Despite the fact that audio conversion has been the subject of many books, a basic understanding of the two concepts is critical to properly operate your computer-based recorder: sample rate vs. bit rate.
The conversion procedure is complicated, and there are few options for completing it. But don’t worry; we’ll just go over sample rate vs. bit rate in terms of linear pulse-code modulation (PCM), which is one of the most popular conversion technologies.
Basics Of Sampling
Computers work by turning a series of switches on and off at a high speed one at a time at their most basic level. Since computers work in discrete stages, converting analog sound to the digital realm necessitates mathematically describing the continuous analog waveform as a series of discrete amplitude values.
This is achieved in an analog-to-digital converter by taking a rapid sequence of short snapshot samples of a given size at a defined rate. Each audio sample includes information that allows the original analog waveform to be reliably reproduced.
This datastream contains information such as dynamic range, frequency content, and so on. The magnitude of the closest measurement increment is assigned to the instant amplitude stage of each sample, a method known as quantization
A digital-to-analog converter generates a nearly equivalent in principle copy of the original amplitude by replicating these variables and playing them again in the very same order and at the same rate as they were recorded.
The sample rate is the rate at which data is captured and played back. The bit depth or word length is the sample size, or more precisely, the number of bits used to characterize each sample. The bit rate is the amount of bits transmitted per second. Let’s look at this in the context of digital audio.
Pulse Code Modulation
For storing analog waves in a digital environment, PCM is at the industry level. The frequency of the sound is sampled at a constant period in a PCM stream. PCM is a completely non-proprietary format, which means that anybody can use it for free. But, due to file size and playback compatibility finding any audio in PCM format is quite rare.
The file size of the captured audio is enormous since PCM is uncompressed. Audio files can be compressed using lossy or even lossless encoding algorithms to maintain audio quality while reducing the size of the file.
Dolby and DTS are lossless audio compressions that are mostly used for this task because they can reduce PCM audio file sizes by up to 90%.
Unfortunately, Dolby and DTS’s methods for converting PCM streams to a bitstream into storage and afterward decoding them for playback aren’t great. While the resulting sound is smaller by storage space, it is not always as crisp and clear as the source, resulting in a loss of clarity and consistency.
Lossless audio formats like Dolby Digital TrueHD and DTS-HD Master Audio can help here. They will decode PCM audio signals in the very same way that they were recorded.
Most operating systems (OS) do not allow native PCM file playback. The Waveform Audio Format (WAV) was specified by IBM and Microsoft for Windows OS, while Apple has used Audio Interchange File Format (AIFF) for Macintosh OS.
Both formats are simply a wrapper over the PCM audio format that contains extra audio material such as the author’s profile and track title, among other things.
The sampling rate of digital audio files is comparable to the framerate of film. The more audio data (samples) collected over time, the similar the captured data gets to the original analog sound. The sampling rate of a standard digital sound CD recording is 44,100 or 44.1 kHz.
The Nyquist-Shannon sampling theorem explains that why frequency is so high, given that the human ear will only detect frequencies up to 20kHz at most.
The Nyquist theorem, also known as the Nyquist frequency, states that when digitally sampling a signal, you must sample at least twice the maximum predicted signal frequency to avoid any loss of detail. In certain cases, such as when capturing animals that emit ultrasonic sound, a sampling rate of 384,000 Hz is used.
The number of bits transmitted or processed per unit of time is used as a measurement is referred to as bit rate. It’s similar to the sampling rate, except the number of bits is counted rather than the number of samples. In a playback/streaming case, bitrate is more widely used than in a video context.
Bitrate is a concept that isn’t just used in the recording industry. It’s also popular in networking and multimedia. In the case of audio, though, a higher bitrate is often correlated with higher output.
Since each bit of an audio file records a piece of data that can be used to recreate the original sound, this is the case. In other words, the more the bits you can cram into a unit of time, the closer you get to replicate the original variable frequency sound wave, and therefore the more effective it is as a song representation.
Unfortunately, a higher bitrate entails a larger file size, which is a no where disk capacity and bandwidth are limited, as they are with music streaming platforms such as Apple Music and Spotify.
Basically, the number of sounds recorded per time unit is known as sample rate, and the number of bits recorded per time unit is known as bit rate.
The expression “high-resolution audio” is often used, but it is poorly described. This is due to the lack of a universally accepted description. For several years, the word “resolution” meant “bit depth,” but in recent years, it has come to mean “both sampling rate and bit depth.”
And the word “high resolution” in particular is debatable. When 8-bit audio was the standard, 16-bit was considered “high resolution.” In today’s world, 24-bit, 96-kHz audio is regarded as “high quality.” This concept is likely to evolve rapidly as 192 kHz converters and audio interfaces become more popular.
So if you’re still confused about which is better, Sample rate or Bit rate, it really depends. They both have their own merits and demerits. And excel at their own field.
You may also like to read: How to Record A Song?