I will name the format aas yak vimovlyati. How to open an .AAC file? And yak on rahunok LOSSLESS

We hope that we have helped you resolve the problem with the AAC file. If you don’t know where you can download a program from our list, click on the message (these are called programs) - You will find information about the place where you can download a safe installation version is required noi programm.

What else can cause problems?

There may be more reasons why you cannot open the AAC file (not just the number of related programs).
According to Pershe- The AAC file may not be linked correctly (inappropriately) due to the installation of an add-on for its maintenance. In this case, you need to change this connection yourself. To do this, press the right mouse button on the AAC file that you want to edit, press the option "Help with help" and then select from the list the program you installed. After such problems, problems with the AAC file are likely to arise.
In a different way- the file you want to open may just be damaged. In this case, it will be best to find a new version of it, or download it again from the same device (it is possible that for any drive in the previous session, downloading the AAC file has not ended and it cannot be opened correctly).

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If you have additional information about the extension of the AAC file, we will be happy for you to share it with members of our site. Complete the form and send us your information about the AAC file.

2009-09-30T20:52

2009-09-30T20:52

Audiophile's Software

The first ideas for using psychoacoustic masking to compress audio data date back to 1979. However, common audio coders began to become widespread in the mid-1990s, as personal computers became more demanding to produce compressed audio in real time, and the MPEG-1 Audio Layer 3 standard emerged. vidomy yak MP3. Audio formats with compression have become indispensable when transmitting sound over the Internet, providing “virtually clear” clarity to stereo sound (so that the encoded signal for most ears is not disturbed by the original) at bit rates higher than 128 kbit /s. You can learn about the basic principles of the MP3 format from the articles of K. Glasman (2...8/2005)

The development of data compression and psychoacoustics methods has gradually led to the MP3 standard becoming “close” for the implementation of new ideas in encoded audio. As a result, until 1997, the Fraunhofer Institute (Fraunhofer IIS), which created MP3 in the early 90s, as well as the companies Dolby, AT&T, Sony and Nokia, developed a new method of audio compression - Advanced Audio Coding (AAC), which increased to MPEG-2 and MPEG-4 standards. The main features of the MP3 standard are:

• support for a wider range of formats (up to 48 channels) and audio sampling frequencies (from 8 kHz to 96 kHz);
• More efficient and simpler filter bank: the hybrid MP3 filter bank has been replaced with the original MDCT (modified discrete cosine diversion);
• wider variations in the frequency-hour resolution of the filter bank - in all cases (in MP3 - in thirds) - led to an increase in the encoding of transients (transient processes) and stationary sections of the audio signal;
• more clearly encoded frequencies above 16 kHz;
• a larger encoding mode for stereo signals, which allows you to switch to the M/S (“joint stereo”) mode regardless of different frequency ranges;
• Adding capabilities to the standard to improve compression efficiency: time-domain noise shaping technology (TNS), hourly MDCT prediction (long term prediction), parametric stereo mode, perceptual noise substitution technology, update high frequency (SBR).

Due to these features, the AAC standard allows for greater flexibility and efficiency, and therefore a clearer sound encoding. As a result of the wide expansion of the MP3 format, the AAC standard has not yet regained the same popularity as MP3. Prote AAC is the main format in the popular online store iTunes Store, iPod, iTunes players, iPhone phones, PlayStation 3, Nintendo Wii game consoles and in DAB+/DRM digital radio.

Let's look at the main features of the AAC report.

Filter bank

Like other psychoacoustic audio coders, AAC operates in an advanced circuit. The input signal is passed through a bank of filters - conversion, which transfers the signal from the time-hour domain to the frequency-hour domain (similarly to spectrographs). In parallel, the psychoacoustic model analyzes the signal and determines the thresholds of psychoacoustic masking. Next, the spectral coefficients of the signal at the output of the filter bank are quantized so that the noise spectrum, as far as possible (as the bit rate allows), appears lower than the masking thresholds and is not sensitive. Quantization coefficients are compressed without wasting the output AAC file. Thus, the filter bank itself does not compress the signal, it is only necessary to transfer them to a form more suitable for compression.

The peculiarity of the skin bank of filters is its frequency separation, the number of frequency bands into which the spectrum of the signal is divided. For most filter banks, which are used to reduce sound, the number of filters can reach hundreds. This means that, through a combination of inconsistencies, such filter banks can last for about a few tens of milliseconds. If the spectral coefficients of the signal are quantized, then the quantization factor during the decoded signal is expanded over the hour throughout the entire filter bank window. In some episodes this can lead to an unusual effect, which is called pre-echo. It manifests itself when the quantization noise from the transient (a sharp burst of energy in the signal) expands over the hour by an fraction of the hour that is transmitted to the transient and becomes noticeable (Fig. 1). To change this effect, seal the filter banks with variable frequency-time separation. For example, MP3 changes the time limit of the filter bank between 26 and 9 ms. For stationary signals, the vicoristic windows are 26 ms long, which gives good frequency separation, and for transients, the vicoristic windows are 9 ms long, which reduces the front-cut effect (div. Fig. 1).

The AAC algorithm also varies the size of the MDCT window. In this case, the difference in the size of the windows is eight times: 6 and 48 ms (256 and 2048 times). This algorithm is designed to adapt to a wider range of signals and achieve short compression.

TNS technology - shaping amplitude-based noise

One of the problems with current psychoacoustic audio encoders is working with transients (transient processes in the audio signal). To ensure accurate coding, it is necessary to ensure that the quantization noise enters the masking threshold so that it remains in place for an hour. However, in practice it is important to satisfy the transition processes, because The quantization noise, which occurs during the encoding hour, expands over the hour when decoding for the entire duration of the MDCT window. This can cause significant noise disturbances due to the quantization of the time-masking thresholds.

The TNS (temporal noise shaping) technology of the AAC standard allows for wider noise quantization over the hour at the boundaries of the MDCT skin window. TNS technology is based on the similarity (frequency-hour dualism) of the amplitude of the signal and the spectrum that it is generated, as well as on the vicinity of linear transfer (LPC) by frequency when quantizing the spectrum.

It is good to know that for signals with a spectrum that is strongly biased towards white (for example, tonal signals), a vicarious linear transfer (LPC) in the clock region allows one to effectively “blind” the spectrum and encode it as such. and the signals are laid out on the transfer coefficient and are equally small for amplitude of propagation (residual). When decoded, the line transfer filter shapes the decoding spectrum consistent with the spectrum of the output signal.

In an AAC encoder, linear transmission is used in the most basic manner: for transmission of parts of the spectrum of the frequency region. The difference in MDCT output and transfer coefficients is quantized according to the masking thresholds (in traditional encoders the MDCT output coefficients are quantized). Line transfer coefficients are also written to the output file. When the signal is decoded, the line transfer filter, which is applied to the differential signal in the frequency domain (which includes quantization reduction), shapes the amplitude of the output signal (and quantization reduction) in the clock domain. Thus, the amplitude of the final quantization becomes close to the amplitude of the output signal (Fig. 2).

TNS technology reduces the effect of pre-moon and quantization on certain harmonic signals with a pulsed nature of sound production (language, wind instruments and stringed-bow instruments). In Fig. 2 The quantization corrections applied to the vocal signal by AAC and MP3 algorithms with the same bit rates will be updated. At the same time, due to the extreme reduction of the quantization cut-off (through the greater efficiency of AAC), the formation of the amplitude of the final quantization cut-off is avoided over an hour in line with the final output signal.

In the AAC standard, TNS technology can be limited to several frequencies in the spectrum, or switched on entirely.

SBR technology – upgrade of high frequencies

Reliable transmission over a wide frequency range is an important benefit for clear coding. However, the transmission of the cutaneous octave of the sound range once or twice increases the capacity of a traditional audio encoder to the bitrate. In order to reduce the bit rate and thereby preserve high frequencies in the material that is being encoded, the technology of custom synthesis of high frequencies SBR (spectral band replication) was created.

The technology relies on the fact that our hearing analyzes high frequencies with less accuracy, lower mids and low frequencies. To create the effect of the presence of high frequencies, it is necessary to accurately reconstruct the shape of the signal, but it is sufficient to renew certain psychoacoustic parameters of the signal at high frequencies. These basic parameters indicate the frequency-hour distribution of the (general) signal energy and the level of its tonality/noise level.

The idea behind the algorithm is this. When encoding, the high frequencies in the output audio signal are analyzed and their parameters are extracted: first of all, the amplitude is expressed in units (eight) of frequency ranges. High frequencies from the recording are removed and only the low and mid frequencies that are missing are encoded. In this case, the output file also contains a relatively small flow of information about the parameters of high-frequency input.

When opened, the signal of low and mid frequencies is decoded. Next (if present in the player) the SBR decoder starts working. The first step is the synthesis of a high-frequency signal through transposition (more precisely, frequency substitution) of the apparent mid-frequencies. Some stages of tonality/noise in the spectrum at mid and high frequencies are approximately the same, which results in a high-frequency signal with a plausible spectrum structure. On the other hand, the SBR decoder uses additional information about high frequencies to provide the required amplitude in the skin frequency mixture. The result is a signal whose high frequencies are synthesized from the mids, while preserving the sound of the output high frequencies.

SBR technology can be extended to a variety of advanced audio encoding techniques. For example, SBR combined with MP3 is called MP3 PRO, and SBR combined with AAC is called HE-AAC (high efficiency AAC). Basically, SBR is favored when encoding at relatively low bit rates: 64 kbps or lower. The technology allows you to significantly expand the frequency range of the audio signal with a minimal increase in bitrate (several kbit/s).

Parametric stereo technology

Transmission of a stereo signal requires the encoder to produce at least twice the higher bitrate, while transmitting a monaural signal. In this case, stereo channels can be encoded independently, and after M/S conversion. The S-channel often consumes less bitrate than the M-channel. This coding mode is also called joint stereo. With the AAC standard, this mode can be turned on and off by the encoder regardless of the skin frequency signal.

For efficient encoding of stereo signals at very low bit rates (16...32 kbit/s), the technology of parametric stereo encoding was developed. What this means is that the stereo signal is reduced to mono before encoding, but the output file is given a small stream (2...3 kbit/s) to contain information about the stereo panorama of the output stereo file. This flow contains (in a simple way) a specific panorama map for the frequency-hour area.

At the stage of decoding the extracted monophonic signal, frequency-dependent panning is established. It is possible to work simultaneously with decoding, setting equal amplitude multipliers to equal MDCT coefficients of the left and right channels.

Parametric stereo technology gives greater clarity about the output stereo sound at the cost of a slight increase in bitrate compared to encoded mono signals. However, it does not allow you to reach the surface of clear sound, since it is impossible to capture all the nuances with stereo panoramas, for example, phase disruption between stereo channels.

Parametric stereo technology is enabled up to the HE-AAC v2 standard.

PNS technology – noise generation

To further increase the efficiency of encoding noise signals, the AAC standard is equipped with PNS (perceptual noise substitution) technology for noise synthesis. It seems that our ear is more sensitive to the spectrum of the amplitude signal, less to the phase one. Therefore, instead of encoding MDCT coefficients for the output signal in noise areas, it is possible to transfer parameters to the noise: its intensity depends on the frequency and time.

This is how PNS technology works. During encoding, sections of the spectrum that are noisy are identified, and specific groups of MDCT coefficients are switched off during the encoding process. The frequency difference is designated as noise, and the hidden energy of the noise is memorized for it.

When decoding the frequency of noise, marked as noise, pseudo-prevalence MDCT coefficients with the necessary neutrality are presented. As a result of the selection of frequency ranges, noise is synthesized that is close to the sounds before the output noise.

Technology Long term prediction - hour forecast

Psychoacoustic encoding of tonal signals results in a higher local signal-to-noise ratio than the encoding of noise signals (for example, 20 dB and 6 dB in parallel). And this, in its turn, will require an increased bitrate. However, MDCT coefficients of tones are transmitted hourly. This arrangement allows you to use its storage capacity depending on the hour of change in bitrate.

The AAC standard has a Long term prediction mode, in which MDCT coefficients are additionally encoded within an hour using additional linear transfer. The term “long term” means that the transmission occurs behind the current outputs, and, according to the outputs, separated by the most significant period of the tone of this frequency.

Quantization and compression of MDCT coefficients

Similar to the MP3 standard, AAC uses nonlinear quantization of MDCT coefficients and their compression by the Huffman method. MDCT coefficients are quantized at a level of 0.75, which allows the quantization to be increased for strong signals and adjusted for weak signals between skin frequencies. In this way, there is an additional implicit shaping of the noise spectrum.

After quantization, the MDCT coefficients are compressed into another set of fixed Huffman tables. The AAC standard has a larger table than MP3, and greater possibilities for grouping factors. This leads to additional compression.

The vigor of the sound

When assessing the sound quality of an audio encoder, subjective tests are used. Listeners receive fragments of recordings compressed by different encoders, and they evaluate the purity of the sound of the skin fragment on a scale of 1 to 5. The final codec takes into account the sound quality that is equal to competitors at a given bit rate. go.

A fairly authoritative Internet resource that provides the results of such tests is the site http://www.rjamorim.com/test/ It presents testing of various codecs for impersonal bitrates. The results achieved by this method work well with other devices. We present a number of results for MP3 and AAC encoders, which help to equalize their brightness.

The most beautiful MP3 encoder is the cost-free Lame. However, most of the high-rate wines comply with the new compression standards. At high bitrates (over 128 kbps), the difference is small, and the leader is the Ogg Vorbis encoder.

At a bitrate of 64 kbps, AAC transmission is already acceptable. For the HE-AAC option, the algorithm rejects a score of 3.68. This is confirmed by Lame with a bitrate of 96 kbps and means that AAC is approximately 1.5 times superior to MP3. Lame's rating with a bitrate of 128 kbps is 4.29.

At a bitrate of 32 kbps, the AAC encoder from the company Nero plays seriously as well as MP3: ratings of 3.23 and 1.72 per hour. However, AAC slightly outperforms the MP3PRO format, giving a score of 3.08. This indicates that SBR technology effectively significantly improves the brightness at low bitrates.

Visnovki

In addition to new technologies in the AAC standard, this format is clearly superior to MPEG-1 Layer 3 (MP3), allowing for higher-quality sound at the same bitrates. Particularly strong winnings are avoided in the area of ​​low bit rates: 96 kbit/s and lower. This confirms the promise of the AAC format for digital radio broadcasting.

The popularity of AAC for distributing music on the Internet today is not equal to that of the MP3 format. Players can benefit from the short portability of MP3 over the powerful compression of AAC. A significant portion of the music archives on sites that offer all types of music are initially in MP3 format, and providers do not have access to compressed recordings. This means that there is no great sense in converting such recordings into the AAC format - the information is often already wasted. However, new players and online stores now support the AAC format, often with verification of the legality of the content (which is also true for merchants who do not want to include copied music among themselves).

Although promising, the AAC format is not the only clear audio compression format. At high bit rates (above 128 kbps), AAC is often supplied to Ogg Vorbis and Musepack format encoders. At the lowest bitrates (less than 32 kbps), AAC can be used by parametric audio encoders, including specialized encoders for compressing video. However, in the medium-low bitrate range, AAC tends to save the palm of the palm.

Oleksiy Lukin
Magazine "Sound Engineer" 2008 #1

To this day, the AAC format has still not achieved widespread audio expansion, but due to the low VIN parameters, it overcomes all types of audio compression that exist today, and therefore, our respect.

What's this?

Let's get this out of the way: AAC is a proprietary (patented) option for compressing an audio file. In this case, there is less wastage of capacity under the hour of encoding equal to MP3 for the same bitrate. In addition, the AAC format is a broad-based audio encoding algorithm that combines two main encoding principles by significantly reducing the amount of data required to transmit clear digital audio. This solution is recognized as one of the most effective implementations of cost-saving technology. The format encourages more immediate, even portable use. Please note that ringtones in AAC format can be added to the iTunes Store, and in this store music is presented, compressed inclusive of the specified solution. It is also necessary to say that the AAC format has initially emerged as a successor to MP3, which can give increased coding power. The decision began in 1997 as a new, 7th, part of the MPEG-2 family.

Robot principle

When encoded into this format, the following processes are completed: components that are not received are removed from the signal, the encoded audio signal is cleared of supernumerary content. After this, the data are processed according to the MDCT method according to their complexity. At the next stage, correction codes for various internal damages are added. If it is found, the signal is transmitted and saved.

All details

It is important that the AAC format has a sampling frequency between 8-96 kHz, as well as a number of channels in the segment 1-48. MP3 is a hybrid set of filters. At its core, AAC expands to Modified Discrete Cosine Reversal with a larger window size, reaching 2048 points.

Thus, AAC is much more suitable for encoding audio that contains a stream of complex impulses, as well as direct signals similar to MP3. The format is based on the creation of dynamic alternation at the end of MDCT blocks at the intervals of 2048-256 points. If there is a short-term or single change, a small “window” of 256 points will be fixed in order to achieve the shortest permission. When doing this, a 2048-point large window is used in order to maximize the efficiency of coding. AAC has a number of advantages over the standard MP3. Among them are the following: the implementation of a large number of audio channels (up to 48), the effectiveness of encoding a constant and variable bitrate, as well as sampling frequencies ranging from 8 Hz to 96 kHz (for MP3 this is shown Nickname set from 8 Hz to 48 kHz) And a larger mode called Joint stereo. The solution “AAC+” is a codec oriented towards low-bitrate work. With a combination of SBR and AAC LC, you can achieve better sound quality in the range of 32-48 kbps.

For obvious reasons, it is installed on the computer antivirus programs possible, perhaps scan all files onto your computer, as well as the other file. You can scan any file by right-clicking on the file and selecting the appropriate option to scan the file for viruses.

For example, on this little baby you can see file my-file.aac, then you need to right-click on this file, and select the option in the file menu "Go get help AVG". When you select this option, AVG Antivirus will open, which will scan the file for viruses.

Sometimes the decision may be canceled as a result incorrect software installation What could be related to the problem that occurred during the installation process? This can affect your operating system Link your AAC file to the correct software, falling into such a title "file extension association".

Sometimes it's simple reinstalled Apple iTunes We can solve your problem by correctly linking AAC to Apple iTunes. In other cases, problems with file associations may arise as a result filthy programming software security a retailer, and you may need to contact a retailer to obtain additional assistance.

Porada: Try updating Apple iTunes to the latest version to make sure the latest fixes and updates are installed.

This may seem obvious, but often The AAC file itself could be causing the problem. If you have received a file through an email attachment or have downloaded it from a website, and the process of downloading has been interrupted (for example, a connection to life or for other reasons), the file may be corrupted. If possible, try creating a new copy of the AAC file and try opening it again.

Carefully: A corrupted file can be caused by additional bugs from previous or existing faulty programs on your PC, so it is important that you regularly run antivirus updates on your computer.

What is your AAC file? related to hardware security on your computer, to open the file you may need update device drivers, related to these possessions.

Qia problem Depend on the types of multimedia files, which lies under the successful release of hardware security in the middle of the computer, for example, sound card or video card. For example, if you are trying to open an audio file but cannot open it, you may need to Update sound card drivers.

Porada: If you try to open an AAC file, you will be deleted notification about the cancellation, linked to the .SYS file, problem, it is clear that you can associated with faulty or outdated device drivers, what needs to be updated. This process can be made easier with the help of software for updating drivers, such as DriverDoc.

The Crocs didn't see the problem, and you are still having problems with AAC file permissions, which may be related to the number of available system resources. Some versions of AAC files may require significant amounts of resources (eg, memory/RAM, processing power) to run properly on your computer. This problem occurs often when you try to use old computer hardware and a new operating system at the same time.

This problem may arise if the computer needs to return from work, and parts of the operating system (and other services that run in the background) may You need a lot of resources to open an AAC file. Try closing all programs on your PC, first opening the Advanced Audio Coding File. By making the most of all available resources on your computer, you will have the best minds to try opening an AAC file.

Yakshcho vi vikonali all descriptions more details, and the AAC file does not open before, you may need to exit renovation of equipment. In most cases, probably with older versions of the device, the calculation intensity may be less than sufficient for most of the customer's benefits (as you do not rely on a richly resource-intensive robotic process sora, such as 3D rendering, financial/scientific modeling or intensive multimedia robot). In such a manner It is absolutely certain that your computer does not run out of unnecessary memory.(often called “RAM” or random access memory) is the file storage name.

Free Download AAC Codec for Windows 10/8/7 etc, to Decode/Encode AAC on PC

AAC Codecs Information about Windows 10/8/7 etc is available.

AAC codec can be used as a library/project or set of software tools for encoding or decoding AAC audio. AAC codes can manage both AAC encoding and decoding, while others can only decode AAC with purpose of playback or conversion. So, what AAC codecs can we download for Windows 10/8/7 etc? Check the list below.

Nero AAC Codec
Supported OS: Windows, Linux
Type: Encoder/Decoder
License: Free
Download Link: https://www.videohelp.com/software/Nero-AAC-Codec

Nero AAC Codec, developed and distributed by Nero AG, is an all-in-one AAC encoder and decoder supporting MPEG-4 AAC LC, HE-AAC and HE-AACv2 with up to 96kHz additional frequency and up to 6 channels (5.1 surround). This is very high because it can produce high-quality results in a small volume. This AAC codec gave this update on 18 February 2010 with version 1.5.4.0. Moreover, this is not a long version, you can finish on Windows 10, 8, 7 and lower versions.

Fraunhofer FDK AAC
Supported OS: Android, Windows, macOS, Linux
Type: Encoder/Decoder
License: Liberal
Download Link: https://github.com/mstorsjo/fdk-aac

Fraunhofer FDK AAC has been expanded to include AAC coding for Android and continues to expand to other platforms. This supports a wide range of AAC audio types, including MPEG-2/4 AAC LC, HE-AAC (AAC LC + SBR), HE-AACv2 (LC + SBR + PS) and AAC-LD/ELD. Audio files with sample rate ranging from 8kHz to 96kHz and up to 8 channels (7.1 surround) can be customized, vikorystyogo library. The source-code distribution Fraunhofer library is called fdk-aac and can be compiled with other programs for the HandBrake application. >>Further Reading:

FAAC
Supported OS: Windows, macOS, Linux
Type: Encoder (FAAC)/Decoder (FAAD2)
License: Proprietary (FAAC)/Free (GNU General Public License Version 2 or Later for FADD2)
Download Link: http://www.audiocoding.com/downloads.html

FAAC AAC codec includes FAAC encoder and FAAD2 decoder. Back-end recorders are mainly used to generate AAC (MPEG-2/4 AAC) files from other formats during front-end recording, which can be inserted into MPEG-2/4 AAC files with the support of audio objects LC, Main, LTP, S BR i PS. library (libfaad) includes a notebook in FAAD2 that can be used with other programs for AAC recording, for the AAC ACM Codec application.

It's amazing that the codecs that are unique to AAC audio and video on Windows are several comprehensive codec packs/libraries achieving the same purpose.

The Smartest AAC Audio Downloader and Extractor – WinX HD Video Converter Deluxe

Quickly download AAC/MP3 Music from SoundCloud, Audiomack and other Music Sites without moisture.
Convert local and online (for YouTube/Vevo music) video for AAC or other video formats such as MP3, WMA, FLAC, ALAC, M4R, OGG, DTS, etc.

Related External Source from Wiki:
Audio codec - A device or computer program capable of coding or decoding a digital data stream of audio...