Voice Record and Playback devices are integrated circuits that connect to microphones and speakers, or the amplification circuits connected to microphones and speakers. They enable stored audio samples in the form of sampled data to be converted to analog signals and output to a speaker or associated amplifier for playback purposes. They also allow microphone signals to be amplified, converted to digital samples and stored for recording purposes.

 

There are devices that operate standalone, incorporating programmable internal memory for message storage. There are also devices that provide microprocessor (MPU) or microcontroller (MCU) system the ability to transfer audio to the analog audio system, called Audio Interface Chips.

 

The conversion between analog and digital circuits is performed by an Analog to Digital Converter (ADC) in the record path and a Digital to Analog Converter (DAC) in the playback path. These converters work at a period rate called a sample rate. Typical sample rates are 4 kHz through to 192 kHz – the higher the sample rate, the higher the frequency that can be recorded. They also have the ability to sum in configurable tones or alarms to facilitate the integration of services like telephone DTMF dialing. They can typically drive 8 or 16 Ohm Speakers and may support mono or stereo applications.

 

In the record path, the microphone will have a special analog circuit for amplifying and adjusting signal levels from the microphone that are usually very low amplitude. This circuit is called an Automatic Level Control (ALC). Once signal levels are in the ADCs range, they can be converted to digital values by using a process called sampling. Sampling measures and stores instantaneous values of the signal at a sample rate more than twice the maximum frequency of the audio signal. The audio signal is filtered to prevent higher frequencies from being incorrectly sampled (aliasing). The digital samples are then able to be stored in a memory device. This can be an SRAM, DRAM or if the recording needs to be held even if the power is lost, non-volatile storage such as a flash memory.

 

The playback path starts with digital samples in memory. These samples are sent to a DAC and converted to a pulse stream of instantaneous values that are smoothed using an anti-aliasing filter and converted into an analog signal. The analog signal is then passed to a programmable gain amplifier (PGA) that can drive a speaker or subsequent audio amplifier. The PGA setting can determine the volume.