AD1836

AD1836 Sound CODEC Linux Driver.

Supported devices

Evaluation Boards

Source Code

Status

Source

Mainline?

git

Yes

Files

Function

File

driver

sound/soc/codecs/ad1836.c

include

sound/soc/codecs/ad1836.h

Example device initialization

Unlike PCI or USB devices, SPI devices are not enumerated at the hardware level. Instead, the software must know which devices are connected on each SPI bus segment, and what slave selects these devices are using. For this reason, the kernel code must instantiate SPI devices explicitly. The most common method is to declare the SPI devices by bus number.

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

For more information see: Overview of Linux kernel SPI support

You need to set the modalias of your SPI info according to your codec. Valid values are ad1835, ad1836, ad1837, ad1838 and ad1839. You’ll also have to adjust bus_num and chip_select according to your board setup.

static struct spi_board_info board_spi_board_info[] __initdata = {
    [--snip--]
    {
        .modalias = "ad1836",
        .max_speed_hz = 3125000,     /* max spi clock (SCK) speed in HZ */
        .bus_num = 0,
        .chip_select = 4, /* CS, change it for your board */
        .mode = SPI_MODE_3,
    },
    [--snip--]
};

static int __init board_init(void)
{
    [--snip--]

    spi_register_board_info(board_spi_board_info, ARRAY_SIZE(board_spi_board_info));

    [--snip--]

    return 0;
}
arch_initcall(board_init);

ASoC DAPM Widgets

Name

Description

Model

DAC1OUT

DAC Channel1 Output

AD1835A, AD1836A, AD1838A

DAC2OUT

DAC Channel2 Output

AD1835A, AD1836A, AD1838A

DAC3OUT

DAC Channel3 Output

AD1835A, AD1836A, AD1838A

DAC4OUT

DAC Channel4 Output

AD1835A

ADC1IN

ADC Channel1 Input

AD1835A, AD1836A, AD1838A

ADC2IN

ADC Channel2 Input

AD1836A

ALSA Controls

Name

Description

Model

ADC High Pass Filter Switch

Enable/Disable ADC high-pass filter

AD1835A, AD1836A, AD1838A

Playback Deemphasis

Select playback de-emphasis. Possible Values: None, 44.1kHz, 32kHz, 48kHz

AD1835A, AD1836A, AD1838A

DAC1 Playback Volume

DAC Channel 1 volume

AD1835A, AD1836A, AD1838A

DAC2 Playback Volume

DAC Channel 2 volume

AD1835A, AD1836A, AD1838A

DAC3 Playback Volume

DAC Channel 3 volume

AD1835A, AD1836A, AD1838A

DAC4 Playback Volume

DAC Channel 4 volume

AD1835A

DAC1 Playback Switch

Mute/Unmute DAC Channel 1

AD1835A, AD1836A, AD1838A

DAC2 Playback Switch

Mute/Unmute DAC Channel 2

AD1835A, AD1836A, AD1838A

DAC3 Playback Switch

Mute/Unmute DAC Channel 3

AD1835A, AD1836A, AD1838A

DAC4 Playback Switch

Mute/Unmute DAC Channel 4

AD1835A

ADC1 Capture Switch

Mute/Unmute ADC Channel1

AD1835A, AD1836A, AD1838A

ADC2 Capture Switch

Mute/Unmute ADC Channel2

AD1836A

ADC2 Capture Volume

Gain for ADC Channel 2

AD1836A

DAI Configuration

The CODEC driver registers one DAI named depending on the chip model used.

DAI name

Model

ad1835-hifi

AD1835, AD1837

ad1836-hifi

AD1836

ad1838-hifi

AD1838, AD1839

Supported DAI formats

Name

Supported by driver

Description

SND_SOC_DAIFMT_I2S

no

I2S mode

SND_SOC_DAIFMT_RIGHT_J

no

Right Justified mode

SND_SOC_DAIFMT_LEFT_J

no

Left Justified mode

SND_SOC_DAIFMT_DSP_A

yes

data MSB after FRM LRC

SND_SOC_DAIFMT_DSP_B

no

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

no

Normal bit- and frameclock

SND_SOC_DAIFMT_NB_IF

no

Normal bitclock, inverted frameclock

SND_SOC_DAIFMT_IB_NF

no

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

no

Codec bit- and frameclock slave

Example DAI Configuration

static int bf5xx_ad1836_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 ret;

    /* set cpu DAI configuration */
    ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
        SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
    if (ret < 0)
        return ret;

    /* set codec DAI configuration */
    ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
        SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
    if (ret < 0)
        return ret;

    return 0;
}

static struct snd_soc_ops bf5xx_ad1836_ops = {
    .hw_params = bf5xx_ad1836_hw_params,
};

static struct snd_soc_dai_link bf5xx_ad1836_dai = {
    .name = "ad1836",
    .stream_name = "AD1836",
    .cpu_dai_name = "bfin-tdm.0",
    .codec_dai_name = "ad1836-hifi",
    .platform_name = "bfin-tdm-pcm-audio",
    .codec_name = "spi0.4",
    .ops = &bf5xx_ad1836_ops,
};

AD1836 evaluation board driver

Adding Kernel Support - As a module

To add support for the built-in codec AD183X of BF5XX to the kernel build system, a few things must be enabled properly for things to work.The configuration is as following:

Linux Kernel Configuration
  Device Drivers  --->
    Sound  --->
      <M> Sound card support
        Advanced Linux Sound Architecture  --->
          <M> Advanced Linux Sound Architecture
          < > Sequencer support
          <M> OSS Mixer API
          <M> OSS PCM (digital audio) API
          <M>   ALSA for SoC audio support  --->
              <M>   SoC I2S(TDM mode) Audio for the ADI BF5xx chip
              <M>   SoC AD183X Audio support for BF5xx

Doing this will create modules (outside the kernel). The modules will be inserted automatically when it is needed. You can also build sound driver into kernel.

Testing the built in kernel driver

If audio is configured as modules, skip this section. If audio is built into kernel and you have booted the kernel, there are a few things to check to ensure audio is working:

  1. Check the boot messages to see if you have booted the correct kernel. During kernel boot, it should print out:

Advanced Linux Sound Architecture Driver Version 1.0.12rc1 (Thu Jun 22 13:55:50 2006 UTC).
ASoC version 0.13.1
dma rx:3 tx:4, err irq:45, regs:
asoc: AD183X <-> bf5xx-tdm mapping ok
ALSA device list:
  #0: bf5xx_ad183x (AD183X)

Testing the audio module

root:/> modprobe snd-ad183x
dma rx:3 tx:4, err irq:45, regs:ffc00800
asoc: AD183X <-> bf5xx-tdm mapping ok
root:/> modprobe snd-pcm-oss
root:/> lsmod
Module                  Size  Used by
snd_pcm_oss            28414  0
snd_mixer_oss          10215  1 snd_pcm_oss
snd_ad183x               801  0
snd_bf5xx_tdm           1857  1 snd_ad183x
snd_soc_ad183x          8033  1 snd_ad183x
snd_soc_bf5xx_tdm       2157  1 snd_ad183x
snd_soc_bf5xx_sport     9392  2 snd_bf5xx_tdm,snd_soc_bf5xx_tdm
snd_soc_core           33839  4 snd_ad183x,snd_bf5xx_tdm,snd_soc_ad183x,snd_soc_bf5xx_tdm
snd_pcm                44936  3 snd_pcm_oss,snd_bf5xx_tdm,snd_soc_core
snd_page_alloc          2753  1 snd_pcm
snd_timer              12412  1 snd_pcm
snd                    32171  5 snd_pcm_oss,snd_mixer_oss,snd_soc_core,snd_pcm,snd_timer
input_core             15713  1 snd
soundcore               3591  1 snd

root:~> tone
TONE: generating sine wave at 1000 Hz...

Driver testing

  1. Check the output

root:~> tone
TONE: generating sine wave at 1000 Hz...

You should hear something out of the headphone Jack.

  1. Check and set the audio mixer:

root:/> amixer
Simple mixer control 'Playback Deemphasis',0
  Capabilities: enum
  Items: 'None' '44.1kHz' '32kHz' '48kHz'
  Item0: '48kHz'
Simple mixer control 'ADC High Pass Filter',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'ADC1',0
  Capabilities: pswitch
  Playback channels: Front Left - Front Right
  Mono:
  Front Left: Playback [on]
  Front Right: Playback [on]
Simple mixer control 'ADC2',0
  Capabilities: pswitch
  Playback channels: Front Left - Front Right
  Mono:
  Front Left: Playback [on]
  Front Right: Playback [on]
Simple mixer control 'DAC1',0
  Capabilities: volume pswitch
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1023
  Front Left: 1023 [100%] Playback [on]
  Front Right: 1023 [100%] Playback [on]
Simple mixer control 'DAC2',0
  Capabilities: volume pswitch
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1023
  Front Left: 1023 [100%] Playback [on]
  Front Right: 1023 [100%] Playback [on]
Simple mixer control 'DAC3',0
  Capabilities: volume pswitch
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1023
  Front Left: 1023 [100%] Playback [on]
  Front Right: 1023 [100%] Playback [on]

root:/> amixer sset 'DAC3' 200
Simple mixer control 'DAC3',0
  Capabilities: volume pswitch
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1023
  Front Left: 200 [20%] Playback [on]
  Front Right: 200 [20%] Playback [on]

Check to make sure mp3s work (assuming you have built mp3play). The first step is to download a mp3 file onto the platform. The wget command assumes that networking is properly configured (you have an IP number, the default gateway is set, and DNS servers can be accessed), and working.

root:/> cd /var
root:/var> wget http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3

Next, play it with mp3play:

root:/var> mp3play ABCOWhosOnFirstclip.mp3

You can play it in one step with:

root:~> mp3play http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3
%%http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3: MPEG2-III (0 ms)%%

Optionally check to make sure the mic and headphone are working properly:

root:~> arecord -d 10 test.wav
Recording WAVE "test.wav" : Unsigned 8 bit, Rate 8000 Hz, Mono
root:~> aplay test.wav

This should record 10 seconds of whatever is on the Line, and then play it back over the output. You should also be able to do a talkthrough, and hear on the speakers anything you put on the line.

root:~> arecord | aplay