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This version (16 Nov 2012 14:27) was approved by larsc.The Previously approved version (05 Sep 2012 14:23) is available.Diff

AD1836 Sound CODEC Linux Driver

Supported devices

Evaluation Boards

Source Code

Status

Source Mainline?
git Yes

Files

Example device initialization

Declaring SPI slave devices

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: Documentation/spi/spi-summary

21 Oct 2010 16:10 · Michael Hennerich

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 Justified 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]
Also you can run “alsamixer” to get graphic configuration interface, OSS-based “mixer” can work too.

  1. Check to make sure mp3s work (assuming you have built mp3play),
    1. 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. See the network setup page for more info.
      root:/> cd /var
      root:/var> wget http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3
      
    2. Next, play it with mp3play:
      root:/var> mp3play ABCOWhosOnFirstclip.mp3
  2. 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)
    
  3. 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.
  4. You should also be able to do a “talkthrough”, and hear on the speakers anything you put on the line.
    root:~> arecord | aplay