TLV320AIC22CPTR

       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 D Two 16-Bit Analog-to-Digital Converters D D D D D D D D D Supports 8- and 16-kHz Sampling Rates D Preamplifiers for Microphone, Handset, (ADCs) Two 16-Bit Digital-to-Analog Converters (DACs) Programmable Input/Output Gain Analog Crosspoint to Connect the Two Coders/Decoders (Codecs) to Any of the I/O Ports – Controlled Through the Serial Port or the Inter-Integrated Circuit (I2C) Bus 8-Bit A-Law/µ-Law Companded Data or 16-Bit Linear Data Complying With G.711 Standard Filters Comply With G.712 and G.722 Standards Programmable Analog-to-Digital and Digital-to-Analog Conversion Rate Typical 77-dB Signal-to-Noise + Distortion for ADC Typical 78-dB Signal-to-Noise + Distortion for DAC D D D D D D D D D D D Headset, and Speakerphone Gain Selectable Via the Serial Port or I2C Bus 2.5-V Microphone Bias Voltage Seamless Interface to a Single Multichannel Buffered Serial Port (McBSP) of a C54x or a C6x Digital Signal Processor (DSP) Four TLV320AIC22C ICs Can Be Cascaded Together to Allow up to Eight Channels 2s-Complement Data Format Differential Outputs Typical Low Crosstalk < –85 dB Hardware/Software Power Down Independent Power Down for Drivers Single 3.3-V Supply Operation 120-mW Typical Power Consumption Available in 48-Terminal Low-Profile Plastic Quad Flatpack (LQFP) Package description The TLV320AIC22C contains two coders/decoders (codecs) for voice applications, including voice over internet protocol (VoIP). It features two analog-to-digital converter (ADC) channels and two digital-to-analog converter (DAC) channels that can be connected to a handset, headset, speaker, microphone, or a subscriber line via an analog crosspoint. The TLV320AIC22C has a flexible serial interface that allows the two channels of the TLV320AIC22C to be interfaced to a single multichannel buffered serial port (McBSP) of the external digital signal processor (DSP). The two channels share the digital interface at different time slots. Up to four TLV320AIC22C units can be cascaded together to obtain eight channels. For control purposes, either the serial interface or the inter-integrated circuit (I2C) interface can be used. Programmable-gain amplifiers (PGAs), preamp gain, microphone bias voltages, and analog crosspoint are programmed through the serial interface or the I2C interface. The TLV320AIC22C can be powered down, via a dedicated terminal or by using software control, to reduce power dissipation. The TLV320AIC22C is available in a 48-terminal LQFP package and is characterized for operation from –40°C to 85°C. ORDERING INFORMATION PACKAGE TA PLASTIC QUAD FLATPACK (PT) –40°C to 85°C TLV320AIC22CPT Please be aware that an important notice concerning availability, standard warranty, and use in critical applications of Texas Instruments semiconductor products and disclaimers thereto appears at the end of this data sheet. C54x and TMS320C54x are trademarks of Texas Instruments Incorporated. Copyright  2003, Texas Instruments Incorporated  
  
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   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 CIINM AVSS3 SPOUTM AVDD3 SPOUTP AVSS3 MCBIAS MCINP MCINM AVDD1 AVSS1 CIINP PT PACKAGE (TOP VIEW) 36 35 34 33 32 31 30 29 28 27 26 25 LCDOUT HSOUTM HSOUTP HSINM HSINP FILT2 FILT1 LNINP LNINM LNOUTM LNOUTP VCOM 37 24 38 23 39 22 40 21 41 20 42 19 43 18 44 17 45 16 46 15 47 14 13 48 2 3 4 5 6 7 8 9 10 11 12 HDINM HDINP HDOUTM HDOUTP AVDD2 AVSS2 NC NC PWRDWN SDA SCL AD1 1 NC – No internal connection 2 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 VSS RESET MCLK M/S BCLK FSYNC DIN DOUT DVSS DVDD I2C/SPI AD0 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 functional block diagram Codec 1 HSOUTP Handset Output HSOUTM 150 Ω PGA –36 dB to 12 dB DAC HSINP HSIN DIN DOUT BCLK FSYNC M/S MCLK PGA –36 dB to 12 dB 1.5-dB Steps (see Note A) Handset Input HSINM HSIN HDIN MCIN LNIN CIIN ADC Preamp (23 dB/14 dB/ 0 dB/Mute) HDOUTP Headset Output HDOUTM 150 Ω Codec 2 Serial Interface HDINP PGA –36 dB to 12 dB 1.5-dB Steps DAC Headset Input HDINM HDIN Preamp (23 dB/14 dB/0 dB/Mute) HSIN PGA –36 dB to 12 dB 1.5-dB Steps (see Note A) 4-Bit DAC (Shared) AD1 SCL SDA MCINP LCD OUT Microphone Input MCINM MCIN Preamp (42 dB/32 dB/ 20 dB/ 0 dB/Mute) MCBIAS LNOUTP Line Output LNOUTM 600 Ω I2C/SPI AD0 SPOUTP Speaker Output SPOUTM 8 Ω HDIN MCIN LNIN CIIN ADC Control/ Status Registers VCOM 1.5 V LNINP I2C Logic LNIN Line Input LNINM CIINP CIIN CIINM Caller ID Amplifier Input NOTES: A. The attenuation on the ADC PGA (0 dB to –36 dB) is done after the analog-to-digital conversion. This attenuation cannot prevent clipping. To prevent clipping, both the preamp gain and the PGA should be lowered to the required value. B. Input and output analog signals are differential. All switches are register controlled. functional block diagram (one of two codecs shown) Digital Output Buffer Decimation Filter Sigma-Delta ADC Antialiasing Filter Analog Input PGA Digital Loopback Vref VCOM Digital Input Interpolation Filter Sigma-Delta DAC POST OFFICE BOX 655303 Analog Loopback Low-Pass Filter • DALLAS, TEXAS 75265 PGA Analog Output 3 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Terminal Functions TERMINAL NAME NO. I/O DESCRIPTION Address. In I2C mode, AD1 is used with AD0 to form the lower two bits of the 7-bit I2C chip address. The upper five bits are fixed at 11100. AD1 is used in conjunction with AD0 to assign the two time slots for the codec in serial-port mode. AD1 is the MSB. Address. In I2C mode, AD0 is used with AD1 to form the lower two bits of the 7-bit I2C chip address. The upper five bits are fixed at 11100. AD0 is used in conjunction with AD1 to assign the two time slots for the codec in serial-port mode. AD0 is the LSB. AD1 12 I AD0 13 I AVDD1 AVDD2 33 I Analog power supply. Connect to AVDD2 (see Note 1). 5 I Analog power supply. Connect to AVDD1 (see Note 1). AVDD3 27 I Analog power supply for 8-Ω speaker driver. AVDD3 can be connected to AVDD1 and AVDD2. Because this signal requires large amounts of current, it is recommended that a separate PCB trace be run to this terminal and connected to the main supply at the power-supply connection to the PC board (see Note 1). AVSS1 AVSS2 32 I Analog ground. Connect to AVSS2 (see Note 1). 6 I Analog ground. Connect to AVSS1 (see Note 1). AVSS3 25 29 I Analog ground for 8-Ω speaker driver. AVSS3 can be connected to AVSS1 and AVSS2. Because this signal requires large amounts of current, it is recommended that a separate PCB trace be run to this terminal and connected to the main supply at the power-supply connection to the PC board (see Note 1). BCLK 20 I/O Bit clock. BCLK clocks serial data into DIN and out of DOUT. When configured as an output (master mode), BCLK is generated internally by multiplying the frame-synchronization signal frequency by 256. When configured as an input (slave mode), BCLK is an input and must be synchronous with the master clock and frame synchronization. CIINM 30 I Caller ID amplifier analog inverting input CIINP 31 I Caller ID amplifier analog noninverting input DIN 18 I Data input. DIN receives the DAC input data and register data from the external DSP or controller and is synchronized to BCLK. Data is latched on the falling edge of BCLK in the two time slots that are specified by the AD1 and AD0 bits. Codec 1 receives data in the first assigned time slot, followed by codec 2 receiving data in the second assigned time slot. DOUT 17 O Data output. DOUT transmits the ADC output bits and the register data. It is synchronized to BCLK. Data is transmitted on the rising edge of BCLK in the two time slots that are specified by the AD1 and AD0 bits. DOUT is at high impedance during time slots not assigned to the codec. Codec 1 transmits data in the first assigned time slot, followed by codec 2 in the second assigned time slot. DVDD 15 I Digital power supply (see Note 1) DVSS 16 I Digital ground (see Note 1) FILT1 43 O Reference filter node. FILT1 and FILT2 provide decoupling of the reference voltage. This reference is 2.25 V. The optimal capacitor value is 0.1 µF (ceramic) and is connected between FILT1 and FILT2. FILT1 should not be used as a voltage source. FILT2 42 O Reference filter node. FILT1 and FILT2 provide decoupling of the reference voltage. This reference is 0 V. The optimal capacitor value is 0.1 µF (ceramic) and is connected between FILT1 and FILT2. FSYNC 19 I/O Frame synchronization. FSYNC indicates the beginning of a frame and the start of time slot 0. When FSYNC is sampled high on the rising edge of BCLK, the codec receives or transmits data in its specified time slot (specified by AD0 and AD1) in the frame. FSYNC is generated by the master device (output) and is an input to the slave devices. Codec 1 communicates in the first assigned time slot, followed by codec 2 communicating in the second assigned time slot. HDINM 1 I Headset amplifier analog inverting input. A connection between HDIN and HDOUT occurs, with selected echo gain, unless the echo gain is muted (see register 14). HDINP 2 I Headset amplifier analog noninverting input HDOUTM 3 O Inverting headset output. The HDOUTM terminal, together with the HDOUTP terminal form the differential output. With HDOUTP, a 150-Ω load can be driven differentially. HDOUTM can be used alone for single-ended operation. HDOUTP 4 O Noninverting headset output. HDOUTP can be used alone for single-ended operation. With HDOUTM, a 150-Ω load can be driven differentially. NOTE 1: This device has separate analog and digital power and ground terminals. For best operation and results, the PC board design should utilize separate analog and digital power supplies, as well as separate analog and digital ground planes. Mixed-signal design practices should be used. 4 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Terminal Functions (Continued) TERMINAL NAME NO. I/O DESCRIPTION HSINM 40 I Handset amplifier analog inverting input. A connection between HSIN and HSOUT occurs, with selected echo gain, unless the echo gain is muted (see register 13). HSINP 41 I Handset amplifier analog noninverting input. A connection between the HSIN and HSOUT occurs, with programmed echo gain, unless the echo gain is muted (see register 13). HSOUTM 38 O Inverting handset output. The HSOUTM terminal, together with the HSOUTP terminal, forms the differential output. With HSOUTP, a 150-Ω load can be driven differentially. HSOUTM can be used alone for single-ended operation. HSOUTP 39 O Noninverting handset output. With HSOUTM, a 150-Ω load can be driven differentially. HSOUTP can be used alone for single-ended operation. I2C/SPI 14 I I2C/serial-port interface select. Setting this terminal high allows the user to program the registers using the I2C interface. A low state configures the serial interface for control register programming during normal data transmission, using time slots 0 and 1. When set high (I2C interface selected), time slots 0 and 1 in the normal data transmission are ignored. LCDOUT 37 O 4-bit DAC output voltage, programmed through the control interface. LCDOUT can be used to provide the bias voltage for an LCD display. LNINP 44 I Line-port amplifier analog noninverting input (see Note 2). LNINM 45 I Line-port amplifier analog inverting input (see Note 2). LNOUTM 46 O Inverting line-port output. The LNOUTM terminal, together with the LNOUTP terminal, form the differential output. With LNOUTP, a 600-Ω load can be driven differentially. LNOUTM can be used alone for single-ended operation (see Note 2). LNOUTP 47 O Noninverting line-port output. With LNOUTM, a 600-Ω load can be driven differentially. LNOUTP can be used alone for single-ended operation (see Note 2). MCLK 22 I Master clock input. All internal clocks are derived from this clock. This clock typically is 32.768 MHz or 24.576 MHz. MCBIAS 36 O MCBIAS provides a bias voltage and current to operate Electret microphones. The bias voltage is specified across the microphone at 2.5 V. MCINM 34 I Microphone amplifier analog inverting input MCINP 35 I Microphone amplifier analog noninverting input M/S 21 I Master/slave select input. When M/S is high, the device is the master, and when it is low, it is a slave. Power down. PWRDWN is active high and when PWRDWN is pulled high, the device goes into a power-down mode that disables the output drivers and most of the high-speed clocks. The serial interface and I2C interface are enabled. However, all register values are sustained and the device resumes full-power operation without reinitialization when PWRDWN is pulled low again. PWRDWN resets the counters only and preserves the programmed register contents. PWRDWN 9 I RESET 23 I SCL 11 I SDA 10 I/O Bidirectional control data I/O line for the I2C interface. Data is clocked into and out of the device by SCL. Tie this terminal to DVDD when not used. SPOUTP 26 O Inverting analog output from 8-Ω speaker amplifier SPOUTM 28 O Noninverting analog output from 8-Ω speaker amplifier NC 7, 8 VCOM VSS 48 O 24 I Codec device reset. RESET initializes all device internal registers to default values when pulled low. SCL is the serial control interface clock for the I2C interface and is used to clock control bits into and out of the device through the SDA terminal. Tie this terminal to DVDD when not used. Reserved. Leave unconnected. VCOM provides a reference voltage of 1.5 V. The maximum source or sink current at this terminal is 2.5 mA. Internal substrate connection. VSS should be tied to AVSS1 and AVSS2 (see Note 1). NOTES: 1. This device has separate analog and digital power and ground terminals. For best operation and results, the PC board design should utilize separate analog and digital power supplies as well as separate analog and digital ground planes. Mixed-signal design practices should be used. 2. The LNINP and LNINM are sensitive to crosstalk from LNOUTP and LNOUTM. Keep the LNOUT and LNIN signals separated on the printed circuit board. Do not route the LNOUT signals parallel to the LNIN signals. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 5 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 absolute maximum ratings over operating free-air temperature range (unless otherwise noted)† Supply voltage, AVDD1, AVDD2, AVDD3 to AVSS1, AVSS2, AVSS3, DVDD to DVSS . . . . . . . . . –0.3 V to 4.5 V Analog input voltage range to AVSS1, AVSS2, and AVSS3 . . . . . . . . . . . . . . . . . . . . . . –0.3 V to AVDD + 0.3 V Digital input voltage range . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –0.3 V to DVDD + 0.3 V Operating virtual junction temperature range, TJ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –40°C to 150°C Operating free-air temperature range, TA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –40°C to 85°C Storage temperature range, Tstg . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –65°C to 150°C Lead temperature 1,6 mm (1/16 inch) from case for 10 seconds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260°C † Stresses beyond those listed under “absolute maximum ratings” may cause permanent damage to the device. These are stress ratings only, and functional operation of the device at these or any other conditions beyond those indicated under “recommended operating conditions” is not implied. Exposure to absolute-maximum-rated conditions for extended periods may affect device reliability. recommended operating conditions Supply voltage, AVDD1, AVDD2, AVDD3, DVDD (3.3-V supply) Analog signal peak-to-peak input voltage, VI(analog) differential Analog signal peak-to-peak input voltage, VI(analog), differential MIN NOM MAX 3 3.3 3.6 V LNINP, LNINM, CIINP, CIINM Preamp gain set to 0 dB 4 V HSINP, HSINM, HDINP, HDINM, MCINP, MCINM Preamp gain set to 0 dB 4 (scaled by the selected gain) V High-level input voltage, any digital input, VIH 2 V Low-level input voltage, any digital input, VIL 0.8 LNOUTP, LNOUTM Differential output o tp t load resistance, resistance RL Input impedance for hybrid amplifiers UNIT V 600 SPOUTP, SPOUTM 8 HDOUTP, HDOUTM 150 HSOUTP, HSOUTM 150 LNINP, LNINM Ω 68 kΩ Master clock input 32.768 Load capacitance, CL (unless otherwise specified) MHz 20 pF ADC or DAC conversion rate 7.2 16 kHz Operating free-air temperature, TA –40 85 °C electrical characteristics over recommended operating free-air temperature range, DVDD = 3.3 V, AVDD1 = AVDD2 = AVDD3 = 3.3 V (unless otherwise noted) digital inputs and outputs PARAMETER 6 TEST CONDITIONS VOH VOL High-level output voltage, any digital output IIH IIL High-level input current, any digital input Ci Input capacitance, any digital input Co Output capacitance, any digital output Ilkg1 Input leakage current, any digital input (except DIN) Low-level output voltage, any digital output Low-level input current, any digital input IOH = –360 µA IOL = 2 mA Inp t leakage ccurrent, Input rrent DIN IOZ Output leakage current, any digital output POST OFFICE BOX 655303 TYP • DALLAS, TEXAS 75265 MAX 2.4 UNIT V VIH = 3.3 V VIL = 0.6 V VIH = 3.3 V VIL = 0.6 V Ilkg2 lk 2 MIN 0.4 V 10 µA 10 µA 5 pF 10 pF 10 µA 20 µA 60 µA 10 µA 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 ADC dynamic performance characteristics (see Notes 3 and 4) PARAMETER TEST CONDITIONS VI = –3 3 dBr at input in ut to ADC ADC dynamic performance with line in input, ut, handset or headset selected. VI = –13 13 dBr at in input ut to ADC VI = –43 43 dBr at in input ut to ADC Preamp gain = 0 dB, PGA gain = 0 dB VI = –33 33 dBr, Pream ADC dynamic performance with microphone micro hone input in ut selected VI = –43 43 dBr, Pream Preamp gain = 0 dB, PGA gain = 0 dB VI = –58 58 dBr, Pream Preamp gain = 0 dB, PGA gain = 0 dB VI = –13 13 dBr, PGA gain = 0 dB ADC dynamic d i performance f with caller ID input selected VI = –33 33 dBr, PGA gain = 0 dB VI = –43 dBr,, PGA g gain = 0 dB MIN THD 72 SNR 76 THD + N 70 THD 72 SNR 68 THD + N 66 THD 48 SNR 39 THD + N 38 THD 60 SNR 52 THD + N 45 THD 48 SNR 43 THD + N 41 THD 34 SNR 28 THD + N 26 THD 72 SNR 70 THD + N 68 THD 56 SNR 50 THD + N 48 THD 46 SNR 38 THD + N 36 TYP MAX UNIT dB dB dB NOTES: 3. The test condition is a 1020-Hz input signal with an 8-kHz conversion rate. Input and output common mode is 1.5 V. 4. The input level corresponds to TSNR mask corner points in specification G.712. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 7 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 ADC channel transfer response characteristics over recommended ranges of supply voltage and operating free-air temperature, when selecting handset or headset as input PARAMETER TEST CONDITIONS MIN TYP MAX 0 Hz to 60 Hz 200 Hz Preamp gain = 0 dB, PGA gain = 0 dB, Pream Sampling g rate = 8 kHz (see Note 5) Gain relative to gain at 1020 Hz –1.8 0.35 300 Hz to 3 kHz –0.25 0.25 3.3 kHz –0.35 0.25 3.4 kHz –0.9 –0.25 4 kHz –25 4.6 kHz to 8 kHz –60 Above 8 kHz –55 0 Hz to 120 Hz –26 400 Hz Pream gain = 0 dB, PGA gain = 0 dB, Preamp Sampling g rate = 16 kHz (see Note 6) UNIT –26 –1.8 0.35 600 Hz to 6 kHz –0.25 0.25 6.6 kHz –0.35 0.25 6.8 kHz –0.9 0.25 8 kHz –25 9.2 kHz to 16 kHz –60 Above 16 kHz –55 dB NOTES: 5. When the high-pass filter (HPF) is bypassed, the passband is 0 Hz to 3 kHz. When the HPF is inserted, the passband is 300 Hz to 3 kHz. 6. When the HPF is bypassed, the passband is 0 Hz to 6 kHz. When the HPF is inserted, the passband is 600 Hz to 6 kHz. ADC channel passband frequency characteristics with microphone selected as input PARAMETER TEST CONDITIONS MIN 300 Hz to 3 kHz Gain G i relative l ti to t gain i att 1020 Hz H (see Note 5) Preamp gain P i = 0 dB, dB PGA gain i = 0 dB dB, Sampling rate = 8 kHz Preamp gain P i = 0 dB, dB PGA gain i = 0 dB dB, Sampling rate = 16 kHz –0.25 MAX UNIT 0.25 3.3 kHz –0.4 0.25 3.4 kHz –0.9 –0.25 600 Hz to 6 kHz Gain G i relative l ti to t gain i att 1020 Hz H (see Note 6) TYP –0.25 0.25 6.6 kHz –0.5 0.25 6.8 kHz –1 0.25 dB dB NOTES: 5. When the high-pass filter (HPF) is bypassed, the passband is 0 Hz to 3 kHz. When the HPF is inserted, the passband is 300 Hz to 3 kHz. 6. When the HPF is bypassed, the passband is 0 Hz to 6 kHz. When the HPF is inserted, the passband is 600 Hz to 6 kHz. ADC channel passband frequency characteristics with line input selected PARAMETER Gain G i relative l ti to t gain i att 1020 Hz H (see Note 5) Gain G i relative l ti tto gain i att 1020 H Hz (see Note 6) TEST CONDITIONS MIN PGA gain = 0 dB, Sampling Sam ling rate = 8 kHz, Pole select = 64, 32, 21.3, or 16 kHz 300 Hz to 3 kHz PGA gain = 0 dB, Sampling Sam ling rate = 16 kHz, Pole select = 64, 32, 21.3, or 16 kHz 600 Hz to 6 kHz TYP MAX –0.25 0.25 3.3 kHz –0.4 0.25 3.4 kHz –0.9 0.25 –0.65 0.25 6.6 kHz –0.9 0.25 6.8 kHz –1.5 –0.25 UNIT dB dB NOTES: 5. When the high-pass filter (HPF) is bypassed, the passband is 0 Hz to 3 kHz. When the HPF is inserted, the passband is 300 Hz to 3 kHz. 6. When the HPF is bypassed, the passband is 0 kHz to 6 kHz. When the HPF is inserted, the passband is 600 Hz to 6 kHz. 8 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 ADC characteristics PARAMETER VI(PP) TEST CONDITIONS Peak-input voltage, differential Preamp gain = 0 dB Interchannel and intrachannel isolation (see Note 2) Any input to any input, any input to any output Valid for HSIN, HDIN, and MCIN EG Gain error (with respect res ect to ideal gain) EO(ADC) ADC channel offset error CMRR Common-mode rejection ratio Preamp = 0 dB PGA = 0 dB Idle channel noise Preamp = 0 dB PGA = 0 dB VI = 1020 Hz H PGA gain = 0 dB Valid for LNIN Valid forCIIN MIN TYP MAX 4 80 UNIT V dB –1 1 dB –4 1.5 dB –1.5 4 dB 50 mV 40 dB 30 Channel delay (HPF bypassed) 75 20/fs µV rms s PGA step error VI = 1020 Hz –0.5 0.5 dB NOTE 2. The LNINP and LNINM are sensitive to crosstalk from LNOUTP and LNOUTM. Keep the LNOUT and LNIN signals separated on the printed circuit board. Do not route the LNOUT signals parallel to the LNIN signals. callerID frequency response characteristics (see Figure 1) PARAMETER fco(L) Low-cutoff frequency Ap TEST CONDITIONS Connected as shown in Figure 23 (see Note 7) MIN TYP MAX UNIT 570 Hz Passband gain at 2 kHz 1.5 dB Attenuation from input to IC terminal at 60 Hz –44 dB NOTE 7: All values are applicable when used with external components as shown in Figure 23. AP –3 dB fco(L) 2 kHz Figure 1. Caller ID Frequency Response POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 9 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 DAC dynamic performance characteristics (THD and SNR calculated with bandwidth = Fs/2) PARAMETER TEST CONDITIONS VI = –3 3 dBr (see Note 8) DAC dynamic performance with handset or headset drivers (HSOUT or HDOUT) (see Note 4) VI = –9 9 dBr (see Note 8) VI = –43 43 dBr (see Note 8) VI = –3 3 dBr (see Note 9) DAC dynamic performance with 8-Ω 8 Ω driver (SPOUT) (see Note 4) VI = –13 13 dBr (see Note 9) VI = –43 43 dBr (see Note 9) VI = –3 3 dBr (see Note 8) DAC dynamic y performance with i h line-output li driver di (LNOUT), (LNOUT) 16-kHz pole selected, selected 1000-Hz input signal (see Note 4) VI = –13 13 dBr (see Note 8) VI = –43 43 dBr (see Note 8) VI = –3 3 dBr (see Note 8) DAC dynamic performance y i h line-output li di (LNOUT) with driver (LNOUT), 64-kHz pole selected, selected 1000-Hz input signal (see Note 4) VI = –13 13 dBr (see Note 8) VI = –43 dBr ((see Note 8)) MIN THD 72 SNR 76 THD + N 70 THD 72 SNR 68 THD + N 66 THD 50 SNR 37 THD + N 36 THD 60 SNR 65 THD + N 60 THD 55 SNR 58 THD + N 54 THD 38 SNR 36 THD + N 35 THD 72 SNR 76 THD + N 70 THD 72 SNR 72 THD + N 70 THD 50 SNR 44 THD + N 42 THD 72 SNR 76 THD + N 70 THD 72 SNR 70 THD + N 68 THD 50 SNR 44 THD + N 42 TYP MAX UNIT dB dB dB dB NOTES: 4. The input level corresponds to TSNR mask corner points in specification G.712. 8. The input signal is the digital equivalent of a sine wave (digital full scale = 0 dBr). A 0-dBr or full-scale digital input results in a 4-V(P-P) differential output. 9. The input signal is the digital equivalent of a sine wave (digital full scale = 0 dBr). A 0-dBr or full-scale digital input results in a 5-V(P-P) differential output. 10 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 DAC channel transfer response characteristics over recommended ranges of supply voltage and operating free-air temperature (see Note 8), with DAC connected to handset (HSOUT) or headset (HDOUT) drivers PARAMETER TEST CONDITIONS Sampling Sam ling rate = 8 kHz Gain relative to gain at 1020 Hz Sampling g rate = 16 kHz MIN TYP MAX 0 Hz to 3 kHz –0.25 0.25 3.3 kHz –0.35 0.25 3.4 kHz –0.9 –0.25 4 kHz –25 4.6 kHz and above –68 0 Hz to 6 kHz –0.25 0.25 6.6 kHz –0.35 0.25 6.8 kHz –0.9 –0.25 8 kHz –25 9.2 kHz and above –68 UNIT dB NOTES: 8. The input signal is the digital equivalent of a sine wave (digital full scale = 0 dBr). A 0-dBr or full-scale digital input results in a 4-V(P-P) differential output. DAC channel passband frequency characteristics with DAC connected to 8-Ω speaker driver (SPOUT) (see Note 9) PARAMETER TEST CONDITIONS MIN 0 Hz to 3 kHz Sampling Sam ling rate = 8 kHz Gain relative to gain at 1020 Hz Sampling g rate = 16 kHz TYP MAX –0.28 0.25 3.3 kHz –0.4 0.15 3.4 kHz –1.2 0.25 0 Hz to 6 kHz –0.7 0.25 6.6 kHz –0.8 0.25 6.8 kHz –1.35 –0.1 UNIT dB NOTES: 9. The input signal is the digital equivalent of a sine wave (digital full scale = 0 dBr). A 0-dBr or full-scale digital input results in a 5-V(P-P) differential output. DAC channel passband frequency characteristics with DAC connected to line output driver (LNOUT) (see Note 8 and Note 10) PARAMETER TEST CONDITIONS Sampling rate = 8 kHz, Pole select = 64 kHz, 32 kHz, 21.3 kHz, or 16 kHz Gain relative to gain at 1.02 kHz g Sampling S li rate t = 16 kH kHz, Pole select = 64 kHz kHz, 32 kHz kHz, or 21 21.3 3 kHz Sampling g rate = 16 kHz,, Pole select = 16 kHz MIN TYP MAX 0 Hz to 3 kHz –0.25 0.25 3.3 kHz –0.35 0.25 3.4 kHz –1.0 0.25 0 Hz to 6 kHz –0.47 0.25 6.6 kHz –0.63 0.25 6.8 kHz –1.16 0.25 0 Hz to 6 kHz –0.7 0.25 6.6 kHz –0.9 0.25 6.8 kHz –1.45 0.25 UNIT dB NOTES: 8. The input signal is the digital equivalent of a sine wave (digital full scale = 0 dBr). A 0-dBr or full-scale digital input results in a 4-V(P-P) differential output. 10. The filter gain is measured with respect to the gain at 1020 Hz. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 11 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 line output out-of-band performance characteristics PARAMETER TEST CONDITIONS MIN Noise measured in 1-kHz bandwidth from 4.6 4 6 kHz to 300 kHz; –10-dB 10 dB input signal; PGA gain = 0 dB, Output Out ut load = 600 Ω Line output out ut out-of-band out of band performance TYP Pole select = 64 kHz 32 Pole select = 32 kHz 20 Pole select = 21.3 kHz 14 Pole select = 16 kHz 11 MAX UNIT µV/√Hz DAC characteristics PARAMETER TEST CONDITIONS Interchannel and intrachannel isolation EG Gain error (with respect to ideal gain) Any input to any output, any output to any output VI = 1020 Hz fs/2 Idle channel noise MIN MAX 80 UNIT dB –0.5 30 Channel delay VO TYP 0.5 dB 75 µV rms 21/fs Analog output voltage (SPOUTP–SPOUTM) Differential for full-scale digital input (see Note 11 and Note 12) ±2.5 Analog output voltage (handset/headset and line interfaces) Differential for full-scale digital input (see Note 11 and Note 12) ±2 PGA step error Input signal = 1020 Hz s V(P–P) (P P) –0.5 0.5 dB NOTES: 11 This amplifier should be used only in differential mode. 12 Common mode: 1.5 V power-supply rejection characteristics (see Note 13) PARAMETER TEST CONDITIONS VDD(1) Supply-voltage rejection ratio, ADC channel, AVDD1, and AVDD2 VDD(4) Supply-voltage rejection ratio, DAC channel MIN TYP MAX UNIT fI = 0 to fs/2 –50 dB fI = 0 to 30 kHz –50 dB NOTE 13 Power-supply rejection measurements are made with both the ADC and the DAC channels idle and a 200-mV peak-to-peak signal applied to the appropriate supply. power supply characteristics PARAMETER TEST CONDITIONS MIN TYP MAX UNIT 30.8 61.5 mA IDD(analog) Codec power-supply current, analog (including drivers); AVDD1, AVDD2 Operating IDD(analog) Codec power-supply current, analog Analog master power down 1.5 mA IDD(digital) Codec power-supply current, digital; DVDD Operating 6.5 mA IDD(digital) Codec power-supply current, digital (hardware power-down mode) PWRDWN terminal = logic 1 2.2 5 mA IDD(speaker) Power-supply current, 8-Ω speaker driver; AVDD3 Operating 200 400 mA IDD(quiescent) 8-Ω driver dc current without swing at output; AVDD3 2 mA IDD(analog) Codec power-supply current, analog (hardware power-down mode) 100 µA 12 POST OFFICE BOX 655303 PWRDWN terminal = logic 1 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 speaker driver characteristics PARAMETER TEST CONDITIONS VN(PP) Output peak-to-peak voltage (between SPOUTP and SPOUTM) AVDD3 = 3.3 V, Fully differential, 8-Ω load, 0 dBr = full-scale digital input VOO Output offset voltage Fully differential Output power (peak) RI = 8 Ω, AVDD3 = 3.3 V Mute MIN TYP MAX 5 UNIT V(P–P) ±5 mV 390 80 mW dB Maximum capacitive load 25 pF handset and headset driver characteristics PARAMETER VN(PP) VOO TEST CONDITIONS Output peak-to-peak voltage MIN AVDD1, AVDD2 = 3.3 V, Fully differential, 150-Ω load Fully differential Output offset voltage TYP MAX 4 V ±5 Maximum capacitance load mV 100 Mute UNIT 80 pF dB Maximum resistive load 150 Ω MAX UNIT line driver characteristics PARAMETER TEST CONDITIONS VN(PP) Output peak-to-peak voltage AVDD1, AVDD2 = 3.3 V, Fully differential, 600-Ω load, 0-dB gain VOO Output offset voltage Fully differential MIN TYP 4 V ±5 Maximum capacitive load for LNOUT mV 25 Maximum resistive load for LNOUT 600 Mute (neither DAC connected to line driver) 80 pF Ω dB 4-bit DAC characteristics PARAMETER VO Output voltage TEST CONDITIONS AVDD1, AVDD2 = 3.3 V MIN TYP 0 3 ±0.5 Linearity O tp t load Output ts MAX Settling time 50 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 UNIT V LSB 600 Ω 20 pF µs 13 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 mic bias characteristics PARAMETER VO IO TEST CONDITIONS Output voltage At MCBIAS terminal, Sourcing 4 mA Output current, max Source only Output noise 20 Hz to 20 kHz Output PSRR Up to 8 kHz MIN TYP 2.3 MAX 2.7 4 UNIT V mA 60 µVrms –60 dB timing requirements MCLK PARAMETER f TEST CONDITIONS MIN Frequency MAX UNIT 32.768 MHz ±200 Accuracy Duty cycle tr tf TYP 32.768 or 24.576 40% 50% ppm 60% Rise time 8 ns Fall time 8 ns timing requirements digital I/O timing (see Figure 2) PARAMETER TEST CONDITIONS MIN TYP MAX UNIT td(1) Delay time, BCLK↑ to FSYNC↑ (slave mode) 15 ns td(3) Delay time, MCLK↑ to BCLK↑ (slave mode) 29 ns tsu(1) Setup time, DIN valid before BCLK↓ th(1) Hold time, DIN valid after BCLK↓ 10 ns 9 ns switching characteristics digital I/O timing (see Figure 2) PARAMETER TEST CONDITIONS td(1) td(2) Delay time, BCLK↑ to FSYNC↑ (master mode) td(6) Delay time, BCLK↑ to FSYNC↓ td(4) Delay time, BCLK↓ to to DOUT invalid td(5) Delay time, BCLK↑ to DOUT high impedance following last data-bit transfer td(3) Delay time, MCLK↑ to BCLK↑ (master mode) Delay time, BCLK↑ to DOUT valid MIN TYP MAX 5 CL = 20 pF UNIT 10 ns 25 ns 3 ns BCLK low time/ 2 + td(2) ns 25 ns 29 ns 13 reset timing PARAMETER tw th(r) 14 TEST CONDITIONS RESET pulse width MIN TYP 2/MCLK Wait time after RESET • DALLAS, TEXAS 75265 UNIT ns 10 POST OFFICE BOX 655303 MAX µs 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 MCLK (input) td(3) BCLK (input/output) td(1) td(6) td(4) FSYNC (input/output) td(2) DOUT (output) td(5) D15 D14 tsu(1) DIN (input) D15 D14 th(1) Figure 2. Digital I/O Timing for Data Channel POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 15 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 detailed description codecs There are two codecs on the TLV320AIC22C that can be connected to any of the analog inputs or outputs via the internal analog crosspoint. The codecs are full 8-bit pulse-coded modulation (PCM) companded or 16-bit linear codecs that meet G.711 standards and include transmit band-pass and receive low-pass filters (LPFs). A-law/µ-law companding or linear coding and –36 dB to 12 dB of analog gain adjustment, in steps of 1.5 dB for each path, are selectable via the I2C or serial interface. These modes can be selected by programming the appropriate register. In the 8-bit PCM companded mode, the data is zero padded to 15 bits and the 16th bit serves as the valid data bit. analog crosspoint The internal analog crosspoint is a lossless analog switch matrix controlled via the I2C or serial interface. The analog crosspoint allows any source device to be connected to any sink device. Additionally, special summing connections with adjustable loss are included to implement sidetone for the handset and headset ports. A muting function is included on any of the sink devices. The control for the analog crosspoint, defined in the register map, is implemented in such a way that a particular analog input or output can be connected to a codec by setting a single bit. This implies that more than one analog input or output can be connected to a codec at one time. Full performance is ensured for two or fewer inputs and outputs connected to a codec, except in the case of the line output. Connecting the output of both codecs to the line output (LNOUTP and LNOUTM) is not allowed. ADC channel The ADC channel consists of a PGA, an antialiasing filter, a sigma-delta ADC, and a decimation filter. The ADC is an oversampling sigma-delta modulator. The ADC provides high resolution and low-noise performance using oversampling techniques and the noise-shaping advantages of sigma-delta modulators. The analog input signals are amplified and filtered by on-chip buffers and an antialiasing filter before being applied to ADC input. The ADC converts the signal into discrete-output digital words in 2s-complement format, corresponding to the analog signal value at the sampling time. The decimation filter reduces the digital data rate to the sampling rate. This is accomplished by decimating with a ratio equal to the oversampling ratio. The output of this filter is a 15-bit 2s-complement data word, clocking at the selected sample rate. The 16th bit is a data-valid flag. These 15-bit digital words, representing sampled values of the analog input signal, are sent to the host via the serial-port interface. If the ADC reaches its maximum value, a control register flag is set. This bit can be read only via the serial port. The analog-to-digital and digital-to-analog conversions are synchronous. The digital conversion data is transmitted out of the device via the serial interface, with the data-valid flag being transmitted first, followed by the MSB of the conversion data. Data is transmitted on the rising edge of BCLK. The bandwidth of the codec is 3.6 kHz for a sampling rate of 8 kHz, and 7.2 kHz for a sampling rate of 16 kHz. The gain of the ADC input amplifier is programmed in register 3 for codec 1 and register 8 for codec 2. The ADC channel contains an HPF that suppresses power-line frequencies, which can be bypassed by programming the appropriate bits in registers 15 and 16 for codec 1 or codec 2, respectively. 16 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 DAC channel The DAC channel consists of an interpolation filter, a sigma-delta DAC, LPF, and a PGA. The DAC is an oversampling sigma-delta modulator. The DAC performs high-resolution, low-noise, digital-to-analog conversion using oversampling sigma-delta techniques. The DAC receives 16-bit data words (2s complement) from the host via the serial-port interface. Data is latched on the falling edge of BCLK. The most significant bit (MSB) of the digital data is transmitted to the DAC first, ending with the LSB as the last bit. The data is converted to an analog voltage by the sigma-delta DAC, comprising a digital interpolation filter and a digital modulator. The interpolation filter resamples the digital data at a rate of N times the incoming sample rate, where N is the oversampling ratio. The high-speed data output from this filter is applied to the sigma-delta DAC. The DAC output is passed to an internal LPF to complete the signal reconstruction, resulting in an analog signal. This analog signal is buffered and amplified by a differential output driver capable of driving the required load. The gain of the DAC output amplifier is programmed in register 4 for codec 1 and register 9 for codec 2. analog and digital loopback The test capabilities include an analog loopback and a digital loopback. The loopbacks allow the user to test the ADC/DAC channels and can be used for in-circuit system-level tests. The digital loopback feeds the ADC output to the DAC input on the device. The analog loopback loops the DAC output back into the ADC input. power down and reset When the power-down (PWRDWN) terminal is pulled high, the device goes into a power-down mode, where the required analog power-supply current drops to approximately 100 µA and the digital power-supply current drops to approximately 2 mA. This is called the hardware power-down mode†. The serial interface and I2C interface are still enabled. All register values are sustained and the device resumes full-power operation without reinitialization when PWRDWN is pulled low again. PWRDWN resets the counters only and preserves the programmed register contents. After PWRDWN has been pulled low, the user must wait at least two frame synchronizations before communicating control or conversion information. Software control can be used to power down individual codecs. Each codec contains an ADC, a DAC, and a digital filter. Codec power down resets all internal counters, but leaves the contents of the programmable control registers unchanged. Analog circuitry and the analog power-supply current are not affected when programming codec power-down mode. Codec power down is achieved by programming register 2 for codec 1 and register 7 for codec 2. An analog master power down can be initiated via software control by programming register 14. Analog master power down is used to power down all of the analog circuitry within the device. This mode is similar to hardware power down in that the required analog power-supply current drops to approximately 100 µA. Table 1 shows the state of the terminals during codec power down and hardware power down. †To obtain the low analog power-down current, the clock should not be running. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 17 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Table 1. Terminal States During Hardware and Codec Power Down TERMINAL NAME 18 STATE DURING CODEC POWER DOWN STATE DURING HARDWARE POWER DOWN HDOUTM Internal common-mode voltage (1.5 V) Floating HDOUTP Internal common-mode voltage (1.5 V) Floating SPOUTP Internal common-mode voltage (1.5 V) Floating SPOUTM Internal common-mode voltage (1.5 V) Floating HSOUTM Internal common-mode voltage (1.5 V) Floating HSOUTP Internal common-mode voltage (1.5 V) Floating LNOUTM Internal common-mode voltage (1.5 V) Floating LNOUTP Internal common-mode voltage (1.5 V) Floating HDINP Normal operation Floating HDINM Normal operation Floating MCINP Normal operation Floating MCINM Normal operation Floating HSINM Normal operation Floating HSINP Normal operation Floating LNIN Normal operation Floating LNINM Normal operation Floating LCDOUT Normal operation Floating MCBIAS Normal operation Floating PWRDWN Normal operation Normal operation SDA Normal operation Normal operation SCL Normal operation Normal operation AD1 Normal operation Normal operation AD0 I2C/SPI Normal operation Normal operation Normal operation Normal operation DOUT Normal operation Normal operation DIN Normal operation Logic high FSYNC Normal operation Normal operation BCLK Normal operation Normal operation VCOM Normal operation Floating M/S Normal operation Normal operation CIINP Pulled to AVSS1/AVSS2 through 40 kΩ CIINM Pulled to AVSS1/AVSS2 through 40 kΩ POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 power down and reset (continued) The capability to individually power down each output driver also is present. Table 2 shows the typical power savings that can be achieved if the associated driver is powered down. Table 2. Powering Down Individual Drivers DRIVER POWERED DOWN REGISTER USED TO POWER DOWN DRIVER TYPICAL POWER SAVINGS WHEN OUTPUT POWERED DOWN Handset 13 3.2 mA Headset 14 3.2 mA Speaker 11 1 mA Line output 14 2.5 mA There are two ways to reset the TLV320AIC22C: D By pulling RESET low, or D By writing to the software reset bits in control registers 2 and/or 7 to reset either codec Asserting RESET low puts the device into a default state with default register settings. After deasserting RESET, the user should wait a minimum of 10 µs before sending control or conversion data to the device. The default register settings are described in the sections titled suggested configuration sequence and register map. After a software reset has been removed, control and conversion data can be sent in the next frame. Asserting a software reset by programming register 2 puts registers 1–5 and 15 in their default settings and resets codec 1. Asserting a software reset by programming register 7 puts registers 6–14, 16, and 17 in their default settings and resets codec 2. microphone bias To operate Electret microphones properly, a bias voltage and current are provided. Typically, the current drawn by the microphone is on the order of 100 µA to 800 µA, and the bias voltage is specified across the microphone at 2.5 V. The bias has good power-supply noise rejection in the audio band, can source 4-mA maximum current, and can be shared between all the microphones. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 19 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 microphone amplifiers There are three microphone preamplifiers, one each for the handset, headset, and speakerphone microphones. The input signals for the handset and headset amplifiers typically are less than 20 mVrms, 100 mV max. The input signals for the speakerphone amplifier typically are less than 2 mVrms, 20 mV max. The amplifiers have a differential input to minimize noise and electromagnetic compatibility (EMC) immunity problems. Three values for the gain for the handset and headset microphones and four values for the gain for the speakerphone microphone are selectable via the I2C or serial interface to meet the requirements in Europe and North America. The frequency response is flat, up to 8 kHz. Table 3. Gain Settings INPUT GAIN SETTINGS Handset microphone preamplifier (HSINP,M) 0 dB, 14 dB, 23 dB, or mute Headset microphone preamplifier (HDINP,M) 0 dB, 14 dB, 23 dB, or mute Speakerphone microphone preamplifier (MCINP,M) 0 dB, 20 dB, 32 dB, 42 dB, or mute By default, the echo gain for the handset and headset are 14 dB. Therefore, a connection exists between the handset and headset inputs (microphones) and their respective outputs (speakers) to implement sidetone. driver amplifiers There are two driver amplifiers that are meant to drive a 150-Ω handset or headset speaker, differentially. The drive amplifier is differential to minimize noise and EMC immunity problems. The frequency response is flat, up to 8 kHz. speakerphone amplifiers The speakerphone speaker impedance is 8 Ω. The drivers are capable of providing a 5-V peak-to-peak differential signal, which means that the peak power is about 390 mW. To achieve this and to minimize noise and EMC immunity problems, the drive amplifier is differential. The frequency response is flat, up to 8 kHz. 4-bit DAC The 4-bit DAC can be used to provide bias to any component on the board, such as a liquid crystal display (LCD). The output of the 4-bit DAC is controlled through the I2C or the serial interface by writing to the four LSBs of control register 12. The register uses 2s-complement data. The DAC has a settling time of about 5 µs, a linearity of ±0.5 LSB, and is a voltage-output DAC. It provides a maximum output of 3 V. For a 16-character by 2-line LCD display module, the contrast control requires 0.2 mA. The input codes and the corresponding output voltages at the LCDOUT terminal are shown in Table 4. 20 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Table 4. 4-Bit DAC Input Code vs Output Voltage INPUT VALUE (DECIMAL) INPUT CODE (2S COMPLEMENT) D3–D0 OUTPUT VOLTAGE 7 0111 2.8125 6 0110 2.625 5 0101 2.4375 4 0100 2.25 3 0011 2.0625 2 0010 1.875 1 0001 1.6875 0 0000 1.5 –1 1111 1.3125 –2 1110 1.125 –3 1101 0.9375 –4 1100 0.75 –5 1011 0.5625 –6 1010 0.375 –7 1001 0.1875 –8 1000 0 caller ID amplifier The caller ID amplifier has a fixed 0-dB gain (typ), attenuates the low-frequency ring signal, and isolates from the line. This input also can be connected to the ADC via the analog crosspoint. line ports The line ports can be connected, via a transformer, to a telephone line. The driver stage is capable of driving a 600-Ω load, differentially, to near rail-to-rail swing. This stage is implemented such that the resistors and capacitors are integrated. Signal levels at the input terminals can be as high as 1.4 Vrms (2 V). The analog pole select option (register 14) allows the user to select the position of the filter pole for the line input and output. serial interface The serial interface is designed to provide glueless interface to the McBSP of a TMS320C54x or TMS320C6x DSP. This interface is used primarily for transferring ADC and DAC data. However, control register information also can be transferred, refer to register programming using the serial interface. The serial interface is a 4-line interface consisting of: D D D D BCLK – bit clock used to transmit and receive data bits FSYNC – frame-synchronization signal that denotes the start of a new frame of data DOUT – output serial data used to transfer ADC data and register information to the attached DSP DIN – input serial data used to transfer DAC data and register control information from the attached DSP The TLV320AIC22C can be configured as a master or a slave (see the master/slave functionality section for a detailed description). When configured as a master device, FSYNC and BCLK are generated by the master codec and input to the DSP. Data is received and transmitted in frames consisting of 256 BCLKs, which is 16, 16-bit time slots. Each frame is subdivided into time slots consisting of 16 BCLKs per time slot. In each frame, two time slots are reserved for control register information and eight time slots are reserved for codec data. The remaining six time slots are unused. A pulse on FSYNC indicates the beginning of a frame. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 21 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 serial interface (continued) The control information is valid only when the serial interface has been selected by connecting the I2C/SPI terminal to logic 0. The frame format is shown in Figure 3, and the timing diagram for the frame is shown in Figure 4. Control Register Address and Data Time Slot 0 Time Slot 1 Data Time Slot 2 Time Slot 3 Time Slot 4 Time Slot 5 Vacant Time Slot 6 Time Slot 7 Time Slot 8 Time Slot 9 Time Slots 10–15 (unused) 256 BCLK Cycles Figure 3. Frame Format Used by the TLV320AIC22C FSYNC One FSYNC Cycle 256 BCLKs BCLK MSB LSB MSB LSB DIN 15 LSB MSB 0 15 0 15 0 15 0 DOUT 15 0 15 0 15 0 15 0 Slot 0 LSB MSB Slot 1 Slot 2 Slot 9 Vacant Figure 4. TLV320AIC22C Frame Format Timing When the serial interface is selected for control (I2C/SPI set to logic 0), the first two time slots after the FSYNC pulse (time slots 0 and 1) are used for sending and receiving control data. The next eight slots are used for actual conversion data sent and received by the codec. Each time slot is 16 bits wide. Data bytes are sent with the first bit representing the MSB. Transmitted data is sent on the rising edge of BCLK, and data being received is latched on the falling edge of BCLK. 22 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register address and data When I2C/SPI is tied to a logic low, the serial interface is selected for controlling the device. Control information is sent and received in time slots 0 and 1. An active-high pulse on FSYNC indicates the start of a frame. The structure of time slots 0 and 1 is shown in Figures 5 and 6. Bit 15 (the MSB) is transmitted or received first. Transmitted data is sent on the rising edge of BCLK and data being received is latched on the falling edge of BCLK. Time slot 0 indicates: D If a read or write operation is occurring D Which device is being accessed D The register address within the device being accessed AD0 (LSB) and AD1 (MSB) form the device address. Up to four TLV320AIC22C devices can be addressed, with addresses ranging from 0 to 3. The five LSBs in time slot 0 are unused. Slot 0: B15 B14 B13 B12–B5 B4–B0 Read/write control 1 = Write, 0 = Read AD1 device address bit (MSB) AD0 device address bit (LSB) Register address (8 bits) Unused MSB 15 LSB 14 13 R/W AD1 AD0 12 11 10 9 8 7 6 5 Register Address 4 3 2 1 0 Unused NOTE A: The register address is the binary equivalent of the register number. Figure 5. Bit Assignment and Definition for Slot 0 Word If bit 15 in slot 0 is a 1, a write operation has been requested by the DSP. The DSP drives data onto the DIN terminal in the next time slot (time slot 1) as follows: D The eight bits of data to be written into the register appear on the first eight bits, with the MSB appearing first. D The next eight bits (eight LSBs) are unused. If bit 15 in slot 0 is a 0, a read operation has been requested. The TLV320AIC22C compares the values of the device address bits, bits 14 and 13 of time slot 0 (AD1 and AD0 bits) to the configuration of the AD1 and AD0 terminals on the device to determine if it is the device being addressed. The device drives data on DOUT if it is the addressed device as follows: D The 8 bits of data from the addressed register appear in the first eight bits, with the MSB appearing first. D The next eight bits (eight LSBs) are unused. Slot 1: B15–B8 B7–B0 Control register data Unused POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 23 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 MSB LSB 15 14 13 12 11 10 9 8 7 6 5 4 Control Register Data 3 2 1 0 Unused NOTES: A. If the register address is 0x00h, no register is updated. B. The default condition is for control information to be updated every frame. If control information is not to be updated every frame, register 17 can be programmed to cause the control slots to appear with N frames of empty control slots between them. The contents of register 17 are equal to N. In this condition, the data in slots 0 and 1 that appear in the N frames between frames with valid control slots are ignored. The default setting for register 17 is 0; control slots appear in every frame. After register 17 is programmed with a nonzero value, the first sequence has N – 1 frames with empty control slots. Figure 6. Bit Assignment and Definition for Slot 1 Word ADC data word A data word occupies one time slot and is 16 bits long. The ADC data word (output on DOUT) can be any of the following: D D D D Data-valid flag + 15 bits of linear data 16 bits of linear data (no data-valid flag) Data-valid flag + A-law or µ-law coded PCM data A-law or µ-law coded PCM data (no data-valid flag) The selection of linear, A-law, or µ-law coding is programmed in register 15, bits 6 and 7. The selection for providing the data-valid flag bit is programmed in register 13 (see the ADC and DAC channel data section for a detailed description of the valid and invalid data). The structure of a data word is shown in Figures 7 and 8. LSB MSB 15 14 Data-Valid Flag Bit 13 12 11 10 9 8 7 6 5 4 3 2 1 0 15-Bit Codec Data (Linear Mode) Zeros When A-Law or µ-Law PCM Mode Selected 8-Bit Codec Data (A-Law, µ-Law PCM Mode) NOTE A: The MSB of the codec data is bit 14 for linear mode and bit 7 for A-law and µ-law. Figure 7. Bit Assignment and Definition for ADC Data Word When the Data-Valid Flag Is Enabled Figure 7 shows the ADC data word format when the data-valid flag is used. The data-valid flag is positioned in bit 15 (the MSB of the data word) and is transmitted first. The flag bit is enabled by programming register 13. Bit 14 of the data word is the MSB of the 15-bit codec data when the linear mode is selected and the data-valid flag is enabled. When A-law or µ-law PCM coding is selected, the eight bits of the PCM data are located with the MSB in the bit-7 location and the LSB in the bit-0 location of the data word. Unused bits are zero when PCM coding is enabled. Bit 15 always is the data-valid flag for both the PCM and linear coding when the data-valid flag is enabled. The selection of linear, A-law, or µ-law coding is programmed in register 15, bits 6 and 7. 24 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 ADC data word (continued) Figure 8 describes the ADC data word format when the data-valid flag is disabled. MSB 15 LSB 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 16-Bit Codec Data (Linear Mode) 8-Bit Codec Data (A-Law, µ-Law PCM Mode) NOTE A: The MSB of the codec data is bit 15 for linear mode and bit 7 for A-law and µ-law. Figure 8. Bit Assignment and Definition for ADC Data Word When the Data-Valid Flag Is Disabled When the data-valid flag is disabled, 16 bits of data are presented in the linear mode, with the MSB in bit 15. When A-law or µ-law PCM coding is selected, the eight bits of the PCM data are located with the MSB in the bit-7 position and the LSB in the bit-0 position. The upper byte of the 16-bit word is ignored for PCM coding and contains zeros. DAC data word The DAC data word (input on DIN) can be any of the following: D 16 bits of linear data D A-law or µ-law coded PCM data The structure of the DAC data word is shown in Figure 9. MSB 15 LSB 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 16-Bit Codec Data (Linear Mode) All Zeros When A-Law or µ-Law PCM Coding Selected 8-Bit Codec Data (A-Law, µ-Law PCM Mode) NOTE A: The MSB of the DAC data is bit 15 for linear mode and bit 7 for A-law and µ-law. Figure 9. Bit Assignment and Definition for DAC Data Word POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 25 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 address terminals The AD1 and AD0 terminals are used to define the address of the codec in the I2C mode and for the serial interface (see the register programming using the I 2C bus section for a detailed description of the I2C mode). For the serial interface, the address determines the time slot used by a certain codec. Provisions are made to support up to four TLV320AIC22Cs connected to a single DSP. With four TLV320AIC22Cs, there are eight slots used for data. Table 5 shows how the time slots used are related to the AD1 and AD0 address lines. Codec 1 in a TLV320AIC22C communicates during the first assigned time slot, based on the AD0 and AD1 configuration, while codec 2 in the same TLV320AIC22C communicates during the second assigned time slot, based on that same AD0/AD1 configuration. Table 5. AD0 and AD1 vs Time-Slot Assignment TLV320AIC22C DEVICE AD1 AD0 TIME SLOT CODEC 1 TIME SLOT CODEC 2 0 0 0 2 3 1 0 1 4 5 2 1 0 6 7 3 1 1 8 9 This address description is used to make the codec register address map unique across the codecs. This is explained further in the following paragraphs. master/slave functionality The TLV320AIC22C can be configured as a master or a slave. A particular codec is configured as the master by tying the M/S terminal (terminal 21) high. Tying M/S low configures the device as a slave. This functionality can be used for connecting multiple TLV320AIC22C devices to a single McBSP port (see Figure 20). Only one device can be a master in such a system. The master device generates the FSYNC and the BCLK signals that are used by the DSP and the remaining TLV320AIC22Cs in the system. The slave devices input the FSYNC and BCLK signals generated by the master device. The TLV320AIC22C also can be used as a stand-alone slave. In this configuration, there is no master TLV320AIC22C device providing the FSYNC and BCLK signals. FSYNC and BCLK are provided by some other device such as a DSP or ASIC. Careful attention must be paid to the relationship between MCLK, FSYNC, and BCLK when using stand-alone slave configurations. When operating the device as a stand-alone slave, the configurations shown in Table 6 must be met. Table 6. Slave-Mode Clock Inputs MCLK INPUT FREQUENCY (MHz) BCLK INPUT FREQUENCY (MHz) FSYNC INPUT FREQUENCY (kHz) REGISTER 12 VALUE (BITS D6–D4, DECIMAL) 24.576 2.048 8 0 24.576 4.096 16 2 32.768 2.048 8 1 32.768 4.096 16 3 All of these signals (BCLK, MCLK, and FSYNC) must be synchronous. The appropriate values for register 12, bits D6–D4, as well as the I values for codec 1 (register 2) and codec 2 (register 7), must be loaded prior to transmitting and receiving valid conversion data to obtain the desired sampling rate (see the channel sampling rates section for a detailed description). 26 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 zero crossing block The zero crossing functionality (programmed in registers 15 and 16) applies whenever the user changes a preamplifier or PGA gain setting. When the user wishes to change a gain setting in a particular channel (ADC or DAC path), the changed gain takes effect when the signal level coming from the particular channel crosses a programmed threshold. The threshold can be specified in registers 15 and 16 for either channel. For example, if the user is talking on the handset and wishes to mute it, the zero crossing block checks the ADC input to see whether the input falls within the programmed range before making the mute effective internally. This is to avoid noise if a sharp change is implemented. Note, in the transmit path, the zero crossing block checks only the ADC input value. If both the handset and the microphone are in use with one ADC channel, and the user wishes to mute only the handset, the zero crossing block does not prevent noise when muting the handset. If the user mutes both the handset and the microphone, then zero crossing is evaluated properly. On the DAC side, the zero crossing is effective in a similar manner. The DAC output is checked to see whether the value is within the programmed range. The mute then becomes effective in the driver where mute has been selected. Deselecting mute is taken care of in the same way. If the user wants to deselect mute, the TLV320AIC22C internally checks to see if the signal level is within the programmed limit and then allows the device to leave mute. Internally, a change in gain setting becomes effective only after the signal level has reached a value near zero. If the signal does not cross the programmed zero-crossing threshold, the gain change automatically occurs after 64/fs seconds. channel sampling rates The TLV320AIC22C can be configured to have standard sampling rates (8 kHz and 16 kHz). The sampling rate (fs) equals the frame synchronization rate (FSYNC). Examples of master clock frequencies, with the derivations of the sampling rates, bit clocks, and the frame synchronization frequencies, are shown in Table 7. The default setting is for a case in which the channel sampling rate and FSYNC are at 8 kHz when an MCLK of 24.576 MHz is provided. The default setting is for register 12 to have bits D6–D4 equal to 000 and registers 1 and 7 to be left in their default configuration. The various parameters for the sampling rates and bit shift clock rates are determined using the following equations: BCLK = See Table 7 FSYNC = BCLK/256 Sample rate = MCLK/(512 × I) Table 7. FSYNC, BCLK, and Sample Rate Derivations With Register Settings D6 D5 D4 MCLK INPUT (MHz) 0† 0† 0† 24.576 BCLK/256 or 8 kHz MCLK/12 or 2.048 MHz MCLK/(512 × I) or 8 kHz 6† 0 0 1 32.768 BCLK/256 or 8 kHz MCLK/16 or 2.048 MHz MCLK/(512 × I) or 8 kHz 8 0 1 0 24.576 BCLK/256 or 16 kHz MCLK/6 or 4.096 MHz MCLK/(512 × I) or 16 kHz 3 0 1 1 32.768 MCLK/(512 × 4) or 16 kHz MCLK/8 or 4.096 MHz MCLK/(512 × I) or 16 kHz 4 REGISTER 12 FSYNC BCLK SAMPLE RATE I † Default setting POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 27 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 ADC and DAC channel data The ADC channel produces 15 bits of 2s complement conversion data in linear mode or 7 bits of zeros and 8 bits of PCM coded data in A-law or µ-law mode, plus a data-valid flag bit which, by default, is enabled. The ADC places a 1 in the data-valid bit for all conversion data, if the valid data flag is enabled in register 13. The DAC uses 16 bits of 2s-complement data or 8 bits of zeros, followed by 8 bits of PCM data as input. No data-valid flag is required for the DAC data. register programming The TLV320AIC22C contains 18 registers that are used to configure the device for the desired operation. Register programming is accomplished in two different ways: D Serial interface (time slots 0 and 1) D I2C bus The I2C/SPI terminal is used to select either interface for programming the device. register programming using the serial interface To program the control registers using the serial interface, I2C/SPI should be tied to logic 0. The frame format and control word description are discussed previously in serial interface section. Time slots 0 and 1 are used for codec register programming and are configured as follows: D Slot 0 – Read/write, physical address of codec register This is the register address appended to the codec address derived from terminals AD0 and AD1. D Slot 1 – Value to be written in the codec register for a write operation For a read operation, the DIN slot 1 is zero stuffed. Depending on the register to be read, the codec puts the register contents on the slot 1 of DOUT in the same frame. The following are examples of programming a TLV320AIC22C whose device address is set to 0 (AD0 = AD1 = 0). example 1: write operation (R/W = 1) Programming control register 15 of a device with address 0x00h, with the data set to 0x23h, results in the following data being driven on the DIN terminal for time slots 0 and 1: Slot 0: 1 00 0000 1111 00000 Slot 1: 0010 0011 0000 0000 Bit 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 1 0 0 0 0 0 0 1 1 1 1 0 0 0 0 0 R/W AD1 AD0 Register Address = 15 Unused Figure 10. DIN Data Stream for Programming Example 1, Slot 0 28 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Bit 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 0 0 1 0 0 0 1 1 0 0 0 0 0 0 0 0 Control Register Data = 0 × 23h Unused Figure 11. DIN Data Stream for Programming Example 1, Slot 1 The data seen on DOUT in these two time slots is: Time slot 0: 0000 0000 0000 0000 Time slot 1: 0000 0000 0000 0000 example 2: read operation (R/W = 0) Requesting a read operation from the device, with address 0x00h and reading control register 15, results in the following data being driven on DIN for time slot 0 and 1: Time slot 0: 0 00 0000 1111 00000 Time slot 1: 0000 0000 0000 0000 Bit 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 0 0 0 0 0 0 0 1 1 1 1 0 0 0 0 0 R/W AD1 AD0 Register Address = 15 Unused Figure 12. DIN Data Stream for Programming Example 2, Slot 0 Bit 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 Control Register Data = 0 (Not Writing to a Register) Unused Figure 13. DIN Data Stream for Programming Example 2, Slot 1 DOUT provides the register data in slot 1. If register 15 had been programmed as in example 1, then DOUT would drive the following data: Time slot 0: 0000 0000 0000 0000 (data is always 0 in time slot 0 on DOUT) Time slot 1: 0010 0011 0000 0000 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 29 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 Bit 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 0 0 1 0 0 0 1 1 0 0 0 0 0 0 0 0 Control Register READ Data = Contents of Register 15 Unused Figure 14. DOUT Data Stream for Programming Example 2, Slot 1 register programming using the I2C bus The I2C interface is provided to program the registers of the TLV320AIC22C in situations where programming them through the serial interface is not convenient. The I2C interface is selected by setting the I2C/SPI terminal to logic high. When the I2C interface is selected, data contained in time slots 0 and 1 in the normal serial data transmission is ignored. The I2C interface consists of the following terminals: D SCL – I2C-bus serial clock. This input is used to synchronize the data transfer from and to the codec. A maximum clock frequency of 400 kHz is allowed. D SDA – I2C-bus serial address/data input/output. This is a bidirectional terminal used to transfer register control address and data into and out of the codec. It is an open-drain terminal and, therefore, requires a pullup resistor to DVDD (typical 10 kΩ for 100 kHz). D AD0 – In I2C mode, AD0 is a chip address bit. D AD1 – In I2C mode, AD1 is a chip address bit. Terminals AD0 and AD1 form the partial chip address. The upper 5 bits (A6–A2) of the 7-bit address field must be 11100. To communicate with a TLV320AIC22C, the LSBs of the chip address field (A1–A0), which is the first byte sent to the TLV320AIC22C, should match the settings of the AD1, AD0 terminals. For normal data transfer, SDA is allowed to change only during SCL low. Changes during SCL high are reserved for indicating the start and stop conditions. Data transfer can be initiated only when the bus is not busy. During data transfer, the data line must remain stable whenever the clock line is high. Changes in the data line while the clock line is high are interpreted as a start or stop condition. Table 8. I2C Bus Status CONDITION 30 STATUS DESCRIPTION A Bus not busy Both data and clock lines remain high. B Start data transfer A high-to-low transition of the SDA line while the clock (SCL) is high determines a start condition. All commands must proceed from a start condition. C Stop data transfer A low-to-high transition of the SDA line while the clock (SCL) is high determines a stop condition. All operations must end with a stop condition. D Data valid The state of the data line represents valid data when, after a start condition, the data line is stable for the duration of the high period of the clock signal. POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 I2C-bus conditions The data on the line must be changed during the low period of the clock signal. There is one clock pulse per bit data and each data transfer is initiated with a start condition and terminated with a stop condition. The host device determines the number of data bytes transferred between the start and stop conditions. When addressed, the TLV320AIC22C generates an acknowledge after the reception of each byte. The host device (microprocessor or DSP) must generate an extra clock pulse, which is associated with this acknowledge bit. The TLV320AIC22C must pull the SDA line down during the acknowledge clock pulse, so that the SDA line is stable low during the high period of the acknowledge-related clock pulse. Setup and hold times must be taken into account. During reads, a host device must signal an end of data to the slave by not generating an acknowledge bit on the last byte that has been clocked out of the slave. In this case, the slave (TLV320AIC22C) must leave the data line high to enable the host device to generate the stop condition. SCL SDA A6 A5 Start A4 A0 R/W ACK 0 0 R7 R6 R5 R0 ACK D7 D6 D5 D0 ACK 0 Slave Address 0 Register Address Data Stop Figure 15. I2C-Bus Write to TLV320AIC22C SCL SDA A6 A5 A0 R/W ACK Start 0 R7 R6 R0 ACK A6 A0 0 R/W ACK 1 Slave Address Register Address D7 D6 D0 0 Slave Drives the Data Slave Address Repeated Start ACK Stop Master Drives ACK and Stop Figure 16. I2C Read From TLV320AIC22C (Protocol A) SCL SDA Start A6 A5 A0 R/W ACK 0 R7 R6 R0 ACK A5 A0 R/W ACK D7 Stop Start 0 Slave Address A6 Register Address 1 Slave Address NOTE A: Slave = TLV320AIC22C 0 D0 ACK Stop Slave Drives Master the Data Drives ACK and Stop Figure 17. I2C Read From TLV320AIC22C (Protocol B) POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 31 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 PGA and preamp gain setting The ability to control when ADC and DAC PGA and preamp gain settings take effect is provided. Programming bit 2 in register 21 controls this feature. When bit 2 in register 21 is set to 0, the PGA and preamp gain settings take effect after the desired gain is programmed in register 3, 4, 8, 9, or 11. To control when a gain change takes place, register 21, bit 2 must be set to 1. No gain change made in registers 3, 4, 8, 9, or 11 takes place after the register is programmed. All previously recognized settings remain in place as long as register 21, bit 2 is set to 1. The gain change is not made until after register 21, bit 2 is set to 0. The timing of when the new ADC PGA and preamp gains actually are initiated is a fixed delay (only when the zero crossing feature is not active). The sequence is: D D D D Register 21, bit D2 is set to 1. New ADC or DAC PGA or preamp gain data is written to registers 3, 4, 8, 9, or 11. Register 21, bit D2 is set to 0. The new gains are applied to the internal circuitry within a maximum of one sample frequency time period (125 ms for an 8-kHz sample rate or 62.5 ms for a 16-kHz sample rate). This feature allows the user to apply preamp and PGA gains simultaneously. This avoids any delays incurred as a result of the PGA and preamp gains being spread across multiple registers. 32 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 register functional summary The following features are register programmable: D D D D D D Software reset Software power down Selection of digital loopback for both channels Selection of analog loopback for both channels Selection of I values for both channels Analog crosspoint control • • • • Analog input for codec 1, selectable from five possible inputs Analog input for codec 2, selectable from five possible inputs Analog output for codec 1, selectable from four possible outputs Analog output for codec 2, selectable from four possible outputs D D D D D Handset input amplifier gain select (mute, 0/14/23 dB) D D D D D D D Linear/A-law/µ-law mode select for both codecs Headset input amplifier gain select (mute, 0/14/23 dB) Handset and headset echo gain select (mute, –12 dB to –24 dB in steps of 2 dB) Microphone input amplifier gain select (mute, 42 dB, 32 dB, 20 dB, or 0 dB) Gain selection for the ADC input PGA (mute, 12 dB to –36 dB in steps of 1.5 dB) and DAC output PGA (mute, 12 dB to –36 dB in steps of 1.5 dB) for both channels Independent power down for drivers 4-bit DAC voltage control HPF bypass for both channels Analog filter pole select (16 kHz, 21.3 kHz, 32 kHz, 64 kHz) Zero crossing enable and threshold Number of frames after which control information is to be sent register map Registers 1–5 and 15 are used to control codec 1. Registers 6–10 and 16 are used to control codec 2. Registers 11–14 and 17 are used to configure the device inputs, outputs, and clocking. Register 21 – Device ID and preamp control POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 33 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 1 (for codec 1) register address = 00000001 D7 D6 D5 D4 X X X X D3 D2 D1 D0 X X X X D1 D0 DESCRIPTION Reserved; this register must remain in the default setting. Reserved; this register must remain in the default setting. Default value: 0101 0000 (D = 6 and N = 0) control register 2 (for codec 1) register address = 00000010 D7 D6 D5 D4 X X X X D3 D2 DESCRIPTION Binary number representing the I register for codec 1 (loaded as I–1) 1 Analog loopback asserted 0 Analog loopback not asserted 1 Digital loopback asserted 0 Digital loopback not asserted 1 Codec 1 power down asserted 0 Codec 1 power down not asserted 1 Software reset (registers 1–5 and 15 are reset to default setting) 0 Software reset not asserted Default value: 0101 0000 (I = 6) 34 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 3 (for codec 1) register address = 00000011 D7 D6 D5 D4 D3 D2 D1 D0 1 1 0 0 0 0 Codec 1 ADC input PGA gain = mute 1 0 0 0 0 1 Codec 1 ADC input PGA gain = 12 dB 1 0 0 0 0 0 Codec 1 ADC input PGA gain = 10.5 dB 0 1 1 1 1 1 Codec 1 ADC input PGA gain = 9 dB 0 1 1 1 1 0 Codec 1 ADC input PGA gain = 7.5 dB 0 1 1 1 0 1 Codec 1 ADC input PGA gain = 6 dB 0 1 1 1 0 0 Codec 1 ADC input PGA gain = 4.5 dB 0 1 1 0 1 1 Codec 1 ADC input PGA gain = 3 dB 0 1 1 0 1 0 Codec 1 ADC input PGA gain = 1.5 dB 0 1 1 0 0 1 Codec 1 ADC input PGA gain = 0 dB 0 1 1 0 0 0 Codec 1 ADC input PGA gain = –1.5 dB 0 1 0 1 1 1 Codec 1 ADC input PGA gain = –3 dB 0 1 0 1 1 0 Codec 1 ADC input PGA gain = –4.5 dB 0 1 0 1 0 1 Codec 1 ADC input PGA gain = –6 dB 0 1 0 1 0 0 Codec 1 ADC input PGA gain = –7.5 dB 0 1 0 0 1 1 Codec 1 ADC input PGA gain = –9 dB 0 1 0 0 1 0 Codec 1 ADC input PGA gain = –10.5 dB 0 1 0 0 0 1 Codec 1 ADC input PGA gain = –12 dB 0 1 0 0 0 0 Codec 1 ADC input PGA gain = –13.5 dB 0 0 1 1 1 1 Codec 1 ADC input PGA gain = –15 dB 0 0 1 1 1 0 Codec 1 ADC input PGA gain = –16.5 dB 0 0 1 1 0 1 Codec 1 ADC input PGA gain = –18 dB 0 0 1 1 0 0 Codec 1 ADC input PGA gain = –19.5 dB 0 0 1 0 1 1 Codec 1 ADC input PGA gain = –21 dB 0 0 1 0 1 0 Codec 1 ADC input PGA gain = –22.5 dB 0 0 1 0 0 1 Codec 1 ADC input PGA gain = –24 dB 0 0 1 0 0 0 Codec 1 ADC input PGA gain = –25.5 dB 0 0 0 1 1 1 Codec 1 ADC input PGA gain = –27 dB 0 0 0 1 1 0 Codec 1 ADC input PGA gain = –28.5 dB 0 0 0 1 0 1 Codec 1 ADC input PGA gain = –30 dB 0 0 0 1 0 0 Codec 1 ADC input PGA gain = –31.5 dB 0 0 0 0 1 1 Codec 1 ADC input PGA gain = –33 dB 0 0 0 0 1 0 Codec 1 ADC input PGA gain = –34.5 dB 0 0 0 0 0 1 Codec 1 ADC input PGA gain = –36 dB 0 0 0 0 0 0 Codec 1 ADC input PGA gain = 0 dB X DESCRIPTION ADC codec overflow indicator 1 Line output (LNOUT) selected for analog output 0 Line output (LNOUT) not selected for analog output Default value: x0000000 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 35 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 4 (for codec 1) register address = 00000100 D7 D6 X X D5 D4 D3 D2 D1 D0 DESCRIPTION 1 1 0 0 0 0 Codec 1 DAC output PGA gain = mute 1 0 0 0 0 1 Codec 1 DAC output PGA gain = 12 dB 1 0 0 0 0 0 Codec 1 DAC output PGA gain = 10.5 dB 0 1 1 1 1 1 Codec 1 DAC output PGA gain = 9 dB 0 1 1 1 1 0 Codec 1 DAC output PGA gain = 7.5 dB 0 1 1 1 0 1 Codec 1 DAC output PGA gain = 6 dB 0 1 1 1 0 0 Codec 1 DAC output PGA gain = 4.5 dB 0 1 1 0 1 1 Codec 1 DAC output PGA gain = 3 dB 0 1 1 0 1 0 Codec 1 DAC output PGA gain = 1.5 dB 0 1 1 0 0 1 Codec 1 DAC output PGA gain = 0 dB 0 1 1 0 0 0 Codec 1 DAC output PGA gain = –1.5 dB 0 1 0 1 1 1 Codec 1 DAC output PGA gain = –3 dB 0 1 0 1 1 0 Codec 1 DAC output PGA gain = –4.5 dB 0 1 0 1 0 1 Codec 1 DAC output PGA gain = –6 dB 0 1 0 1 0 0 Codec 1 DAC output PGA gain = –7.5 dB 0 1 0 0 1 1 Codec 1 DAC output PGA gain = –9 dB 0 1 0 0 1 0 Codec 1 DAC output PGA gain = –10.5 dB 0 1 0 0 0 1 Codec 1 DAC output PGA gain = –12 dB 0 1 0 0 0 0 Codec 1 DAC output PGA gain = –13.5 dB 0 0 1 1 1 1 Codec 1 DAC output PGA gain = –15 dB 0 0 1 1 1 0 Codec 1 DAC output PGA gain = –16.5 dB 0 0 1 1 0 1 Codec 1 DAC output PGA gain = –18 dB 0 0 1 1 0 0 Codec 1 DAC output PGA gain = –19.5 dB 0 0 1 0 1 1 Codec 1 DAC output PGA gain = –21 dB 0 0 1 0 1 0 Codec 1 DAC output PGA gain = –22.5 dB 0 0 1 0 0 1 Codec 1 DAC output PGA gain = –24 dB 0 0 1 0 0 0 Codec 1 DAC output PGA gain = –25.5 dB 0 0 0 1 1 1 Codec 1 DAC output PGA gain = –27 dB 0 0 0 1 1 0 Codec 1 DAC output PGA gain = –28.5 dB 0 0 0 1 0 1 Codec 1 DAC output PGA gain = –30 dB 0 0 0 1 0 0 Codec 1 DAC output PGA gain = –31.5 dB 0 0 0 0 1 1 Codec 1 DAC output PGA gain = –33 dB 0 0 0 0 1 0 Codec 1 DAC output PGA gain = –34.5 dB 0 0 0 0 0 1 Codec 1 DAC output PGA gain = –36 dB 0 0 0 0 0 0 Codec 1 DAC output PGA gain = 0 dB Not used Default value: xx000000 36 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 5 (for codec 1) register address = 00000101 D7 D6 D5 D4 D3 D2 D1 D0 1 DESCRIPTION Handset input (HSIN) selected for analog input 0 Handset input (HSIN) not selected for analog input 1 Headset input (HDIN) selected for analog input 0 Headset input (HDIN) not selected for analog input 1 Microphone input (MCIN) selected for analog input 0 Microphone input (MCIN) not selected for analog input 1 Line input (LNIN) selected for analog input 0 Line input (LNIN) not selected for analog input 1 CallerID amplifier input (CIIN) selected for analog input 0 CallerID amplifier input (CIIN) not selected for analog input 1 Handset output (HSOUT) selected for analog output 0 Handset output (HSOUT) not selected for analog output 1 Headset output (HDOUT) selected for analog output 0 Headset output (HDOUT) not selected for analog output 1 Speaker output (SPOUT) selected for analog output 0 Speaker output (SPOUT) not selected for analog output Default value: 1000 0100 control register 6 (for codec 2) register address = 00000110 D7 D6 D5 D4 X X X X D3 D2 D1 D0 DESCRIPTION Reserved; this register must remain in the default setting. X X X X D1 D0 Reserved; this register must remain in the default setting. Default value: 0101 0000 (D = 6 and N = 0) control register 7 (for codec 2) register address = 00000111 D7 D6 D5 D4 X X X X D3 D2 DESCRIPTION Binary number representing the I register for codec 2 (loaded as I – 1) 1 Analog loopback asserted 0 Analog loopback not asserted 1 Digital loopback asserted 0 Digital loopback not asserted 1 Codec 2 power down asserted 0 Codec 2 power down not asserted 1 Software reset (registers 6–14, 16, and 17 are reset to their default settings) 0 Software reset not asserted Default value: 0101 0000 (I = 6) POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 37 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 8 (for codec 2) register address = 00001000 D7 D6 D5 D4 D3 D2 D1 D0 1 1 0 0 0 0 Codec 2 ADC input PGA gain = mute 1 0 0 0 0 1 Codec 2 ADC input PGA gain = 12 dB 1 0 0 0 0 0 Codec 2 ADC input PGA gain = 10.5 dB 0 1 1 1 1 1 Codec 2 ADC input PGA gain = 9 dB 0 1 1 1 1 0 Codec 2 ADC input PGA gain = 7.5 dB 0 1 1 1 0 1 Codec 2 ADC input PGA gain = 6 dB 0 1 1 1 0 0 Codec 2 ADC input PGA gain = 4.5 dB 0 1 1 0 1 1 Codec 2 ADC input PGA gain = 3 dB 0 1 1 0 1 0 Codec 2 ADC input PGA gain = 1.5 dB 0 1 1 0 0 1 Codec 2 ADC input PGA gain = 0 dB 0 1 1 0 0 0 Codec 2 ADC input PGA gain = –1.5 dB 0 1 0 1 1 1 Codec 2 ADC input PGA gain = –3 dB 0 1 0 1 1 0 Codec 2 ADC input PGA gain = –4.5 dB 0 1 0 1 0 1 Codec 2 ADC input PGA gain = –6 dB 0 1 0 1 0 0 Codec 2 ADC input PGA gain = –7.5 dB 0 1 0 0 1 1 Codec 2 ADC input PGA gain = –9 dB 0 1 0 0 1 0 Codec 2 ADC input PGA gain = –10.5 dB 0 1 0 0 0 1 Codec 2 ADC input PGA gain = –12 dB 0 1 0 0 0 0 Codec 2 ADC input PGA gain = –13.5 dB 0 0 1 1 1 1 Codec 2 ADC input PGA gain = –15 dB 0 0 1 1 1 0 Codec 2 ADC input PGA gain = –16.5 dB 0 0 1 1 0 1 Codec 2 ADC input PGA gain = –18 dB 0 0 1 1 0 0 Codec 2 ADC input PGA gain = –19.5 dB 0 0 1 0 1 1 Codec 2 ADC input PGA gain = –21 dB 0 0 1 0 1 0 Codec 2 ADC input PGA gain = –22.5 dB 0 0 1 0 0 1 Codec 2 ADC input PGA gain = –24 dB 0 0 1 0 0 0 Codec 2 ADC input PGA gain = –25.5 dB 0 0 0 1 1 1 Codec 2 ADC input PGA gain = –27 dB 0 0 0 1 1 0 Codec 2 ADC input PGA gain = –28.5 dB 0 0 0 1 0 1 Codec 2 ADC input PGA gain = –30 dB 0 0 0 1 0 0 Codec 2 ADC input PGA gain = –31.5 dB 0 0 0 0 1 1 Codec 2 ADC input PGA gain = –33 dB 0 0 0 0 1 0 Codec 2 ADC input PGA gain = –34.5 dB 0 0 0 0 0 1 Codec 2 ADC input PGA gain = –36 dB 0 0 0 0 0 0 Codec 2 ADC input PGA gain = 0 dB X DESCRIPTION ADC codec overflow indicator 1 Line output (LNOUT) selected for analog output 0 Line output (LNOUT) not selected for analog output Default value: x0000001 38 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 9 (for codec 2) register address = 00001001 D7 D6 X X D5 D4 D3 D2 D1 D0 DESCRIPTION 1 1 0 0 0 0 Codec 2 DAC output PGA gain = mute 1 0 0 0 0 1 Codec 2 DAC output PGA gain = 12 dB 1 0 0 0 0 0 Codec 2 DAC output PGA gain = 10.5 dB 0 1 1 1 1 1 Codec 2 DAC output PGA gain = 9 dB 0 1 1 1 1 0 Codec 2 DAC output PGA gain = 7.5 dB 0 1 1 1 0 1 Codec 2 DAC output PGA gain = 6 dB 0 1 1 1 0 0 Codec 2 DAC output PGA gain = 4.5 dB 0 1 1 0 1 1 Codec 2 DAC output PGA gain = 3 dB 0 1 1 0 1 0 Codec 2 DAC output PGA gain = 1.5 dB 0 1 1 0 0 1 Codec 2 DAC output PGA gain = 0 dB 0 1 1 0 0 0 Codec 2 DAC output PGA gain = –1.5 dB 0 1 0 1 1 1 Codec 2 DAC output PGA gain = –3 dB 0 1 0 1 1 0 Codec 2 DAC output PGA gain = –4.5 dB 0 1 0 1 0 1 Codec 2 DAC output PGA gain = –6 dB 0 1 0 1 0 0 Codec 2 DAC output PGA gain = –7.5 dB 0 1 0 0 1 1 Codec 2 DAC output PGA gain = –9 dB 0 1 0 0 1 0 Codec 2 DAC output PGA gain = –10.5 dB 0 1 0 0 0 1 Codec 2 DAC output PGA gain = –12 dB 0 1 0 0 0 0 Codec 2 DAC output PGA gain = –13.5 dB 0 0 1 1 1 1 Codec 2 DAC output PGA gain = –15 dB 0 0 1 1 1 0 Codec 2 DAC output PGA gain = –16.5 dB 0 0 1 1 0 1 Codec 2 DAC output PGA gain = –18 dB 0 0 1 1 0 0 Codec 2 DAC output PGA gain = –19.5 dB 0 0 1 0 1 1 Codec 2 DAC output PGA gain = –21 dB 0 0 1 0 1 0 Codec 2 DAC output PGA gain = –22.5 dB 0 0 1 0 0 1 Codec 2 DAC output PGA gain = –24 dB 0 0 1 0 0 0 Codec 2 DAC output PGA gain = –25.5 dB 0 0 0 1 1 1 Codec 2 DAC output PGA gain = –27 dB 0 0 0 1 1 0 Codec 2 DAC output PGA gain = –28.5 dB 0 0 0 1 0 1 Codec 2 DAC output PGA gain = –30 dB 0 0 0 1 0 0 Codec 2 DAC output PGA gain = –31.5 dB 0 0 0 0 1 1 Codec 2 DAC output PGA gain = –33 dB 0 0 0 0 1 0 Codec 2 DAC output PGA gain = –34.5 dB 0 0 0 0 0 1 Codec 2 DAC output PGA gain = –36 dB 0 0 0 0 0 0 Codec 2 DAC output PGA gain = 0 dB Don’t care Default value: xx000000 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 39 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 10 (for codec 2) register address = 00001010 D7 D6 D5 D4 D3 D2 D1 D0 1 DESCRIPTION Handset input (HSIN) selected for analog input 0 Handset input (HSIN) not selected for analog input 1 Headset input (HDIN) selected for analog input 0 Headset input (HDIN) not selected for analog input 1 Microphone input (MCIN) selected for analog input 0 Microphone input (MCIN) not selected for analog input 1 Line input (LNIN) selected for analog input 0 Line input (LNIN) not selected for analog input 1 Caller ID amplifier input (CIIN) selected for analog input 0 Caller ID amplifier input (CIIN) not selected for analog input 1 Handset output (HSOUT) selected for analog output 0 Handset output (HSOUT) not selected for analog output 1 Headset output (HDOUT) selected for analog output 0 Headset output (HDOUT) not selected for analog output 1 Speaker output (SPOUT) selected for analog output 0 Speaker output (SPOUT) not selected for analog output Default value: 0001 0000 control register 11 register address = 00001011 D7 D6 0 0 D5 D4 D3 D2 D1 D0 Handset input amplifier gain = 14 dB DESCRIPTION 0 1 Handset input amplifier gain = 23 dB 1 0 Handset input amplifier gain = mute 1 1 Handset input amplifier gain = 0 dB 0 0 Headset input amplifier gain = 14 dB 0 1 Headset input amplifier gain = 23 dB 1 0 Headset input amplifier gain = mute 1 1 Headset input amplifier gain = 0 dB 0 0 0 Microphone input amplifier gain = 32 dB 0 0 1 Microphone input amplifier gain = 20 dB 0 1 0 Microphone input amplifier gain = 42 dB 0 1 1 Microphone input amplifier gain = 0 dB 1 1 1 Microphone input amplifier gain = mute 1 Speaker output powered down (mute) 0 Speaker output enabled Default value: 0000 0001 40 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 12 register address = 00001100 D7 D6 D5 D4 D3 D2 D1 D0 0 0 0 FSYNC is 8 kHz for MCLK = 24.576 MHz 0 0 1 FSYNC is 8 kHz for MCLK = 32.768 MHz 0 1 0 FSYNC is 16 kHz for MCLK = 24.576 MHz 0 1 1 FSYNC is 16 kHz for MCLK = 32.768 MHz 1 0 0 Reserved 1 0 1 FSYNC is 64 kHz for MCLK = 32.768 MHz and 48 kHz for MCLK = 24.576 MHz 1 1 0 Reserved 1 1 1 Reserved X DESCRIPTION Don’t care X X X X D2 D1 D0 LCD DAC output voltage = 1.5 + (3/16) × (decimal value) Default value: x000 0000 control register 13 register address = 00001101 D7 D6 D5 D4 D3 1 DESCRIPTION Handset output powered down (mute) 0 Handset output enabled X Don’t care X Don’t care 0 0 0 Handset echo gain = –12 dB 0 0 1 Handset echo gain = –14 dB 0 1 0 Handset echo gain = –16 dB 0 1 1 Handset echo gain = –18 dB 1 0 0 Handset echo gain = –20 dB 1 0 1 Handset echo gain = –22 dB 1 1 0 Handset echo gain = –24 dB 1 1 1 Handset echo gain = mute 1 Data-valid flag is disabled. 0 Data-valid flag is enabled. The fifteenth bit (MSB) of the 16-bit data word transmitted from the TLV320AIC22C indicates that the data is valid (bit 15 = 1) or invalid (bit 15 = 0). 1 Version ID bit. Device is a TLV320AIC22C. 0 Version ID bit. Device is a TLV320AIC22C (see Note 14). NOTE 14: Bit 0 is not a read-only bit in the older TLV320AIC22 devices. Therefore, the version ID bit should be read immediately after power is applied or the device is reset. The version ID bit is a read-only bit in the TLV320AIC22C. Default value: 0xx0 0001 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 41 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 14 register address = 00001110 D7 D6 D5 D4 D3 D2 D1 D0 1 DESCRIPTION Headset output powered down (mute) 0 Headset output enabled 0 0 Analog pole-select for line amplifier. Filter pole at 64 kHz. 0 1 Analog pole-select for line amplifier. Filter pole at 32 kHz. 1 0 Analog pole-select for line amplifier. Filter pole at 21.3 kHz. 1 1 Analog pole-select for line amplifier. Filter pole at 16 kHz. 0 0 0 Headset echo gain = –12 dB 0 0 1 Headset echo gain = –14 dB 0 1 0 Headset echo gain = –16 dB 0 1 1 Headset echo gain = –18 dB 1 0 0 Headset echo gain = –20 dB 1 0 1 Headset echo gain = –22 dB 1 1 0 Headset echo gain = –24 dB 1 1 1 Headset echo gain = mute Analog master power down active. Entire analog section is powered down. 1 0 Analog master power down not active 1 Line output, line input amplifiers powered down; VCOM is floating. 0 Line output, line input, and VCOM are enabled. Default value: 0000 0000 control register 15 (for codec 1) register address = 00001111 D7 D6 0 X D5 D4 D3 D2 D1 D0 Linear mode selected 1 0 A-law mode selected 1 1 µ-law mode selected 1 DESCRIPTION Zero crossing disabled for ADC 0 Zero crossing enabled for ADC 1 Zero crossing disabled for DAC 0 Zero crossing enabled for DAC 1 ADC channel HPF bypassed 0 ADC channel HPF not bypassed 1 Zero crossing disabled 0 Zero crossing enabled 0 0 Number of LSBs used to determine the zero crossing threshold = 6 0 1 Number of LSBs used to determine the zero crossing threshold = 4 1 0 Number of LSBs used to determine the zero crossing threshold = 8 1 1 Number of LSBs used to determine the zero crossing threshold = 10 Default value: 00xx 0000 42 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 16 (for codec 2) register address = 00010000 D7 D6 D5 D4 D3 D2 D1 D0 DESCRIPTION 0 X Linear mode selected 1 0 A-law mode selected 1 1 µ-law mode selected 1 Zero crossing disabled for ADC 0 Zero crossing enabled for ADC 1 Zero crossing disabled for DAC 0 Zero crossing enabled for DAC 1 ADC channel HPF bypassed 0 ADC channel HPF not bypassed 1 Zero crossing disabled 0 Zero crossing enabled 0 0 Number of LSBs used to determine the zero crossing threshold = 6 0 1 Number of LSBs used to determine the zero crossing threshold = 4 1 0 Number of LSBs used to determine the zero crossing threshold = 8 1 1 Number of LSBs used to determine the zero crossing threshold = 10 Default value: 00xx 0000 control register 17 register address = 00010001 D7 D6 D5 D4 D3 D2 D1 D0 X X X X X X X X DESCRIPTION Number of frames that do not contain control information present between frames containing control information. Loading zero makes control information present in every frame. Default value: 0000 0000 control register 18 register address = 0001 0010 D7 D6 D5 D4 D3 D2 D1 D0 0 0 0 0 0 0 0 0 DESCRIPTION Reserved Default value: 0000 0000 control register 19 register address = 0001 0011 D7 D6 D5 D4 D3 D2 D1 D0 0 0 0 0 0 0 0 0 DESCRIPTION Reserved Default value: 0000 0000 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 43 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 control register 20 register address = 0001 0100 D7 D6 D5 D4 D3 D2 D1 D0 0 0 0 0 0 0 0 0 D2 D1 D0 DESCRIPTION Reserved Default value: 0000 0000 control register 21 register address = 0001 0101 D7 V D6 V D5 V D4 D3 DESCRIPTION AIC22C version. Binary representation of the version of the AIC22C. 0000 is not a valid combination. V X Reserved 1 PGA/preamplification update control. If there have been any changes to the PGA or preamp gain settings in register 3, 4, 8, 9, or 11 after this bit is set to a 1, then the new values are not immediately read by the device. The previous values remain active until this bit is set to 0. 0 PGA/preamplification update control. The PGA and preamp gain settings in registers 3, 4, 8, 9, and 11 are read by the device immediately after being written. X X Reserved Default value: 0001 1000 44 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 APPLICATION INFORMATION TLV320AIC22C-to-DSP interface The TLV320AIC22C interfaces gluelessly to the McBSP port of a C54x or C6x TI DSP. Figure 18 shows a single TLV320AIC22C connected to a C54x or C6x TI DSP. Figure 19 shows multiple TLV320AIC22Cs connected to a single McBSP port (master/slave functionality). DX DR FSX FSR CLKX CLKR MCLK FSYNC From Oscillator DIN TLV320AIC22C (Slots 2, 3) DOUT BCLK M/S DVDD AD1 AD0 TMS320C54x TMS320C6x Figure 18. TLV320AIC22Cs Interface to McBSP Port of C54x or C6x DSP POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 45 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 APPLICATION INFORMATION From Oscillator DR DX FSX FSR MCLK FSYNC DIN TLV320AIC22C (Master Slots 2, 3) DOUT CLKX 3.3 V BCLK CLKR AD1 AD0 M/S TMS320C54x TMS320C6x MCLK FSYNC DIN TLV320AIC22C (Slave Slots 4, 5) DOUT BCLK M/S AD1 AD0 3.3 V MCLK FSYNC DIN TLV320AIC22C (Slave Slots 6, 7) DOUT BCLK M/S AD1 AD0 3.3 V MCLK FSYNC DIN TLV320AIC22C (Slave Slots 8, 9) DOUT BCLK M/S AD1 AD0 3.3 V Figure 19. Four TLV320AIC22Cs Cascaded to Provide Eight Channels 46 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 APPLICATION INFORMATION hybrid-circuit external connections The TLV320AIC22C, connected to the telephone line using the LNIN and LNOUT hybrid circuit, is shown in Figure 20. 68 kΩ 68 kΩ LNINP 136 kΩ 300 Ω HYBRID LNINM 68 kΩ Line LNI 68 kΩ 136 kΩ LNOUTP 600 Ω 300 Ω LNOUTM Figure 20. Hybrid-Circuit External Connections POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 47 
       


   SPAS041B – OCTOBER 2001 – REVISED JANUARY 2003 APPLICATION INFORMATION microphone, handset, and headset external connections The microphone, handset, and headset external connections are shown in Figure 21. The suggested discrete components, with their values, also are included. 220 Ω MCBIAS 4 mA Max, 2.5 V MIC Preamplifier 0.22 µF 33 µF AVSS 20 kΩ MCINM