Data lost due to compression is irreversible

In this series, we will focus on the basic knowledge about “sound” that is necessary for video production, and we will make it easy to understand by omitting small and difficult things as much as possible, such as a little general knowledge and sound, including music. . I look forward to delivering it, so I look forward to working with you!

Now, let’s talk about the first memorable event under the name [Digital Audio Basics]. There are several types of digital audio. Among them, I have summarized the main ones.
[Format types and functions]
◉ Uncompressed format: linear PCM (WAV, BWF, AIFF)
→ The most basic format for digital audio. BWF is a commercial WAV that can contain metadata.
◉ Lossy compression format: P3, AAC (MP4), MQA, etc.
→ Format used mainly for general purposes. In many cases, the information in the uncompressed data is shrunk and compressed. The data capacity is reduced, but the sound quality also deteriorates accordingly. MQA is a new format that is irreversible in terms of data, but reversible in terms of sound quality.
◉ Lossless compression format: FLAC, ALAC, etc.
→ Format mainly used for high-quality listening. It has the reversibility of being able to reproduce exactly the same sound quality as before compression, but the data capacity is not that small.
◉ Others: DSD (DSF, DSDIFF, etc.)
→ It is also called 1-bit audio, but since the concept is fundamentally different from multi-bit audio like linear PCM, it can be compared to “24bit” WAV, etc. in the same line I have not. Currently, it is one of the highest quality formats, but it has the weakness of not being editable.
How is it? I think there are several things, from the familiar ones to the ones you see for the first time, but among them, the one that is most suitable for today’s video production is “Linear PCM”! The reason is as follows.
1. Since it is an uncompressed format, it has excellent sound quality.
2. You can edit like cut and paste.
3. The digital voice tracker is the most popular Ma ‘around the world because the bet, any device, can be managed by software.
Since MP3 and AAC (MP4) are compressed formats, there is a considerable loss in sound quality. Depending on the compression ratio, it may not be obvious at first glance, but it is not suitable as processing-based material such as video production and music production. FLAC and ALAC are lossless compression formats that do not deteriorate sound quality, but do not significantly reduce capacity, and there is no software that can be edited natively (without conversion to other formats), so it is still unsuitable for the production. . DSD was adopted from SACD which appeared in 1999, and is said to be the most analog digital audio today, and it has a smooth texture that is different from linear PCM in terms of sound quality. This format has finally attracted attention in recent years, but due to its mechanism, it has the weakness that it cannot be edited as is, so on the production site, mainly one-shot music recording (recording without editing) and mixing (long-playing recording without editing) and mixing (often used as a master recorder when combining multiple sounds into one stereo or surround sound (also called track down). “Almost Ichi 択 linear PCM” video production, I think I could understand that you can refer to. Of course, if the compressed format does not make you uncomfortable, you can use it, but consider it as an emergency. If you still want quality, you must use linear PCM. The data lost by compression is irreversible. The file that will be the master of the work must be of the highest possible quality. By the way, whether you use WAV or AIFF, the sound quality is almost the same. However, co Considering compatibility, even Mac users can be relieved to use WAV for data transfer.
“16 bit / 44.1 kHz” is
the lowest line of CD quality
Now let’s dive a little deeper into linear PCM. There are “number of quantization bits” (bit depth) and “sample rate” (sample rate) that represent linear PCM specifications. Have you ever seen the notation “16 bit / 44.1 kHz”? This means that the original (analog) audio is sampled (digitized) 44,100 times per second at the 16-bit volume stage (2 raised to 16 = 65,536)! Still, I think it’s “what is this?”, So I tried to sum up the points by comparing it to the video!