/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "control_sgtl5000.h"
#include "Wire.h"
#define CHIP_ID 0x0000
// 15:8 PARTID 0xA0 - 8 bit identifier for SGTL5000
// 7:0 REVID 0x00 - revision number for SGTL5000.
#define CHIP_DIG_POWER 0x0002
// 6 ADC_POWERUP 1=Enable, 0=disable the ADC block, both digital & analog,
// 5 DAC_POWERUP 1=Enable, 0=disable the DAC block, both analog and digital
// 4 DAP_POWERUP 1=Enable, 0=disable the DAP block
// 1 I2S_OUT_POWERUP 1=Enable, 0=disable the I2S data output
// 0 I2S_IN_POWERUP 1=Enable, 0=disable the I2S data input
#define CHIP_CLK_CTRL 0x0004
// 5:4 RATE_MODE Sets the sample rate mode. MCLK_FREQ is still specified
// relative to the rate in SYS_FS
// 0x0 = SYS_FS specifies the rate
// 0x1 = Rate is 1/2 of the SYS_FS rate
// 0x2 = Rate is 1/4 of the SYS_FS rate
// 0x3 = Rate is 1/6 of the SYS_FS rate
// 3:2 SYS_FS Sets the internal system sample rate (default=2)
// 0x0 = 32 kHz
// 0x1 = 44.1 kHz
// 0x2 = 48 kHz
// 0x3 = 96 kHz
// 1:0 MCLK_FREQ Identifies incoming SYS_MCLK frequency and if the PLL should be used
// 0x0 = 256*Fs
// 0x1 = 384*Fs
// 0x2 = 512*Fs
// 0x3 = Use PLL
// The 0x3 (Use PLL) setting must be used if the SYS_MCLK is not
// a standard multiple of Fs (256, 384, or 512). This setting can
// also be used if SYS_MCLK is a standard multiple of Fs.
// Before this field is set to 0x3 (Use PLL), the PLL must be
// powered up by setting CHIP_ANA_POWER->PLL_POWERUP and
// CHIP_ANA_POWER->VCOAMP_POWERUP. Also, the PLL dividers must
// be calculated based on the external MCLK rate and
// CHIP_PLL_CTRL register must be set (see CHIP_PLL_CTRL register
// description details on how to calculate the divisors).
#define CHIP_I2S_CTRL 0x0006
// 8 SCLKFREQ Sets frequency of I2S_SCLK when in master mode (MS=1). When in slave
// mode (MS=0), this field must be set appropriately to match SCLK input
// rate.
// 0x0 = 64Fs
// 0x1 = 32Fs - Not supported for RJ mode (I2S_MODE = 1)
// 7 MS Configures master or slave of I2S_LRCLK and I2S_SCLK.
// 0x0 = Slave: I2S_LRCLK an I2S_SCLK are inputs
// 0x1 = Master: I2S_LRCLK and I2S_SCLK are outputs
// NOTE: If the PLL is used (CHIP_CLK_CTRL->MCLK_FREQ==0x3),
// the SGTL5000 must be a master of the I2S port (MS==1)
// 6 SCLK_INV Sets the edge that data (input and output) is clocked in on for I2S_SCLK
// 0x0 = data is valid on rising edge of I2S_SCLK
// 0x1 = data is valid on falling edge of I2S_SCLK
// 5:4 DLEN I2S data length (default=1)
// 0x0 = 32 bits (only valid when SCLKFREQ=0),
// not valid for Right Justified Mode
// 0x1 = 24 bits (only valid when SCLKFREQ=0)
// 0x2 = 20 bits
// 0x3 = 16 bits
// 3:2 I2S_MODE Sets the mode for the I2S port
// 0x0 = I2S mode or Left Justified (Use LRALIGN to select)
// 0x1 = Right Justified Mode
// 0x2 = PCM Format A/B
// 0x3 = RESERVED
// 1 LRALIGN I2S_LRCLK Alignment to data word. Not used for Right Justified mode
// 0x0 = Data word starts 1 I2S_SCLK delay after I2S_LRCLK
// transition (I2S format, PCM format A)
// 0x1 = Data word starts after I2S_LRCLK transition (left
// justified format, PCM format B)
// 0 LRPOL I2S_LRCLK Polarity when data is presented.
// 0x0 = I2S_LRCLK = 0 - Left, 1 - Right
// 1x0 = I2S_LRCLK = 0 - Right, 1 - Left
// The left subframe should be presented first regardless of
// the setting of LRPOL.
#define CHIP_SSS_CTRL 0x000A
// 14 DAP_MIX_LRSWAP DAP Mixer Input Swap
// 0x0 = Normal Operation
// 0x1 = Left and Right channels for the DAP MIXER Input are swapped.
// 13 DAP_LRSWAP DAP Mixer Input Swap
// 0x0 = Normal Operation
// 0x1 = Left and Right channels for the DAP Input are swapped
// 12 DAC_LRSWAP DAC Input Swap
// 0x0 = Normal Operation
// 0x1 = Left and Right channels for the DAC are swapped
// 10 I2S_LRSWAP I2S_DOUT Swap
// 0x0 = Normal Operation
// 0x1 = Left and Right channels for the I2S_DOUT are swapped
// 9:8 DAP_MIX_SELECT Select data source for DAP mixer
// 0x0 = ADC
// 0x1 = I2S_IN
// 0x2 = Reserved
// 0x3 = Reserved
// 7:6 DAP_SELECT Select data source for DAP
// 0x0 = ADC
// 0x1 = I2S_IN
// 0x2 = Reserved
// 0x3 = Reserved
// 5:4 DAC_SELECT Select data source for DAC (default=1)
// 0x0 = ADC
// 0x1 = I2S_IN
// 0x2 = Reserved
// 0x3 = DAP
// 1:0 I2S_SELECT Select data source for I2S_DOUT
// 0x0 = ADC
// 0x1 = I2S_IN
// 0x2 = Reserved
// 0x3 = DAP
#define CHIP_ADCDAC_CTRL 0x000E
// 13 VOL_BUSY_DAC_RIGHT Volume Busy DAC Right
// 0x0 = Ready
// 0x1 = Busy - This indicates the channel has not reached its
// programmed volume/mute level
// 12 VOL_BUSY_DAC_LEFT Volume Busy DAC Left
// 0x0 = Ready
// 0x1 = Busy - This indicates the channel has not reached its
// programmed volume/mute level
// 9 VOL_RAMP_EN Volume Ramp Enable (default=1)
// 0x0 = Disables volume ramp. New volume settings take immediate
// effect without a ramp
// 0x1 = Enables volume ramp
// This field affects DAC_VOL. The volume ramp effects both
// volume settings and mute When set to 1 a soft mute is enabled.
// 8 VOL_EXPO_RAMP Exponential Volume Ramp Enable
// 0x0 = Linear ramp over top 4 volume octaves
// 0x1 = Exponential ramp over full volume range
// This bit only takes effect if VOL_RAMP_EN is 1.
// 3 DAC_MUTE_RIGHT DAC Right Mute (default=1)
// 0x0 = Unmute
// 0x1 = Muted
// If VOL_RAMP_EN = 1, this is a soft mute.
// 2 DAC_MUTE_LEFT DAC Left Mute (default=1)
// 0x0 = Unmute
// 0x1 = Muted
// If VOL_RAMP_EN = 1, this is a soft mute.
// 1 ADC_HPF_FREEZE ADC High Pass Filter Freeze
// 0x0 = Normal operation
// 0x1 = Freeze the ADC high-pass filter offset register. The
// offset continues to be subtracted from the ADC data stream.
// 0 ADC_HPF_BYPASS ADC High Pass Filter Bypass
// 0x0 = Normal operation
// 0x1 = Bypassed and offset not updated
#define CHIP_DAC_VOL 0x0010
// 15:8 DAC_VOL_RIGHT DAC Right Channel Volume. Set the Right channel DAC volume
// with 0.5017 dB steps from 0 to -90 dB
// 0x3B and less = Reserved
// 0x3C = 0 dB
// 0x3D = -0.5 dB
// 0xF0 = -90 dB
// 0xFC and greater = Muted
// If VOL_RAMP_EN = 1, there is an automatic ramp to the
// new volume setting.
// 7:0 DAC_VOL_LEFT DAC Left Channel Volume. Set the Left channel DAC volume
// with 0.5017 dB steps from 0 to -90 dB
// 0x3B and less = Reserved
// 0x3C = 0 dB
// 0x3D = -0.5 dB
// 0xF0 = -90 dB
// 0xFC and greater = Muted
// If VOL_RAMP_EN = 1, there is an automatic ramp to the
// new volume setting.
#define CHIP_PAD_STRENGTH 0x0014
// 9:8 I2S_LRCLK I2S LRCLK Pad Drive Strength (default=1)
// Sets drive strength for output pads per the table below.
// VDDIO 1.8 V 2.5 V 3.3 V
// 0x0 = Disable
// 0x1 = 1.66 mA 2.87 mA 4.02 mA
// 0x2 = 3.33 mA 5.74 mA 8.03 mA
// 0x3 = 4.99 mA 8.61 mA 12.05 mA
// 7:6 I2S_SCLK I2S SCLK Pad Drive Strength (default=1)
// 5:4 I2S_DOUT I2S DOUT Pad Drive Strength (default=1)
// 3:2 CTRL_DATA I2C DATA Pad Drive Strength (default=3)
// 1:0 CTRL_CLK I2C CLK Pad Drive Strength (default=3)
// (all use same table as I2S_LRCLK)
#define CHIP_ANA_ADC_CTRL 0x0020
// 8 ADC_VOL_M6DB ADC Volume Range Reduction
// This bit shifts both right and left analog ADC volume
// range down by 6.0 dB.
// 0x0 = No change in ADC range
// 0x1 = ADC range reduced by 6.0 dB
// 7:4 ADC_VOL_RIGHT ADC Right Channel Volume
// Right channel analog ADC volume control in 1.5 dB steps.
// 0x0 = 0 dB
// 0x1 = +1.5 dB
// ...
// 0xF = +22.5 dB
// This range is -6.0 dB to +16.5 dB if ADC_VOL_M6DB is set to 1.
// 3:0 ADC_VOL_LEFT ADC Left Channel Volume
// (same scale as ADC_VOL_RIGHT)
#define CHIP_ANA_HP_CTRL 0x0022
// 14:8 HP_VOL_RIGHT Headphone Right Channel Volume (default 0x18)
// Right channel headphone volume control with 0.5 dB steps.
// 0x00 = +12 dB
// 0x01 = +11.5 dB
// 0x18 = 0 dB
// ...
// 0x7F = -51.5 dB
// 6:0 HP_VOL_LEFT Headphone Left Channel Volume (default 0x18)
// (same scale as HP_VOL_RIGHT)
#define CHIP_ANA_CTRL 0x0024
// 8 MUTE_LO LINEOUT Mute, 0 = Unmute, 1 = Mute (default 1)
// 6 SELECT_HP Select the headphone input, 0 = DAC, 1 = LINEIN
// 5 EN_ZCD_HP Enable the headphone zero cross detector (ZCD)
// 0x0 = HP ZCD disabled
// 0x1 = HP ZCD enabled
// 4 MUTE_HP Mute the headphone outputs, 0 = Unmute, 1 = Mute (default)
// 2 SELECT_ADC Select the ADC input, 0 = Microphone, 1 = LINEIN
// 1 EN_ZCD_ADC Enable the ADC analog zero cross detector (ZCD)
// 0x0 = ADC ZCD disabled
// 0x1 = ADC ZCD enabled
// 0 MUTE_ADC Mute the ADC analog volume, 0 = Unmute, 1 = Mute (default)
#define CHIP_LINREG_CTRL 0x0026
// 6 VDDC_MAN_ASSN Determines chargepump source when VDDC_ASSN_OVRD is set.
// 0x0 = VDDA
// 0x1 = VDDIO
// 5 VDDC_ASSN_OVRD Charge pump Source Assignment Override
// 0x0 = Charge pump source is automatically assigned based
// on higher of VDDA and VDDIO
// 0x1 = the source of charge pump is manually assigned by
// VDDC_MAN_ASSN If VDDIO and VDDA are both the same
// and greater than 3.1 V, VDDC_ASSN_OVRD and
// VDDC_MAN_ASSN should be used to manually assign
// VDDIO as the source for charge pump.
// 3:0 D_PROGRAMMING Sets the VDDD linear regulator output voltage in 50 mV steps.
// Must clear the LINREG_SIMPLE_POWERUP and STARTUP_POWERUP bits
// in the 0x0030 (CHIP_ANA_POWER) register after power-up, for
// this setting to produce the proper VDDD voltage.
// 0x0 = 1.60
// 0xF = 0.85
#define CHIP_REF_CTRL 0x0028 // bandgap reference bias voltage and currents
// 8:4 VAG_VAL Analog Ground Voltage Control
// These bits control the analog ground voltage in 25 mV steps.
// This should usually be set to VDDA/2 or lower for best
// performance (maximum output swing at minimum THD). This VAG
// reference is also used for the DAC and ADC voltage reference.
// So changing this voltage scales the output swing of the DAC
// and the output signal of the ADC.
// 0x00 = 0.800 V
// 0x1F = 1.575 V
// 3:1 BIAS_CTRL Bias control
// These bits adjust the bias currents for all of the analog
// blocks. By lowering the bias current a lower quiescent power
// is achieved. It should be noted that this mode can affect
// performance by 3-4 dB.
// 0x0 = Nominal
// 0x1-0x3=+12.5%
// 0x4=-12.5%
// 0x5=-25%
// 0x6=-37.5%
// 0x7=-50%
// 0 SMALL_POP VAG Ramp Control
// Setting this bit slows down the VAG ramp from ~200 to ~400 ms
// to reduce the startup pop, but increases the turn on/off time.
// 0x0 = Normal VAG ramp
// 0x1 = Slow down VAG ramp
#define CHIP_MIC_CTRL 0x002A // microphone gain & internal microphone bias
// 9:8 BIAS_RESISTOR MIC Bias Output Impedance Adjustment
// Controls an adjustable output impedance for the microphone bias.
// If this is set to zero the micbias block is powered off and
// the output is highZ.
// 0x0 = Powered off
// 0x1 = 2.0 kohm
// 0x2 = 4.0 kohm
// 0x3 = 8.0 kohm
// 6:4 BIAS_VOLT MIC Bias Voltage Adjustment
// Controls an adjustable bias voltage for the microphone bias
// amp in 250 mV steps. This bias voltage setting should be no
// more than VDDA-200 mV for adequate power supply rejection.
// 0x0 = 1.25 V
// ...
// 0x7 = 3.00 V
// 1:0 GAIN MIC Amplifier Gain
// Sets the microphone amplifier gain. At 0 dB setting the THD
// can be slightly higher than other paths- typically around
// ~65 dB. At other gain settings the THD are better.
// 0x0 = 0 dB
// 0x1 = +20 dB
// 0x2 = +30 dB
// 0x3 = +40 dB
#define CHIP_LINE_OUT_CTRL 0x002C
// 11:8 OUT_CURRENT Controls the output bias current for the LINEOUT amplifiers. The
// nominal recommended setting for a 10 kohm load with 1.0 nF load cap
// is 0x3. There are only 5 valid settings.
// 0x0=0.18 mA
// 0x1=0.27 mA
// 0x3=0.36 mA
// 0x7=0.45 mA
// 0xF=0.54 mA
// 5:0 LO_VAGCNTRL LINEOUT Amplifier Analog Ground Voltage
// Controls the analog ground voltage for the LINEOUT amplifiers
// in 25 mV steps. This should usually be set to VDDIO/2.
// 0x00 = 0.800 V
// ...
// 0x1F = 1.575 V
// ...
// 0x23 = 1.675 V
// 0x24-0x3F are invalid
#define CHIP_LINE_OUT_VOL 0x002E
// 12:8 LO_VOL_RIGHT LINEOUT Right Channel Volume (default=4)
// Controls the right channel LINEOUT volume in 0.5 dB steps.
// Higher codes have more attenuation.
// 4:0 LO_VOL_LEFT LINEOUT Left Channel Output Level (default=4)
// Used to normalize the output level of the left line output
// to full scale based on the values used to set
// LINE_OUT_CTRL->LO_VAGCNTRL and CHIP_REF_CTRL->VAG_VAL.
// In general this field should be set to:
// 40*log((VAG_VAL)/(LO_VAGCNTRL)) + 15
// Suggested values based on typical VDDIO and VDDA voltages.
// VDDA VAG_VAL VDDIO LO_VAGCNTRL LO_VOL_*
// 1.8 V 0.9 3.3 V 1.55 0x06
// 1.8 V 0.9 1.8 V 0.9 0x0F
// 3.3 V 1.55 1.8 V 0.9 0x19
// 3.3 V 1.55 3.3 V 1.55 0x0F
// After setting to the nominal voltage, this field can be used
// to adjust the output level in +/-0.5 dB increments by using
// values higher or lower than the nominal setting.
#define CHIP_ANA_POWER 0x0030 // power down controls for the analog blocks.
// The only other power-down controls are BIAS_RESISTOR in the MIC_CTRL register
// and the EN_ZCD control bits in ANA_CTRL.
// 14 DAC_MONO While DAC_POWERUP is set, this allows the DAC to be put into left only
// mono operation for power savings. 0=mono, 1=stereo (default)
// 13 LINREG_SIMPLE_POWERUP Power up the simple (low power) digital supply regulator.
// After reset, this bit can be cleared IF VDDD is driven
// externally OR the primary digital linreg is enabled with
// LINREG_D_POWERUP
// 12 STARTUP_POWERUP Power up the circuitry needed during the power up ramp and reset.
// After reset this bit can be cleared if VDDD is coming from
// an external source.
// 11 VDDC_CHRGPMP_POWERUP Power up the VDDC charge pump block. If neither VDDA or VDDIO
// is 3.0 V or larger this bit should be cleared before analog
// blocks are powered up.
// 10 PLL_POWERUP PLL Power Up, 0 = Power down, 1 = Power up
// When cleared, the PLL is turned off. This must be set before
// CHIP_CLK_CTRL->MCLK_FREQ is programmed to 0x3. The
// CHIP_PLL_CTRL register must be configured correctly before
// setting this bit.
// 9 LINREG_D_POWERUP Power up the primary VDDD linear regulator, 0 = Power down, 1 = Power up
// 8 VCOAMP_POWERUP Power up the PLL VCO amplifier, 0 = Power down, 1 = Power up
// 7 VAG_POWERUP Power up the VAG reference buffer.
// Setting this bit starts the power up ramp for the headphone
// and LINEOUT. The headphone (and/or LINEOUT) powerup should
// be set BEFORE clearing this bit. When this bit is cleared
// the power-down ramp is started. The headphone (and/or LINEOUT)
// powerup should stay set until the VAG is fully ramped down
// (200 to 400 ms after clearing this bit).
// 0x0 = Power down, 0x1 = Power up
// 6 ADC_MONO While ADC_POWERUP is set, this allows the ADC to be put into left only
// mono operation for power savings. This mode is useful when
// only using the microphone input.
// 0x0 = Mono (left only), 0x1 = Stereo
// 5 REFTOP_POWERUP Power up the reference bias currents
// 0x0 = Power down, 0x1 = Power up
// This bit can be cleared when the part is a sleep state
// to minimize analog power.
// 4 HEADPHONE_POWERUP Power up the headphone amplifiers
// 0x0 = Power down, 0x1 = Power up
// 3 DAC_POWERUP Power up the DACs
// 0x0 = Power down, 0x1 = Power up
// 2 CAPLESS_HEADPHONE_POWERUP Power up the capless headphone mode
// 0x0 = Power down, 0x1 = Power up
// 1 ADC_POWERUP Power up the ADCs
// 0x0 = Power down, 0x1 = Power up
// 0 LINEOUT_POWERUP Power up the LINEOUT amplifiers
// 0x0 = Power down, 0x1 = Power up
#define CHIP_PLL_CTRL 0x0032
// 15:11 INT_DIVISOR
// 10:0 FRAC_DIVISOR
#define CHIP_CLK_TOP_CTRL 0x0034
// 11 ENABLE_INT_OSC Setting this bit enables an internal oscillator to be used for the
// zero cross detectors, the short detect recovery, and the
// charge pump. This allows the I2S clock to be shut off while
// still operating an analog signal path. This bit can be kept
// on when the I2S clock is enabled, but the I2S clock is more
// accurate so it is preferred to clear this bit when I2S is present.
// 3 INPUT_FREQ_DIV2 SYS_MCLK divider before PLL input
// 0x0 = pass through
// 0x1 = SYS_MCLK is divided by 2 before entering PLL
// This must be set when the input clock is above 17 Mhz. This
// has no effect when the PLL is powered down.
#define CHIP_ANA_STATUS 0x0036
// 9 LRSHORT_STS This bit is high whenever a short is detected on the left or right
// channel headphone drivers.
// 8 CSHORT_STS This bit is high whenever a short is detected on the capless headphone
// common/center channel driver.
// 4 PLL_IS_LOCKED This bit goes high after the PLL is locked.
#define CHIP_ANA_TEST1 0x0038 // intended only for debug.
#define CHIP_ANA_TEST2 0x003A // intended only for debug.
#define CHIP_SHORT_CTRL 0x003C
// 14:12 LVLADJR Right channel headphone short detector in 25 mA steps.
// 0x3=25 mA
// 0x2=50 mA
// 0x1=75 mA
// 0x0=100 mA
// 0x4=125 mA
// 0x5=150 mA
// 0x6=175 mA
// 0x7=200 mA
// This trip point can vary by ~30% over process so leave plenty
// of guard band to avoid false trips. This short detect trip
// point is also effected by the bias current adjustments made
// by CHIP_REF_CTRL->BIAS_CTRL and by CHIP_ANA_TEST1->HP_IALL_ADJ.
// 10:8 LVLADJL Left channel headphone short detector in 25 mA steps.
// (same scale as LVLADJR)
// 6:4 LVLADJC Capless headphone center channel short detector in 50 mA steps.
// 0x3=50 mA
// 0x2=100 mA
// 0x1=150 mA
// 0x0=200 mA
// 0x4=250 mA
// 0x5=300 mA
// 0x6=350 mA
// 0x7=400 mA
// 3:2 MODE_LR Behavior of left/right short detection
// 0x0 = Disable short detector, reset short detect latch,
// software view non-latched short signal
// 0x1 = Enable short detector and reset the latch at timeout
// (every ~50 ms)
// 0x2 = This mode is not used/invalid
// 0x3 = Enable short detector with only manual reset (have
// to return to 0x0 to reset the latch)
// 1:0 MODE_CM Behavior of capless headphone central short detection
// (same settings as MODE_LR)
#define DAP_CONTROL 0x0100
#define DAP_PEQ 0x0102
#define DAP_BASS_ENHANCE 0x0104
#define DAP_BASS_ENHANCE_CTRL 0x0106
#define DAP_AUDIO_EQ 0x0108
#define DAP_SGTL_SURROUND 0x010A
#define DAP_FILTER_COEF_ACCESS 0x010C
#define DAP_COEF_WR_B0_MSB 0x010E
#define DAP_COEF_WR_B0_LSB 0x0110
#define DAP_AUDIO_EQ_BASS_BAND0 0x0116 // 115 Hz
#define DAP_AUDIO_EQ_BAND1 0x0118 // 330 Hz
#define DAP_AUDIO_EQ_BAND2 0x011A // 990 Hz
#define DAP_AUDIO_EQ_BAND3 0x011C // 3000 Hz
#define DAP_AUDIO_EQ_TREBLE_BAND4 0x011E // 9900 Hz
#define DAP_MAIN_CHAN 0x0120
#define DAP_MIX_CHAN 0x0122
#define DAP_AVC_CTRL 0x0124
#define DAP_AVC_THRESHOLD 0x0126
#define DAP_AVC_ATTACK 0x0128
#define DAP_AVC_DECAY 0x012A
#define DAP_COEF_WR_B1_MSB 0x012C
#define DAP_COEF_WR_B1_LSB 0x012E
#define DAP_COEF_WR_B2_MSB 0x0130
#define DAP_COEF_WR_B2_LSB 0x0132
#define DAP_COEF_WR_A1_MSB 0x0134
#define DAP_COEF_WR_A1_LSB 0x0136
#define DAP_COEF_WR_A2_MSB 0x0138
#define DAP_COEF_WR_A2_LSB 0x013A
#define SGTL5000_I2C_ADDR 0x0A // CTRL_ADR0_CS pin low (normal configuration)
//#define SGTL5000_I2C_ADDR 0x2A // CTRL_ADR0_CS pin high
bool AudioControlSGTL5000::enable(void)
{
muted = true;
Wire.begin();
delay(5);
//Serial.print("chip ID = ");
//delay(5);
//unsigned int n = read(CHIP_ID);
//Serial.println(n, HEX);
write(CHIP_ANA_POWER, 0x4060); // VDDD is externally driven with 1.8V
write(CHIP_LINREG_CTRL, 0x006C); // VDDA & VDDIO both over 3.1V
write(CHIP_REF_CTRL, 0x01F2); // VAG=1.575, normal ramp, +12.5% bias current
write(CHIP_LINE_OUT_CTRL, 0x0F22); // LO_VAGCNTRL=1.65V, OUT_CURRENT=0.54mA
write(CHIP_SHORT_CTRL, 0x4446); // allow up to 125mA
write(CHIP_ANA_CTRL, 0x0137); // enable zero cross detectors
write(CHIP_ANA_POWER, 0x40FF); // power up: lineout, hp, adc, dac
write(CHIP_DIG_POWER, 0x0073); // power up all digital stuff
delay(400);
write(CHIP_LINE_OUT_VOL, 0x1D1D); // default approx 1.3 volts peak-to-peak
write(CHIP_CLK_CTRL, 0x0004); // 44.1 kHz, 256*Fs
write(CHIP_I2S_CTRL, 0x0130); // SCLK=32*Fs, 16bit, I2S format
// default signal routing is ok?
write(CHIP_SSS_CTRL, 0x0010); // ADC->I2S, I2S->DAC
write(CHIP_ADCDAC_CTRL, 0x0000); // disable dac mute
write(CHIP_DAC_VOL, 0x3C3C); // digital gain, 0dB
write(CHIP_ANA_HP_CTRL, 0x7F7F); // set volume (lowest level)
write(CHIP_ANA_CTRL, 0x0036); // enable zero cross detectors
//mute = false;
semi_automated = true;
return true;
}
unsigned int AudioControlSGTL5000::read(unsigned int reg)
{
unsigned int val;
Wire.beginTransmission(SGTL5000_I2C_ADDR);
Wire.write(reg >> 8);
Wire.write(reg);
if (Wire.endTransmission(false) != 0) return 0;
if (Wire.requestFrom(SGTL5000_I2C_ADDR, 2) < 2) return 0;
val = Wire.read() << 8;
val |= Wire.read();
return val;
}
bool AudioControlSGTL5000::write(unsigned int reg, unsigned int val)
{
if (reg == CHIP_ANA_CTRL) ana_ctrl = val;
Wire.beginTransmission(SGTL5000_I2C_ADDR);
Wire.write(reg >> 8);
Wire.write(reg);
Wire.write(val >> 8);
Wire.write(val);
if (Wire.endTransmission() == 0) return true;
return false;
}
unsigned int AudioControlSGTL5000::modify(unsigned int reg, unsigned int val, unsigned int iMask)
{
unsigned int val1 = (read(reg)&(~iMask))|val;
if(!write(reg,val1)) return 0;
return val1;
}
bool AudioControlSGTL5000::volumeInteger(unsigned int n)
{
if (n == 0) {
muted = true;
write(CHIP_ANA_HP_CTRL, 0x7F7F);
return muteHeadphone();
} else if (n > 0x80) {
n = 0;
} else {
n = 0x80 - n;
}
if (muted) {
muted = false;
unmuteHeadphone();
}
n = n | (n << 8);
return write(CHIP_ANA_HP_CTRL, n); // set volume
}
bool AudioControlSGTL5000::volume(float left, float right)
{
unsigned short m=((0x7F-calcVol(right,0x7F))<<8)|(0x7F-calcVol(left,0x7F));
return write(CHIP_ANA_HP_CTRL, m);
}
bool AudioControlSGTL5000::micGain(unsigned int dB)
{
unsigned int preamp_gain, input_gain;
if (dB >= 40) {
preamp_gain = 3;
dB -= 40;
} else if (dB >= 30) {
preamp_gain = 2;
dB -= 30;
} else if (dB >= 20) {
preamp_gain = 1;
dB -= 20;
} else {
preamp_gain = 0;
}
input_gain = (dB * 2) / 3;
if (input_gain > 15) input_gain = 15;
return write(CHIP_MIC_CTRL, 0x0170 | preamp_gain)
&& write(CHIP_ANA_ADC_CTRL, (input_gain << 4) | input_gain);
}
// CHIP_ANA_ADC_CTRL
// Actual measured full-scale peak-to-peak sine wave input for max signal
// 0: 3.12 Volts p-p
// 1: 2.63 Volts p-p
// 2: 2.22 Volts p-p
// 3: 1.87 Volts p-p
// 4: 1.58 Volts p-p
// 5: 1.33 Volts p-p
// 6: 1.11 Volts p-p
// 7: 0.94 Volts p-p
// 8: 0.79 Volts p-p
// 9: 0.67 Volts p-p
// 10: 0.56 Volts p-p
// 11: 0.48 Volts p-p
// 12: 0.40 Volts p-p
// 13: 0.34 Volts p-p
// 14: 0.29 Volts p-p
// 15: 0.24 Volts p-p
bool AudioControlSGTL5000::lineInLevel(uint8_t left, uint8_t right)
{
if (left > 15) left = 15;
if (right > 15) right = 15;
return write(CHIP_ANA_ADC_CTRL, (left << 4) | right);
}
// CHIP_LINE_OUT_VOL
// Actual measured full-scale peak-to-peak sine wave output voltage:
// 0-12: output has clipping
// 13: 3.16 Volts p-p
// 14: 2.98 Volts p-p
// 15: 2.83 Volts p-p
// 16: 2.67 Volts p-p
// 17: 2.53 Volts p-p
// 18: 2.39 Volts p-p
// 19: 2.26 Volts p-p
// 20: 2.14 Volts p-p
// 21: 2.02 Volts p-p
// 22: 1.91 Volts p-p
// 23: 1.80 Volts p-p
// 24: 1.71 Volts p-p
// 25: 1.62 Volts p-p
// 26: 1.53 Volts p-p
// 27: 1.44 Volts p-p
// 28: 1.37 Volts p-p
// 29: 1.29 Volts p-p
// 30: 1.22 Volts p-p
// 31: 1.16 Volts p-p
unsigned short AudioControlSGTL5000::lineOutLevel(uint8_t n)
{
if (n > 31) n = 31;
else if (n < 13) n = 13;
return modify(CHIP_LINE_OUT_VOL,(n<<8)|n,(31<<8)|31);
}
unsigned short AudioControlSGTL5000::lineOutLevel(uint8_t left, uint8_t right)
{
if (left > 31) left = 31;
else if (left < 13) left = 13;
if (right > 31) right = 31;
else if (right < 13) right = 13;
return modify(CHIP_LINE_OUT_VOL,(right<<8)|left,(31<<8)|31);
}
unsigned short AudioControlSGTL5000::dacVolume(float n) // set both directly
{
if ((read(CHIP_ADCDAC_CTRL)&(3<<2)) != ((n>0 ? 0:3)<<2)) {
modify(CHIP_ADCDAC_CTRL,(n>0 ? 0:3)<<2,3<<2);
}
unsigned char m=calcVol(n,0xC0);
return modify(CHIP_DAC_VOL,((0xFC-m)<<8)|(0xFC-m),65535);
}
unsigned short AudioControlSGTL5000::dacVolume(float left, float right)
{
unsigned short adcdac=((right>0 ? 0:2)|(left>0 ? 0:1))<<2;
if ((read(CHIP_ADCDAC_CTRL)&(3<<2)) != adcdac) {
modify(CHIP_ADCDAC_CTRL,adcdac,1<<2);
}
unsigned short m=(0xFC-calcVol(right,0xC0))<<8|(0xFC-calcVol(left,0xC0));
return modify(CHIP_DAC_VOL,m,65535);
}
unsigned short AudioControlSGTL5000::adcHighPassFilterEnable(void)
{
return modify(CHIP_ADCDAC_CTRL, 0, 3);
}
unsigned short AudioControlSGTL5000::adcHighPassFilterFreeze(void)
{
return modify(CHIP_ADCDAC_CTRL, 2, 3);
}
unsigned short AudioControlSGTL5000::adcHighPassFilterDisable(void)
{
return modify(CHIP_ADCDAC_CTRL, 1, 3);
}
// DAP_CONTROL
unsigned short AudioControlSGTL5000::audioPreProcessorEnable(void)
{
// audio processor used to pre-process analog input before Teensy
return write(DAP_CONTROL, 1) && write(CHIP_SSS_CTRL, 0x0013);
}
unsigned short AudioControlSGTL5000::audioPostProcessorEnable(void)
{
// audio processor used to post-process Teensy output before headphones/lineout
return write(DAP_CONTROL, 1) && write(CHIP_SSS_CTRL, 0x0070);
}
unsigned short AudioControlSGTL5000::audioProcessorDisable(void)
{
return write(CHIP_SSS_CTRL, 0x0010) && write(DAP_CONTROL, 0);
}
// DAP_PEQ
unsigned short AudioControlSGTL5000::eqFilterCount(uint8_t n) // valid to n&7, 0 thru 7 filters enabled.
{
return modify(DAP_PEQ,(n&7),7);
}
// DAP_AUDIO_EQ
unsigned short AudioControlSGTL5000::eqSelect(uint8_t n) // 0=NONE, 1=PEQ (7 IIR Biquad filters), 2=TONE (tone), 3=GEQ (5 band EQ)
{
return modify(DAP_AUDIO_EQ,n&3,3);
}
unsigned short AudioControlSGTL5000::eqBand(uint8_t bandNum, float n)
{
if(semi_automated) automate(1,3);
return dap_audio_eq_band(bandNum, n);
}
void AudioControlSGTL5000::eqBands(float bass, float mid_bass, float midrange, float mid_treble, float treble)
{
if(semi_automated) automate(1,3);
dap_audio_eq_band(0,bass);
dap_audio_eq_band(1,mid_bass);
dap_audio_eq_band(2,midrange);
dap_audio_eq_band(3,mid_treble);
dap_audio_eq_band(4,treble);
}
void AudioControlSGTL5000::eqBands(float bass, float treble) // dap_audio_eq(2);
{
if(semi_automated) automate(1,2);
dap_audio_eq_band(0,bass);
dap_audio_eq_band(4,treble);
}
// SGTL5000 PEQ Coefficient loader
void AudioControlSGTL5000::eqFilter(uint8_t filterNum, int *filterParameters)
{
// TODO: add the part that selects 7 PEQ filters.
if(semi_automated) automate(1,1,filterNum+1);
modify(DAP_FILTER_COEF_ACCESS,(uint16_t)filterNum,15);
write(DAP_COEF_WR_B0_MSB,(*filterParameters>>4)&65535);
write(DAP_COEF_WR_B0_LSB,(*filterParameters++)&15);
write(DAP_COEF_WR_B1_MSB,(*filterParameters>>4)&65535);
write(DAP_COEF_WR_B1_LSB,(*filterParameters++)&15);
write(DAP_COEF_WR_B2_MSB,(*filterParameters>>4)&65535);
write(DAP_COEF_WR_B2_LSB,(*filterParameters++)&15);
write(DAP_COEF_WR_A1_MSB,(*filterParameters>>4)&65535);
write(DAP_COEF_WR_A1_LSB,(*filterParameters++)&15);
write(DAP_COEF_WR_A2_MSB,(*filterParameters>>4)&65535);
write(DAP_COEF_WR_A2_LSB,(*filterParameters++)&15);
write(DAP_FILTER_COEF_ACCESS,(uint16_t)0x100|filterNum);
}
/* Valid values for dap_avc parameters
maxGain; Maximum gain that can be applied
0 - 0 dB
1 - 6.0 dB
2 - 12 dB
lbiResponse; Integrator Response
0 - 0 mS
1 - 25 mS
2 - 50 mS
3 - 100 mS
hardLimit
0 - Hard limit disabled. AVC Compressor/Expander enabled.
1 - Hard limit enabled. The signal is limited to the programmed threshold (signal saturates at the threshold)
threshold
floating point in range 0 to -96 dB
attack
floating point figure is dB/s rate at which gain is increased
decay
floating point figure is dB/s rate at which gain is reduced
*/
unsigned short AudioControlSGTL5000::autoVolumeControl(uint8_t maxGain, uint8_t lbiResponse, uint8_t hardLimit, float threshold, float attack, float decay)
{
//if(semi_automated&&(!read(DAP_CONTROL)&1)) audioProcessorEnable();
if(maxGain>2) maxGain=2;
lbiResponse&=3;
hardLimit&=1;
uint8_t thresh=(pow(10,threshold/20)*0.636)*pow(2,15);
uint8_t att=(1-pow(10,-(attack/(20*44100))))*pow(2,19);
uint8_t dec=(1-pow(10,-(decay/(20*44100))))*pow(2,23);
write(DAP_AVC_THRESHOLD,thresh);
write(DAP_AVC_ATTACK,att);
write(DAP_AVC_DECAY,dec);
return modify(DAP_AVC_CTRL,maxGain<<12|lbiResponse<<8|hardLimit<<5,3<<12|3<<8|1<<5);
}
unsigned short AudioControlSGTL5000::autoVolumeEnable(void)
{
return modify(DAP_AVC_CTRL, 1, 1);
}
unsigned short AudioControlSGTL5000::autoVolumeDisable(void)
{
return modify(DAP_AVC_CTRL, 0, 1);
}
unsigned short AudioControlSGTL5000::enhanceBass(float lr_lev, float bass_lev)
{
return modify(DAP_BASS_ENHANCE_CTRL,((0x3F-calcVol(lr_lev,0x3F))<<8) | (0x7F-calcVol(bass_lev,0x7F)), (0x3F<<8) | 0x7F);
}
unsigned short AudioControlSGTL5000::enhanceBass(float lr_lev, float bass_lev, uint8_t hpf_bypass, uint8_t cutoff)
{
modify(DAP_BASS_ENHANCE,(hpf_bypass&1)<<8|(cutoff&7)<<4,1<<8|7<<4);
return enhanceBass(lr_lev,bass_lev);
}
unsigned short AudioControlSGTL5000::enhanceBassEnable(void)
{
return modify(DAP_BASS_ENHANCE, 1, 1);
}
unsigned short AudioControlSGTL5000::enhanceBassDisable(void)
{
return modify(DAP_BASS_ENHANCE, 0, 1);
}
unsigned short AudioControlSGTL5000::surroundSound(uint8_t width)
{
return modify(DAP_SGTL_SURROUND,(width&7)<<4,7<<4);
}
unsigned short AudioControlSGTL5000::surroundSound(uint8_t width, uint8_t select)
{
return modify(DAP_SGTL_SURROUND,((width&7)<<4)|(select&3), (7<<4)|3);
}
unsigned short AudioControlSGTL5000::surroundSoundEnable(void)
{
return modify(DAP_SGTL_SURROUND, 3, 3);
}
unsigned short AudioControlSGTL5000::surroundSoundDisable(void)
{
return modify(DAP_SGTL_SURROUND, 0, 3);
}
unsigned char AudioControlSGTL5000::calcVol(float n, unsigned char range)
{
// n=(n*(((float)range)/100))+0.499;
n=(n*(float)range)+0.499;
if ((unsigned char)n>range) n=range;
return (unsigned char)n;
}
// DAP_AUDIO_EQ_BASS_BAND0 & DAP_AUDIO_EQ_BAND1 & DAP_AUDIO_EQ_BAND2 etc etc
unsigned short AudioControlSGTL5000::dap_audio_eq_band(uint8_t bandNum, float n) // by signed percentage -100/+100; dap_audio_eq(3);
{
n=(n*48)+0.499;
if(n<-47) n=-47;
if(n>48) n=48;
n+=47;
return modify(DAP_AUDIO_EQ_BASS_BAND0+(bandNum*2),(unsigned int)n,127);
}
void AudioControlSGTL5000::automate(uint8_t dap, uint8_t eq)
{
//if((dap!=0)&&(!(read(DAP_CONTROL)&1))) audioProcessorEnable();
if((read(DAP_AUDIO_EQ)&3) != eq) eqSelect(eq);
}
void AudioControlSGTL5000::automate(uint8_t dap, uint8_t eq, uint8_t filterCount)
{
automate(dap,eq);
if (filterCount > (read(DAP_PEQ)&7)) eqFilterCount(filterCount);
}
// if(SGTL5000_PEQ) quantization_unit=524288; if(AudioFilterBiquad) quantization_unit=2147483648;
void calcBiquad(uint8_t filtertype, float fC, float dB_Gain, float Q, uint32_t quantization_unit, uint32_t fS, int *coef)
{
// I used resources like http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
// to make this routine, I tested most of the filter types and they worked. Such filters have limits and
// before calling this routine with varying values the end user should check that those values are limited
// to valid results.
float A;
if(filtertype<FILTER_PARAEQ) A=pow(10,dB_Gain/20); else A=pow(10,dB_Gain/40);
float W0 = 2*3.14159265358979323846*fC/fS;
float cosw=cosf(W0);
float sinw=sinf(W0);
//float alpha = sinw*sinh((log(2)/2)*BW*W0/sinw);
//float beta = sqrt(2*A);
float alpha = sinw / (2 * Q);
float beta = sqrtf(A)/Q;
float b0,b1,b2,a0,a1,a2;
switch(filtertype) {
case FILTER_LOPASS:
b0 = (1.0F - cosw) * 0.5F; // =(1-COS($H$2))/2
b1 = 1.0F - cosw;
b2 = (1.0F - cosw) * 0.5F;
a0 = 1.0F + alpha;
a1 = 2.0F * cosw;
a2 = alpha - 1.0F;
break;
case FILTER_HIPASS:
b0 = (1.0F + cosw) * 0.5F;
b1 = -(cosw + 1.0F);
b2 = (1.0F + cosw) * 0.5F;
a0 = 1.0F + alpha;
a1 = 2.0F * cosw;
a2 = alpha - 1.0F;
break;
case FILTER_BANDPASS:
b0 = alpha;
b1 = 0.0F;
b2 = -alpha;
a0 = 1.0F + alpha;
a1 = 2.0F * cosw;
a2 = alpha - 1.0F;
break;
case FILTER_NOTCH:
b0=1;
b1=-2*cosw;
b2=1;
a0=1+alpha;
a1=2*cosw;
a2=-(1-alpha);
break;
case FILTER_PARAEQ:
b0 = 1 + (alpha*A);
b1 =-2 * cosw;
b2 = 1 - (alpha*A);
a0 = 1 + (alpha/A);
a1 = 2 * cosw;
a2 =-(1-(alpha/A));
break;
case FILTER_LOSHELF:
b0 = A * ((A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw));
b1 = 2.0F * A * ((A-1.0F) - ((A+1.0F)*cosw));
b2 = A * ((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
a0 = (A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw);
a1 = 2.0F * ((A-1.0F) + ((A+1.0F)*cosw));
a2 = -((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
break;
case FILTER_HISHELF:
b0 = A * ((A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw));
b1 = -2.0F * A * ((A-1.0F) + ((A+1.0F)*cosw));
b2 = A * ((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
a0 = (A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw);
a1 = -2.0F * ((A-1.0F) - ((A+1.0F)*cosw));
a2 = -((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
default:
b0 = 0.5;
b1 = 0.0;
b2 = 0.0;
a0 = 1.0;
a1 = 0.0;
a2 = 0.0;
}
a0=(a0*2)/(float)quantization_unit; // once here instead of five times there...
b0/=a0;
*coef++=(int)(b0+0.499);
b1/=a0;
*coef++=(int)(b1+0.499);
b2/=a0;
*coef++=(int)(b2+0.499);
a1/=a0;
*coef++=(int)(a1+0.499);
a2/=a0;
*coef++=(int)(a2+0.499);
}