Comparison DACs

- DAC. Actually, digital-to-analog converters in the original sense of the word are devices designed to convert digital audio transmitted via an optical, coaxial or USB interface into an analog line-level audio signal, usually in stereo format. Sometimes switching of a digital signal may also be provided (output unchanged to one or another digital output), rarely also reverse, analog-to-digital conversion and/or sound processing using built-in filters and regulators.

— DAC with amplifier. Digital-to-analog converters (see corresponding paragraph), complemented by a built-in headphone amplifier and headphone output. The use of this feature can vary: some devices use the “ears” to control the sound coming to the DAC outputs, while others are actually high-end compact headphone amplifiers that connect to the digital output of a PC, game console or other similar device.

The number of audio channels supported by the device.

The standard option for modern DACs is classic 2-channel stereo sound. However, in addition to this, there may be other options. Thus, there are multi-channel (up to 32 channels) devices; they are usually referred to as audio interfaces (see "Type") and are intended for mixing multiple sound sources - for example, a microphone and several instruments. The opposite option is single-channel converters; They are used mainly in hi-end audio systems on the “one per channel” principle and are intended for maximum channel separation (which, in turn, allows you to very accurately configure the operating parameters of each channel).

Sample rate of the digital-to-analogue converter used in the device.

DAC in this case means the “heart” of the device, the main circuit that directly provides the conversion of digital audio to analogue. And the sampling rate is initially one of the characteristics of digital sound. In this case, its meaning is as follows: the sampling rate of the DAC must not be lower than the corresponding indicator in the incoming audio signal, otherwise the device will not be able to effectively cope with the conversion.

The bit depth of the digital-to-analogue converter used in the device.

DAC in this case means the “heart” of the device, the main circuit that directly provides the conversion of digital audio to analogue. Bit depth is initially one of the characteristics of a digital audio signal. In this case, its meaning is as follows: the bit depth of the DAC must be no lower than the bit depth of the audio signal with which the converter is planned to be used, otherwise the device will not be able to effectively cope with the conversion.

The audio frequency range supported by the device. Most often, we are talking about the frequency range that the device can output in an analogue audio signal at the output.

In general, the wider the frequency range — the fuller the sound, the lower the likelihood that the transducer will “cut off” the upper or lower frequencies. However, note that the human ear is able to hear sounds at frequencies from 16 to 22,000 Hz, and the upper limit decreases with age. So from a practical point of view, it does not make sense to provide a wider range in audio technology. And the impressive numbers found in high-end devices (for example, 1 – 50,000 Hz) are more of a "side effect" of advanced electronic circuits and are given in the characteristics mainly for the purpose of advertising. Also recall that the overall sound quality is affected by many other factors, in addition to the frequency range.

The dynamic range of a transducer is defined as the ratio between the maximum signal level it is capable of delivering and the level of its own noise when a low amplitude signal is applied. Quite simply, this parameter can be described as the difference between the quietest and loudest sound that the device can produce.

The wider the dynamic range, the more advanced the DAC is considered, the better sound it can produce, all other things being equal. The minimum value for modern devices is about 90 dB, in top models this figure can reach 140 dB.

Also note that this parameter is similar in its meaning to the signal-to-noise ratio, however, these characteristics are measured in different ways; see below for more on this.

The coefficient of harmonic distortion produced by the converter during operation.

The lower this indicator, the clearer the sound produced by the device is, the less distortion is introduced into the audio signal. It is impossible to completely avoid such distortions, but it is possible to reduce them to a level that is not perceived by a person. It is believed that the human ear does not hear harmonics, the level of which is 0.5% and below. However, in high-end audio applications, distortion rates can be much lower — 0.005%, 0.001% or even less. This makes quite a practical sense: the distortions from the individual components of the system are summed up, and the lower the harmonic coefficient of each component, the less distortion there will be in the audible sound as a result.

The capacity of the analogue-to-digital converter installed in the device.

The principle of converting analogue audio to digital is that the sinusoid of the audio signal is divided into separate "steps", and the data about each "step" is encoded in digital form. This encoding has two parameters: sampling rate and bit depth. For the first, see the relevant paragraph; and it depends on the bit depth how accurately the height of each “step” will correspond to the height of the point of the sinusoid on which it falls. In fact, the higher the bit depth of the ADC, the higher the quality of its work, the more accurately the device converts analogue sound into digital.

The minimum indicator for ADCs used in modern audio interfaces is 16 bits — this corresponds to the bit depth of audio in the Audio CD format and is considered quite sufficient even for high-quality sound. However, there are also more advanced converters — 24 and even 32 bit.

Sample rate of the analogue-to-digital converter installed in the device.

The principle of converting analogue audio to digital is that the sinusoid of the audio signal is divided into separate "steps", and the data about each "step" is encoded in digital form. This encoding has two parameters: bit depth and sampling rate. For the first, see the relevant paragraph; and the sample rate describes how many digital "steps" the sine wave is divided into. The higher it is, the more “steps” will fall on every second of the sound and the closer to the original will be the analogue sound restored from the digital format.

The minimum indicator necessary for the full reproduction of sound in the entire range of audible frequencies is 44.1 kHz; this matches the sound quality of an AudioCD. And in the most advanced ADCs, the sampling rate can be 192 kHz (DVD-Audio level) and even 384 kHz.