This version is outdated by a newer approved version.
This version (29 Aug 2011 14:58) was approved by Michael Hennerich.The Previously approved version (29 Aug 2011 14:46) is available.
This version (29 Aug 2011 14:58) was approved by Michael Hennerich.The Previously approved version (29 Aug 2011 14:46) is available.This is an old revision of the document!
Table of Contents
ADAU1373 Sound CODEC Linux Driver
Supported Devices
This driver supports the
ADAU1373
Source Code
Status
| Source | Mainlined? |
|---|---|
| git | In progress |
Files
| Function | File |
|---|---|
| driver | sound/soc/codecs/adau1373.c |
| include | sound/soc/codecs/adau1373.h |
| include | include/sound/adau1373.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
Declaring I2C devices
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: Documentation/i2c/instantiating-devices.rst
static struct i2c_board_info __initdata bfin_i2c_board_info[] = { [--snip--] { I2C_BOARD_INFO("adau1373", 0x1a), }, [--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 |
|---|---|
| AIN1L | Left Channel Input 1 |
| AIN1R | Right Channel Input 1 |
| AIN2L | Left Channel Input 2 |
| AIN2R | Right Channel Input 2 |
| AIN3L | Left Channel Input 3 |
| AIN3R | Right Channel Input 3 |
| AIN4L | Left Channel Input 4 |
| AIN4R | Right Channel Input 4 |
| DMIC1DAT | Serial Data Input Digital Microphone 1 and 2 |
| DMIC2DAT | Serial Data Input Digital Microphone 3 and 4 |
| LOUT1L | Left Channel Line Output 1 |
| LOUT1R | Right Channel Line Output 1 |
| LOUT2L | Left Channel Line Output 2 |
| LOUT2R | Right Channel Line Output 2 |
| HPL | Left Headphone Output |
| HPR | Right Headphone Output |
| SPKL | Left Speaker Output |
| SPKR | Right Speaker Output |
| EP | Eearpiece Output |
| MICBIAS1 | Micbias 1 supply |
| MICBIAS2 | Micbias 2 supply |
ALSA Controls
| Name | Description | Configuration |
|---|---|---|
| AIF1 Capture Volume | ||
| AIF2 Capture Volume | ||
| AIF3 Capture Volume | ||
| ADC Capture Volume | ||
| DMIC Capture Volume | ||
| AIF1 Playback Volume | ||
| AIF2 Playback Volume | ||
| AIF3 Playback Volume | ||
| DAC1 Playback Volume | ||
| DAC2 Playback Volume | ||
| Lineout1 Playback Volume | ||
| Speaker Playback Volume | ||
| Headphone Playback Volume | ||
| Input 1 Capture Volume | ||
| Input 2 Capture Volume | ||
| Input 3 Capture Volume | ||
| Input 4 Capture Volume | ||
| Earpiece Playback Volume | ||
| AIF3 Boost Playback Volume | ||
| AIF2 Boost Playback Volume | ||
| AIF1 Boost Playback Volume | ||
| AIF3 Boost Capture Volume | ||
| AIF2 Boost Capture Volume | ||
| AIF1 Boost Capture Volume | ||
| DMIC Boost Capture Volume | ||
| ADC Boost Capture Volume | ||
| DAC2 Boost Playback Volume | ||
| DAC1 Boost Playback Volume | ||
| Input 1 Boost Capture Volume | ||
| Input 2 Boost Capture Volume | ||
| Input 3 Boost Capture Volume | ||
| Input 4 Boost Capture Volume | ||
| Speaker Boost Playback Volume | ||
| Lineout1 LR Mux | ||
| Speaker LR Mux | ||
| HPF Cutoff | ||
| HPF Switch | ||
| HPF Channel | ||
| Bass HPF Cutoff | ||
| Bass Clip Level Threshold | ||
| Bass LPF Cutoff | ||
| Bass Playback Switch | ||
| Bass Playback Volume | ||
| Bass Channel | ||
| 3D Freq | ||
| 3D Level | ||
| 3D Playback Switch | ||
| 3D Playback Volume | ||
| 3D Channel | ||
| Zero Cross Switch | ||
| Lineout2 Playback Volume | Single-ended lineout | |
| Lineout2 LR Mux | Single-ended lineout | |
| DRC1 Channel | ||
| DRC2 Channel | ||
| DRC3 Channel |
Supported DAI formats
| Name | Supported by driver | Description |
|---|---|---|
| SND_SOC_DAIFMT_I2S | yes | I2S Justified mode |
| SND_SOC_DAIFMT_RIGHT_J | yes | Right Justified mode |
| SND_SOC_DAIFMT_LEFT_J | yes | Left Justified mode |
| SND_SOC_DAIFMT_DSP_A | no | 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 |
ADAU1373 evaluation board driver
Source
Status
| Source | Mainlined? |
|---|---|
| In progress |
Files
| Function | File |
|---|---|
| driver | sound/soc/blackfin/bfin-eval-adau1373.c |
Adding Kernel Support - As a module
Skip this section if you want to build the driver into the kernel directly. To add support for codec ADAU1373 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
System on Chip audio support --->
<M> ALSA for SoC audio support
<M> SoC I2S Audio for the ADI BF5xx chip
<M> SoC ADAU1373 Audio support
< > SoC AC97 Audio support for BF5xx
(0) Set a SPORT for Sound chip
I2C bus is used to configure the codec. So, if the audio driver is built into kernel, the I2c driver is also built into kernel automatically. But if the audio driver is built as module, then make sure that the I2C driver is loaded before the audio module.
Linux Kernel Configuration
Device Drivers --->
<*> I2C support --->
--- I2C support
I2C Hardware Bus Support --->
<*> Blackfin TWI I2C support
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:15, regs:ffc00800 adau1371 Audio Codec 0.1<6>dma_alloc_init: dma_page @ 0x03011000 - 512 pages at 0x03e00000 asoc: ADAU1373 <-> bf5xx-i2s-0 mapping ok ALSA device list: #0: bf5xx_adau1373 (ADAU1373)
Testing the audio module
root:~> modprobe snd-adau1373 root:~> modprobe snd-pcm-oss root:~> lsmod Module Size Used by snd_pcm_oss 31968 0 snd_mixer_oss 11360 1 snd_pcm_oss snd_adau1373 1412 0 snd_soc_adau1373 8528 1 snd_adau1373 snd_soc_bf5xx 2784 1 snd_adau1373 snd_soc_bf5xx_i2s 10916 2 snd_adau1373,snd_soc_bf5xx snd_soc_core 17120 3 snd_adau1373,snd_soc_adau1373,snd_soc_bf5xx snd_pcm 48356 3 snd_pcm_oss,snd_soc_bf5xx,snd_soc_core snd_page_alloc 4232 1 snd_pcm snd_timer 13796 1 snd_pcm snd 31092 6 snd_pcm_oss,snd_mixer_oss,snd_soc_adau1373,snd_soc_core,snd_pcm,snd_timer soundcore 3940 1 snd root:~> tone TONE: generating sine wave at 1000 Hz...
Testing Audio
- Check the output
root:~> tone TONE: generating sine wave at 1000 Hz...
You should hear something out of the headphone Jack on the top of J8. - Select audio input to INPB (the default is INPA, assuming you have built ALSA utils):
root:/> amixer sset 'Input Mux' 'INPB' Simple mixer control 'Input Mux',0 Capabilities: enum Items: 'INPA' 'INPB' 'INPD' 'INPD' Item0: 'INPB'
Also you can run “alsamixer” to get graphic configuration interface. - 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 audio out is right:
root:/> amixer sset 'Output Mixer' 'Line' Simple mixer control 'Output Mixer',0 Capabilities: enum Items: 'Line' 'Class D' 'HeadPhone' Item0: 'Line' root:~> arecord -d 10 test.wav Recording WAVE "test.wav" : Unsigned 8 bit, Rate 8000 Hz, Mono root:~> aplay test.wav Adjust playback volume can be done through command:amixer sset Master 80% here 80 is the ratio.
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/adau1373.1314622713.txt.gz · Last modified: 29 Aug 2011 14:58 by Michael Hennerich