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Sound file formats

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Sound file formats

Sound is a natural phenomenon and propagates through air vibrations, so we can say that it is only wave characteristics.

The task of converting sound into electronic form is to repeat all of these same wave characteristics. But the electronic signal converted into an ADC (analog-digital converter) is not analog and is recorded by short discrete values. Let them have a small interval between each other and be practically imperceptible, but we must always remember that it is only the emulation of a natural phenomenon called sound.

This recording is called pulse code modulation and is a sequential recording of discrete values. The length of the device, calculated in bits, indicates how many values are taken simultaneously in a recorded sample. The higher the bit depth, the closer the sound will be to the original.

Any sound file can be presented (for your understanding) as a database. It has its own structure, the parameters of which are usually indicated in the file header. Next comes a structured list of values for specific fields. Sometimes, instead of values, there are formulas to reduce file size.

RSM

PCM stands for Pulse Code Modulation, which stands for Pulse Code Modulation. Files with this extension are quite rare (I’ve only seen them in 3D Audio). But PCM is essential for all sound files. I wouldn’t say that this is a very inexpensive method of storing data on a disk, but I think you will never escape this, and modern hard drive volumes already allow you to ignore a couple of tens of megabytes.

DPCM

Research on economical storage of audio data on disk. If you come across this abbreviation, know that it is Differential PCM. This method is based on the quite justified idea that the calculations are much more cumbersome, compared to the fact that you can simply enter the difference values.

ADDPCM

Adaptive DPCM. Agree that by specifying only the difference values, there may be a problem with the fact that there are very small and very large values. As a result, no matter how super accurate the measurements are, there is still a distortion of reality. Therefore, a scalability factor has been added to the adaptive method.

Wav

The simplest discrete data storage. I would say direct. One of the file types in the RIFF family. In addition to the usual discrete values, bitness, number of channels, and volume level values, wav can contain many more parameters that you probably didn’t even know about. These are placemarks for timing, the total number of discrete values, the order of playing various parts of the sound file, and there is also a place for you to put text information there.

RIFF

Resource interchange file format. A single storage system for any structured data.

IFF

This storage technology is derived from the Amiga systems. Interchange file format. Almost the same as RIFF, only there are some nuances. Let’s start with the fact that the Amiga system is one of the first who started thinking about software-sampler emulation of musical instruments. As a result, the sound in this file is divided into two parts: what should sound at the beginning and the element of what follows at the beginning. That is, the beginning sounds once, then the second piece is repeated as many times as necessary, and the note can be played indefinitely.

MODIFICATION

The file contains a small sample of the sound that you can then use as a template for the instrument. Simply put, a sample stitched onto the synthesizer.

AIF or AIFF

Audio interchange file format. This format is common on Apple Macintosh and Silicon Graphics systems. Contains a combination of MOD and WAV.

AIFС or AIFF-С

The same AIFF, only with the specified compression parameters (compression).

Again, the race to save space. The structure of the file is much simpler than in wav, but there is a method to encode the data. The files are very light, so they are quite widespread on the Internet. Most of the time you can find parameters * -Law 8 kHz – mono. But there are also 16-bit stereo files with frequencies of 22050 and 44100 Hz. This audio format is designed to work with audio on SUN, Linux, and FreeBCD production systems.

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Description of the main audio formats. Audio file formats

Description of the main audio formats. Audio file formats

Surely, many users prefer to use their home computer not only as a workhorse, but also as a multimedia center where they can watch movies or family photos, as well as listen to their favorite music. Although compact digital players or mobile phones are certainly more suitable for listening to musical compositions, but unlike them, a computer can not only play music.

No matter how big the built-in memory of your music player is, it will most likely be difficult to store your entire music library on it. Plus, you can create, edit, organize, and search for music with your PC. Also, don’t forget that there are around three dozen common digital audio formats today, and most players are far from omnivorous and can only play a few of them.

So why do you need to create so many music formats to store one type of content? The fact is that, in the vast majority of cases, the sound is stored in “compressed” form, since one minute of uncompressed composition occupies about 10 MB on the hard disk. On the one hand, this seems not to be much, but on the other, if you are a music lover and your collection consists of several hundred or even thousands of songs, then it is clear that the sound must be compressed to reduce the space it occupies in electronic media.

Various special algorithms are used to compress music files, which subsequently determine the structure and presentation of the audio data, or so-called digital audio file formats. All audio formats can be divided into three groups: uncompressed audio formats, lossless compression, and lossy compression.

No compression
One of the most widespread formats related to this type is the well-known WAV. The sound of files with this extension is stored without compression or changes. It is true that much more space is required to store uncompressed files and therefore WAV is more widely used only in professional audio and video applications, where the sound should not have a loss of quality before processing. Keeping ordinary musical compositions in this form is unwarranted waste.

To play WAV files, you do not need any special software, as all media players understand this format, including the standard Windows Media audio player built into the Windows system.

Another format used to store uncompressed audio that is worth mentioning is Apple’s development called AIFF (Audio Interchange File Format). As you may have guessed, it is most commonly used on Macintosh computers running Mac OS X.

Lossless compression (lossless)
Lossless compression algorithms for audio files work on the principle of conventional file cabinets. They do not provide the highest level of compression (40 to 60%), while they have virtually no effect on sound quality. It is also worth noting that in this case, the encrypted data can be fully restored to its original form. Therefore, the use of lossless compression is most often used in cases where it is important to preserve the identity of the compressed data with respect to the original.

The most popular audio formats in this group are FLAC (Free Lossless Audio Codec), APE (Monkey’s Audio), WMA (Windows Media Lossless), and ALAC (Apple Lossless Audio Codec). Each has its own pros and cons. For example, the APE codec offers slightly better compression gains, while FLAC is more common. In general, all true music lovers store their music collections in lossless formats, since they do not remove any data from the audio stream and files created with these codecs can be listened to even on high-quality stereos.

Files with digitized audio

Files with digitized audio

Digital audio

Sound files in which the original continuous (“analog”) waveform is recorded as a sequence of short discrete values of the amplitudes of the sound signal, measured (“selected”) at equal time intervals and with an interval very small between them.

DIGITAL AUDIO

The process of replacing a continuous signal with a sequence of its values is called sampling, and this form of recording is pulse code. The hardware implementation of digital audio processing is that an analog-to-digital converter (ADC) converts an analog signal into a set of digital measurements and, during playback, a digital-to-analog converter (DAC) performs the reverse process: convert a digital signal into analog. There are two types of files with digitized audio: header and no header.

Files with music notation (song file, music file): sound files that contain a sequence of commands indicating which note and by which instrument and for how long to play at any given time. The format can foresee the simultaneous execution of several musical instruments, in this case it speaks of the corresponding number of voices.
Edit Basic standards for multichannel audio

Dolby Stereo is a standard for digital movie sound recording / playback technology for cinemas that allows four channels to be encoded into two movie soundtracks: left, center, right, and rear. The signal read from the film is converted by the decoder into four channels, which gives a surround sound effect. Without a decoder, the sound is played as normal two-channel stereo. The standard was proposed by Dolby Laboratories in 1976.

DDS (Dolby Surround Sound) is a standard for digital recording / playback of movie soundtracks in the frequency range 100-7000 Hz for home theater systems. The standard allows encoding three channels in two soundtracks of a movie: left, right and rear. The signal read from the film is decoded into three channels. Without a decoder, the sound is played as normal two-channel stereo. The standard was proposed by Dolby Laboratories in 1982.
DPL (Dolby Surround Pro Logic) is an evolution of the DDS standard for home theater systems with three to four sound channels: left, center, right and surround. The standard was proposed by Dolby Laboratories in 1987.
Dolby Digital is a standard for encoding / decoding six-channel (5 + 1) audio recording in the 20 Hz to 20 kHz range: 5 surround channels and one low-frequency channel (subwoofer). The standard was proposed by Dolby Laboratories in 1992. The frequency range of the five channels is 3 Hz to 20 kHz, the subwoofer is 3 Hz to 120 kHz.
Dolby Digital AC3 is an addition to the Dolby Digital standard with a scheme that provides an audio recording compression density of 12: 1 or more at a 64 to 640 Kbps bit rate with high quality playback.
Dolby Surround AC3 is a simplified version of the Dolby Digital home theater standard with reduced bit rates.
DTS (Digital Theater System) is a standard for six-channel (5 + 1) sound recording on music DVDs, close to Dolby Digital, with a lower compression ratio (4: 1) and a faster data rate. high (bit rate – 882 Kbps). Due to this, in addition to the use of a perfect compression algorithm, it is characterized by high-quality sound recording and reproduction. The recording uses a 48 kHz sample rate, making it the highest quality DVD audio standard ever recorded.
Dolby Pro Logic II is an evolution of the Dolby Surround Pro Logic standard, which breaks down normal stereo sound into six channels: 5 + 1.
Dolby Pro Logic Iix is an evolution of the Dolby Surround Pro Logic standard, which provides stereo sound decomposition into 7 (6 + 1) or 8 channels (7 + 1). Possible decoding modes: Movie: mirroring the center channel or rear channels; game (Play): the signal is also sent to the “new channels”; Music).
Dolby Digital EX is a home theater variant of the Dolby Pro Logic Iix standard.
Dolby Digital Surround EX is an expanded version of up to 7 channels (6 + 1) of the Dolby Digital Surround standard, in which there is an additional rear channel (rear) that doubles the center channel if the sound is recorded in 5 + 1 format. If the sound is recorded in 6 + 1 format, the additional channel becomes a full surround channel.
DTS-ES is an analog of the Dolby Digital EX standard developed by DTS; allows you to encode audio in 6 + 1 and 7 + 1 formats and decompose audio encoded in DTS (5 + 1) format into 7 (6 + 1) or 8 (7 + 1) channels.

DETECTION AND ANALYSIS OF COMPRESSION TRACKS OF SOUND SIGNALS USING MP3, AAC, WMA AND VORBIS CODES

DETECTION AND ANALYSIS OF COMPRESSION TRACKS OF SOUND SIGNALS USING MP3, AAC, WMA AND VORBIS CODES

audio signal

DETECTION AND ANALYSIS OF MP3, WMA, OGG AND VORBIS CODECS IN AUDIO SIGNALS

audio signals

The article describes the method of MP3, WMA, OGG and Vorbis codec trace detection in the audio signal.

The method reveals digital audio editing, sample rate change, and multi-encoding traces. Keywords: digital audio and video forensics, codec trace detection, psychoacoustic codecs, MP3, AAC, WMA, Vorbis. Introduction Today, digital phonograms or video phonograms, the audio signal of which has been compressed, quite often become the subject of examination of video and sound recordings.

The purpose of compression, as a rule, is to reduce voice traffic on communication channels, or to reduce the amount of data stored. Devices and programs that implement algorithms to compress audio and video signals are called codecs.

Increased recognition in the field of digital recording and storage of audio signals received the so-called psychoacoustic codecs, which provide compression of the signal by removing from it spectral components inaudible to humans (frequency and time masking). The use of such codecs significantly reduces the amount of memory required to render the signal, leaving the sound quality at an acceptable level for everyday use, which is why psychoacoustic codecs are widely used in the media industry.

The most famous and widespread representative of the psychoacoustic codec family is MPEG 1/2 / 2.5 Layer 3, better known as the MP3 codec. Developed more than 20 years ago, the MP3 codec is now implemented in almost any device with the function of recording and reproducing phonograms or video phonograms at the software or hardware level.

In the last decade, psychoacoustic codecs have become increasingly common, using more advanced psychoacoustic models: Advanced Audio Codec (AAC), WMA (Windows Media Audio), and Ogg Vorbis (OGG). Theoretical foundations When analyzing the dynamic spectrogram of a signal that has been encoded using psychoacoustic codecs, it is often easy to notice rectangular outliers (Fig. 1), which is one of the signs of using one of the psychoacoustic codecs. Figure 1. Dynamic spectrogram with MP3 codec coding traces. These dropouts are the result of encoding the signal using the psychoacoustic codec, the operation of which is described below using the MP3 codec as an example.

In the first stage of MP3 encoding, the spectrum of the signal is calculated using the Modified Discrete Cosine Transform (MDCT). Furthermore, based on a psychoacoustic frequency and time masking model, the inaudible components of the MDCT spectrum are reset to zero. The spectrum of the signal is then quantized and encoded using the Huffman method. To simplify further description, the description of the coding step associated with band-pass filtering and reduced sampling of the signal “bands” is omitted before calculating the MDCT spectra, as this is irrelevant in the context under consideration. In connection with this simplification in the work, the sizes of the analysis windows will be indicated for the original signal, and not for the “bands” of signals, as indicated in the specifications.

For convenience, MP3 spectra are called MDKP spectra, which are calculated in the same way as with MP3 encoding. The calculation of MP3 spectra can be performed using four types of analysis windows: with a standard window of 1152 counts in size (indicated in blue), a small window of 384 counts in size (indicated in red) and two types of windows transition (indicated in green) color).

In this case, the window sizes do not depend on the sample rate of the original signal. During the encoding process, the original signal is divided into fragments that intersect with a step of 576 samples (step of the MP3 encoding window). The size of the fragment, depending on the type of window, can vary from 1152 samples for the standard window, 960 for the transition window and 768 for the small window (three small windows with a 50% intersection), but the step between the “centers” of the fragments in all cases is 576 samples.

What exactly is a codec?

What exactly is a codec?

Codec

Today there are about three dozen common digital audio formats. Why you need to create so many types of sound files to store one type of content and how to manage all this, you will learn from this material.

Codecs

Surely many users prefer to use their home computer not only as a workhorse, but also as a multimedia center, where they can watch movies or family photos, as well as listen to their favorite music. Although compact digital players or mobile phones are certainly more suitable for listening to musical compositions, but unlike them, a computer can not only play music.

To play WAV files, you do not need any special software, as all media players understand this format, including the standard Windows Media audio player built into the Windows system.

To play lossless compressed formats, as a rule, third-party players (except WMA) are used, such as MPlayer, foobar, AIMP, Winamp, VLC and others, since all the necessary codecs are already built into them. Another option is to separately install an additional codec pack (for example, K-Lite), after which you can listen to files in lossless format from almost any audio player.

Lossy compression
This is the most popular group of algorithms that provides the maximum audio compression ratio (up to 10 times or more). However, unlike previous formats, the audio file loses quality.

What are video codecs and audio codecs?

What are video codecs and audio codecs?

Video Codecs

Almost any computer user periodically listens to music on it, which is stored electronically. There are many formats for storing music, each of them was developed for specific tasks:

Video Codec

Bit rate is the amount of information used during encoding for 1 second playback. The higher it is, the less the distortion and the sound matches the original as much as possible.
Lossless – Lossless audio encoding. By converting to lossless formats and vice versa, we get exactly the same sound.
Lossy: compression formats designed for the fact that a person simply cannot physically hear certain frequencies that are skipped during conversion. At the same time, it can significantly save the amount of disk space.

Audio CD
The format that ushered in the era of digital sound after the transition from vinyl records. It was adopted as a standard in 1979 by Philips and Sony. In the audio CD format, music can be physically stored only on optical media; when recording to a hard disk, the audio track must be converted.

Due to the highest sound quality and the ability to play on any player, the format is still very popular, even though it is quite old.

Flac
Perhaps the most common format for storing lossless music. Compared to other lossless audio compression codecs, flac developed by xiph.org is completely free and offers the smallest output file size.

MP3
The most popular music format accepted as an unofficial standard for any playback device. Its popularity is based on the fact that because it cuts frequencies inaudible to the ear with practically the same sound quality, an mp3 file is 30% of the original lossless file.

The first audio track in mp3 format appeared in 1994. One of the reasons for its popularity is the ability to store a variety of additional information on audio file tags and the convenience of organizing a music library.

Ogg
A new lossy format that was launched in 2002 as a free alternative to paid formats. Unlike its predecessors, mp3 in particular, it allows the possibility of multi-channel encoding and multi-channel audio storage. It is most used in video games.

The term “audio” today means everything that is somehow connected with sound. This is processing, playing, mixing and simply listening to audio recordings. Few people know that during their existence all popular audio formats have undergone significant changes, sometimes for the better and sometimes even for the worse.

The problem is that when the creators tried to improve the recording quality by using the new format, the size of the result increased significantly. Reducing the size of the final file resulted in a significant loss of quality. But this was not always the case.

The first mention of computer sound is associated with the creation of several primitive video games. Then the sound was played back using the speaker of the system. As the software developers of that time did not try, they failed to achieve the level of quality that would be compatible with tape and reel recorders. This is what got many developers thinking about how to change the audio format to make the sound more natural and natural. It is this problem that has led to the current competition in the audio market. As a result, the formats used strongly affect the quality of the reproduced material and the configuration of the basic playback parameters.

WAV format

The first full quality of audio formats is associated with this particular format. The WAV extension designation was derived from the English word “wave”, which means wave in Russian. It was this format that became the first audio format to be processed with computer programs at a highly professional level. Files with a WAV extension had the following characteristics:

– depth of sound;
– sampling frequency;
– bit rate, etc.

This format was even compatible with the sound that could be obtained after processing an audio CD with an equalizer and other tools. However, the file size in this case was completely unwarranted. For example, the most common 3 minute long track could be up to 50 megabytes long.

Vorbis

Vorbis

OGG Vorbis

Vorbis is a free and lossy audio compression format that was officially released in the summer of 2002. In terms of functionality and quality, it is similar to codecs like AAC, AC3 and VQF, superior to MP3. The psychoacoustic model used in Vorbis is similar in principle to MP3 and the like, but the mathematical processing and practical implementation of this model are significantly different, which allowed the authors to declare its format completely independent of all predecessors.

ogg vorbis

Container

For storing audio data in Vorbis format, the Ogg media container is most frequently used, such a file usually has the extension .ogg and is named with the double name “Ogg / Vorbis” [1] or ” Ogg Vorbis “[2]. However, “Ogg Vorbis” is also called a codec by itself without a container, as it is part of the Ogg project [1].

Request

In 2020, it is much less widespread than MP3. Used in computer games, for podcasts.

Vorbis is ideal for use as movie soundtracks, as it does not change in length when the bit rate is variable, allowing you to stay in sync with the video track and is applicable for multi-channel sound (e.g. 6-channel audio). channels).

It is used for the audio track of WebM files in conjunction with the VP8 video codec (since VP9, support for the new free Opus codec has been added to the format).

Metadata

The format was originally designed with streaming capabilities. This gives the format a rather useful side effect: multiple songs can be stored in one file with their own tags. When loading such a file into the player, all songs should be displayed as having been loaded from several different files.

The format has a flexible labeling system. The tag header can easily be expanded to include lyrics of any length and complexity (such as song lyrics), interspersed with images (such as album cover photos). Text labels are stored in UTF-8, which allows writing in multiple languages at the same time and eliminates potential encoding problems.

Bitrate
Ogg Vorbis uses a variable bitrate by default, while the latter is not limited to hard values and can vary even by 1 kbps. It should be noted that the maximum bit rate is not strictly limited by the format, and with the maximum encoding setting it can range from 500 to 1000 kbps. The sample rate has the same flexibility: users can choose between 2 and 192 kHz.

Development objective
Vorbis was developed by the Xiph.Org community to replace all paid proprietary audio formats. Despite being the youngest format of all MP3 competitors, Ogg Vorbis has full support on all popular platforms (Microsoft Windows, Linux, Apple Mac OS, Android [3], PocketPC, Palm, Symbian, DOS, FreeBSD, BeOS and etc.), as well as a large number of hardware implementations.

Video and audio ad production: is there a difference?

Video and audio ad production: is there a difference?

Audio and Video differences

Since audio ads and video ads are different types of content, the process for creating them is slightly different. This is also true for dubbing, but not entirely. And today we tell you why.

Technical highlights
Generally, all the difference comes down to glitches – the most common sample rate when recording an audio track to copy video is 48 kHz. You can work with others, of course, but you have to remember that few video codecs work with an audio sample rate of 44.1 kHz. There are some other nuances of video advertising production that are well known to technical specialists (for example, the characteristics of the same frequency processing), and if they are all observed, the quality of the sound series will be the best.

With audio clips, everything is a little easier, but in this case, the quality of the final material will depend directly on the professionalism of the announcer and the sound engineer.

There are no subtle nuances in video ads – everything is important, from the choice of microphone to the nuances of voice processing.
There are no subtle nuances in video ads – everything is important, from the choice of microphone to the nuances of voice processing.
If we talk about the frequency of sound, then audio advertising is as open as possible for experiments: record the voice of the announcer at a whisper frequency: 100 Hz (for men) or 120 Hz (for women), and video it will inspire more confidence. The 2.4 kHz frequency is subconsciously perceived as alarming and is a great way to attract attention at times. And sound at a frequency of 15 kHz can attract the attention of young people without disturbing the elderly, since the former are susceptible and the latter are not.

Psychoacoustics opens up wide opportunities for creating audio advertising
Psychoacoustics opens up wide opportunities for creating audio advertising
These examples are the simplest “tricks” in psychoacoustics, with which you can “pump” your audio ad to achieve the desired effect.

However, all of this doesn’t mean that audio plays the last role in video ads.

Visual and sound range: two parts of a single whole
Yes, these parts are not at all equivalent, however a fly in the ointment (in our case, poor quality sound) is more than enough to ruin the whole barrel of honey (the impression of a perfect visual series). In other words, video advertising is a product that consists of a large number of ingredients, and if you don’t pay enough attention to each of them, you won’t get a really “tasty” video. It’s like making a complicated dish but forgetting to add salt to it. Or put on a tux and sandals – you won’t be sent to jail for this, but the effect will be unexpected, and of course not with a plus sign.

Everything in the video should be fine: both visual and sound
Everything in the video should be fine: both visual and sound
There is a difference?
I mean, does it matter where in the future you plan to play our video: on TV, YouTube, radio, or even on your personal blog? The answer may surprise you, but generally, no. Video is audio: the distribution channel is not important, only the format of the content itself is important: visual or auditory. The sound and video recording studio will take care of the rest on its own.

In other words, the difference exists only for the technical specialists, but for the client and other artists there is simply no difference.

HEVC Audio

H265/HEVC

More efficiency, more channels, more functions.

H.264 vs H.265

When we talk about video compression standards, such as MPEG-2 and H.264, most of us prefer to think only about the video aspects, without really thinking about the sound. Sound is certainly important too, but it just … is.

Today, with the world striving to offer something better than the current MPEG-4 / H.264 compression used on Blu-ray discs and most digital camcorders, it is worth spending a little more time learning the features of audio from next-generation video compression standards. The most likely contender for the number one next generation video compression codec is HEVC (High Efficiency Video Coding), also known as H.265 in the ITU rankings. , but remember this is a video compression codec, not sound. The development of the audio compression system that accompanies the video in the HEVC standard is being carried out by a completely different team, not the one that works in the HEVC / H.265 codec.

Google is also developing a competitive compression standard called VP9 that will be integrated into many web browsers. The use of the VP9 codec is royalty-free and Google believes that the codec will provide better performance than HEVC / H.265 in terms of compression efficiency and image quality. However, it appears that H.265 will be used in professional TV and video broadcasts, despite the need to pay royalties when using the standard.

Don’t forget about another video compression standard: the newly released Daala codec, jointly developed by the Xiph.Org Foundation and Mozilla Corp. The founder of Xiph.Org claims that the Daala codec will perform better than HEVC and VP9. , But this standard will not be ready this year. It’s funny that the Xiph.Org Foundation ever developed the FLAC (Free Codec for Lossless Audio Compression) standard, which has earned an excellent reputation for performance.

TWICE MORE EFFICIENT

In terms of video parameters, the efficiency of the H.265 codec is roughly twice that of H.264, which in turn was about twice as efficient as the MPEG-2 codec. In other words, a video stream compressed at 20 megabits per second using the MPEG-2 codec can be compressed to 10 megabits per second when using the H.264 codec and 5 megabits per second using the H.265 codec. Of course, this is a very simplified approach, but it’s fine as a practical example.

MP3 Logo The MPEG-2 codec gave most of us an idea of the MP3 standard used for audio encoding. The term MP3, first introduced with MPEG-1 compression, refers to the MPEG Audio Layer III codec. It has become a popular audio compression standard, but other standards are used in parallel. As in the case of the main video compression standard, the MP3 standard provides lossy compression, which means that this codec changes the audio during the compression process and these changes are irreversible.

The MP3 standard has a wide range of settings that affect the final audio quality, including the sample rate and bit rate settings. In most cases, MP3 audio is sampled at sample rates of 32, 44.1, and 48 kHz and compressed at 56 to 384 kilobits per second. At a bit rate of 128 kilobits per second and a sample rate of 44.1 kHz, the resulting MP3 is approximately 9.1% of the uncompressed CD file. Compressing MP3 at a bit rate of 320 kilobits per second creates a bit stream that is approximately 23% the size of an uncompressed CD file.

The AAC (Advanced Audio Coding) codec was developed on the basis of MP3, taking into account the experience gained during the development and operation of the initially popular format. In general, using the AAC codec offers better sound quality compared to MP3 at the same bit rate. The AAC codec also has a kind of “fork” known as the High Efficiency Advanced Audio Coding (HE-AAC) codec, which is used in mobile TV standards such as DVB-H and ATSC-M / H. Just like MP3, the AAC codec is a lossy compression format with a number of MP3-like settings.

Choosing an audio codec for online streaming and recording.

Choosing an audio codec for online streaming and recording.

Audio Codec

Are you interested in what is an audio codec and how to choose the right one to get the best result from online streaming or recording?

Audio Codecs

Imagine that we live in a completely analog world. Then there would be no need for audio codecs. What is it, you ask? It is an algorithm used to convert analog audio to digital. This is what is needed in the world of digital devices, media players and the Internet.

The quality of audio codecs has improved significantly over the years. Let’s go back, for example, to the 80s, when the first digital amplifiers appeared. Compared to the reproduction quality of a modern digital amp, the difference will be obvious. The best audio codecs offer better and more realistic sound.

But now there are so many different audio codecs. Which to choose?
Many codecs are quite specific. Some of them are proprietary, while others were created for specific applications, most often telecommunications. For voice signals, such as on your phone, you do not need to use high-fidelity audio codecs, as the reproduction of a signal with a limited audio range is more suitable in this case. But for music playback, a high-quality audio signal is certainly preferable.

If you dig deeper, you will find that different audio codecs serve different purposes in processing the original analog signal. For example, an audio codec like PCM is a lossless compression algorithm. This means that the signal is reproduced in digital form without losing a single bit of original information. Other audio codecs, such as AAC and MP3, compress audio with some loss.

Compression reduces the bits of the original content and therefore reduces the file size. If you are listening to songs on a mobile device, you can be sure that these files have been compressed to take up less space. And that is why you can save a large number of music files on your device, but their quality will differ from optimal.

Audio codecs for Epiphan Pearl and Pearl-2
Of course, it is impossible to tell in detail all the characteristics of audio codecs in one article, but it can still help to clarify some of the nuances in choosing the correct audio codec for live streaming or recording using Epiphan Pearl or Pearl- 2 .

There are 3 audio codecs available:

-PCM – Uncompressed audio codec, which may be the best option if you plan to record shows for further editing and if you are not limited by network bandwidth.

-AAC: audio codec with compression algorithm best suited for live streaming or content recording with immediate playback on media players or for uploading to the Internet. Experts believe that AAC plays better audio than MP3 with the same audio bit rate. As a rule, the newer codecs reproduce the analog signal better than their predecessors, you can trust the experts on this.

-MP3: a fairly old, but still very popular audio code compression algorithm, also suitable for live streaming or recording content with immediate playback on media players or uploading to the Internet.
Choosing the correct audio codec is important when setting up live streaming or recording with the Epiphan Pearl or Pearl-2. Sample rate and audio oversampling effects are other important parameters for improving sound quality.