The most common misconceptions about digital audio

What is video and audio bitrate?

Bitrate is the number of bits per second, that is, the data transfer rate. The bitrate symbol is bps. Typically, it determines the size and quality of video and audio files: the higher the bitrate, the better the quality and the larger the file size, since File Size = Bitrate (kilobits per second) x Duration.

In most cases, 1 byte per second (1 byte/s) corresponds to 8 bits/s.

To find out what bitrate is, it’s worth understanding how video transmission of information works.

Any video is a changing series of pictures. To avoid the “slide show feeling”, the image changing speed should be at least 24 frames/sec. Each frame has parameters: width and height. The higher they are, the more pixels fit in one picture - the higher the quality.

Each “point” that makes up the frame has a weight, and it is equal to 1 byte.

Let's take one Full HD picture and calculate its weight - it will come out to about two megabytes (1920x1080=2,073,600). Thus, one second of video, containing 24 frames, would weigh 48 MB. This is where the concept of bitrate comes into play - this is the strength of video compression.

Those. The required file, when encoded, becomes lighter in weight. But from strong compression it can also lose quality. Of course, in reality everything is not so simple - a lot depends on the codec that is used for encoding. This is the name of the direct compression method. So, videos in different formats, but with the same bitrate, can produce pictures of different quality. The concept of “audio bitrate” is also applicable, which indicates the compression strength, but for an audio stream.

What it is

Bitrate (stream width/bandwidth) is the number of bits (units of information) used to transmit and process data per second of playback time. In other words, it is the minimum value of the channel capable of passing the flow without delays. The measurement unit is bits/sec and its derivatives – kbit/sec, Mbit/sec and others.

1. Audio bitrate is the amount of audio information that is output to a sound source per second of time.
2. Similarly with video files, the bitrate is the amount of image displayed on the screen per second.

Did you figure out what kind of video/audio bitrate this is? You might be surprised, but there are several varieties of bandwidth. We clearly explain what types can be divided into.

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The higher the video bitrate, the higher its quality?

Generally, a higher bitrate will result in higher image quality in the video output, but this is only relevant when comparing the same video at the same resolution.

The bitrate is expected to increase whenever the resolution increases as more data is processed.

So, a higher video bitrate may provide superior quality, but it can also put a serious strain on your hardware, which can cause it to freeze.

Kinds

There are several stream compression modes - it's worth exploring each of them. It is important to understand all the subtleties - this is the only way to finally understand what it is - bitrate.

As we said above, there are only three types. Let's go through each one.

Constant bitrate (CBR)

The given value does not change throughout the file: each second corresponds to a certain - and the same - number of encoded bits of data (even when talking about silence). It is used more often when processing files with constant sound and image characteristics.

Pros:

• Can be used to transmit streaming media over a limited communication channel;
• The size of the final document can be predicted with high accuracy.

There is also a drawback:

• Not suitable for works in which audio and video streams change dynamically, as it does not provide an optimal ratio of quality and size.

Variable (VBR)

The number specified by the user is automatically changed by the encoder program during the encoding process (for each frame, according to its characteristics). This technology is used to achieve the best ratio of size and quality.

Advantages:

• Files of varying saturation can be encoded at a certain quality (much higher than the average value);
• The document size is significantly reduced by fragments where a high flow width is not required.

The downside is this:

• It is impossible to predict the final file size - it could be significantly larger or smaller.

Average (ABR)

Average bitrate is a combination of the first two types! Used when creating large projects - with a small file size, you can achieve the highest quality encoding, while the accuracy of calculating the size is much higher. The value in kbit/sec is set by the user (as with CBR), and the program already varies it within certain limits (as with VBR).

Studying the existing types allows you to understand what the bitrate of video and audio affects. The higher this indicator, the higher the quality of the finished multimedia product! This is the degree of compression - the more the signal is compressed, the worse the quality and smaller the size. And vice versa!

What is the best video bitrate?

For stable video streaming, the resolution must match the correct video bitrate. For streaming on YouTube, Facebook or any other channels, the following video bitrates are most suitable:

• For full HD standard definition, set the bitrate to between 3500 and 5000 kbps.
• For regular standard definition HD videos, set the bitrate to between 2500 and 4000 kbps.
• For full HD high resolution, set the bitrate to between 4500 and 6000 kbps.
• For regular high-resolution HD videos, set the bitrate to between 3500 and 5000 kbps.

How to change the indicator

You already know everything about stream width - it’s time to figure out how to find out the video bitrate and change it. In most cases, you don't need third-party software.

1. If we are talking about files downloaded to your computer, just look at the desired value through the document properties;
2. If you want to know what bitrate is used on different services (for example, bitrate on Twitch), look at the technical specifications page in Help or Support.

Third-party services are needed to change the value - there are a lot of online converters or desktop applications. We recommend paying attention to Online Audio Converter, Diktorov.net, AIMP, online-convert.com.

They explained in detail what video and audio bitrates are, gave standard values, and talked about the varieties. Now you can choose the optimal number and change it if necessary!

Standard audio bitrates

To better understand the quality of music or voice recordings, we suggest you familiarize yourself with the table of average values. This includes encoding options with compression (MP3) and without (FLAC, WAVE).

• 6 Kbps - the lowest quality, used only for transmitting information, without aesthetic value, such as recordings from a voice recorder.
• 32 Kbps - also a low-quality recording, but a slightly higher level, used in more expensive voice recorders.
• 128 or 160 Kbps is the lowest quality threshold for music; usually, at this bitrate, noticeable distortion of the audio track is heard.
• 192 Kbps is an option in which deterioration can only be noticed occasionally, and this is how files were distributed in the early era of the Internet.
• 256 Kbps - with this compression, flaws in the sound can only be heard on professional equipment or good headphones.
• 320 Kbps is the maximum available bitrate for “lossy compression” formats, such as MP3. The difference in sound is only noticeable when compared with Lossless formats.
• 400–1411 Kbps - used for music and high-quality recordings in the “lossless compression” format (also called FLAC).
• 1411 Kbps - Uncompressed audio, usually in LPCM format when transferred to CDs.
• 5644 Kbps - beyond this mark is professional studio sound.
• 6 Mbit/s - fans of Blu-ray films can pamper themselves with this bitrate.
• 18 mbps is the standard for HDTV, the highest available quality.

Average bitrate

This type is a compromise between the above. The problem with variable bitrate is that the changes are automated and sometimes over-compression can occur. The average bitrate allows the user to set the range within which compression variation will occur. True, the technique of using it is not so simple and is used mainly in professional studios when working on serious projects. An additional advantage of the average bitrate is that it allows you to more accurately calculate the weight of the file, even when the compression strength changes.

Standard video bitrates

• 400 Kbps - low-quality videos shot on low-resolution cameras.
• 750 Kbps - used by YouTube algorithms for 360p videos.
• 1 Mbit/s is the same for 480p videos.
• 2.5 Mbit/s - videos in 720p.
• 3.8 Mbit/s - the same YouTube, only video in 720p and 60 frames/s.
• 4.5 Mbit/s - YouTube sets a similar bitrate for 1080p videos.
• 6.8 Mbps - the same thing, only at 60 fps.
• 9.8 Mbit/s - used in the production of DVDs with films.
• 19 Mbps - used by HDV format at 720p and 1080p resolution.
• 24 Mbit/s - often used when recording Blu-ray media.
• 29 Mbit/s - can be found on HD DVD discs.
• 40 Mbps is the highest quality for 1080p Blu-Ray recording.
• 45 Mbps - bitrate for 4K video

How to find out the bitrate of a file

How to calculate the compression strength of a video, and accordingly find out the degree of its quality. We're happy to say that you don't need any special programs; the data on the file's bitrate is basic and available for viewing by any user. To do this, right-click on the object and select “Properties”. In the window that opens, find the “Details” tab, and then “Data transfer speed”. The video bitrate will be indicated there. If you need information about an audio file, look for the “Bit Rate” item. True, there are several formats in which access to the bitrate is complicated - for example, MKV files. For them you need to use specialized software, for example MediaInfo.

What bitrate value should I set?

Now the most interesting part: what bitrate to choose when transcoding movies and music? What bitrate is better for 1920x1080 video, for example?

Naturally, there is no definite answer. Many factors need to be taken into account: for what audience the content is being made, what codecs will be involved in compression, etc. Based on average values, for Full HD video you can narrow the range to approximately 5-8 Mbps for a standard frame rate (FPS). For increased (48-60 FPS) it is worth taking values ​​from 7.5 to 12 Mbit/s. For an audio track, the lower quality threshold is 192 kbit/s, after which negative changes already audible to the naked ear will begin.

Lossy audio encoding. What is what?

Attention

: This is the old version of the article, the new one is available on my website. The year is 2011, 17 years have passed since the appearance of the first MP3 encoder. But the fact that most of us still quietly listen to music in MP3 format does not mean that progress has been marking time all this time. And this applies not only to the development of the MP3 encoding algorithm, but also to the evolution of lossy audio encoding in general - in the form of new, more advanced codecs that really allow you to get better quality at a smaller size. Formats such as OGG Vorbis, AAC, WMA, Musepack have long left behind the outdated MP3 with its many limitations and disadvantages.

At the same time, lossless coding is gaining momentum. But due to the large volumes of data today, it is still not suitable for full-scale use - especially for portable devices with limited memory, for network streaming, and simply for quickly sharing music on the Internet (admittedly, not everyone has 100-megabit Internet access is not always at hand).

And so, MP3 is outdated, and a replacement is definitely ripe for it. But what should an uninitiated user do, but who wants to achieve the highest quality sound with minimal memory consumption? After all, there are quite a lot of alternative codecs (at least 3 of them are really worthy of attention): Apple is promoting the AAC format (Advanced Audio Coding - positioned as a successor to MP3) using its iTunes Store, Microsoft is promoting its own licensed WMA (Windows Media Audio), in addition , OGG Vorbis is becoming increasingly famous, and especially enlightened ones even use such a format as Musepack. Which of these codecs should you choose?

There is no clear answer to this question - and that is why I am writing this article.

The choice of a particular codec depends on the specific task. Namely:

1. From the equipment and software with which the sound will be played. Those. on the availability of support for a particular audio format, as well as the quality of playback (it is advisable to use this as a guide when choosing a bitrate).

2. The amount of memory that will be allocated for the final material. A higher or lower target bitrate/quality is selected accordingly.

And, of course, in addition to the format and bitrate, it is necessary to select the optimal encoder and encoding parameters. It should be understood that different formats/encoders behave differently at different bitrate ranges.

So the algorithm is something like this:

1) Find out what formats the target device supports. 2) Decide how much space you can allocate for audio material, and also determine the total duration of audio intended for encoding. 3) Calculate the required bitrate using the formula: bitrate = disk_space (in kilobits) / total_duration (in seconds). 4) Depending on the bitrate, select the optimal one from the supported formats (more on this later). 5) Select the best encoder and parameters for it.

A.A.C.

The development of data compression methods and psychoacoustics gradually led to the fact that the MP3 standard became “cramped” for the implementation of new ideas in audio encoding. As a result, by 1997, the Fraunhofer Institute (Fraunhofer IIS), which created MP3 in the early 90s, as well as Dolby, AT&T, Sony and Nokia, developed a new audio compression method - Advanced Audio Coding (AAC), which became part of the standards MPEG-2 and MPEG-4. The main differences from the MP3 standard are:

• support for a wider range of formats (up to 48 channels) and audio sampling frequencies (from 8 kHz to 96 kHz);
• more efficient and simpler filter bank: the hybrid MP3 filter bank has been replaced by the conventional MDCT (modified discrete cosine transform);
• wider limits for varying the frequency-time resolution in the filter bank - eight times (in MP3 - three times) - led to improved coding of transients (transient processes) and stationary sections of the audio signal;
• better encoding of frequencies above 16 kHz;
• a more flexible stereo coding mode, allowing you to switch to M/S (“joint stereo”) mode independently in different frequency bands;
• additional features of the standard that increase compression efficiency: time-domain noise generation technology (TNS), long-term prediction of MDCT coefficients, parametric stereo coding mode, perceptual noise substitution, high-speed restoration technology frequencies (SBR).

Thanks to these features, the AAC standard is able to achieve more flexible and efficient, and therefore higher quality audio encoding. As a result of the widespread use of the MP3 format, the AAC standard has not yet gained popularity comparable to MP3. However, AAC is the dominant format in the popular iTunes Store, iPod, iTunes, iPhone, PlayStation 3, Nintendo Wii and DAB+/DRM digital radio.

OGG Vorbis

Ogg Vorbis is a relatively new universal audio compression format, officially released in the summer of 2002. It belongs to the same type of formats as MP3, AAC, VQF and WMA, that is, lossy compression formats. The psychoacoustic model used in Ogg Vorbis is close in operating principles to MP3 and others like them, but only the mathematical processing and practical implementation of this model are fundamentally different, which allows the authors to declare their format completely independent from all predecessors. The main indisputable advantage of the Ogg Vorbis format is its complete openness and freedom. Moreover, it uses the latest and highest quality psychoacoustic model, which is why the bitrate/quality ratio is significantly lower than other formats. The result is better sound quality, but smaller file sizes. The format has a large number of advantages. For example, the Ogg Vorbis format does not limit the user to only two audio channels (stereo - left and right). It supports up to 225 individual channels with sampling rates up to 192kHz and bit depths up to 32bit (which no lossy compression format allows), making Ogg Vorbis great for encoding 6-channel DVD-Audio audio. In addition, the OGG Vorbis format is sample accurate. This ensures that the audio data before encoding and after decoding will not have offsets or extra/lost samples relative to each other. This is easy to appreciate when you're encoding non-stop music (where one track fades into another) - ultimately maintaining the integrity of the sound. The possibility of streaming is no surprise to anyone these days, but this format has it built in from the very foundations. This gives the format a rather useful side effect - several compositions with their own tags can be stored in one file. When loading such a file into the player, all songs should be displayed as if they were loaded from several different files. Separately, it is worth mentioning the fairly flexible tag system. Header tags are easily expandable to include text of any length and complexity (for example, song lyrics) interspersed with images (for example, a photo of an album cover). Text tags are stored in UTF-8, which allows you to write in all languages ​​at the same time and eliminates possible problems with encodings. This is much more convenient than various tricks such as id3 tags. Ogg Vorbis uses a variable bitrate by default, but the latter is not limited to any fixed values, and it can vary by even 1kbps. It is worth noting that the maximum bitrate is not strictly limited by the format, and with maximum encoding settings it can vary from 400kbps to 700kbps. The sampling frequency has the same flexibility - users have any choice ranging from 2000Hz to 192000Hz. Ogg Vorbis was developed by the Xiphophorus community to replace all paid proprietary audio formats. Despite the fact that this is the youngest format of all MP3 competitors, Ogg Vorbis has full support on all known platforms (Windows, PocketPC, Symbian, DOS, Linux, MacOS, FreeBSD, BeOS, etc.), as well as a large number of hardware implementations . Its popularity today significantly exceeds all alternative solutions. It's worth noting that Ogg Vorbis is just a small part of the Ogg Squish multimedia project, which also includes free encoders: Speex - for voice compression; FLAC - for lossless audio compression; Theora - for video compression.

Musepack

MusePack (mpp, mp+, mpc, MPEG+) is an unlicensed file format for storing audio information, distributed under the GNU General Public License. The encoding quality of MPC at high bitrates (160 Kbps and above) is noticeably (if not significantly) higher than the quality provided by MP3. Main advantages:

• The format does not perform a second DCT conversion, and does not actually suffer from pre-echo artifacts, unlike formats such as MP3, Vorbis, AAC and WMA.
• More efficient variable bitrate algorithms. If you monitor how the bitrate changes while playing MPC tracks, you will notice that for simpler sections the encoder allocates a lower bitrate, and for complex sections it allocates a much higher bitrate, sometimes above 400 (!) kbps. It is also worth mentioning one interesting fact here: the MP3 encoder in VBR mode for silence allocates a bitrate of 32 kbit/s (at a sampling frequency of 44100 Hz), AAC and OGG Vorbis - 2 kbit/s, Musepack encodes silence with minimal costs, <1 kbit /s (for example, a minute of silence will take some 514 bytes). All this speaks to the extreme “efficiency” of this encoder.
• Powerful and flexible psychoacoustic model. Here we can mention, for example, a frame-based dynamic low-pass filter (other encoders have a fixed bandwidth for each quality preset).
• More advanced compression based on optimized Huffman tables (the same LAME MP3 wastes about 20% of the bitrate - just because of imperfect mathematical compression)

WMA

Windows Media Audio is a licensed file format developed by Microsoft for storing and broadcasting audio information.

Initially, the WMA format was advertised as an alternative to MP3, but today Microsoft is contrasting it with the AAC format. Nominally, the WMA format has good compression capabilities, which allows it to “bypass” the MP3 format and compete in terms of parameters with the Ogg Vorbis and AAC formats. But as has been shown by independent tests, as well as by subjective assessment, the quality of the formats is still not clearly equivalent, and the advantage even over MP3 is clear, as claimed by Microsoft.

Now straight to the heart of the matter.

To make your choice easier, I would like to share my experience gained through numerous comparisons, auditions, and analysis of the results of open hearing tests.

And so, below I will talk about the most suitable encoders for each individual case, as well as the correct choice of parameters. For conversion, I recommend using foobar2000 (configuring the converter is described in detail here), the parameters are indicated just for it. In addition, foobar2000 has a large number of useful DSPs that can be useful for audio pre-processing.

For those who are going to convert via the console or another program: the variable %s should be replaced with the name of the source file (or a similar variable), and %d with the name of the output file.

Please note that for each bitrate range, possible format options are indicated: the first is the highest priority. If your player does not support the first option, pay attention to the next one, and so on. As I already wrote, in fact, today only three codecs are worthy of attention - AAC, OGG Vorbis and Musepack. WMA, due to its closed nature, does not differ in special quality, but in most cases it is still better than MP3. Given that some alternative devices only support WMA, I will make recommendations for each of the four formats.

Regarding bitrates: you need to understand that the optimal encoding mode is the so-called. True VBR, i.e. mode with target quality, not bitrate. Ideally, the result is a track with variable bitrate, but constant quality (do not equate these two concepts - more complex parts of the track need more bits to maintain quality). Thus, the output bitrate is difficult to predict. Therefore, the bitrate values ​​below are indicated only as approximate, if possible - average for a large number of compositions of varying complexity.

The encoders mentioned in this article, as well as some other encoders, with Russian descriptions of the main parameters and recommendations can be found here.

Ultra-low bitrates (~25-40 kbps)

This range is great for encoding audiobooks.
And here there can only be one option - AAC, or more precisely, Nero AAC. The parameters are as follows: -lc -q 0.35 -ignorelength -if — -of %d

In this case, the material must first be converted to mono and resampled to a frequency of 22050 Hz (preferably with a SoX resampler). The output will be the usual Low Complexity AAC with a bitrate of about 25 kbps.

There are also options for music in this range:

1) Nero AAC. No transformations are needed here:

-q 0.15 -ignorelength -if — -of %d

Output: High Efficiency AAC v2 (with parametric stereo and HF synthesis), ~35 kbit/s. An excellent option for any Internet radio. Only here we must not forget that the decoder in the player must support HE-AACv2, otherwise you will get a complete absence of HF and monophony.

2) OGG Vorbis AoTuV - this modification of libvorbis includes an improvement in the low bitrate encoding algorithm and, even without SBR technology, is not much inferior to HE-AACv2. Command line:

-s %r -Q -q-2 — -o %d

The resulting files should be fully compatible with standard OGG Vorbis decoders. Bitrate is similar - about 35 kbps.

3) WMA 10 Pro. For such cases, Microsoft also has something like SBR (RF synthesis), which doesn’t sound as bad as it could. True, the bitrate is slightly out of bounds - 48 kbps.

-silent -a_codec WMA9PRO -a_mode 3 -a_setting 48_44_2_16 -input %s -output %d

Please note that older (especially hardware) decoders do not support WMA 10. For this case, you can use WMA 9.2 (the encoder is the same), although its quality at low bitrates is much worse.

-silent -a_codec WMA9STD -a_mode 3 -a_setting 48_44_2 -input %s -output %d

Low bitrate, ~64 kbps

Initially, I thought about going straight to higher speeds.
But since hydrogenaudio.org recently conducted a comparison of encoders at this particular bitrate, it would be a sin to miss it. 1) QuickTime AAC is the winner (except for the newly created Opus/CELT) of that very test. Below are the settings for the QAAC encoder:

-s -v 64 —he -q 2 —ignorelength — -o %d

The output is HE-AAC (with SBR, but without Parametric Stereo), which should be supported by various iPods and the like.

2) OGG Vorbis AoTuV - although it turned out to be quite far from QAAC, but still:

-s %r -Q -q0 — -o %d

3) And just in case, WMA 10 Pro:

-silent -a_codec WMA9PRO -a_mode 3 -a_setting 64_44_2_16 -input %s -output %d

For older decoders - WMA 9 Standard:

-silent -a_codec WMA9STD -a_mode 3 -a_setting 64_44_2 -input %s -output %d

Slightly higher, ~80-100 kbps

And I’m considering this bitrate because of Vorbis.
1) As tests have shown, the OGG Vorbis AoTuV encoder copes best with it:

-s %r -Q -q1 — -o %d

2) Nero AAC - very good result. In places where the highs are not so pronounced, it may sound even better than Vorbis (at highs it loses due to synthesis). 30 -ignorelength -if — -of %d

The profile used is HE-AAC.

De facto standard, 128 kbit/s

Interesting fact: many argue that for MP3 128 kbps is the “borderline bitrate”, from which quality indistinguishable from the original begins.
Perhaps this is so... for plastic Chinese speakers with blatnyak. In reality, this threshold is somewhere around 200 kbit/s, and new formats provide more stable quality at this bitrate. Modern coders have managed to lower this level of 128 kbit/s by almost half (again, according to the developers). But, nevertheless, if you have more or less decent acoustics (or headphones), on complex fragments the difference can be heard even at 128 kbit/s.

1) Nero AAC:

-q 0.40 -ignorelength -if — -of %d

The profile is regular AAC LC.

2) OGG Vorbis AoTuV:

-s %r -Q -q2.8 — -o %d

3) WMA 10 Pro:

-silent -a_codec WMA9PRO -a_mode 3 -a_setting 128_44_2_24 -input %s -output %d

For older decoders - WMA 9 Standard:

-silent -a_codec WMA9STD -a_mode 3 -a_setting 128_44_2 -input %s -output %d

~160-192 kbit/s

In this range, the difference between Nero, QuickTime AAC and Vorbis encoders practically disappears.
But here the same Musepack already comes onto the scene. It is at these bitrates that its advantage begins to manifest itself (due to the unusually flexible VBR mode, as well as a fundamentally different compression algorithm): 1) Musepack —silent —quality 5 — %d

2) Nero AAC -q 0.50 -ignorelength -if — -of %d

3) OGG Vorbis AoTuV: -s %r -Q -q5 — -o %d

4) WMA 9 Standard:

-silent -a_codec WMA9STD -a_mode 3 -a_setting 160_44_2 -input %s -output %d

Transparency threshold: ~200-225 kbps

What I was talking about.
At this bitrate, almost all encoders provide sound that is transparent to most listeners. And it is this range that is optimal in terms of size/quality. By the way, LAME MP3 also has a similar threshold in this area (VBR V2), but this codec has very big problems with pre-echo (distortion preceding sharp bursts of the signal), and Noise Shaping is often felt by ear (noise from quantization errors in this way transferred to the high-frequency region).

For codecs such as Vorbis, AAC and MPC, at this threshold even background noise begins to be clearly depicted in compositions.

1) Musepack —silent —quality 6 — %d

2) Nero AAC -q 0.55 -ignorelength -if — -of %d

3) OGG Vorbis AoTuV: -s %r -Q -q6 — -o %d

4) WMA 10 Pro:

-silent -a_codec WMA9PRO -a_mode 3 -a_setting 192_44_2_24 -input %s -output %d

WMA 9 Standard, maximum bitrate perceived by older decoders:

-silent -a_codec WMA9STD -a_mode 3 -a_setting 192_44_2 -input %s -output %d

Reasonable maximum: ~320-350 kbps

I should draw your attention: after ~225 kbit/s, increasing the bitrate most often no longer gives an audible increase in quality, and the file size naturally increases.
But still, for particularly complex compositions (and good equipment/ears) there are higher quality settings. At these bitrates, for encoders such as Museppack and Vorbis, I couldn’t even find killer samples (problem samples in which the shortcomings of the encoding algorithm are clearly manifested). And so: 1) OGG Vorbis AoTuV -s %r -Q -q9 — -o %d

2) Musepack —silent —quality 10 — %d

3) QAAC -s -V 127 -q 2 —ignorelength — -o %d

4) WMA 10 Pro -silent -a_codec WMA9PRO -a_mode 3 -a_setting 384_44_2_24 -input %s -output %d

Ahead of your questions: yes, for some of these encoders there are higher quality settings, but further increasing them no longer makes any sense.
Unless you really don’t care about the amount of memory occupied by music, and your device doesn’t have lossless support. That, in fact, is all I wanted to share with you. Try it, comment, ask questions.

SQ what does this mean in music?

SQ is not format/quality. Despite its slight similarity to the above abbreviations, SQ is from the world of car audio.

SQ is a competition in the design and sound of car acoustics. A well-designed audio system with balanced sound across all frequencies increases the participant’s chances of winning. Perhaps SQ stands for Sound Quality .

In car audio, there is also a popular abbreviation SPL, which stands for Sound Pressure Level, which means sound pressure level.

The volume of sound at competitions often goes off scale, the car literally rattles all sorts of parts. It is unlikely that anywhere else you will hear as much bass (low frequencies) as at car audio competitions.

I hope this information was useful. Good luck and good luck!