CS42L51 Low Power, Stereo CODEC with Headphone Amp DIGITAL to ANALOG FEATURES ANALOG to DIGITAL FEATURES 98 dB Dynamic Range (A-wtd) 98 dB Dynamic Range (A-wtd) -86 dB THD+N -88 dB THD+N Analog Gain Controls - +32 dB or +16 dB MIC Pre-Amplifiers Headphone Amplifier - GND Centered - - - - - On-Chip Charge Pump Provides -VA_HP No DC-Blocking Capacitor Required 46 mW Power Into Stereo 16 @ 1.8 V 88 mW Power Into Stereo 16 @ 2.5 V -75 dB THD+N - +20 dB Digital Boost Programmable Automatic Level Control (ALC) Digital Signal Processing Engine - - - - Analog Programmable Gain Amplifier (PGA) Bass & Treble Tone Control, De-Emphasis PCM + ADC Mix w/Independent Vol Control Master Digital Volume Control Soft Ramp & Zero Cross Transitions - Noise Gate for Noise Suppression - Programmable Threshold and Attack/Release Rates Independent Channel Control Digital Volume Control Beep Generator - Tone Selections Across Two Octaves High-Pass Filter Disable for DC Measurements - Separate Volume Control Dual MIC Inputs - Programmable On & Off Time Intervals - Programmable, Low Noise MIC Bias Levels - Continuous, Periodic or One-Shot Beep Selections - Differential MIC Mix for Common Mode Noise Rejection Stereo 3:1 Analog Input MUX Very Low 64 Fs Oversampling Clock Reduces Programmable Peak-Detect and Limiter Power Consumption Pop and Click Suppression 1.8 V to 3.3 V 1.8 V to 2.5 V 1.8 V to 2.5 V MUX Serial Audio Input Serial Audio Output Switched Capacitor DAC and Filter Headphone Amp - GND Centered Left HP Out Switched Capacitor DAC and Filter Headphone Amp - GND Centered Right HP Out Charge Pump ALC Volume Controls High Pass Filters Multibit Oversampling ADC MUX Stereo Input 1 Stereo Input 2 PGA MUX Multibit Oversampling ADC Register Configuration http://www.cirrus.com Multibit Modulator MUX PCM Serial Interface Reset Level Translator Hardware Mode or I2C & SPI Software Mode Control Data Beep Generator Digital Signal Processing Engine 1.8 V to 2.5 V MUX ALC Copyright (c) Cirrus Logic, Inc. 2006 (All Rights Reserved) PGA +32 dB Stereo Input 3 / Mic Input 1 & 2 +32 dB MIC Bias JULY '06 DS679F1 CS42L51 SYSTEM FEATURES GENERAL DESCRIPTION 24-bit Converters The CS42L51 is a highly integrated, 24-bit, 96 kHz, low power stereo CODEC. Based on multi-bit, delta-sigma modulation, it allows infinite sample rate adjustment between 4 kHz and 96 kHz. Both the ADC and DAC offer many features suitable for low power, portable system applications. 4 kHz to 96 kHz Sample Rate Multi-bit Delta Sigma Architecture Low Power Operation - Stereo Playback: 12.93 mW @ 1.8 V - Stereo Record and Playback: 20.18 mW @ 1.8 V Variable Power Supplies - 1.8 V to 2.5 V Digital & Analog - 1.8 V to 3.3 V Interface Logic Power Down Management - ADC, DAC, CODEC, MIC Pre-Amplifier, PGA Software Mode (IC(R) & SPITM Control) Hardware Mode (Stand-Alone Control) Digital Routing/Mixes: - Analog Out = ADC + Digital In - Digital Out = ADC + Digital In - Internal Digital Loopback - Mono Mixes Flexible Clocking Options - Master or Slave Operation - High-Impedance Digital Output Option (for easy MUXing between CODEC and Other Data Sources) - Quarter-Speed Mode - (i.e. Allows 8 kHz Fs while maintaining a flat noise floor up to 16 kHz) APPLICATIONS HDD & Flash-Based Portable Audio Players MD Players/Recorders PDAs Personal Media Players Portable Game Consoles Digital Voice Recorders Digital Camcorders The ADC input path allows independent channel control of a number of features. An input multiplexer selects between line-level or microphone level inputs for each channel. The microphone input path includes a selectable programmable-gain pre-amplifier stage and a low noise MIC bias voltage supply. A PGA is available for line or microphone inputs and provides analog gain with soft ramp and zero cross transitions. The ADC also features a digital volume attenuator with soft ramp transitions. A programmable ALC and Noise Gate monitor the input signals and adjust the volume levels appropriately. The DAC output path includes a digital signal processing engine. Tone Control provides bass and treble adjustment of four selectable corner frequencies. The Mixer allows independent volume control for both the ADC mix and the PCM mix, as well as a master digital volume control for the analog output. All volume level changes may be configured to occur on soft ramp and zero cross transitions. The DAC also includes de-emphasis, limiting functions and a beep generator delivering tones selectable across a range of two full octaves. The stereo headphone amplifier is powered from a separate positive supply and the integrated charge pump provides a negative supply. This allows a ground-centered analog output with a wide signal swing and eliminates external DC-blocking capacitors. In addition to its many features, the CS42L51 operates from a low-voltage analog and digital core, making this CODEC ideal for portable systems that require extremely low power consumption in a minimal amount of space. The CS42L51 is available in a 32-pin QFN package in both Commercial (-10 to +70 C) and Automotive grades (-40 to +85 C). The CDB42L51 Customer Demonstration board is also available for device evaluation and implementation suggestions. Please see "Ordering Information" on page 85 for complete details. Digital Cameras Smart Phones 2 DS679F1 CS42L51 TABLE OF CONTENTS 1. PIN DESCRIPTIONS - SOFTWARE (HARDWARE) MODE .................................................................. 7 1.1 Digital I/O Pin Characteristics ........................................................................................................... 9 2. TYPICAL CONNECTION DIAGRAMS ................................................................................................. 10 3. CHARACTERISTIC AND SPECIFICATION TABLES ......................................................................... 12 SPECIFIED OPERATING CONDITIONS ............................................................................................. 12 ABSOLUTE MAXIMUM RATINGS ....................................................................................................... 12 ANALOG INPUT CHARACTERISTICS (COMMERCIAL - CNZ) .......................................................... 13 ANALOG INPUT CHARACTERISTICS (AUTOMOTIVE - DNZ) .......................................................... 14 ADC DIGITAL FILTER CHARACTERISTICS ....................................................................................... 15 ANALOG OUTPUT CHARACTERISTICS (COMMERCIAL - CNZ) ...................................................... 16 ANALOG OUTPUT CHARACTERISTICS (AUTOMOTIVE - DNZ) ...................................................... 17 LINE OUTPUT VOLTAGE CHARACTERISTICS ................................................................................. 18 HEADPHONE OUTPUT POWER CHARACTERISTICS ...................................................................... 19 COMBINED DAC INTERPOLATION & ON-CHIP ANALOG FILTER RESPONSE .............................. 20 SWITCHING SPECIFICATIONS - SERIAL PORT ............................................................................... 20 SWITCHING SPECIFICATIONS - IC(R) CONTROL PORT .................................................................. 22 SWITCHING CHARACTERISTICS - SPITM CONTROL PORT ............................................................ 23 DC ELECTRICAL CHARACTERISTICS .............................................................................................. 24 DIGITAL INTERFACE SPECIFICATIONS & CHARACTERISTICS ..................................................... 24 POWER CONSUMPTION .................................................................................................................... 25 4. APPLICATIONS ................................................................................................................................... 26 4.1 Overview ......................................................................................................................................... 26 4.1.1 Architecture ........................................................................................................................... 26 4.1.2 Line & MIC Inputs .................................................................................................................. 26 4.1.3 Line & Headphone Outputs ................................................................................................... 26 4.1.4 Signal Processing Engine ..................................................................................................... 26 4.1.5 Beep Generator ..................................................................................................................... 26 4.1.6 Device Control (Hardware or Software Mode) ...................................................................... 26 4.1.7 Power Management .............................................................................................................. 26 4.2 Hardware Mode .............................................................................................................................. 27 4.3 Analog Inputs ................................................................................................................................. 28 4.3.1 Digital Code, Offset & DC Measurement ............................................................................... 28 4.3.2 High-Pass Filter and DC Offset Calibration ........................................................................... 29 4.3.3 Digital Routing ....................................................................................................................... 29 4.3.4 Differential Inputs .................................................................................................................. 29 4.3.4.1 External Passive Components ................................................................................... 29 4.3.5 Analog Input Multiplexer ........................................................................................................ 31 4.3.6 MIC & PGA Gain ................................................................................................................... 31 4.3.7 Automatic Level Control (ALC) .............................................................................................. 32 4.3.8 Noise Gate ............................................................................................................................ 33 4.4 Analog Outputs ............................................................................................................................... 34 4.4.1 De-Emphasis Filter ................................................................................................................ 34 4.4.2 Volume Controls .................................................................................................................... 35 4.4.3 Mono Channel Mixer ............................................................................................................. 35 4.4.4 Beep Generator ..................................................................................................................... 35 4.4.5 Tone Control .......................................................................................................................... 36 4.4.6 Limiter .................................................................................................................................... 36 4.4.7 Line-Level Outputs and Filtering ........................................................................................... 37 4.4.8 On-Chip Charge Pump .......................................................................................................... 38 4.5 Serial Port Clocking ........................................................................................................................ 38 4.5.1 Slave ..................................................................................................................................... 39 4.5.2 Master ................................................................................................................................... 39 DS679F1 3 CS42L51 4.5.3 High-Impedance Digital Output ............................................................................................. 40 4.5.4 Quarter- and Half-Speed Mode ............................................................................................. 40 4.6 Digital Interface Formats ................................................................................................................ 40 4.7 Initialization ..................................................................................................................................... 41 4.8 Recommended Power-Up Sequence ............................................................................................. 41 4.9 Recommended Power-Down Sequence ........................................................................................ 42 4.10 Software Mode ............................................................................................................................. 43 4.10.1 SPI Control .......................................................................................................................... 43 4.10.2 IC Control ........................................................................................................................... 43 4.10.3 Memory Address Pointer (MAP) .......................................................................................... 45 4.10.3.1 Map Increment (INCR) ............................................................................................. 45 5. REGISTER QUICK REFERENCE ........................................................................................................ 46 6. REGISTER DESCRIPTION .................................................................................................................. 49 6.1 Chip I.D. and Revision Register (Address 01h) (Read Only) ......................................................... 49 6.2 Power Control 1 (Address 02h) ...................................................................................................... 49 6.3 MIC Power Control & Speed Control (Address 03h) ...................................................................... 50 6.4 Interface Control (Address 04h) ..................................................................................................... 52 6.5 MIC Control (Address 05h) ............................................................................................................. 53 6.6 ADC Control (Address 06h) ............................................................................................................ 54 6.7 ADCx Input Select, Invert & Mute (Address 07h) ........................................................................... 56 6.8 DAC Output Control (Address 08h) ................................................................................................ 57 6.9 DAC Control (Address 09h) ............................................................................................................ 58 6.10 ALCX & PGAX Control: ALCA, PGAA (Address 0Ah) & ALCB, PGAB (Address 0Bh) ............... 59 6.11 ADCx Attenuator: ADCA (Address 0Ch) & ADCB (Address 0Dh) ................................................ 60 6.12 ADCx Mixer Volume Control: ADCA (Address 0Eh) & ADCB (Address 0Fh) .............................. 61 6.13 PCMX Mixer Volume Control: PCMA (Address 10h) & PCMB (Address 11h) ..................................................................................... 62 6.14 Beep Frequency & Timing Configuration (Address 12h) .............................................................. 62 6.15 Beep Off Time & Volume (Address 13h) ...................................................................................... 63 6.16 Beep Configuration & Tone Configuration (Address 14h) ............................................................ 64 6.17 Tone Control (Address 15h) ......................................................................................................... 65 6.18 AOUTx Volume Control: AOUTA (Address 16h) & AOUTB (Address 17h) ................................................................................. 66 6.19 PCM Channel Mixer (Address 18h) .............................................................................................. 67 6.20 Limiter Threshold SZC Disable (Address 19h) ............................................................................. 67 6.21 Limiter Release Rate Register (Address 1Ah) .............................................................................. 69 6.22 Limiter Attack Rate Register (Address 1Bh) ................................................................................. 70 6.23 ALC Enable & Attack Rate (Address 1Ch) ................................................................................... 70 6.24 ALC Release Rate (Address 1Dh) ................................................................................................ 71 6.25 ALC Threshold (Address 1Eh) ...................................................................................................... 71 6.26 Noise Gate Configuration & Misc. (Address 1Fh) ......................................................................... 72 6.27 Status (Address 20h) (Read Only) ............................................................................................... 73 6.28 Charge Pump Frequency (Address 21h) ...................................................................................... 74 7. ANALOG PERFORMANCE PLOTS .................................................................................................... 75 7.1 Headphone THD+N versus Output Power Plots ............................................................................ 75 7.2 Headphone Amplifier Efficiency ...................................................................................................... 77 7.3 ADC_FILT+ Capacitor Effects on THD+N ...................................................................................... 78 8. EXAMPLE SYSTEM CLOCK FREQUENCIES .................................................................................... 79 8.1 Auto Detect Enabled ....................................................................................................................... 79 8.2 Auto Detect Disabled ...................................................................................................................... 80 9. PCB LAYOUT CONSIDERATIONS ..................................................................................................... 81 9.1 Power Supply, Grounding ............................................................................................................... 81 9.2 QFN Thermal Pad .......................................................................................................................... 81 10. ADC & DAC DIGITAL FILTERS ........................................................................................................ 82 4 DS679F1 CS42L51 11. PARAMETER DEFINITIONS .............................................................................................................. 83 12. PACKAGE DIMENSIONS ................................................................................................................. 84 THERMAL CHARACTERISTICS ........................................................................................................ 84 13. ORDERING INFORMATION ............................................................................................................. 85 14. REFERENCES .................................................................................................................................... 85 15. REVISION HISTORY ......................................................................................................................... 86 LIST OF FIGURES Figure 1.Typical Connection Diagram (Software Mode) ........................................................................... 10 Figure 2.Typical Connection Diagram (Hardware Mode) .......................................................................... 11 Figure 3.Headphone Output Test Load ..................................................................................................... 19 Figure 4.Serial Audio Interface Slave Mode Timing .................................................................................. 21 Figure 5.Serial Audio Interface Master Mode Timing ................................................................................ 21 Figure 6.Control Port Timing - IC ............................................................................................................. 22 Figure 7.Control Port Timing - SPI Format ................................................................................................ 23 Figure 8.Analog Input Architecture ............................................................................................................ 28 Figure 9.MIC Input Mix w/Common Mode Rejection ................................................................................. 30 Figure 10.Differential Input ........................................................................................................................ 30 Figure 11.ALC ........................................................................................................................................... 32 Figure 12.Noise Gate Attenuation ............................................................................................................. 33 Figure 13.Output Architecture ................................................................................................................... 34 Figure 14.De-Emphasis Curve .................................................................................................................. 35 Figure 15.Beep Configuration Options ...................................................................................................... 36 Figure 16.Peak Detect & Limiter ............................................................................................................... 37 Figure 17.Master Mode Timing ................................................................................................................. 39 Figure 18.Tri-State Serial Port .................................................................................................................. 40 Figure 19.IS Format ................................................................................................................................. 40 Figure 20.Left-Justified Format ................................................................................................................. 41 Figure 21.Right-Justified Format (DAC only) ............................................................................................ 41 Figure 22.Initialization Flowchart ............................................................................................................... 42 Figure 23.Control Port Timing in SPI Mode .............................................................................................. 43 Figure 24.Control Port Timing, IC Write ................................................................................................... 44 Figure 25.Control Port Timing, IC Read ................................................................................................... 44 Figure 26.AIN & PGA Selection ................................................................................................................ 56 Figure 27.THD+N vs. Output Power per Channel at 1.8 V (16 load) .................................................... 75 Figure 28.THD+N vs. Output Power per Channel at 2.5 V (16 load) .................................................... 75 Figure 29.THD+N vs. Output Power per Channel at 1.8 V (32 load) .................................................... 76 Figure 30.THD+N vs. Output Power per Channel at 2.5 V (32 load) .................................................... 76 Figure 31.Power Dissipation vs. Output Power into Stereo 16 ......................................................................77 Figure 32.Power Dissipation vs. Output Power into Stereo 16 (Log Detail) .......................................... 77 Figure 33.ADC THD+N vs. Frequency w/Capacitor Effects ...................................................................... 78 Figure 34.ADC Passband Ripple .............................................................................................................. 82 Figure 35.ADC Stopband Rejection .......................................................................................................... 82 Figure 36.ADC Transition Band ................................................................................................................ 82 Figure 37.ADC Transition Band Detail ...................................................................................................... 82 Figure 38.DAC Passband Ripple .............................................................................................................. 82 Figure 39.DAC Stopband .......................................................................................................................... 82 Figure 40.DAC Transition Band ................................................................................................................ 82 Figure 41.DAC Transition Band (Detail) .................................................................................................... 82 DS679F1 5 CS42L51 LIST OF TABLES Table 1. I/O Power Rails ............................................................................................................................. 9 Table 2. Hardware Mode Feature Summary ............................................................................................. 27 Table 3. MCLK/LRCK Ratios .................................................................................................................... 39 6 DS679F1 CS42L51 Pin Name # SDIN SCLK MCLK SDOUT (M/S) DGND VD VL RESET 1. PIN DESCRIPTIONS - SOFTWARE (HARDWARE) MODE 32 31 30 29 28 27 26 25 LRCK 1 24 AIN1B SDA/CDIN (MCLKDIV2) 2 23 AIN1A SCL/CCLK (IS/LJ) 3 22 AFILTB ADO/CS (DEM) 4 21 AFILTA VA_HP 5 20 AIN2B/BIAS FLYP 6 19 AIN2A GND_HP 7 18 MICIN2/BIAS/AIN3B FLYN 8 17 MICIN1/AIN3A 9 10 11 12 13 14 15 16 VSS_HP AOUTB AOUTA VA AGND DAC_FILT+ VQ ADC_FILT+ CS42L51 Pin Description Left Right Clock (Input/Output) - Determines which channel, Left or Right, is currently active on the serial audio data line. LRCK 1 SDA/CDIN (MCLKDIV2) 2 SCL/CCLK (IS/LJ) 3 AD0/CS (DEM) 4 VA_HP 5 Analog Power For Headphone (Input) - Positive power for the internal analog headphone section. FLYP 6 Charge Pump Cap Positive Node (Input) - Positive node for the external charge pump capacitor. GND_HP 7 Analog Ground (Input) - Ground reference for the internal headphone/charge pump section. FLYN 8 Charge Pump Cap Negative Node (Input) - Negative node for the external charge pump capacitor. VSS_HP 9 Negative Voltage From Charge Pump (Output) - Negative voltage rail for the internal analog headphone section. DS679F1 Serial Control Data (Input/Output) - SDA is a data I/O in IC Mode. CDIN is the input data line for the control port interface in SPI Mode. MCLK Divide by 2 (Input) - Hardware Mode: Divides the MCLK by 2 prior to all internal circuitry. Serial Control Port Clock (Input) - Serial clock for the serial control port. Interface Format Selection (Input) - Hardware Mode: Selects between IS & Left-Justified interface formats for the ADC & DAC. Address Bit 0 (IC) / Control Port Chip Select (SPI) (Input) - AD0 is a chip address pin in IC Mode; CS is the chip-select signal for SPI format. De-Emphasis (Input) - Hardware Mode: Enables/disables the de-emphasis filter. 7 CS42L51 AOUTB AOUTA 10 11 Analog Audio Output (Output) - The full-scale output level is specified in the DAC Analog Characteristics specification table VA 12 Analog Power (Input) - Positive power for the internal analog section. AGND 13 Analog Ground (Input) - Ground reference for the internal analog section. DAC_FILT+ 14 Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits. VQ 15 Quiescent Voltage (Output) - Filter connection for internal quiescent voltage. ADC_FILT+ 16 Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits. MICIN1/ AIN3A 17 Microphone Input 1 (Input) - The full-scale level is specified in the ADC Analog Characteristics specification table. MICIN2/ BIAS/AIN3B 18 Microphone Input 2 (Input/Output) - The full-scale level is specified in the ADC Analog Characteristics specification table. This pin can also be configured as an output to provide a low noise bias supply for an external microphone. Electrical characteristics are specified in the DC Electrical Characteristics table. AIN2A 19 Analog Input (Input) - The full-scale level is specified in the ADC Analog Characteristics specification table. AIN2B/BIAS 20 Analog Input (Input/Output) - The full-scale level is specified in the ADC Analog Characteristics specification table. This pin can also be configured as an output to provide a low noise bias supply for an external microphone. Electrical characteristics are specified in the DC Electrical Characteristics table. AFILTA AFILTB 21 22 Filter Connection (Output) - Filter connection for the ADC inputs. AIN1A AIN1B 23 24 Analog Input (Input) - The full-scale level is specified in the ADC Analog Characteristics specification table. RESET 25 Reset (Input) - The device enters a low power mode when this pin is driven low. VL 26 Digital Interface Power (Input) - Determines the required signal level for the serial audio interface and host control port. Refer to the Recommended Operating Conditions for appropriate voltages. VD 27 Digital Power (Input) - Positive power for the internal digital section. DGND 28 Digital Ground (Input) - Ground reference for the internal digital section. SDOUT (M/S) 29 Serial Port Master/Slave (Input/Output) - Hardware Mode Startup Option: Selects between Master and Slave Mode for the serial port. MCLK 30 Master Clock (Input) - Clock source for the delta-sigma modulators. SCLK 31 Serial Clock (Input/Output) - Serial clock for the serial audio interface. SDIN 32 Serial Audio Data Input (Input) - Input for two's complement serial audio data. Thermal Pad 8 Serial Audio Data Output (Output) - Output for two's complement serial audio data. - Thermal relief pad for optimized heat dissipation. See "QFN Thermal Pad" on page 81. DS679F1 CS42L51 1.1 Digital I/O Pin Characteristics The logic level for each input should not exceed the maximum ratings for the VL power supply. Pin Name SW/(HW) I/O Driver Receiver RESET Input - 1.8 V - 3.3 V SCL/CCLK (IS/LJ) Input - 1.8 V - 3.3 V, with Hysteresis SDA/CDIN (MCLKDIV2) Input/Output 1.8 V - 3.3 V, CMOS/Open Drain 1.8 V - 3.3 V, with Hysteresis AD0/CS (DEM) Input - 1.8 V - 3.3 V MCLK Input - 1.8 V - 3.3 V LRCK Input/Output 1.8 V - 3.3 V, CMOS 1.8 V - 3.3 V SCLK Input/Output 1.8 V - 3.3 V, CMOS 1.8 V - 3.3 V SDOUT (M/S) Input/Output 1.8 V - 3.3 V, CMOS 1.8 V - 3.3 V SDIN Input - 1.8 V - 3.3 V Table 1. I/O Power Rails DS679F1 9 CS42L51 2. TYPICAL CONNECTION DIAGRAMS See Note 4 +1.8 V or +2.5 V 1 F 0.1 F 0.1 F VD VA 0.1 F +1.8 V or +2.5 V 1 F Note 4: Series resistance in the path of the power supplies must be avoided. Any voltage drop on VA_HP will directly impact the negative charge pump supply (VSS_HP) and result in clipping on the audio output . VA_HP AOUTB ** 1.5 F 1 F ** Headphone Out Left & Right FLYP AOUTA FLYN 0.022 F See Note 5 VSS_HP 1.5 F ** 51.1 ** 1 F GND_HP 470 * *Use low ESR ceramic capacitors. C Note 2 : CS42L51 For best response to Fs/2 : C Rext + 470 C= 4Fs (Rext x 470 ) Speaker Driver Note 5 : Larger capacitors, such as 1.5 F, improves the charge pump performance (and subsequent THD+N) at the full scale output power achieved with gain (G) settings greater than default. AIN1A 1800 pF * 1800 pF * 100 k 100 k 100 AIN2A LRCK SDIN SDOUT 1800 pF * 1 F 1800 pF * AIN2B BIAS1 RESET Right Analog Input 1 MICIN1 AIN3A SDA/CDIN Left Analog Input 2 100 100 k 100 k 100 Right Analog Input 2 1 F SCL/CCLK Microphone Input 100 k 1 F AD0/CS BIAS2 AIN3B/MICIN2 Microphone Bias RL 0.1 F 2k See Note 1 Left Analog Input 1 100 1 F SCLK Digital Audio Processor 1 F AIN1B MCLK +1.8 V, +2.5 V or +3.3 V Rext 470 This circuitry is intended for applications where the CS42L51 connects directly to an unbalanced output of the device. For internal routing applications please see the DAC Analog Output Characteristics section for loading limitations. 2k Rext Line Level Out Left & Right See Note 2 See Note 3 Note 3: The value of R L is dictated by the microphone cartridge. VL ADC_FILT+ 0.1 F Note 1: Resistors are required for IC control port operation DAC_FILT+ 1 F 10 F AGND * AFILTA AFILTB VQ DGND * 150 pF 150 pF 1 F * Capacitors must be C0G or equivalent Figure 1. Typical Connection Diagram (Software Mode) 10 DS679F1 CS42L51 See Note 4 +1.8V or +2.5V 1 F 0.1 F 0.1 F VD VA +1.8V or +2.5V 1 F 0.1 F Note 4: Series resistance in the path of the power supplies (typically used for added filtering) must be avoided. Any voltage drop on VA_HP will directly impact the negative charge pump supply (VSS_HP) and result in clipping on the audio output . VA_HP AOUTB 1 F AOUTA FLYN 1 F Headphone Out Left & Right FLYP ** 0.022 F 51.1 VSS_HP ** GND_HP 470 C * *Use low ESR ceramic capacitors. CS42L51 C Rext Line Level Out Left & Right See Note 2 Rext 470 Speaker Driver MCLK SCLK LRCK AIN1A SDIN VL or DGND (1) SDOUT/ M/S Digital Audio Processor Left Analog Input 1 1800 pF * 1 F 100 100 k 1800 pF * 100 k 100 AIN1B Right Analog Input 1 1 F RESET IS/LJ ADC_FILT+ MCLKDIV2 DAC_FILT+ DEM 1 F AGND * +1.8V, 2.5 V or +3.3V AFILTA AFILTB VQ VL 0.1 F 150 pF 1 F * Capacitors must be C0G or equivalent DGND (1) Pull-up to VL (47 k for Master Mode. Pull-down to DGND for Slave Mode. * 150 pF 10 F Note 2 : This circuitry is intended for applications where the CS 42L51 connects directly to an unbalanced output of the device . For internal routing applications please see the DAC Analog Output Characteristics section for loading limitations . For best response to Fs/2 : C= Rext + 470 4Fs (Rext x 470 ) Figure 2. Typical Connection Diagram (Hardware Mode) DS679F1 11 CS42L51 3. CHARACTERISTIC AND SPECIFICATION TABLES (All Min/Max characteristics and specifications are guaranteed over the Specified Operating Conditions. Typical performance characteristics and specifications are derived from measurements taken at nominal supply voltages and TA = 25 C.) SPECIFIED OPERATING CONDITIONS (AGND=DGND=0 V, all voltages with respect to ground.) Parameters Symbol Min Nom Max Units 1.65 2.37 1.65 2.37 1.65 2.37 1.65 2.37 3.14 1.8 2.5 1.8 2.5 1.8 2.5 1.8 2.5 3.3 1.89 2.63 1.89 2.63 1.89 2.63 1.89 2.63 3.47 V V V V V V V V V -10 -40 - +70 +85 C C DC Power Supply (Note 1) VA Analog Core VA_HP Headphone Amplifier Digital Core VD Serial/Control Port Interface VL Ambient Temperature Commercial - CNZ Automotive - DNZ TA ABSOLUTE MAXIMUM RATINGS (AGND = DGND = 0 V; all voltages with respect to ground.) Parameters Min Max Units -0.3 -0.3 -0.3 3.0 3.0 4.0 V V V Iin - 10 mA Analog Input Voltage VIN AGND-0.7 VA+0.7 V Digital Input Voltage (Note 3) VIND -0.3 VL+ 0.4 V Ambient Operating Temperature (power applied) TA -50 +115 C Storage Temperature Tstg -65 +150 C DC Power Supply Symbol Analog VA, VA_HP VD Digital VL Serial/Control Port Interface Input Current (Note 2) WARNING: Operation at or beyond these limits may result in permanent damage to the device. Normal operation is not guaranteed at these extremes. Notes: 1. The device will operate properly over the full range of the analog, headphone amplifier, digital core and serial/control port interface supplies. 2. Any pin except supplies. Transient currents of up to 100 mA on the analog input pins will not cause SCR latch-up. 3. The maximum over/under voltage is limited by the input current. 12 DS679F1 CS42L51 ANALOG INPUT CHARACTERISTICS (COMMERCIAL - CNZ) (Test Conditions (unless otherwise specified): Input sine wave (relative to digital full scale): 1 kHz through passive input filter; Measurement Bandwidth is 10 Hz to 20 kHz unless otherwise specified. Sample Frequency = 48 kHz) VA = 2.5 V (nominal) Min Typ Max Parameter (Note 5) VA = 1.8 V (nominal) Min Typ Max Unit Analog In to ADC (PGA bypassed) Dynamic Range A-weighted unweighted Total Harmonic Distortion + Noise -1 dBFS -20 dBFS -60 dBFS 93 90 99 96 - 90 87 96 93 - dB dB - -86 -76 -36 -80 - - -84 -73 -33 -78 - dB dB dB Analog In to PGA to ADC Dynamic Range PGA Setting: 0 dB A-weighted unweighted 92 89 98 95 - 89 86 95 92 - dB dB PGA Setting: +12 dB A-weighted unweighted 85 82 91 88 - 82 79 88 85 - dB dB -1 dBFS -60 dBFS - -88 -35 -81 - - -86 -32 -80 - dB dB -1 dBFS - -85 -79 - -83 -77 dB Total Harmonic Distortion + Noise PGA Setting: 0 dB PGA Setting: +12 dB Analog In to MIC Pre-Amp (+16 dB) to PGA to ADC Dynamic Range PGA Setting: 0 dB A-weighted unweighted - 86 83 - - 83 80 - dB dB -1 dBFS - -76 - - -74 - dB Total Harmonic Distortion + Noise PGA Setting: 0 dB Analog In to MIC Pre-Amp (+32 dB) to PGA to ADC Dynamic Range A-weighted unweighted - 78 74 - - 75 71 - dB dB -1 dBFS - -74 - - -71 - dB Interchannel Gain Mismatch - 0.2 - - 0.2 - dB Gain Drift - 100 - - 100 - ppm/C - 352 - - 352 - LSB - 90 - - 90 - dB 0.74*VA 0.75*VA 70 0.78*VA 0.794*VA 0.129*VA 0.022*VA 0.82*VA 0.83*VA - 20 39 50 - PGA Setting: 0 dB Total Harmonic Distortion + Noise PGA Setting: 0 dB Other Characteristics DC Accuracy Offset Error SDOUT Code with HPF On Input Interchannel Isolation DAC Isolation (Note 4) Full-scale Input Voltage Input Impedance (Note 6) DS679F1 0.74*VA ADC PGA (0 dB) 0.75*VA MIC (+16 dB) MIC (+32 dB) ADC PGA MIC 70 0.78*VA 0.82*VA 0.794*VA 0.83*VA 0.129*VA 0.022*VA 20 39 50 - dB Vpp Vpp Vpp Vpp k k k 13 CS42L51 ANALOG INPUT CHARACTERISTICS (AUTOMOTIVE - DNZ) (Test Conditions (unless otherwise specified): Input sine wave (relative to full scale): 1 kHz through passive input filter; Measurement Bandwidth is 10 Hz to 20 kHz unless otherwise specified. Sample Frequency = 48 kHz) VA = 2.5 V (nominal) Min Typ Max Parameter (Note 5) VA = 1.8 V (nominal) Min Typ Max Unit Analog In to ADC Dynamic Range A-weighted unweighted Total Harmonic Distortion + Noise -1 dBFS -20 dBFS -60 dBFS 91 78 99 96 - 88 85 96 93 - dB dB - -86 -76 -36 -78 - - -84 -73 -33 -76 - dB dB dB Analog In to PGA to ADC Dynamic Range PGA Setting: 0 dB A-weighted unweighted 90 87 98 95 - 87 84 95 92 - dB dB PGA Setting: +12 dB A-weighted unweighted 83 80 91 88 - 80 77 88 85 - dB dB -1 dBFS -60 dBFS - -88 -35 -80 - - -86 -32 -78 - dB dB -1 dBFS - -85 -77 - -83 -75 dB Total Harmonic Distortion + Noise PGA Setting: 0 dB PGA Setting: +12 dB Analog In to MIC Pre-Amp (+16 dB) to PGA to ADC Dynamic Range PGA Setting: 0 dB A-weighted unweighted - 86 83 - - 83 80 - dB dB -1 dBFS - -76 - - -74 - dB Total Harmonic Distortion + Noise PGA Setting: 0 dB Analog In to MIC Pre-Amp (+32 dB) to PGA to ADC Dynamic Range A-weighted unweighted - 78 74 - - 75 71 - dB dB -1 dBFS - -74 - - -71 - dB Interchannel Gain Mismatch - 0.1 - - 0.1 - dB Gain Drift - 100 - - 100 - ppm/C - 352 - - 352 - LSB - 90 - - 90 - dB 70 0.78*VA 0.794*VA 0.129*VA 0.022*VA 0.82*VA 0.83*VA 0.74*VA 0.75*VA 70 0.78*VA 0.794*VA 0.129*VA 0.022*VA 0.82*VA 0.83*VA - - 18 40 50 - - PGA Setting: 0 dB Total Harmonic Distortion + Noise PGA Setting: 0 dB Other Characteristics DC Accuracy Offset Error SDOUT Code with HPF On Input Interchannel Isolation DAC Isolation (Note 4) Full-scale Input Voltage Input Impedance (Note 6) ADC 0.74*VA PGA (0 dB) 0.75*VA MIC (+16 dB) MIC (+32 dB) 18 ADC 40 PGA 50 MIC dB Vpp Vpp Vpp Vpp k k k 4. Measured with DAC delivering full-scale output power into 16 . 14 DS679F1 CS42L51 5. Referred to the typical full-scale voltage. Applies to all THD+N and Dynamic Range values in the table. 6. Measured between AINxx and AGND. ADC DIGITAL FILTER CHARACTERISTICS Parameter (Note 7) Min Typ Max Unit 0 - 0.4948 Fs -0.09 - 0.17 dB Stopband 0.6 - - Fs Stopband Attenuation 33 - - dB - 7.6/Fs - s Passband (Frequency Response) Passband Ripple Total Group Delay to -0.1 dB corner High-Pass Filter Characteristics (48 kHz Fs) Frequency Response -3.0 dB -0.13 dB - 3.7 24.2 - Hz Hz Phase Deviation @ 20 Hz - 10 - Deg - - 0.17 dB - 5/Fs 0 s Passband Ripple Filter Settling Time 10 7. Response is clock-dependent and will scale with Fs. Note that the response plots (Figure 33 to Figure 41) have been normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs. HPF parameters are for Fs = 48 kHz. DS679F1 15 CS42L51 ANALOG OUTPUT CHARACTERISTICS (COMMERCIAL - CNZ) (Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load RL = 10 k, CL = 10 pF for the line output (see Figure 3), and test load RL = 16 , CL = 10 pF (see Figure 3) for the headphone output. HP_GAIN[2:0] = 011.) VA = 1.8V (nominal) VA = 2.5V (nominal) Min Typ Max Min Typ Max Parameter (Note 8) Unit RL = 10 k Dynamic Range 18 to 24-Bit A-weighted unweighted A-weighted unweighted 16-Bit 92 89 - 98 95 96 93 - 89 86 - 95 92 93 90 - dB dB dB dB - -86 -75 -35 -86 -73 -33 -78 - - -88 -72 -32 -88 -70 -30 -82 - dB dB dB dB dB dB 92 89 - 98 95 96 93 - 89 86 - 95 92 93 90 - dB dB dB dB - -75 -75 -35 -75 -73 -33 -69 - - -75 -72 -32 -75 -70 -30 -69 - dB dB dB dB dB dB Total Harmonic Distortion + Noise 18 to 24-Bit 0 dB -20 dB -60 dB 0 dB -20 dB -60 dB 16-Bit RL = 16 Dynamic Range 18 to 24-Bit A-weighted unweighted A-weighted unweighted 16-Bit Total Harmonic Distortion + Noise 18 to 24-Bit 0 dB -20 dB -60 dB 0 dB -20 dB -60 dB 16-Bit Other Characteristics for RL = 16 or 10 k Output Parameters (Note 9) Modulation Index (MI) Analog Gain Multiplier (G) Full-scale Output Voltage (2*G*MI*VA) (Note 9) Full-scale Output Power (Note 9) Interchannel Isolation (1 kHz) 16 10 k 0.6787 0.6787 0.6047 0.6047 Refer to Table "Line Output Voltage Characteristics" on page 18 Refer to Table "Headphone Output Power Characteristics" on page 19 80 80 95 93 - Interchannel Gain Mismatch - 0.1 0.25 - 0.1 0.25 Gain Drift - 100 - - 100 - Vpp mW dB dB dB ppm/ C AC-Load Resistance (RL) (Note 10) 16 - - 16 - - Load Capacitance (CL) (Note 10) - - 150 - - 150 pF 16 DS679F1 CS42L51 ANALOG OUTPUT CHARACTERISTICS (AUTOMOTIVE - DNZ) (Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz and 96 kHz; test load RL = 10 k, CL = 10 pF for the line output (see Figure 3), and test load RL = 16 , CL = 10 pF (see Figure 3) for the headphone output. HP_GAIN[2:0] = 011.) VA = 2.5V (nominal) Min Typ Max Parameter (Note 8) VA = 1.8V (nominal) Min Typ Max Unit RL = 10 k Dynamic Range 18 to 24-Bit A-weighted unweighted A-weighted unweighted 16-Bit 90 87 - 98 95 96 93 - 87 84 - 95 92 93 90 - dB dB dB dB - -86 -75 -35 -86 -73 -33 -73 - - -88 -72 -32 -88 -70 -30 -80 - dB dB dB dB dB dB 90 87 - 98 95 96 93 - 87 84 - 95 92 93 90 - dB dB dB dB - -75 -75 -35 -75 -73 -33 -67 - - -75 -72 -32 -75 -70 -30 -67 - dB dB dB dB dB dB Total Harmonic Distortion + Noise 18 to 24-Bit 0 dB -20 dB -60 dB 0 dB -20 dB -60 dB 16-Bit RL = 16 Dynamic Range 18 to 24-Bit A-weighted unweighted A-weighted unweighted 16-Bit Total Harmonic Distortion + Noise 18 to 24-Bit 0 dB -20 dB -60 dB 0 dB -20 dB -60 dB 16-Bit Other Characteristics for RL = 16 or 10 k Output Parameters (Note 9) Modulation Index (MI) Analog Gain Multiplier (G) Full-scale Output Voltage (2*G*MI*VA) (Note 9) Full-scale Output Power (Note 9) Interchannel Isolation (1 kHz) 16 10 k 0.6787 0.6787 0.6047 0.6047 Refer to Table "Line Output Voltage Characteristics" on page 18 Refer to Table "Headphone Output Power Characteristics" on page 19 80 80 95 93 - Interchannel Gain Mismatch - 0.1 0.25 - 0.1 0.25 Gain Drift - 100 - - 100 - Vpp mW dB dB dB ppm/ C AC-Load Resistance (RL) (Note 10) 16 - - 16 - - Load Capacitance (CL) (Note 10) - - 150 - - 150 pF DS679F1 17 CS42L51 LINE OUTPUT VOLTAGE CHARACTERISTICS Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load RL = 10 k, CL = 10 pF (see Figure 3). Parameter VA = 2.5V (nominal) Min Typ Max VA = 1.8V (nominal) Min Typ Max 1.95 - 2.15 - 1.41 - - - Unit AOUTx Voltage Into RL = 10 k HP_GAIN[2:0] Analog Gain (G) 000 0.3959 001 0.4571 010 0.5111 011 (default) 0.6047 100 0.7099 101 0.8399 110 1.0000 111 1.1430 18 VA_HP 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V - 1.34 1.34 1.55 1.55 1.73 1.73 2.05 2.05 2.41 2.41 2.85 2.85 3.39 3.39 (See (Note 11) 3.88 - 0.97 0.97 1.12 1.12 1.25 1.25 1.48 1.48 1.73 1.73 2.05 2.05 2.44 2.44 2.79 2.79 1.55 - Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp Vpp DS679F1 CS42L51 HEADPHONE OUTPUT POWER CHARACTERISTICS Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load RL = 16 , CL = 10 pF (see Figure 3). VA = 1.8V (nominal) Min Typ Max VA = 2.5V (nominal) Min Typ Max Parameter Unit AOUTx Power Into RL = 16 HP_GAIN[2:0] Analog Gain (G) 000 0.3959 001 0.4571 010 0.5111 011 (default) 0.6047 100 0.7099 101 0.8399 110 1.0000 111 1.1430 VA_HP 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V 1.8 V 2.5 V - 14 14 19 19 23 23 (Note 11) 32 (Note 11) 44 - - - - (Note 9, 11) 7 7 10 10 12 12 17 17 23 23 (Note 9) 32 - mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms mWrms 8. One-half LSB of triangular PDF dither is added to data. 9. Full-scale output voltage and power is determined by the gain setting, G, in register "Headphone Analog Gain (HP_GAIN[2:0])" on page 57. High gain settings at certain VA and VA_HP supply levels may cause clipping when the audio signal approaches full-scale, maximum power output, as shown in Figures 27 - 30 on page 76. 10. See Figure 3. RL and CL reflect the recommended minimum resistance and maximum capacitance required for the internal op-amp's stability and signal integrity. In this circuit topology, CL will effectively move the band-limiting pole of the amp in the output stage. Increasing this value beyond the recommended 150 pF can cause the internal op-amp to become unstable. 11. VA_HP settings lower than VA reduces the headroom of the headphone amplifier. As a result, the DAC may not achieve the full THD+N performance at full-scale output voltage and power. AOUTx 51 0.022 F C L R L AGND Figure 3. Headphone Output Test Load DS679F1 19 CS42L51 COMBINED DAC INTERPOLATION & ON-CHIP ANALOG FILTER RESPONSE Parameter (Note 12) Frequency Response 10 Hz to 20 kHz Passband to -0.05 dB corner to -3 dB corner StopBand StopBand Attenuation (Note 13) Group Delay De-emphasis Error Fs = 32 kHz Fs = 44.1 kHz Fs = 48 kHz Min Typ Max Unit -0.01 - +0.08 dB 0 0 - 0.4780 0.4996 Fs Fs 0.5465 - - Fs 50 - - dB - 10.4/Fs - s - - +1.5/+0 +0.05/-0.25 -0.2/-0.4 dB dB dB Notes: 12. Response is clock dependent and will scale with Fs. Note that the response plots (Figure 38 to Figure 41 on page 82) have been normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs. 13. Measurement Bandwidth is from Stopband to 3 Fs. SWITCHING SPECIFICATIONS - SERIAL PORT (Inputs: Logic 0 = DGND, Logic 1 = VL, SDOUT CLOAD = 15 pF.) Parameters RESET pin Low Pulse Width Symbol (Note 14) MCLK Frequency MCLK Duty Cycle (Note 15) Min Max Units 1 - ms 1.024 38.4 MHz 45 55 % 4 8 4 50 12.5 25 50 100 kHz kHz kHz kHz 45 55 % - 64*Fs Hz 45 55 % Slave Mode Input Sample Rate (LRCK) Quarter-Speed Mode Half-Speed Mode Single-Speed Mode Double-Speed Mode Fs Fs Fs Fs LRCK Duty Cycle SCLK Frequency 1/tP SCLK Duty Cycle LRCK Setup Time Before SCLK Rising Edge ts(LK-SK) 40 - ns LRCK Edge to SDOUT MSB Output Delay td(MSB) - 52 ns SDOUT Setup Time Before SCLK Rising Edge ts(SDO-SK) 20 - ns SDOUT Hold Time After SCLK Rising Edge th(SK-SDO) 30 - ns SDIN Setup Time Before SCLK Rising Edge ts(SD-SK) 20 - ns th 20 - ns SDIN Hold Time After SCLK Rising Edge 20 DS679F1 CS42L51 Parameters Symbol Min Fs - Max Units Master Mode (Note 17) Output Sample Rate (LRCK) All Speed Modes (Note 17) MCLK ----------------128 Hz 45 55 % - 64*Fs Hz 45 55 % td(MSB) - 52 ns SDOUT Setup Time Before SCLK Rising Edge ts(SDO-SK) 20 - ns SDOUT Hold Time After SCLK Rising Edge th(SK-SDO) 30 - ns SDIN Setup Time Before SCLK Rising Edge ts(SD-SK) 20 - ns th 20 - ns LRCK Duty Cycle 1/tP SCLK Frequency SCLK Duty Cycle LRCK Edge to SDOUT MSB Output Delay SDIN Hold Time After SCLK Rising Edge 14. After powering up the CS42L51, RESET should be held low after the power supplies and clocks are settled. 15. See "Example System Clock Frequencies" on page 79 for typical MCLK frequencies. 16. See 17. "Master" on page 39 18. "MCLK" refers to the external master clock applied. // LRCK ts(LK-SK) // tP // SCLK // td(MSB) SDOUT ts(SD-SK) SDIN th(SK-SDO) // MSB // th // MSB // ts(SDO-SK) MSB-1 MSB-1 Figure 4. Serial Audio Interface Slave Mode Timing // LRCK // tP // SCLK // td(MSB) SDOUT ts(SD-SK) SDIN th(SK-SDO) // MSB // th // MSB // ts(SDO-SK) MSB-1 MSB-1 Figure 5. Serial Audio Interface Master Mode Timing DS679F1 21 CS42L51 SWITCHING SPECIFICATIONS - IC(R) CONTROL PORT (Inputs: Logic 0 = DGND, Logic 1 = VL, SDA CL = 30 pF) Parameter Symbol Min Max Unit SCL Clock Frequency fscl - 100 kHz RESET Rising Edge to Start tirs 500 - ns Bus Free Time Between Transmissions tbuf 4.7 - s Start Condition Hold Time (prior to first clock pulse) thdst 4.0 - s Clock Low time tlow 4.7 - s Clock High Time thigh 4.0 - s Setup Time for Repeated Start Condition tsust 4.7 - s thdd 0 - s tsud 250 - ns Rise Time of SCL and SDA trc - 1 s Fall Time SCL and SDA tfc - 300 ns Setup Time for Stop Condition tsusp 4.7 - s Acknowledge Delay from SCL Falling tack 300 3450 ns SDA Hold Time from SCL Falling (Note 19) SDA Setup time to SCL Rising 19. Data must be held for sufficient time to bridge the transition time, tfc, of SCL. RST t irs Stop Repeated Start Start Stop SDA t buf t t high t hdst tf hdst t susp SCL t low t hdd t sud t sust tr Figure 6. Control Port Timing - IC 22 DS679F1 CS42L51 SWITCHING CHARACTERISTICS - SPITM CONTROL PORT (Inputs: Logic 0 = DGND, Logic 1 = VL) Parameter Symbol Min Max Units CCLK Clock Frequency fsck 0 6.0 MHz RESET Rising Edge to CS Falling tsrs 20 - ns CS Falling to CCLK Edge tcss 20 - ns CS High Time Between Transmissions tcsh 1.0 - s CCLK Low Time tscl 66 - ns CCLK High Time tsch 66 - ns CDIN to CCLK Rising Setup Time tdsu 40 - ns CCLK Rising to DATA Hold Time (Note 20) tdh 15 - ns Rise Time of CCLK and CDIN (Note 21) tr2 - 100 ns Fall Time of CCLK and CDIN (Note 21) tf2 - 100 ns 20. Data must be held for sufficient time to bridge the transition time of CCLK. 21. For fsck <1 MHz. RST tsrs CS tcsh tcss tsch tscl tr2 CCLK tf2 tdsu tdh CDIN Figure 7. Control Port Timing - SPI Format DS679F1 23 CS42L51 DC ELECTRICAL CHARACTERISTICS (AGND = 0 V; all voltages with respect to ground.) Parameters Min Typ Max Units Nominal Voltage Output Impedance DC Current Source/Sink (Note 22) - 0.5*VA 23 - 10 V k A DAC_FILT+ Nominal Voltage ADC_FILT+ Nominal Voltage - VA VA - V V - -0.8*(VA_HP) 10 V A 0.8*VA 0.7*VA 0.6*VA 0.5*VA 50 1 - V V V V mA dB 60 - dB VQ Characteristics VSS_HP Characteristics Nominal Voltage DC Current Source MIC BIAS Characteristics DC Current Source Power Supply Rejection Ratio (PSRR) 1 kHz - Power Supply Rejection Ratio (PSRR) (Note 23) 1 kHz - Nominal Voltage MICBIAS_LVL[1:0] = 00 MICBIAS_LVL[1:0] = 01 MICBIAS_LVL[1:0] = 10 MICBIAS_LVL[1:0] = 11 22. The DC current draw represents the allowed current draw from the VQ pin due to typical leakage through electrolytic de-coupling capacitors. 23. Valid with the recommended capacitor values on DAC_FILT+, ADC_FILT+ and VQ. Increasing the capacitance will also increase the PSRR. DIGITAL INTERFACE SPECIFICATIONS & CHARACTERISTICS Parameters (Note 24) Input Leakage Current Symbol Min Max Units Iin - 10 A - 10 pF Input Capacitance 1.8 V - 3.3 V Logic High-Level Output Voltage (IOH = -100 A) VOH VL - 0.2 - V Low-Level Output Voltage (IOL = 100 A) VOL - 0.2 V High-Level Input Voltage VIH 0.68*VL - V Low-Level Input Voltage VIL - 0.32*VL V 24. See "Digital I/O Pin Characteristics" on page 9 for serial and control port power rails. 24 DS679F1 CS42L51 POWER CONSUMPTION See (Note 25) PDN_DACB PDN_DACA PDN_PGAB PDN_PGAA PDN_ADCB PDN_ADCA PDN PDN_MICB PDN_MICA PDN_MICBIAS Power Ctl. Registers 02h 03h Operation Typical Current (mA) iVA_HP iVA iVD iVL (Note 28) V Total Power (mWrms) 1 x x x x x x x x x x 1.8 2.5 0 0 0 0 0 Off (Note 26) 0 0 0 0 0 2 Standby (Note 27) x x x x x x 1 x x x 1.8 2.5 0 0.01 0.02 0 0.05 0 0.01 0.03 0 0.10 3 Mono Record ADC 1 1 1 1 1 0 0 1 1 1 1.8 2.5 PGA to ADC 1 1 1 0 1 0 0 1 1 1 1.8 0 1.85 2.03 0.03 7.05 0 2.07 3.05 0.05 12.94 0 2.35 2.03 0.03 7.95 2.5 0 2.58 3.08 0.05 14.29 MIC to PGA to ADC 1 1 1 0 1 0 0 1 0 0 1.8 (with Bias) 2.5 MIC to PGA to ADC 1 1 1 0 1 0 0 1 0 1 1.8 (no Bias) 2.5 0 3.67 2.05 0.03 10.36 0 3.95 3.09 0.05 17.71 0 3.27 2.03 0.03 9.61 0 3.52 3.08 0.05 16.62 ADC 1 1 1 1 0 0 0 1 1 1 1.8 2.5 PGA to ADC 1 1 0 0 0 0 0 1 1 1 1.8 0 2.69 2.12 0.03 8.72 0 2.93 3.18 0.04 15.40 0 3.65 2.12 0.03 10.45 2.5 0 3.91 3.17 0.04 17.84 MIC to PGA to ADC 1 1 0 0 0 0 0 0 0 1 1.8 (no Bias) 2.5 0 5.48 2.11 0.03 13.73 0 5.76 3.17 0.04 22.45 1 0 1 1 1 1 0 1 1 1 1.8 2.5 1.66 1.40 2.35 0.01 9.74 2.03 1.71 3.48 0.02 18.08 4 5 Stereo Record Mono Playback 6 Stereo Playback 0 0 1 1 1 1 0 1 1 1 1.8 2.5 2.77 2.05 2.35 0.01 12.93 3.21 2.50 3.49 0.02 23.02 7 Mono Record & Playback PGA in (no MIC) to Mono Out 1 0 1 0 1 0 0 1 1 1 1.8 2.5 1.66 3.63 2.73 0.03 14.49 2.03 4.16 4.08 0.05 25.79 8 Phone Monitor MIC (w/bias) in to Mono Out 1 0 1 0 1 0 0 1 0 0 1.8 2.5 1.66 4.95 2.75 0.03 16.90 2.03 5.52 4.08 0.05 29.20 9 Stereo Record & Playback PGA in (no MIC) to Stereo Out 0 0 0 0 0 0 0 1 1 1 1.8 2.5 2.77 5.59 2.82 0.03 20.18 3.21 6.28 4.19 0.04 34.30 25. Unless otherwise noted, test conditions are as follows: All zeros input, slave mode, sample rate = 48 kHz; No load. Digital (VD) and logic (VL) supply current will vary depending on speed mode and master/slave operation. 26. RESET pin 25 held LO, all clocks and data lines are held LO. 27. RESET pin 25 held HI, all clocks and data lines are held HI. 28. VL current will slightly increase in master mode. DS679F1 25 CS42L51 4. APPLICATIONS 4.1 4.1.1 Overview Architecture The CS42L51 is a highly integrated, low power, 24-bit audio CODEC comprised of stereo analog-to-digital converters (ADC), and stereo digital-to-analog converters (DAC) designed using multi-bit delta-sigma techniques. The DAC operates at an oversampling ratio of 128Fs and the ADC operates at 64Fs, where Fs is equal to the system sample rate. The different clock rates maximize power savings while maintaining high performance. The CODEC operates in one of four sample rate speed modes: Quarter, Half, Single and Double. It accepts and is capable of generating serial port clocks (SCLK, LRCK) derived from an input Master Clock (MCLK). 4.1.2 Line & MIC Inputs The analog input portion of the CODEC allows selection from and configuration of multiple combinations of stereo and microphone (MIC) sources. Six line inputs with configuration for two MIC inputs (or one MIC input with common mode rejection), two MIC bias outputs and independent channel control (including a high-pass filter disable function) are available. A Programmable Gain Amplifier (PGA), MIC boost, and Automatic Level Control (ALC), with noise gate settings, provide analog gain and adjustment. Digital volume controls, including gain, boost, attenuation and inversion are also available. 4.1.3 Line & Headphone Outputs The analog output portion of the D/A includes a headphone amplifier capable of driving headphone and line-level loads. An on-chip charge pump creates a negative headphone supply allowing a full-scale output swing centered around ground. This eliminates the need for large DC-Blocking capacitors and allows the amplifier to deliver more power to headphone loads at lower supply voltages. Eight gain settings for the headphone amplifier are available. 4.1.4 Signal Processing Engine A signal processing engine is available to process serial input D/A data before output to the DAC. The D/A data has independent volume controls and mixing functions such as mono mixes and left/right channel swaps. A Tone Control provides bass and treble at four selectable corner frequencies. An automatic level control provides limiting capabilities at programmable attack and release rates, maximum thresholds and soft ramping. A 15/50 s de-emphasis filter is also available at a 44.1 kHz sample rate. 4.1.5 Beep Generator A beep may be generated internally at select frequencies across approximately two octave major scales and configured to occur continuously, periodically or at single time intervals controlled by the user. Volume may be controlled independently. 4.1.6 Device Control (Hardware or Software Mode) In Software Mode, all functions and features may be controlled via a two-wire IC or three-wire SPI control port interface. In Hardware Mode, a limited feature set may be controlled via stand-alone control pins. 4.1.7 Power Management Two Software Mode control registers provide independent power-down control of the ADC, DAC, PGA, MIC pre-amp and MIC bias, allowing operation in select applications with minimal power consumption. 26 DS679F1 CS42L51 4.2 Hardware Mode A limited feature-set is available when the CODEC powers up in Hardware Mode (see "Recommended Power-Up Sequence" on page 41) and may be controlled via stand-alone control pins. Table 2 shows a list of functions/features, the default configuration and the associated stand-alone control available. Hardware Mode Feature/Function Summary Feature/Function Default Configuration Stand-Alone Control Note Powered Up Powered Up Powered Up Powered Up Powered Down Powered Down - - Enabled - - Auto-Detect Speed Mode Single-Speed Mode - - MCLK Divide (Selectable) "MCLKDIV2" pin 2 Serial Port Master / Slave Selection (Selectable) "M/S" pin 29 (Selectable) "IS/LJ" pin 3 Disabled Disabled Disabled Disabled 0 dB 0 dB Disabled Disabled - - - - - - - - Power Control CODEC PGAx ADCx DACx MIC Bias MICx Pre-amplifier Auto-Detect Speed Mode Serial Port Slave Serial Port Master Interface Control ADC DAC ADC Volume & Gain Digital Boost Soft Ramp Zero Cross Invert PGAx Attenuator ALC Noise Gate ADCx High-Pass Filter ADCx High-Pass Filter Freeze Line/MIC Input Select DAC Volume & Gain HP Gain AOUTx Volume Invert Soft Ramp Zero Cross DAC De-Emphasis Signal Processing Engine (SPE) Mix Beep Tone Control Peak Detect and Limiter Data Selection Channel Mix Charge Pump Frequency ADC DAC Enabled Continuous DC Subtraction AIN1A to PGAA AIN1B to PGAB G = 0.6047 0 dB Disabled Enabled Disabled see Section 4.5 on page 38 see Section 4.5 on page 38 see Section 4.6 on page 40 see Section (Selectable) "DEM" pin 4 Disabled Disabled Disabled Disabled - - Data Input (PCM) to DAC - - ADCA = L; ADCB = R PCMA = L; PCMB = R - - (64xFs)/7 - - on page 34 Table 2. Hardware Mode Feature Summary DS679F1 27 CS42L51 4.3 Analog Inputs AINxA and AINxB are the analog inputs, internally biased to VQ, that accepts line-level and MIC-level signals, allowing various gain and signal adjustments for each channel. ADCA_MUTE ADCA_DBOOST +20dB Digital Boost MUX MUX ADCA_ATT[7:0] 0/-96dB 1dB steps PDN_ADCA Multibit Oversampling ADC Attenuator ADCA_HPF FREEZE ADCA_HPF ENABLE INV_ADCA SOFTA PCM Serial Interface MICMIX DIGMIX Noise Gate NG_ALL NG_EN THRESH[3:0] NGDELAY[1:0] MICA_BOOST MICBIAS PDN_MICBIAS ALC_ENB ALCB_SRDIS ALCB_ZCDIS MICBIAS_SEL PDN_ADCB SOFTB +20dB Digital Boost ADCB_HPF FREEZE ADCB_HPF ENABLE AIN3A/ MICIN1 +16/ 32 dB PDN_PGAA PDN_MICA ALC MUX AIN1A AIN2A MUX MICBIAS_LVL[1:0] ADCB_DBOOST MUX PGA AINA_MUX[1:0] ALCA_SRDIS ALCA_ZCDIS ALC_ENA ALC_ARATE[5:0] ALC_RRATE[5:0] MAX[2:0] MIN[2:0] PGAA_VOL[5:0] ADC_SNGVOL SOFTA ZCROSSA +12/-3dB 0.5dB steps Attenuator Multibit Oversampling ADC PGAB_VOL[5:0] ADC_SNGVOL SOFTB ZCROSSB +12/-3dB 0.5dB steps PGA PDN_PGAB ADCB_MUTE AIN1B AIN2B/MICBIAS MUX AIN3B/ MICIN2/ MICBIAS +16/ 32 dB INV_ADCB ADCB_ATT[7:0] 0/-96dB 1dB steps AINB_MUX[1:0] MICB_BOOST PDN_MICB TO SIGNAL PROCESSING ENGINE (SPE) FROM SIGNAL PROCESSING ENGINE (SPE) Figure 8. Analog Input Architecture 4.3.1 Digital Code, Offset & DC Measurement The ADC output data is in two's complement binary format. For inputs above positive full scale or below negative full scale, the ADC will output 7FFFFFH or 800000H, respectively and cause the ADC overflow bit to be set to a `1'. Given the two's complement format, low-level signals may cause the MSB of the serial data to periodically toggle between `1' and `0', possibly introducing noise into the system as the bit switches back and forth. To prevent this phenomena, a constant DC offset is added to the serial data bringing the low-level signal just above the point at which the MSB would normally toggle, thus reducing the noise introduced. Note that this offset is not removed (refer to "Analog Input Characteristics (Commercial - CNZ)" on page 13 and/or "Analog Input Characteristics (Automotive - DNZ)" on page 14 for the specified offset level). The CODEC may be used to measure DC voltages by disabling the high-pass filter for the designated channel. DC levels are measured relative to VQ and will be decoded as positive two's complement binary numbers above VQ and negative two's complement binary numbers below VQ. Software Controls: 28 "Status (Address 20h) (Read Only)" on page 73, "ADC Control (Address 06h)" on page 54. DS679F1 CS42L51 4.3.2 High-Pass Filter and DC Offset Calibration The high-pass filter continuously subtracts a measure of the DC offset from the output of the decimation filter. If the high-pass filter is "frozen" during normal operation, the current value of the DC offset for the corresponding channel is held. It is this DC offset that will continue to be subtracted from the conversion result. This feature makes it possible to perform a system DC offset calibration by: 1. Running the CODECwith the high-pass filter enabled and the DC offset not "frozen" until the filter settles. See the Digital Filter Characteristics for filter settling time. 2. Freezing the DC offset. The high-pass filters are controlled using the ADCx_HPFRZ and ADCx_HPFEN bits. If a particular ADC channel is used to measure DC voltages, the high-pass filter may be disabled using the ADCx_HPFEN bit. Software Controls: 4.3.3 "ADC Control (Address 06h)" on page 54. Digital Routing The digital output of the ADC may be internally routed to the signal processing engine (SPE) for playback of analog input signals. Volume to the DAC may be controlled using the ADCMIX[6:0] bits. The serial input data may also be routed to the ADC serial interface using the DIGMIX bit. This is useful for recording a digital mix along with the analog input. Software Controls: 4.3.4 "ADCx Mixer Volume Control: ADCA (Address 0Eh) & ADCB (Address 0Fh)" on page 61, "Interface Control (Address 04h)" on page 52. Differential Inputs The stereo pair inputs act as a single differential input when the MICMIX bit is enabled. This provides common mode rejection of noise in digitally intense PCB's where the microphone signal traverses long traces, or across long microphone cables as illustrated in Figure 9. Since the mixer provides a differential combination of the two signals, the potential input mix may exceed the maximum full-scale input and result in clipping. The level out of the mixer, therefore, is automatically attenuated 6 dB. Gain may be applied using either the analog PGA or MIC Pre-amp or the digital ADCMIX volume control to re-adjust a small signal to desired levels. The analog inputs may also be used as a differential input pair as illustrated in Figure 10. The two channels are differentially combined when the MICMIX bit is enabled. 4.3.4.1 External Passive Components The microphone input is internally biased to VQ. Input signals must be AC coupled using external capacitors with values consistent with the desired high-pass filter design. The MICINx input resistance of 50 k may be combined with an external capacitor of 1 F to achieve the cutoff frequency defined by the equation, An electrolytic capacitor must be placed such that the positive terminal is positioned relative to the side with the greater bias voltage. The MICBIAS voltage level is controlled by the MICBIAS_LVL[1:0] bits. DS679F1 29 CS42L51 1 fc = ----------------------------------------------- = 3.18 Hz 2 ( 50 k ) ( 1 F ) The MICBIAS series resistor must be selected based on the requirements of the particular microphone used. The MICBIAS output pin is selected using the MICBIAS_SEL bit. Software Controls: "Interface Control (Address 04h)" on page 52, "MIC Control (Address 05h)" on page 53. MICBIAS 20 MICIN1 + // 17 + // MICIN2 18 Figure 9. MIC Input Mix w/Common Mode Rejection 2.5 V 2.15 V VA AINxA 1.25 V 0.35 V 2.15 V AINxB 1.25 V 0.35 V Full-Scale Differential Input Level (MICMIX=1) = (AINxA - AINxB) = 3.6 VPP = 1.27 VRMS Figure 10. Differential Input 30 DS679F1 CS42L51 4.3.5 Analog Input Multiplexer A stereo 4-to-1 analog input multiplexer selects between a line-level input source, or a mic-level input source, depending on the PDN_PGAx and AINx_MUX[1:0] bit settings. Signals may be routed to or bypassed around the PGA. To conserve power, the PGA's may be powered down allowing the user to select from multiple line-level sources and route the stereo signal directly to the ADC. When using the MIC preamp, however, the PGA must be powered up. Analog input channel B may also be used as an output for the MIC bias voltage. The MICBIAS_SEL bit routes the bias voltage to either of two pins. The multiplexer must then select from the remainder of the two input channels. The ADC, PGA and MIC pre-amplifier each has an associated input resistance. When selecting between these paths, the input resistance to the CODEC will change accordingly. Refer to the input resistance characteristics in the Characteristic and Specification Tables for the input resistance of each path. Software Controls: 4.3.6 "Power Control 1 (Address 02h)" on page 49, "MIC Control (Address 05h)" on page 53 "ADCx Input Select, Invert & Mute (Address 07h)" on page 56. MIC & PGA Gain The MIC-level input passes through a +16 dB or +32 dB analog gain stage prior to the input multiplexer, allowing it to be used for microphone level signals without the need for any external gain. The PGA must be powered up when using the MIC pre-amp. The PGA stage provides an additional +12 dB to -3 dB of analog gain in 0.5 dB steps. Software Controls: DS679F1 "Power Control 1 (Address 02h)" on page 49, "ADCx Input Select, Invert & Mute (Address 07h)" on page 56, "ALCX & PGAX Control: ALCA, PGAA (Address 0Ah) & ALCB, PGAB (Address 0Bh)" on page 59, "MIC Control (Address 05h)" on page 53. 31 CS42L51 4.3.7 Automatic Level Control (ALC) When enabled, the ALC monitors the analog input signal after the digital attenuator, detects when peak levels exceed the maximum threshold settings and lowers, first, the PGA gain settings and then increases the digital attenuation levels at a programmable attack rate and maintains the resulting level below the maximum threshold. When input signal levels fall below the minimum threshold, digital attenuation levels are decreased first and the PGA gain is then increased at a programmable release rate and maintains the resulting level above the minimum threshold. Attack and release rates are affected by the ADC soft ramp/zero cross settings and sample rate, Fs. ALC soft ramp and zero cross dependency may be independently enabled/disabled. Recommended settings: Best level control may be realized with the fastest attack and slowest release setting with soft ramp enabled in the control registers. Note: 1.) The maximum realized gain must be set in the PGAx_VOL register. The ALC will only apply the gain set in the PGAx_VOL. 2.) The ALC maintains the output signal between the MIN and MAX thresholds. As the input signal level changes, the level-controlled output may not always be the same but will always fall within the thresholds. "ALC Enable & Attack Rate (Address 1Ch)" on page 70, "ALC Release Rate (Address 1Dh)" on page 71, "ALC Threshold (Address 1Eh)" on page 71, "ALCX & PGAX Control: ALCA, PGAA (Address 0Ah) & ALCB, PGAB (Address 0Bh)" on page 59. Software Controls: Input MAX[2:0] MIN[2:0] below full scale below full scale ALC ADCx_ATT[7:0] and PGAx_VOL[4:0] volume controls should NOT be adjusted manually when ALCx is enabled. PGA Gain and/or Attenuator Output (after ALC) MAX[2:0] MIN[2:0] below full scale below full scale RRATE[5:0] ARATE[5:0] Figure 11. ALC 32 DS679F1 CS42L51 4.3.8 Noise Gate The noise gate may be used to mute signal levels that fall below a programmable threshold. This prevents the ALC from applying gain to noise. A programmable delay may be used to set the minimum time before the noise gate attacks the signal. Maximum noise gate attenuation levels will depend on the gain applied in either the PGA or MIC pre-amplifier. For example: If both +32 dB pre-amplification and +12 dB programmable gain is applied, the maximum attenuation that the noise gate achieves will be 52 dB (-96 + 32 + 12) below full-scale. Ramp-down time to the maximum setting is affected by the SOFTx bit. Recommended settings: For best results, enable soft ramp for the digital attenuator. When the analog inputs are configured for differential signals (see "Differential Inputs" on page 29), enable the NG_ALL bit to trigger the noise gate only when both inputs fall below the threshold. Software Controls: "Noise Gate Configuration & Misc. (Address 1Fh)" on page 72, "ADC Control (Address 06h)" on page 54. Output (dB) N =1 EN G N G =0 EN -96 Maximum Attenuation* -52 dB -64 dB -80 dB -40 Input (dB) THRESH[2:0] Figure 12. Noise Gate Attenuation DS679F1 33 CS42L51 4.4 Analog Outputs AOUTA and AOUTB are the ground-centered line or headphone outputs. Various signal processing options are available, including digital mixes with the ADC signal and an internal Beep Generator. The desired path to the DAC must be selected using the DATA_SEL[1:0] bits. Software Controls: "DAC Control (Address 09h)" on page 58. ARATE[7:0] RRATE[7:0] MAX[2:0] MIN[2:0] LIM_SRDIS LIM_ZCDIS LIMIT_EN SIGNAL PROCESSING ENGINE (SPE) OUTA_VOL[7:0] OUTB_VOL[7:0] +12dB/-102dB 0.5dB steps PCM Serial Interface Chnl Vol. Settings DEEMPH MUTE_PCMMIXA MUTE_PCMMIXB PCMMIXA_VOL[6:0] PCMMIXB_VOL[6:0] PCMA[1:0] PCMB[1:0] +12dB/-51.5dB 0.5dB steps Demph Limiter VOL TC_EN Channel Swap VOL DAC_SZC[1:0] DACA_MUTE DACB_MUTE INV_DACA INV_DACB DAC_SNGVOL AMUTE BPVOL[4:0] 0dB/-50dB 2.0dB steps Peak Detect Bass/ Treble/ Control DATA_SEL[1:0] 01 PDN_DACA PDN_DACB Switched Capacitor DAC and Filter HP_GAIN[2:0] Headphone Amp - GND Centered Left/Right HP Out 00 BASS_CF[1:0] TREB_CF[1:0] Charge Pump BASS[3:0] TREB[3:0] +12.0dB/-10.5dB 1.5dB steps CHRG_FREQ[3:0] VOL OFFTIME[2:0] ONTIME[3:0] FREQ[3:0] REPEAT BEEP Beep Generator Figure 13. Output Architecture 4.4.1 De-Emphasis Filter The CODEC includes on-chip digital de-emphasis optimized for a sample rate of 44.1 kHz. The filter response is shown in Figure 14. The de-emphasis feature is included to accommodate audio recordings that utilize 50/15 s pre-emphasis equalization as a means of noise reduction. De-emphasis is only available in Single-Speed Mode. Software Controls: "DAC Control (Address 09h)" on page 58. Pin Hardware Control: 34 Setting "DEM" pin 4. LO HI Selection No De-Emphasis De-Emphasis Applied DS679F1 CS42L51 Gain dB T1=50 s 0dB T2 = 15 s -10dB F1 3.183 kHz F2 Frequency 10.61 kHz Figure 14. De-Emphasis Curve 4.4.2 Volume Controls Three digital volume control functions are implemented, offering independent control over the ADC and PCM signal paths into the mixer as well as a combined control over the mixed signals. All volume controls are programmable to ramp in increments of 0.125 dB at a rate controlled by the DAC soft ramp/zero cross settings. All signal paths may also be independently muted via mute control bits. When enabled, each bit attenuates the signal to its maximum value. When the mute bit is disabled, the signal returns to the attenuation level set in the respective volume control register. The attenuation is ramped up and down at the rate specified by the DAC_SZC[1:0] bits. Software Controls: 4.4.3 "ADCx Mixer Volume Control: ADCA (Address 0Eh) & ADCB (Address 0Fh)" on page 61, "PCMX Mixer Volume Control: PCMA (Address 10h) & PCMB (Address 11h)" on page 62, "AOUTx Volume Control: AOUTA (Address 16h) & AOUTB (Address 17h)" on page 66, "DAC Output Control (Address 08h)" on page 57. Mono Channel Mixer A channel mixer may be used to create a mix of the left and right channels for either the PCM or ADC signals. This mix allows the user to produce a MONO signal from a stereo source. The mixer may also be used to implement a left/right channel swap. Software Controls: 4.4.4 "PCM Channel Mixer (Address 18h)" on page 67. Beep Generator The Beep Generator generates audio frequencies across approximately two octave major scales. It offers three modes of operation: Continuous, multiple and single (one-shot) beeps. Sixteen on and eight off times are available. Note: The Beep is generated before the limiter and may affect desired limiting performance. If the limiter function is used, it may be required to set the Beep volume sufficiently below the threshold to prevent the peak detect from triggering. Since the master volume control, AOUTx_VOL[7:0], will affect the Beep volume, DAC volume may alternatively be controlled using the PCMMIXx_VOL[6:0] bits. Software Controls: DS679F1 "Beep Frequency & Timing Configuration (Address 12h)" on page 62, "Beep Off Time & Volume (Address 13h)" on page 63, "Beep Configuration & Tone Configuration (Address 14h)" on page 64 35 CS42L51 REPEAT = '1' BEEP = '1' CONTINUOUS BEEP: Beep turns on at a configurable frequency (FREQ) and volume (BPVOL) and remains on until REPEAT is cleared. REPEAT = '1' BEEP = '0' MULTI-BEEP: Beep turns on at a configurable frequency (FREQ) and volume (BPVOL) for the duration of ONTIME and turns off for the duration of OFFTIME. On and off cycles are repeated until REPEAT is cleared. REPEAT = '0' BEEP = '1' SINGLE-BEEP: Beep turns on at a configurable frequency (FREQ) and volume (BPVOL) for the duration of ONTIME. BEEP must be cleared and set for additional beeps. ... BPVOL[4:0] FREQ[3:0] ONTIME[3:0] OFFTIME[2:0] Figure 15. Beep Configuration Options 4.4.5 Tone Control Shelving filters are used to implement bass and treble (boost and cut) with four selectable corner frequencies. Boosting will affect peak detect and limiting when levels exceed the maximum threshold settings. Software Controls: 4.4.6 "Tone Control (Address 15h)" on page 65. Limiter When enabled, the limiter monitors the digital input signal before the DAC modulator, detects when levels exceed the maximum threshold settings and lowers the AOUT volume at a programmable attack rate below the maximum threshold. When the input signal level falls below the maximum threshold, the AOUT volume returns to its original level set in the Volume Control register at a programmable release rate. Attack and release rates are affected by the DAC soft ramp/zero cross settings and sample rate, Fs. Limiter soft ramp and zero cross dependency may be independently enabled/disabled. Recommended settings: Best limiting performance may be realized with the fastest attack and slowest release setting with soft ramp enabled in the control registers. The "cushion" bits allow the user to set a threshold slightly below the maximum threshold for hysteresis control - this cushions the sound as the limiter attacks and releases. Note: 1. When the Limiter is enabled, the AOUT Volume is automatically controlled and should not be adjusted manually. Alternative volume control may be realized using the PCMMIXx_VOL[6:0] bits. 2. The Limiter maintains the output signal between the CUSH and MAX thresholds. As the digital input signal level changes, the level-controlled output may not always be the same but will always fall within the thresholds. Software Controls: 36 "Limiter Release Rate Register (Address 1Ah)" on page 69, "Limiter Attack Rate Register (Address 1Bh)" on page 70, "DAC Control (Address 09h)" on page 58 DS679F1 CS42L51 Input MAX[2:0] Limiter AOUTx_VOL[7:0] volume control should NOT be adjusted manually when Limiter is enabled. ATTACK/RELEASE SOUND CUSHION Volume Output (after Limiter) CUSH[2:0] MAX[2:0] ARATE[5:0] RRATE[5:0] Figure 16. Peak Detect & Limiter 4.4.7 Line-Level Outputs and Filtering The CODEC contains on-chip buffer amplifiers capable of producing line level single-ended outputs on AOUTA and AOUTB. These amplifiers are ground centered and do not have any DC offset. A load stabilizer circuit, shown in the "Typical Connection Diagram (Software Mode)" on page 10 and the "Typical Connection Diagram (Hardware Mode)" on page 11, is required on the analog outputs. This allows the DAC amplifiers to drive line or headphone outputs. Also shown in the Typical Connection diagrams is the recommended passive output filter to support higher impedances such as those found on the inputs to operational amplifiers. "Rext", shown in the typical connection diagrams, is the input impedance of the receiving device. The invert and digital gain controls may be used to provide phase and/or amplitude compensation for an external filter. The delta-sigma conversion process produces high frequency noise beyond the audio passband, most of which is removed by the on-chip analog filters. The remaining out-of-band noise can be attenuated using an off-chip low pass filter. Software Controls: DS679F1 "DAC Output Control (Address 08h)" on page 57, "AOUTx Volume Control: AOUTA (Address 16h) & AOUTB (Address 17h)" on page 66. 37 CS42L51 4.4.8 On-Chip Charge Pump An on-chip charge pump derives a negative supply voltage from the VA_HP supply. This provides dual rail supplies allowing a full-scale output swing centered around ground and eliminates the need for large, DC-blocking capacitors. Added benefits include greater pop suppression and improved low frequency (bass) response. Note: Series resistance in the path of the power supplies must be avoided. Any voltage drop on the VA_HP supply will directly impact the derived negative voltage on the charge pump supply, VSS_HP, and may result in clipping. The FLYN and FLYP pins connect to internal switches that charges and discharges the external capacitor attached, at a default switching frequency. This frequency may be adjusted in the control port registers. Increasing the charge-pumping capacitor will slightly decease the pumping frequency. The capacitor connected to VSS_HP acts as a charge reservoir for the negative supply as well as a filter for the ripple induced by the charge pump. Increasing this capacitor will decrease the ripple on VSS_HP. Refer to the typical connection diagrams in Figure 1 on page 10 or Figure 2 on page 11 for the recommended capacitor values for the charge pump circuitry. Software Controls: 4.5 "Charge Pump Frequency (Address 21h)" on page 74. Serial Port Clocking The CODEC serial audio interface port operates either as a slave or master. It accepts externally generated clocks in slave mode and will generate synchronous clocks derived from an input master clock (MCLK) in master mode. The frequency of the MCLK must be an integer multiple of, and synchronous with, the system sample rate, Fs. The LRCK frequency is equal to Fs, the frequency at which audio samples for each channel are clocked into or out of the device. The SPEED and MCLKDIV2 software control bits or the SDOUT/(M/S) and MCLKDIV2 stand-alone control pins, configure the device to generate the proper clocks in Master Mode and receive the proper clocks in Slave Mode. The value on the SDOUT pin is latched immediately after powering up in Hardware Mode. Software Control: "MIC Power Control & Speed Control (Address 03h)" on page 50, "DAC Control (Address 09h)" on page 58. Pin Hardware Control: "SDOUT, M/S" pin 29 "MCLKDIV2" pin 2 38 Setting Selection 47 k Pull-down Slave 47 k Pull-up Master LO No Divide HI MCLK is divided by 2 prior to all internal circuitry. DS679F1 CS42L51 4.5.1 Slave LRCK and SCLK are inputs in Slave Mode. The speed of the CODEC is automatically determined based on the input MCLK/LRCK ratio when the Auto-Detect function is enabled. Certain input clock ratios will then require an internal divide-by-two of MCLK* using either the MCLKDIV2 bit or the MCLKDIV2 standalone control pin. Additional clock ratios are allowed when the Auto-Detect function is disabled; but the appropriate speed mode must be selected using the SPEED[1:0] bits. Auto-Detect QSM HSM SSM DSM Disabled (Software Mode only) 512, 768, 1024, 1536, 2048, 3072 256, 384, 512, 768, 1024, 1536 128, 192, 256, 384, 512, 768 128, 192, 256, 384 512, 768, 1024*, 1536* 256, 384, 512*, 768* 128, 192, 256*, 384* 1024, 1536, 2048*, 3072* *MCLKDIV2 must be enabled. Enabled Table 3. MCLK/LRCK Ratios 4.5.2 Master LRCK and SCLK are internally derived from the internal MCLK (after the divide, if MCLKDIV2 is enabled). In Hardware Mode the CODEC operates in single-speed only. In Software Mode, the CODEC operates in either quarter-, half-, single- or double-speed depending on the setting of the SPEED[1:0] bits. /1 0 /2 1 / 128 Double Speed 00 / 128 Single Speed 01 / 256 Half Speed 10 / 512 Quarter Speed 11 MCLK LRCK Output (Equal to Fs) SPEED[1:0] MCLKDIV2 /2 Double Speed 00 /2 Single Speed 01 /4 Half Speed 10 /8 Quarter Speed 11 SCLK Output Figure 17. Master Mode Timing DS679F1 39 CS42L51 4.5.3 High-Impedance Digital Output The serial port may be placed on a clock/data bus that allows multiple masters for the serial port I/O without the need for external buffers. The 3ST_SP bit places the internal buffers for these I/O in a high-impedance state, allowing another device to transmit serial port data without bus contention. CS42L51 Transmitting Device #2 Transmitting Device #1 SDOUT 3ST_SP SCLK/LRCK Receiving Device Figure 18. Tri-State Serial Port 4.5.4 Quarter- and Half-Speed Mode Quarter-Speed Mode (QSM) and Half-Speed Mode (HSM) allow lower sample rates while maintaining a relatively flat noise floor in the typical audio band of 20 Hz - 20 kHz. Single-Speed Mode (SSM) will allow lower frequency sample rates; however, the DAC's noise floor, that normally rises out-of-band, will scale with the lower sample rate and begin to rise within the audio band. QSM and HSM corrects for most of this scaling, effectively increasing the dynamic range of the CODEC at lower sample rates, relative to SSM. 4.6 Digital Interface Formats The serial port operates in standard IS, Left-Justified or Right-Justified (DAC only) digital interface formats with varying bit depths from 16 to 24. Data is clocked out of the ADC or into the DAC on the rising edge of SCLK. Figures 19-21 illustrate the general structure of each format. Refer to "Switching Specifications - Serial Port" on page 20 for exact timing relationship between clocks and data. Software Control: "Interface Control (Address 04h)" on page 52. Pin Hardware Control: LRCK "IS/LJ" pin 3 Setting Selection LO Left-Justified Interface HI IS Interface L eft C h a n n e l R ig ht C h a n n el SCLK SDIN MSB LSB MSB AOUTA / AINxA LSB MSB AOUTB / AINxB Figure 19. IS Format 40 DS679F1 CS42L51 LRCK L eft C h a n n e l R ig ht C h a n n e l SCLK SDIN MSB LS B MSB LSB MSB AOUTB / AINxB AOUTA / AINxA Figure 20. Left-Justified Format LRCK L eft C h a n n el R ig ht C h a n n el SCLK MSB SDIN M SB LSB AOUTA LS B AOUTB Figure 21. Right-Justified Format (DAC only) 4.7 Initialization The initialization and Power-Down sequence flowchart is shown in Figure 22 on page 42. The CODEC enters a Power-Down state upon initial power-up. The interpolation and decimation filters, delta-sigma modulators and control port registers are reset. The internal voltage reference, multi-bit DAC and ADC and switched-capacitor low-pass filters are powered down. The device will remain in the Power-Down state until the RESET pin is brought high. The control port is accessible once RESET is high and the desired register settings can be loaded per the interface descriptions in "Software Mode" on page 43. If a valid write sequence to the control port is not made within approximately 10 ms, the CODEC will enter Hardware Mode. Once MCLK is valid, the quiescent voltage, VQ, and the internal voltage references, DAC_FILT+ and ADC_FILT+ will begin powering up to normal operation. The charge pump slowly powers up and charges the capacitors. Power is then applied to the headphone amplifiers and switched-capacitor filters, and the analog/digital outputs enter a muted state. Once LRCK is valid, MCLK occurrences are counted over one LRCK period to determine the MCLK/LRCK frequency ratio and normal operation begins. 4.8 Recommended Power-Up Sequence 1. Hold RESET low until the power supplies are stable. 2. Bring RESET high. After approximately 10 ms, the device will enter Hardware Mode. 3. For Software Mode operation, set the PDN bit to `1'b in under 10 ms. This will place the device in "standby". 4. Load the desired register settings while keeping the PDN bit set to `1'b. 5. Start MCLK to the appropriate frequency, as discussed in Section 4.5. 6. Set the PDN bit to `0'b. 7. Apply LRCK,SCLK and SDIN for normal operation to begin. 8. Bring RESET low if the analog or digital supplies drop below the recommended operating condition to prevent power glitch related issues. DS679F1 41 CS42L51 4.9 Recommended Power-Down Sequence To minimize audible pops when turning off or placing the CODEC in standby, 1. Mute the DAC's and ADC's. 2. Set the PDN bit in the power control register to `1'b. The CODEC will not power down until it reaches a fully muted sate. Do not remove MCLK until after the part has fully muted. Note that it may be necessary to disable the soft ramp and/or zero cross volume transitions to achieve faster muting/power down. 3. Bring RESET low. No Power 1. No audio signal generated. Off Mode (Power Applied) 1. No audio signal generated. 2. Control Port Registers reset to default. PDN bit = '1'b? Standby Mode 1. No audio signal generated. 2. Control Port Registers retain settings. Yes No No RESET = Low? Valid MCLK Applied? Yes No 20 ms delay Control Port Active Charge Caps 1. VQ Charged to quiescent voltage. 2. Filtx+ Charged. No Control Port Valid Write Seq. within 10 ms? Yes ADC Initialization 2048 internal MCLK cycle delay DAC Initialization Digital/Analog Output Muted Charge Pump Powered Up 50 ms delay Headphone Amp Powered Down Hardware Mode Minimal feature set support. Power Off Transition 1. Audible pops. Software Mode Registers setup to desired settings. Sub-Clocks Applied 1. LRCK valid. 2. SCLK valid. 3. Audio samples processed. 20 s delay 20 s delay (DAC only) Headphone Amp Powered Up Stand-By Transition 1. Pops suppressed. No Reset Transition 1. Pops suppressed. Valid MCLK/LRCK Ratio? Yes RESET = Low ERROR: MCLK/LRCK ratio change ERROR: Power removed Normal Operation Audio signal generated per control port or standalone settings. PDN bit set to '1'b (software mode only) ERROR: MCLK removed Analog Output Freeze 1. Aout bias = last audio sample. 2. DAC Modulators stop operation. 3. Audible pops. Figure 22. Initialization Flowchart 42 DS679F1 CS42L51 4.10 Software Mode The control port is used to access the registers allowing the CODEC to be configured for the desired operational modes and formats. The operation of the control port may be completely asynchronous with respect to the audio sample rates. However, to avoid potential interference problems, the control port pins should remain static if no operation is required. The control port operates in two modes: SPI and IC, with the CODEC acting as a slave device. Software Mode is selected if there is a high-to-low transition on the AD0/CS pin after the RESET pin has been brought high. IC Mode is selected by connecting the AD0/CS pin through a resistor to VL or DGND, thereby permanently selecting the desired AD0 bit address state. 4.10.1 SPI Control In Software Mode, CS is the CS42L51 chip-select signal, CCLK is the control port bit clock (input into the CS42L51 from the microcontroller), CDIN is the input data line from the microcontroller. Data is clocked in on the rising edge of CCLK. The CODEC will only support write operations. Read request will be ignored. Figure 23 shows the operation of the control port in Software Mode. To write to a register, bring CS low. The first seven bits on CDIN form the chip address and must be 1001010. The eighth bit is a read/write indicator (R/W), which should be low to write. The next eight bits form the Memory Address Pointer (MAP), which is set to the address of the register that is to be updated. The next eight bits are the data which will be placed into the register designated by the MAP. There is MAP auto-increment capability, enabled by the INCR bit in the MAP register. If INCR is a zero, the MAP will stay constant for successive read or writes. If INCR is set to a 1, the MAP will auto-increment after each byte is read or written, allowing block reads or writes of successive registers. CS 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 CCLK CHIP ADDRESS (WRITE) CDIN 1 0 0 1 0 1 0 MAP BYTE 0 INCR 6 5 4 3 DATA +n DATA 2 1 0 7 6 1 0 7 6 1 0 Figure 23. Control Port Timing in SPI Mode 4.10.2 IC Control In IC Mode, SDA is a bidirectional data line. Data is clocked into and out of the part by the clock, SCL. There is no CS pin. Pin AD0 forms the least significant bit of the chip address and should be connected through a resistor to VL or DGND as desired. The state of the pin is sensed while the CS42L51 is being reset. The signal timings for a read and write cycle are shown in Figure 24 and Figure 25. A Start condition is defined as a falling transition of SDA while the clock is high. A Stop condition is a rising transition while the clock is high. All other transitions of SDA occur while the clock is low. The first byte sent to the CS42L51 after a Start condition consists of a 7-bit chip address field and a R/W bit (high for a read, low for a write). The upper 6 bits of the 7-bit address field are fixed at 100101. To communicate with a CS42L51, the chip address field, which is the first byte sent to the CS42L51, should match 100101 followed by the setting of the AD0 pin. The eighth bit of the address is the R/W bit. If the operation is a write, the next byte is the Memory Address Pointer (MAP) which selects the register to be read or written. If the operation is a read, DS679F1 43 CS42L51 the contents of the register pointed to by the MAP will be output. Setting the auto-increment bit in MAP allows successive reads or writes of consecutive registers. Each byte is separated by an acknowledge bit. The ACK bit is output from the CS42L51 after each input byte is read and is input to the CS42L51 from the microcontroller after each transmitted byte. 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 24 25 26 27 28 19 SCL CHIP ADDRESS (WRITE) 1 SDA 0 0 1 0 1 AD0 MAP BYTE 0 INCR 6 5 4 3 1 0 ACK 7 6 1 ACK DATA +n DATA +1 DATA 2 0 7 6 1 0 7 6 1 ACK 0 ACK STOP START Figure 24. Control Port Timing, IC Write 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 SCL CHIP ADDRESS (WRITE) SDA 1 0 0 1 STOP MAP BYTE 0 1 AD0 0 INCR 6 5 4 3 2 1 0 ACK CHIP ADDRESS (READ) 1 0 0 1 0 ACK START DATA 1 AD0 1 START 7 ACK DATA +1 0 7 ACK 0 DATA + n 7 0 NO ACK STOP Figure 25. Control Port Timing, IC Read Since the read operation cannot set the MAP, an aborted write operation is used as a preamble. As shown in Figure 25, the write operation is aborted after the acknowledge for the MAP byte by sending a stop condition. The following pseudocode illustrates an aborted write operation followed by a read operation. Send start condition. Send 100101x0 (chip address & write operation). Receive acknowledge bit. Send MAP byte, auto-increment off. Receive acknowledge bit. Send stop condition, aborting write. Send start condition. Send 100101x1 (chip address & read operation). Receive acknowledge bit. Receive byte, contents of selected register. Send acknowledge bit. Send stop condition. Setting the auto-increment bit in the MAP allows successive reads or writes of consecutive registers. Each byte is separated by an acknowledge bit. 44 DS679F1 CS42L51 4.10.3 Memory Address Pointer (MAP) The MAP byte comes after the address byte and selects the register to be read or written. Refer to the pseudo code above for implementation details. 4.10.3.1 Map Increment (INCR) The device has MAP auto-increment capability enabled by the INCR bit (the MSB) of the MAP. If INCR is set to 0, MAP will stay constant for successive IC writes or reads and SPI writes. If INCR is set to 1, MAP will auto-increment after each byte is read or written, allowing block reads or writes of successive registers. DS679F1 45 CS42L51 5. REGISTER QUICK REFERENCE Software mode register defaults are as shown. "Reserved" registers must maintain their default state. Addr 01h Function ID p 49 default 02h Power Ctl. 1 p 49 default 03h Speed Ctl. & Power Ctl. 2 p 50 default 04h Interface Ctl. p 52 default 05h MIC Control & Misc. p 53 default 06h ADC Control p 54 default 07h ADC Input Select , Invert, Mute 7 6 5 4 3 2 1 0 Chip_ID4 Chip_ID3 Chip_ID2 Chip_ID1 Chip_ID0 Rev_ID2 Rev_ID1 Rev_ID0 1 1 0 1 1 0 0 1 PDN_ADCB PDN_ADCA PDN Reserved PDN_DACB PDN_DACA dPDN_PGAB PDN_PGAA 0 0 0 0 0 0 0 0 AUTO SPEED1 SPEED0 3-ST_SP PDN_MICB PDN_MICA PDN_ MICBIAS MCLKDIV2 1 0 1 0 1 1 1 0 SDOUT->SDIN M/S DAC_DIF2 DAC_DIF1 DAC_DIF0 ADC_IS/LJ DIGMIX MICMIX 0 0 0 0 0 0 0 0 ADC_SNGVOL ADCB_ DBOOST ADCA_ DBOOST MICBIAS_ SEL MICBIAS_ LVL1 MICBIAS_ LVL0 MICB_ BOOST MICA_ BOOST 0 0 0 0 0 0 0 0 ADCB_HPF EN ADCB_HP FRZ ADCA_HPF EN ADCA_HP FRZ SOFTB ZCROSSB SOFTA ZCROSSA 1 0 1 0 0 0 0 0 INV_ADCB INV_ADCA ADCB_ MUTE ADCA_ MUTE AINB_MUX1 AINB_MUX AINA_MUX1 AINA_MUX0 0 0 0 0 0 0 0 0 0 HP_GAIN2 HP_GAIN1 HP_GAIN0 DAC_SNG VOL INV_PCMB INV_PCMA DACB_ MUTE DACA_ MUTE 0 1 1 0 0 0 0 0 DATA_SEL1 DATA_SEL0 FREEZE Reserved DEEMPH AMUTE DAC_SZC1 DAC_SZC0 0 0 0 0 0 1 1 0 ALCA_SR DIS ALCA_ZC DIS Reserved PGAA VOL4 PGAA VOL3 PGAA VOL2 PGAA VOL1 PGAA VOL0 0 0 0 0 0 0 0 0 ALCB_SR DIS ALCB_ZC DIS Reserved PGAB VOL4 PGAB VOL3 PGAB VOL2 PGAB VOL1 PGAB VOL0 p 59 default 0 0 0 0 0 0 0 0 0Ch ADCA Attenuator ADCA_ ATT7 ADCA_ ATT6 ADCA_ ATT5 ADCA_ ATT4 ADCA_ ATT3 ADCA_ ATT2 ADCA_ ATT1 ADCA_ ATT0 p 60 default 0 0 0 0 0 0 0 0 0Dh ADCB Attenuator ADCB_ ATT7 ADCB_ ATT6 ADCB_ ATT5 ADCB_ ATT4 ADCB_ ATT3 ADCB_ ATT2 ADCB_ ATT1 ADCB_ ATT0 p 56 default 08h DAC Output Control p 57 default 09h DAC Control p 58 default 0Ah ALCA SZC & PGAA Volume p 59 default 0Bh 46 ALCB SZC & PGAB Volume DS679F1 CS42L51 Addr Function p 60 default 0Eh Vol. Control ADCMIXA p 61 default 0Fh Vol. Control ADCMIXB p 61 default 10h Vol. Control PCMMIXA p 62 default 11h Vol. Control PCMMIXB p 62 default 12h BEEP Freq. & OnTime p 62 default 13h BEEP Off Time & Vol. p 63 default 14h BEEP Control & Tone Config p 64 default 15h Tone Control p 65 default 16h Vol. Control AOUTA p 66 default 17h Vol. Control AOUTB p 66 default 18h PCM & ADC Channel Mixer p 67 default 19h Limiter Threshold & SZC Disable p 67 default DS679F1 7 6 5 4 3 2 1 0 0 0 0 0 0 0 0 0 MUTE_ADC MIXA ADCMIXA VOL6 ADCMIXA VOL5 ADCMIXA VOL4 ADCMIXA VOL3 ADCMIXA VOL2 ADCMIXA VOL1 ADCMIXA VOL0 1 0 0 0 0 0 0 0 MUTE_ADC MIXB ADCMIXB VOL6 ADCMIXB VOL5 ADCMIXB VOL4 ADCMIXB VOL3 ADCMIXB VOL2 ADCMIXB VOL1 ADCMIXB VOL0 1 0 0 0 0 0 0 0 MUTE_PCM MIXA PCMMIXA VOL6 PCMMIXA VOL5 PCMMIXA VOL4 PCMMIXA VOL3 PCMMIXA VOL2 PCMMIXA VOL1 PCMMIXA VOL0 1 0 0 0 0 0 0 0 MUTE_PCM MIXB PCMMIXB VOL6 PCMMIXB VOL5 PCMMIXB VOL4 PCMMIXB VOL3 PCMMIXB VOL2 PCMMIXB VOL1 PCMMIXB VOL0 1 0 0 0 0 0 0 0 FREQ3 FREQ2 FREQ1 FREQ0 ONTIME3 ONTIME2 ONTIME1 ONTIME0 0 0 0 0 0 0 0 0 OFFTIME2 OFFTIME1 OFFTIME0 BPVOL4 BPVOL3 BPVOL2 BPVOL1 BPVOL0 0 0 0 0 0 0 0 0 REPEAT BEEP Reserved TREB_CF1 TREB_CF0 BASS_CF1 BASS_CF0 TC_EN 0 0 0 0 0 0 0 0 TREB3 TREB2 TREB1 TREB0 BASS3 BASS2 BASS1 BASS0 1 0 0 0 1 0 0 0 AOUTA_ VOL7 AOUTA_ VOL6 AOUTA_ VOL5 AOUTA_ VOL4 AOUTA_ VOL3 AOUTA_ VOL2 AOUTA_ VOL1 AOUTA_ VOL0 0 0 0 0 0 0 0 0 AOUTB_ VOL7 AOUTB_ VOL6 AOUTB_ VOL5 AOUTB_ VOL4 AOUTB_ VOL3 AOUTB_ VOL2 AOUTB_ VOL1 AOUTB_ VOL0 0 0 0 0 0 0 0 0 PCMA1 PCMA0 PCMB1 PCMB0 ADCA1 ADCA0 ADCB1 ADCB0 0 0 0 0 0 0 0 0 MAX2 MAX1 MAX0 CUSH2 CUSH1 CUSH0 LIM_SRDIS LIM_ZCDIS 0 0 0 0 0 0 0 0 47 CS42L51 Addr Function 7 6 5 4 3 2 1 0 1Ah Limiter Config & Release Rate LIMIT_EN LIMIT_ALL LIM_RRATE 5 LIM_RRATE 4 LIM_RRATE 3 LIM_RRATE 2 LIM_RRATE 1 LIM_RRATE 0 0 1 1 1 1 1 1 1 Reserved Reserved p 70 default 0 0 1Ch ALC Enable & Attack Rate ALC_ENB ALC_ENA p 70 default 0 0 1Dh ALC Release Rate Reserved Reserved p 71 default 0 0 1 1 1 1 1 1 MAX2 MAX1 MAX0 MIN2 MIN1 MIN0 Reserved Reserved 0 0 0 0 0 0 0 0 NG_ALL NG_EN NG_BOOST THRESH2 THRESH1 THRESH0 NGDELAY1 NGDELAY0 p 72 default 0 0 0 0 0 0 0 0 Status Reserved p 73 default 0 0 0 0 0 0 0 0 CHRG_ FREQ3 CHRG_ FREQ2 CHRG_ FREQ1 CHRG_ FREQ0 Reserved Reserved Reserved Reserved 0 1 0 1 0 0 0 0 p 69 default 1Bh 1Eh Limiter Attack Rate ALC Threshold p 71 default 1Fh 20h 21h Noise Gate Config Charge Pump Frequency p 74 default 48 LIM_ARATE5 LIM_ARATE4 LIM_ARATE3 LIM_ARATE2 LIM_ARATE1 LIM_ARATE0 0 0 0 0 0 0 ALC_ARATE AALC_RATE ALC_ARATE ALC_ARATE ALC_ARATE ALC_ARATE 5 4 3 2 1 0 0 0 0 0 0 0 ALC_RRATE ALC_RRATE ALC_RRATE ALC_RRATE ALC_RRATE ALC_RRATE 5 4 3 2 1 0 SP_CLKER SPEB_OVFL SPEA_OVFL PCMA_OVFL PCMB_OVFL ADCA_OVFL ADCB_OVFL R DS679F1 CS42L51 6. REGISTER DESCRIPTION All registers are read/write except for the chip I.D. and Revision Register and Interrupt Status Register which are read only. See the following bit definition tables for bit assignment information. The default state of each bit after a power-up sequence or reset is listed in each bit description. All "Reserved" registers must maintain their default state. Note: Certain functions are only available when the "Signal Processing Engine to DAC" option is selected using the DATA_SEL[1:0] bits, as described in section "DAC Data Selection (DATA_SEL[1:0])" on page 58. 6.1 Chip I.D. and Revision Register (Address 01h) (Read Only) 7 Chip_ID4 6 Chip_ID3 5 Chip_ID2 4 Chip_ID1 3 Chip_ID0 2 Rev_ID2 1 Rev_ID1 0 Rev_ID0 Chip I.D. (Chip_ID[4:0]) Default: 11011 Function: I.D. code for the CS42L51. Permanently set to 11011. Chip Revision (Rev_ID[2:0]) Default: 001 Function: CS42L51 revision level. Revision B is coded as 001. Revision A is coded as 000. 6.2 Power Control 1 (Address 02h) 7 Reserved 6 PDN_DACB 5 PDN_DACA 4 PDN_PGAB 3 PDN_PGAA 2 PDN_ADCB 1 PDN_ADCA 0 PDN Notes: 1. To activate the power-down sequence for individual channels (A or B,) both channels must first be powered down either by enabling the PDN bit or by enabling the power-down bits for both channels. Enabling the power-down bit on an individual channel basis after the CODEC has fully powered up will mute the selected channel without achieving any power savings. Recommended channel power-down sequence: 1.) Enable the PDN bit, 2.) enable power-down for the select channels, 3.) disable the PDN bit. Power Down DAC X (PDN_DACX) Default: 0 0 - Disable 1 - Enable Function: DAC channel x will either enter a power-down or muted state when this bit is enabled. See Note 1 above. DS679F1 49 CS42L51 Power Down PGA X (PDN_PGAX) Default: 0 0 - Disable 1 - Enable Function: PGA channel x will either enter a power-down or muted state when this bit is enabled. See Power Control 1 (Address 02h) on page 49 above. This bit is used in conjunction with AINx_MUX bits to determine the analog input path to the ADC. Refer to "ADCX Input Select Bits (AINX_MUX[1:0])" on page 56 for the required settings. Power Down ADC X (PDN_ADCX) Default: 0 0 - Disable 1 - Enable Function: ADC channel x will either enter a power-down or muted state when this bit is enabled. See Note 1 on page 49. Power Down (PDN) Default: 0 0 - Disable 1 - Enable Function: The entire CODEC will enter a low-power state when this function is enabled. The contents of the control port registers are retained in this mode. 6.3 MIC Power Control & Speed Control (Address 03h) 7 AUTO 6 SPEED1 5 SPEED0 4 3-ST_SP 3 PDN_MICB 2 PDN_MICA 1 PDN_MICBIAS 0 MCLKDIV2 Auto-Detect Speed Mode (AUTO) Default: 1 0 - Disable 1 - Enable Function: Enables the auto-detect circuitry for detecting the speed mode of the CODEC when operating as a slave. When AUTO is enabled, the MCLK/LRCK ratio must be implemented according to Table 3 on page 39. The SPEED[1:0] bits are ignored when this bit is enabled. Speed is determined by the MCLK/LRCK ratio. 50 DS679F1 CS42L51 Speed Mode (SPEED[1:0]) Default: 01 11 - Quarter-Speed Mode (QSM) - 4 to 12.5 kHz sample rates 10 - Half-Speed Mode (HSM) - 12.5 to 25 kHz sample rates 01 - Single-Speed Mode (SSM) - 4 to 50 kHz sample rates 00 - Double-Speed Mode (DSM) - 50 to 100 kHz sample rates Function: Sets the appropriate speed mode for the CODEC in Master or Slave Mode. QSM is optimized for 8 kHz sample rate and HSM is optimized for 16 kHz sample rate. These bits are ignored when the AUTO bit is enabled (see Auto-Detect Speed Mode (AUTO) above). Tri-State Serial Port Interface (3ST_SP) Default: 0 0 - Disable 1 - Enable Function: When enabled and the device is configured as a master, all serial port outputs are place in a high impedance state. If the serial port is configured as a slave, only the SDOUT pin will be placed in a high-impedance state. The other signals will remain as inputs. Power Down MIC X (PDN_MICX) Default: 1 0 - Disable 1 - Enable Function: When enabled, the microphone pre-amplifier for channel x will be in a power-down state. Power Down MIC BIAS (PDN_MICBIAS) Default: 1 0 - Disable 1 - Enable Function: When enabled, the microphone bias circuit will be in a power-down state. MCLK Divide By 2 (MCLKDIV2) Default: 0 0 - Disabled 1 - Divide by 2 Function: Divides the input MCLK by 2 prior to all internal circuitry. This bit is ignored when the AUTO bit is disabled in Slave Mode. DS679F1 51 CS42L51 6.4 Interface Control (Address 04h) 7 SDOUT->SDIN 6 M/S 5 DAC_DIF2 4 DAC_DIF1 3 DAC_DIF0 2 ADC_IS/LJ 1 DIGMIX 0 MICMIX SDOUT to SDIN Loopback (SDOUT->SDIN) Default: 0 0 - Disabled; SDOUT internally disconnected from SDIN 1 - Enabled; SDOUT internally connected to SDIN Function: Internally loops the signal on the SDOUT pin to SDIN. Master/Slave Mode (M/S) Default: 0 0 - Slave 1 - Master Function: Selects either master or slave operation for the serial port. DAC Digital Interface Format (DAC_DIF[2:0]) Default = 000 DAC_DIF[2:0] 000 001 010 011 100 101 110 100 Description Left-Justified, up to 24-bit data IS, up to 24-bit data Right-Justified, 24-bit data Right-Justified, 20-bit data Right-Justified, 18-bit data Right-Justified, 16-bit data Reserved Reserved Figure 20 on page 41 19 on page 40 21 on page 41 21 on page 41 21 on page 41 21 on page 41 - Function: Selects the digital interface format used for the data in on SDIN. The required relationship between the Left/Right clock, serial clock and serial data is defined by the Digital Interface Format and the options are detailed in the section "Digital Interface Formats" on page 40. 52 DS679F1 CS42L51 ADC IS or Left-Justified (ADC_IS/LJ) Default: 0 0 - Left-Justified 1 - IS Function: Selects either the IS or Left-Justified digital interface format for the data on SDOUT. The required relationship between the Left/Right clock, serial clock and serial data is defined by the Digital Interface Format and the options are detailed in this section . Digital Mix (DIGMIX) Default: 0 DIGMIX 0 DATA_SEL[1:0] xx 00 01 10 11 1 Mix Selected No Mix: ADC to ADC serial port, SDOUT data. No Mix: SDIN data to ADC serial port, SDOUT data. Mix: ADC + SDIN data to ADC serial port, SDOUT data. No Mix: ADC to ADC serial port, SDOUT data. Reserved Function: Selects between the ADC or a digital mix of the ADC and DAC into the serial port to the SDOUT pin. This mix function is affected by the data select bits DATA_SEL[1:0]. Microphone Mix (MICMIX) Default: 0 0 - Disabled; No Mix: Left/Right Channel to ADC serial port, SDOUT. 1 - Enabled; Mix: Differential mix ((A-B)/2)to ADC serial port, SDOUT. Function: Selects between the ADC stereo mix or a differential mix of analog inputs A and B. 6.5 MIC Control (Address 05h) 7 ADC_SNGVOL 6 5 4 3 2 ADCB_DBOOST ADCA_DBOOST MICBIAS_SEL MICBIAS_LVL1 MICBIAS_LVL0 1 0 MICB_BOOST MICA_BOOST ADC Single Volume Control (ADC_SNGVOL) Default: 0 0 - Disabled 1 - Enabled Function: The individual PGA Volume (PGAx_VOLx) and ADC channel attenuation (ADCx_ATTx) levels as well as the ALC A and B enable (ALC_ENx) are independently controlled by their respective control registers when this function is disabled. When enabled, the volume on both channels is determined by the ADCA Attenuator Control register, or the PGAA Control register, and the ADCB Attenuator and PGAB Control registers are ignored. The ALC enable control for channel B is controlled by the ALC A enable when the ADC_SNGVOL bit is enabled and the ALC_ENB control register is ignored. DS679F1 53 CS42L51 ADCx 20 dB Digital Boost (ADCx_DBOOST) Default: 0 0 - Disabled 1 - Enabled Function: Applies a 20 dB digital gain to the input signal on ADC channel x, regardless of the input path. MIC Bias Select (MICBIAS_SEL) Default: 0 0 - MICBIAS on AIN3B/MICIN2 pin 1 - MICBIAS on AIN2B pin Function: Determines the output pin for the internally generated MICBIAS signal. If set to `0'b, the MICBIAS is output on the AIN3B/MICIN2 pin. If set to `1'b, the MICBIAS is output on the AIN2B pin. MIC Bias Level (MICBIAS_LVL[1:0]) Default: 00 00 - 0.8 x VA 01 - 0.7 x VA 10 - 0.6 x VA 11 - 0.5 x VA Function: Determines the output voltage level of the MICBIAS output. MIC X Preamplifier Boost (MICX_BOOST) Default: 0 0 - +16 dB Gain 1 - +32 dB Gain Function: Determines the amount of gain applied to the microphone preamplifier for channel x. 6.6 ADC Control (Address 06h) 7 6 5 4 ADCB_HPFEN ADCB_HPFRZ ADCA_HPFEN ADCA_HPFRZ 3 SOFTB 2 ZCROSSB 1 SOFTA 0 ZCROSSA ADCX High-Pass Filter Enable (ADCX_HPFEN) Default: 1 0 - High-pass filter is disabled 1 - High-pass filter is enabled Function: When this bit is set, the internal high-pass filter will be enabled for ADCx. When set to `0', the high-pass filter will be disabled. For DC measurements, this bit must be cleared to `0'. See "ADC Digital Filter Characteristics" on page 15. 54 DS679F1 CS42L51 ADCX High-Pass Filter Freeze (ADCX_HPFRZ) Default: 0 0 - Continuous DC Subtraction 1 - Frozen DC Subtraction Function: The high-pass filter works by continuously subtracting a measure of the DC offset from the output of the decimation filter. If the ADCx_HPFRZ bit is taken high during normal operation, the current value of the DC offset is frozen, and this DC offset will continue to be subtracted from the conversion result. For DC measurements, this bit must be set to `1'. See "ADC Digital Filter Characteristics" on page 15. Soft Ramp CHX Control (SOFTX) Default: 0 0 - Disabled 1 - Enabled Function: Soft Ramp allows level changes to be implemented via an incremental ramp. ADCx_ATT[7:0] digital attenuation changes are ramped from the current level to the new level at a rate of 0.125 dB per LRCK period. PGAx_VOL[4:0] gain changes are ramped in 0.5 dB steps every 16 LRCK periods. Soft Ramp & Zero Cross Enabled When used in conjunction with the ZCROSSx bit, the PGAx_VOL[4:0] gain changes will occur in 0.5 dB steps and be implemented on a signal zero crossing. Zero Cross CHX Control (ZCROSSX) Default: 0 0 - Disabled 1 - Enabled Function: Zero Cross Enable dictates that signal level changes will occur on a signal zero crossing to minimize audible artifacts. The requested level change will occur after a timeout period of 1024 sample periods (approximately 10.7 ms at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is independently monitored and implemented for each channel. Soft Ramp & Zero Cross Enabled When used in conjunction with the SOFTx bit, the PGAx_VOL[4:0] gain changes will occur in 0.5 dB steps and be implemented on a signal zero crossing. The ADC Attenuator ADCx_ATT[7:0] is not affected by the ZCROSSx bit. DS679F1 SOFTx ZCROSSx Analog PGA Volume (PGAx_VOL[4:0]) Digital Attenuator (ADCx_ATT[7:0]) 0 0 Volume changes immediately. Volume changes immediately. 0 1 Volume changes at next zero cross time. Volume changes immediately. 1 0 Volume changes in 0.5 dB steps. Change volume in 0.125 dB steps. 1 1 Volume changes in 0.5 dB steps at every signal zero-cross. Change volume in 0.125 dB steps. 55 CS42L51 6.7 ADCx Input Select, Invert & Mute (Address 07h) 7 AINB_MUX1 6 AINB_MUX0 5 AINA_MUX1 4 AINA_MUX0 3 INV_ADCB 2 INV_ADCA 1 0 ADCB_MUTE ADCA_MUTE ADCX Input Select Bits (AINX_MUX[1:0]) Default: 00 PDN_PGAx AINx_MUX[1:0] 0 0 0 0 1 1 1 1 00 01 10 11 00 01 10 11 Selected Path to ADC AIN1x-->PGAx AIN2x-->PGAx AIN3x/MICINx-->PGAx AIN3x/MICINx-->Pre-Amp(+16/+32 dB Gain)-->PGAx AIN1x AIN2x AIN3x/MICINx Reserved Function: Selects the specified analog input signal into ADCx. The microphone pre-amplifier is only available when PDN_PGAx is disabled. See Figure 26. AIN1x AIN2x AIN3x AIN1x AIN2x AIN3x / MICINx MUX MUX ADC PGA +16/ 32 dB Decoder AINx_MUX[1:0] PDN_PGAx Figure 26. AIN & PGA Selection ADCX Invert Signal Polarity (INV_ADCX) Default: 0 0 - Disabled 1 - Enabled Function: When enabled, this bit will invert the signal polarity of the ADC x channel. ADCX Channel Mute (ADCX_MUTE) Default: 0 0 - Disabled 1 - Enabled Function: The output of channel x ADC will mute when enabled. The muting function is affected by the ADCx Soft bit (SOFT). 56 DS679F1 CS42L51 6.8 DAC Output Control (Address 08h) 7 6 5 HP_GAIN2 HP_GAIN1 HP_GAIN0 4 DAC_ SNGVOL 3 2 INV_PCMB INV_PCMA 1 0 DACB_MUTE DACA_MUTE Headphone Analog Gain (HP_GAIN[2:0]) Default: 011 HP_GAIN[2:0] Gain Setting 000 001 010 011 100 101 110 111 0.3959 0.4571 0.5111 0.6047 0.7099 0.8399 1.0000 1.1430 Function: These bits select the gain multiplier for the headphone/line outputs. See "Line Output Voltage Characteristics" on page 18 and "Headphone Output Power Characteristics" on page 19. DAC Single Volume Control (DAC_SNGVOL) Default: 0 Function: The individual channel volume levels are independently controlled by their respective Volume Control registers when this function is disabled. When enabled, the volume on all channels is determined by the AOUTA Volume Control register and the AOUTB Volume Control register is ignored. PCMX Invert Signal Polarity (INV_PCMX) Default: 0 0 - Disabled 1 - Enabled Function: When enabled, this bit will invert the signal polarity of the PCM x channel. DACX Channel Mute (DACX_MUTE) Default: 0 0 - Disabled 1 - Enabled Function: The output of channel x DAC will mute when enabled. The muting function is affected by the DACx Soft and Zero Cross bits (DACx_SZC[1:0]). DS679F1 57 CS42L51 6.9 DAC Control (Address 09h) 7 DATA_SEL1 6 DATA_SEL0 5 FREEZE 4 Reserved 3 DEEMPH 2 AMUTE 1 DAC_SZC1 0 DAC_SZC0 DAC Data Selection (DATA_SEL[1:0]) Default: 00 00 - PCM Serial Port to DAC 01 - Signal Processing Engine to DAC 10 - ADC Serial Port to DAC 11 - Reserved Function: Selects the digital signal source for the DAC. Note: Certain functions are only available when the "Signal Processing Engine to DAC" option is selected using these bits. Freeze Controls (FREEZE) Default: 0 Function: This function will freeze the previous settings of, and allow modifications to be made to all control port registers without the changes taking effect until the FREEZE is disabled. To have multiple changes in the control port registers take effect simultaneously, enable the FREEZE bit, make all register changes, then disable the FREEZE bit. DAC De-Emphasis Control (DEEMPH) Default: 0 0 - No De-Emphasis 1 - De-Emphasis Enabled Function: Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control. Enables the digital filter to apply the standard 15s/50s digital de-emphasis filter response for a sample rate of 44.1 kHz. Analog Output Auto MUTE (AMUTE) Default: 0 0 - Auto Mute Disabled 1 - Auto Mute Enabled Function: Enables (or disables) Automatic Mute of the analog outputs after 8192 "0" samples on each digital input channel. 58 DS679F1 CS42L51 DAC Soft Ramp and Zero Cross Control (DAC_SZC[1:0]) Default = 10 00 - Immediate Change 01 - Zero Cross 10 - Soft Ramp 11 - Soft Ramp on Zero Crossings Function: Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control Immediate Change When Immediate Change is selected all volume-level changes will take effect immediately in one step. Zero Cross This setting dictates that signal-level changes, either by gain changes, attenuation changes or muting, will occur on a signal zero crossing to minimize audible artifacts. The requested level change will occur after a timeout period between 1024 and 2048 sample periods (21.3 ms to 42.7 ms at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is independently monitored and implemented for each channel. Note: The LIM_SRDIS bit is ignored. Soft Ramp Soft Ramp allows level changes, either by gain changes, attenuation changes or muting, to be implemented by incrementally ramping, in 1/8 dB steps, from the current level to the new level at a rate of 0.5 dB per 4 left/right clock periods. Soft Ramp on Zero Crossing This setting dictates that signal-level changes, either by gain changes, attenuation changes or muting, will occur in 1/8 dB steps and be implemented on a signal zero crossing. The 1/8 dB level change will occur after a timeout period between 512 and 1024 sample periods (10.7 ms to 21.3 ms at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is independently monitored and implemented for each channel. Note: The LIM_SRDIS bit is ignored. 6.10 ALCX & PGAX Control: ALCA, PGAA (Address 0Ah) & ALCB, PGAB (Address 0Bh) 7 6 ALCX_SRDIS ALCX_ZCDIS 5 Reserved 4 PGAX_VOL4 3 PGAX_VOL3 2 PGAX_VOL2 1 PGAX_VOL1 0 PGAX_VOL0 ALCX Soft Ramp Disable (ALCX_SRDIS) Default: 0 0 - Off 1 - On Function: Overrides the SOFTx bit setting for the ADC. When this bit is set, the ALC attack rate in the PGA will not be dictated by the soft ramp setting. ALC volume-level changes will take effect in one step. DS679F1 59 CS42L51 ALCX Zero Cross Disable (ALCX_ZCDIS) Default: 0 0 - Off 1 - On Function: Overrides the ZCROSSx bit setting for the ADC. When this bit is set, the ALC attack rate in the PGA will not be dictated by the zero cross setting. ALC volume-level changes will take effect immediately in one step. PGA X Gain Control (PGAX_VOL[4:0]) Default: 00000 Binary Code Volume Setting 11000 *** 01010 *** 00000 11111 11110 *** 11001 11010 +12 dB *** +5 dB *** 0 dB -0.5 dB -1 dB *** -3 dB -3 dB Function: The PGAx Gain Control register allows independent setting of the signal levels in 0.5 dB increments as dictated by the ADCx Soft and Zero Cross bits (SOFTx & ZCROSSx) from +12 dB to -3 dB. Gain settings are decoded as shown in the table above. The gain changes are implemented as dictated by the ALCX Soft & Zero Cross bits (ALCX_SZC). Levels are decoded as described in the table above. Note: 6.11 When the ALC is enabled, the PGA is automatically controlled and should not be adjusted manually. ADCx Attenuator: ADCA (Address 0Ch) & ADCB (Address 0Dh) 7 ADCx_ATT7 6 ADCx_ATT6 5 ADCx_ATT5 4 ADCx_ATT4 3 ADCx_ATT3 2 ADCx_ATT2 1 ADCx_ATT1 0 ADCx_ATT0 ADCX Attenuation Control (ADCX_ATT[7:0]) Default: 00h 60 Binary Code Volume Setting 0111 1111 *** 0000 0000 1111 1111 1111 1110 *** 1010 0000 *** 1000 0000 0 dB *** 0 dB -1 dB -2 dB *** -96 dB *** -96 dB DS679F1 CS42L51 Function: The level of ADCX can be adjusted in 1.0 dB increments as dictated by the ADCx Soft and Zero Cross bits (SOFTx & ZCROSSx) from 0 to -96 dB. Levels are decoded in two's complement, as shown in the table above. Note: 6.12 When the ALC is enabled, the Attenuator and PGA volume is automatically controlled and should not be adjusted manually. ADCx Mixer Volume Control: ADCA (Address 0Eh) & ADCB (Address 0Fh) 7 6 5 4 3 2 1 0 MUTE_ADCMIXx ADCMIXx_VOL6 ADCMIXx_VOL5 ADCMIXx_VOL4 ADCMIXx_VOL3 ADCMIXx_VOL2 ADCMIXx_VOL1 ADCMIXx_VOL0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. ADCX Mixer Channel Mute (MUTE_ADCMIXX) Default: 1 0 - Disabled 1 - Enabled Function: The ADC channel X input to the output mixer will mute when enabled. The muting function is affected by the DACX Soft and Zero Cross bits (DACX_SZC[1:0]). ADCX Mixer Volume Control (ADCMIXX_VOL[6:0]) Default = 000 0000 Binary Code Volume Setting 001 1000 *** 000 0000 111 1111 111 1110 *** 001 1001 +12.0 dB *** 0 dB -0.5 dB -1.0 dB *** -51.5 dB Function: The level of the ADCX input to the output mixer can be adjusted in 0.5 dB increments as dictated by the DACX Soft and Zero Cross bits (DACX_SZC[1:0]) from +12 to -51.5 dB. Levels are decoded as shown in the table above. DS679F1 61 CS42L51 6.13 PCMX Mixer Volume Control: PCMA (Address 10h) & PCMB (Address 11h) 7 MUTE_ PCMMIXx 6 PCMMIXx_ VOL6 5 PCMMIXx_ VOL5 4 PCMMIXx_ VOL4 3 PCMMIXx_ VOL3 2 PCMMIXx_ VOL2 1 PCMMIXx_ VOL1 0 PCMMIXx_ VOL0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. PCMX Mixer Channel Mute (MUTE_PCMMIXX) Default = 1 0 - Disabled 1 - Enabled Function: The PCM channel X input to the output mixer will mute when enabled. The muting function is affected by the DACX Soft and Zero Cross bits (DACX_SZC[1:0]). PCMX Mixer Volume Control (PCMMIXX_VOL[6:0]) Default: 000 0000 Binary Code Volume Setting 001 1000 *** 000 0000 111 1111 111 1110 *** 001 1001 +12.0 dB *** 0 dB -0.5 dB -1.0 dB *** -51.5 dB Function: The level of the PCMX input to the output mixer can be adjusted in 0.5 dB increments as dictated by the DACX Soft and Zero Cross bits (DACX_SZC[1:0]) from +12 to -51.5 dB. Levels are decoded as described in the table above. 6.14 Beep Frequency & Timing Configuration (Address 12h) 7 FREQ3 6 FREQ2 5 FREQ1 4 FREQ0 3 ONTIME3 2 ONTIME2 1 ONTIME1 0 ONTIME0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Beep Frequency (FREQ[3:0]) Default: 0000 FREQ[3:0] 0000 0001 0010 0011 0100 62 Frequency Pitch Fs = 12, 24, 48 or 96 kHz 260.87 Hz 521.74 Hz 585.37 Hz 666.67 Hz 705.88 Hz C4 C5 D5 E5 F5 DS679F1 CS42L51 FREQ[3:0] Frequency Pitch Fs = 12, 24, 48 or 96 kHz 0101 0110 0111 1000 1001 1010 1011 1100 1101 1110 1111 774.19 Hz 888.89 Hz 1000.00 Hz 1043.48 Hz 1200.00 Hz 1333.33 Hz 1411.76 Hz 1600.00 Hz 1714.29 Hz 2000.00 Hz 2181.82 Hz G5 A5 B5 C6 D6 E6 F6 G6 A6 B6 C7 Function: The frequency of the beep signal can be adjusted from 260.87 Hz to 2181.82 Hz. Beep frequency will scale directly with sample rate, Fs, but is fixed at the nominal Fs within each speed mode. Refer to Figure 15 on page 36 for single, multiple and continuous beep configurations using the REPEAT and BEEP bits. Beep On Time Duration (ONTIME[3:0]) Default: 0000 TIME[3:0] On Time Fs = 12, 24, 48 or 96 kHz 0000 *** 1111 86 ms *** 5.2 s Function: The on-duration of the beep signal can be adjusted from approximately 86 ms to 5.2 s. The on-duration will scale inversely with sample rate, Fs, but is fixed at the nominal Fs within each speed mode. Refer to Figure 15 on page 36 for single-, multiple- and continuous-beep configurations using the REPEAT and BEEP bits. 6.15 Beep Off Time & Volume (Address 13h) 7 OFFTIME2 6 OFFTIME1 5 OFFTIME0 4 BPVOL4 3 BPVOL3 2 BPVOL2 1 BPVOL1 0 BPVOL0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Beep Off Time (OFFTIME[2:0]) Default: 000 DS679F1 OFFTIME[2:0] Off Time Fs = 12, 24, 48 or 96 kHz 000 001 010 011 100 101 1.23 s 2.58 s 3.90 s 5.20 s 6.60 s 8.05 s 63 CS42L51 OFFTIME[2:0] Off Time Fs = 12, 24, 48 or 96 kHz 110 111 9.35 s 10.80 s Function: The off-duration of the beep signal can be adjusted from approximately 75 ms to 680 ms. The off-duration will scale inversely with sample rate, Fs, but is fixed at the nominal Fs within each speed mode. Refer to Figure 15 on page 36 for single-, multiple- and continuous-beep configurations using the REPEAT and BEEP bits. Beep Volume (BPVOL[4:0]) Default: 00000 Binary Code Volume Setting 00110 *** 00000 11111 11110 *** 00111 +12.0 dB *** 0 dB -2 dB -4 dB *** -50 dB Function: The level of the beep into the output mixer can be adjusted in 2.0 dB increments from +12 dB to -50 dB. Refer to Figure 15 on page 36 for single-, multiple- and continuous-beep configurations using the REPEAT and BEEP bits. Levels are decoded as described in the table above. 6.16 Beep Configuration & Tone Configuration (Address 14h) 7 REPEAT 6 BEEP 5 Reserved 4 TREB_CF1 3 TREB_CF0 2 BASS_CF1 1 BASS_CF0 0 TC_EN Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Repeat Beep (REPEAT) Default: 0 0 - Disabled 1 - Enabled Function: This bit is used in conjunction with the BEEP bit to mix a continuous or periodic beep with the analog output. Refer to Figure 15 on page 36 for a description of each configuration option. Beep (BEEP) Default: 0 0 - Disabled 1 - Enabled Function: 64 DS679F1 CS42L51 This bit is used in conjunction with the REPEAT bit to mix a continuous or periodic beep with the analog output. Note: Re-engaging the beep before it has completed its initial cycle will cause the beep signal to remain ON for the maximum ONTIME duration. Refer to Figure 15 on page 36 for a description of each configuration option. Treble Corner Frequency (TREB_CF[1:0]) Default: 00 00 - 5 kHz 01 - 7 kHz 10 - 10 kHz 11 - 15 kHz Function: The treble corner frequency is user selectable as shown above. Bass Corner Frequency (BASS_CF[1:0]) Default: 00 00 - 50 Hz 01 - 100 Hz 10 - 200 Hz 11 - 250 Hz Function: The bass corner frequency is user-selectable as shown above. Tone Control Enable (TC_EN) Default = 0 0 - Disabled 1 - Enabled Function: The Bass and Treble tone control features are active when this bit is enabled. 6.17 Tone Control (Address 15h) 7 TREB3 6 TREB2 5 TREB1 4 TREB0 3 BASS3 2 BASS2 1 BASS1 0 BASS0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Treble Gain Level (TREB[3:0]) Default: 1000 dB (No Treble Gain) DS679F1 Binary Code Gain Setting 0000 *** 0111 1000 1001 *** 1111 +12.0 dB *** +1.5 dB 0 dB -1.5 dB *** -10.5 dB 65 CS42L51 Function: The level of the shelving treble gain filter is set by Treble Gain Level. The level can be adjusted in 1.5 dB increments from +12.0 to -10.5 dB. Bass Gain Level (BASS[3:0]) Default: 1000 dB (No Bass Gain) Binary Code Gain Setting 0000 *** 0111 1000 1001 *** 1111 +12.0 dB *** +1.5 dB 0 dB -1.5 dB *** -10.5 dB Function: The level of the shelving bass gain filter is set by Bass Gain Level. The level can be adjusted in 1.5 dB increments from +10.5 to -10.5 dB. 6.18 AOUTx Volume Control: AOUTA (Address 16h) & AOUTB (Address 17h) 7 6 5 4 3 2 1 0 AOUTx_VOL7 AOUTx_VOL6 AOUTx_VOL5 AOUTx_VOL4 AOUTx_VOL3 AOUTx_VOL2 AOUTx_VOL1 AOUTx_VOL0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. AOUTX Volume Control (AOUTX_VOL[7:0]) Default = 00h Binary Code Volume Setting 0001 1000 *** 0000 0000 1111 1111 1111 1110 *** 0011 0100 *** 0001 1001 +12.0 dB *** 0 dB -0.5 dB -1.0 dB *** -102 dB *** -102 dB Function: The analog output levels can be adjusted in 0.5 dB increments from +12 to -102 dB as dictated by the DAC Soft and Zero Cross bits (DACX_SZC[1:0]). Levels are decoded in unsigned binary as described in the table above. Note: 66 When the limiter is enabled, the AOUT Volume is automatically controlled and should not be adjusted manually. Alternative volume control may be achieved using the PCMMIXx_VOL[6:0] bits. DS679F1 CS42L51 6.19 PCM Channel Mixer (Address 18h) 7 PCMA1 6 PCMA0 5 PCMB1 4 PCMB0 3 ADCA1 2 ADCA0 1 ADCB1 0 ADCB0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Channel Mixer (PCMx[1:0] & ADCx[1:0]) Default: 00 AOUTA PCMB[1:0] and/or AOUTB ADCB[1:0] PCMA[1:0] and/or ADCA[1:0] 00 L 01 00 L+R -----------2 10 11 R 01 10 11 R L+R -----------2 L Function: Implements mono mixes of the left and right channels as well as a left/right channel swap. 6.20 Limiter Threshold SZC Disable (Address 19h) 7 MAX2 6 MAX1 5 MAX0 4 CUSH2 3 CUSH1 2 CUSH0 1 LIM_SRDIS 0 LIM_ZCDIS Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Maximum Threshold (MAX[2:0]) Default: 000 MAX[2:0] Threshold Setting (dB) 000 0 001 -3 010 -6 011 -9 101 -12 101 -18 110 -24 111 -30 Function: Sets the maximum level, below full scale, at which to limit and attenuate the output signal at the attack rate. Bass, Treble and digital gain settings that boost the signal beyond the maximum threshold may trigger an attack. DS679F1 67 CS42L51 Cushion Threshold (CUSH[2:0]) Default: 000 CUSH[2:0] Threshold Setting (dB) 000 0 001 -3 010 -6 011 -9 101 -12 101 -18 110 -24 111 -30 Function: Sets a cushion level below full scale. This setting is usually set slightly below the maximum (MAX[2:0]) threshold. The Limiter uses this cushion as a hysteresis point for the input signal as it maintains the signal below the maximum as well as below the cushion setting. This provides a more natural sound as the limiter attacks and releases. Limiter Soft Ramp Disable (LIM_SRDIS) Default: 0 0 - Off 1 - On Function: Overrides the DAC_SZC setting. When this bit is set, the Limiter attack and release rate will not be dictated by the soft ramp setting. Note: This bit is ignored when the zero-cross function is enabled (i.e. when DAC_SZC[1:0] = `01'b or `11'b.) Limiter Zero Cross Disable (LIM_ZCDIS) Default: 0 0 - Off 1 - On Function: Overrides the DAC_SZC setting. When this bit is set, the Limiter attack and release rate will not be dictated by the zero-cross setting. 68 DS679F1 CS42L51 6.21 Limiter Release Rate Register (Address 1Ah) 7 LIMIT_EN 6 LIMIT_ALL 5 RRATE5 4 RRATE4 3 RRATE3 2 RRATE2 1 RRATE1 0 RRATE0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Peak Detect and Limiter Enable (LIMIT_EN) Default: 0 0 - Disabled 1 - Enabled Function: Limits the maximum signal amplitude to prevent clipping when this function is enabled. Peak Signal Limiting is performed by digital attenuation. Note: When the limiter is enabled, the AOUT Volume is automatically controlled and should not be adjusted manually. Alternative volume control may be realized using the PCMMIXx_VOL[6:0] bits. Peak Signal Limit All Channels (LIMIT_ALL) Default: 1 0 - Individual Channel 1 - Both channel A & B Function: When set to 0, the peak signal limiter will limit the maximum signal amplitude to prevent clipping on the specific channel indicating clipping. The other channels will not be affected. When set to 1, the peak signal limiter will limit the maximum signal amplitude to prevent clipping on both channels in response to any single channel indicating clipping. Limiter RELEASE Rate (RRATE[5:0]) Default: 111111 Binary Code Release Time 000000 *** 111111 Fastest Release *** Slowest Release Function: Sets the rate at which the limiter releases the digital attenuation from levels below the minimum setting in the limiter threshold register, and returns the analog output level to the AOUTx_VOL[7:0] setting. The limiter release rate is user selectable but is also a function of the sampling frequency, Fs, and the DAC_SZC setting unless the disable bit is enabled. DS679F1 69 CS42L51 6.22 Limiter Attack Rate Register (Address 1Bh) 7 Reserved 6 Reserved 5 ARATE5 4 ARATE4 3 ARATE3 2 ARATE2 1 ARATE1 0 ARATE0 Note: The DATA_SEL[1:0] bits in reg09h must be set to `01'b to enable function control in this register. Limiter Attack Rate (ARATE[5:0]) Default: 000000 Binary Code Attack Time 000000 *** 111111 Fastest Attack *** Slowest Attack Function: Sets the rate at which the limiter attenuates the analog output from levels above the maximum setting in the limiter threshold register. The limiter attack rate is user-selectable but is also a function of the sampling frequency, Fs, and the DAC_SZC setting unless the disable bit is enabled. 6.23 ALC Enable & Attack Rate (Address 1Ch) 7 ALC_ENB 6 ALC_ENA 5 4 3 2 1 0 ALC_ARATE5 ALC_ARATE4 ALC_ARATE3 ALC_ARATE2 ALC_ARATE1 ALC_ARATE0 ALC Enable (ALC_ENX) Default: 0 0 - Disabled 1 - Enabled Function: Enables automatic level control for ADC channel x. Note: When the ALC is enabled, the Attenuator and PGA volume is automatically controlled and should not be adjusted manually. ALC Attack Rate (ARATE[5:0]) Default: 000000 Binary Code Attack Time 000000 *** 111111 Fastest Attack *** Slowest Attack Function: Sets the rate at which the ALC attenuates the analog input from levels above the maximum setting in the ALC threshold register. The limiter attack rate is user-selectable but is also a function of the sampling frequency, Fs, and the SOFTx & ZCROSSx bit settings unless the disable bit for each function is enabled. 70 DS679F1 CS42L51 6.24 ALC Release Rate (Address 1Dh) 7 Reserved 6 Reserved 5 4 3 2 1 0 ALC_RRATE5 ALC_RRATE4 ALC_RRATE3 ALC_RRATE2 ALC_RRATE1 ALC_RRATE0 ALC Release Rate (RRATE[5:0]) Default: 111111 Binary Code Release Time 000000 *** 111111 Fastest Release *** Slowest Release Function: Sets the rate at which the ALC releases the PGA & digital attenuation from levels below the minimum setting in the ALC threshold register, and returns the input level to the PGA_VOL[4:0] & ADCx_ATT[7:0] setting. The ALC release rate is user selectable, but is also a function of the sampling frequency, Fs, and the SOFTx & ZCROSS bit settings unless the disable bit for each function is enabled. 6.25 ALC Threshold (Address 1Eh) 7 MAX2 6 MAX1 5 MAX0 4 MIN2 3 MIN1 2 MIN0 1 Reserved 0 Reserved Maximum Threshold (MAX[2:0]) Default: 000 MAX[2:0] Threshold Setting (dB) 000 0 001 -3 010 -6 011 -9 100 -12 101 -18 110 -24 111 -30 Function: Sets the maximum level, relative to full scale, at which to limit and attenuate the input signal at the attack rate. Minimum Threshold (MIN[2:0]) Default: 000 MIN[2:0] DS679F1 Threshold Setting (dB) 000 0 001 -3 010 -6 71 CS42L51 MIN[2:0] Threshold Setting (dB) 011 -9 100 -12 101 -18 110 -24 111 -30 Function: Sets the minimum level at which to disengage the ALC's attenuation or amplify the input signal at a rate set in the release rate register until levels again reach this minimum threshold. The ALC uses this minimum as a hysteresis point for the input signal as it maintains the signal below the maximum as well as below the minimum setting. This provides a more natural sound as the ALC attacks and releases. 6.26 Noise Gate Configuration & Misc. (Address 1Fh) 7 NG_ALL 6 NG_EN 5 NG_BOOST 4 THRESH2 3 THRESH1 2 THRESH0 1 NGDELAY1 0 NGDELAY0 Noise Gate Channel Gang (NG_ALL) Default: 0 0 - Disabled 1 - Enabled Function: Gangs the noise gate function for channel A and B. When enabled, both channels must fall below the threshold setting for the noise gate attenuation to take effect. Noise Gate Enable (NG_EN) Default: 0 0 - Disabled 1 - Enabled Function: Enables the noise gate. Maximum attenuation is relative to all gain settings applied. Noise Gate Boost (NG_BOOST) and Threshold (THRESH[3:0]) Default: 000 72 THRESH[2:0] Minimum Setting (NG_BOOST = `0'b) Minimum Setting (NG_BOOST = `1'b) 000 001 010 011 100 101 110 111 -64 dB -67 dB -70 dB -73 dB -76 dB -82 dB Reserved Reserved -34 dB -37 dB -40 dB -43 dB -46 dB -52 dB -58 dB -64 dB DS679F1 CS42L51 Function: Sets the threshold level of the noise gate. Input signals below the threshold level will be attenuated to -96 dB. NG_BOOST = `1'b adds 30 dB to the threshold settings. Noise Gate Delay Timing (NGDELAY[1:0]) Default: 00 00 - 50 ms 01 - 100 ms 10 - 150 ms 11 - 200 ms Function: Sets the delay time before the noise gate attacks. Noise gate attenuation is dictated by the SOFTx & ZCROSS bit settings unless the disable bit for each function is enabled. 6.27 Status (Address 20h) (Read Only) 7 Reserved 6 SP_CLKERR 5 SPEA_OVFL 4 SPEB_OVFL 3 2 PCMA_OVFL PCMB_OVFL 1 ADCA_OVFL 0 ADCB_OVFL For all bits in this register, a "1" means the associated error condition has occurred at least once since the register was last read. A "0" means the associated error condition has NOT occurred since the last reading of the register. Reading the register resets all bits to 0. Serial Port Clock Error (SP_CLK Error) Default: 0 Function: Indicates an invalid MCLK to LRCK ratio. See "Serial Port Clocking" on page 38 for valid clock ratios. Note: On initial power up and application of clocks, this bit will be high as the serial port re-synchronizes. Signal Processing Engine Overflow (SPEX_OVFL) Default: 0 Function: Indicates a digital overflow condition within the data path after the signal processing engine. PCMX Overflow (PCMX_OVFL) Default: 0 Function: Indicates a digital overflow condition within the data path of the PCM mix. ADC Overflow (ADCX_OVFL) Default = 0 Function: Indicates that there is an over-range condition anywhere in the CS42L51 ADC signal path of each of the associated ADC's. DS679F1 73 CS42L51 6.28 Charge Pump Frequency (Address 21h) 7 6 5 4 CHRG_FREQ CHRG_FREQ CHRG_FREQ CHRG_FREQ 3 2 1 0 3 2 1 0 Reserved Reserved Reserved Reserved Charge Pump Frequency (CHRG_FREQ[3:0]) Default: 0101 N CHRG_FREQ[3:0] 0 ... 15 0000 ... 1111 Frequency 64xFs ----------------N+2 Function: Alters the clocking frequency of the charge pump in 1/(N+2) fractions of the DAC oversampling rate, 128Fs, should the switching frequency interfere with other system frequencies such as those in the AM radio band. Note: 74 Distortion performance may be affected. DS679F1 CS42L51 7. ANALOG PERFORMANCE PLOTS 7.1 Headphone THD+N versus Output Power Plots Test conditions (unless otherwise specified): Input test signal is a 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Fs = 48 kHz. Plots were taken from the CDB42L51 using an Audio Precision analyzer. -10 G = 0.6047 -15 VA_HP = VA = 1.8 V G = 0.7099 -20 G = 0.8399 -25 -30 G = 1.0000 -35 G = 1.1430 -40 Legend -45 d B r A -50 NOTE: Graph shows the output power per channel (i.e. Output Power = 23 mW into single 16 and 46 mW into stereo 16 with THD+N = 75 dB). -55 -60 -65 -70 -75 -80 -85 -90 -95 -100 0 10m 20m 30m 40m 50m 60m 70m 80m W Figure 27. THD+N vs. Output Power per Channel at 1.8 V (16 load) -10 -15 G = 0.6047 VA_HP = VA = 2.5 V G = 0.7099 -20 -25 G = 0.8399 -30 G = 1.0000 -35 G = 1.1430 -40 Legend -45 d B r A NOTE: Graph shows the output power per channel (i.e. Output Power = 44 mW into single 16 and 88 mW into stereo 16 with THD+N = 75 dB). -50 -55 -60 -65 -70 -75 -80 -85 -90 -95 -100 0 10m 20m 30m 40m 50m 60m 70m 80m W Figure 28. THD+N vs. Output Power per Channel at 2.5 V (16 load) DS679F1 75 CS42L51 G = 0.6047 VA_HP = VA = 1.8 G = 0.7099 -20 -30 G = 0.8399 -35 G = 1.0000 -40 G = 1.1430 -45 Legend -50 NOTE: Graph shows the output power per channel (i.e. Output Power = 22 mW into single 32 and 44 mW into stereo 32 with THD+N = 75 dB). -55 d B r -60 A -65 -70 -75 -80 -85 -90 -95 -100 0 6m 12m 18m 24m 30m 36m 42m 48m 54m 60m W Figure 29. THD+N vs. Output Power per Channel at 1.8 V (32 load) G = 0.6047 -20 VA_HP = VA = 2.5 V -25 G = 0.7099 -30 G = 0.8399 -35 G = 1.0000 -40 G = 1.1430 -45 Legend -50 -55 d B r NOTE: Graph shows the output power per channel (i.e. Output Power = 42 mW into single 32 and 84 mW into stereo 32 with THD+N = 75 dB). -60 A -65 -70 -75 -80 -85 -90 -95 -100 0 5m 10m 15m 20m 25m 30m 35m 40m 45m 50m 55m 60m W Figure 30. THD+N vs. Output Power per Channel at 2.5 V (32 load) 76 DS679F1 CS42L51 7.2 Headphone Amplifier Efficiency The architecture of the headphone amplifier is that of typical class AB amplifiers. Test conditions (unless otherwise specified): Input test signal is a 997 Hz sine wave; Power Consumption Mode 6 - Stereo Playback w/16 load. HP_GAIN = 1.1430. Best efficiency is realized when the amplifier outputs maximum power. VA_HP = VA = 1.8 V Figure 31. Power Dissipation vs. Output Power into Stereo 16 VA_HP = VA = 1.8 V Figure 32. Power Dissipation vs. Output Power into Stereo 16 (Log Detail) DS679F1 77 CS42L51 7.3 ADC_FILT+ Capacitor Effects on THD+N The value of the capacitor on the ADC_FILT+ pin, 16, affects the low frequency total harmonic distortion + noise (THD+N) performance of the ADC. Larger capacitor values yield significant improvement in THD+N at low frequencies. Figure 33 shows the THD+N versus frequency for the ADC analog input. Plots were taken from the CDB42L51 using an Audio Precision analyzer. -60 1 F -64 -68 10 F -72 22 F Legend - Capacitor Value on ADC_FILT+ -76 d B F S -80 -84 -88 -92 -96 -100 20 50 100 200 500 1k 2k 5k 10k 20k Hz Figure 33. ADC THD+N vs. Frequency w/Capacitor Effects 78 DS679F1 CS42L51 8. EXAMPLE SYSTEM CLOCK FREQUENCIES 8.1 Auto Detect Enabled Sample Rate LRCK (kHz) 1024x MCLK (MHz) 1536x 2048x* 8 11.025 12 3072x* 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 24.5760 33.8688 36.8640 Sample Rate LRCK (kHz) 512x MCLK (MHz) 768x 1024x* 16 22.05 24 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 16.3840 22.5792 24.5760 16.3840 22.5792 24.5760 Sample Rate LRCK (kHz) MCLK (MHz) 384x 512x* 256x 32 44.1 48 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 Sample Rate LRCK (kHz) 128x 192x 64 88.2 96 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 16.3840 22.5792 24.5760 MCLK (MHz) 256x* 16.3840 22.5792 24.5760 1536x* 24.5760 33.8688 36.8640 768x* 24.5760 33.8688 36.8640 384x* 24.5760 33.8688 36.8640 *The"MCLKDIV2" pin 4 must be set HI. DS679F1 79 CS42L51 8.2 80 Auto Detect Disabled Sample Rate LRCK (kHz) 512x 8 11.025 12 6.1440 768x MCLK (MHz) 1024x 1536x 2048x 3072x 6.1440 8.4672 9.2160 8.1920 11.2896 12.2880 16.3840 22.5792 24.5760 24.5760 33.8688 36.8640 Sample Rate LRCK (kHz) 256x 384x 512x 16 22.05 24 6.1440 6.1440 8.4672 9.2160 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 MCLK (MHz) 768x 12.2880 16.9344 18.4320 Sample Rate LRCK (kHz) 256x 32 44.1 48 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 Sample Rate LRCK (kHz) 128x 192x 64 88.2 96 8.1920 11.2896 12.2880 12.2880 16.9344 18.4320 1024x 1536x 16.3840 22.5792 24.5760 24.5760 33.8688 36.8640 MCLK (MHz) 384x 512x 16.3840 22.5792 24.5760 MCLK (MHz) 256x 16.3840 22.5792 24.5760 768x 24.5760 33.8688 36.8640 384x 24.5760 33.8688 36.8640 DS679F1 CS42L51 9. PCB LAYOUT CONSIDERATIONS 9.1 Power Supply, Grounding As with any high-resolution converter, the CS42L51 requires careful attention to power supply and grounding arrangements if its potential performance is to be realized. Figure 1 on page 10 shows the recommended power arrangements, with VA and VA_HP connected to clean supplies. VD, which powers the digital circuitry, may be run from the system logic supply. Alternatively, VD may be powered from the analog supply via a ferrite bead. In this case, no additional devices should be powered from VD. Extensive use of power and ground planes, ground plane fill in unused areas and surface mount decoupling capacitors are recommended. Decoupling capacitors should be as close to the pins of the CS42L51 as possible. The low value ceramic capacitor should be closest to the pin and should be mounted on the same side of the board as the CS42L51 to minimize inductance effects. All signals, especially clocks, should be kept away from the DAC_FILT+/ADC_FILT+ and VQ pins in order to avoid unwanted coupling into the modulators. The DAC_FILT+/ADC_FILT+ and VQ decoupling capacitors, particularly the 0.1 F, must be positioned to minimize the electrical path from DAC_FILT+/ADC_FILT+ and AGND. The CDB42L51 evaluation board demonstrates the optimum layout and power supply arrangements. 9.2 QFN Thermal Pad The CS42L51 is available in a compact QFN package. The under side of the QFN package reveals a large metal pad that serves as a thermal relief to provide for maximum heat dissipation. This pad must mate with an equally dimensioned copper pad on the PCB and must be electrically connected to ground. A series of vias should be used to connect this copper pad to one or more larger ground planes on other PCB layers. In split ground systems, it is recommended that this thermal pad be connected to AGND for best performance. The CS42L51 evaluation board demonstrates the optimum thermal pad and via configuration. DS679F1 81 CS42L51 10.ADC & DAC DIGITAL FILTERS 82 Figure 34. ADC Passband Ripple Figure 35. ADC Stopband Rejection Figure 36. ADC Transition Band Figure 37. ADC Transition Band Detail Figure 38. DAC Passband Ripple Figure 39. DAC Stopband Figure 40. DAC Transition Band Figure 41. DAC Transition Band (Detail) DS679F1 CS42L51 11.PARAMETER DEFINITIONS Dynamic Range The ratio of the rms value of the signal to the rms sum of all other spectral components over the specified bandwidth. Dynamic Range is a signal-to-noise ratio measurement over the specified band width made with a -60 dBFS signal. 60 dB is added to resulting measurement to refer the measurement to full-scale. This technique ensures that the distortion components are below the noise level and do not affect the measurement. This measurement technique has been accepted by the Audio Engineering Society, AES17-1991, and the Electronic Industries Association of Japan, EIAJ CP-307. Expressed in decibels. Total Harmonic Distortion + Noise The ratio of the rms value of the signal to the rms sum of all other spectral components over the specified band width (typically 10 Hz to 20 kHz), including distortion components. Expressed in decibels. Measured at -1 and -20 dBFS as suggested in AES17-1991 Annex A. Frequency Response A measure of the amplitude response variation from 10 Hz to 20 kHz relative to the amplitude response at 1 kHz. Units in decibels. Interchannel Isolation A measure of crosstalk between the left and right channel pairs. Measured for each channel at the converter's output with no signal to the input under test and a full-scale signal applied to the other channel. Units in decibels. Interchannel Gain Mismatch The gain difference between left and right channel pairs. Units in decibels. Gain Error The deviation from the nominal full-scale analog output for a full-scale digital input. Gain Drift The change in gain value with temperature. Units in ppm/C. Offset Error The deviation of the mid-scale transition (111...111 to 000...000) from the ideal. Units in mV. DS679F1 83 CS42L51 12.PACKAGE DIMENSIONS 32L QFN (5 X 5 mm BODY) PACKAGE DRAWING e b D Pin #1 Corner Pin #1 Corner E2 E A1 L D2 A Top View DIM MIN A A1 b D D2 E E2 e L -0.0000 0.0071 0.1280 0.1280 0.0118 Bottom View Side View INCHES NOM --0.0091 0.1969 BSC 0.1299 0.1969 BSC 0.1299 0.0197 BSC 0.0157 MAX MIN 0.0394 0.0020 0.0110 -0.00 0.18 0.1319 3.25 0.1319 3.25 0.0197 0.30 MILLIMETERS NOM --0.23 5.00 BSC 3.30 5.00 BSC 3.30 0.50 BSC 0.40 NOTE MAX 1.00 0.05 0.28 3.35 3.35 0.50 1 1 1,2 1 1 1 1 1 1 JEDEC #: MO-220 Controlling Dimension is Millimeters. 1. Dimensioning and tolerance per ASME Y 14.5M-1995. 2. Dimensioning lead width applies to the plated terminal and is measured between 0.20 mm and 0.25 mm from the terminal tip. THERMAL CHARACTERISTICS Parameter Junction to Ambient Thermal Impedance 84 2 Layer Board 4 Layer Board Symbol Min Typ Max Units JA - 52 38 - C/Watt DS679F1 CS42L51 13.ORDERING INFORMATION Product CS42L51 CDB42L51 CRD42L51 Description Low-Power Stereo CODEC w/HP Amp for Portable Apps CS42L51 Evaluation Board CS42L51 Reference Design Package Pb-Free Grade Temp Range Commercial -10 to +70 C 32L-QFN Yes Automotive -40 to +85 C Container Order # Rail CS42L51-CNZ Tape & Reel CS42L51-CNZR Rail CS42L51-DNZ Tape & Reel CS42L51-DNZR - No - - - CDB42L51 - No - - - CRD42L51 14.REFERENCES 1. Cirrus Logic, AN18: Layout and Design Rules for Data Converters and Other Mixed Signal Devices, Version 6.0, February 1998. 2. Cirrus Logic, Techniques to Measure and Maximize the Performance of a 120 dB, 96 kHz A/D Converter Integrated Circuit, by Steven Harris, Steven Green and Ka Leung. Presented at the 103rd Convention of the Audio Engineering Society, September 1997. 3. Cirrus Logic, A Stereo 16-bit Delta-Sigma A/D Converter for Digital Audio, by D.R. Welland, B.P. Del Signore, E.J. Swanson, T. Tanaka, K. Hamashita, S. Hara, K. Takasuka. Paper presented at the 85th Convention of the Audio Engineering Society, November 1988. 4. Cirrus Logic, The Effects of Sampling Clock Jitter on Nyquist Sampling Analog-to-Digital Converters, and on Oversampling Delta Sigma ADC's, by Steven Harris. Paper presented at the 87th Convention of the Audio Engineering Society, October 1989. 5. Cirrus Logic, An 18-Bit Dual-Channel Oversampling Delta-Sigma A/D Converter, with 19-Bit Mono Application Example, by Clif Sanchez. Paper presented at the 87th Convention of the Audio Engineering Society, October 1989. 6. Cirrus Logic, How to Achieve Optimum Performance from Delta-Sigma A/D and D/A Converters, by Steven Harris. Presented at the 93rd Convention of the Audio Engineering Society, October 1992. 7. Cirrus Logic, A Fifth-Order Delta-Sigma Modulator with 110 dB Audio Dynamic Range, by I. Fujimori, K. Hamashita and E.J. Swanson. Paper presented at the 93rd Convention of the Audio Engineering Society, October 1992. 8. Philips Semiconductor, The IC-Bus Specification: Version 2.1, January 2000. http://www.semiconductors.philips.com DS679F1 85 CS42L51 15.REVISION HISTORY Revision 86 Changes A1 Initial Release A2 Renamed pin 14, FILT1+, to DAC_FILT+ and pin 16, FILT2+, to ADC_FILT+. Added 1.5 F capacitor recommendation to figure "Typical Connection Diagram (Software Mode)" on page 10. Removed the 0.1F capacitors from pins DAC_FILT+, ADC_FILT+ and VQ on the figures "Typical Connection Diagram (Software Mode)" on page 10 and "Typical Connection Diagram (Hardware Mode)" on page 11. Added DAC Isolation specification to "Analog Input Characteristics (Commercial - CNZ)" on page 13 and "Analog Input Characteristics (Automotive - DNZ)" on page 14. Corrected specification table "Headphone Output Power Characteristics" on page 19. Removed td timing specification from table in section "Switching Specifications - Serial Port" on page 20. Added ts(SDO-SK) and th(SK-SDO) timing specification to table in section "Switching Specifications - Serial Port" on page 20. Adjusted timing specifications ts(SD-SK) from 0 ns to 20 ns and th from 50 ns to 20 ns in table in section "Switching Specifications - Serial Port" on page 20. Added MIC Bias PSRR specification to "DC Electrical Characteristics" on page 24. Adjusted specification table "Power Consumption" on page 25. Removed QSM clock ratios 128, 192, 256, 384 and HSM ratios 128, 192 from Table 3 on page 39. Modified Digital Mix description in section "Digital Mix (DIGMIX)" on page 53. Corrected DAC Zero Cross timeout period in section "Zero Cross" on page 59. Adjusted BEEP off time settings in section "Beep Off Time (OFFTIME[2:0])" on page 63. Modified BEEP description in section "Beep (BEEP)" on page 64. Adjusted the minimum settings for the "Noise Gate Boost (NG_BOOST) and Threshold (THRESH[3:0])" on page 72. Swapped bits PCMA_OVFL w/PCMB_OVFL and ADCA_OVFL w/ADCB_OVFL in register "Status (Address 20h) (Read Only)" on page 73. Corrected Charge Pump Frequency setting in section "Charge Pump Frequency (CHRG_FREQ[3:0])" on page 74. Added sections "Headphone THD+N versus Output Power Plots" on page 75 and "ADC_FILT+ Capacitor Effects on THD+N" on page 78. DS679F1 CS42L51 Revision Changes PP1 Adjusted the minimum voltage specification in "Specified Operating Conditions" section on page 12. Adjusted Ambient Operating Temp. specification in "Absolute Maximum Ratings" section on page 12. Adjusted maximum "Analog In to PGA to ADC" THD+N performance specification in "Analog Input Characteristics (Commercial - CNZ)" on page 13. Added Offset Error specification to "Analog Input Characteristics (Commercial - CNZ)" on page 13 and "Analog Input Characteristics (Automotive - DNZ)" on page 14. Corrected Interchannel Gain Mismatch specification in "Analog Input Characteristics (Commercial - CNZ)" on page 13 and "Analog Input Characteristics (Automotive - DNZ)" on page 14. Adjusted ADC full scale input voltage specification in "Analog Input Characteristics (Commercial - CNZ)" on page 13 and "Analog Input Characteristics (Automotive - DNZ)" on page 14. Corrected Group Delay characteristic in table in section "ADC Digital Filter Characteristics" on page 15. Adjusted maximum "RL = 10k" THD+N performance specification in "Analog Output Characteristics (Commercial - CNZ)" on page 16 and "Analog Output Characteristics (Automotive - DNZ)" on page 17. Corrected Group Delay characteristic in table in section "Combined DAC Interpolation & on-Chip Analog FIlter Response" on page 20. Adjusted timing specifications td(MSB) from 40 ns to 52 ns and ts(SDO-SK) from 30 ns to 20 ns in table in section "Switching Specifications - Serial Port" on page 20. Adjusted IC timing specification tack from 1000 ns to 3450 ns in table in section "" on page 21. Adjusted High-Level Input Voltage specifications VIH from 0.65VL to 0.68VL and VIL from 0.35VL to 0.32VL in table in section "Digital Interface Specifications & Characteristics" on page 24. Adjusted the +20 dB Digital Boost block before the ALC feedback path in Figure 8 on page 28. Modified ALC Recommended Settings in section "Automatic Level Control (ALC)" on page 32. Modified step 2 of the "Recommended Power-Down Sequence" on page 42. Corrected default values for ALC and Limiter Release Rates shown in "Register Quick Reference" on page 46. Corrected default value for the DAC_SZC bits and Added AMUTE bit and description in "DAC Control (Address 09h)" on page 58. Added section "Headphone Amplifier Efficiency" on page 77. Corrected ADC Filter Response shown in Figures 34, 35, 36, and 37 on page 82. Corrected ADC_SNGVOL description in "MIC Control (Address 05h)" on page 53. F1 DS679F1 Final Release 87 CS42L51 Contacting Cirrus Logic Support For all product questions and inquiries, contact a Cirrus Logic Sales Representative. To find the one nearest to you, go to www.cirrus.com. IMPORTANT NOTICE Cirrus Logic, Inc. and its subsidiaries ("Cirrus") believe that the information contained in this document is accurate and reliable. However, the information is subject to change without notice and is provided "AS IS" without warranty of any kind (express or implied). Customers are advised to obtain the latest version of relevant information to verify, before placing orders, that information being relied on is current and complete. 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