ADAU1361

ADAU1361 Sound CODEC Linux Driver.

Supported Devices

Reference Circuits

Evaluation Boards

Source Code

Status

Source

Mainlined?

git

Yes

Files

Function

File

driver

sound/soc/codecs/adau1761.c

driver

sound/soc/codecs/adau17x1.c

include

sound/soc/codecs/adau17x1.h

include

include/linux/platform_data/adau17x1.h

Example device initialization

For compile time configuration, it’s common Linux practice to keep board- and application-specific configuration out of the main driver file, instead putting it into the board support file.

For devices on custom boards, as typical of embedded and SoC-(system-on-chip) based hardware, Linux uses platform_data to point to board-specific structures describing devices and how they are connected to the SoC. This can include available ports, chip variants, preferred modes, default initialization, additional pin roles, and so on. This shrinks the board-support packages (BSPs) and minimizes board and application specific #ifdefs in drivers.

I2C

Unlike PCI or USB devices, I2C devices are not enumerated at the hardware level. Instead, the software must know which devices are connected on each I2C bus segment, and what address these devices are using. For this reason, the kernel code must instantiate I2C devices explicitly. There are different ways to achieve this, depending on the context and requirements. However the most common method is to declare the I2C devices by bus number.

This method is appropriate when the I2C bus is a system bus, as in many embedded systems, wherein each I2C bus has a number which is known in advance. It is thus possible to pre-declare the I2C devices that inhabit this bus. This is done with an array of struct i2c_board_info, which is registered by calling i2c_register_board_info().

So, to enable such a driver one need only edit the board support file by adding an appropriate entry to i2c_board_info.

For more information see: How to instantiate I2C devices

The I2C device id depends on the ADDR0 and ADDR1 pin settings and needs to be set according to your board setup.

ADDR1

ADDR0

I2C device id

0

0

0x38

0

1

0x39

1

0

0x3a

1

1

0x3b

In this example we assume ADDR0=0 and ADDR1=0.

static struct i2c_board_info __initdata bfin_i2c_board_info[] = {

    [--snip--]
    {
        I2C_BOARD_INFO("adau1761", 0x38),
    },
    [--snip--]
}

static int __init stamp_init(void)
{
    [--snip--]
    i2c_register_board_info(0, bfin_i2c_board_info,
                ARRAY_SIZE(bfin_i2c_board_info));
    [--snip--]

    return 0;
}
arch_initcall(board_init);

ASoC DAPM Widgets

Name

Description

Configuration

LAUX

Left Channel Single-Ended Auxiliary Input

RAUX

Right Channel Single-Ended Auxiliary Input

LINP

Left Channel Noninverting Input or Single-Ended Input 0

LINN

Left Channel Inverting Input or Single-Ended Input 1

RINP

Right Channel Noninverting Input or Single-Ended Input 2

RINN

Right Channel Inverting Input or Single-Ended Input 3

LOUT

Left Line Output

ROUT

Right Line Output

LHP

Left Headphone Output

RHP

Right Headphone Output

MONOOUT

Mono Output

Headphone mode not capless

DMIC

Digital Microphone in

JACKDET/MICIN pin configured for DMIC

MICBIAS

Bias Voltage for Electret Microphone

ALSA Controls

Name

Description

Configuration

Digital Capture Volume

Digital volume attenuation for input from either the ADC or the digital microphone input.

Digital Playback Volume

Digital volume attenuation for output from the DAC

ADC High Pass Filter Switch

Enable/Disable ADC high-pass-filter

Playback De-emphasis

Enable/Disable Playback de-empahsis

Capture Boost

Mixer amplifier bias boost Valid values: Normal operation, Boost Level 1, Boost Level 2, Boost Level 3

Mic Bias Mode

Microphone bias. Valid values: Normal operation, High performance

DAC Mono Stereo

DAC mono mode. Valid values: Stereo, Mono Left Channel (L+R), Mono Right Channel (L+R), Mono (L+R)

Capture Mux

Selects the source for the capture I2S audio data stream, can either be the DSP or the ADC/DMIC

ADAU1761 only

DAC Playback Mux

Selects the source for the DACs, can either be from the I2S audio interface or the DSP

ADAU1761 only

Input 1 Capture Volume

Gain for single-ended input from the LINP pin

Single-ended inputs

Input 2 Capture Volume

Gain for single-ended input from the LINN pin

Single-ended inputs

Input 3 Capture Volume

Gain for single-ended input from the RINP pin

Single-ended inputs

Input 4 Capture Volume

Gain for single-ended input from the RINN pin

Single-ended inputs

Capture Volume

Differential PGA input volume

Differential inputs

Capture Switch

Mute/Unmute differential input

Differential inputs

Aux Capture Volume

Single-ended auxiliary input gain in the record path

PGA Boost Capture Volume

Differential PGA input gain boost

Differential inputs

Headphone Playback Volume

Headphone volume

Headphone Playback Switch

Mute/Unmute Headphone signal

Lineout Playback Volume

Lineout volume

Lineout Playback Switch

Mute/Unmute Lineout signal

ADC Bias

ADC bias. Valid values: Normal operation, Extreme powersaving, Enhanced performance, Power saving

DAC Bias

DAC bias. Valid values: Normal operation, Extreme powersaving, Enhanced performance, Power saving

Capture Bias

Record path bias. Valid values: Normal operation, Enhanced performance, Power saving

Playback Bias

Playback path bias. Valid values: Normal operation, Enhanced performance, Power saving

Headphone Bias

Headphone bias. Valid values: Normal operation, Extreme powersaving, Enhanced performance, Power saving

Right LR Playback Mixer Left Volume

Left Playback Mixer gain to the Right LR Playback Mixer

Right LR Playback Mixer Right Volume

Right Playback Mixer gain to the Right LR Playback Mixer

Left LR Playback Mixer Left Volume

Left Playback Mixer gain to the Left LR Playback Mixer

Left LR Playback Mixer Right Volume

Right Playback Mixer gain to the Left LR Playback Mixer

Right Playback Mixer Left DAC Switch

Mix Left DAC signal into the Right Playback Mixer

Right Playback Mixer Right DAC Switch

Mix Right DAC signal into the Right Playback Mixer

Right Playback Mixer Aux Bypass Volume

Auxiliary input gain to the Right Playback Mixer

Right Playback Mixer Right Bypass Volume

Right Record Mixer gain to the Right Playback Mixer

Right Playback Mixer Left Bypass Volume

Left Record Mixer gain to the Right Playback Mixer

Left Playback Mixer Left DAC Switch

Mix Left DAC signal into the Left Playback Mixer

Left Playback Mixer Right DAC Switch

Mix Right DAC signal into the Left Playback Mixer

Left Playback Mixer Aux Bypass Volume

Auxiliary input gain to the Left Playback Mixer

Left Playback Mixer Right Bypass Volume

Right Record Mixer gain to the Left Playback Mixer

Left Playback Mixer Left Bypass Volume

Left Record Mixer gain to the Left Playback Mixer

Mono Playback Switch

Mute/Unmute Mono out signal

Headphone mode not capless

Input Select

Select capture path source. Valid values: ADC, DMIC

DMIC/JACKDETECT pin configured as DMIC

Jack Detect Switch

Enable/Disable Jack insertion detection. If enable the Lineout signals are muted if a headphone is inserted.

DMIC/JACKDETECT pin configured as JACKDETECT

PLL configuration

The ADAU1761 features one PLL:

enum adau17x1_pll {
    ADAU17X1_PLL
};

The PLL input signal is the MCLK signal.

enum adau17x1_pll_src {
    ADAU17X1_PLL_SRC_MCLK,
};

(The input frequency must configured to be between 8000000 and 27000000 Hz (8MHz - 27MHz). The output frequency must be configured to be between 45158000 and 49152000. Configuring the PLL with other input or output frequency will fail.)

The PLL runs at 1024 times the base sample rate. So for a 48000 Hz based sample rate you’d normally choose 49152000 Hz for the PLL output frequncey and for a 44100 Hz based sample rate 45158400 Hz.

DAI configuration

The codec driver registers one DAI: adau-hifi

Supported DAI formats

Name

Supported by driver

Description

SND_SOC_DAIFMT_I2S

yes

I2S mode

SND_SOC_DAIFMT_RIGHT_J

yes

Right Justified mode

SND_SOC_DAIFMT_LEFT_J

yes

Left Justified mode

SND_SOC_DAIFMT_DSP_A

yes

data MSB after FRM LRC

SND_SOC_DAIFMT_DSP_B

yes

data MSB during FRM LRC

SND_SOC_DAIFMT_AC97

no

AC97 mode

SND_SOC_DAIFMT_PDM

no

Pulse density modulation

SND_SOC_DAIFMT_NB_NF

yes

Normal bit- and frameclock

SND_SOC_DAIFMT_NB_IF

yes

Normal bitclock, inverted frameclock

SND_SOC_DAIFMT_IB_NF

yes

Inverted frameclock, normal bitclock

SND_SOC_DAIFMT_IB_IF

yes

Inverted bit- and frameclock

SND_SOC_DAIFMT_CBM_CFM

yes

Codec bit- and frameclock master

SND_SOC_DAIFMT_CBS_CFM

no

Codec bitclock slave, frameclock master

SND_SOC_DAIFMT_CBM_CFS

no

Codec bitclock master, frameclock slave

SND_SOC_DAIFMT_CBS_CFS

yes

Codec bit- and frameclock slave

DAI sysclk

The DAIs can either use the PLL or the MCLK signal as source.

When using the PLL the DAIs rate should be set to the rate of the PLL. When using MCLK the rate should be set to frequency of the external MCLK signal.

enum adau17x1_clk_src {
    ADAU17X1_CLK_SRC_MCLK,
    ADAU17X1_CLK_SRC_PLL,
};

Example DAI configuration

static int bfin_eval_adau1x61_hw_params(struct snd_pcm_substream *substream,
    struct snd_pcm_hw_params *params)
{
    struct snd_soc_pcm_runtime *rtd = substream->private_data;
    struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
    struct snd_soc_dai *codec_dai = rtd->codec_dai;
    int pll_rate;
    int ret;

    ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
            SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
    if (ret)
        return ret;

    ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
            SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
    if (ret)
        return ret;

    switch (params_rate(params)) {
    case 48000:
    case 8000:
    case 12000:
    case 16000:
    case 24000:
    case 32000:
    case 96000:
        pll_rate = 48000 * 1024;
        break;
    case 44100:
    case 7350:
    case 11025:
    case 14700:
    case 22050:
    case 29400:
    case 88200:
        pll_rate = 44100 * 1024;
        break;
    default:
        return -EINVAL;
    }

    ret = snd_soc_dai_set_pll(codec_dai, ADAU17X1_PLL,
            ADAU17X1_PLL_SRC_MCLK, 12288000, pll_rate);
    if (ret)
        return ret;

    ret = snd_soc_dai_set_sysclk(codec_dai, ADAU17X1_CLK_SRC_PLL, pll_rate,
            SND_SOC_CLOCK_IN);

    return ret;
}

static struct snd_soc_ops bfin_eval_adau1x61_ops = {
    .hw_params = bfin_eval_adau1x61_hw_params,
};

static struct snd_soc_dai_link bfin_eval_adau1x61_dai = {
    .name = "adau1x61",
    .stream_name = "ADAU1X61",
    .cpu_dai_name = "bfin-i2s.0",
    .codec_dai_name = "adau-hifi",
    .platform_name = "bfin-i2s-pcm-audio",
    .codec_name = "adau1761.0-0038",
    .ops = &bfin_eval_adau1x61_ops,
};

TDM configuration

The ADAU1361 and ADAU1761 chips have basic TDM support.

  • The number of slots can be either 2, 4 or 8 (ADAU1761 only).

  • The slot width can be 32 (ADAU1361 only), 64, 48, 128 or 256 (ADAU1761 only)

  • The slot mask must be either 0x03 (slot 0 and 1), 0x0c (slot 2 and 3), 0x30 (slot 4 and 5), 0xc0 (slot 6 and 7)

Example TDM configuration:

ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 0x0c, 8, 64);

SigmaDSP Firmware

In order to use the SigmaDSP core of the ADAU1761 you need to provide a firmware file. Please refer to the SigmaDSP Firmware Utility for Linux page on how to generate a firmware file. The firmware file for ADAU1761 driver has to be named adau1761.bin.

ADAU1X61 evaluation board driver

There is no dedicated Blackfin STAMP evaluation board for the ADAU1361 or ADAU1761. During test and driver development we used the EVAL-ADAU1361Z and EVAL-ADAU1761Z boards.

It can be easily wired to the Blackfin STAMP SPORT header.

Source

Status

Source

Mainlined?

git

Yes

Files

Function

File

driver

sound/soc/blackfin/bfin-eval-adau1x61.c

Kernel configuration

Device Drivers  --->
[*] I2C support  --->
[*]   I2C Hardware Bus support  --->
***     I2C system bus drivers (mostly embedded / system-on-chip) ***
<*>       Blackfin TWI I2C support
(100)     Blackfin TWI I2C clock (kHz)

Enable ALSA SoC evaluation board driver:

Device Drivers  --->
<M> Sound card support  --->
<M>   Advanced Linux Sound Architecture  --->
<M>     ALSA for SoC audio support  --->
<M>       Support for the EVAL-ADAU1X61 boards on Blackfin eval boards

Hardware configuration

Connect the STAMP SPORT 0 port (P6) to the EVAL-ADAU1X61 J1 and J6 headers.

Note that the SPORT has separate signals for the capture and playback clocks, while the ADAU1361 uses the same clock signals for both, so the EVAL-ADAU1X61 clock signal pins need to be connected to two STAMP pins each.

STAMP pin

EVAL-ADAU1X61 pin

Function

P6-26 (SPORT 0 - PJ2_SCL)

J1-1

I2C SCL

P6-24 (SPORT 0 - PJ3_SDA)

J1-3

I2C SDA

P6-6 (SPORT 0 - PJ9_TSCLK0), P6-16 (SPORT 0 - PJ6_RSCLK0)

J6-6

BCLK

P6-11 (SPORT 0 - PJ10_TFS0), P6-7 (SPORT 0 - PJ7_RFS0)

J6-8

LRCLK

P6-14 (SPORT 0 - PJ11_DT0PRI

J6-4

Playback data

P6-8 (SPORT 0 - PJ8_DR0PRI)

J6-2

Captrue data

P6-33

J6-1

GND

Driver testing

Load the driver and make sure the sound card is properly instantiated.

root:/> modprobe snd-bf5xx-i2s
root:/> modprobe snd-soc-bf5xx-i2s
root:/> modprobe snd-soc-adau1761
root:/> modprobe snd-soc-bfin-eval-adau1x61
bfin-i2s bfin-i2s.0: dma rx:3 tx:4, err irq:45, regs:ffc00800
dma_alloc_init: dma_page @ 0x02791000 - 256 pages at 0x03f00000
asoc: adau-hifi <-> bfin-i2s.0 mapping ok
ALSA device list:
  #0: bfin-eval-adau1x61

root:/> modprobe snd-pcm-oss
root:/> tone
TONE: generating sine wave at 1000 Hz...

root:/> arecord -f cd | aplay
Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo

Important

If you do not hear any sound during testing make sure you have enabled the necessary Switches and Volumes. E.g. for playback on the Headphone output you need to enable the Left Playback Mixer Left DAC, Right Playback Mixer Right DAC and the Headphone Playback switches and set the DAC Playback Mux to AIFIN.

More information

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