uncompressed audio formats example Archives

THE MOST COMMON FORMATS FOR MUSIC AND OTHER AUDIO FILES, AND HOW THEY ARE RELATED TO EACH OTHER

Free Download Mp4Gain

THE MOST COMMON FORMATS FOR MUSIC AND OTHER AUDIO FILES, AND HOW THEY ARE RELATED TO EACH OTHER

And for the direct competitors of the universal MP3 format, they can count on a lot today.

Due to continuing inconsistencies in home storage of the WAV format, it was eventually discontinued. But for professional studios, he says, the basics of the job. Especially when recording live vocals or instruments. Just convert the recorded material from WAV to MP3 at the final stage.

music format

However, music can be represented in some other popular formats nowadays. For example, many times (especially the Internet) they use these data types like OGG, AIFF, AMR, etc. But the real competitor of MP3 has become the newest and best audio FLAC etc. Of course, for MP3 you can convert all parameters to the maximum, but the playback quality of FLAC represents much higher. Also, it is a single file and the separation occurs directly on the track due to the player or startup software. In other words, listeners see each track individually, but can switch between playback tracks. For the MP3 format, this also seems possible to merge multiple tracks through it, thus creating a single file. But here it is in this version fast switching between tracks will not be possible (normal fast forward should be used, that’s all).

However, not everything is bad. The fact that music or audiobooks are all popular formats today allows them to be easily converted, even keeping the original parameters of the audio material. Based on this, and for sound processing and conversion and audio editors, almost all programs call converters. Any program of this type (MP3 editor or converter) detects the original and final type of audio files, is unambiguous and can produce direct and reverse transformations. Let’s explain this specific example.

WAVE THEORY AUDIO EDITOR FOR MP3 FILES
Many types of software are used in audio processing today. First, look at the narrow application of so-called audio editors. The most prominent representatives of these can be called giants Sony Audio Forge, Sintrillium Cool Editing Pro, which was later acquired by Adobe and renamed Audition, Acoustica Mixcraft, ACID Pro and many others.

mp3 editor

The principle on which they operate is that, for convenience, all MP3 audio programs have a typical waveform, as originally used for WAV files. This method determines the appearance and opportunity enough to edit any type of conventional audio material in WAV format. Other than that, the fact that you can do basic copy, cut, paste, etc. E., it’s just a matter of getting the frequency characteristics and bitrate changes, not to mention using a lot of extra effects that plug into VSTs via DirectX or a generic host bridge studio thing.

In its simplest form, the conversion can be done using the standard file menu, which contains the line “Save As…” (Save As…) or the export function present in MP3 format. Thus, all the process is reduced to just the final selection of the format (MP3 here as an example) and activation of the recording mode. In this case the conversion will be done automatically saving the current configuration parameters and the frequency characteristics. I don’t like the original version? ?Nothing is easier than changing the format to MP3, pre-specified with higher settings. However, one thing needs to be considered here: if the raw material is of such poor quality that special remediation or even professional tools will not work for audio it is necessary to use Repairs here, the intervention of various filters, etc. D. For the layman, it will cause great difficulties.

As is clear, there is absolutely no difference between the audiobooks we are dealing with: MP3, music or just recorded voice or noise. By the way, audiobooks are supposed to have a much lower sound quality by default. This is understandable, since the file has to take up minimal space and, in general, the perceived sound characteristics of speech are not that important. Finally, is this a professional recording of a particular set of albums?

However, if you use some standard operations, even without specific knowledge, it’s fine to achieve good results, especially since there are such built-in templates, based on any application for specific operations. Of course, it will be very difficult for the first time to achieve a perfect sound, but if you study the plan and understand how it works, it will work like clockwork, and as a result, it will take a lot of time.

Free Download Mp4Gain

Mp4Gain Main Window

Mp4Gain Features

Free Download Mp4Gain

Compressed audio formats

Compressed audio formats

Compressed Audio File Formats

Understanding compressed audio formats
The digital age dictates its own laws, according to which, in particular, audio and video information is more convenient to store and transmit in compressed form. Let’s briefly discuss the principle of sound compression.

Compressed Audio file formats

As you know, the music we listen to consists of a set of signals, each of which has its own characteristics, including loudness. The human auditory system is designed so that we do not distinguish or misdirect a weak (low) signal from the background of a strong (strong) signal. This principle forms the basis of modern means of compression (compression) of audio data.

If we imagine that a signal of a certain length is divided into many parts, and each part is processed in such a way that a weaker signal, which is difficult to distinguish from a strong one, falls under the knife and a stronger signal remains, then this will be a rough model of audio signal compression. … Consequently, the level of data compression will depend on how many parts (samples) the original file will be divided into and how many weak signals from each individual sample will be removed (what the bit rate will be: the number of bits in a sample of a specified duration).

The first versions of codecs for data compression acted quite crudely: they just cut off a weak signal and did not take into account the type of music, therefore, rather energetic music, without special nuances, in a compressed form does not it sounded worse than the original, whereas more complex classical and acoustic music simply lost all color and depth.

As a result of this, a transition to a more intelligent compression algorithm, with a variable bit rate, was made. Depending on the musical texture, that is, the ratio of weak and strong signals, the codec changes the amount of weak signals cut, so that we hear a more believable sound.

Obviously, with a higher sample rate (sampling) of 44.1-48.0 KHz and a higher bit rate (160-192 Kbps), we will get a sound more consistent with the original than with a sample rate 22 KHz and 64 Kbps bit rate. However, the size of the final compressed file is directly proportional to the selected sample rate and bit rate, and this is what people who distribute music in the form of compressed (compressed).

It should also be remembered that most algorithms cut the upper part of the audible range as well, starting at around 15 kHz.

There are currently several original compression algorithms, most of which are compatible with Linux.

Ogg Vorbis
Ogg Vorbis is a completely open audio format that allows you to store and transmit audio information with high sound quality (44.1-48.0 kHz sample rate, 16+ bits, polyphony (multi-channel audio)) and bit rates ranging from 16 to 512 kbps per channel. The number of channels processed can be as high as 255. This allows Vorbis to be on par with MPEG-4 (AAC and TwinVQ), WMA and PAC audio, and clearly superior to MPEG-1 Layer 3 (MP3) audio. .

Ogg Vorbis is also a streaming format, allowing it to be used, for example, for Internet broadcasts, especially since this format is compatible with Icecast. The characteristics of the codec algorithm allow you to get the final file smaller than MP3 files of similar quality.

For the reproduction the console program ogg123 is used, to encode – oggenc; both have graphic housings. More details on both are in the following sections.

MP3
MP3 or MPEG-1 audio layer 3 is by far the most popular format for storing and transmitting compressed data. This format was developed by the Frauenhofer Institut, Germany. However, despite the ubiquity of the format, it should not be forgotten that the patent for MP3 encoding and decoding algorithms belongs to a single company, so the end user at any time may find themselves in a very disadvantageous environment, such as It has already happened with the developers of free MP3 data compression tools …. You can get details about the license conditions on the developers website.

WMA
The WMA format is a proprietary product of Microsoft. It failed to occupy a market segment comparable to MP3, but it has some popularity despite serious security concerns identified. At the moment, only the universal MPlayer player can play WMA files. There are no free data compression tools for this algorithm and its appearance is unlikely.

Uncompressed audio formats

Below, we list the various types of uncompressed linear audio formats. The first three are the ones that have become the main ones supported by most of today’s professional audio applications and are: audio swap file format :

-AIF, AIFF The audio exchange file format is of Apple (1985) origin and allows the storage of mono or multichannel samples of 8 or 16 bits and various sampling rates. Being a format designed to be portable, it can also be easily converted (as we will see, Microsoft RIFF is similar) and is therefore often used. Extensions RIFF WAVE :

-WAV Developed by Microsoft and IBM in 1992, the RIFF WAVE (whose full name is Resource Sharing File Format Waveform Audio Format) follows the specifications of the more general rich information file format. It was introduced starting with Windows 3.1 and quickly became the most popular format for PC applications. WAV files support various types of sampling: they are single or multi-channel, 8 or 16 bits at different sampling rates with various encoding systems, even if the most common are PCM and ADPCM. Extensions TRANSMISSION WAVE FORMAT (BWF) :

-WAV It is an extension of the popular WAVE format and was created by the European Broadcasting Union (EBU) in 1997 and updated in 2001 and 2003. The purpose of this format is to add to the normal .WAV metadata format to facilitate the exchange of data between different platforms and different audio applications, allowing the files thus encoded to identify themselves autonomously and allow synchronization with other recordings. Since the only difference from “normal” WAVE files is in the extended information written in the file header area, the two formats are absolutely compatible and a particular player is not required for playback. In order to overcome the limitation on the maximum size of the WAVE file (2 Gb) in 2006, it was specified as an extension of the BWF l! RF-64, SD2 Sound Designer II Format SD2f file extension The Sound Designer 2 format is proprietary to Digidesign and is the evolution of the original Sound Designer 1 format, unlike which it structures the data so that all audio samples are stored in the file’s data fork and all parameters in place . resource holder. This is extremely convenient in files where the data fork can become hundreds of Mb and more, because it is possible to modify, add, cut the parameters of the audio file without having to modify the sample data, a feature that saves a lot of time, especially at a time when computers and hard drives were vastly smaller in capacity and slower in data processing and writing. It can also be monophonic or multichannel (interleaved), Sampling depth of up to 24 bits at different sampling frequencies of up to 192 KHz, as in the case of Pro Tools HD. It should be noted that in multitrack applications such as Pro Tools, the standard professional market software created for a DSP based card system, each track is recorded in a separate mono file, even for stereo or multi-channel tracks. In the case of a stereo track, the “split stereo” file is used, which consists of two monophonic files with the same name but with the two suffixes .L and .R (or in previous versions (L) and (R) ), that remain physically separated but that the application treats as a single stereophonic file, operating all the editing operations on both files simultaneously in perfect phase coherence; In native applications (ie fully dependent on CPU processing power), all those in practice outside of Pro Tools software (not Pro Tools LE or Pro Tools M-Powered who are also “native” even if they depend of specific Digidesign software or M-Audio production hardware), instead, the use of “interleaved” files is generalized, that is, stereo or multichannel files in which all the channels are stored in a single file. The stereo track is represented by a single file containing the two channels, which are “written” in blocks (first a number n of blocks from the left channel followed by an equal number n of blocks from the right channel, and so on). This type of file, created to dominate applications, used in multitrack applications still generates a bit of confusion, because although it is more practical (in theory) to be used within the native application, it needs to become a “split” file when imported into Pro Tools.

What are the digital audio formats?

PCM, Wav, Aiff. Compression. Mp3, Ogg, Wma.

Working with digital audio is almost a chore for puzzle specialists. Since audio is saved on the computer and all computer files have extensions, we have to interpret each acronym and abbreviation.

The extension is the end of the file after the name and period. It is used to know what type of file it is, whether it is a text, a video or an audio. There are many extensions and they are all sure to sound familiar to you: WAV, RM, MP3, WMA, OGG … Let’s play, then, to decipher puzzles and see what each of these acronyms means.

UNCOMPRESSED DIGITAL AUDIO FILES

.PCM

It is not a file type or format, but a technique of transforming analog to digital audio without any compression. (1) Therefore, we do not see audios with the pcm extension. We work with PCM when digitizing, but we always keep files with one of these extensions:

.WAV: (Wave, wave in English)

It is the most widely used uncompressed digital audio format. It belongs to Microsoft / IBM.

.AIFF: (Audio Interchange File Format)

It is similar to WAV but for Apple Macintosh or MAC computers.

.CDA

: These are the audio tracks recorded on Compact Disc that also use the PCM system.

All uncompressed files are large. Approximately 10 megabytes for every minute of audio. These are the formats used to store audio at a professional level since the quality is very good. But when we don’t need that much quality and we’re short on space, it’s time to use file compression.

AUDIO COMPRESSION

Compressing is reducing and whenever we reduce we lose something. The same is true for digital audio. The latest advances have allowed compression to be done with the least possible loss of quality, but there always are. Against that, much has been gained in reducing the size of the files.

While a 4-minute audio in WAV format takes approximately 40 megabytes, that same audio, compressed to MP3, can reduce its weight to 4 megabytes, 10 times less. And apparently, they sound the same. (2)

SAVE WITHOUT COMPRESSING

When working in production, it is always recorded in WAV, without compression. In that same way it is edited and mixed. If the final result of the edition is an audio to be uploaded on the Web or saved on the hard drive of a computer, we can compress it to mp3 but with a quality of no less than 160 kbps.

If, on the contrary, the production has as its final destination to be recorded on a CD, never compress, always leave the audio in WAV and burn it that way on the CD.

1. How does compression work?

It is not about wrinkling or crushing the audio. Most audio compression systems take advantage of a “defect” in our ears to reduce file size. It is called masking.

Masking is a property of the human ear that prevents it from distinguishing two frequencies close together within the same range, one masking the other. For example, if a sound with a frequency of 12 Khz and another of 12.2 Khz sounds at the same time in a song, we could remove one of the two without being noticed when listening to it.

In this way, the compressor “subtracts” the masked frequencies, which reduces the number of bytes. And fewer bytes in computing translates into smaller files, but not shorter. The song, when compressed, lasts as long as it is uncompressed.

2. Quality of compressed files

We saw in the previous question that digital audio has two parameters: the sampling frequency (the optimum is 44.1 Khz.) And the resolution or size of each sample (8 or 16 bits). By compressing, we add a third parameter to these two, the bitrate. It is the amount of kilobytes per second (kbps) and refers to the quality of the compression.

• A lower number of Kbps, more compression, smaller file size, but lower quality.

• A higher number of Kbps, less compression, larger file size and more quality.

A compressed audio at 128 Kbps has a higher compression level than a 256 Kbps one. That means that 128 is a smaller file and less quality than 256. Although you must have a cat’s ear to distinguish between both!

VARIABLE OR CONSTANT BIT

Some files have a constant bit rate per second (CBR Constant Bit Rate) and others have a variable one (VBR Variable Bit Rate). The constant is always the same for all audio, for example 128 kilobytes per second. In the variable method, what the compressor does is use more bits when there are parts of the audio where there are more frequencies and it cannot mask all of them.

COMPRESSED FILE FORMATS

Mp3 (MPEG-1 Audio Layer 3)

It achieves high compressions without much loss, although it all depends on the quality of the compression we use. 128 Kbps and below is not recommended.

Although mp3 is the most widely used compression standard, especially for audio on Web pages, the great drawback is its patent. So any player or editing software that wants to use it has to pay for it.

.OGG (Vorbis)

As a result of this patent, the Xiph.org Foundation developed in 2002 a completely free codec (5) for audio compression. Similar in characteristics to mp3, it is beginning to be used a lot on the Web and in some players since manufacturers do not have to pay the costs of the patent. At this point, it is difficult to completely replace the mp3 but it is eating up a lot of ground.

.AAC (Advanced Audio Coding)

The compression level is higher than mp3 (MPEG-1) without major loss of quality. AAC is one of the codecs used in the new MPEG-4 compression standard. This audio format is used in players like the iPod and in some of the new digital radio systems. AAC is shaping up to be the successor to the mp3.

.RAM (also RM or RA)

They are the files of the Real Network company for audio. The problem is that its reproduction and edition is very limited to software from the same company and few others.

.WMA (Windows Media Audio)

It is Windows’ bet on compressed formats. It is like a WAV, but smaller and less quality. While mp3 and ogg files are played by almost all players and editors, the same is not the case with wma files, so it is rarely used.

. AA3 (ATRAC – Adaptive Transform Acoustic Coding)

Format invented by Sony. It is the one used by minidisc recorder-players.