8.3.2.1 Audio Serial Interface

The audio interface port is a 3-wire serial port with the signals LRCK, BCK, and DIN. BCK is the serial audio bit clock, used to clock the serial data present on DIN into the serial shift register of the audio interface. Serial data is clocked into the PCM5252 on the rising edge of BCK. LRCK is the serial audio left/right word clock. LRCK polarity for left/right is given by the format selected.

The PCM5252 requires the synchronization of LRCK and system clock, but does not need a specific phase relation between LRCK and system clock.

If the relationship between LRCK and system clock changes more than ±5 SCK, internal operation (using an onchip oscillator) is initialized within one sample period and analog outputs are forced to the bipolar zero level until resynchronization between LRCK and system clock is completed.

If the relationship between LRCK and BCK are invalid more than 4 LRCK periods, internal operation (using an onchip oscillator) is initialized within one sample period and analog outputs are forced to the bipolar zero level until resynchronization between LRCK and BCK is completed.

8.3.2.2 PCM Audio Data Formats

The PCM5252 supports industry-standard audio data formats, including standard I²S and left-justified. Data formats are selected via Register (Pg0Reg40). All formats require binary twos-complement, MSB-first audio data; up to 32-bit audio data is accepted.

The PCM5252 also supports right-justified and TDM/DSP in software control mode. I²S, LJ, RJ, and TDM/DSP are selected using Register (Pg0Reg40). All formats require binary twos-complement, MSB-first audio data. Up to 32 bits are accepted. Default setting is I²S and 24-bit word length.

Figure 18. Left-Justified Audio Data Format

I²S Data Format; L-channel = LOW, R-channel = HIGH

Figure 19. I²S Audio Data Format

TDM/DSP Data Format; L-channel = FIRST, R-channel = LAST with OFFSET = 1

Figure 20. TDM/DSP 2 Audio Data Format

TDM/DSP Data Format; L-channel = FIRST, R-channel = LAST with OFFSET = N

Figure 21. TDM/DSP 3 Audio Data Format

8.3.2.3 Zero Data Detect

The PCM5252 has a zero-detect function. When the device detects the continuous zero data for both left and right channels, or separate channels, Analog mutes are set to both OUTL and OUTR, or separate OUTL and OUTR. These are controlled by Page 0, Register 65, D(2:1) as shown in Table 5.

8.3.4.1 PCM5252 Audio Processing Options

8.3.4.2 Interpolation Filter

The PCM5252 provides 4 types of interpolation filters, selectable by writing to Page 0, Register 43, D(4:0).

Additional RAM based Hybrid Flows can be implemented by selecting Program 31, and downloading instructions and coefficients to the device.

The PCM5252 supports four sampling modes (single rate, dual rate, quad rate, and octal rate) which produce different oversampling rates (OSR) in the interpolation digital filter operation. These are shown in Table 8.

SPACE

Figure 22. Normal x8 Interpolation Filter
Frequency Response

Figure 24. Normal x8 Interpolation Filter Passband Ripple

Figure 23. Normal x8 Interpolation Filter
Impulse Response

SPACE

Figure 25. Normal x4 Interpolation Filter
Frequency Response

Figure 27. Normal x4 Interpolation Filter Passband Ripple

Figure 26. Normal x4 Interpolation Filter
Impulse Response

SPACE

Figure 28. Normal x2 Interpolation Filter
Frequency Response

Figure 30. Normal x2 Interpolation Filter Passband Ripple

Figure 29. Normal x2 Interpolation Filter
Impulse Response

SPACE

Figure 31. Low Latency x8 Interpolation Filter
Frequency Response

Figure 33. Low Latency x8 Interpolation Filter Passband Ripple

Figure 32. Low Latency x8 Interpolation Filter
Impulse Response

SPACE

Figure 34. Low Latency x4 Interpolation Filter
Frequency Response

Figure 36. Low Latency x4 Interpolation Filter Passband Ripple

Figure 35. Low Latency x4 Interpolation Filter
Impulse Response

SPACE

Figure 37. Low Latency x2 Interpolation Filter
Frequency Response

Figure 39. Low Latency x2 Interpolation Filter Passband Ripple

Figure 38. Low Latency x2 Interpolation Filter
Impulse Response

SPACE

Figure 40. Asymmetric FIR x8 Interpolation Filter Frequency Response, Single Rate

Figure 42. Asymmetric FIR x8 Interpolation Filter Passband Ripple, Single Rate

Figure 41. Asymmetric FIR x8 Interpolation Filter Impulse Response, Single Rate

SPACE

Figure 43. Asymmetric FIR x4 Interpolation Filter Frequency Response, Dual Rate

Figure 45. Asymmetric x4 Interpolation Filter Passband Ripple, Dual Rate

Figure 44. Asymmetric FIR x4 Interpolation Filter Impulse Response, Dual Rate

SPACE

Figure 46. Asymmetric FIR x2 Interpolation Filter Frequency Response, Quad Rate

Figure 48. Asymmetric x2 Interpolation Filter Passband Ripple, Quad Rate

Figure 47. Asymmetric FIR x2 Interpolation Filter Impulse Response, Quad Rate

SPACE

Figure 49. High-Attentuation x8 Interpolation Filter Frequency Response, Single Rate

Figure 51. High-Attentuation x8 Interpolation Filter Passband Ripple, Single Rate

Figure 50. High-Attentuation x8 Interpolation Filter Impulse Response, Single Rate

SPACE

Figure 52. High-Attentuation x4 Interpolation Filter Frequency Response, Dual Rate

Figure 54. High-Attentuation x4 Interpolation Filter Passband Ripple, Dual Rate

Figure 53. High-Attentuation x4 Interpolation Filter Impulse Response, Dual Rate

SPACE

Figure 55. High-Attentuation x2 Interpolation Filter Frequency Response, Quad Rate

Figure 57. High-Attentuation x2 Interpolation Filter Passband Ripple, Quad Rate

Figure 56. High-Attentuation x2 Interpolation Filter Impulse Response, Quad Rate

8.3.4.3 Overview

The PCM5252 features a configurable miniDSP core. The algorithms for the miniDSP are loaded into the device after power up. The miniDSP has direct access to the digital stereo audio stream, offering the possibility for advanced DSP algorithms with very low group delay. The miniDSP can run up to 1024 instructions on every audio sample at a 48 kHz sample rate.

The PCM5252 Smart Amplifier uses a mix of code sources. ROM based process flow and RAM based process flow. In the program, different algorithms are called from ROM – such as EQ, DRC and Zero Crossing volume control enabling a faster program load.

8.3.4.4 Smart SOA

The "Safe Operating Area" (SOA) for a loudspeaker is based on its electro-mechanical-thermal model. Depending on a speaker's inefficiency, some of the power is dissipated as heat rather than mechanical/acoustic energy. By understanding the characteristics of the speaker, Smart Amp is able to drive the speaker harder, without causing the speaker to thermally overload; or, suffer voice coil over-exclusion and fail. SMART SOA are parameters that are differentiated by a PPC GUI into coefficients that the algorithm uses.

8.3.4.5 Smart BASS

Smart Bass is an intelligent True Bass Alignment algorithm. Smart Bass uses the combination of the speaker model and a desired target response selected by the user to equalize the speaker in the bass region. This target response is critical for the sound character and the user can apply the same target response to very different speakers and get the same sound.

In conventional adaptive Bass Boost Algorithms, designers need to vary the amount of bass boost whenever the output volume is changed. This approach is very much an "open loop" process. Smart Bass is a new proprietary algorithm that combines: True bass extension (in bandwidth and amplitude) and Psycho-acoustic bass extension, with a smart adaptive control.

Smart Bass varies the mix of True Bass extension and Psycho-acoustic bass extension in real time, depending on the loudspeakers position in its SOA.

Smart Bass dynamically switches between True Bass and Psycho-acoustic extension based on a number of parameters such as:

• Capabilities and properties of the speaker, including Q compensation
• Music type
• Volume setting
• Temperature
• User preferences
• Designer preferences

8.3.4.6 Smart Protection

The two main failure mechanisms for loudspeakers are over temperature and over excursion. By modeling the current state of the speaker, Smart Protection adaptively changes various settings in Smart Amplifier to avoid over temperature and over excursion. Design engineers must first provide details of the loudspeaker (driver and enclosure) into the GUI. From there the appropriate coefficients are generated for the algorithm.

8.3.4.7 Implementing a Real World Design

Traditionally, system developers and hardware engineers use graphic equalizers in trial-and-error fashion to boost the bass for each new speaker until the sound is right (or "good enough" in many cases). However, this typically results in a strange combined response with too much phase shift. This process must be repeated every time a new speaker is selected. The Smart Bass concept uses the GUI to select a desired target response takes the speaker out of the equation. By this approach users can obtain a target response with minimum phase warp and time domain ringing which gives a speedy and tight bass. Conversely, users can select a target response that has lots of ringing to give a classical heavy ‘oomph’ bass.

8.3.4.8 Digital Output

The PCM5252 supports an SDOUT output. This can be selected within the process flow, and driven out of a GPIO pin selected in the register map (e.g. Page 0 / Register 80). The I2S output can be fed back to the signal host and used for echo cancellation.

8.3.4.9 Software

Software selection for the PCM5252 is supported through TI's comprehensive PurePath^TM Console Development Environment; a powerful, easy-to-use tool designed specifically to simplify development on the PCM5252 platform. Visit the PCM5252 product folder on www.ti.com to learn more about PurePath™ Console and the latest status on available, ready-to-use DSP algorithms.

8.3.4.10 Process Flow

An example of the default Process Flow available for the PCM5252 in the PurePath™ Console target is shown below:

Figure 58. Example Processflow

This process flow has from input to output:

• Volume block, from -110 db to +6 dB with 0.5 dB steps, including a fixed gain block of 0dB to 12 dB gain
• monobass mixer – mixes the bass into mono below the set frequency, useful for systems where left and right speaker shares the same cabinet volume, bypassed when not needed
• 10 Biquads for filtering and EQ. The PPC GUI have an advanced biquad control where various filter and eq options can be set and controlled.
• SmartAmp block, containing all the blocks for bass Q compensation, bass alignment, excursion control and power limited
• Digital monitor output enabled on GPIO3

8.3.5.1 Analog Outputs

The PCM5252 devices include a two-channel DAC, with differential outputs. Each pin has a full-scale output voltage is 2.1 V_rms with ground center output. This equates to a 4.2 V_rms differential output. A DC-coupled load is supported in addition to an AC-coupled load, if the load resistance conforms to the specification. The PCM5252 DAC outputs on the OUTLP, OUTLN, OUTRP, and OUTRN terminals have market-leading low out-of-band noise, which offer up to 20-dB lower out-of-band noise compared with existing DAC technology.

Many applications require an external low-pass RC filter (470 Ω + 1.2 nF) to provide sufficient out-of-band noise rejection. This RC filter provides the added advantage of improved protection against ESD damage.

The PCM5252 can also support single ended outputs, using OUTLP and OUTRP respectively. A single 470-Ω and 2.2-nF capacitor can be used on each pin in single ended mode.

The choice between VREF and VCOM modes affects the maximum output level. This is explained in Recommended Operating Conditions.

Figure 59. Optional Low Pass Filters

8.3.5.2 Choosing Between VREF and VCOM Modes

VREF mode is the default configuration. This mode allows full 2.1-V_rms signal output. As shown in Recommended Operating Conditions, the minimum AVDD to avoid clipping is 3.2 V.

VCOM mode allows setting a custom common-mode voltage when required by the application. This somewhat limits the output signal swing before clipping.

8.3.5.3 Digital Volume Control

A basic digital volume control with range from 24 dB to –103 dB and mute is available on each channels by Page 0, Resister 61, D(7:0) for L-ch and Register 62, D(7:0) for R-ch. These volume controls all have 0.5-dB step programmability over most gain and attenuation ranges. Table 21 lists the detailed gain versus programmed setting for this basic volume control. Volume can be changed for both L-ch and R-ch at the same time or independently by Page 0, Register 60, D(1:0). When D(1:0) set 00 (default), independent control is selected. When D(1:0) set 01, R-ch accords with L-ch volume. When D(1:0) set 10, L-ch accords with R-ch volume. To set D(1:0) to 11 is prohibited.

Ramp-up frequency and ramp-down frequency can be controlled by Page 0, Register 63, D(7:6) and D(3:2) as shown in Table 22. Also Ramp-up step and ramp-down step can be controlled by Page 0, Register 63 D(5:4) and D(1:0) as shown in Table 23.

8.3.5.4 Analog Gain Control

Analog gain control can be selected between 2-V_rms FS (0 dB) or 1-V_rms FS (–6 dB). Gain is controlled through hardware by the AGNS pin, and through software (SPI/I²C), Page 1, Register 2, D4(L-ch) / D0(R-ch).

8.3.6.1 Clocking Overview

The PCM5252 devices have flexible systems for clocking. Internally, the device requires a number of clocks, mostly at related clock rates to function correctly. All of these clocks can be derived from the serial audio interface in one form or another.

Figure 60. Audio Flow with Respective Clocks

As shown in Figure 60 the data flows at the sample rate (f_S). Once the data is brought into the serial audio interface, it gets processed, interpolated and modulated all the way to 128 × f_S before arriving at the current segments for the final digital to analog conversion.

The clock tree is shown in Figure 61.

Figure 61. PCM5252 Clock Distribution Tree

The serial audio interface typically has 4 connections: SCK (system master clock), BCK (bit clock), LRCK (left right word clock), and DIN (data). The device has an internal PLL that is used to take either SCK or BCK and create the higher rate clocks required by the interpolating processor and the DAC clock. This allows the device to operate with or without an external SCK.

In situations where the highest audio performance is required, it is suggested that the SCK is brought to the device, along with BCK and LRCK. The device should be configured so that the PLL is only providing a clock source to the miniDSP. By ensuring that the DACCK (DAC Clock) is being driven by the external SCK source, jitter evident in the PLL (in all PLLs) is kept out of the DAC, charge pump, and oversampling system.

Everything else should be a division of the incoming SCK. This is done by setting DAC CLK Source Mux (SDAC in Figure 61) to use SCK as a source, rather than the output of the SCK/PLL Mux. Code examples for this are available in SLASE12.

When the Auto Clock Configuration bit is set (Page 0/ Register 0x25), no additional clocks configuration is required. However, when setting custom PLL values and so forth, the target output rates should match those shown in the recommended PLL values of Table 122.

8.3.6.2 Clock Slave Mode With Master and System Clock (SCK) Input (4 Wire I²S)

The PCM5252 requires a system clock to operate the digital interpolation filters and advanced segment DAC modulators. The system clock is applied at the SCK input and supports up to 50 MHz. The PCM5252 system-clock detection circuit automatically senses the system-clock frequency. Common audio sampling frequencies in the bands of 8 kHz, 16 kHz, (32 kHz - 44.1 kHz - 48 kHz), (88.2kHz - 96kHz), (176.4 kHz - 192 kHz), and 384 kHz with ±4% tolerance are supported. Values in the parentheses are grouped when detected, (for example, 88.2 kHZ and 96 kHz are detected as double rate, and 32 kHz, 44.1 kHz and 48 kHz are detected as single rate.)

In the presence of a valid bit SCK, BCK and LRCK in software mode, the device will auto-configure the clock tree and PLL to drive the miniDSP as required.

The sampling frequency detector sets the clock for the digital filter, Delta Sigma Modulator (DSM) and the Negative Charge Pump (NCP) automatically. Table 24 shows examples of system clock frequencies for common audio sampling rates.

SCK rates that are not common to standard audio clocks, between 1 MHz and 50 MHz, are only supported in software mode by configuring various PLL and clock-divider registers. This programmability allows the device to become a clock master and drive the host serial port with LRCK and BCK, from a non-audio related clock (for example, using 12 MHz to generate 44.1 kHz [LRCK] and 2.8224 MHz [BCK]).

See Timing Requirements: PCM Audio Data for clock timing requirements.

8.3.6.3 Clock Slave Mode With BCK PLL to Generate Internal Clocks (3-Wire PCM)

The system clock PLL mode allows designers to use a simple 3-wire I²S audio source. The 3-wire source reduces the need for a high frequency SCK, making PCB layout easier, and reduces high frequency electromagnetic interference.

In hardwired mode, the internal PLL is disabled as soon as an external SCK is supplied.

In hardwired mode, the device starts up expecting an external SCK input, but if BCK and LRCK start correctly while SCK remains at ground level for 16 successive LRCK periods, then the internal PLL starts, automatically generating an internal SCK from the BCK reference. Specific BCK rates are required to generate an appropriate master clock. Table 25 describes the minimum and maximum BCK per LRCK for the integrated PLL to automatically generate an internal SCK.

In software mode, the user must set all the PLL registers and clock divider registers for referencing BCK. See Clock Generation Using the PLL for more information. Recommended values can be found in Table 122.

8.3.6.4 Clock Generation Using the PLL

The PCM5252 supports a wide range of options to generate the required clocks for the DAC section as well as interface and other control blocks as shown in Figure 61.

The clocks for the PLL require a source reference clock. This clock is sourced as the incoming BCK or SCK. In software mode, a GPIO can also be used.

The source reference clock for the PLL reference clock is selected by programming the SRCREF value on Page 0, Register 13, D(6:4). The PCM5252 provides several programmable clock dividers to achieve a variety of sampling rates for the DAC and clocks for the NCP, OSR and the miniDSP. OSRCK for OSR must be set at 16 f_S frequency by DOSR on Page0, Register 30, D(6:0). See Figure 61.

If PLL functionality is not required, set the PLLEN value on Page 0, Register 4, D(0) to 0. In this situation, an external SCK is required.

8.3.6.5 PLL Calculation

The PCM5252 has an on-chip PLL with fractional multiplication to generate the clock frequency needed by the audio DAC, Negative Charge Pump, Modulator and Digital Signal Processing blocks. The programmability of the PLL allows operation from a wide variety of clocks that may be available in the system. The PLL input (PLLCKIN) supports clock frequencies from 1 MHz to 50 MHz and is register programmable to enable generation of required sampling rates with fine precision.

The PLL is enabled by default. The PLL can be turned on by writing to Page 0, Register 4, D(0). When the PLL is enabled, the PLL output clock PLLCK is given by Equation 1.

Equation 1.

where

• R = 1, 2, 3,4, ... , 15, 16
• J = 4,5,6, . . . 63, and D = 0000, 0001, 0002, . . . 9999
• K = [J value].[D value]
• P = 1, 2, 3, ... 15

R, J, D, and P are programmable. J is the integer portion of K (the numbers to the left of the decimal point), while D is the fractional portion of K (the numbers to the right of the decimal point, assuming four digits of precision).

8.3.6.6 Clock Master Mode from Audio Rate Master Clock

In Master Mode, the device generates bit clock (BCK) and left-right clock (LRCK) and outputs them on the appropriate pins. To configure the device in this mode, first put the device into reset, then use registers BCKO and LRKO (Pg 0, Reg 9 0x09). Then reset the LRCK and BCK divider counters using bits RBCK and RLRK (Pg 0, Reg 12 0x0C). Finally, exit reset.

An example of this is given in register programming examples in the PCM5242 data sheet (SLASE12.)

Figure 62 shows a simplified serial port clock tree for the device in master mode.

Figure 62. Simplified Clock Tree for SCK Sourced Master Mode

In master mode, SCK is an input and BCK/LRCK are outputs. BCK and LRCK are integer divisions of SCK. Master mode with a non-audio rate master clock source will require external GPIOs to use the PLL in standalone mode.

The PLL will also need to be configured to ensure that the onchip miniDSP processor can be driven at its maximum clock rate.

Register changes that need to be done include switching the device into master mode, and setting the divider ratio.

Here is an example of using 24.576 MCLK as a master clock source and driving the BCK and LRCK with integer dividers to create 48 kHz.

In this mode, the DAC section of the device is also running from the PLL output. While the PLL inside the PCM5252 is one that has been specified to achieve the stated performance, using the SCK CMOS Oscillator source will have less jitter.

To switch the DAC clocks (SDAC in the Figure 61) the following registers should be modified.

• Clock Tree Flex Mode (Page 253, Registers 0x3F and 0x40)
• DAC and OSR Source Clock Register (Page 0, Reg 14) – set to 0x30 (SCK input, and OSR is set to whatever the DAC source is)
• The DAC clock divider should be 16 F_S.
  • 16 × 48 kHz = 768 kHz
  • 24.576 MHz (SCK in) / 768 kHz = 32
  • Therefor, divide ratio for register DDAC (Page 0, Reg 28 0x1C) should be set to 32. The may the register is mapped gives 0x00 = 1, so 32 must be converted to 0x1F.

An example configuration can be found in the PCM5242 data sheet (SLASE12).

8.3.6.7 Clock Master from a Non-Audio Rate Master Clock

The classic example here is running a 12-MHz Master clock for a 48-kHz sampling system. Given the clock tree for the device (shown in Figure 61), a non-audio clock rate cannot be brought into the SCK to the PLL in master mode. Therefore, the PLL source must be configured to be a GPIO pin, and the output brought back into another GPIO pin.

Figure 63. Application Diagram for Using Non-Audio Clock Sources to Generate Audio Clocks

The clock flow through the system is shown in Figure 63. The newly-generated SCK must be brought out of the device on a GPIO pin, then brought into the SCK pin for integer division to create BCK and LRCK outputs.

A code example for configuring this mode is provided in the PCM5242 data sheet (SLASE12).

8.4.1.1 Software Control

8.4.1.1.1 SPI Interface

The SPI interface is a 4-wire synchronous serial port which operates asynchronously to the serial audio interface and the system clock (SCK). The serial control interface is used to program and read the on-chip mode registers.

The control interface includes MISO (pin 24), MOSI (pin 11), MC (pin 12), and MS (pin 18). MISO (Master In Slave Out) is the serial data output, used to read back the values of the mode registers; MOSI (Master Out Slave In) is the serial data input, used to program the mode registers.

MC is the serial bit clock, used to shift data in and out of the control port by falling edge of MC, and MS is the mode control enable with LOW active, used to enable the internal mode register access. If feedback from the device is not required, the MISO pin can be assigned to GPIO1 by register control.

8.4.1.1.1.1 Register Read and Write Operation

All read/write operations for the serial control port use 16-bit data words. Figure 64 shows the control data word format. The most significant bit is the read/write bit. For write operations, the bit must be set to 0. For read operations, the bit must be set to 1. There are seven bits, labeled IDX[6:0], that hold the register index (or address) for the read and write operations. The least significant eight bits, D[7:0], contain the data to be written to, or the data that was read from, the register specified by IDX[6:0].

Figure 64 and Figure 65 show the functional timing diagram to write or read through the serial control port. MS is held at a logic-1 state until a register access. To start the register write or read cycle, set MS to logic 0. Sixteen clocks are then provided on MC, corresponding to the 16 bits of the control data word on MOSI and read-back data on MISO. After the eighth clock cycle has completed, the data from the indexed-mode control register appears on MISO during the read operation. After the sixteenth clock cycle has completed, the data is latched into the indexed-mode control register during the write operation. To write or read subsequent data, MS is set to logic 1 once (see t_MHH in Figure 69).

Figure 64. Control Data Word Format; MDI

NOTE

B8 is used for selection of Write or Read. Setting = 0 indicates a Write, while = 1 indicates a Read. Bits 15–9 are used for register address. Bits 7–0 are used for register data. Multiple-byte write or read (up to 8 bytes) is supported while MS is kept low. The address field becomes the initial address, automatically incrementing for each byte.

Figure 65. Serial Control Format; Write, Single Byte

Figure 66. Serial Control Format; Write, Multiple Byte

Figure 67. Serial Control Format; Read

Figure 68. Serial Control Format; Read, Multiple Byte

Figure 69. Control Interface Timing

Table 29. Control Interface Timing

		MIN	MAX	UNIT
tMCY	MC Pulse Cycle Time	100		ns
tMCL	MC Low Level Time	40		ns
tMCH	MC High Level Time	40		ns
tMHH	MS High Level Time	20		ns
tMSS	MS ↓ Edge to MC ↑ Edge	30		ns
tMSH	MS Hold Time(1)	30		ns
tMDH	MDI Hold Time	15		ns
tMDS	MDI Set-up Time	15		ns
tMOS	MC Rise Edge to MDO Stable		20	ns

(1) MC falling edge for LSB to MS rising edge.

8.4.1.1.2 I²C Interface

The PCM5252 supports the I²C serial bus and the data transmission protocol for standard and fast mode as a slave device.

In I²C mode, the control terminals are changed as follows.

Table 30. I²C Pins and Functions

SIGNAL	PIN	I/O	DESCRIPTION
SDA	11	I/O	I2C data
SCL	12	I	I2C clock
ADR2	16	I	I2C address 2
ADR1	24	I	I2C address 1

8.4.1.1.2.1 Slave Address

Table 31. I²C Slave Address

MSB							LSB
1	0	0	1	1	ADR2	ADR1	R/ W

The PCM5252 has 7 bits for its own slave address. The first five bits (MSBs) of the slave address are factory preset to 10011 (0x9x). The next two bits of the address byte are the device select bits which can be user-defined by the ADR1 and ADR0 terminals. A maximum of four devices can be connected on the same bus at one time. This gives a range of 0x98, 0x9A, 0x9C and 0x9E. Each PCM5252 responds when it receives its own slave address.

8.4.1.1.2.2 Register Address Auto-Increment Mode

Figure 70. Auto Increment Mode

Auto-increment mode allows multiple sequential register locations to be written to or read back in a single operation, and is especially useful for block write and read operations.

8.4.1.1.2.3 Packet Protocol

A master device must control packet protocol, which consists of start condition, slave address, read/write bit, data if write or acknowledge if read, and stop condition. The PCM5252 supports only slave receivers and slave transmitters.

Figure 71. Packet Protocol

Table 32. Write Operation - Basic I²C Framework

Transmitter	M	M	M	S	M	S	M	S		S	M
Data Type	St	slave address	R/	ACK	DATA	ACK	DATA	ACK		ACK	Sp

Table 33. Read Operation - Basic I²C Framework

Transmitter	M	M	M	S	S	M	S	M		M	M
Data Type	St	slave address	R/	ACK	DATA	ACK	DATA	ACK		NACK	Sp

M = Master Device; S = Slave Device; St = Start Condition; Sp = Stop Condition

8.4.1.1.2.4 Write Register

A master can write to any PCM5252 registers using single or multiple accesses. The master sends a PCM5252 slave address with a write bit, a register address with auto-increment bit, and the data. If auto-increment is enabled, the address is that of the starting register, followed by the data to be transferred. When the data is received properly, the index register is incremented by 1 automatically. When the index register reaches 0x7F, the next value is 0x0. Table 34 shows the write operation.

Table 34. Write Operation

Transmitter

M

M

M

S

M

S

M

S

M

S

S

M

Data Type

St

slave addr

W

ACK

inc

reg addr

ACK

write data 1

ACK

write data 2

ACK

ACK

Sp

M = Master Device; S = Slave Device; St = Start Condition; Sp = Stop Condition; W = Write; ACK = Acknowledge

8.4.1.1.2.5 Read Register

A master can read the PCM5252 register. The value of the register address is stored in an indirect index register in advance. The master sends a PCM5252 slave address with a read bit after storing the register address. Then the PCM5252 transfers the data which the index register points to. When auto-increment is enabled, the index register is incremented by 1 automatically. When the index register reaches 0x7F, the next value is 0x0. Table 35 shows the read operation.

Table 35. Read Operation

Transmitter

M

M

M

S

M

S

M

M

M

S

S

M

M

M

Data Type

St

slave addr

W

ACK

inc

reg addr

ACK

Sr

slave addr

R

ACK

data

ACK

NACK

Sp

M = Master Device; S = Slave Device; St = Start Condition; Sr = Repeated Start Condition; Sp = Stop Condition; W = Write; R = Read; NACK = Not acknowledge

8.4.1.1.2.6 Timing Characteristics

Figure 72. Register Access Timing

Table 36. I²C Bus Timing

			MIN	MAX	UNIT
fSCL	SCL clock frequency	Standard		100	kHz
		Fast		400	kHz
tBUF	Bus free time between a STOP and START condition	Standard	4.7		µs
		Fast	1.3
tLOW	Low period of the SCL clock	Standard	4.7		µs
		Fast	1.3
tHI	High period of the SCL clock	Standard	4.0		µs
		Fast	600		ns
tRS-SU	Setup time for (repeated)START condition	Standard	4.7		µs
		Fast	600		ns
tS-HD	Hold time for (repeated)START condition	Standard	4.0		µs
tRS-HD		Fast	600		ns
tD-SU	Data setup time	Standard	250		ns
		Fast	100
tD-HD	Data hold time	Standard	0	900	ns
		Fast	0	900
tSCL-R	Rise time of SCL signal	Standard	20 + 0.1CB	1000	ns
		Fast	20 + 0.1CB	300
tSCL-R1	Rise time of SCL signal after a repeated START condition and after an acknowledge bit	Standard	20 + 0.1CB	1000	ns
		Fast	20 + 0.1CB	300
tSCL-F	Fall time of SCL signal	Standard	20 + 0.1CB	1000	ns
		Fast	20 + 0.1CB	300
tSDA-R	Rise time of SDA signal	Standard	20 + 0.1CB	1000	ns
		Fast	20 + 0.1CB	300
tSDA-F	Fall time of SDA signal	Standard	20 + 0.1CB	1000	ns
		Fast	20 + 0.1CB	300
tP-SU	Setup time for STOP condition	Standard	4.0		µs
		Fast	600		ns
CB	Capacitive load for SDA and SCL line			400	pF
tSP	Pulse width of spike suppressed	Fast		50	ns
VNH	Noise margin at High level for each connected device (including hysteresis)		0.2 × VDD		V