Tag: mp2 normalize

CBR and VBR What are they and what is the difference?


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Both acronyms correspond to two coding modes used for audio and video and their meaning is as follows:

CBR (Constant Bit Rate): Constant bit rate.
VBR (Variable Bit Rate): Variable bit rate.
Constant bit rate
In CBR mode, the bit rate per second that will be used in the coding process is set numerically and this will be maintained constantly for the entire duration of the audio or video clip.

Variable bit rate

When we use VBR, an average of the bit rate per second that will be used in the coding process is established numerically and this, according to analysis of the characteristics of each image frame, varies decreasing and increasing according to the information needs that occur during the audio or video clip.

Which of the two is recommended to use?
The use of one method or another depends fundamentally on two factors that cannot be analyzed separately since they are co-dependent:

The intended quality
available capacity

Let’s say we are going to make a video compilation on a double layer DVD with the capacity to store 8.5 GB. The video clips are in HD (720p) and although the figures that will be used for the example cannot be precise because they depend on the type of compression used, we will assume that in total, putting together all the clips we add 10 minutes.

The result of the compilation made in VBR to the standard commonly used for this quality (6-8 Mbit / s), would only be occupying 0.7GB of the total capacity of the disk, then then, according to our capacity budget, we can still increase the bit rate to increase the amount of information and consequently the image quality.

In this specific case, we could use the CBR mode to the maximum quality that the software / hardware that we are using allows us to increase and increase the bit rate for example to 9 Mbit / s, thus maintaining a constant good quality at all times of the film without any risk that the disc is not enough to record the total 10 minutes.

Returning to the example, suppose now that instead of 10 minutes, our clips total 90 minutes. Beforehand, we know that the 8.5GB disk will not be enough to hold that amount of information at constant maximum quality and that is when we use the VBR mode to compile.

Modality of one and two passes

The VBR mode can be configured in one or two pass mode and this refers to the fact that if we choose 1 pass, each image frame will be analyzed in fractions of a second (on the fly) and according to the information obtained, the rate of bits to apply during a certain number of frames in the sequence. This method encodes more quickly but sometimes, you get to notice the variations in image quality because in some way, the program tries to “guess” the behavior of the pixels during the following frames and when it varies unexpectedly in a cut of scene, sudden color variations or an increase in the action of the image, the bit rate applied is lower than required.

In the 2-pass mode, the first one dedicated exclusively to image analysis, then the software makes a budget and applies during the second pass the bit rate variation with much better result and virtually imperceptible quality transitions. When the scenes are relatively stable and static, the bit rate decreases and when variations in the intensity of brightness, colors or the action on the screen intensify, the bit rate increases. In this way, the coding program makes an optimal distribution by subtracting information where it is not necessary and adding it where the image requires it to finally be able to make the highest quality compilation in less capacity.


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Explanation of advanced mp3 conversion settings

 

In this article we are going to address the audio coding settings that affect the sound quality. Understanding how conversion settings work can help you select the optimal sound coding properties in terms of file size relative to sound quality.

What is the bit rate?

The bit rate is the amount of data consumed to transmit the audio sequence per unit of time. For example, a bit rate of 128 kbps (kilobits per second) means that a second sound is encoded with 128,000 bits (1 byte = 8 bits). If you convert this into kilobytes, a second of sound occupies about 16 KB.

Therefore, the higher the bit rate of a track, the more space it will occupy on the computer. However, with the same format, a higher bit rate allows you to record the best quality sound. For example, if you convert an audio CD to MP3, the 256 kbps bit rate will provide much better sound quality than the 64 kbps bit rate.

Because today’s hard disk space is relatively cheap, it is recommended to convert to MP3 with a bit rate of at least 192 kbps or higher.

The bit rate can also be classified as constant or variable.

The difference between constant bit rate (CBR) and variable bit rate (VBR)

The constant bit rate means that the encoding of each audio segment consumes a constant amount of bits. However, the structure of the sound may be different, and the coding of a segment of silence requires much less bits than the coding of a segment of intense sound. Unlike the constant bit rate, the variable bit rate adjusts the quality of the coding at various intervals. Thus, intervals that are simple in terms of coding will use a lower bit rate, while more complex intervals will be coded with a higher bit rate. The use of a variable bit rate allows for better sound quality without increasing the file size.

What is the sampling frequency?

This term is used in the conversion of analog signal to digital form and defines the number of samples (signal level sample measurements) per second needed to convert a signal.

CBR vs VBR – which one to choose?

When you are going to pass a music CD to MP3 or AAC format you will have seen two different encoding options, the CBR and the VBR. Do you know the diference?

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CBR (Constant bitrate) encoding

CBR is a type of encoding in which a fixed bit rate is always used, so if we encode a song at 192 Kbps, the resulting file will have a bitrate of 192 Kbps for the entire duration of the song.

It is the speed at which data is processed or transferred.
It is usually measured in seconds and the most common units are:
Kb / s or Kbps (remember that the lower case “b” is bits, not bytes).
Mb / s or Mbps.
Also called: bitrate, bit-rate and BR.
The main advantage of using CBR is that the coding is a bit faster (compared to VBR). However, the resulting files are not as well optimized in size and quality.

 

CBR coding also has another advantage and we know in advance the transfer rate we need. For example, if we set a bitrate of 300 Kbps, we already know that with a 320 Kbps connection we will be able to transmit the data without suffering cuts, so it is usually used in real-time transmissions or streaming.

VBR encoding (bitrate variable)

VBR is an encoding method that allows a variable bit rate, this means that the bitrate of an audio file can increase or decrease dynamically depending on the complexity of the sound.

If the music is very simple or there is silence for a few seconds the bitrate can go down and then go back up in the more complex areas of a song.

What is bitrate? Bitrate video, audio, internet and more

HomeAudio Y VideoWhat is Bitrate? Bitrate video, audio, internet and more …
What is bitrate? Bitrate video, audio, internet and more …

Surely we have heard the word bitrate countless times when an expert user refers to a video or audio in digital format, and we have come to know that it is the element that defines the flow of data. But what exactly is bitrate? The question arises because not only in these fields is this parameter used.

Like the resolution and the final format of the digital video or audio, another determining factor to obtain excellent quality in an image or sound is, without a doubt, bitrate, a parameter that perhaps is not always taken into account and that not only applies to the field of audio or video. That is why in this article we will find a lot of information to perfectly understand what bitrate is.

Bitrate: Why it is so important in our digital life

Electronic devices have reached unthinkable operating speeds just a few years ago, and that is why today we hope that our device, be it a smartphone or a tablet, a computer or a hard disk, will respond to us at the moment and without hesitation. In this they have to see many and varied factors, but one of the most important is the bit rate at which it can exchange or process information.

The term bit rate, used in computing and telecommunications systems, basically refers to the amount of bits that can be transmitted in a given unit of time through a transmission system or between two digital devices. Depending on the context in which the term is used, the bit rate, or bitrate in English, is measured in Kbit / s or Mbps, kilobits per second or megabits per second, respectively.

Regardless of the unit of measurement for defining bitrate, higher numbers always mean better and higher quality values, although we must not forget that low bit rate values ​​can also mean less signal processing by the hardware, very convenient in equipment such as smartphones, tablets or netbooks.

Bit rate on the Internet

In the case of the bit rate applicable to the Internet, the higher bit rate is better, since the content we receive from the network arrives faster. In other words, the higher the bitrate we get from our ISP, the better the connection and we can work much more comfortably.

A higher bitrate in an Internet connection means streaming movies and video in high definition, playing online with no delay and downloading really large files without problems and in a few seconds.

In the event that we want to know exactly what the bitrate of our connection is, we can do so easily and comfortably by accessing with our browser a site that is responsible for performing this test. One of the best in the market is speedtest.net.

Bit rate in audio and video

If we talk about audio and video, the meaning of the term bit rate differs a bit from what we use for the Internet. In this context, the bit rate refers to the amount of data stored for every second of data that they reproduce. To take an example, an MP3 file of a 320 kbps song offers a much higher quality than the same 128 kbps encoded file, obviously as long as both files have been created from the same source.

But we must always remember that if the source from which we obtained the files was of poor quality, then the copy will also be of poor quality, it has been encoded at 128 kbps or 320 kbps.

This also happens with videos, a much higher bit rate will offer a much better viewing quality than a video with the same resolution but at a lower bit rate.

The bit rate could be expected to increase each time the resolution grows as a larger amount of data is being processed. This means that while high bitrate rates can offer excellent display quality, they also require much more effort to process part of the hardware, forcing it, especially in modest and older hardware, to produce pauses and cuts.

Another aspect that we must also take into account since it is very important, is that video file formats use different sets of compression algorithms, which could also offer high quality with a more discrete bit rate. However, the extra process load for these types of videos can also complicate the processor and the systems involved in decoding.

Digital audio

 

Digital audio is the representation of sound signals through a set of binary data. A complete digital audio system usually begins with a transceiver (microphone) that converts the pressure wave that represents the sound to an analog electrical signal.

This analog signal goes through an analog signal processing system, in which limitations in frequency, equalization, amplification and other processes such as compaction can be performed. The equalization aims to counteract the particular frequency response of the transceiver used so that the analog signal closely resembles the original audio signal.

After analog processing the signal is sampled, quantified and encoded. Sampling takes a discrete number of analog signal values ​​per second (sampling rate) and quantification assigns discrete analog values ​​to those samples, which means a loss of information (the signal is no longer the same as the original). The coding assigns a sequence of bits to each discrete analog value. The length of the bit sequence is a function of the number of analog levels used in the quantization. The sampling rate and the number of bits per sample are two of the fundamental parameters to choose when you want to digitally process a certain audio signal.

The digital audio formats try to represent that set of digital samples (or a modification) of them efficiently, so that it is optimized depending on the application, either the volume of the data to be stored or the processing capacity necessary to obtain the starting samples. In this sense there is a very widespread audio format that is not considered digital audio: the MIDI format. MIDI does not start from digital samples of sound, but stores the musical description of the sound, being a representation of the score of the same.

The digital audio system usually ends the reverse process to that described. The set of samples they represent are obtained from the stored digital representation. These samples go through a digital-analog conversion process providing an analog signal that after a processing (filtering, amplification, equalization, etc.) affects the output transceiver (speaker) that converts the electrical signal to a pressure wave that represents Sound.

Digital audio quality

The quality of the digital audio depends strongly on the parameters with which that sound signal has been acquired, but they are not the only important parameters for determining the quality.

One way to estimate the quality of digital sound is to analyze the signal difference between the original sound and the sound reproduced from its digital representation. According to this strategy we can talk about a specific signal to noise ratio. For audio systems that perform lossless digital compressions, this measure will be determined by the number of bits per sample and the sampling rate.

The number of bits per sample determines a number of quantification levels and these a signal-to-noise ratio of carrier peak that depends quadratically on the number of bits per sample in the case of uniform quantification. The sampling rate establishes a higher level for the spectral components that can be represented, and linear distortion may appear in the output signal and aliasing (or spectral overlap) if the signal filtering is not adequate.

For digital systems with another type of compression, the signal to noise ratio can indicate very small values ​​even if the signals are identical to the human ear.

The reason is that the signal to noise ratio is not a good parameter of sound quality measurement because the quality perceived by the listener is determined by the response of the human ear to the sound waves, which does not perceive many of the possible differences Logically, if the signals are very similar, the ear cannot differentiate them, but they can also be very different and can be perceived as the original signal. Therefore, the evaluation of the quality of a digital system through sensitivity parameters of the human ear and specific tests with specialized listeners seems more appropriate.

It is in this sense that the quality of digital audio systems is evaluated today. Both MPEG and Dolby Digital (AC-3), which establish perceptual compressions, perform test benches to estimate the quality of the encodings.

The truth about audio formats and their quality

As you well know, there are three digital ways to play a song. From the original recording, through a copy with lossless compression (what we usually find when we buy a CD) or through a copy with lossy compression (which we usually download legally from the G.G internet).

The three files differ basically in the same property, which is none other than bitrate, that is, the information it contains per second. As more bitrate, more weight, so an audio file of a song without compression can quietly occupy 200mbs. In the second case (lossless understanding), the weight is reduced a lot (over 40, 50mbs), and is obtained by reducing the bitrate in those parts with silence, or with a wave oscillation in a single spectrum. To understand each other, the healthy young ear recognizes between 20Hz and 20KHz. A file with lossless compression (usually .flac files or those found on a CD) maintains this spectrum, reducing it when it is not necessary (silences). And finally there are the files with compression and loss (.mp3, .mp4, .flv, …) files that reduce this spectrum to the one that most ears recognize, leaving it around between 15Hz and 15KHz, obtaining a weight per file of song around 4mb, 5mb.
The latter has always been questioned, especially in musical circles, which assured that compression with loss greatly diminished the quality of what was reproduced, not allowing to admire the most serious or the most acute, thus losing all the completeness of the work.
As if that were not enough, mp3 files have different encodings (64, 128, 192 or 320 Kbps), with a greater or worse loss, and even constant (CBR) or variable bitrate (VBR) that is usually optimal when compressing with various bitrates Different moments of the songs.

Loud speaker and sound wave

Well, it has been more than 50 years before a good music lover programmer named Jeff Atwood decided to see if there really is a substantial change for the human ear between the different formats. In his blog, after several entries and several weeks of study in what would be called The Great Experiment of bitrate in MP3, we have finally obtained an empirical version of this eternal question.

But let’s make a brief summary of what we have in hand.
To test his hypothesis, Atwood decided to hang five audio files from his website, one of them being the original (without any digital treatment that modifies the bitrate), and another four tablets at various bitrates between 128 and 320 Kbps. The objective was that the user entered, listened to the five, and chose which one seemed to him to have higher or lower quality. Best of all, he obtained a not insignificant opinion of 3,500 visitors, hanging the results weeks later.

And from his observations you can get some gold reefs:
No doubt people knew how to differentiate the worst of all, such as the mp3 encoded with the worst bitrate 128 Kbps CBR.

The variable bit rate coding proved to be higher than the constant.
The most positive audio obtained was that of 160Kbps VBR, even higher than 320 Kbps CBR, and paradoxically also superior to the original audio of the CD.

From all this follows a corollary:
People are unable to ensure that it has more quality above 160Kbps, so it sends lossless formats to the horn that occupy one more scumbag, and that in practice, our ear cannot discern.
So you know. That’s over 15, 20 songs for a CD. There is no excuse.

Mp4 Gain : Video & Audio Normalizer

Mp4 Gain : Video & Audio Normalizer

 

The name can be misleading Mp4 Gain that one can believe that dse is simply a software to normalize volume of MP4 format files when in fact many more standardized formats:

Video Formats:
mp4, flv, avi
mpeg, mpg
3gp, wmv

Audio Formats:
mp3, mp2, flac
ogg, m4a, acc
wav, ac3

In all these formats Mp4 Gain normalizes the volume for all sound at the same volume, also with great audio quality.

This is a milestone in the field of audio volume and video files, as there was nothing like so far, no such software could standardize formats and much less with the quality Mp4 Gain get it done.

The way he does is analyzing each frame by multiple forms of cdada one of the files (video or 3-minute song usually has about 7 million frames), so the setting is 40,000 VECs per second, making every instrument, every sound, every voice, stand properly, the color, brightness, volume and intensity required.

Of course the analysis is unique, profiundo and latest technology, which achieves a surprising result.

Until now it was very incomod have files or videos that have such different volumes and the appearance is finally getting Mp4Gain the repair this problem and avoid so many troubles. You can now listen to your music or video in any condition and on any computer, even using any software and sound with the same volume level.