Principles of MPEG audio compression

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1 Principles of MPEG audio compression Principy komprese hudebního signálu metodou MPEG Petr Kubíček Abstract The article describes briefly audio data compression. Focus of the article is a MPEG standard, which define world wide compatibility of compressed data. There are descriptions of principles audio data compression (masking effects), compatibilities and compress ratios. Finally there are a lot of examples where are MPEG standards used. There is focus for real time applications which are working in the world at present time. INTRODUCTION TO AUDIO MPEG The article is based on standard ISO/IEC (ISO/IEC 11172:1993 Information technology-coding of moving pictures and associated audio for digital storage media at up to about 1,5Mbit/s). Actually it is standard for MPEG-1 compression. It consist from five parts: Systems, Video, Audio, Conformance testing, Software simulation. Next paragraphs are used only Part 3: Audio. In fact, there are several standards for MPEG compression. Later, in 1996, was set up other standard for MPEG2 standard. It is ISO/IEC 13818:1996 Information technology-generic coding of moving pictures and associated audio information. Figure 1 1: ISO standards of MPEG 1 Ing. Petr Kubíček, Department of Telecommunications, Brno University of Technology, Purkyňova 118, Brno, tel.: 05/ fax.: 05/

2 Figure 1 shows similarity of MPEG-1 and MPEG-2. More over MPEG-2 is built for low bit rates and for more audio channels compare to MPEG-1. Low bit rate is a demand by industry of electronics (for example radio connection). On the other hand high bit rate is caused with modern trends in house acoustic (home video, surround sound). The MPEG-1 coding algorithms have mainly been designed for high quality audio compression (CD-like quality) of mono- or stereophonic signals. When the MPEG-1 standard was established, it became obvious that two extensions of the standard would open new applications for digital audio coding: Better quality for low bit rates. Many applications don't need CD-like quality, but lowest possible amount of data. Although it's possible to generate bitstreams down to 32 kbit/sec with MPEG-1, they don't offer the best possible quality, because they were designed for high quality audio. Coding of up to 5 audio channels (and an optional low frequency enhancement channel) to support multichannel surround sound. LAYERS The MPEG group chose to recommend three compression methods and named them Audio Level I, II and III. Level I (sometimes named Layer I) is the simplest, a subband-coder with a psychoacoustic model. Layer II adds more advanced bit allocation techniques and greater accuracy. Layer III adds a hybrid filter bank and non-uniform quantization. Layer I, II and III gives increasing quality/compression ratios with increasing complexity and demand on processing power. The reason for recommending three methods was based on testers, whose felt that none of coders was 100% transparent to all material and partly that the best coder (layer III) was so heavy in computing that it would seriously impact the acceptance of the standard. LAYER COMPATIBILITY The specifications say, that a valid Layer III decoder shall be able to decode any Layer I, II or III MPEG audio stream. A Layer II decoder shall be able to decode Layer I and Layer II MPEG audio stream. In order to ensure the backward compatibility, an "old" MPEG-1 decoder must be able to decode the basic stereo signal of the MPEG-2 multi-channel signal, that are the L0 and R0 signals as defined in MPEG-1. This may be achieved by sending a suitable

3 mix of the multiple channels as L0 and R0, or by sending L and R as L0 and R0. The rest of the information is transmitted in the ancillary data channel, while other ancillary data may still be transmitted. AUDIO COMPRESSION Exists two alternatives how to simplify audio signal data stream. Either is sampled less often (less than times per second) or is sampled with less resolution (less than 16 bits per sample). If is demanded quality, sample frequency cannot be changed. Humans can hear sounds with frequencies from 20 Hz to 20 khz approximately (depends on person`s sence). According to the Nyquist theorem (equation 1) sample must be taken at least two times the highest frequency as is asked for producing. Allowing to imperfect filters, a 44.1 khz sampling rate is a fair minimum. So is either set out to prove the Nyquist theorem is wrong or go to work on reducing resolution. Fs > 2 * Fmax (1) Now, the real reason for using 16 bits is to get a good signal-to-noise (s/n) ratio. The noise is quantization noise from the digitizing process. For each added bit is signalto-noise ratio better about 6 db. To the ear, 6 db corresponds to a doubling of the sound level. CD audio quality achieves about 90 db s/n. This matches the dynamic range of the ear fairly well. That is no noise for listeners coming from the system. If samples has only eight bits as sampling resolution, the noise floor in recording is increased. Now it is noticeable for humans. It is possible to hear it in silent moments in the music between words or sentences. But in loud passages is not noticed it. This is the masking effect and it is the key of MPEG audio coding. In develop a coding method, the basis had to be the human ear. Unfortunately this is not a perfect device for acoustic reception but is the best that is known. Advantage was taken of one of the humane ear's shortcomings: its non-linear and adaptive threshold of hearing. MASKING EFFECT The threshold of hearing is the level below which a sound is not heard. It varies with frequency and, of course, between individuals. Whether a person hears a sound or not depends on the frequency of the sound and whether its amplitude is above or below that persons hearing threshold at that frequency.

4 The threshold of hearing is also adaptive, being constantly changed by the sounds heard. For example, an ordinary conversation in a room is perfectly audible under normal conditions. However, the same conversation in the vicinity of a loud noise, such as an aircraft passing low overhead, is impossible to hear due to the distortions introduced to the hearing thresholds of the individuals concerned. When the aircraft has gone the hearing thresholds return to normal. Sounds that are inaudible due to dynamic adaptation of the hearing threshold are said to be "masked". This effect is universal but is of particular relevance in music. An orchestra instrument playing fortissimo will, to a greater or lesser extent, make the sound of some other instruments inaudible to the human ear. When the music is recorded, however, all the frequencies go on the medium because the response of the recording device is flat, i.e. it is not dynamically adaptive. When the recording is played the masked instruments will not be audible to the listener, so might as well not be there. A linear recording, as used on CD, is inefficient in this respect. To make the most use of a recording medium the parts of the medium that contain inaudible data can better be used for audible data. In this way the amount of the recording medium needed to contain the music can be considerably reduced without any loss of audio quality. Because it is the main aspect of the MPEG encoding process to understand, next paragraph explains masking effect in an other angle of sight, on concrete signal. When there is a strong tone with a frequency of 1000 Hz nearby a tone with 1100 Hz, but 18 db lower, second tone is not heard anymore. It is completely masked by the first 1000 Hz tone. As a matter of fact, any relatively weak sound near a strong sound is masked. So the further you get from a sound the less masking effect it has. The masking effect means possibility to raise the noise floor around a strong sound, because the noise is masked by stronger sound. Raising the noise is the same as using less bits. And using less bits is the same as compression.

5 Figure 2: Masking effect The change of the hearing threshold so that a weak tone cannot be heard in presence of a strong tone is what we call masking. In the figure 2 the dotted line represents the unmodified hearing threshold. The area below the masking thresholds is where we find the inaudible information. MPEG AUDIO ENCODER A perceptual subband audio encoder constantly analyses the incoming audio signal and determines the masking curve, the threshold under which additional noise will not be audible by the human auditory system. MPEG audio coder in first step divide the frequency audio spectrum (from 20 Hz to 20 khz) into 32 subbands. Each subband holds a little slice of the audio spectrum. Later the coder calculates the masking effect of the sound in each subband. It means, if tone of 1000 Hz is in eigth subband, it will have the effect through entire subband too. The coder find appropriate masking threshold. So that acceptable s/n ratio is difference between level of strong tone and threshold looked by coder. This difference point to appropriate number of bit resolution. In addition, there are masking effects on near bands. The effect decrease with the distance from basic band (now No. 8). In real-life situation you have sounds in most bands and the masking effects are additive. In addition, the coder considers the sensitivity of the ear for various frequencies. The ear is a lot sensitive in the high and low frequencies. Peak sensitivity is around 2-4 khz, the same region that human voice occupies.

6 The information used in a quantizer in each subband is transmitted along with the coded subband samples. The decoder can then decode the bit stream without knowing how the encoder determined this information. This allows for encoders of different quality and complexity, and for future improvements of encoders. In MPEG Layer I and II each subband is 625 Hz wide (for Fs=44.1kHz). It would be better, if the sub-bands are narrower in the low frequency range and wider in the high frequency range. To do that it is necessary to use more complex filters (Layer III). To keep the filters simple was chosen to add FFT in parallel with filtering and use the spectral components from the FFT as additional information to the coder. This way gives higher resolution in the low frequencies where the ear is more sensitive. Masking effect also occurs before and after strong sound (pre- and post- masking effect). The brain needs some processing time. Premasking is only about 2 to 5 ms, postmasking can bi up till 100 ms. Other bit reduction techniques involve considering tonal and non tonal components of the sound. For the stereo signal is a lot of redundancy between channels. The last step before formatting is Huffman coding (Layer III only). For Layer II the coder works on 23 ms of sound (1152 samples) at a time and for Layer I it is 8.7 ms (because 384 samples only). It should be a problem. COMPRESS RATIO (BIT RATES) The calculations are done on a basis of a psychoacoustic model, which is the heart of MPEG audio coding. Of course, the encoder does not know what the actual sound pressure is at the listener's premises, but this is for practical purposes not necessary either. Experiments show that the coding, at a given bit rate, gives the same impression of quality for a wide range of sound pressures. One may add that the quantizing noise at low sound pressures and modest compression ratios is much too low to be noticed even by trained ears. Layer I is the simplest layer which is suitable for consumer use. Bit rates range from 32 kbit/s (mono) up to 448 kbit/s (stereo). Depending on the complexity of the encoder, a high (near CD) audio quality requires a bit rate of about kb/s per stereo program. The complexity of the decoder is low, the encoder complexity is about 1.5 to 3 times as high.

7 LAYER BIT RATE FROM [kbps] TO [kbps] LAYER I LAYER II LAYER III Table 1: Bit rates for every layer Layer II offers more compression than Layer I. Bit rates range from kb/s for mono, and from kb/s for stereo. Depending on the complexity of the encoder, a high (near CD) audio quality requires a bit rate of about kb/s per stereo program. The complexity of the decoder is about 25 % higher than for Layer I decoder, the encoder complexity is about 2 to 4 times higher. APPLICATIONS There are two audio coding schemes were the origin for three Layers: MUSICAM and ASPEC. Whereas the ASPEC coding scheme offered the best audio quality, especially at low bit rates or below 64 kbit/sec/channel, MUSICAM had some advantages with respect to implementation complexity and decoding delay. In order to have a generic coding system applicable for various demands, it was decided to establish a standard with three Layers: Layer II is nearly identical with the MUSICAM coding scheme, and Layer I is a simplified version of Layer II resp. MUSICAM. Nevertheless Layer III is mainly based on ASPEC and offers the best audio quality amongst the MPEG audio compression schemes. Layer II has numerous applications in both consumer and professional audio, such as audio broadcasting, television, telecommunications and multi-media. Layer III extends MPEG-1 applications into narrow band ISDN telecommu-nications and certain specialist areas of professional audio. PCM MPEG coded data A / D ENCODER CD Phone microphone Computer data storage Computer Figure 3: Applications for MPEG-1: encoding Satellite dish

8 Audio above the adapted hearing threshold is coded and anything below it is ignored. Using such a coding scheme reduces the amount of a medium required to make an audio recording by a factor of six. MPEG coded data PCM Data storage Phone DECODER D / A Satellite dish CD Computer speaker Figure4: Applications for MPEG-1: decoding Products with MPEG Layer I coding : Digital Compact Disk (DCC) (PASC), Solid State Audio (theme parks, traffic), Disk storage and editing, CD-i Full Motion Video. Products with MPEG Layer II coding : CD-i Full Motion Video, Video CD, Solid State Audio (theme parks, traffic), Disk storage and editing, Digital Audio Broadcast, Digital Versatile Disk (DVD), Cable and satellite radio (e.g. DMX), Cable and satellite TV (e.g. Digital Video Broadcast, Digital Satellite System), Contribution, distribution and emission links (e.g. ISDN, microwave radio), Movie sound tracks. Products with MPEG Layer III coding : ISDN. Literature [ 1 ] ISO/IEC Office, International Standard ISO/IEC , Information technology - Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit / s - Part 3: Audio. Switzerland, Geneve, [ 2 ] MPEG - FAQ: Multimedia compression - version 4.1., [ 3 ] Cornet, O.; Dejeux, I.; Tchong, C.: Etude du Codage MPEG et Conception d un Codeur bas debit. ESIEE, Paris, 1996.

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