a compact flash card. While the file - a jingle, ID or other message - is played, delay is being built up in real time, allowing programming to continue in safety as soon as the jingle is over. This can also be used for painless entry into delay.

The unit shows not only an exact digital readout of delay time, but also has a "quick read" bar graph which instantly confirms "you're safe" at a glance. The BD600 is an 80-second stereo delay unit featuring analog XLR-type inputs and outputs, as well as AES/EBU digital audio inputs and outputs. All front panel switches (except "configure") can be remoted, as can the status indicators. Both RS-232 in and out are included; four configurable control inputs and two configurable relay-isolated control outputs are provided via an easy-to-wire Euroblock barrier strip."

If you want 10 minutes of delay, you are talking some serious jack for memory. This is about as good as it's going to get for delays.

He's not looking for 10 min. of delay, he's looking for something that can hold 10 min. worth of speech in a buffer so that it can be output at a rate set by the operator.

Non-intelligent pitch shifters usually slow things down when you shift the pitch down. Other than that I would contact somewhere that uses interpreter services a lot. Try big news stations, the UN building, places like that, see what they're using.

McCheese wrote: He's not looking for 10 min. of delay, he's looking for something that can hold 10 min. worth of speech in a buffer so that it can be output at a rate set by the operator.

Non-intelligent pitch shifters usually slow things down when you shift the pitch down. Other than that I would contact somewhere that uses interpreter services a lot. Try big news stations, the UN building, places like that, see what they're using.

OK, either way, he storing in RAM and needing RAM based processing, which isn't available in any pro audio box. I misunderstood him about the buffer/delay.

What about this:
http://www.bbn.com/For_Government_Customers/Speech_Recognition/Audio_Monitoring_System.html

From my dealings in the past, the UN is in real time on the floor.