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AAC: Lossy Encoding Is Getting Better – AAC Format Summary

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AAC: Lossy Encoding Is Getting Better – AAC Format Summary

AAC Music File Format

At the time of writing this article, the MP3 codec is over 23 years old. So as not to repeat myself with the article (its most recent version), which already describes the OGG Vorbis codecs (and again hello to the Xiph organization, this is also its development), MPC (Musepack), WMA (Windows Media Audio) and AAC, I will briefly describe the format here. AAC in terms of technologies that until recently were at the forefront of lossy coding.

In my humble opinion, AAC (Advanced Audio Codec) is one of the most advanced formats in the field of data encoding. I will describe the main features of this format, starting with the popular profiles that can be represented by a matryoshka (see the figure below):

– Low Complexity Advanced Audio Coding (LC-AAC)

Low decoding complexity is great for implementing a hardware codec; The hardware requirements for CPU and RAM are also low, which has gained a lot of popularity for this profile. It encodes the 96 kbps signal efficiently.

– High Efficiency Advanced Audio Coding (HE-AAC).

The HE-AAC profile is an extension of LC-AAC and is complemented by patented SBR (Spectral Band Replication, thick – “spectral repetition”) technology. It is spectral repetition technology that allows you to “preserve” high frequencies by encoding at low bit rates.

Why is “save” in quotes? Because the king is not real: the codec leaves room for additional information that is used by the codec synthesizer to restore the high frequencies, but since these frequencies are synthesized, that is, they are recreated by the codec, they are, in fact, a rough copy of the high frequencies that existed in the original file. … In practice, a signal encoded at 48 kbps will sound, for example, as mp3 at 98 kbps if it is supported by the decoder; otherwise, the file will simply be played without restoring the high frequencies and its bit rate will correspond to its mp3-like quality.

– High Efficiency Advanced Audio Coding Version 2 (HE-AACv2)

This profile is relatively young (described in 2006), it was created for a more efficient audio coding in low bandwidth conditions.
The second version of the profile is an extension of the first profile, the changes are in the addition of PS (Parametric Stereo) technology. The principle is somewhat similar to SBR technology: the codec also makes room for recovery information from the stereo base, sacrificing precision.

The operating conditions for this profile are the same as for the HE-AAC described above; The lack of profile support from the decoder will make the recording sound in mono.

– AAC-LD (advanced audio coding – low delay)

The AAC-LD profile has advanced coding algorithms to reduce delays (up to 20 ms);

– AAC-ELD (Advanced Audio Coding – Enhanced Low Delay)

This profile, which inherits all the capabilities of HE-AACv2 (analogous SBR and PS technologies are used, but designed for low latencies);

– AAC main profile

This profile was introduced as MPEG-2 AAC or HC-AAC (High Complexity Advanced Audio Coding). Not compatible with LC-AAC;

– AAC-LTP (Advanced Audio Coding – Long Term Prediction)

This profile is more complex and resource intensive (but also of higher quality) than all the others. It is also not compatible with LC-AAC.

That’s all I wanted to write about this codec. I put the greatest emphasis on the technologies that are used in various AAC profiles (which, by the way, generate a lot of abbreviations: AAC, LC-AAC, eAAC +, aacPlus, HE-AAC, etc.), as I will compare them with the from Opus, but the codec does its job: it is widely used in Internet radio, as well as in digital radio transmission technologies: DRM (Digital Radio Mondiale) and DAB (Digital Audio Broadcasting) (you can see these technologies here), YouTube , as an audio track for many videos in mp4, mkv, etc.

2. Introduction to Opus: description of the format

On December 21, 2017, Xiph announced the beta version of the Opus audio codec version 1.3. I will not go into important matters when I describe this codec, since such information is freely available (for example, here, here, and for those who know English, here and here). The release notes for this beta version can be found here. Here I will point out that this codec is an excellent candidate to replace other codecs. It has many advantages:

bit rate from 6 to 510 Kbit / s;
sampling frequency from 8 to 48 KHz;
support for constant bitrates (CBR) and variables (VBR);
support for narrowband and wideband audio;
support for voice and music;
support for stereo and mono;

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AAC: improved compression of audio files

AAC Format

The Advanced Audio Coding audio format, or AAC for short, offers very good sound quality even at low bit rates. While you need a bit rate of at least 192 kbit / s for CD quality with MP3, you can achieve the same quality with AAC from a bit rate of at least 128 kbit / s. As a result, AAC files are much smaller than MP3 files and are often used, for example, when streaming music or Internet radio.

AAC AUDIO FORMAT

AAC became known especially through iTunes. The music player developed by Apple preferably uses this format. One reason for this is that copy protection can be built into AAC, preventing uncontrolled transfer of music files.

Better than MP3

As with MP3, the AAC format removes sounds that are not perceptible to humans and therefore unnecessary. This applies both to inaudible frequency ranges and to redundant tones that arise, for example, by overlapping. Therefore, a low data rate does not necessarily mean a deterioration of the sound. A decisive advantage over MP3: AAC files have the same quality and a significantly smaller file size than MP3 files.

AAC has other advantages as well. AAC supports up to 48 channels, while MP3 only supports six. Also, the scanning range is significantly wider with AAC. It ranges from 8 to 96 kHz. AAC achieves good sound quality starting at 94 kbit / s. Bit rates of 192 kbit / s or 224 kbit / s correspond to almost lossless CD quality. By the way, AAC files mainly use the .mp4 file extension and are used by Apple iTunes, Quicktime, mobile phones, car radios, Nintendo and Playstation, among others. AAC is also used for Internet radio.

Compress AAC

If you want to compress a file according to AAC, you can use various software programs, so-called encoders, for this. Both free and paid software are available on the market, and there are few programs that work really well. Some examples are Nero or Liquifier.

Where does the AAC come from?

AAC was developed at the Fraunhofer Institute, with other companies such as Sony and Nokia also involved. The goal was to develop a standardized audio format that offered very good sound quality in the smallest possible size. AAC is a further development of the MPEG-2 standard. Later more AAC profiles were developed under MPEG4. If you want to use AAC commercially, you have to pay for a license, which is why most encoders are paid.

AAC comes with different profiles

AAC has different profiles that can be selected for compression. Typically the LC profile is used, which is suitable for medium to high data rates. Other profiles are the HE profile, the SSR profile and the LD profile. Due to the different profiles, in some players it may happen that data such as the artist or the title is not displayed correctly because the existing profile is not supported correctly.

Audio file format AAC

AAC files are designed to replace MP3 files. Lossy compression gives you better sound at the same bit rates. AAC files are standardized by ISO / IEC as part of the MPEG-2 and MPEG-4 file families (originally they were members of the MPEG-2 Part 7 filegroup). AAS files contain a higher number of sample rates (compared to MP3) and up to 48 channels. Significantly improved encoding efficiency, including an increased number of filter sets. Enhanced transient signal coding accuracy. Like MP3 files, AAS files suppress sound at frequencies that a person cannot hear. This way you can reduce the file size. Compared to MP3, AAS files are much smaller.

AAC Format

Technical details of AAC files

MPEG-2 Part 7 files are represented by three types of files: Low Complexity AAC-LC, Main Type (AAC Main), and Variable Rate Files (AAC-SSR). AAS files allow temporary noise generation, uneven sampling, and repeated generation of the bitstream format (for 16 stereo channels, 16 mono channels, 16 low-frequency channels, and 16 channels for single bitstream annotation). In 1999, the MPEG-2 Part-7 format was incorporated into the MPEG-4 Part-3 format, allowing the introduction of audio object types as well as constant noise replacement technology. The AAC format is currently described in ISO / IEC 14496-3. Lossy compression uses sound masking to remove unnecessary information while maintaining quality.

Aac format

Additional information about the AAC format

Until now, the AAC format has not yet achieved mass distribution in audio media. However, in a number of parameters it exceeds all existing types of audio compression, which means that it deserves our attention.

What is this?

Let’s start with the definition: AAC is the proprietary (proprietary) option for compressing an audio file. At the same time, the quality loss when encoding is less than with MP3 at the same bit rate. Furthermore, the AAC format is a wideband algorithm for encoding audio that uses two fundamental encoding principles to greatly reduce the amount of data required to transmit high-quality digital audio. This solution is recognized as one of the highest quality implemented using lossy compression technology. The format is compatible with most modern devices, including laptops. It should be noted that ringtones in AAC format can be purchased from the iTunes store and that this store presents music that is only compressed with this solution. It should also be said that the AAC format was originally created as a successor to MP3, which can offer improved encoding quality. The solution was published in 1997 as the new seventh part of the MPEG-2 family.

All the details

Interestingly, the AAC format has a sample rate in the range of 8 to 96 kHz, as well as the number of channels in the range of 1 to 48. MP3 uses a set of hybrid filters. AAC, in turn, refers to the modified discrete cosine transform with an expanded “window” size that reaches 2048 points.

Therefore, compared to MP3, AAC is much better suited for encoding audio with complex pulse stream as well as square wave signals. The format was given the ability to dynamically change MDCT block lengths within 2048-256 points. In the case of a short or simple movement, a small 256-point “window” is used to achieve better resolution. In this case, a default 2048-point “window” is used to maximize encoding efficiency. AAC has several advantages over regular MP3. Noteworthy is the implementation of a large number of audio channels (up to 48), significant coding efficiency at constant and variable bit rates, and sample rates from 8 Hz to 96 kHz (for MP3 from 8 Hz to 48 kHz). and a more flexible special mode called Joint Stereo. The solution is “AAS +”, a codec designed for a low bit rate. It is a combination of SBR and AAC