• SSM2602
• SSM2603
• SSM2604

• SSM2603-EVALZ
• SSM2604-EVALZ

• CN0282

SPI can be used for the SSM2602 if in SPI mode (MODE pin set to 1).

static struct spi_board_info board_spi_board_info[] __initdata = {
	[--snip--]
	{
		.modalias = "ssm2602",
		.max_speed_hz = 25000000,     /* max spi clock (SCK) speed in HZ */
		.bus_num = 0,
		.chip_select = GPIO_PF10 + MAX_CTRL_CS, /* 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);

I2C can be used for the SSM2602, SSM2603, SSM2604. For the SSM2602 make sure that it is in I2C mode (MODE pin set to 0).

static struct i2c_board_info __initdata bfin_i2c_board_info[] = {
 
	[--snip--]
	{
		I2C_BOARD_INFO("ssm2604", 0x1b),
	},
	[--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);

The codec driver registers one DAI named “ssm2602-hifi”.

The codecs system clock can be configured for various input rates. When configuring the codec system clock use SSM2602_SYSCLK for the clock id.

The following list contains the supported system clock rates and their resulting sample-rates.

static struct snd_soc_card bf5xx_ssm2602;
 
static int bf5xx_ssm2602_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 *codec_dai = rtd->codec_dai;
	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
	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 < 0)
		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 < 0)
		return ret;
 
	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, 12000000,
		SND_SOC_CLOCK_IN);
	if (ret < 0)
		return ret;
 
	return 0;
}
 
static struct snd_soc_ops bf5xx_ssm2602_ops = {
	.hw_params = bf5xx_ssm2602_hw_params,
};
 
static struct snd_soc_dai_link bf5xx_ssm2602_dai_link[] = {
	{
		.name = "ssm2602",
		.stream_name = "SSM2602",
		.cpu_dai_name = "bfin-i2s.0",
		.codec_dai_name = "ssm2602-hifi",
		.platform_name = "bfin-i2s-pcm-audio",
		.codec_name = "ssm2602.0-001b",
		.ops = &bf5xx_ssm2602_ops,
	},
};

To add support for the built-in codec SSM2602 of BF52xC 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 card support --->
         Advanced Linux Sound Architecture  --->
          < > Sequencer support
           OSS Mixer API 
           OSS PCM (digital audio) API
           ALSA for SoC audio support  --->
             SoC I2S Audio for the ADI BF5xx chip 
               SoC SSM2602 Audio support for BF52x ezkit
            < > 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:

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:15, regs:ffc00800
   ssm2602 Audio Codec 0.1<6>dma_alloc_init: dma_page @ 0x03011000 - 512 pages at 0x03e00000
   asoc: SSM2602 <-> bf5xx-i2s-0 mapping ok
   ALSA device list:
     #0: bf5xx_ssm2602 (SSM2602)

root:~> modprobe snd-ssm2602
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_ssm2602             1412  0 
snd_soc_ssm2602         8528  1 snd_ssm2602
snd_soc_bf5xx           2784  1 snd_ssm2602
snd_soc_bf5xx_i2s      10916  2 snd_ssm2602,snd_soc_bf5xx
snd_soc_core           17120  3 snd_ssm2602,snd_soc_ssm2602,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_ssm2602,snd_soc_core,snd_pcm,snd_timer
soundcore               3940  1 snd

root:~> tone
TONE: generating sine wave at 1000 Hz...

1. 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.
2. Set the audio mixer to Mic (the default is Line, assuming you have built ALSA utils):

   root:/> amixer sset 'Input Mux' 'Mic'
   Simple mixer control 'Input Mux',0
     Capabilities: enum
     Items: 'Line' 'Mic'
     Item0: 'Mic'

   Also you can run “alsamixer” to get graphic configuration interface, OSS-based “mixer” can work too.
3. 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.

      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

4. 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)
   

5. Optionally check to make sure the Line and headphone are working properly:

   root:/> amixer sset 'Input Mux' 'Line'
   Simple mixer control 'Input Mux',0
     Capabilities: enum
     Items: 'Line' 'Mic'
     Item0: 'Line'
   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.
6. You should also be able to do a “talkthrough”, and hear on the speakers anything you put on the line.

   root:~> arecord | aplay