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Bitrate or Kilobytes per second (kbps)

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What corresponds to talk now is about the quality of these paintings. Let’s take a frame and see its quality and weight in kb … This is called bitrate or kbps which means KiloBytes per Second or Kilobytes per second, and is the measure used to determine the image quality, the higher the bitrate, the image quality will be better, but the weight of the file will be greater too. There are standards and limits for bitrate, this for the MPEG format. For a DVD, which is an MPEG2 video format, the limitation is more than 1500 and less than 8000 kbps, for VCD it is exactly 1150 kbps, and for example a MiniDV tape of a handycam uses a bitrate of over 30000, this is a lot of quality and when you go to hard disk you see the large file size. For the other formats it depends on the codec and its configuration, being able to compare in quality an MPEG2 format with 3000 kbps with an AVI with Xvid codec at 1500 kbps, this difference is given by the way in which the video is compressed, the Xvid codec It does better than the MPEG2 format.

Now, bitrate can be constant throughout the video or it can be adjusted to changing needs, this is Constant Bitrate (CBR) and Variable Bitrate (VBR), when we use Constant Bitrate all video always uses the same amount of information to show Each frame, whether you need it or not, this can make the video heavier than it could be or that it does not have the right quality in certain parts or has plenty of information where it is not needed.

Let’s see the following example, where a small video clip is shown where it requires a lot of bitrate or kbps and another part where it does not need so much, encoded in MPEG CBR and VBR. We see the difference in the weight of the files and their comparison with an Avi with DivX codec with a lower bitrate. In minute 2 another video is shown where very little bitrate is needed and as when coding in VBR or CBR it does not greatly affect the final result.

When we use Bitrate Variable we can define the maximum and minimum to use and let the coding program scan the video and adjust the amount of information as necessary, for this you can use one or two passes (usually no more than two are used), that is, the program goes through the video once, calculates and in the next pass applies the bitrate correctly. It can also be done in a single pass, but the calculation can be wrong, this because to make the necessary bitrate changes makes use of something called keyframes or keyframes. In addition there are programs that detect scene changes and there apply keyframes to change the amount of bitrate occupied, this may be in one pass, but it may also not be as accurate as two passes.

To prevent the file from being excessively complex and heavy to process, when encoding an amount of information is allocated for video blocks, these blocks are defined by the keyframes, for example, there may be 90 frames between each keyframe, and if the video run at 30 frames per second, then the bitrate can change only every 3 seconds of video (3 seconds x
30 frames = 90 frames), and if there is a change of scene, for example, from a dark and slow shot to a scene with light and moving, if we are in the middle of a block between keyframes we will notice that the scene in motion is seen bad, with large video blocks and little definition, this can happen when the coding is done in a single pass using Bitrate Variable, so it is preferred to use 2 passes, in this way the program is responsible for detecting the change of scenes where there is need to adjust the bitrate, and accommodate the keyframes of
correct way, so that in each change the quality of the video is adjusted and seen
all right. And what happens when we have a whole video with movement, well, the bitrate remains high for most of the time, this will make a video that is heavier than another of the same duration but where the content is more static while maintaining good quality . If we use Constant Bitrate for both videos, one with a lot of movement, and another with more static content, the result will be very similar, it will only change the quality of the content, unless we use a very high bitrate to give quality to the video with movement, which it would be excessive for a video of more static scenes, which would generate a heavy file without any need. Therefore, we have videos that can be heavier than others, and videos with different qualities. This is where we need to know what should be adjusted when encoding a video and how automatic adjustments do not always give the same results, this because each video may have different coding needs.

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Normalize audio: Normalization and normalizer

Normalization – Normalize audio

Normalization is an atypical dynamic process, very different from compression, limitation, expansion or noise reduction:

It does not reduce the relative dynamic range of the audio signal.
It is not applied in “real time”, or at the moment, but it is a process that is carried out “a posteriori”, on the previously recorded material.
The normalizer is a device that falls into the category of dynamics processors.

Analyze the target signal, detect its highest volume peak and increase its gain to the maximum level possible without distorting.
With the same proportion the level of the rest of the signal increases.
The signal, in general, will sound with a greater volume.

It is very unpleasant to have a playlist with mp3 files playing at different volumes. However, it is possible to normalize (equalize) the volume of our mp3s.

Loudness normalizer

You can use the Loudness Normalizer to obtain a specific loudness.

Increasing the loudness to a specific value may cause clipping. To solve it, a peak limiter can be part of the process. The Loudness Normalizer increases the loudness and limits the peaks of the signal at the same time if necessary, to obtain the desired loudness.

This process takes place in several stages, first an analysis and then the final rendering.

What is an audio compressor?

When we talk about compression we also talk about dynamic range. Recall that the dynamic range is the difference in amplitude between the lowest and highest part of a signal. In the compression process, basically and technically it consists in decreasing the dynamic range of a signal.

These are the most common controls or points to control in an audio compressor:

Threshold: This is the point from which the compressor will start working. Any signal that exceeds this point will be compressed.

Ratio: This is the reason for the compression that will be applied to the signal that exceeds the threshold. We see it and find it as 2: 1, 4: 1, etc. The first number means the number of input decibels that exceed the threshold, and the second number the output decibels. In other words, for “X” number of decibels that exceed the threshold, “Y” number of decibels will be output.

Attack: It is the time it takes for the signal to compress after having exceeded the threshold. We see it and find it commonly expressed in milliseconds (ms).

Release: How soon the compressor stops working after the signal falls below the threshold.

Knee: This is a parameter that subtly modifies the compressor threshold. A low setting (Hard knee or 0) means that the compressor will act only from the set threshold. A high setting (Soft knee) will allow the compressor to act gradually from before the signal reaches the threshold. In this way, the threshold can be treated as a range and not as a specific point. As the signal approaches the threshold compression increases. When it exceeds it, it continues to increase until the entire compression ratio is applied.

WebM: everything you need to know about the Google format

What is the WebM?

WebM is a container format (with extension * .webm) for multimedia files, that is, for videos and audio files. In the same container the video codecs VP8 and VP9 are used, as well as the Vorbis and Opus audio codecs. At the Google I / 0 2010 conference, the company announced its plan for WebM to be an alternative to the existing MP4 format with its H.264 codec from the beginning. The consumer can use the latter at no cost when watching a video, but developers who want to work with the codec must pay the license fees. On the contrary, WebM is an open source project with which anyone can work without paying rights for it.

WebM is designed for use with HTML5. The VP8 and VP9 codecs are designed so that in those cases where considerable compression must be carried out, the extraction can still occur with little computing power. The objective of this design is to allow the reproduction of Internet videos on virtually any device (regardless of whether it is a desktop computer, a tablet, a smartphone or a multimedia device such as a Smart TV). It is not surprising that YouTube, being a subsidiary of Google, converts all its videos to the WebM format, regardless of the format of the original file. Despite everything, YouTube still supports H.264 for those who cannot play WebM.

WebM has become a political issue within the Internet community. While Google tries hard to consolidate this audio and video format, other important market players such as Apple or Microsoft cling to formats like MP4. The main reason is, above all, the patent system: both software companies use a group of MPEG-LA patents, since it is responsible for maintaining the patents of the used codecs and charging royalties for them. Google is trying to circumvent these patents with WebM.

This situation has already led to legal problems in the past, the VP8 codec being the point of contention. Several companies have criticized that their codec patent has been ignored. Google would have reached an agreement with MPEG LA, however, Nokia is not part of this patent pool and believes its rights have been ignored. A first lawsuit, in which the company faced its competitor HTC before the courts, whose devices support V8, was dismissed by the Mannheim regional court.

WebM vs. MP4: advantages and disadvantages

While WebM is relatively young, MP4 (MPEG-4 Part 14) and H.264 have been used for many years. Due to its age, this format and the codec have become a standard: you will find few applications that do not support MP4. In addition to Internet services and PC and MAC software, many other devices (such as camcorders) can also use MP4. The high degree of acceptance makes the format interesting for both manufacturers and users.

But Google has been marked somewhat with the open source character of WebM: using the format is no cost to manufacturers, developers or end users. In addition, the software is distributed under an open BSD license.

The fabric behind the MP4 or H.264 license is opaque: most users, even those who create videos in a professional way, do not know if they have a valid license with the purchase of hardware or software or if any video violates The license right. WebM eliminates this confusion. The MPEG LA already announced in 2010 that the use of the H.264 codec would also be free in the future, provided that the videos created were already free for users.

For many users, the performance of both formats is more important than the controversies surrounding their patents: it is for some reason that H.264 has positioned itself as the leader of the codecs in recent years. The quality of MP4 videos of this encoding is generally considered very good. H.265 exceeds it in some aspects. WebM also convinces with the image and audio quality, but VP8 does not reach the level of H.264. To what extent the image quality of VP9 approaches H.265 (also known as HEVC) is a controversial issue; some believe that both are equal, while others say that the quality of VP9 does not reach that of H.264.

Two other determining characteristics when comparing codecs are the file size and the speed of encoding and decoding. Both directly influence the utility: for fast data transmission over the Internet, the size should be kept as small as possible. This is especially relevant in the mobile Internet field. H.264 has a bad reputation for creating, in comparison, large files. At the same time, decoding on the user’s site

How MP3 files work

The MP3 movement is one of the most incredible phenomena that the music industry has ever seen. Unlike other similar phenomena, such as the introduction of cassette tape or CD, MP3 technology did not start with the industry, but with a huge audience of music lovers on the Internet. The digital MP3 music format has had, and will continue to have a great impact on how people collect, listen and distribute the music.

If you have wondered how MP3 files work, or simply want to know what uses can be given, read on. This article will give some features of this popular sound format.

MP3 format

If you know something about how CD’s work, then you know how they store music. A CD stores a song in the form of digital information. The data on a CD uses a decompressed high resolution format. This is what happens when a CD is created:

The music is sampled (fractionated) 44,100 times per second. Each of these parts has a size of 16 bits.
Pieces of these fractions or “samples” are taken from the left and right channels in a stereo system.
With a simple formula we realize how great a single song can be.

Fractions * bits * channels = X bits per second

In our case it would be 44,100 for 16 bits per 2 channels, which would give us 1,411,200 bits per second. 1.4 million bits per second equals 176,000 bytes per second. If the average of a song is 3 minutes, then the average of a song on a CD is 32 million bytes of space. That is a lot of space for a song, and it is especially great if we consider that we are downloading music with a 56K Modem, which will take us a few hours.

The MP3 format is a compression system for music. This format allows you to reduce the number of bytes in a song without damaging the sound quality. The goal of the MP3 format is to compress a CD quality song without letting you see the difference. With MP3, a 32 MB song from a CD, compresses up to 3 MB. This allows you to download a song in minutes instead of hours, and store hundreds of songs on your computer’s hard drive.

Compression and quality

Is it possible to compress a song without damaging the quality? To perform this compression, the use of algorithms is needed, in the same way that we use them to compress other formats, such as graphics, text files, applications, etc. A very popular algorithm for compressing sound is the “perceptual noise shaping” technique. This algorithm uses characteristics of the human ear such as:

There are certain sounds that the human ear cannot hear.
There are certain sounds that the human ear hears better than others.
Its there are two sounds playing at the same time, we can hear the one that is louder, and not the lowest.
Using factors like these, certain parts of the song can be eliminated without significantly damaging the quality of the song for the listener. When you have created the MP3 file, what you have is music with a quality close to that of a conventional CD. It doesn’t sound exactly the same because some things have been removed, but it’s very close.

Using the MP3 format

The MP3 movement – consisting of the MP3 format itself and the ability of websites to distribute it – have done several things in the music world:

It has made it easy for anyone to distribute music at a low cost, or even for free.
It has made accessing music simple and instant.
He has taught people to manipulate music on a computer.
One of the strengths of this format is the ability to edit, create and modify music files thanks to powerful computer software tools. Thanks to these tools, it is extremely easy for anyone:

Download an MP3 file from a website and play it instantly.
Transform or “rip” a song from a CD, to the MP3 format, and listen to it later.
Record a song yourself, convert it to MP3, and make it available to everyone on the Internet.
Convert MP3 files into CD files and make your own audio CD’s with MP3 files downloaded from the Internet.
Have thousands of hours of music stored on one or more hard drives.
Upload MP3 files to portable players and listen to them wherever you want.
To do all this, all you need is a computer with a sound card, speakers, an Internet connection, a CD / DVD player / recorder, and an MP3 player.

What is an audio compressor.

In the field of professional sound, a compressor is an electronic sound processor designed to reduce the dynamic range of the signal without noticing its presence too much. This task is done by reducing the system gain, when the signal exceeds a certain threshold.

Traditionally, compressors have been electronic equipment with one or two rack units, but software versions of them have appeared for some years.

A compressor acts in such a way that it attenuates the electrical signal by a certain amount (normally measured in decibels) and from a certain input level. The objective is to ensure that the resulting dynamic excursion is lower than the original, to protect certain equipment against possible signal peaks or, if it is a saturated sound, to try to hide the error.

Reasons to compress a signal

-Control the energy of the signal: The human ear is very sensitive, so the compression must be smooth and subtle so as not to capture it. This type of compression is used when there is a signal in which the intensity varies, so it is compressed to achieve a more constant signal within the values assigned to it.

-Control the peak level of the signal: Often the equipment is limited, so the amplifiers can saturate and therefore be damaged. In this case the compression is used to control the signal and thus protect the equipment.

-Reduce the dynamic range of the signal: By attenuating the peaks of a signal, we reduce its dynamic range. Many devices are limited by the peaks, and this allows the RMS level of the signal to be raised.

Compressor Uses

In the field of music, its use ranges from applications for musical recordings to live sound. For example, it is often used to add more glued to the sound, an effect that is achieved by compressing the signal to subsequently apply a gain to the output of the device, which usually conceals possible interpretation failures by the artist, at least as Dynamic control refers. A compressor is highly recommended (and with certain musical styles, indispensable) for when using an electric bass. The slapping effect (hitting the strings with the finger) produces extremely high output peaks (20 dB or 10 times more than normal), which at low output levels generate distortion, and at high volumes (as in recitals) they can cause serious damage to the amplifier, and even the speaker (an excess of “excursion” can cause the speaker to tear from its suspension). Even in the (theoretical) case of a musical system with an infinite dynamic range, the difference, auditory speaking, using or not the compressor is imperceptible. Its use is also very frequent in voices, since not all singers use the appropriate technique so the signal level varies constantly.

-It is widely used in broadcasting, to improve the speaker’s diction.
-Compress during mastering improves the sound definition of the final mix.
-To protect the equipment (speakers).

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.

The quality of YouTube videos leaves much to be desired: they need an update

When we watch a video on the platform, we can usually appreciate that, despite finding videos in 1080p resolution, the compression applied by the platform is too aggressive. This causes the final quality of the video we are watching to differ greatly from that of the original file. The codec that YouTube uses is H.264 / MPEG-4 AVC, using various profiles or “levels” that specify the maximum resolution, frames per second and maximum bitrate of each quality.

We have analyzed a few videos, and we have taken a fairly representative one that is available on both Vimeo and YouTube to see how both platforms compress the videos. In addition, we have seen the maximum and minimum bitrate that each video can have according to the YouTube Help page for each resolution. The audio, as we discussed in summer, reaches 128 Kbps, leaving 320 Kbps only for YouTube Red users.

What sound quality (bitrate) do YouTube videos have?

The bitrate for 1080p videos is too low: 4K is the way to go
The bitrates that YouTube says it assigns to each video are the following, with the profile level in parentheses:

4K / 2160p
60 fps: Between 20,000 and 51,000 Kbps (L5.2)
30 fps: Between 13,000 and 34,000 Kbps (L5.1)
1440p
60 fps: Between 9,000 and 18,000 Kbps (L5.1)
30 fps: Between 6,000 and 13,000 Kbps (L5.0)
1080p
60 fps: Between 4,500 and 9,000 Kbps (L4.2)
30 fps: Between 3,000 and 6,000 Kbps (L4.1)
720p
60 fps: Between 2,250 and 6,000 Kbps.
30 fps: Between 1,500 and 4,000 Kbps.
480p: Between 500 and 2,000 Kbps.
360p: Between 400 and 1,000 Kbps.
240p: Between 300 and 700 Kbps.

In our tests, the bitrates we obtained for the previous video were the following:

4K at 30 fps
Vimeo: 19.4 Mbps (file size: 943 MB) (capture)
YouTube: 17 Mbps (file size: 821 MB) (capture)
1080p at 30 fps
Vimeo: 4.31 Mbps (file size: 219 MB) (capture)
YouTube: 3.2 Mbps (file size: 160 MB) (capture)
vimeo vs youtube compression

As we see, Vimeo files occupy more not only because of the lower compression of the videos, whose quality is superior to the naked eye, but that Vimeo’s sound quality doubles that of YouTube, since it reaches 256 Kbps by 128 Kbps from YouTube. So that you can see the difference in image quality, you can open the same New Zealand Ascending video on YouTube and Vimeo, and we have also left four captures at the same moment of each video so you can save them and see comfortably the video difference.

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.

What audio formats exist? All you need to know

FLAC, WAV, AIFF, DSD … these are just some of the acronyms you can find when looking for a digital format. They are also accompanied by technical data such as sample rates and bit depth. So many terms can leave you more misplaced than a chicken in a dance. And unless you are an expert in digital sound, the process to choose the audio format that best suits your needs can be a mess. But if they explain it to you, the subject is relatively simple. That is why in Culturasonora we have prepared a complete guide on the different audio formats used. This will prevent any acronym from taking you on the dark side, dear Padawan.

Sample Rate and Bit Depth.
MP3s vs WAVs vs AIFF.
OGG vs FLAC vs ALAC.
What is the DSD format?
How to listen to the DSD?
MQA audio Hi-Res.
What is Bit Depth and Sample Rate?

These two concepts are basic. To understand how audio formats work, you need to know what Bit Depth and Sample Rate are. They are two measures that indicate the quality of a digital audio file. We will try to summarize it so that you stay with the general idea

When you read the specifications of the audio formats you find a couple of figures. For example: 32-bit / 192kHz or 24-bit / 96kHz. These numbers indicate the bit depth and the sample rate. These references tell us how much information the different formats transmit and the sound quality. For example, the audio we hear on a normal CD, or on a Spotify stream, is 16bit / 44.1kHz. Samples are always measured in Hertz (or hertz) and bit depth in Bits.
Softwares or hardwares do not usually work with a continuous flow of information but often use pieces, samples or samples to effectively manage the data that is transmitted. The sample rate is the number of samples per second that are obtained from a recording. The higher the number of times a device plays the samples, the higher the sound quality. Each of these extracts or samples has a certain amount of information, which is the bit depth, or bit depth.
To understand it better, we are going to make a slightly beast analogy, which is not entirely true, but which will help you to make sense of all this. What interests us. If you control a bit of photography and image you will get it right away: the sample rate would be something similar to the frames or frames per second of a video, and the bit rate would be similar to the pixels of a photograph. The higher the bit depth number, the more information each sample will have. The more pixels an image has, the more resolution each frame of a video will have. The more frames per second a movie has, the greater the definition. In short: the higher the number of the Bit Depth and the Sample Rate, the higher the quality of the audio file.

Audio formats: MP3 vs WAV vs AIFF

What is the MP3 format?
If you are interested in getting some audio fidelity and decent sound from your files, you will want to avoid this format. Why? Because basically an MP3 is a file that sacrifices audio quality to minimize size. They weigh very little for any device to read. The negative? The compression of these files provides a poor, almost lifeless sound. Nowadays almost nobody uses that format seriously. Even its creators recently finished the license declaring her dead. But surely every now and then you find a zombie file with this format.
What is the WAV format?
WAV (Waveform Audio File Format) are equally common but better for anyone who wants a decent audio format. They are higher resolution files than MP3s. A WAV is an audio piece that is encoded with something known as Pulse Code Modulation (PCM), a medium that encodes analog audio parts and converts them into digital so that they can have the Sample rates and the Bit Depth of the that we have talked about before.
What is the AIFF format?
The audio format AIFF (Audio Interchange File Format) is very similar to WAV, since it also uses the PCM to encode analog audio pieces and present them in digital format. This format was born as an answer from Apple to the Microsoft WAV, and at the beginning it could only work on MAC computers. Currently, the AIFF and WAV are more or less interchangeable.
In summary…
To close this topic we will tell you that if you have a file in WAV or AIFF audio formats you will hear a piece of good quality sound. Normally these formats are used in files that we play through our services, such as the iTunes music library. We will not see them in online streaming services, which tend to use special types of files. Now we will review that point

The great experiment on MP3 quality: no, there really isn’t that much difference with CDs

This article was originally published in Cooking Ideas, a Vodafone blog where we collaborate weekly with the goal of creating stories that “feed the mind of ideas.”

A programmer named Jeff Atwood said some time and several entries from his blog, the always recommended Coding Horror, to a healthy entertainment he called The Great Experiment of bitrate in MP3. Its objective: to verify empirically if for ordinary people there are really qualitative differences when listening to music in various MP3 formats compared to traditional ones.

The contestants were the traditional formats called “no loss of quality”, basically CD (Compact Disc) and FLAC versus compression formats with loss of quality: MP3 with different bitrates. The bit rate, better known by its name in English, is a key feature because it basically determines how much information is transmitted per unit of time: in this case it is the waves that define the music and become human voices and instrument notes . In the world of MP3 encodings of 64, 128, 192 or 320 Kbps (kilobits per second) are usually used.

Like everything in life, music coding is a compromise between quality and quantity: a song stored in the best possible format – for almost all experts, that is the CD – can occupy about 50 MB (megabytes), maybe 40 or 35 only using some of the lossless compressors that save some space without loss of quality (FLAC, Apple Lossless, etc.). That same song in MP3 can vary between 4, 8 and 12 MB depending on the bitrate (64, 128 and 192 Kbps). To further complicate the matter, you can also choose between a constant (CBR) or variable (VBR) bitrate that is usually optimal when compressing different moments of the songs with various bitrates.

For many users, being able to store between 5 and 10 times more music in the same space is an important saving, easy to translate if one takes into account the price of hard drives, flash memories or storage on iPods, tablets and the like. But there have always been two schools confronted: that of audiophiles who believe that nothing can equal the maximum quality of the CD and that of those who, with a more practical sense, consider the differences between an MP3 and CD ridiculous, if at all there are.

Atwood’s experimental study sought precisely to shed some light on these theories based on the basics: listening to music, quantifying its “quality” and deciding which is the best format based on the various variables. For this, he prepared five different audio files: one of them uncompressed and another four tablets at different bitrates between 128 and 320 Kbps. He put them on his server so that people could listen to them and vote (with a quality “note” of 1 to 5) without knowing which was which. And in total he got more than 3,500 people to contribute to the results – hundreds more than for many of the “quality studies” mentioned in the TV commercials.

The results were analyzed with a spreadsheet and various statistical tools, which showed trends and conclusions quite clearly:

The only sample that could really be considered very different from the rest was the MP3 at 128 Kbps CBR, the worst quality. That quality is not enough to compare with the rest. The best simply ignore it.

The MP3 at 160 Kbps VBR is the highest quality sample, even better than the MP3 at 320 Kbps CBR. This indicates that the coding with a variable bit rate is higher than the fixed one even at those values, and that 160 Kbps VBR up is impossible to improve qualitatively.
Ironically, this would indicate that there are MP3s that are heard “better” than audio CDs. Several things can happen here: that the “artifacts” created by compression seem to improve the audio or that when testing people “imagine things,” which could also happen. The truth is that the data serves to feed the theory that from 160 Kbps people no longer distinguish one quality from another, as it is deduced from the data.

The conclusion of the study confirms the hypothesis that an MP3 at 192 Kbps VBR has such quality that not even the ultrasensitive and powerful ear of a dog would notice the difference with an audio CD. Wow!
In conclusion, we already know at what rate to code and compress if we want a good saving in storage without losing quality: a MP3 of 192 Kbps VBR, the winning format of the test.