antees that, regardless of origin, any fully compli- ant MPEGiaudio decoder will be able to decode any MPEG/audio bitstream with a predictable Tutorial on result. Designers are free to try new and different A implementations of the encoder or decoder within the bounds of the standard. The encoder especial- ly offers good potential for diversity. Wide accep- MPEG/ Audio tance of this standard will permit manufacturers to produce and sell, at reasonable cost, large num- bers of MPEG/audio codecs. Lompression Features and applications MPEGiaudio is a generic audio compression standard. Unlike vocal-tract-model coders special- Davis Pan ly tuned for speech signals, the MPEG/audio coder Motorola gets its compression without making assumptions about the nature of the audio source. Instead, the coder exploits the perceptual limitations of the his tutorial covers the theory behind human auditory system. Much of the compression -This tutorial covers MI’EGiaudio compression. It is written results from the removal of perceptually irrelevant the theory behind for people with a modest background parts of the audio signal. Since removal of such MPEG/audio in digital signal processing and does parts results in inaudible distortions, MPEG/audio compression. While T not assume prior experience in audio compression can compress any signal meant to be heard by the lossy, the algorithm or psychoacoustics. The goal is to give a broad, pre- human ear. often can provide liminary understanding of MPEG/audio compres- In keeping with its generic nature, MI’EGiaudio “transparent,” sion, so I omitted many of the details. Wherever offers a diverse assortment of compression modes, perceptually lossless possible, figures and illustrative examples present as follows. In addition, the MPEG/audio bitstream compression even the intricacies of the algorithm. makes features such as random access, audio fast- with factors of 6-to-1 The MPEGiaudio compression algorithm is the forwarding, and audio reverse possible. or more. It exploits first international standard’,’ for the digital the perceptual compression of high-fidelity audio. Other audio Sampling rate. The audio sampling rate can be properties of the compression algorithms address speech-only appli- 32, 41.1, or 48 kHz. human auditory cations or provide only medium-fidelity audio system. The article compression performance. Audio channel support. The compressed bit- also covers the basics The MPEGiaudio standard results from more stream can support one or two audio channels in of psychoacoustic than three years of collaborative work by an inter- one of four possible modes: modeling and the national committee of high-fidelity audio com- methods the pression experts within the Moving Picture Experts 1. a monophonic mode for a single audio channel, algorithm uses to Group (MPEGiaudio). The International Organ- compress audio data ization for Standards and the International 2. a dual-monophonic mode for two indepen- with the least Electrotechnical Commission (ISO/IEC) adopted dent audio channels (functionally identical to perceptible this standard at the end of 1992. the stereo mode), degradation. Although perfectly suitable for audio-only applications, MPEG/audio is actually one part of a 3. a stereo mode for stereo channels that share three-part compression standard that also includes bits but do not use joint-stereo coding, and video and systems. The MPEG standard addresses the compression of synchronized video and audio 4. a joint-stereo mode that takes advantage of at a total bit rate of about 1.5 megabits per second either the correlations between the stereo (Mbps). channels or the irrelevancy of the phase dif- The MPEG standard is rigid only where neces- ference between channels, or both. sary to ensure interoperability. It mandates the syntax of the coded bitstream, defines the decod- Predefined bit rates. The compressed bit- ing process, and provides compliance tests for stream can have one of several predefined fixed bit assessing the accuracy of the decoder.‘ This guar- rates ranging from 32 to 224 kilobits per second 1070-986X/95/$4 00 0 1995 IEEE (Kbps) per channel. Depending on the audio sam- pling rate, this translates to compression factors Bit/noise bitstream ranging from 2.7 to 24. In addition, the standard PCM frequency Bitstream audio quantizer, and formatting provides a “free” bit rate mode to support fixed bit input filter bank rates other than the predefined rates. Compression layers. MPEG/audio offers a Psychoacoustic Ancillary data choice of three independent layers of compression. model (optional) This provides a wide range of trade-offs between (a) codec complexity and compressed audio quality. Layer I, the simplest, best suits bit rates above 128 Kbps per channel. For example, Philips’ bitstream PCM Digital Compact Cassette (DCC)’ uses Layer I Frequency audio sample to time b compression at 192 Kbps per channel. unpacking reconstruction Layer I1 has an intermediate complexity and I I targets bit rates around 128 Kbps per channel. ‘i Possible applications for this layer include the Ancillary data (b) (if included) coding of audio for digital audio broadcasting (DAB),” the storage of synchronized video-and- audio sequences on CD-ROM, and the full-motion extension of CD-interactive, Video CD. Layer 111 is the most complex but offers the best audio quality, particularly for bit rates around 64 the input into multiple subbands of frequency. Figure 1. MPEG/audio Kbps per channel. This layer suits audio transmis- The input audio stream simultaneously passes compression and sion over ISDN. through a psychoacoustic model that determines decompression. (a) All three layers are simple enough to allow sin- the ratio of the signal energy to the masking MPEG/audio encoder. gle-chip, real-time decoder implementations. threshold for each subband. The bit- or noise- (b)MPEG/audio allocation block uses the signal-to-mask ratios to decoder. Error detection. The coded bitstream supports decide how to apportion the total number of code an optional cyclic redundancy check (CRC) error- bits available for the quantization of the subband detection code. signals to minimize the audibility of the quanti- zation noise. Finally, the last block takes the rep- Ancillary data. MPEG/audio provides a means resentation of the quantized subband samples and of including ancillary data within the bitstream. formats this data and side information into a coded bitstream. Ancillary data not necessarily Overview related to the audio stream can be inserted Mth- The key to MPEG/audio compression-quanti- in the coded bitstream. The decoder deciphers this zation-is lossy. Nonetheless, this algorithm can bitstream, restores the quantized subband values, give “transparent,” perceptually lossless compres- and reconstructs the audio signal from the sub- sion. The MPEG/audio committee conducted band values. extensive subjective listening tests during devel- First we’ll look at the time to frequency map- opment of the standard. The tests showed that ping of the polyphase filter bank, then implemen- even with a 6-to-1 compression ratio (stereo, 16 tations of the psychoacoustic model and more bits per sample, audio sampled at 48 kHz com- detailed descriptions of the three layers of pressed to 256 Kbps) and under optimal listening MPEG/audio compression. That gives enough conditions, expert listeners could not distinguish background to cover a brief summary of the differ- between coded and original audio clips with sta- ent bit (or noise) allocation processes used by the tistical significance. Furthermore, these clips were three layers and the joint stereo coding methods. specially chosen as difficult to compress. Grewin and Ryden: gave the details of the setup, proce- The polyphase filter bank dures, and results of these tests. The polyphase filter bank is common to all lay- Figure 1 shows block diagrams of the MPEG/ ers of MPEGiaudio compression. This filter bank audio encoder and decoder. The input audio divides the audio signal into 32 equal-width stream passes through a filter bank that divides frequency subbands. Relatively simple, the filters Figure 2. Flow diagram Shift in 32 new samples into 512- analysis polyphase filter outputs that closely of the MPEG/audio I point FIFO buffer, Xi I resembles a method described by Rothweiler.# encoder filter bank. Figure 2 shows the flow diagram from the IS0 MPEG/audio standard for the MPEG-encoder fil- ter bank based on Rothweiler’s proposal. I Window samples: for i = 0 to 511 do Z, = C, XI By combining the equations and steps shown by Figure 2, we can derive the following equation for the filter bank outputs: I I I Partial calculation: for i = 0 to 63 do Y, = Z’ + 641 st[;] = /=o A2 - 1 c M[i][k]x (C[k+64j] x x[k+64j]) (1) 63 k=O j=O Calculate 32 samples by matrixing SI= YI MI k=O where i is the subband index and ranges from 0 to 31; s,[i] is the filter output sample for subband i at time where tis an integer multiple of 32 audio Output 32 subband samples t, I I sample intervals; C[H]is one of 512 coefficients of the analysis window defined in the standard; x[n] is an audio input sample read from a 512-sample buffer: and provide good time resolution with reasonable frequency resolution. The design has three (2 x i + 1)x (k- 16)x E notable concessions. M[i][k]= cos First, the equal widths of the subbands do not 1 accurately reflect the human auditory system’s fre- quency-dependent behavior. Instead, the width are the analysis matrix coefficients. of a “critical band” as a function of frequency is a Equation 1 is partially optimized to reduce the good indicator of this behavior. Many psycho- number of computations. Because the function acoustic effects are consistent with a critical-band within the parentheses is independent of the value frequency scaling.