Audio codec technology

The development of voice codec technology is still to a certain extent, but the audio codec technology has been evolving (see Figure 1). For example, the trend towards more surround sound channels is a trend. One of the biggest technological trends is the addition of technologies for simulating multi-channel audio in stereo systems to recreate the "liveness" of a special live concert venue. So the challenge of completing all these processes arises because you can no longer use a DSP engine that relies on a high-power power supply in an AV receiver or DVD player to implement it, but must use a battery-powered DSP engine in a mobile phone or PDA to fulfill. All of the above factors bring an interesting story to two different but fused application areas (personal audio and home theater).

Personal audio

Personal audio

Randy Cole, chief technology officer of Texas Instruments ’portable audio and infotainment business, pointed out that the most common audio codec in the personal audio field is MP3 (see Figure 2). MP3 was once limited to PCs and portable multimedia players, but it is also now everywhere in the mobile phone industry, the reason is that mobile phone manufacturers are constantly chasing new features to lure end users to replace products every six months .

MP3 is a standard published by ISO. It is the third in a series of MPEG-1 encoding and decoding standards developed by MPEG. MPEG-1 has three layers, and each layer includes the previous layer. Thus, layer 3 implements layers 1 and 2. For this reason, MPEG-1 and layer 3 are known as MP3, which provides a bandwidth and data rate suitable for portable applications, although lower is sufficient.

In the past 10 years, MP3 has been the dominant audio codec for downloading music. But Apple ’s iPod broke this convention in a new MPEG development project called Advanced Audio Encoder (AAC). MPEG began to develop AAC standards in the mid-1990s as part of the next-generation MPEG-2 research and realized the best new design concepts developed by AT & T, Dolby, Fraunhof and Sony. At first, it was intended to be backward compatible with MP3, but this goal could not be achieved.

Therefore, because the entire industry is deeply involved in MP3, no company is willing to allocate resources to produce a large number of audio programs for the new program of AAC, so it has declined. In other words, until Apple chose the MPEG-4 version of AAC for the iPod, the technology was in a state of decline. (The next MPEG working group established after MPEG-2 is MPEG-4, which surpassed MPEG-3. The MPEG-4AAC used by Apple is an enhanced version of MPEG-2AAC, with a slightly lower data rate and improved quality. )

Other proprietary encoders exist outside the scope of MP3 and AAC. They have some penetration in the field of PCs and personal multimedia devices, but they are not very important in the mobile phone industry, because mobile phone manufacturers prefer standard encoders and their fixed royalties. A well-known proprietary encoder is WindowsMediaAudio (WMA). It is mainly used for PC, and competes with MP3 and AAC in it. However, in terms of data rate, it is very flexible (with appropriate quality differences from low to high). At present, there is also WMA-Pro, a multi-channel version of WMA, and Microsoft also released a lower loss WMA in June 2005.

The other important proprietary encoder is DolbyDigital, also known as AC3. This encoder is used for DVDs and (US) digital TV. Until recently, its operating rate was too high for the Internet and mobile phones. However, what needs to be remedied is a new version that reduces the data rate even lower.

According to MohsinImTIaz, the marketing manager of TI ’s high-performance audio business, the main codecs in the home theater are Dolby and DTS. DolbyDigitalPlus, released by Dolby, targets the high-definition DVD and broadcast markets. However, there is a certain crossover between portable standards such as MP3, AAC, and WMA. For the next generation of DVD, Microsoft is pushing WMA.

Parsing a codec

Let's break down a codec. In order to make the whole thing clear, we look at a paper submitted at the annual meeting of the Audio Engineering Society in October 2004, which describes DolbyDigitalPlus technology.

The paper says that the new DolbyDigitalPlus codec is based on an earlier version of DolbyDigital, also called AC-3. DolbyDigitalPlus or enhanced AC-3 (E-AC-3) retains the metadata carrier, filter library and frame structure. Current data rates range from 32Kb / s to 6.144Mb / s. Under the conditions of sampling rate 32KHz and six modules conversion frame, the resolution of data rate control can reach 1/3 bits per second. (The resolution of the data rate is proportional to the sampling rate and inversely proportional to the frame size.)

E-AC-3 retains AC-3's six 256 coefficient conversion frame structure, but it allows shorter frames that contain one, two, and three 256 coefficient conversion modules. As a result, audio transmission can be performed at a rate higher than 6? 0Kb / s, which is suitable for some DVDs that limit the amount of data per frame.

E-AC-3 can support the current 5.1, 6.1 or 7.1 channel, and then to the 13.1 channel of the cinema. The main audio program bit stream plus up to eight additional sub-streams are multiplexed into a single E-AC-3 bit stream. Channel replacement eliminates coding errors caused by matrix subtraction. Compared with AC-3, E-AC-3 can transmit seven independent bit streams.

The coding efficiency can also be improved by a new filter library, better quantization, enhanced channel coupling, spectral spreading, and a technique called "transient pre-noise processing".

When audio with stable characteristics appears, the filter bank adds a secondary DCT after the existing AC-3 filter bank. This converts the six 256-coefficient conversion modules into a single 1536-coefficient hybrid conversion module, and improves the frequency resolution. This increased frequency resolution combined with the six-dimensional vector component (VQ) and the gain adaptive component (GAQ) can improve the coding efficiency of "difficult to encode" signals, such as clarinet and harpsichord.

VQ is used in frequency band regions that require lower accuracy. When higher accuracy components are needed, GAQ is more efficient. In addition, through the coupling of the channel and the phase preservation, the coding efficiency can be improved. Where AC-3 uses a high-frequency single-synthesis channel as the high-frequency portion of each channel, adding phase information and the spectral amplitude processing controlled by the encoder can make this high-frequency single-synthesis channel handle lower frequencies, thereby reducing It increases the effective coding bandwidth and increases the coding efficiency.

Spectral spreading replaces the frequency conversion coefficients of higher layers with lower spectrum segments up-converted in the frequency domain. The spectral characteristics of the converted spectrum segment are matched with the original form through the spectral modulation of the conversion coefficient.

To improve audio quality at low data rates, E-AC-3 uses transient pre-noise processing technology. This post-decoding process minimizes the pre-noise error. Its approach is to use a time-scale synthesis technique that shortens the duration of the pre-noise, thereby reducing the audibility of transient disturbances. The metadata calculated by the encoder and sent in the E-AC-3 bit stream provides the parameters required for the post-decoding process and time-scale synthesis processing, which uses auditory situation analysis technology.

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