What format is flac? Can you play mp3 in flac format?


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What format is flac? Can you play mp3 in flac format?

flac file

How to play flac format?

FLAC

FLAC is short for Free Lossless Audio Codec, which can be interpreted as lossless audio compression coding in Chinese. FLAC is a well-known free audio compression codec, which is characterized by lossless compression. Unlike other lossy compression codes such as MP3 and AAC, it does not destroy any original audio data, so it can restore the sound quality of music discs. It is now compatible with many software and hardware audio products. To play flac format in MP3, you need to convert the format. The specific operation method is as follows: 1. Download the format factory class format conversion software, install it and open it. 2. Click the FLAC format on the open interface, and open the FLAC format file to convert on this page. It is equivalent to inputting the file in this format into the software. 3. On the opened page, browse and select the MP3 format to be output, and then select the sound quality effect to be output, and then click Start or OK. 4. After the conversion is complete, connect the MP3 player to the computer. 5. Copy the converted files to MP3 storage. 6. Safely eject the device, remove the MP3 player and start playing the music.

FLAC stands for Free Lossless Audio Codec – Free lossless audio compression. In short, FLAC is similar to MP3, but it has lossless compression, which means that the audio is compressed in FLAC without losing any information. This compression is similar to Zip, but FLAC will give you a higher compression ratio, because FLAC is a compression method specially designed for the characteristics of audio, and you can use the player to play FLAC compressed files like you normally play MP3 files (there are already there are many FLAC-compatible car players and home audio equipment, and you can find links to these equipment manufacturers on the FLAC website). General hi-fi players can be used, and normal MP3 generally supports MP3 and WAV formats. Common lossless formats are: FLAC, APE, TTA, TAK, ALAC APE is the most popular lossless audio, APE is featured on major resource stations and even in many people’s minds, only APE is lossless. This is thanks to the promotion of the APE encoder by Monkey’s Audio. In fact, the compression rate of APE is very good and the encoding speed is fast enough.


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Normalize FLAC – FLAC loudness normalizer

Normalize FLAC – FLAC loudness normalizer

FLAC

Normalize the volume of a FLAC file

flac

Mp4Gain is capable of normalizing the loudness of FLAC, Ogg, etc. files. And also videos.

You can normalize the volume level (volume leveler, volume enhancer) of your FLAC files.

All this with the push of a button.

What is a FLAC?

FLAC can be interpreted as lossless audio compression coding. FLAC is a well-known free audio compression codec, which is characterized by lossless compression. Unlike other lossy compression codes, such as MP3 and AAC, it does not destroy any original audio information, so it can restore the sound quality of music discs [1] . It has been compatible with many software and hardware audio products (such as CDs, etc.) since 2012.

FLAC is different from MP3 .MP3 is a lossy audio compression encoding, but FLAC is a lossless compression, which means no information will be lost after the audio is compressed with the FLAC encoding. After the file is restored FLAC to a WAV file, the contents of the WAV file before compression itself. This compression is similar to ZIP, but FLAC has a higher compression rate than ZIP and RAR because FLAC is a compression method specially designed for the characteristics of PCM audio. And you can use the player to play FLAC compressed files directly, just like MP3 files (there are many car players and home audio equipment that support FLAC, you can find links to these equipment manufacturers on the FLAC website) .

FLAC is free and is supported by most operating systems, including Windows.

Now major websites have FLAC music downloads, and publishers usually take the .cda audio track directly into .flac after buying the CD to ensure the original lossless quality of the CD.
Lossless formats work very well with good headphones.

Mp4 Normalize – Mp4 Loudness Normalization

Mp4 Normalize – Mp4 Loudness Normalization

Mp4Gain can normalize the volume and loudness of the most popular video formats.

Initially when trying to do a volume leveler or volume enhancer (volume louder), it was only possible to do it for an mp3.

It was not possible to normalize mp3 for other audio formats. Moreover, initially there were no formats such as Ogg, FLAC, etc.

For some time now, we have not only made Mp4Gain able to normalize the loudness of the most important audio formats, but we have also achieved something incredible: normalizing mp4 files.

In fact, we have managed to normalize videos of the main formats. We have also got Mp4Gain to be used as an audio and video converter.

You can take a video and convert it from one format to another and convert an audio file from one format to another. It can also extract the audio from a video file and convert it to an audio format, getting audio in any of its main formats.

Mp4Gain’s volume normalization is so advanced that not only does it make everyone sound at the same level of audio gain, but each video (or each song in the case of audio files) has volume consistency, preventing there are parts that sound louder and other parts that sound barely audible.

We recommend you download and try Mp4Gain on your windows computer so that you can enjoy the results, without a doubt it is the best and most powerful normalizer with the possibility of normalizing audio and video files.

Mp3 – Subjective perception of sound: timbre

Mp3 – Subjective perception of sound: timbre

Audio Timbre mp3

Tone is a concept that we often come into contact with in our daily lives, and it can even be said that it is an essential concept.

Mp3 timbre

In high school physics textbooks, timbre, along with pitch and volume, are considered the three basic attributes of sound. Although we often deal with the word doorbell, and doorbell has more mature applications in many respects, doorbell is actually a very vague concept. What physical phenomenon determines the timbre? Why can we only use vague words like light and shadow, thickness and warmth to describe timbre?

Pitch and volume are well understood and correspond to physical phenomena: pitch corresponds to the vibrational frequency of an object, and volume corresponds to the object’s vibrational amplitude. However, we were unable to find a physical phenomenon that corresponds to the timbre. There are also no precise words to describe the timbre in our language.

We can create great music by precisely controlling the combination of pitch, duration, and volume. So can we control timbre like we control pitch and duration?

What exactly is pitch? What physical processes are involved in the timbre?

This is something that many people have been exploring and researching for the last few hundred years. But to this day, there is still no perfect answer.

In this article, I have classified some of the research results of my predecessors and presented some of my own thoughts. I hope that it can give everyone a systematic understanding of timbre, and I hope to inspire those who are interested in studying timbre in the future.

This article will start with the definition of timbre, discuss what timbre is, the subjective perception of timbre, the main factors that affect timbre, the description of timbre in language, the application of timbre in music, and personal perspectives for timbre. doorbell investigation. This article also proposes a sound classifier model for the timbre. At the same time, there are some conjectures that I have not been able to study in the article. I will check them out in the future.

The definition of sound
There are many definitions of timbre.

The word timbre comes from the French   . In old French it referred to the sounds produced by different musical instruments.

In his music textbook “Fundamentals of Music Theory,” Li Chongguang noted that “timbres are different due to the nature, shape, and number of harmonics of the sounding body.”

The American Standards Association (ASA) defines ringing from the opposite perspective. It does not define what timbre is, but what timbre it is not: “timbre is any other difference between sounds of the same loudness, pitch, and duration.”

All the above definitions tell us what tone it is. But none of these definitions is as clearly linked to the corresponding physical laws as other physical properties.

Like pitch and frequency of vibration, like speed and time and displacement, like color and wavelength.

Why can’t the word timbre clearly correspond to a certain physical phenomenon? I think this is due to the complexity of the timbre.

How does the normalization of an mp3 work?

How does the normalization of an mp3 work?

mp3 volume normalizer

The normalization of the loudness of an mp3 is based on perception.

How MP3 Compression Works

The human ear has its peculiarities and the study of these have allowed the development of the normalization algorithm for mp3 files.

Initially what was needed was simply to reduce the space an audio file took up, while maintaining high quality.

We must remember that in the early years of the internet, it was very slow and hard drives had very little capacity. What made it impossible to download a wav file (this option is still not used today), which is the one with the “original” quality because the wav takes up a lot of space.

So the normalization algorithm of an mp3 was based on being able to understand how the human ear works to be able to discard information without sacrificing quality, based on the way we perceive music and thus, the mp3 file, which occupies much less space ( a tenth of the original) sounds almost the same as the original.

But… what information can be discarded without the human ear noticing? How is it achieved that even discarding information, an mp3 sounds almost the same as the original?

The first thing is to discard all the sounds that are not perceived by the human ear, but are nevertheless present in the original music wav file. Because the human ear can only perceive sounds in a range of frequencies, any lower frequency will not be heard and any higher frequency will not be heard either. So it’s safe to rule out silences, and also to rule out low and high frequencies that are beyond our ability to perceive.

But there are other phenomena in human hatred, such as the so-called masking. It happens that if we listen to a frequency at high volume and immediately after (or even at the same time) we listen to another instrument or sound with a similar frequency, our ear will not perceive this second sound… Then it is also possible to discard these sounds that are masked by first. Almost all frequency redundancy can be discarded without the human ear being able to perceive it.

So if we take a wav file and remove all the frequencies that are outside of our hearing range and also the masking, we will have been able to reduce the new audio considerably. If we also add a compression (like the one used for a .zip file), we will have a reduction so great that the new file will weigh one tenth of the original.

This allows the new file to be much more manageable than the original and to take up much less space on the hard drive, which was of crucial importance in those years.

This method that we are explaining in general was the one that was used two decades ago or more. Today Mp4Gain uses much more modern and complex methods, which provide more surprising results.

In addition, Mp4Gain is capable of normalizing not only mp3 files, but all popular audio formats and it can also normalize videos. And this is based on more modern, efficient and state-of-the-art algorithms, which provides a much better result.

Generally speaking, we have provided a non-technical explanation of how audio files were compressed to make an mp3.

We advise you to download Mp4Gain and check its quality for yourself. You guys are going to love it.

Loudness Normalization – Mp4Gain

Loudness Normalization – Mp4Gain

Loudness Normalization

Mp4Gain is the most advanced normalizer and practically the only loudness normalizer that is still current and updated according to new technology and new formats.

Loudness Normalization

In fact, it is the only one that handles the most popular audio and video formats. You can even extract the audio from any cvideo and save it as an audio format (mp3, acc, m4a, flac, ogg, etc).

These are the formats that Mp4Gain handles:

Video formats:

mp4, flv, avi
mpg, 3gp, wmv

Audio formats:

mp3, mp2, flac
ogg, m4a, acc
wav ac3

And very soon we will add more formats…

In fact, Mp4Gain offers other features, such as the ability to alter the pitch of a song without changing its speed and vice versa.

In fact, normalization offers a whole series of parameters that will allow you to achieve exactly what you are looking for. You can add the option to include Replay Gain, for example…

It also offers the option that, within the same song or video, all its parts have a similar sound. Avoiding, in this way, the existence in the audio or video file, parts that are very loud and others that have a very low volume.

Of course you can equalize as you like, to improve the quality of the audio.

For all these reasons, we believe that it is best that you download a copy of Mp4Gain and try it for free on your computer.

The program is very simple to use, you could practically say that it is intuitive. In other words, it is possible to use Mp4Gain without needing to read any manual, since it usually only loads the songs or videos and clicks the normalize button and that’s it.

Never again will you suffer from the situation that your different audio or video files have a very different loudness, forcing you to manipulate the volume knob to manually compensate for the difference in volume levels.

Download it today and give it a try.

Loudness Normalization: Why is it necessary to Normalize the loudness of an audio or a video?

Loudness Normalization: Why is it necessary to Normalize the loudness of an audio or a video?

Loudness

The war of volume or loudness war.

Already in the 1940s and in later decades, in the middle of the vinyl record era, a volume war was experienced.

The goal was to make a song sound louder on the radio, louder than other songs and louder than advertising.

Sure, the limitations of vinyl didn’t allow the ability to indiscriminately increase volume to be possible.

Loudness normalization

But with the advent of CDs and digital music it was possible to push the loudness of a song to the max. The situation is that the digitization of the audio allowed it to be manipulated quite precisely, achieving dynamic normalizations that actually ended the dynamics of the music and then played all the time at maximum volume.

By the 90s, groups like Red Hot Chilli Peppersm and their album Californication took this war of loudness to levels rarely seen.

But why did they do that?

Some research on human hearing showed that people did not find that a song sounded better if it had louder loudness.

Every artist, every producer, and every hardware manufacturer has figured out a way to make their production sound louder, louder.

Digitally many limiters and compressors pointed in that direction and made a lot of music sound almost to the point of distortion.

Each one wanted their music to stand out, among other things for being louder and having a greater sound, a higher volume level.

If to this recipe we add the appearance of the mp3 and a great variety of encoders, and also that ordinary people did not understand the effect that the bit rate could produce, then many mp3s with different qualities were generated.

The possibility of sharing these mp3s filled people with mp3s that each had very different sounds. Both for its production and for its coding.

Then a new need appeared: normalize the music to avoid these disparities in loudness, in the volume of the songs.

The holy grail of normalization had to be found.

Many ideas were found, many experiments. The situation matured and certain products like Mp3Doctor and Mp4Gain matured to the point where they actually managed to find the solution: a dynamic standardization that will work well with today’s advanced player equipment.

Then Mp4Gain made the leap, achieving that even videos could not be normalized.

Audio could already be normalized in its main formats (mp34, aac, ogg, floac, etc) with Mp3Doctor, but Mp4Gain added the possibility of these dynamic normalization to video in its main formats (mp4, 3gp, flv, avi, etc. )

What exactly is normalizing?

Music is distinguished by what is often called “dynamic” and which refers to the changes (more or less abrupt) of the “effort” with which certain notes or passages are interpreted.
Whether it is an instrument or the voice.

singer

Any vocal performance that has been considered virtuous, in general terms, will have a dynamic that goes from very soft passages, almost whispered, to intense passages, with a high volume, singing at full voice.

At the time when vinyl existed as the option to listen to music, it was not felt (at least it went almost unnoticed) the fact of noticeable differences between the loudness or the volume of a song.

It was with the advent of digitization and the possibility of its variants (opting for different bitrates, sample rates, bitdepths, etc.) that this difference became very evident.

And with the appearance of mp3 and its distribution or exchange, at the same time that winamp and distribution lists arose, when it was inevitable and it was even started to look for solutions.

Napster

These first ones were based on the sound peaks and their results were very inefficient.

Returning to the mention of the mp3, situations such as masking (where information is removed) further marked the problem of differences in volume.

Then began to use the RMS that rather mediates the average power that the song had, more than the peaks.

Initially, it was enough to put a slower reaction level to the volume meter, to have a more general idea and less impacted by the volume peaks.

And so, the way of listening to music and considering what normalization was evolved.

Finally it appears to be somewhat closer to a mixture of a volume limiter and a compressor.

What is a volume limiter? It is a hardware (although lately there are also limiters in software version) that ensures that no peak exceeds a maximum limit.

A compressor, on the other hand, is a device or software that is used to “compact” the volume, preventing the parts with the lowest volume from being too low and at the same time preventing the high parts from exceeding a range that has been assigned. We would say that the compressor dampens the increases and decreases in volume.

To this we can add an equalization that differentiates the bands and treats them differently both in the limitation and in the compression. Each frequency band has a different treatment in the Mp4gain and that produces a very efficient result. It is NOT the only improvement offered by Mp4Gain, but this is described here. In other articles we will deal with other differences.

Mp4Gain is the best normalizer of 2020 and this is clear when using it.

Audio Level normalization

The audio levels of the material produced in a radio station
In general, in radio they do not tend to stay within standardized levels for their audio editions (spots), it is not necessary to know much about levels, since an audio processor compresses and limits everything on air.

Radio Studio Compressor

The console operator does not understand anything about dynamic range, something that has no practical use in the air. And this is how many radios work with adjustments that “work” in the air by trial and error, and not always with the most demanding criteria. successful.

Dynamic range compression

Level normalization

In radio, an editor does not know or manage any level convention, so it could be said that level normalization is not widely used. However, a good professional practice would be that all the material generated by a station “sounds” at the same level. Not to the air, because to the air if it is transmitted normalized or compressed and limited, but inside the station. And for this, there are two ways:

The material is processed “by ear” by comparison.
An RMS value is defined and all publishers normalize their mixes to that average level.

Regarding the first point, differences of up to +/- 2 dB will be absolutely acceptable. But a very common vice is to overcompress the edits, or sometimes the voices, seeking to hear the compact and aggressive sound of the FM on studio monitoring. That sound should be determined on-air by the streaming processor, not the publisher. Editors generally abuse processes like Normalize RMS (Sound Forge) and “maximizers”; Wave Hammer (Sound Forge / Vegas) Ultramaximizer and L1 (Waves). Ideally, how much to “squeeze” the dynamics of the edited material should be a function of the type of processor the radio has. At this point it is possible to clarify a fairly common confusion: STANDARDIZATION has nothing to do with making an audio sound “strong” or “powerful”. Using normalization for that purpose is a beginner’s mistake.

The second option is the most accurate way of working -although this precision is not necessary- normalizing all the editions to a given RMS value. This does not impact the sound in the air but it does the internal prolixity of the station. RMS is not an accurate measurement of loudness or “volume”, but for what you need in radio it is enough.

The streaming audio processor knows nothing about the level of the audio file. The processor receives an audio level from the console and works accordingly. What affects the behavior of the processor is the dynamics of the material, if it has dynamics or is super-compressed / limited.

Normal working values

The level at which operator-editors generate material has two well-defined extremes to avoid: very high levels of compression / cliping and excessively low material (less than 24 dB RMS). When we talk about level, we must be clear about the differences between peak level and average level.

PEAK level

Regarding the peak level, the logical maximum limit is digital cliping. Needless to say, a cliping mix is ​​unacceptable.
It is advisable that the maximum peak level is not 0 dBfs, as this will generate overshoot cliping in the D / A converters and especially if the compressed material (MP3) is exported.
An appropriate value for the material on a radio is maximum peak – 1dBfs (the recommendation if using mp3 compression is -3 dBfs). But this does not mean that it should be -1 dB. If no peak reaches the established maximum it is not a problem as long as the material complies with the appropriate working level. The peak level does not matter, but in general the signal will always reach the maximum peak level.

Listening level (RMS)

The “listening level” or mix level is determined by the RMS or “average” value of the material. This is true even if the publisher has never measured the RMS value of their audios. In general the radio editor “compresses”, “maximizes” or -conception error by- “normalizes” your edits “so that they sound”. And in that “so that they sound”, it is taking the cuts to a certain value.

The question that arises is what should that value be? How much should the final mix “squeeze”? The final value should not be a value that generates excessive compression, as this is the task of the transmission processor. How to compress is a topic of discussion for another article, since it is fine spinning and the radios in general do not take into account these aspects. In general lines we will say:

If the radio has a simple analog processor, type M31 or Solidyne 362, they will perform better with material that has a more compact sound (more compression).
If the station has a high-end digital processor, and especially if it works with a highly processed sound in the air, it is not recommended or necessary to excessively maximize the material generated by the station, because these audio equipment respond better when the material is origin is not over compressed.

 

But what if the file level is very low? It depends. Depending on the PC-Console connection, the operator typically has at least 15 dB of gain range for level correction from the PC. In turn, if the level is low with the fader on, the AGC of the processor has between 10 and 20 dB more correction to compensate the level in the air. But if the file were generated too low, it could fall outside the operator / processor correction range and go low on air.

GENERAL AND ELEMENTARY CONCLUSIONS:

Different materials generated in the radio must sound at the same level, either by ear or measured RMS.
It should not be overcompressed, much less cliping.
The peak level should not exceed -1 dB.
It should not be too low as it may fall outside the processor’s AGC / operator correction ranges.

Put in values:

RMS values ​​between -16 to -13 dB RMS are acceptable.
Values ​​between -13 and -10 dB RMS generally indicate strong compression.
Values ​​less than -10 dB RMS indicate excessive compression, not recommended as it generates a very loud but “muffled” sound that cannot be “improved” by the air processor.

Normalization of an audio file.

Normalization of an audio file.

Normalization is used to increase or decrease the level of the song as a whole, so that its maximum volume peaks assume the indicated level.

Loudness Normalization

For example, if the maximum intensity points of the song are -3 dB (therefore well below 0, which should represent the maximum before distortion), normalizing to 0 dB means increasing the level of the entire song so that these peaks reach 0 dB.

This is the typical normalization of the peaks.

There is also RMS normalization (which takes into account not the peaks but the actual average level of the song).

Audio Normalization

AUDIO CDs, which have good dynamic possibilities (various intensity tones, from pianissimo to fortissimo), are generally recorded so that the maximum volume points are at 0 dB.

Normalizing your WAV recordings can be helpful in adjusting them to the average level of a CD in case they are too low (because you had been careful in level during recording) but one important thing to note:

Normalization of this type alters the original dynamics, that is, the reciprocal relationships between weak and strong sounds.

Although all levels are raised by the same amount, the relationship between 2 levels changes (small mathematical example:
2/5 = 0.4 ma (2 + 1) / (5 + 1) = 0.5 …

The result is that the weaker sounds, after abrupt normalization, sound much louder and those that were already playing only sound a little louder … altering the dynamic relationships that had been envisioned by those who originally recorded the music and making the sound output to lose depth.

Some types of music, generally already deficient dynamics (rock, metal, etc.) since the excursions between the minimum and maximum volume are almost never very consistent, are more “normalizable” without problems, while the genres in which there may be Large Dynamic excursions (classical music or music with passages from pianissimi to fortissimi) are more problematic.

In addition, it is necessary to take into account that if you normalize a large wav file that contains many songs (not yet divided) there can still be, even in genres with little dynamics, substantial differences, in this case between one song and another and not between different points of the same song.

So a light normalization can do and is actually used (to raise the level of the part), but it would be better to make sure you don’t need it (recording from the beginning with a good level) or at least not have too much. remember, however, that the dynamics are somewhat flattened.

Normalize with Mp4Gain

This software is capable (it is the only one that can do this) of normalizing the main audio and video formats and its standardization algorithm is by far the most efficient and the one that produces the best results.
For this reason it is used by musicians, radio broadcasters, universities, television stations, producers, etc.