What Is Advanced Audio Coding?

AAC (Advanced Audio Coding), Chinese name: Advanced Audio Coding, appeared in 1997, based on MPEG-2 audio coding technology. Developed by Fraunhofer IIS, Dolby Laboratories, AT & T, Sony and other companies, the goal is to replace the MP3 format. In 2000, after the emergence of the MPEG-4 standard, AAC reintegrated its features, adding SBR technology and PS technology. In order to distinguish it from the traditional MPEG-2 AAC, it is also called MPEG-4 AAC.

aac

(Advanced Audio Coding)

AAC (Advanced Audio Coding), Chinese name: Advanced
Improved compression rate: higher sound quality can be obtained with a smaller file size;
Support multi-channel: can provide up to 48 full-range channels;
Higher resolution: Supports a maximum sampling frequency of 96KHz;
Improved decoding efficiency: fewer resources are consumed for decoding and playback;
Conclusions from Dolby Laboratories
128Kbps AAC stereo music is considered by experts to be difficult to detect the difference from the original uncompressed sound source;
The performance of AAC format at 96Kbps code rate exceeds the MP3 format of 128Kbps;
The same is 128Kbps, the sound quality of AAC format is obviously better than MP3;
AAC is the only one that can obtain "excellent" webcast format in all EBU audition test projects.
In general, AAC can be said to be an extremely comprehensive encoding method. On the one hand, the characteristics of multi-channel and high sampling rate make it very suitable for future DVD-Audio; on the other hand, the high sound quality at low bit rate makes It is also suitable for mobile communication, Internet telephony, online broadcasting and other fields. It is a truly versatile encoding method.
The key different from MP3
AAC was developed based on MP3, so there are some similarities between the two encoding systems. However, comparing the encoding flowcharts of the two, we will find that the AAC encoding process is more complicated.
Filter bank: mainly completes the time-frequency conversion of the signal. Thus, the spectral coefficients in the frequency domain are obtained.
Temporal Noise Shaping (TNS): This magical technique can trim the distribution of quantized noise in the time domain through prediction in the frequency domain. In some special speech and quantification of drastically changing signals, TNS technology has made great contributions to the improvement of sound quality!
Prediction: Prediction of audio signals can reduce the processing of repeated redundant signals and improve efficiency.
Quantization: AAC's quantization process uses two nested loops to perform repeated operations. With good control of quantization analysis, the bit rate can be used more efficiently.
Bit-stream format: In AAC, the transmission of information must be entropy coded to ensure as little redundancy as possible. In addition, AAC has a flexible bit stream structure, which further improves the coding efficiency.
Long Term Prediction (LTP): This is a tool unique to MPEG-4 AAC. It is used to reduce signal redundancy between two consecutive coded sound frames. It is very useful for processing low-bit-rate speech. effective.
Perceptual Noise Substitution (PNS): This is also a unique tool in MPEG-4 AAC. When the encoder finds a signal like noise, it does not quantize it, but ignores the past by marking it. When It is restored when decoding, which improves the efficiency.
AAC + is also called HE-AAC.
HE means "high efficiency". HE-AAC combines AAC and SBR technologies. SBR stands for Spectral Band Replication. The key of SBR is to provide full-bandwidth coding without generating unnecessary signals under low code streams. Traditionally, audio coding at low bitstreams means reduced bandwidth and reduced sampling rate (see MP3 FAQ # 7) or produces unpleasant noise signals. SBR solves the problem by letting the core code encode low-frequency signals, and the SBR decoder analyzes the low-frequency signals to generate high-frequency signals and some guidance signals retained in the bit stream (usually the code stream is extremely low, ~ 2 kbps). This is why no SBR decoder is used, so your bandwidth response (frequency response) will be seriously wasted. This is why it is called Spectral Band Replication, it just increases the audio bandwidth, not reconstruction.

IN OTHER LANGUAGES

Was this article helpful? Thanks for the feedback Thanks for the feedback

How can we help? How can we help?