Audio codecs are one of the fundamental cores of modern media systems. Without an audio codec, there would be no current digital broadcasts, streaming services, and music distribution. The first and most mainstream MPEG audio codec was mp3, which was released in 1998. Since then, Fraunhofer IIS and other ISO-MPEG member participants have developed and developed multiple audio codecs. Each MPEG audio codec has or will change the way we consume media. This article introduces the MPEG audio codec and its applications, and shows Fraunhofer IIS, the most successful creator of modern audio coding solutions.
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MPEG L3: mp3
Mp3 has revolutionized the music industry and has changed the way consumers buy and enjoy music. Mp3 is still the main format for music distribution, because mp3 files can be played anytime, anywhere on any device. The mp3 technology was developed in the late 1980s. In 1995, with the birth of a file suffixed with ".mp3", the technology reached its peak. In the same year, Fraunhofer IIS introduced the first hardware prototype of the mp3 player. The file suffix mp3 quickly became the replacement for the "MPEG Layer 3" standard name, but it was not until three years later, in 1998, the first mp3 player was released.
Mp3 is a perceptual audio codec based on the perceptual model of the human auditory system. These models describe the elements of the audio signal that the human ear can perceive and are not aware of, regardless of whether the listener's ear has been trained. By analyzing the audio signal, mp3 and other perceptual audio codecs confirm the fact that the sound quality indicators can be prioritized by the human ear's perception and finely represented in the final audio file. Therefore, if the bit rate (ie at least 192 kbps) is chosen properly, the listener will not be able to discern the difference between the mp3 file and the source file.
Not only is mp3 based on a perceptual model, but most of the current MPEG series of audio codecs are also wise to use the human auditory system to reduce data rates and file sizes. The audio codec of the AAC series is no exception.
AAC series
AAC-LC
MPEG has developed another audio codec before the market adopted mp3 on a large scale. The goal is to achieve the same high quality audio quality as mp3 while significantly reducing the data rate. Since then, the R&D prelude has been opened, from the AAC in 1994 to the extended HE-AAC in 2012. The entire codec series is prologue.
In 1994, the first new AAC codec was developed according to the MPEG-2 format and named Advanced Audio Coding (AAC). Based on the experience of mp3 and other codec patents, major players such as AT&T, Dolby, Fraunhofer IIS, and Sony have designed a state-of-the-art new audio codec from scratch. Extend the MPEG-2 AAC codec to MPEG-4 by adding tools such as Perceptual Noise Shaping (PNS), Spectral Band Replication (SBR), and Parametric Stereo (PS) standard.
The basic MPEG-4 AAC configuration is called AAC-LC (low complexity) configuration. It provides "crystal" audio quality. In the audio coding domain, although the encoded signal of "crystal" audio is mathematically different from the source file, even a hearing expert with a "golden ear" cannot distinguish the difference from the source file. Therefore, AAC-LC can meet the highest audio quality requirements of broadcasters. The stereo AAC-LC bit rate is typically 128-192 kbps, and the 5.1 multi-channel AAC-LC bit rate is 320 kbps. Both AACs are encoded in vertical channels. AAC-LC is one of the most flexible audio codecs available today, with sample rates from 8 kHz to 192 kHz, bit rates up to 256 kbps per channel, and support for 48 channels. The most famous application for this configuration is Apple iTunes, which has been used in the Japanese ISDB digital TV standard.
HE-AAC and HE-AACv2
MPEG-4 "High Efficiency Profile (HE-AAC)" combines MPEG-4 AAC-LC and Spectral Band Replication (SBR) tools to further reduce overall bit rate while maintaining excellent audio quality. When the bit rate of the stereo signal is lower than 128 kbps, the HE-AAC has a bit rate reduction of 30% compared to the AAC-LC of the same audio quality. For HE-AAC, the low audio spectrum is encoded using AAC-LC and the high spectrum is encoded by the SBR tool. Spectral replication is a parametric method that recreates the entire audio spectrum of a signal using the height of the spectrum. To further reduce the bit rate, the AAC-LC code uses a 50% sample of the total signal for low frequency encoding. The typical data rate used for HE-AAC stereo is 48-64 kbps, and the typical data rate for HE-AAC 5.1 multi-channel is 160 kbps. Like the AAC-LC, the HE-AAC supports sample rates from 8 to 192 kHz, up to 48 channels, and audio-specific metadata.
The "Efficient AAC v2 Configuration (HE-AACv2)" adds a Parametric Sound (PS) tool to HE-AAC. The HE-AACv2 applies parameters for stereo signal encoding and further reduces the bit rate. Instead of transmitting two channels, the parametric sound encoder extracts parameters from the stereo signal, reconstructs the stereo signal on the decoder side, and then generates a HE-AAC encoded mono mix. The parameter data and the spectral data are transmitted in the auxiliary data field of the AAC bitstream. The decoder decodes the mono signal and the parametric decoder reconstructs the stereo. For stereo, a mono signal encoded with HE-AAC with parametric data transmission is more efficient than a two-channel HE-AAC encoded signal. For stereo signals, HE-AACv2 typically has a bit rate of 24 to 32 kbps.
Currently, AAC and HE-AAC are widely used. Especially in Internet applications, AAC and HE-AAC are the main audio codecs outside of mp3.
HE-AACv2 is widely used in state-of-the-art television broadcasting systems. It is part of the DVB toolbox or has recently introduced a second-generation terrestrial TV-specific codec for use in Spain, the United Kingdom, France, Ireland, Sweden, Austria, Italy, Denmark, Finland and Norway. In most countries in Brazil and South America, HE-AAC is the only designated audio codec for terrestrial television broadcasting. In addition, HE-AAC is also a designated component in smart TVs. For example, it is the designated codec for the European Hybrid Broadcast Broadband TV (HbbTV) data service. Therefore, all HDTV receiver devices, such as TVs and set-top boxes currently sold in Europe and South America, support HE-AAC. All major broadcast encoder vendors have long deployed HE-AAC to their devices. Of course, HE-AACv2 supports all relevant broadcast metadata.
HE-AAC is the mainstream audio streaming media codec. All major streaming platforms support HE-AAC, including Flash, Silverlight, Windows Media Player, Winamp, and iTunes. HE-AAC is available in operating systems such as Mac OS X and Windows, as well as mobile systems such as iOS, Android, Windows Phone, Symbian, and BlackBerry.
Currently, mature http-adaptive streaming media systems such as Apple HLS, Microsoft Smooth Streaming, and Adobe Dynamic Streaming are also based on the AAC family of codecs.
HE-AACv2 is also an important part of the streaming media standard in the consumer electronics arena, playing an indispensable role in the electronics field of the Open IPTV Forum, ATIS, HbbTV and DLNA. Therefore, the codec is supported by almost all digital TVs, Blu-ray players , set-top boxes, and game consoles. The extensive support of HE-AACv2 makes it the best codec of choice for content providers. Therefore, most webcasts such as Pandora, Aupeo, Hulu, and BBC iPlayer are based on HE-AACv2.
MPEG Surround
MPEG Surround technology can be seen as an extension of the parametric stereo principle from stereo to multi-channel. Unlike parametric stereo tools, MPEG Surround is more scalable in terms of bit rate and quality. MPEG Surround can be combined with the AAC family of codecs to provide higher coding efficiency. Another advantage of MPEG Surround is its backward compatibility with stereo signals. The bitstream always contains the AAC encoded core stereo signal and MPEG Surround. The stereo decoder extracts the core stereo signal and decodes it, while the MPEG Surround-enabled decoder reconstructs the entire multi-channel audio signal. In this way, MPEG Surround can be used in a mixed-signal receiver with a cheap or traditional stereo receiver or multi-channel receiver without the need to play both stereo and multi-channel signals simultaneously.
Low latency audio codecs: AAC-LD, AAC-ELD and AAC-ELDv2
MPEG codecs are not only widely used in broadcast, streaming and music distribution, but also in communication applications. AAC Series communication codecs are especially popular in high-quality conferencing and video telephony systems because they help service providers and operators deliver full HD voice services. Full HD voice is the highest audio quality that a communication system can achieve. Traditional narrowband phones can only transmit audio bandwidths up to 3.5 kHz, while full HD voice systems can transmit all audio spectrum from 14 kHz to the human ear. In this way, a full HD voice call sounds as clear as a conversation with someone in the same room. The AAC Series Full HD speech codecs include Low Latency AAC (Alow-ATC), Enhanced Low Delay AAC (AAC-ELD), and Enhanced Low Latency AACv2 (AAC-ELDv2).
AAC-LD is the industry standard for high quality video conferencing, providing full bandwidth, low latency audio encoding. It has an algorithmic delay of only 20 milliseconds while providing good compression and high sound quality for all types of audio signals.
AAC-ELD is an enhanced version of AAC-LD that combines MPEG-4 AAC-LD and spectrum replication. AAC-ELD is also the best choice for all delay-sensitive applications that require full audio bandwidth at low data rates of 24 kbps.
AAC-LD and AAC-ELD are currently used in professional and consumer video conferencing applications. For example, Apple's FaceTime application is based on AAC-ELD.
AAC-ELDv2 is the latest extension of the superior AAC-ELD audio codec. AAC-ELDv2 combines the advantages of AAC-ELD optimized delay parameter multi-channel encoding. This method allows only one mono and other information to be transmitted instead of two separate channels.
Extended HE-AAC
The newest member of the AAC series is the extended HE-AAC. In early 2012, MPEG standardization was just completed. The extended HE-AAC significantly improves the audio quality of music and speech, especially at very low bit rates such as 8 kbps, and is compatible with HE-AAC streams. The new codec combines two previously separated universal audio and speech codes to provide the advantages of existing voice and music codecs. After adding a new set of encoding tools to the HE-AACv2 audio codec, the performance of the extended HE-AAC greatly exceeds the dedicated speech and general audio coding schemes, making up for the shortcomings of both, providing consistent high for all signal types. Quality audio.
Conclusion
Whether for entertainment or communications applications, MPEG audio codecs are used in all state of the art consumer electronics, IT and communications equipment. Since the beginning of the mp3 in the late 1990s, the development of MPEG audio codecs has never stopped. Mp3 and its successor AAC are well known in the consumer field. For other MPEG codecs running in the background, such as HE-AAC or AAC-ELD, only professionals know it, but most people use these in their daily lives. Codecs, for example, watch internet videos or make calls using Apple FaceTime. Fraunhofer IIS plays an important role in the development and marketing of each audio codec.
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