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MC14LC5480SD Scheda tecnica(PDF) 2 Page - Freescale Semiconductor, Inc |
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MC14LC5480SD Scheda tecnica(HTML) 2 Page - Freescale Semiconductor, Inc |
2 / 24 page FREQ FREQ 2.4 V REFERENCE RO + RO - PI PO - PO + VDD VSS VAG TG TI - TI + - + - 1 1 - + SHARED DAC DAC 1.575 V REF ADC TRANSMIT SHIFT REGISTER SEQUENCE AND CONTROL DR FSR BCLKR Mu/A PDI MCLK BCLKT FST DT RECEIVE SHIFT REGISTER Figure 1. MC14LC5480 PCM Codec–Filter Block Diagram DEVICE DESCRIPTION A PCM Codec–Filter is used for digitizing and reconstruct- ing the human voice. These devices are used primarily for the telephone network to facilitate voice switching and trans- mission. Once the voice is digitized, it may be switched by digital switching methods or transmitted long distance (T1, microwave, satellites, etc.) without degradation. The name codec is an acronym from ‘‘COder’’ for the analog–to–digital converter (ADC) used to digitize voice, and ‘‘DECoder’’ for the digital–to–analog converter (DAC) used for reconstruct- ing voice. A codec is a single device that does both the ADC and DAC conversions. To digitize intelligible voice requires a signal–to–distortion ratio of about 30 dB over a dynamic range of about 40 dB. This may be accomplished with a linear 13–bit ADC and DAC, but will far exceed the required signal–to–distortion ratio at larger amplitudes than 40 dB below the peak ampli- tude. This excess performance is at the expense of data per sample. Two methods of data reduction are implemented by compressing the 13–bit linear scheme to companded pseudo–logarithmic 8–bit schemes. The two companding schemes are: Mu–255 Law, primarily in North America and Japan; and A–Law, primarily used in Europe. These com- panding schemes are accepted world wide. These compand- ing schemes follow a segmented or ‘‘piecewise–linear’’ curve formatted as sign bit, three chord bits, and four step bits. For a given chord, all sixteen of the steps have the same voltage weighting. As the voltage of the analog input increases, the four step bits increment and carry to the three chord bits which increment. When the chord bits increment, the step bits double their voltage weighting. This results in an effec- tive resolution of six bits (sign + chord + four step bits) across a 42 dB dynamic range (seven chords above 0, by 6 dB per chord). In a sampling environment, Nyquist theory says that to properly sample a continuous signal, it must be sampled at a frequency higher than twice the signal’s highest frequency component. Voice contains spectral energy above 3 kHz, but its absence is not detrimental to intelligibility. To reduce the digital data rate, which is proportional to the sampling rate, a sample rate of 8 kHz was adopted, consistent with a band- width of 3 kHz. This sampling requires a low–pass filter to limit the high frequency energy above 3 kHz from distorting the in–band signal. The telephone line is also subject to 50/60 Hz power line coupling, which must be attenuated from the signal by a high–pass filter before the analog–to– digital converter. The digital–to–analog conversion process reconstructs a staircase version of the desired in–band signal, which has spectral images of the in–band signal modulated about the sample frequency and its harmonics. These spectral images are called aliasing components, which need to be attenuated to obtain the desired signal. The low–pass filter used to at- tenuate these aliasing components is typically called a re- construction or smoothing filter. The MC14LC5480 PCM Codec–Filter has the codec, both presampling and reconstruction filters, a precision voltage reference on–chip, and requires no external components. Freescale Semiconductor, Inc. For More Information On This Product, Go to: www.freescale.com |
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