This page describes the basic parameters of digital audio signals. Digital audio has various formats and transmission methods. These details are not covered on this page, but this page describes three important parameters that determine the sound quality of digital audio. One of the important reasons to know these is to determine whether the sound source you are about to input into the system has sound quality suitable for operation. Even if the quality of the system is high, if the sound quality of the input sound source is poor, the system will faithfully reproduce that poor quality. The quality of the sound source is important as well as the system.
This page explains the conversion method called PCM. In many cases, analog signals are converted to digital signals using this method.
Basic Concepts of Analog to Digital Data Conversion
Analog signals are converted into digital data in a discrete manner as shown in the diagram above. The horizontal axis represents time, and the vertical axis represents amplitude. Each axis is discretized, and the analog signal is approximated and recorded where the lines cross. The finer the spacing between the lines, the closer the approximation is to the shape of the analog signal.
This page introduces three parameters,
This page introduces three parameters,
Bit (Quantization bit depth)
This determines how many parts the amplitude is divided into; the higher the value, the greater the accuracy of the amplitude. CDs are recorded in 16-bit format, and most high-resolution audio sources are recorded in 24-bit format.
Bit are a somewhat complicated unit, and the amount of data when there are N bits is 2 to the power of N (2^N).
For example, the answer to the question of how many times 24 bits is compared to 16 bits is not 24/16, but (2^24)/(2^16).
Bit are a somewhat complicated unit, and the amount of data when there are N bits is 2 to the power of N (2^N).
For example, the answer to the question of how many times 24 bits is compared to 16 bits is not 24/16, but (2^24)/(2^16).
Sampling Frequency
The sampling frequency indicates how much data is recorded per second on the time axis. The higher the number, the better the sound quality. This is often written as "fs", an acronym for sampling frequency. If data is acquired 44,100 times per second, the sampling frequency is 44.1 kHz. The sampling frequency of a CD is 44.1kHz. Hi-Res audio sources also exist with higher sampling frequencies of 96kHz and 192kHz.
For humans to hear digital audio signals, they must ultimately be converted back to analog signals, but they can only be converted back to half the sampling frequency. If the sampling frequency is 44.1 kHz, then theoretically it is possible to convert a signal down to 22.05 kHz into an analog signal.
For humans to hear digital audio signals, they must ultimately be converted back to analog signals, but they can only be converted back to half the sampling frequency. If the sampling frequency is 44.1 kHz, then theoretically it is possible to convert a signal down to 22.05 kHz into an analog signal.
bps (bits per second)
The third parameter is bps (bits per second).
After digitization, audio signals are often decimated to reduce the amount of data. This reduction is often called compression. bps represents the amount of data (bit) per second that remains after reduction. Sound quality degrades after data reduction, so the higher this value, the better the sound quality.. Since the sound quality will deteriorate after data reduction, the higher this value, the better the sound quality.
WAV is an audio source with no reduction (uncompressed audio source), while AAC and mp3 are audio sources with reduction (compressed audio sources).
The bits that indicate the degree of amplitude discreteness mentioned above and the bits in bps are fundamentally different. In uncompressed audio sources like WAV, these match.
After digitization, audio signals are often decimated to reduce the amount of data. This reduction is often called compression. bps represents the amount of data (bit) per second that remains after reduction. Sound quality degrades after data reduction, so the higher this value, the better the sound quality.. Since the sound quality will deteriorate after data reduction, the higher this value, the better the sound quality.
WAV is an audio source with no reduction (uncompressed audio source), while AAC and mp3 are audio sources with reduction (compressed audio sources).
The bits that indicate the degree of amplitude discreteness mentioned above and the bits in bps are fundamentally different. In uncompressed audio sources like WAV, these match.
Summary
However, the higher the values, the more data there is, and the shorter maximum recording time for a fixed storage capacity.
If these three parameters were graphed, they might look like the cube shown below.
Below we have prepared audio files with the specifications indicated by the numbers in the diagram.
Listen to the difference in sound quality.
If these three parameters were graphed, they might look like the cube shown below.
Below we have prepared audio files with the specifications indicated by the numbers in the diagram.
Listen to the difference in sound quality.
Original: WAV: 16bit, 44.1kHz (uncompressed audio source),
(1) mp3: 16bit, 44.1kHz, 192kbps
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(2) mp3: 16bit, 44.1kHz, 128kbps
(3) mp3: 16bit, 44.1kHz, 096kbps (4) mp3: 16bit, 44.1kHz, 064kbps |
(5) mp3: 16bit, 32kHz, 192kbps
(6) mp3: 16bit, 16kHz, 192kbps (7) mp3: 16bit, 08kHz, 192kbps |
(8) mp3: 8bit, 44.1kHz, 192kbps
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(9) mp3: 8bit, 8kHz, 064kbps |
* These audio data are posted using the Weebly service. Due to the mechanism used for posting, the sound played may be slightly different from the original.
