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  • FV-1 Audio Reverb Effect

  • Created: Jun 26, 2020

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The reverb effect is primarily used to create depth in sounds. It can be used to improve an instrument sound or used to make a poor singer sound better. In the past, amplifiers use a spring tank to add a reverb effect to the sound of the instrument. The amplifier usually have a sub-circuit that sends the instrument signal to the spring reverb tank. The springs get excited with the sound and create the reverb sound which is sent back to the amplifier to be mixed with the straight sound. Today, modern amplifiers use a digital IC that integrates an ADC, DSP, DAC, etc. to process the sound digitally and add a reverb effect to it.

This reference design features a reverb effect schematic diagram which is based on an FV-1 digital reverb IC. The FV-1 chip is a device that has an internal DSP and other internal peripherals such as ADC, DAC, etc. that are used to digitally process an audio signal. The device operates with a 3.3V power supply. It has eight internal programs (effects) which is burned in its internal ROM that can be selected through its S0-S2 pins. A designer can still add eight more programs to the device by connecting an external serial EEPROM to the device. The FV-1 chip has three potentiometer inputs that can be used to vary the real-time parameters such as decay time in a reverb, rate and depth in a chorus, or frequency in a filter. These inputs are available as coefficients to the selected program and may be used independently. The internal ROM of the FV-1 device has eight effects such as chorus-reverb, flange-reverb, tremolo-reverb, pitch shift, pitch-echo, test, reverb 1, and reverb 2. In this design, the selected effect is the reverb 1 (7th effect) which is selected by pulling S0 pin to LOW while S1 and S2 pins are pulled HIGH. The reverb 1 effect has three real-time parameters (reverb time, HF filter, and LF filter) that can be adjusted through its POT0-POT2 pins. Only the reverb time parameter can be varied in this design, while the HF and LF filters are put into balanced setting. It’s the RV2 potentiometer that can be used to vary the reverb time parameter.

The audio signal is inserted at the input jack J1 and amplified by U2A op-amp. The amplified signal enters to the input pins of the FV-1 device. The internal ADC converts it into digital and the internal DSP processes the signal to apply a reverb effect. The internal DAC converts the digital signal back into analog form and sends it to the output stage preamp U2B. The amount of reverb effect at the output jack J2 depends on the mixing of the straight sound that passes through R8 to the input of U2B and the processed signal from the FV-1 chip. The potentiometer RV1 controls the magnitude of the signal that comes out from the FV-1 chip, therefore, RV1 controls the reverb effect at the output. The whole circuit operates with a power supply that has an output voltage of 9V up to 12V. Though the FV-1 can accept stereo signals, this circuit was designed in mono so that it can be used for instruments like acoustic guitar, bass guitar, electric guitar or to add a reverb effect to a microphone.



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