The size of a three-hour recording as 192 kbps. The most common misconceptions on the topic of digital sound. Licensed Audio CD sound better than their copies

The bitrate is indicated as one of the main characteristics of video and audio recordings. Most users got used to think that it defines the quality of the file being downloaded. But what is bitrates and how do they actually characterize music files and videos? Consider this in more detail.

What is bitrates?

Bitrate is a value that displays the number of information units (megabit or kilobit), which is fixed in one second of the file playback. Accordingly, it is measured in megabits per second (MBPS) or kilobits per second (KBPS). Otherwise, the bitrate can be described as a bandwidth bandwidth. This feature is important for those who want to convert files, because with the same duration, the greater bit rate will lead to an increase in the file. In addition to the size, the sound quality changes. Reducing the size when lowing the bitrate is called compression.

A common musical is an audio file, compressed to such an extent that the standard disk is placed up to 12 hours of music. At the same time, the quality remains high enough due to psychoacoustic compression: from the entire range, sounds with those frequencies and volumes of volume, which are not captured by the human ear are removed. Selected sounds are formed into separate blocks, called frames. Frames have the same sound duration and compress on a given algorithm. When the music is played, the signal is recreated from decoded blocks in a specific sequence.

What is commonly used compression?

The audio bit rate is most often 256 kbps. With this value, the audio recording is compressed in the amount of approximately 6 times, due to which one disc can be recorded 6 times more music than before compression. If the bitrate is reduced to 128 kbps, then one disk will fit already 12 times more music, but the sound quality will be noticeably lower. Music recorded as 128 kbit / s is most often offered for listening on the Internet, since in pursuit of increasing the page loading speed of the resource owners go to any sacrifices. Many users note that its quality is far from perfect.

Now, when it became clear what bitrates are, it's time to determine their optimal level. Both lovers and professionals are infinitely arguing how the bitrate affects the sound quality and does it affect at all. On musical albums, as a rule, a bitrate is indicated. The same disc recorded as 128 Kbps and 256 Kbps will vary by price twice.

Optimal bitrate under different listening conditions

For many people, twelve-fold compression does not represent any damage, while others argue that they cannot listen to music with bitrate lower than 320 kbps. Paradoxically, but those and others are right. The fact is that ultimately the quality of playback depends not on the reproduction conditions and even from the type of music.

For example, the song is played on the tape recorder installed in the domestic car. In this case, quality at 192 Kbps will be quite sufficient. A higher bitrate will improve the sound quality, but the difference will not be noticeable due to the high level of noise during the trip. If the music plays on a home computer or a portable player, it takes at least 256 kbps. If the signal is not subject to change, is transmitted to external devices and is displayed on expensive imported columns, then it should be resorted to minimal compression. It is possible when bitrate 320 kbps.

Optimal bitrate for various musical styles

Music with a high bit rate is not always needed. Popular music, as a rule, sounds quite good when bitty 192-256 kbps. It is possible to install a higher quality, but there is no point in this: the pop songs are short-lived, so the saving of the disk space should be priority. In addition, the quality of the source records is also mediocre, so the increase in the bitrate may not affect the quality of the playable file. For listening in transport and in unofficial parties, the average quality is enough.

If we are talking about classical music, the works of legendary rock groups or rare copyright songs, then the quality should be above all. When purchasing such music, you need to look at the bitrate specified on the disc packaging. If the song is loaded from the Internet, then such information should be present on the download page. In addition, the bitrate is displayed in the player during playback.

Bitrates video files

It was said above that such a bitrate of audio recordings. But what is a bitrate video? Given that the video is played as a sequence of sounds and images, the definition of the bitrate will be similar. The presence of the video detects the file, but ultimately the image for the processor is the same zeros and units as sounds. The principle of information encryption is the same for all types of files.

Over the past few years, it became a terrible fashionable and popular MP3 format. On any tray selling computer CDs, you can easily find more than a dozen disks of the "complete anthology of the group XXX", and below the modest such inscription - mp3. Most often for the full picture on the covers there is a fashionable phrase CD Quality - then you mean the quality, like Audio-CD. It is about this that will not only be next our story - about MP3, what they happen about the quality of sound in mp3.

About mp3 format

To begin with, we will understand a little with the subject area. What does this represent this mp3 in general?

MP3, more correct name MPEG-1 Layer 3 - Standard for compressing audio information with losses. At the same time, the main purpose of creating the standard was to ensure the maximum "identical" source sound, as well as minimizing the volume of stored data. For this, an original coding scheme was created - in the first stage, the digitized sound is divided into frequency components that pass through a series of filters.

The main difference between MP3 from previously existing standards is in filtration. Standard Developers created the so-called psychoacoustic model - a model that takes into account some features of human hearing, and on the basis of this model from the audio signal, those frequencies are filtered out, the absence of which hearing almost does not notice. At the second stage, the resulting stream is encoded according to the Huffman algorithm with a static table. The result and will be a mp3 stream.

In addition, ID3 tags can also be added to the MP3 file (tags containing the name of the song, performer, other information) and various service information.

Compression and bitrate modes

The width of the stream - the bit rate determines how many bits are necessary for encoding 1 second of music. The MP3 standard regulates the streams from 8kbit / S to 320Kbit / s. The most typical bit rate is 128kbit / s.

Based on the stream, it is easy to calculate how much one minute of music will occupy - you need to divide the bit at 8 (the number of bits in the bate) and multiply by 60 (seconds per minute) - we get the number of kilobytes. For the already mentioned flux of 128Kbit / S it will be 128/8 * 60 \u003d 960 kilobytes or near megabytes per minute of recording.

It is quite natural that the greater the bit rate, the more the sound details can be saved, the more realistic it sounds. In the selection of the bitrate when coding, you have to sacrifice any quality in favor of small size, or the size in favor of quality.

The easiest MP3 compression mode is a constant bitter mode (CBR, Constant Bitrate). Earlier, at almost 100% mp3 assemblies, a 128Kbit / S bit rate was used above, and the CD Quality inscription was present on the disks. Frankly, it's just a slightening lie. In practice, to distinguish the sound of such MP3 from the sound of the audio CD, it is impossible not just on the cheapest acoustics.

Quality level on a bit rate of 128Kbit / s is an approximately the level of the middle tape recorder on not the fresh film can be slightly better. You can also add that this bitrate is widespread in the entries available on the Internet.

To simplify the parsing of higher bitrates, I will write their mesh: 128kbit / s, 160kbit / s, 192kbit / s, 224kbit / s, 256kbit / s, 320kbit / s. So, the bitrates 160 and 192kbit / s are already noticeably better in quality than 128kbit / s, but the received files are still not so high. "Artifacts" (flaws) codec is almost imperceptible (at least on my system).

With a Bitret of 224, I never had to meet in my pure form, so I can't say anything about his quality, but it should be higher than on the previous step of the ladder of bitrates. In addition, I did not meet reviews that cover this bitrate. Apparently it is somehow due to the fact that the first bit rate of 256kbit / s is recognized in terms of the accuracy of the sound transmission, almost a complete lack of distortion. In the instructions for the LAME codec, this bitrate is even named as Studio Quality. And the ceiling - 320kbit / s is designed for those whose quality is more expensive, or for owners of very high-quality Hi-Fi or even Hi-End equipment.

We now turn to a slightly more complex issue - the variable bitrate mode (VBR, VARIABLE BITRATE). Here the concept of bitrate is very blurry, the codecs "for the user" generally use adjustment only in quality (such as Xing Audio Catalyst). Other (LAME) allow you to set additional parameters - the minimum and maximum bitrates, again the quality.

When encoding the VBR codec, it selects the desired bitrate itself, based on the parameters given to it, and during the encoded fragment, the bitrate may change. An already mentioned psychoacoustic model is used to assess the desired bitrate. However, the model (since it is not perfect, nothing in our world is perfect) sometimes shows incorrect results. This leads to inclusion of the assessment, and, accordingly, the fall of actually audible sound quality.

The developers of the LAME codec are advised in this case to set the minimum bitrate threshold to avoid very bad results. The varieties of VBR refers and encoding ABR (Average Bitrate), averaged bitrate. Recently, only positive responses about this mode are heard in the reviews, especially ABR on 256kbit / s. This mode works almost as well as VBR, with the exception that the codec holds the average specified value. Currently, I am known to me only one codec that has the ABR mode is LAME.

Choice Codec

Literally recently, the user who wanted to get a decent quality mp3 was not a very large selection - it is some ISO-based codec (based on the codec codec codec released by International Standarts Organization), or codec from IIS Fraunhofer (Institute - MP3 Developer ). Plus codecs in Xing products.

After reading different reviews, and making small own research, I came to the conclusion about the firm of the firm of the firm of Xing - this ... they are better not to use. Even in relatively new versions, all their products that can create mp3 built-in means do it as much as possible.

There is also a lot of "pioneering" crafts, elbowed on a stovered in the Xing codec (almost all contain a Tompg.exe file as part). For a long time, their main advantage was the speed (to the detriment of quality), but today the LAME codec shows comparable speed with higher quality. In addition, Xing products generally speaking costs money, while Lame is free by definition.

Next, I will go on IIS Fraunhofer products. All their programs for compression mp3, available for free, are highly cut down by the possibilities of the versions of their commercial products. Then, all of their codecs did not develop for a long time, and do not contain new tools, supporting VBR / ABR, in addition, not differing in special speed. The only justified application - compression on bitrates below 128kbit / s - they carried out special optimization for low bitrates (places, however, with a violation of the standard).

Different codecs based on ISO code suffer in principle with the same disadvantage - low-quality compression on bitrates below 192kbit / s. In addition, most of them (including bladeenc) are pretty slow.

In my opinion, the most optimal option is the LAME codec. Started as a free codec based on the ISO code, during the development process it has grown and now all reviews when comparing MP3s with other formats are used precisely as a reference for MP3. Little more than a year ago, the LAME project finally got rid of the ISO code and can now be considered a completely independent codec.

The development of the codec is quite intensive, it is constantly updated, correct errors. In addition, it is possible to use LAME not only under Windows, but also for various options for UNIX systems, it also works in pure DOS. Again, the source code is completely free, the source code is available (for lovers to eat it), already compiled binary files (.exe and.dll) are available from several sites, optimized for different processors.

There is also a slightly trimmed version of the LAME - Gogo-No-Coda encoder, which shows fantastic results (twice as fast as fast LAME).

So what is the bitrate and what mode to use?

Considering all of the above, I would recommend putting a MP3 file either with a 320kbit / s stream, CBR mode, or 256kbit / S, ABR. The first in my opinion is somewhat preferable, because You get the most accessible quality within the format. For recordings to "listen and erase" a couple of times, it is reasonable to use ABR 192Kbit / s.

And one more - it is better not to use the bitrate for some long storage below 192kbit / s - if only the record with which MP3 was made, you are not constantly at hand (although remember that the analog record on the magnetic tape is deteriorated over time) .

Very often, the argument that I hear in favor of low bitrates and the "curve" of compression is "I have bad acoustics, and I still can't hear the difference." Everything can change, or you have to use your archive on a decent equipment, and it will not be possible to get to the initial record. The answer is absolutely not thorough, I can bring the case from my own practice.

In our city Pavlovo was once a small club, where the music was played from the computer (MP3 with a bit rate not higher than 160Kbit / s). The club further passed away, and the computer with music archives moved to another firm engaged in mass events. Imagine that they took to twist this music at the bottom of the city! Horror, when all defects brought by packaging on such a small bit rate were heard on more or less decent acoustics. The sound was worse than with their seaside tape recorder with semi-advanced cassettes. It would be reasonable to avoid the repeat of other people's mistakes, right?

Test equipment and software

Computer: Athlon TB 650MHz, M / B ACORP 7KTA 100MHz FSB, 128MB RAM PC-133, HDD Quantum 40GB 5400RPM, SoundBlaster 16 Vibra, AC97 Codec.
Audio system: Radiotehnika amplifier U-7111, a pair of Radiotehnika S-90B speakers.
By: Windows98 SE, Winamp 2.75, EAC 0.9PB11, LAME 3.90A, GOGO-NO-CODA 3.07A

The triumphal procession of the MPEG-1 Layer 3 sound recording format (in the surroundings of the MP3 designation) is explained by the fact that a simple and effective way to compress the sound file that allows you to store on a standard CD-ROM disk up to 12 hours of admissible quality.

If we say it is simplified, the MPEG-1 algorithm Layer 3 is based on the method of so-called "psycho-acoustic" compression, when the spectrum sounds are excluded not perceived frequency hearing and volume levels. The spectrum is "purified" in such a way, the spectrum is divided into separate blocks (frames) of the same duration and shrinks in accordance with the specified requirements. When playing, the signal is formed from the sequence of decoded frames.

The degree of compression depends on the parameters of the audio stream, which must be obtained at the output after decoding the file.

The main parameter determining the quality of the sound and the degree of compression is the so-called (what is) bitrate - bandwidth, measured in bits per second.

The more this indicator, the better quality Sound and less compression ratio. Since almost all MP3 files are recorded in stereo mode with an encoding frequency 44 kHz and a depth of 16 bits that define clean sound factors become: a source of recording used codec and a selected bitrate.

The word codec is formed by a combination of words encoder + decoder. This software is a module that allows you to encode or decode sound or video files in accordance with your own algorithm.

The average flow rate of 256 kbps provides a compression ratio of approximately 6: 1, for other values, the degree of compression varies proportionally. Thus, with a stream of 256 kbps, you can burn music from six ordinary Audio CD to a CD, and with a 128 kbit / s stream - from twelve conventional musical disks.

Regarding the value of the bit rate, providing good sound quality corresponding to the quality of Audio CD playback, there are endless disputes among lovers and professionals.

Some they consider sufficient The level 128 Kbps, others satisfy only the maximum flow value - 320 kbps. In all likelihood, the right and others are the difference only in what is written and in what conditions is reproduced.

The magnitude of the bitrate with which the digitized sound was encoded is usually indicated on the CD cover. For example, a complete collection of BEATLES group music can be purchased on three discs with a bit rate of 128 kbps or on six discs with a bit rate of 256 kbps.

It is clear that in the second case, the purchase cost will be twice as expensive, but also quality is better.

If the music sounds in the car of domestic production, the thread 192 kbps will provide sufficient sound quality, you still do not hear the best due to extraneous noise. To listen to a computer or stand-alone player ( MPZ-Perter) accept a stream of 256 kbps.

But if the signal changes unchanged to the external device and is displayed on high quality columns, the maximum possible flow is desirable - 320 kbps. Based on the listed considerations, the universal can be considered a stream of 256 kbps: with good quality recording, it will provide adequate playback in most cases.

To broadcast music via the Internet, a stream value is usually 128 kbps. At the same time, the quality of the sound "as if" leaves much to be desired.

To record popular music with a bit rate above 192-256 kbit / s does not make sense: the songs live long, and the original records are often not distinguished by high quality. In the end, it is possible to pay and under the sound of the "tape recorder".

It is quite another class of classics and rare copyrighted works. And under the classics we understand not only Bach or Mozart. Today, the BEATLES, LED, Zeppelin, and Vysotsky, and Tsoi, and many other authors can be considered a classic.

If you do not pay attention to the bit rate specified on the package when buying a CD, then you can view the value in the player string during file playback.

Bitret (from English bitrate.) Audio files call the number of bits (units of information) used for storing one second recording. The most common unit of measurement of the bitrate is the number of kilobit per second (Kbps, Kbps). The bitrate is one of the key characteristics of multimedia files that affects their quality and size. With a large bit rate, music or video was recorded, the better their quality and the "volume" will be records.

Accordingly, the change in the value of the bit rate in one direction or another may increase or decrease the file size. But with the influence on the quality of the recordings, everything is more complicated. Whereas the decrease in the value of the bit rate naturally leads to a deterioration in the quality of the source file, the opposite operation on quality does not affect. Even if you set the maximum bit rate, the sound quality and video deer of your file will remain the same.

As you can see, it is not possible to increase the bit rate of the record of a special meaning: as a result you will receive a larger file with the same quality. But to reduce the bitrate in order to reduce the size of the recording, it is very possible. Want to try to change the bitrate of your songs or movies? Download the Movavi video converter - a convenient program with which you can easily change the video and audio sequence bitrates, be it files in popular MP3, WMA, AVI and MP4 formats or records placed in more exotic containers. The instruction is written on the example of working with audio files.

1. Install the program to change the bitrate

Download and run the movie Converter Movavi distribution. Follow the onscreen instructions to install the program. At the end of the installation, the converter will start automatically.

2. Add files to the program

Press the button Add files, Select Add audio and put in the program files . The program supports many media formats, so the input file format can be almost any. Change the MP3, WMA, AAC audio files bitrate. Try to reduce the video bitrate: Work with video recording in AVI, MP4, DivX and various HD video formats. The program will help you cope with a wide range of media file conversion tasks!

3. Select Conservation Format

Before changing the bit rate, select the format in which your audio recordings will be saved. To do this, click on the tab Audio And select the appropriate format from the list. By choosing in favor of this or that audio format, click on its name and from the list of the list, select one of the available bitrate values \u200b\u200b(option is not available for FLAC, OGG, WAV and M4A formats). If you do not want to change the standard bit rate specified in the selected profile, you can skip the next step and proceed to conversion.

4. Set the desired bit rate

Press the gear button to the right of the field Output format. In the list Type of Bitrate Choose

Reliable and efficient program for recording video from the screen in HD. Capture video from programs, online broadcasts and even conversations in Skype and save clips in any popular format, as well as to view on mobile devices.

The bitrate is taken to use when measuring the effective transmission rate of the data stream through the channel, that is, the minimum channel size, which can skip this stream without delay.

Bitrate is expressed by bits per second (bit / c, bPS.), as well as derived values \u200b\u200bwith kilo- (kbit / s, kbps.), mega- (Mbit / s, MBPS.) etc.

Data transfer rate using bits per second block (symbol: "bit / s"), often applied in combination with consoles from the international unit measurement system (C), such as "Kilo" (1 kbps \u003d 1024 bt / s) , "Mega" (1 Mbps \u003d 1024 kbps), Giga (1 Gb / C \u003d 1024 Mbps) or "Tera" (1 Tbit / C \u003d 1024 Gb / s). Non-standard abbreviation "BPS" is often used to replace the standard "bit / s" symbol, so that, for example, "1 Mbps" is used to designate one million bits per second. One byte per second (1 b / c) corresponds to 8 bits / s.

Characteristics

In streaming video and audio formats (for example, MPEG and MP3) using compression with quality loss, the bit parameter expresses the thread compression ratio and, thereby, determines the channel size for which the data flow is compressed. Most often, the bitrate of sound and video is measured in kilobates per second (eng. kilobit Per Second, Kbps), less likely - in megabits per second (for video only).

There are three streaming data compression modes:

CBR (eng. Constant Bitrate.) - with a constant bitter;
VBR (eng. Variable Bitrate.) - with variable bitrate;
Abr. (eng. Average Bitrate.) - with averaged bit rate.

Information transfer rate

Physical level of pure bitrate, information transfer rate, useful bit rate, payload frequency, net data rate, coded transmission rate, efficient data transfer rate or wire feed rate (unofficial language) digital communication channel is the ability without taking into account the physical laying protocol, For multiplex, an example with a temporary separation of channels (TDM) framing bits, reserved with direct correction of errors (FEC) codes, Equalizer of training characters and other channel coding. Noise-resistant codes are common, especially in systems. wireless communication , broadband modem standards or modern high-speed local networks Based on copper. The physical level of pure bitrate is the data transfer rate, measured at the control point at the interface between the channel level and the physical level, and, therefore, may include a data transfer line, as well as a level load.

In modems and wireless systems, communication lines adaptation (automatic adaptation of data and modulation speed and modulation and / or encoding scheme errors, signal quality) is often used. In this context, the term of the bitrate peak means a pure bitrate of the rapid and least reliable transmission mode, used, for example, [when the distance is very short closure] between the sender and the transmitter. Some operating systems and network equipment can detect the "connection speed" (unofficial language) of a particular network access to a network or communication device, which implies the current clean data transfer rate. It should be noted that the term line speed in some textbooks is defined as gross transmission rate in bits, and in others, as a clean data transfer rate.

The relationship between the cumulative bit rate and the net data transfer rate depends on the rate of the PI code in accordance with the following.

Permanent Bitrate

Permanent Bitrate - A variant of streaming data encoding, in which the user initially specifies the required bitrate, which does not change throughout the file.

His main dignity is the ability to quite accurately predict the size of the destination file.

However, the option with a constant bitter is not very suitable for musical works, the sound of which is dynamically changed over time, since it does not provide an optimal ratio size / quality.

Variable bitrate

FROM variable bitrate The codec selects the value of the bitrate based on the parameters (level of the desired quality), and during the encoded fragment, the bitrate may vary. When compressing sound, the desired bit rate is determined on the basis of a psychoacoustic model. This method gives the best quality / output file ratio, but its exact size is very poorly predictable. Depending on the nature of the sound (or images, in the case of video encoding), the size of the file received may differ several times.

Averaged bitrate

Averaged bitrate It is a hybrid of permanent and variable bitrates: the value in KBIT / C is set by the user, and the program varies it in some limits. However, in contrast to the VBR, the codec with caution uses the maximum and the minimum possible values, not at risking the average value for the user specified by the user. This method allows the most flexibly to set the processing speed (for audio it can be any number between 8 and 320 kbps, against numbers, multiple 16 in the CBR method) and with a much greater (compared to VBR) accuracy to predict the size of the output file.

Mp3

MP3 compression audio compression with data loss. The sound quality is improved with the increase in the bitrate:

32 kbit / s - as a rule, only acceptable for speech
96 kbps - usually used to transmit speech or low-quality streaming sound
128 or 160 kbps - the initial level of music coding
192 Kbps - acceptable music coding quality
256 kbps - high quality Music coding
320 kbps - highest quality coding supported by standard mp3

Other audio

700 bps - the lowest bit rate used by the open source speech codec Codec2; Voice is barely recognized, bitrate 1.2 kbit / s gives much better sound
800 BIT / C - the minimum required level for speech recognition, used in specialized speech codecs FS-1015
2.15 kbit / s - Minimal bit rate SPEEX codec with open source
6 Kbps - Minimal Bitrate OPUS OPUS Codec with open source
8 kbps - telephone sound quality using speech codecs

high-quality audio

Here we will look at how to choose the correct bitrate for your Internet broadcast. And so, bitrate is the video quality. What he is higher, the higher the quality. If you make a quality stream with a magnificent picture, then you just need to raise the bitrate and all? No matter how. Stream stream is in online, respectively, all this high bit rate takes the Internet channel and it will be impossible to watch it. Therefore, you need to take into account your internet and Internet of your audience. Not everyone is stretched fiber. So above 2 Mbps, it is not recommended to put a bitraine.

The second thing to pay attention to is, the so-called bit / pixel ratio. This formula looks simple:

bIT / (pixels * Frames)

What does this formula mean? Suppose we encode stream with a resolution of 100px x 100px, 25 fps (frames per second) and put a bitrate of 250 kbps (kilobit per second). So, for a second, a video size of 10,000 pixels (one hundred I multiply a hundred) highlights 25 frames and 250 kilobit. It turns out 10 kilobit (10,000 bits) for each frame (250/25). We divide the bits allocated to the frame, on the size of pixels - we obtain the relationship of the bit / pixel - how much information is allocated for the "encoding" of one pixel.

The more information stands out - the higher the quality.

In our example, the bit / pixel attitude is: (10,000 bits per frame) / (10,000 pixels) \u003d 1. It will be a bit too much. Full great quality can be obtained with the ratio 0,1 -0,15 . For our example, there would be enough bitrate ~ 32-35 kbps.

Calculate the estimated BIT / Pixel ratios for the most common permits:

720p: 1280 × 720 Points:

Bitrate 1500Kbps - 1500000 / ((1280 * 720) * 25) \u003d 1500000/23040000 \u003d 0.065
Bitrate 2500Kbps - 2500000 / ((1280 * 720) * 25) \u003d 2500000/23040000 \u003d 0.109
Bitrate 3500Kbps - 3500000 / ((1280 * 720) * 25) \u003d 3500000/23040000 \u003d 0.152

1080p: 1920 × 1080 Points:

Bitrate 1500Kbps - 1500000 / ((1920 * 1080) * 25) \u003d 1500000/51840000 \u003d 0.029 ( as you can see, quality with the same bit rate will be worse somewhere 2.5 times, so for 1080r you need a greater bit rate than for 720p)
Bitrate 5000Kbps - 5000000 / ((1920 * 1080) * 25) \u003d 5000000/23040000 \u003d 0.096
Bitrate 7500Kbps - 7500000 / ((1920 * 1080) * 25) \u003d 7500000/23040000 \u003d 0,145
Bitrate 10000Kbps - 10000000 / ((1920 * 1080) * 25) \u003d 10000000/23040000 \u003d 0.192

What conclusions can be done? The first, it is the main thing, you can not provide permission to the necessary bitrate - do not try to hate. Do you want to hate anyway? Reduce or permission or FPS. Deed Bit / Pixel at least to 0.075-0.1, and better more.

Quality

Resolution

Video bitratekbps.

Audio bitratekbps.

FPS frames / sec

Video codec

h.264.profile

Audio codec

Audio-channel

240 p. (426 x 240)

400 (300-700)

Aac or MP3

270p. (480x270)

400 (300-700)

Aac or MP3

360p (640x360)

750 (400-1000)

Aac or MP3

480p. (854x480)

1000 (500-2000)

Aac or MP3

540p (960x540)

1000 (800 - 2000)

Aac or MP3

Mono or
Stereo

720p (1280x720)

2500 (1560-4000)

Aac or MP3

Mono or
Stereo

720p (1280x720)

3800 (2500-6000)

Aac or MP3

Mono or
Stereo

1080p. (1920x1080)

4500 (3000-6000)

Aac or MP3

Mono or
Stereo

1080p. (1920x1080)

6800 (4500-9000)

Aac or MP3

Mono or
Stereo

1440p) (2560x1440)

9000 (6000-13000)

Aac or MP3

Mono or
Stereo

1440p (2560x1440)

13000 (9000-18000)

Aac or MP3

Mono or
Stereo

4k / 2160r. (3840x2106)

23000 (13000-34000)

Aac or MP3

Mono or
Stereo

4k / 2160r. (3840x2106)

35000 (20000-51000)

Aac or MP3

Mono or
Stereo

Home / Instructions / We select a bitrate for streaming

Note: For a better understanding of the following text, I highly recommend you to familiarize yourself with the foundations of digital sound.

S: The more bitrate, the better the track

R: This is not always the case. To begin with, I remind you what bit of t. (Bitrate, not Bitraid). In fact, this is the rate of data flow in kilobits for a second when playing. That is, if we take the size of the track in kilobits and divide on its duration in seconds, we obtain its bitrate - t. N. File-based Bitrate (FBR), it is usually not too different from the audio bit bitrate (the reason for the difference is the presence of metadata - tags, "sewn" images, etc.).

Now take an example: the bitrate of an uncompressed PCM audio recorded on the usual Audio CD is calculated as follows: 2 (channel) * 16 (bit on each sample) * 44100 (samples per second) \u003d 1411200 (bit / s) \u003d 1411.2 Kbps . Now, we will leave the track any Lossless codec ("Lossless" - "Binding", that is, such that does not lead to the loss of any data), for example, the FLAC codec. As a result, we will get a bit rate below the original, but the quality will remain unchanged - here's the first refutation.

This is still worth adding something. The bit rate at the output with lossless compression may turn out to be the most different (but, as a rule, it is less than that of uncompressed audio), it depends on the complexity of the compressible signal, or rather from the redundancy of the data. Thus, simpler signals will be compressed better (i.e. we have a smaller file size with the same duration \u003d\u003e Little bitrate), and more complex - worse. That's why classical music Lossless has a smaller bit rate than, say, rock. But it is necessary to emphasize that the bitrate here is by no means an indicator of the quality of the sound material.

Now let's talk about lossy compression (with losses). First of all, it is necessary to understand that there are many different coders and formats, and even within one format, the quality of coding in different encoders may differ (for example, QuickTime AAC encodes much better than the outdated FAAC), not to mention the superiority of modern formats (OGG Vorbis, AAC , Opus) Over MP3. Simply put, of the two identical tracks encoded by different encoders with one bit rate, some kind of sound better, and some kind of worse.

In addition, there is such a thing as apoplevert. That is, you can take a track in mp3 format with a bitter of 96 kbps and convert it to MP3 320 kbps. Not only does the quality improve (after all, the data lost at the previous coding of 96 kbit / with the data is no longer returning), it will even deteriorate. It is worth indicating that at each stage of lossy encoding (with any bit rate and any encoder), a certain portion of distortion is made in the audio.

And even more. There is another nuance. If, say, the bit rate of the audio stream is 320 kbps, this does not mean that all 320 kbps went to coding that second. This is characteristic of coding with a constant bitter and for those cases when a person, hoping to get the maximum, the quality is forcing too much permanent bitrate (as an example - setting 512 Kbps CBR for Nero AAC). As is known, the number of bits highlighted on this or that frame is regulated by a psychoacoustic model. But in the case when the allocated amount is much lower than the established bit rate, it does not save even a bit reservoir (read about the terms in the article "What is CBR, ABR, VBR?") - As a result, we get useless "zero bits" that simply "finish »Frame size to the desired (i.e., increase the flow size to the specified). By the way, it is easy to check - squeeze the resulting file with an archiver (better than 7z) and look at the degree of compression - the more it is the more zero bits (since they lead to redundancy), the greater the wonderful place.

S: DVD-Audio sounds better than Audio CD (24 Bit VS 16, 96 KHz VS 44.1, etc.)

R: In principle, it is quite logical, and even partly the truth, but only people usually look only on the numbers and very rarely think about the effect of one or another parameter.

So, consider to start the bit. This parameter responds to anything other than the dynamic range, i.e. The difference between the most quiet and loud sounds (in dB). In digital audio, the maximum level is 0 DBFS, and the minimum is limited to noise level, i.e., actually the dynamic range of the module is equal to the noise level. For a 16-bit audio dynamic range calculated as 20 * log (2 ^ 16)? 96.33 (DB). In this case, the dynamic range of the symphony orchestra is up to 75 dB (mostly about 40-50 dB).

And now imagine real conditions. The noise level in the room is about 40 dB (do not forget that dB - the value is relative. In this case, the threshold of hearingness is accepted in 0 dB), the maximum volume of music reaches 110 dB (so that there is no discomfort) - we get a difference of 70 dB. Thus, it turns out that the dynamic range of more than 70 dB in this case is simply useless. That is, when the range above or loud sounds will reach a painful threshold, or quiet sounds will be absorbed by the surrounding noises. Achieve the level of ambient noise less than 15 dB is very difficult (since this level is the volume of human breathing and other noise due to the human factor), as a result, the 95 dB range is completely sufficient for listening to music.

Now about the sampling frequency (sample frequency, Sample Rate). This parameter is responsible for the quantization frequency in time and directly affects the maximum frequency of the signal, which can be described by the audio presentation. On the Kotelnikov Theorem, it is equal to half of the sampling frequency. That is, for the usual frequency of seams in 44100 Hz, the maximum frequency of the components of the signal is 22050 Hz. Maximum frequency. Which is perceived by the human ear - just above 20,000 Hz (and at birth; as the threshold is growing up to 16,000 Hz).

Read the downloads in Format 24/192 - why they do not make sense.

S: Different software players sound differently (e. G. Foobar2000 is better than Winamp, etc.)

R: To understand why this is not the case, it is necessary to figure out what is the program player. In essence, this is a decoder, handlers (optional), output plug-in (one of the interfaces: ASIO, DirectSound, Wasapi. Etc.), and of course the GUI (user). T. K. The decoder in 99.9% of cases works according to a standard algorithm, and the output plugin is just part of the program that transmits the flow of a sound card through one of the interfaces, then the reason for the differences can only be handlers. But the fact is that handlers are usually turned off by default (or must be turned off, since the main thing for good player - Be able to pass the sound in the "pristine" form). As a result, the subject of comparison can only be capabilities Processing and output in which, by the way, there is no need for a very often. But even if such a need is - then this is already a comparison of handlers, and not the players.

Here I would also like to mention my and, perhaps, upset users who admire the "enormous" change in sound after the settings described in it - in 95% of cases it is self-impression (except for those cases when some "was turned off during its configuration" Improvement "or other handler, spoiling the whole picture). How sad, winning from all these tricks with replaygain, resamplers and limites - mezera. Conclusion: Want to really high-quality sound - buy yourself a Hi-Fi acoustics and a professional audio card.

S: Different versions Drivers sound differently

R: Based on this statement lies a banal ignorance of the principles of sound card. Driver is software necessary to effectively interact the device with operating system , as well as usually providing a graphical user interface to manage the device, its parameters, etc. The sound card driver provides a sound card recognition as a sound card, reports OS on the formats supported by card, ensures the transmission of uncompressed PCM (usually) the flow of the card, as well as Gives access to the settings. In addition, in the case of software processing (CPU), the driver may contain various DSP (handlers). Therefore, first, when the effects are disabled and processing, if the driver does not provide accurate PCM transmission to the card, it is considered a gross mistake, a critical bug. And it happens rarely. On the other hand, the differences between the drivers can in updating the processing algorithms (resamplers, effects), although it also happens very rarely. In addition, the effects and any processing by the driver should still turn off / bypass to achieve the highest quality.

Thus, driver updates are mainly focused on improving the stability of the work and the elimination of errors associated with processing. None, nor other in our case on the quality of playback does not affect, because in 999 cases out of 1000, the driver does not affect the sound.

S: Licensed Audio CD sound better than their copies

R: If you have no errors (unreasonable) read / write errors and the optical drive of the device on which the disk-copy will be played, there are no problems with reading, then such a statement is erroneously and easily refuted.

S: Stereo coding mode gives best quality than joint stereo

R: This delusion mainly concerns LAME MP3, as all modern encoders (AAC, Vorbis, Musepack) use onlyjoint Stereo mode (and this already says something)

To begin with, it is worth mentioning that the Joint Stereo mode is successfully used with lossless compression. Its essence lies in the fact that the signal before coding is declined to the sum of the right and left channel (MID) and on their difference (SIDE), and then the same encoding of these signals occurs. The limit (for the same information in the right and left channel) obtains dual data savings. And since in most music information in the right and left channels is rather similar, this method is very effective and allows you to significantly increase the degree of compression.

In the lossy principle the same. But here in the permanent bit rate mode, the quality of fragments with similar information in two channels will increase (in the limit to double), and for VBR mode in such places it will simply decrease the bitrate (do not forget that the main task of the VBR mode is stably maintaining the specified coding quality, Using the minimum possible bit rate). Since during the Lossy encoding priority (when the bit distribution) is given by the amount of channels to avoid the deterioration of the stereopanorama, it is used to dynamically switch between Joint Stereo modes (MID / SIDE) and the usual (Left / Right) stereo on the basis of frames. By the way, the cause of this error was the imperfection of the switching algorithm in old versions of Lame, as well as the presence of a FORCED JOINT mode, in which there is no auto detection. IN recent versions LAME JOINT mode is enabled by default and it is not recommended to change it.

S: The wider spectrum, the better the record (about spectrograms, aucdtect and frequency range)

R: Nowadays, on the forums, unfortunately, the quality of the track on the spectrogram track is very common. Obviously, due to the simplicity of this method. But, as practice shows, in reality everything is much more complicated.

And the point here is what. The spectrogram visually demonstrates the frequency signal power distribution, but cannot give a complete view of the recording sound, the presence of distortion and artifacts of compression. That is, in fact, everything that can be determined by the spectrogram is the frequency range (and partly - the spectrum density in the RF region). That is, at best, by analyzing the spectrogram you can reveal the apart. Comparing the same spectrograms of tracks obtained by coding by various encoders, with the original - the complete absurdity. Yes, you can identify differences in the spectrum, but to determine whether they will (and to what extent) will be perceived by the human ear - almost impossible. We must not forget that the task of Lossy coding is to ensure the result indistinguishable human ear from the original (in no way with the eye).

The same applies to the assessment of the quality of coding by analyzing the tracks at the AUCDTECT, AUCDTect Task Manager, Tau Analyzer, FoocdTect is only a shell for a single AUCDTECT console program). The AUCDTect algorithm also actually analyzes the frequency range and only allows you to determine (with a certain share of probability), whether MPEG compression was applied on any of the encoding stages. The algorithm is sharpened by MP3, therefore it is easy to "deceive" using the codecs Vorbis, AAC and Musepack, so that even if the program writes "100% CDDA" - this does not mean that the encoded audio is 100% corresponding to the source.

And, returning directly to the spectra. Popular also the desire of some "enthusiasts" by anything to disable LowPass (LC) filter in the LAME encoder. Here on the face misunderstanding the principles of coding and psychoacoustics. First, the encoder cuts high frequencies with only one goal - to save data and use them to encode the most audible frequency range. The extended frequency range can fatally affect the overall quality of the sound and lead to audible coding artifacts. Moreover, the disconnection of the cut on 20 kHz is generally completely unnecessary, since the frequency above the person simply does not hear.

S: There is a certain "magic" equalizer preset capable of significantly improving the sound

R: This is not entirely so, first, because each individual configuration (headphones, acoustics, sound card) has its own parameters (in particular, its amplitude-frequency response). And therefore each configuration should be your unique approach. Simply put, such an equalizer preset exists, but it differs for different configurations. Its essence is to adjust the frequency response, namely, in the "alignment" of unwanted failures and bursts.

Also, among people distant from the direct work with sound, the setting of a graphic equalizer "Talk" is very popular, which actually represents the increase in the level of the LF and the RF components, but at the same time leads to the muffition of vocals and tools, the spectrum of the sound of which is located in the middle frequency area .

S: Before converting music to another format, it should "squeeze" it in WAV

R: Immediately note that under WAV implies PCM data (pulse-code modulation) in the WAVE container (file with * .wav extension). These data are nothing but the sequence of bits (zeros and units) by groups of 16, 24 or 32 (depending on the bit), each of which is the binary amplitude code of the corresponding samples (for example, for 16 bits in decimal representation These are values \u200b\u200bfrom -32768 to +32768).

So, the fact is that any sound handler - whether it is a filter or encoder - usually works only with these values, that is only With uncompressed data. This means that to convert the sound, let's say, from Flac in Ape, just necessary First decode FLAC in PCM, and then encode PCM in APE. This is how to repack files from Zip to RAR, you must first unpack Zip.

However, if you use a converter or simply advanced console encoder, an intermediate conversion in PCM occurs on the fly, sometimes even without recording to a temporary WAV file. It is this and introduces people in error - it seems that the formats are converted directly to one to another, but in fact, in such a program, there is a decoder of an input format that performs an overpriced conversion to PCM.

Thus, manual transformation in WAV will not give you absolutely nothing but excess time.

Deposit popular myths about digital sound.

2017-10-01T15: 27.

Audiophile "S Software

Note: For a better understanding of the following text, I highly recommend you to familiarize yourself with the foundations of digital sound.

Also, many of the points affected below are illuminated in my publication "Once again about sad truth: where does the good sound come from?" .

The more bitrate, the better the track

This is not always the case. To begin with, I remind you what bit of t. (Bitrate, not Bitraid). In fact, this is the rate of data flow in kilobits for a second when playing. That is, if we take the size of the track in kilobits and divide on its duration in seconds, we obtain its bitrate - t. N. File-based Bitrate (FBR), it is usually not too different from the audio bit bitrate (the reason for the difference is the presence of metadata - tags, "sewn" images, etc.).

Now take an example: the bit rate of the uncompressed PCM audio recorded on the usual Audio CD is calculated as follows: 2 (channel) × 16 (bit on each sample) × 44100 (samples per second) \u003d 1411200 (bit / s) \u003d 1411.2 kbps . Now, we will leave the track any Lossless codec ("Lossless" - "Binding", i.e., such that does not lead to a loss of any information), such as the FLAC codec. As a result, we will get a bit rate below the original, but the quality will remain unchanged - here's the first refutation.

Here something else is worth adding. The bit rate at the output with lossless compression may turn out to be the most different (but, as a rule, it is less than that of uncompressed audio), it depends on the complexity of the compressible signal, or rather from the redundancy of the data. Thus, simpler signals will be compressed better (i.e. we have a smaller file size with the same duration \u003d\u003e Little bitrate), and more complex - worse. That is why classical music in Lossless has a smaller bit rate than, say, rock. But it is necessary to emphasize that the bitrate here is by no means an indicator of the quality of the sound material.

Now let's talk about lossy compression (with losses). First of all, it is necessary to understand that there are many different coders and formats, and even within one format, the quality of coding in different coders may differ (for example, QuickTime AAC encodes a much better than the outdated FAAC), not to mention the superiority of modern formats (OGG Vorbis, AAC , Opus) Over MP3. Simply put, of the two identical tracks encoded by different encoders with one bit rate, some kind of sound better, and some kind of worse.

In addition, there is such a thing as apoplevert. That is, you can take a track in mp3 format with a bit rate of 96 kbps and convert it to MP3 320 kbps. Not only does the quality improve (after all, the data lost at the previous coding of 96 kbit / with the data is no longer returning), it will even deteriorate. It is worth indicating that at each stage of lossy encoding (with any bit rate and any encoder), a certain portion of distortion is made in the audio.

And even more. There is another nuance. If, say, the bit rate of the audio stream is 320 kbps, this does not mean that all 320 kbps went to the coding of the very second. This is characteristic of coding with a constant bitter and for those cases when a person, hoping to get the maximum quality, forces too much permanent bitrate (as an example - setting 512 Kbps CBR for Nero AAC). As is known, the number of bits highlighted on this or that frame is regulated by a psychoacoustic model. But in the case when the allocated amount is much lower than the established bit rate, it does not save even a bit reservoir (read about the terms in the article "What is CBR, ABR, VBR?") - As a result, we get useless "zero bits" that simply "finish »Frame size to the desired (i.e., increase the flow size to the specified). By the way, it is easy to check - squeeze the resulting file with an archiver (better than 7z) and look at the degree of compression - the more it is the more zero bits (since they lead to redundancy), the greater the wonderful place.

Lossy codecs (MP3 and other) are able to send with modern electronic music, but are not able to qualitatively encode the classic (academic), live, tool music

The "Irony of Fate" here is that in fact everything is exactly the opposite. As is known, academic music in the overwhelming majority of cases should be melodic and harmonic principles, as well as tool composition. From a mathematical point of view, this causes a relatively simple harmonic composition of music. So the predominance of consonances produces a smaller number of side harmonics: for example, for the quint (the interval in which the basic frequencies of two sounds differ one and a half times) the total for two sounds will be each second harmonic, for quarts, where frequencies differ by one third - each third, and etc. In addition, the presence of fixed frequency ratios due to the use of uniformly tempered system also simplifies the spectral composition of classical music. The living instrumental composition of the classics causes noise in it characteristic of electronic music, distortion, sharp amplitude jumps, as well as the absence of an excess of high-frequency components.

The factors listed above lead to the fact that classical music is much easier to compress, first of all, purely mathematically. If you remember, mathematical compression is working at the expense of redundancy (describing similar fragments of information using a smaller number of bits), as well as at the expense of the prediction (T.N. predictors The behavior of the signal is predicted, and then only the deviation of the actual signal from the predicted one is encoded - the more precisely they coincided, the less bits are needed for coding). In this case, a relatively simple spectral composition and harmony cause high redundancy, the elimination of which gives a significant degree of compression, and a small number of bursts and noise components (which are random and unpredictable signals) determines the good mathematical predictability of the overwhelming part of the information. And I am no longer talking about a relatively small average volume of classic tracks and about frequently encountered silence intervals, for the encoding of which information is practically required. As a result, we can cut without loss, for example, some solo instrumental music to bitrates below 320 kbps (TAK and OFR encoders are completely capable of such.).

So, first, the fact is that the mathematical compression underlying the lossless encoding is also one of the Lossy encoding steps (read it is clear about MP3 coding). And secondly, since the Fourier transformation is used in Lossy (decomposition of the signal for harmonics), then the simplicity of the spectral composition even doubly facilitates the code of operation. As a result, comparing the original and encoded samples of classical music in a blind dough, we are surprised to find that we cannot find any differences, even with a relatively low bitrate. And the funny thing is that when we start to completely reduce the coding bit rate, the first thing that detects the differences - background noise in the record.

As for electronic music - with it encoders accounted for very difficult: noise components have minimal redundancy, and together with sharp jumps (some kind of saws) are extremely unpredictable signals (for encoders that are "sharpened" under natural sounds, leading themselves perfectly Otherwise), the direct and reverse transformation of Fourier with the garbage of individual harmonics a psychoacoustic model inevitably gives the effects of pre- and post-echo, the audibility of which the encoder is not always easy to evaluate ... Add a high level of high-level RF components - and get a large number of killer - Semples, with which even the most advanced coders do not cope with the most advanced coders, oddly enough, precisely among electronic music.

Also amazing the opinions of "experienced hearing" and musicians who, with complete lack of understanding of the principles of Lossy coding, begin to argue that they hear how tools in music after coding begin to fake, frequencies are floating, etc. It may have been fair for the doping Cassette players with detonation, but in digital audio everything is accurate: the frequency component either remains or discarded, to shift the tonality here simply there is no need. Moreover: the presence of a musical hearing person does not mean the presence of a good frequency hearing (for example, the ability to perceive frequencies\u003e 16 kHz, which is not coming to no) and does not make it easier for it to search for lossy coding artifacts, since distortion These are characterized by very specific and require the experience of the blind comparison precisely Lossy Audio - you need to know what and where to look.

DVD-Audio sounds better than Audio CD (24 bits against 16, 96 kHz against 44.1, etc.)

Unfortunately, people usually look only on numbers and very rarely think about the effect of one or another parameter to objective quality.

Consider to start the bit. This parameter is not responsible for anything other than the dynamic range, i.e., for the difference between the most quiet and loud sounds (in dB). In digital audio, the maximum level is 0 DBFS (FS - Full Scale), and the minimum is limited to noise level, i.e., actually the dynamic range of the module is equal to the noise level. For a 16-bit audio, the dynamic range is calculated as 20 × log 10 2 16, which is 96.33 WB. In this case, the dynamic range of the symphony orchestra is up to 75 dB (mostly about 40-50 dB).

And now imagine real conditions. The noise level in the room is about 40 dB (do not forget that dB - the value is relative. In this case, the threshold of hearingness is accepted in 0 dB), the maximum volume of music reaches 110 dB (so that there is no discomfort) - we get a difference of 70 dB. Thus, it turns out that the dynamic range of more than 70 dB in this case is simply useless. That is, when the range above or loud sounds will reach a painful threshold, or quiet sounds will be absorbed by the surrounding noises. Achieve the level of surrounding noise less than 15 dB is very difficult (since there is a volume of human breathing and other noise due, human physiology), as a result, the 95 dB range for listening to music is quite sufficient.

Best of all, this topic is disclosed in the download article in Format 24/192 - why they do not make sense.

Different software players sound differently (e. G. Foobar2000 is better than Winamp, etc.)

To understand why this is not the case, we must figure out what is the program player. In essence, this is a decoder, handlers (optional), output plugin (on one of the interfaces: ASIO, DirectSound, Wasapi. Etc.), and of course the GUI (user graphical interface). T. K. The decoder in 99.9% of cases works according to a standard algorithm, and the output plugin is just part of the program that transmits the flow of a sound card through one of the interfaces, then the reason for the differences can only be handlers. But the fact is that handlers are usually turned off by default (or must be turned off, since the main thing for a good player is to be able to pass the sound in the "pristine" form). As a result, the subject of comparison can only be capabilities Processing and output in which, by the way, there is no need for a very often. But even if such a need is - then this is already a comparison of handlers, and not the players.

Different versions of the driver sound differently

Based on this statement lies a banal ignorance of the principles of the sound card. The driver is the software required for efficiently interacting the device with the operating system, also usually providing the graphical user interface to manage the device, its parameters, etc. The sound card driver provides a sound card recognition as a Windows audio device, reports OS on the supported card Formats ensures the transmission of uncompressed PCM (in most cases) the thread on the card, and also gives access to the settings. In addition, in the case of software processing (CPU), the driver may contain various DSP (handlers). Therefore, first, when the effects are disabled and processing, if the driver does not provide accurate PCM transmission to the card, it is considered a gross mistake, a critical bug. And it happens rarely. On the other hand, the differences between the drivers can be in updating the processing algorithms (resamplers, effects), although this does not happen often. In addition, to achieve the highest quality effects and any processing by the driver should still be excluded.

Licensed Audio CD sound better than their copies

If there are no errors (unreasonable) read / write errors during copying and the optical drive of the device on which the disk-copy will be played, there are no problems with its reading, then such a statement is erroneously and easily refuted.

Stereo coding mode gives better quality than Joint Stereo

This error mainly concerns LAME MP3, as all modern coders (AAC, Vorbis, Musepack) use only Joint Stereo mode (and this already says something)

In the lossy principle the same. But here in the permanent bit rate mode, the quality of fragments with similar information in two channels will increase (in the limit to double), and for VBR mode in such places it will simply decrease the bitrate (do not forget that the main task of the VBR mode is stably maintaining the specified coding quality, Using the minimum possible bit rate). Since during the Lossy encoding priority (when the bit distribution) is given by the amount of channels to avoid the deterioration of the stereopanorama, it is used to dynamically switch between Joint Stereo modes (MID / SIDE) and the usual (Left / Right) stereo on the basis of frames. By the way, the cause of this error was the imperfection of the switching algorithm in old versions of Lame, as well as the presence of a FORCED JOINT mode, in which there is no auto detection. In the latest LAME versions, the Joint mode is enabled by default and it is not recommended to change it.

The wider spectrum, the better the record (about spectrograms, aucdtect and frequency range)

In our time on the forums, unfortunately, the measurement of the quality of the track "line by spectrogram" is very common. Obviously, due to the simplicity of this method. But, as practice shows, in reality everything is much more complicated.

The same applies to the assessment of the quality of coding by analyzing tracks at the AUCDTECT, AUCDTect Task Manager, Tau Analyzer, FoocdTect is only a shell for a single AUCDTECT console program). The AUCDTect algorithm also actually analyzes the frequency range and only allows you to determine (with a certain share of probability), whether MPEG compression was applied on any of the encoding stages. The algorithm is sharpened by MP3, therefore it is easy to "deceive" using the codecs Vorbis, AAC and Musepack, so that even if the program writes "100% CDDA" - this does not mean that the encoded audio is 100% corresponding to the source.

And returning directly to the spectra. Popular also the desire of some "enthusiasts" by anything to disable LowPass (LC) filter in the LAME encoder. Here on the face misunderstanding the principles of coding and psychoacoustics. First, the encoder cuts high frequencies with only one goal - to save data and use them to encode the most audible frequency range. The extended frequency range can fatally affect the overall quality of the sound and lead to audible coding artifacts. Moreover, the disconnection of the cut on 20 kHz is generally completely unnecessary, since the frequency above the person simply does not hear.

There is a certain "magic" equalizer preset capable of significantly improving the sound

This is not entirely so, firstly, because each individual configuration (headphones, acoustics, sound card) has its own parameters (in particular, its amplitude-frequency response). And therefore each configuration should be your unique approach. Simply put, such an equalizer preset exists, but it differs for different configurations. Its essence is to adjust the frequency response, namely, in the "alignment" of unwanted failures and bursts.

Also among people are far from direct work with the sound, the setting of a graphic equalizer "check mark" is very popular, which actually represents the increase in the level of the LF and the RF components, but at the same time leads to the muffle of vocals and tools, the sound spectrum of which is located in the middle frequency area.

Before converting music to another format, it should "squeeze" it in WAV

Immediately, I note that under WAV implies PCM data (pulse-code modulation) in the Wave container (file with the * .wav extension). These data are nothing more than a sequence of bits (zeros and units) by groups of 16, 24 or 32 (depending on the bit), each of which is a binary amplitude code of the corresponding samples (for example, for 16 bits in decimal representation These are values \u200b\u200bfrom -32768 to +32768).

However, if you use a converter or simply advanced console encoder, an intermediate conversion in PCM occurs on the fly, sometimes even without recording to a temporary WAV file. That is what enters people in error: it seems that the formats are converted directly to one to another, but in fact, in such a program, there is a decoder of an input format that performs an intermediate conversion to PCM.

Thus, manual transformation in WAV will not give you absolutely nothing but excess time.