 |
Electronic Components Datasheet Search |
|
|
Electronic Components Datasheet Search |
|
CSP1027 Datasheet(PDF) 6 Page - Agere Systems |
|
||||||||||||||||||||||||||||||
CSP1027 Datasheet(HTML) 6 Page - Agere Systems |
|
|
6 / 64 page  CSP1027 Voice Band Codec for Data Sheet Cellular Handset and Modem Applications December 1999 Lucent Technologies Inc. 6 4 Architectural Information (continued) 4.1 Overview The CSP1027 is a complete analog-to-digital and digi- tal-to-analog acquisition and conversion system (see Figure 3 on page 5) that band limits and encodes ana- log input signals into 16-bit PCM, and takes 16-bit PCM inputs and reconstructs and filters the resultant analog output signal. The selectable A/D input circuits, pro- grammable sample rates, and digital filter options allow the user to optimize the codec configuration for either speech coding or voice band data communications. The on-chip digital filters meet the ITU-T G.712 voice band frequency response and signal to distortion plus noise specifications and are suitable for IS-54, GSM, and JDC digital cellular applications. In addition, the small supply current drain, when powered down, extends battery life in mobile communication applica- tions. The CSP1027 is intended for both voice band voice and data communication systems. As a result, this codec has a variety of features not found in standard voice band codecs: s 3.0 V regulated power supply for a condenser micro- phone. s Microphone preamplifier with programmable input ranges. s Mute control of D/A output. s Programmable output gain in 3 dB increments. s Output speaker driver. s Programmable master clock divider to set A/D and D/A conversion rate. s Testability loopback mode. s High-quality dither scheme to eliminate idle channel tones. 4.2 Description of Signal Paths 4.2.1 Sampling Frequency The oversampling ratio of the codec is 125:1; this is the ratio of the frequency of the oversampling clock to the frequency of the sampling clock. Most speech applica- tions specify a sampling frequency of 8 kHz, yielding an oversampling frequency of 8 kHz x 125 = 1.0 MHz. The codec will operate at sampling frequencies up to 24 kHz, with the frequency response of the digital filters being changed proportionally. For this architectural description, the sampling frequency, fS, is assumed to be 8 kHz, with an oversampling frequency, fOS, of 1 MHz, unless otherwise stated. 4.2.2 Analog-to-Digital Path The analog-to-digital (A/D) conversion signal path (see Figure 3 on page 5) begins with the analog input driving the input block. The signal from the input block is then encoded by a second-order ∆-Σ modulator A/D. The bulk of the antialiasing filtering is done in the digital domain in two stages following the ∆-Σ modulator to give a 16-bit result. The blocks will next be covered in more detail. 4.2.3 Analog Input Block The A/D input block operates in two modes: when the external input gain select (EIGS) pin is low or left unconnected, the input goes through a preamplifier and is band limited by a second-order 30 kHz low-pass anti- aliasing filter (see Figure 4 on page 7). When EIGS is high, external resistors, Rin and Rfb, are used to set the gain of an inverting amplifier (see Figure 5 on page 7). These resistors, in combination with Cin and Cfb, cre- ate a bandpass antialiasing filter. Note that EIGS is a digital pin whose input levels are relative to digital power and ground (VDD and VSS). 4.2.4 A/D Modulator and Digital Filters A second-order ∆-Σ modulator quantizes the analog signal to 1 bit (see Figure 3 on page 5). At the same time, the resulting quantization noise is shaped such that most of this noise lies outside of the baseband. The modulator output is then digitally low-pass filtered to remove the out-of-band quantization noise. After this filtering, the output samples are decimated down to the output sampling frequency. In the CSP1027, the filter- ing and decimation are completed in two stages. The first-stage low-pass filter shapes the modulator output according to the sinc-cubic transfer function: The output sampling frequency of the sinc-cubic filter is reduced by a factor of 25 from 1 MHz to 40 kHz. The sinc-cubic filter places nulls in the frequency response at multiples of 40 kHz, and removes most of the quanti- zation noise above 20 kHz so that very little energy is aliased as a result of the decimation. The sinc-cubic filter output is then processed by a seventh-order IIR digital low-pass filter. This filter removes the out-of-band quantization noise between 3.4 kHz and 20 kHz, compensates for the passband droop caused by the sinc-cubic decimator, and deci- mates the sampling frequency by a factor of five from 40 kHz to 8 kHz. Hz () 1 25 ------ 1z 25 – – () 1z 1 – – () ------------------------- × 3 = |
Similar Part No. - CSP1027 |
|
Similar Description - CSP1027 |
|
|
Link URL |
| Privacy Policy |
| ALLDATASHEET.COM |
| Does ALLDATASHEET help your business so far? [ DONATE ] |
About Alldatasheet | Advertisement | Datasheet Upload | Contact us | Privacy Policy | Link Exchange | Manufacturer List All Rights Reserved©Alldatasheet.com |
| Russian : Alldatasheetru.com | Korean : Alldatasheet.co.kr | Spanish : Alldatasheet.es | French : Alldatasheet.fr | Italian : Alldatasheetit.com Portuguese : Alldatasheetpt.com | Polish : Alldatasheet.pl | Vietnamese : Alldatasheet.vn Indian : Alldatasheet.in | Mexican : Alldatasheet.com.mx | British : Alldatasheet.co.uk | New Zealand : Alldatasheet.co.nz |
|
Family Site : ic2ic.com |
icmetro.com |
English ▼
English
Chinese
German
Japanese
Russian
Korean
Spanish
French
Italian
Portuguese
Polish
Vietnamese
Indian
Mexican
British
New Zealand
