Surround delay circuit

The RC Delay Element

The collection of electronic circuit - schematics. Power supply, audio, microcontroller, digital circuits, analog circuits and more This is the easy build surround sound processor circuit using the digital delay process method. This audio processor isn't applying any unique function ICs that tough to obtain parsonaly, and designed in only common purpose ICs. The kind of this Surround Processor is producing the surround impact with processing two channels of stereo supply. The majority of those are generates the surround impact with separates reverberations from supply signal and applies any processes, after which mix it to front channels or output as rear channel. The circuit works and contains three modules: Separating the Reverberatins The distinction in between each channels is separated with distinction amplifier from the op-amp. And greater frequencies inside the distinction signal are cut using the LPF. The A-D converter is basically delta modurator that making use of a comparator plus a D type flip-flop. Following passed digital delay, the bit stream is directly conversion into analog signal using the integrater. It isn't beneficial that signal to noise ratio and distortion as this A-D, D-A converter. Shift register generates Read-Modify-Write cycle, read out old data and save new data in a single cycle. And lower byte in the address counter is assined as row address with the DRAM to increment row address every single cycle, to ensure that refresh cycle is often omitted. Electronic Schematic Diagram. Home audio Surround Sound Processor Circuit. Surround Sound Processor Circuit Penulis schematic diagram. Tags audio. Artikel Terkait. Next Post. Previous Post. Post a Comment. Subscribe to: Post Comments Atom.

Simple Surround Sound Processor Circuit


The Mitsubishi M Digital Delay IC has been discontinued for reasons beyond my comprehensionand for some time there was no suitable replacement. See Project 26A for the replacement project details. You will be unable to build the version shown here unless you happen to have access to some M ICs. The Surround Sound Processor Project 18 is a standard Hafler matrix, and will provide a passable rear channel signal. In order to confuse the brain, we really need to delay the signal, which makes it sound as if it were further away. This principle is used in virtually all commercial units, but they have a tendency to be somewhat more complex than a simple passive unit. The digital delay presented here is expected to be sufficient for most applications, and although the delay is fixed at 20ms, this is the most usable delay period. This is the equivalent of being about 7 metres away from the rear speakers, and they will have a suitably 'distant' sound, even when relatively close to the listening position. Most commercial units will add extra features different delay periods, reverberation, etcwhich are all missing from this circuit quite deliberately. These effects are possibly fine to impress your friends, but will become very tedious after a relatively short time, and end up detracting from the program material - you are listening to the effects instead of the sound. Also included is a complete diagram showing how the Digital Delay Unit, Surround Sound Processor Project 18either of the Stereo Width Controllers Project 21 and even electronic crossover Project 08 or Project 09 can be assembled into a complete unit. All that is needed is a whole bunch of amplifiers 8 of them for the most complete arrangement, but you could survive with only 7. This has been around for a while now, and is simple and effective provided that a fixed delay is acceptable. The serial data required to obtain different delay settings is not easily obtained, and would add considerably to the complexity of the circuit. As such, it would no longer be a simple matter to construct using Veroboard or similar, and would require a printed circuit board. The circuit is almost a direct adaptation from the Mitsubishi data sheet, and as shown will give good performance over a wide frequency range. The filters are tuned to around 9. This seems to be the optimum response for rear channel speakers, so should be left alone. The filter circuits use internal opamps, and only require the external components shown below. Figure 1 - The Digital Delay Unit. As shown, the unit can be constructed as a module quite easily, requiring a 5 Volt supply, analogue and digital earth connections, and an input and output. This is easy to wire up, and will keep expensive mistakes to a minimum. The 2MHz crystal or you can use a ceramic resonator if you prefer is probably the only item apart from the M that may be a little awkward to obtain. The 5 Volt supply must be regulated, as anything over 6 Volts will destroy the delay chip. See Figure 4 for a suitable power supply circuit. The input driver circuit is designed to reduce the signal level applied to the delay chip, to prevent any risk of overload. Since the maximum specified level is 1 Volt RMS, it is important to ensure that the signal is below this at all times. The output from a CD player is generally about 2. Since this must be amplified again after the delay, there is a pre-emphasis circuit included to increase the level of high frequencies. The frequency response is restored to normal with treble cut after the delay, reducing noise as well. This technique is used with FM radio broadcasts, vinyl disks and in many other areas and is effective in minimising noise levels. As can be seen, the circuit is very simple. The resistor marked "SoT" Select on Test is designed to allow for the fact that the gain or loss through the delay circuit is not necessarily unity, but can vary. This is designed only to compensate for units with a lower than normal gain, and might be as low as 15k for the worst case. This is unlikely, so if desired, the resistor may be omitted altogether, or a k pot can be used to allow the gain to be changed easily. A complete surround system would consist of a power supply, the decoder matrix, optionally the stereo width controller, and the delay unit. Figure 3 shows my suggested method of interconnecting the units, which can all be housed in a single case. For the more adventurous, you can add a pair of Linkwitz-Riley crossovers for bi-amping shown in the dotted box for the front left and right speakers. This keeps everything together, and only the switches needs to be accessible in normal use. All level controls should be set to the desired volume, and not fiddled with once you have it the way you want.

The simplest audio delay in the world


We have searched the web to help you find quick design ideas. We make every effort to link to original material posted by the designer. Please let us if you would like us to link to or post your design. VA may assume both positive and negative values. For a first-order allpass circuit, the transfer function is. The circuit in Figure 1 allows you to measure vector magnitudes for example, voltage or impedance by multiplying the measured signal with inphase and quadrature sinusoidal signals of the same frequency. However, when the measurements cover a wide frequency span, it is necessary to compensate phase differences between the measured and instrument circuits by using a programmable delay. This means of course that you will be unable to build this delay circuit. I have left the project in place as reference, and the overall connection schematic is still valid if when? The project will be updated at that time. There is no volume control, audio levels being directly controlled from the sound card itself. The first circuit is a dual ramp generator where the positive and negative ramps are generated separately. This circuit was used as a ramp generator for a transistor curve tracer: the positive going ramp was used for testing NPN transistors and the negative ramp for testing PNP transistors. This is built with only conventional parts, and digital delay block, too. So it is very easy to build. Many systems, even some low cost mini units, already have connections for matrix surround. If you see an additional pair of speaker jacks usually labelled "rear" in the back of the unit, then you most likely have surround sound capabilities built-in to your current gear. Many such systems also have a switch labelled "two-by-two matrix" or "matrix surround". This has to be switched on, after the additional speakers have been attached, to activate the matrix circuit. Custom Search. Schematics Index. Hobby Corner. Dave's Circuits. Electronic Resources. Contact Info. Imagineering Ezine.

How to Phase Your Speakers


This article is written with an intent to explain in detail behind the making of a simple surround-sound decoder circuit. The concept of the decoder was first introduced by David Hafler in the 70s. His research illustrates the way to use two speakers as rear speakers on a surround system. According to Figure 1, Hafler designed the circuit to enable the rear speakers generate the difference of signal between right and left output. While every stereo encoded system maintains difference of signal between the right and left channel, it is that difference of signal when received by the rear speakers gets reproduced. However, it is vital to keep in mind not to earth the negative terminals of the rear speakers, else the rear will behave parallel to the main front speakers. Using individual amplifier for rear speakers is not actually possible. However, there is a way-out which we figured out after some research. Referring to Figure 2, it is totally passive, but it needs an ideal transformer — a transformer with impedance of 10K [ ratio], which is quite rare to find, but available. As an alternative we have tried using a ohm unit. But it is for the impedance the output we received was not good as it lacks bass. It is for this reason that we have used telephonic transformers with ohms, and it worked well. The circuit in Figure 2 illustrates the way we followed. Following this design, it worked, but it has very low impedance on all cases barring solid-state preamp. Using ohm unit, the loss generated is around 3dB. The low frequency is -3dB on Hz. However, it varies based upon the quality of the transformer. Most of the hi-powered transformers are sold in bulk and is therefore hard to procure a single copy. So, the alternative would be to use dual opamp to design the system, and its process is mentioned below in detail. The schematic diagram in Figure 3 gives a detailed view behind this development of the simple surround sound decoder circuit.

Simple Surround Sound Decoder 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.

DIY 3D Sound Audio Processor Module



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