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This driver supports the
ADAU1373
Source | Mainlined? |
---|---|
git | In progress |
Function | File |
---|---|
driver | sound/soc/codecs/adau1373.c |
include | sound/soc/codecs/adau1373.h |
include | include/sound/adau1373.h |
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.
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);
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 |
Name | Description | Configuration |
---|---|---|
AIF1 Capture Volume | Digital Audio Interface A Recording Volume | |
AIF2 Capture Volume | Digital Audio Interface B Recording Volume | |
AIF3 Capture Volume | Digital Audio Interface C Recording Volume | |
ADC Capture Volume | ADC Recording Volume | |
DMIC Capture Volume | DMIC Recording Volume | |
Input 1 Capture Volume | Input 1 Gain | |
Input 2 Capture Volume | Input 2 Gain | |
Input 3 Capture Volume | Input 3 Gain | |
Input 4 Capture Volume | Input 4 Gain | |
Input 1 Boost Capture Volume | Input 1 ADC Boost (+20dB) | |
Input 2 Boost Capture Volume | Input 2 ADC Boost (+20dB) | |
Input 3 Boost Capture Volume | Input 3 ADC Boost (+20dB) | |
Input 4 Boost Capture Volume | Input 4 ADC Boost (+20dB) | |
AIF1 Playback Volume | Digital Audio Interface A Playback Datapath Volume | |
AIF2 Playback Volume | Digital Audio Interface B Playback Datapath Volume | |
AIF3 Playback Volume | Digital Audio Interface C Playback Datapath Volume | |
DAC1 Playback Volume | DAC1 Playback Volume | |
DAC2 Playback Volume | DAC2 Playback Volume | |
Lineout1 Playback Volume | Lineout 1 Volume | |
Lineout2 Playback Volume | Lineout 1 Volume | Single-ended lineout |
Speaker Playback Volume | Speaker Out Volume | |
Headphone Playback Volume | Heaphone Out Volume | |
Earpiece Playback Volume | Earpiece Amplifier Gain | |
AIF1 Boost Playback Volume | Digital Audio Interface A Playback Gain (+6dB) | |
AIF2 Boost Playback Volume | Digital Audio Interface B Playback Gain (+6dB) | |
AIF3 Boost Playback Volume | Digital Audio Interface C Playback Gain (+6dB) | |
AIF1 Boost Capture Volume | Digital Audio Interface C Recording Gain (+6dB) | |
AIF2 Boost Capture Volume | Digital Audio Interface C Recording Gain (+6dB) | |
AIF3 Boost Capture Volume | Digital Audio Interface C Recording Gain (+6dB) | |
ADC Boost Capture Volume | ADC Recording Gain (+6dB) | |
DMIC Boost Capture Volume | DMIC Recording Gain (+6dB) | |
DAC1 Boost Playback Volume | DAC1 Playback Gain (+6dB) | |
DAC2 Boost Playback Volume | DAC1 Playback Gain (+6dB) | |
Speaker Boost Playback Volume | Speaker Output Gain | |
Lineout1 LR Mux | Lineout1 Left-Right Mux (Mono Stereo Control) Valid values: “Mute”, “Right Channel (L+R)”, “Left Channel (L+R)”, “Stereo” | |
Lineout2 LR Mux | Lineout2 Left-Right Mux (Mono Stereo Control) Valid values: “Mute”, “Right Channel (L+R)”, “Left Channel (L+R)”, “Stereo” | Single-ended lineout |
Speaker LR Mux | Speaker Left-Right Mux (Mono Stereo Control) Valid values: “Mute”, “Right Channel (L+R)”, “Left Channel (L+R)”, “Stereo” | |
HPF Cutoff | High-pass-filter cutoff frequency. Valid values: “3.7Hz”, “50Hz”, “100Hz”, … steps of 50Hz …, “800Hz” | |
HPF Switch | Enable/Disable High-pass-filter | |
HPF Channel | Hight-pass-filter channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” | |
Bass HPF Cutoff | Bass High-pass-filter cutoff frequency. Valid values: “158Hz”, “232Hz”, “347Hz”, “520Hz” | |
Bass Clip Level Threshold | Signal Extend Density (Clip Level). Overdrive level for bass enhancement. Valid values: “0.125”, “0.250”, “0.370”, “0.500”, “0.625”, “0.750”, “0.875” | |
Bass LPF Cutoff | Bass Low-pass-filter cutoff frequency. Valid values: “801Hz”, “1001Hz” | |
Bass Playback Switch | Enable/Disable Bass Enhancement | |
Bass Playback Volume | Bass Enhancement Gain | |
Bass Channel | Bass Enhancement Channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” | |
3D Freq | 3D Enhancement cutoff frequency (relative to the sampling rate). Valid values: “No 3D”, “0.03125 fs”, “0.04583 fs”, “0.075 fs”, “0.11458 fs”, “0.16875 fs”, “0.27083 fs” | |
3D Level | 3D Enhancement effect level. Valid values: “0%”, “6.67%”, “13.33%”, “20%”, “26.67%”, “33.33%”, “40%”, “46.67%”, “53.33%”, “60%”, “66.67%”, “73.33%”, “80%”, “86.67”, “93.33%”, “100%” | |
3D Playback Switch | Enable/Disable 3D Enhancement | |
3D Playback Volume | 3D Enhancement Gain | |
3D Channel | 3D Enhancement Channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” | |
Zero Cross Switch | Enable/Disable Zero-Cross-Detection for volume updates | |
DRC1 Channel | Dynamic Range Control 1 Channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” | |
DRC2 Channel | Dynamic Range Control 2 Channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” | |
DRC3 Channel | Dynamic Range Control 3 Channel. Valid values: “Channel1”, “Channel2”, “Channel3”, “Channel4”, “Channel5” |
The ADAU1373 features two PLLs:
enum adau1373_pll { ADAU1373_PLL1 = 0, ADAU1373_PLL2 = 1, };
Each PLLs input frequency can be selected from a variety of signals:
enum adau1373_pll_src { ADAU1373_PLL_SRC_MCLK1 = 0, ADAU1373_PLL_SRC_BCLK1 = 1, ADAU1373_PLL_SRC_BCLK2 = 2, ADAU1373_PLL_SRC_BCLK3 = 3, ADAU1373_PLL_SRC_LRCLK1 = 4, ADAU1373_PLL_SRC_LRCLK2 = 5, ADAU1373_PLL_SRC_LRCLK3 = 6, ADAU1373_PLL_SRC_GPIO1 = 7, ADAU1373_PLL_SRC_GPIO2 = 8, ADAU1373_PLL_SRC_GPIO3 = 9, ADAU1373_PLL_SRC_GPIO4 = 10, ADAU1373_PLL_SRC_MCLK2 = 11, };
The input frequency must configured to be between 7813 and 27000000 Hz. 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.
The codec driver registers three DAIs:
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 |
The DAIs can either use PLL1 or PLL2 as source. When configuring a DAI its rate should be set to the rate of the source PLL.
enum adau1373_clk_src { ADAU1373_CLK_SRC_PLL1 = 0, ADAU1373_CLK_SRC_PLL2 = 1, };
static int bfin_eval_adau1373_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; int pll_rate; 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: pll_rate = 48000 * 1024; break; case 44100: case 7350: case 11025: case 14700: case 22050: case 29400: pll_rate = 44100 * 1024; break; default: return -EINVAL; } ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); if (ret) return ret; ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, SND_SOC_CLOCK_IN); return ret; } static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int pll_rate = 48000 * 1024; int ret; ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); if (ret) return ret; ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, SND_SOC_CLOCK_IN); return ret; } static struct snd_soc_ops bfin_eval_adau1373_ops = { .hw_params = bfin_eval_adau1373_hw_params, }; static struct snd_soc_dai_link bfin_eval_adau1373_dai = { .name = "adau1373", .stream_name = "adau1373", .cpu_dai_name = "bfin-i2s.0", .codec_dai_name = "adau1373-aif1", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1373.0-001a", .ops = &bfin_eval_adau1373_ops, .init = bfin_eval_adau1373_codec_init, };
Source | Mainlined? |
---|---|
In progress |
Function | File |
---|---|
driver | sound/soc/blackfin/bfin-eval-adau1373.c |
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
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.
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:
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)
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...
root:~> tone TONE: generating sine wave at 1000 Hz...You should hear something out of the headphone Jack on the top of J8.
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.
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
root:/var> mp3play ABCOWhosOnFirstclip.mp3
root:~> mp3play http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3 http://www.radiocrazy.com/shows/A/AbbottCostello/ABCOWhosOnFirstclip.mp3: MPEG2-III (0 ms)
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.
root:~> arecord | aplay