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Subband APCM coding has found numerous applications in the professional audio industry and the streaming media industry. We will examine ISO/MPEG-1 Layer 2 (MUSICAM by another name) and ISO/MPEG-1 Layer 3 (the popular MP3 format used by such applications as Winamp, MusicMatch, Napster, and leading streaming media formats). We will also examine Dolby AC-3, which is used in DVD players and the ATSC DTV system, and finally, the MPEG AAC system favored by MPEG-4 streaming media implementations. ISO/MPEG-1 Layer 2 (MUSICAM). This algorithm differs from Layer 1 by adopting more accurate quantizing procedures and by additionally removing redundancy and irrelevancy on the generated scale factors [15]. The ISO/MPEG-1 Layer 2 scheme operates on a block of 1152 PCM samples, which at 48 kHz sampling represents a 24 ms time block of the input audio signal. Simplified block diagrams of the encoding/decoding systems are given in Figure E.16. The incoming linear PCM signal block is divided into 32 equally spaced subbands using a polyphase analysis filter bank (Figure E.16a). At 48 kHz sampling, this equates to the bandwidth of each subband being 750 Hz. The bit allocation for the requantizing of these subband samples is then dynamically controlled by information derived from analyzing the audio signal, measured against a preset psychoacoustic model. The filter bank, which displays manageable delay and minimal complexity, optimally adapts each block of audio to balance between the effects of temporal masking and inaudible pre-echoes.



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Figure E.16 ISO/MPEG-1 Layer 2 system: (a) enoder block diagram, (b) decoder block diagram. (After [15].)

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with, but once you become more confident and want to expand, you can simply add more tiles! To begin with, I recommend that you cut yourself a piece of MDF that is 36 cm square, although please bear in mind that this measurement is wholly arbitrary. Now using a ruler and set square, divide the sheet into a matrix of six squares by six squares. This will give you thirty-six equal squares 6 cm square. Now, using the ruler and set square, draw a line 1 cm either side of each line making up the squares. This will leave you with a sheet that does not look dissimilar to Figure 8-6. You are now going to drill holes for the screws that will support the mirrors. You will need to select a drill that is slightly smaller than the screw that you are going to drill the hole for. However, please note that the screw does not need to be a tight fit in the hole, as it would be if you were joining two pieces of wood. Instead, the screw is only going to be used for light adjustment, so the screw can be a relatively slack fit in the hole. Looking at your board of squares, you are going to be drilling two holes in each 6 cm square. The holes will be at the top left and bottom right, where the lines cross to form the smaller square inside each square. Sounds confusing, well, take a look at the furnace drilling diagram (Figure 8-7), which shows where to drill in each square.

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The PCM signal is also fed to a fast Fourier transform (FFT) running in parallel with the filter bank. The aural sensitivities of the human auditory system are exploited by using this FFT process to detect the differences between the wanted and unwanted sounds and the quantization noise already present in the signal and then to adjust the signal-tomask thresholds, conforming to a preset perceptual model. This psychoacoustic model is only found in the coder, thus making the decoder less complex and permitting the freedom to exploit future improvements in coder design. The actual number of levels for each quantizer is determined by the bit allocation. This is arrived at by setting the signal-to-mask ratio (SMR) parameter, defined as the difference between the minimum masking threshold and the maximum signal

level This minimum masking threshold is calculated using the psychoacoustic model and provides a reference noise level of just noticeable noise for each subband In the decoder, after demultiplexing and deciphering of the audio and side information data, a dual-synthesis filter bank reconstructs the linear PCM signal in blocks of 32 output samples (Figure E-16b) A scale factor is determined for each 12-subband sample block The maximum of the absolute values of these 12 samples generates a scale factor word consisting of 6 bits, a range of 63 different levels Because each frame of audio data in Layer 2 corresponds to 36 subband samples, this process will generate 3 scale factors per frame However, exploiting some redundancy in the data can reduce the transmitted data rate for these scale factors Three successive subband scale factors are analyzed and a pattern is determined.

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