This question could have two different approaches:
1.- An existing mp3 has a certain quality and one could wonder what determined that quality
2.- We are going to encode an mp3 from a video from which we will extract the audio and how will we get that new mp3 to have the highest possible audio quality?
In this case we will focus on option 2, how to create a video from original audio and make it have good audio quality.
The first step, inevitably, comes from getting original audio that is of high quality. If we do not start from that point, we will no longer achieve our goal.
The quality of the audio is very difficult to “restore” if you do not have as a starting point an original audio that will have a high quality.
But even if you do have that high quality original source, it’s important against using advanced software like Mp4Gain and taking the right steps to make sure you don’t lose that quality by making a bad decision when extracting and encoding the audio a long time ago. mp3, a flac, etc.
In order to convert or extract an audio from a video, it is important to understand what points are decisive:
-Bit rate. No, to save space, we will use a bitrate lower than 192 and if possible we will use 320.
Nor does it mean that if we have a 128 mp3 and convert it to 320 we will be able to restore its quality.
What it means is that if we originally convert to a 320 mp3, we will discard very little information and thus we will get a very high quality mp3. Virtually indistinguishable from a CD.
-The following will be to use a samplerate of 48000 or at least 44100, not a smaller one.
In order to capture the full range audible by the human ear, it is necessary to use at least 44100.
Mp4Gain is the right normalizer/converter for the job and to give you a high quality output.
Everything we talk about mp3 in this article actually applies to other formats like FLACC, ogg, etc.
But how do you make it sound as good as possible?
In the first place we start from a source file, which is the one that you have obtained. The more quality it has, the better.
And besto wants to say that it must have a good birate and a good samplerate, that it be stereo and that it sound as good as possible.
If we start from that, we will be able to normalize the volume and give a consistent quality with Mp4Gain.
Because we will be able to use a high quality bitrate, an adequate samplerate, make the “color” of the audio clear, bright, but with powerful and defined bass, etc.
Mp4Gain can extract audio tracks from videos. Nowadays it is one of the best options since it is relatively easy to obtain good audio with the music we are looking for and Mp4Gain can extract the audio and convert it into an mp3, a FLAC, etc.
Mp4Gain is the best prepared normalizer for these times where every time we have devices with better hardware for music and video playback quality.
Because we are living in a time where devices and computers are rapidly improving their capacity for audio and video and only Mp4Gain is prepared for it.
Today, with Mp4Gain you have the perfect tool to get your audio and video with the best quality and volume.
Which is better, an mp3 or a lossless music file like FLAC?
lossless maudio quality mp3 flac
Lossless vs lossy
lossless maudio quality mp3 flac
The evolution of devices or equipment that reproduce audio, as well as that of headphones or speakers, have made people wonder more and more if the time has come to resort to lossless music, for example FLAC.
Before, a few years ago, it was a bit absurd to talk about lossless formats when we used equipment that had low audio capabilities on a daily basis.
But now, as many phones and computers contain DACS that have high capacity, one can already think about it without it being nonsense.
And it is not so obvious nor is the question trivial.
For one, there is a lot of bloof in this area. Where many people repeat slogans that take for granted the automatic superiority of lossless. On the other hand, we are entering a stage where true high audio quality is being achieved, including the audio that accompanies videos.
Mp4Gain, it should be noted, is perfectly compatible with this world of high quality audio and video (including video audio).
And Mp4Gain can not only normalize the volume of audio and videos perfectly, but can easily convert to lossless formats with great ease and great results.
It may be worthwhile to try and convert some files to formats such as FLAC and, incidentally, normalizing the volume and listening to them on our computer or device.
Undoubtedly sooner rather than later we will find ourselves fully in the world of high fidelity audio, if not you are already there. It is quite possible that your phone or your computer can already handle this high level of audio perfectly.
What do the bits, bit rate and sample rate of an audio file mean?
bits, bit rate and sample rate
For example, the common mp3 format audio source
bits, bit rate and sample rate
In order to store a continuous physical signal (well, tell me about Planck’s constant…) in a computer, it must be converted to a digital signal. In acoustics, a digital signal is a digital representation of the amplitude of the sound wave at any moment.
Sound waves are longitudinal waves, which are difficult to draw. The following figure is replaced by transverse waves (the concept of longitudinal waves is the phenomenon that the density of air or other media changes regularly due to energy. The peaks represent high density, the troughs represent low density, and the horizontal line is the average density, i.e. silent state)
Using high school physics, waves contain two dimensions, one is intensity and the other is time. “Number of digits” indicates how many levels sound waves are divided into from the strongest to the weakest; “Sampling Rate” determines the precision of the time axis or the sampling density, that is, the length of time represented by each red dot, and the code rate is one second The number of dots on the clock, multiplied by the space that each point occupies.
So the so-called 24 bits consist of dividing the intensity of the sound wave by 2 at power level 24, occupying 3 bytes of space. Obviously, the finer the grade, the more details are restored.
The sample rate is generally 44100 Hz for CD (Hertz = times/second), 48000 Hz for DVD, and 96000 Hz as standard. As with the number of digits, the more points you get in a single second, the more details you retrieve. Why does CD take this value? Because the hearing range of the human ear is generally believed to be between 20 and 20,000 Hz. A peak and a trough need to be represented, and at least two sampling points are required. Therefore, the CD can represent the sound of 22050 Hz at most, but this sound does not have any detail, because if there are only two peak and valley points, the average waveform is completely lost. Therefore, there will be a higher sampling rate.
If it’s in a lossless uncompressed format, the bit rate is strictly equal to the number of bits * sample rate * number of channels. And typically, the MP3 bitrate you can see just represents how much capacity the format needs to describe this one second of audio.
MP3 is lossy compression. In the compression process, some information is lost, but the lost information cannot be represented by the number of bits and the sampling rate. Generally, the higher the code rate, the less information is lost. Mathematically, bitrate and sound quality are proportional. As for whether you can hear it or not, it depends on many factors. The MP3 algorithm is not complicated, of course, to understand it you have to learn what the Fourier transform is.
There is also lossless compression (representing APE, FLAC, etc.), which also has a bitrate, and this bitrate has nothing to do with sound quality. It also describes how much capacity the file uses to describe one second of audio content, but the same audio content can be compressed to different sizes (compression ratios), similar to zip compression ratios. No matter how big you compress it, in the end it can be restored to the same file. So if you see someone looking for a lossless bitrate, you can basically conclude that the product is a bad pen.
Does MP3 quality depend on how much KBPS is the bitrate?
MP3 quality
KBPS = fast bitrate, the read speed must be to play this file smoothly,
because mp3, a common streaming format on the internet, can be downloaded while listening.
MP3 quality
If the download speed is slower than the playback speed, it will stop. (LAG), and the bit rate
refers to the minimum required download speed, but since the lower the required download speed,
the higher the compression required, and MP3 is a destructive compression format, so the bitrate
also
will affect the quality of the file. Bitrate is not the biggest influencer on overall sound quality, but the main influencing factors are sample rate and bit depth. The
sample rate refers to the number of times your computer records the sound per unit of time. Usually,
the sample rate used for a CD is 44100MHz, so
you can get good quality by setting the file to this, but remember that the bitrate should be set to 96KBPS or higher.
Reduce distortion.
Normalize the volume and loudness of an mp3 or a video easily
Normalize the volume and loudness of an mp3 or a video easily
It’s absolutely easy if we use Mp4Gain, it only takes one click of a button and all audio and video files are volume normalized.
Normalize the volume and loudness of an mp3 or a video easily
Today we find many problems with this volume issue because they are compressed by different compressors and above all using different bitrate and sample rate settings.
People don’t realize how important this whole issue is, but Mp4Gain solves it automatically. Not only through bitrate and samplerate, but also by making a deep analysis of each frame and optimizing each frequency band, so that the result is magnificent.
The largest number of inquiries we receive by email refer to that difference in volume levels in the mp3s and also between the mp4s.
And what we have been able to corroborate is that, to a large extent, many are due to having been encoded with wrong settings, for example a very low bitrate.
Because the bitrate implies the amount of information or detail that the audio or video can pass per second and this translates into the detail that a video has, for example. Which immediately affects the quality of the aforementioned video.
Mp4Gain is the solution to normalization problems.
Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless – Part 2
sample rate, bit depth, bit rate and lossless
Bit rate kbps (kp/s)
sample rate, bit depth, bit rate and lossless
In lossless uncompressed formats (such as .wav), bit rate = sample rate x bit depth x number of channels. In lossy compression (for example, .mp3), the bitrate does not equal this formula, because the original information has been destroyed. The bitrate describes the amount of information about the audio in one second, so the total size of the sound file is the bitrate x the total duration. The bit rate is also called the bit rate and the unit is the bit rate (bps, bit per second). Usually 128kbps and 320kbps are bit rates when listening to songs, of which 320kbps is the highest bit rate of mp3 format. But compared to wav file with 44.1 kHz sample rate and 16 bit bit depth (calculate two channel bit rate is 44.1 x 16 x 2 = 1411.2 kbps), it is far from the same. After compression, the bit rate has changed. Bitrate in lossless compression has nothing to do with sound quality, and bitrate in lossy compression is positively correlated with sound quality.
lossless compression
Lossless compression refers to compression (conversion) between formats without loss. Regardless of the format that is compressed (converted), the sound quality is the same and can be restored to the same original file. Lossless generally refers to lossless compression, and there is no such thing as lossless code rate. The compression of various formats corresponds to an algorithm (or encoding), and a decoder is required to decode during playback, and different decoders can also affect the integrity of the decompressed file. Common lossless formats are:
wav – A Microsoft sound file format, which is the closest uncompressed format to real sound (followed by midi), supporting multiple sample rates and multiple quantization precisions. All lossless formats are essentially wav compression, which is converted back to wav when played.
flac: Free Lossless Audio Coded, which is an international general format, characterized by high compression ratio and mature encoding algorithm. When the flac file is damaged, it can still be played normally. Furthermore, this format is also the first lossless format widely supported by hardware.
monkey: The file format converted from CD ripping using Monkey’s audio software, but the advantage is not prominent and decoding is slow.
wma-lossless: It is also produced by Microsoft. It is characterized by a high compression ratio, but it has not become mainstream.
aiff: Produced by Apple, it is the standard audio format on Apple computers.
DSD: I don’t know much about Sony Dafa and I can’t appreciate the spicy culture.
lossy compression
Lossy compression refers to the loss of sound information during the compression process, and the lost sound cannot be represented by the sample rate and number of bits. But the feature is that the compressed file becomes very small and is often used in streaming media. Common lossy formats are:
mp3: A complex algorithm developed to simulate human hearing, known as a “psychoacoustic model”. It improves the compression ratio, lowers the bit rate, and reduces the footprint by extracting some frequency bands in the audio, but at the same time, the details of the sound, such as the emotion of the human voice, the reverberation in the later stage, etc., have been deformed. It is also difficult to distinguish wav and mp3 quickly if you listen blindly and need to use equipment. MP3 is currently the most popular audio compression format, which can best preserve the sound quality before compression.
wma: Microsoft’s masterpiece, characterized by lower bitrate (such as 64kbps), wma can get smaller volume under the same sound quality conditions as mp3. And at ultra-low bit rates (like 16 kbps), wma sound quality is much better than mp3.
aac: The storage format for sound files on Apple computers.
ogg – Completely free, open, and patent-free, but less popular.
Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless
sample rate, bit depth, bit rate and lossless
HZ sampling rate
sample rate, bit depth, bit rate and lossless
The sound from the outside world is an analog signal, which is converted to a digital signal represented by 0 and 1 in the digital device and then stored. Digital signals are discrete, so sampling rate refers to the number of samples per second. The higher the sample rate, the more realistic the restored sound will be. Since the hearing range of the human ear is 20 Hz to 20 kHz, according to Shannon’s sampling theorem (also called Nyquist’s sampling theorem), in theory, audio formats with a sampling frequency greater than 40 kHz they can be called lossless formats. However, the sound obtained at the 40 kHz sampling rate does not have any detail and all frequencies are only sampled with a peak and a valley. The sample rate of general professional equipment is 44.1 kHz. 44.1 kHz is the lowest sample rate in professional audio, also known as “CD-quality sound” (22.05 kHz sample rate is broadcast-quality sound). There are 96kHz, 192kHz, etc., more detailed of course, hearing the details at these higher sample rates is ear and equipment dependent.
bit depth
To reproduce sound as accurately as possible, a high sample rate is not enough. Describes a sample point, the horizontal axis (time) represents the sample rate and the vertical axis (amplitude) represents the bit depth. 16bit means that 16 bits (2 bytes) are used to represent the level of the sample point (in general, it is proportional to volume) The degree of precision that can be achieved when encoding, i.e. the vertical axis is divided into 16 parts Describe the level, such as -3dB and -3.1415926dB accuracy difference. Similarly, there are 20 bits and 24 bits. 16-bit is considered to be the lowest bit depth standard in the field of professional audio and, like the 44.1 kHz sample rate, is the common standard for consumer and professional audio products. Bit depth is also directly related to the size of the signal-to-noise ratio, which directly affects the overall dynamic range of the recorded signal.
We know that there are two types of compression, called lossy, which discards information to make the file smaller, and lossless, which simply uses zip-type compression to reduce the size.
The quality of the audio depends on the number of bits that can be transmitted per second.
Let’s understand this concept well, it’s simple.
If the audio file has a certain amount of information and we can transmit ALL of it, then we will obtain a very high quality and there will be no loss of information.
On the other hand, if we must discard more data because we can only transmit a small amount, quality will necessarily be lost.
For this reason, the amount of information collected by the sample rate and the number of bits that can be transmitted per second go hand in hand. It will be useless to have a very good samplerate if the bitrate is low and forces us to transmit only a small portion of the abundant data available.
Many times when encoding a file, of any format, information is lost either by using a poor encoder or by mutilating the wrong settings.
Mp4Gain is highly efficient at all of this and uses the best settings for you to get the best quality, both when encoding and converting.
Today’s mp3 market can be described as a hundred flowers in bloom, and the competition between the major manufacturers is splendid. In order to take the lead and gain the most market share, various manufacturers, especially those with strength, have their own unique tricks. In terms of appearance, mpio’s fl200 has won the award with its mini round cake design. unique pendant. The latest mp3 title, Truly’s mp379, also created the trend of big screen mp3, and Samsung also released its sports mp3 concept, which is absolutely stunning in the market. Some small manufacturers have also adopted imitation methods, making the market of all kinds of mp3 like a sky full of countless stars. Simple repeat, record and fm functions can also meet the needs of the market. Line-in, features, e-book reading, gaming, and colorful backlights are slowly creeping into the new mp3 design. However, the author believes that no matter how cool it looks and how perfect its functions are, the mp3 is used to enjoy music in the final analysis. An mp3 without good sound quality is at least not a qualified mp3. If you just go for looks and function (actually, we rarely use some functions), and ignore its sound quality performance, you will feel a bit like buying a scorpion.
A decoder chip inherent in mp3
The decoder chip used by the Mp3 itself is the key to its sound quality. The sound quality displayed by high quality decoder chips is unmatched by those of the poorest. Friends who have listened to the famous iriver series of mp3s will know that its sound quality characteristics are very obvious, the bass is strong and powerful, the vocals are restored to truth, the high-frequency field is wide, and the increase is enough, which is very pleasing to the ears. This series mp3 adopts Philips SAA7750, the most advanced decoder chip in the market, and its quality and performance are excellent. The major Korean manufacturer MPIPO (Dewei Technology) also uses this decoder chip. The reputation and market feedback of these two mp3 sound quality brands are very good, and they are highly praised by the industry and outside the industry. It proves that mp3 sound quality is better than md’s ace gun. Due to the relatively high cost of the chip and the control chip, it is rarely used except for some brands. The cost is high, and the price is of course expensive. This is also the threshold that restricts many mp3 lovers from enjoying the beautiful sound quality.
