Surround delay circuit

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SURROUND SOUND

This surround-sound decoder is based on the 'Hafler' principle, first discovered by David Hafler sometime in the early s. The original idea was to connect a pair of speakers as shown in Figure 1, for use as the rear speakers in the surround setup. There is also no way to control the level reproduced, since it will always simply be the difference signal between left and right channels. If the signal is mono, then the signal in both channels will always be more or less identical, and there will be no output from the rear speakers at all. This circuit works by allowing the rear speakers to reproduce only the difference signal between the left and right outputs. All stereo encoded material has some difference between left and right channels if it didn't, it would be monoand it is this difference signal that is reproduced by the rear speakers. It is important to ensure that the connection between the rear speaker negative terminals is not earthed, or they will simply be in parallel with the main speakers. So, if you want to use separate amps for the rear speakers, basically you can't - unless you get sneaky. The first circuit in Figure 2 is completely passive, but requires that a suitable transformer is available. A suitable transformer means a line level, 10k impedance unit with a ratio - these are scarce, but are available after a search. You might be able to get away with a Ohm unit, but because of the impedances you need, its performance will be very ordinary, with an extreme lack of bass there is not enough inductance for a Ohm transformer to work satisfactorily at high impedances. Loading the transformer will give back some of the bass, but the preamp is unlikely to be very happy with the resulting impedance. Having said this, I have used telecommunications transformers for this application ohms and they seem to work fine. The circuit shown is not a bad compromise, although the impedances are too low for anything other than a solid state preamp preferably using opamps. Using a telephony transformer Ohmthe loss overall is about 3dB, with a low frequency -3dB point around Hz. This will vary depending on the quality of the transformer used, so experimentation will be needed. Although Ohm telephony transformers are reasonably readily available, some of them are pretty ordinary. My tests were on a really good one, built by an Australian company called Transcap. I think I can say with some certainty they will be rather unwilling to sell one-off quantities. Another manufacturer of really nice transformers is Midcom in the US, but you will have the same problem with them. These manufacturers are set up to deal with large orders from other companies, not the likes of you and me wanting one "You want As a result you will have to take whatever you can get. Since it is unlikely that this will be viable for most constructors, the alternative is to go active, using a dual opamp to perform the functions. This is described next.

Digital Delay Unit For Surround Sound, Reverb, Echo & PA


We use Cookies to give you best experience on our website. By using our website and services, you expressly agree to the placement of our performance, functionality and advertising cookies. Please see our Privacy Policy for more information. Abstract: 5. D olby and th e do. Abstract: echo sound processors 5. Abstract: amplifier 5. Note: Dolby and the double-D symbol are trademarks of Dolby. Abstract: No abstract text available Text: Dolby and the double-D symbol are trademarks of Dolby Laboratories0. Personal Surround Technology. Virtual surround technology has beenreasonable for spaces used by small numbers of people such as homes. Thus virtual surround technology can. However, attention must be paid to. Table 2 shows the mostsystem can support 5. Abstract: MFP sorround sound processor ic noise cancelling mic circuit circuit diagram of 7. This LSI contains all sorts of functions including delay circuit function. Abstract: TA woofer circuit diagram 2. Set the bit to 0 bL-R signal is shifted by 1 phase shift stage. No need MCU controlradio is reduced by original variable carrier circuit. Abstract: 2. Abstract: No abstract text available Text: decoder 2 3 RAM for digital delay Surround delay time 4 Circuit for space surround 5 6surround Note: Dolby and the double-D symbol are trademarks of Dolby laboratories. Abstract: woofer circuit diagram sub woofer 5. OK, Thanks We use Cookies to give you best experience on our website. Try Findchips PRO for 5. Previous 1 2 Coilcraft Inc. Not Available Abstract: No abstract text available Text: LV dolby sound system circuit diagrams all 5. LXE 5.

How to Phase Your Speakers


Forums New posts Search forums. Articles Top Articles Search resources. Members Current visitors. Log in Register. Search titles only. Search Advanced search…. New posts. Search forums. Log in. Welcome to our site! Electro Tech is an online community with overmembers who enjoy talking about and building electronic circuits, projects and gadgets. To participate you need to register. Registration is free. Click here to register now. JavaScript is disabled. For a better experience, please enable JavaScript in your browser before proceeding. The simplest audio delay in the world. Thread starter scrawny git Start date Oct 9, I'd like to have a go at building some very straightforward audio delay circuits. No variables, just a fixed time delay upwards of 1 second which I assume is achieved with resistor values, and continual one time playback. Does anyone know a good source of simple schematics for this? So far I've found this one:. Are you going to use 1 Bit ADC? Audio can be a very complex waveform and it is no trivial feat to time delay it. I take your point, but I am not necessarily interested in reproducing the sound with any attention to fidelity. Could I not make a test circuit with a 1-bit adc and replace it with a higher bit one if I can get it to do what I want?

Audio Delay Module


The RC delay element is a way to create a time delay in your circuit by connecting a resistor and a capacitor. And very useful. A capacitor is kinda like a tiny little battery. You can charge it with a voltage. And you can use this voltage for a short time until the capacitor is discharged. A capacitor with a higher Farad value can store more energy than one with a smaller value. Therefore it also takes more time to charge a high-value capacitor versus a small-value capacitor. If we connect the capacitor directly to the battery, there is no restriction on the amount of current that flows through the capacitor other than the batteries own maximum current capacity. The task of the resistor is to reduce the flow of current to the capacitor to slow down the time it takes to charge it. It gives you the time it takes for the voltage to rise from zero to approximately Skip to main content Skip to primary sidebar Skip to footer The RC delay element is a way to create a time delay in your circuit by connecting a resistor and a capacitor. The time it takes for the voltage to rise across the capacitor becomes our time delay. The more current that flows, the faster it charges. So a lot of current flows, the capacitor charges really quick, and the delay becomes very small. Social: Facebook Twitter YouTube.

PT2399 Basic Surround Delay Circuit

At any rate, the device described here can be a lot of fun to play and experiment with. The input range is controllable from 0 seconds to the maximum delay by means of an analog input. This analog input can simply be driven by a potentiometer, or from an external signal. Modulating the delay input with different waveforms can produce interesting sound effects. Did you use this instructable in your classroom? Add a Teacher Note to share how you incorporated it into your lesson. The block diagram shows how the circuit works. First, the input signal is amplified as needed. Next, the amplified signal is sampled and converted to a digital value by an analog to digital converter. The digital sample value is then stored in a large RAM buffer. The delayed waveform is created by retrieving older samples from the buffer and sending them to a digital to analog converter DAC. The output of the DAC is low pass filtered to remove unwanted frequencies that are an inherent part of the sampling and reconstruction process. This process of analog to digital conversion, storage, retrieval, and digital to analog conversion runs in a continuous loop, controlled by a microprocessor. In this manner, a delayed replica of the input signal is produced at the output. The steps that follow provide in depth detail of how each sub circuit works. A PDF of the complete schematic is attached here, with each subcircuit highlighted a different color. By biasing the non inverting input at VREF and capacitively coupling the audio, the input voltage at the non inverting input will be equal to the audio waveform offset by VREF, instead of swinging about 0 volts. This allows a waveform with positive and negative portions to be processed by the rest of the circuitry, which only works with positive voltages. The reference voltage is set by the voltage divider created by R1 and R2, and the op amp buffers the voltage. These are essentially the same part, only one is a dual and the other is a quad. These parts have a full scale output that can swing from rail to rail 0V to 3. The input audio is then capacitvely coupled via C1 to the input of op amp U6-A. The result is that the voltage at the input to the op amp will be the audio input signal swinging above and below VREF. Biasing resistor R5 and the AC coupling capacitor also form a high pass filter to the input audio. The values must be chosen such that the cutoff frequency is below any frequencies you want to pass. Depending on the source of the input audio, the gain of the pre amp may need to be altered. If the input is from a source such as the line out of a computer, radio, DVD player, etc, then it may be of sufficient amplitude that it requires no further amplification. If the input is from a microphone, then amplification will be required. R7 must be chosen to configure the amplifier for the needed gain. In this circuit, that means a maximum peak to peak voltage of 3. If the input signal is amplified too much, the peak to peak value will exceed maximum input range and it will be clipped, introducing distortion. I included a switch SW2 in my design so that the gain could easily be changed from 1 for higher level sources to about 34 for use with a microphone. The switch shorts out R7 so that the gain is reduced to 1. I have found that this works well with sources like the line out from a computer, radio, CD player, etc. R4 is used to supply power to an electret microphone if that is the input audio source.

Circuit PT2399 Reverd , Delay & Casio PT-1



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