
Audio encoding

I wrote over audio files last time, but if you reduce the file size (code at a lower bit rate), the sound quality tends to deteriorate. How much should it really be? .. ..

When compressing using audio encoding (AAC, MP3, etc.), the compression rate is determined by the bit rate at the time of encoding. Specifically, if you set a low bit rate, the compression rate will be higher and the file size when saved will be smaller, but first of all, what is the bit rate for uncompressed original sound source (PCM) ?
If you save it as PCM, the sound quality will be that of the original sound, but it can be a bit awkward to save without worrying about the file size. Also, depending on the application, I think the memory capacity is sufficient even for the original sound size and the communication speed is fine. Therefore, I would like to write about the sample rate and bit rate that are often heard in digital audio.
The bit rate of digital audio is determined by the sampling frequency, the number of bits assigned to a sample (number of quantization bits), and the number of channels (stereo, monaural, etc.).
PCM bit rate (uncompressed) = sample rate x number of quantization bits x number of channels
As I wrote a bit last time, in file containers like wav and mp4 format, this information is attached as a header, so that the application can see the header and play it back. The compression rate of the encoding is determined by the bit rate specified at the time of encoding for this PCM (uncompressed) bit rate.
For example, as many of you know about music CDs, with 44.1 kHz stereo, this is the next bit rate.
Music CD bit rate: 44100Hz x 16bit x 2ch (stereo) = 1411.2kbps
When encoding this with MP3, AAC, etc., it is natural to specify a bit rate lower than 1,411.2 kbps. For example, when encoding at 256 kbps, the compression rate is around 18% when the original sound is 100% and the file size is 1/5 or less.
Encode a music CD at 256 kbps: 256 kbps / 1,411.2 kbps = about 18%
In general, the sample rates of audio devices connected to PCs are 48 kHz and 44.1 kHz for music, 16 kHz and 8 kHz for audio such as microphones and headphones, and 32 kHz, 24 kHz, 22.05 kHz. , etc.
The bit rate of PCM (uncompressed sound source) with 16-bit quantization bits is as follows.
Stereo (for music) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
48kHz 16 16 2 1536 kbps
44.1 kHz 16 16 2 1,411.2 kbps Music CD
32kHz 16 16 2 1,024 kbps
24kHz 16 16 2 768 kbps
22.05 kHz 16 16 2 705.6 kbps
Monaural (for audio) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
32kHz 16 16 1 512 kbps Super Wide Band
24kHz 16 16 1 384 kbps
16kHz 16 16 1 256 kbps broadband
8kHz 16 16 1 128 kbps Narrow band
Sampling rate
If you check the web, there are explanations such as the sampling required to convert analog waveforms to digital conversion. For example, it shows how many samples of an audio signal input from a microphone are taken per second and digitized. The larger the sample, the greater the range that can be recorded. When an analog waveform is digitized, the frequency that can be expressed is half the sampling frequency (sampling theorem). For example, with a sample rate of 48kHz, it is possible to express up to 24kHz. At 8kHz (narrowband) and 16kHz (wideband), which are often used for audio, you can only hear up to 4kHz and 8kHz, respectively. The higher the sample rate, the higher the bit rate.
sampling theorem
It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when talking about signal processing or other time ).



















