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ADAU1361声音编解码器Linux驱动程序

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This version (15 Jul 2016 18:57) was approved by Lars-Peter Clausen.The Previously approved version (13 Jul 2016 18:30) is available.Diff

Table of Contents

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.

21 Oct 2010 16:10

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

21 Oct 2010 16:10

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  --->
 Sound card support  --->
   Advanced Linux Sound Architecture  --->
     ALSA for SoC audio support  --->
       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.

This specifies any shell prompt running on the target 

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

This specifies any shell prompt running on the target 

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

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

Need Help?

31 Jul 2012 17:53 · Lars-Peter Clausen
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