This version is outdated by a newer approved version.
This version (09 Nov 2011 14:20) was approved by Lars-Peter Clausen.The Previously approved version (29 Aug 2011 14:56) is available.
This version (09 Nov 2011 14:20) was approved by Lars-Peter Clausen.The Previously approved version (29 Aug 2011 14:56) is available.This is an old revision of the document!
Table of Contents
AD1836 Sound CODEC Linux Driver
Supported devices
Source Code
Status
Files
| Function | File |
|---|---|
| driver | sound/soc/codecs/ad1836.c |
| include | sound/soc/codecs/ad1836.h |
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.rst
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:
- 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
- Check the output
root:~> tone TONE: generating sine wave at 1000 Hz...You should hear something out of the headphone Jack.
- 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.
- Check to make sure mp3s work (assuming you have built mp3play),
- The first step is to download a mp3 file onto the platform. The
wgetcommand 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
- 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
resources/tools-software/linux-drivers/sound/ad1835a.1320844819.txt.gz · Last modified: 09 Nov 2011 14:20 by Lars-Peter Clausen