Misconceptions about digital audio
The higher the bitrate, the better the track
This is not always the case. For starters, let me remind you what bitrate t (bitrate, instead of bitraid). In fact, this is the data rate in kilobits per second during playback. That is, if we take the size of the track in kilobits and divide it by its duration in seconds, we get its bit rate, the call. File-based bitrate (FBR), usually not too different from the bitrate of the audio stream (the reason for the differences is the presence of metadata on the track: tags, “embedded” images, etc.) .
Now let’s take an example: the uncompressed PCM audio bit rate recorded on a normal audio CD is calculated as follows: 2 (channels) × 16 (bits per sample) × 44100 (samples per second) = 1411200 (bps ) = 1411.2 kbps … Now let’s grab and compress the track with any lossless codec (“lossless” – “lossless”, that is, one that does not lead to data loss), for example, the FLAC codec. As a result, we will get a lower bit rate than the original, but the quality will remain unchanged; here is your first rebuttal.
Something else is worth adding here. The lossless compression output bitrate can be very different (but is generally lower than uncompressed audio); It depends on the complexity of the compressed signal, or rather on data redundancy. So simpler signals will compress better (ie we have smaller file size for the same duration => lower bitrate), and more complex signals will be worse. That’s why lossless classical music has a lower bitrate than, say, rock. But it must be emphasized that the bit rate here is in no way an indicator of the quality of the sound material.
Now let’s talk about lossy compression. First of all, you need to understand that there are many different encoders and formats, and even within the same format, the encoding quality for different encoders can differ (for example, QuickTime AAC encodes much better than outdated FAAC), not to mention the superiority of modern formats (OGG Vorbis, AAC, Opus) in MP3. Simply put, from two identical tracks encoded by different encoders with the same bit rate, some will sound better and some will sound worse.
Also, there is upconversion. That is, you can take a track in MP3 format with 96 kbps bit rate and convert it to 320 kbps MP3. Not only will the quality not improve (after all, data lost during the previous 96 kbit / s encoding cannot be returned), it will even get worse. It’s worth noting that at each lossy encoding stage (at any bit rate and any encoder), a certain amount of distortion is introduced into the audio.
And even more. There is one more nuance. If, say, the bitrate of an audio stream is 320 kbps, this does not mean that the 320 kbps was spent encoding that very second. This is typical for constant bit rate encoding and for those cases where a person, hoping to get the highest quality, forces a constant bit rate too high (for example, setting CBR to 512 kbps for Nero AAC ). As you know, the number of bits assigned to a particular frame is regulated by the psychoacoustic model. But in case the allocated amount is much lower than the set bitrate, even the bit deposit is not saved (for terms see the article “What is CBR, ABR, VBR?”) – as a result, we get useless “zero bits” that simply “wrap up” the frame size to the desired one (that is, increase the size of the stream to the specified size). By the way, this is easy to check: compress the resulting file with a filing cabinet (preferably 7z) and look at the compression ratio – the more, the more zero bits (as they lead to redundancy), the more space wasted.
Lossy codecs (MP3 and others) can cope with modern electronic music, but cannot efficiently encode classical (academic), live and instrumental music.
The “irony of fate” here is that, in fact, everything is the exact opposite. As you know, academic music in the vast majority of cases follows melodic and harmonic principles, as well as instrumental composition. From a mathematical point of view, this leads to a relatively simple harmonic composition of the music.
