[alsa-devel] [PATCH] ASoC: AK4671: add ak4671 codec driver

[Mark Brown]

On Thu, Sep 03, 2009 at 09:25:43PM +0900, Joonyoung Shim wrote:
> The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
> Receiver-Amplifier and Headphone-Amplifier.
> 
> The datasheet for the ak4671 can find at the following url:
> http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf
> 
> Signed-off-by: Joonyoung Shim <jy0922.shim at samsung.com>

This looks very good, everything below is relatively minor.

> +/* write to the ak4671 register space */
> +static int ak4671_write(struct snd_soc_codec *codec, unsigned int reg,
> +		unsigned int value)
> +{
> +	u8 data[2];
> +
> +	/* data is
> +	 *   D15..D8 AK4671 register offset
> +	 *   D7...D0 register data
> +	 */
> +	data[0] = reg & 0xff;
> +	data[1] = value & 0xff;
> +
> +	ak4671_write_reg_cache(codec, reg, value);
> +	if (codec->hw_write(codec->control_data, data, 2) == 2)
> +		return 0;
> +	else
> +		return -EIO;
> +}

It would be nice (but not essential) to move over to soc-cache if
possible - see the current for-2.6.32 code.

> +static const struct snd_kcontrol_new ak4671_snd_controls[] = {
> +	/* Common playback gain controls */
> +	SOC_SINGLE_TLV("Stereo Line Output1 Playback Volume",
> +			AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv),
> +	SOC_SINGLE_TLV("Headphone Output2 Playback Volume",
> +			AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv),
> +	SOC_SINGLE_TLV("Stereo Line Output3 Playback Volume",
> +			AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv),

Could drop the "Stereo" from the control names - it'll probably read
better in applications.

> +       switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> +       case SND_SOC_DAIFMT_CBM_CFM:
> +               ak4671->pll_on = 1;
> +               mode |= AK4671_M_S;
> +               break;
> +       case SND_SOC_DAIFMT_CBM_CFS:
> +               ak4671->pll_on = 1;
> +               mode &= ~(AK4671_M_S);
> +               break;
> +       default:
> +               return -EINVAL;
> +       }

Looks like pll_on is always set?  If that's true then there's not much
gain from keeping track of it.

> +/* event handlers */
> +static int ak4671_set_bias_level(struct snd_soc_codec *codec,
> +		enum snd_soc_bias_level level)
> +{
> +	struct ak4671_priv *ak4671 = codec->private_data;
> +	u8 reg;
> +
> +	switch (level) {
> +	case SND_SOC_BIAS_ON:
> +	case SND_SOC_BIAS_PREPARE:
> +	case SND_SOC_BIAS_STANDBY:
> +		if (ak4671->pll_on) {
> +			reg = ak4671_read_reg_cache(codec,
> +					AK4671_PLL_MODE_SELECT1);
> +			reg |= AK4671_PMPLL;
> +			ak4671_write(codec, AK4671_PLL_MODE_SELECT1, reg);
> +			/* pll lock time: max 40ms */
> +			mdelay(40);
> +		}

I suspect this will run into trouble with bypass paths (which do appear
to exist if I read the DAPM routes correctly).  If a bypass path is
active then the CODEC will be brought up to full bias out of sync with
any configuration of pll_on by the DAI format configuration.

I've not looked at the datasheet but I think what you need here is to
make the PLL a SND_SOC_DAPM_SUPPLY for the DACs and ADCs, then use an
event on that to turn on and off the PLL.  DAPM will then keep the PLL
powered so long as one of the DACs or ADCs is in use.

> +static struct snd_soc_dai_ops ak4671_dai_ops = {
> +	.hw_params	= ak4671_hw_params,
> +	.set_sysclk	= ak4671_set_dai_sysclk,
> +	.set_fmt	= ak4671_set_dai_fmt,
> +
> +	/* TODO */

Could just remove the comment here.

> +#ifdef CONFIG_PM
> +static int ak4671_suspend(struct platform_device *pdev, pm_message_t state)
> +{
> +	/* TODO */
> +
> +	return 0;
> +}
> +
> +static int ak4671_resume(struct platform_device *pdev)
> +{
> +	/* TODO */
> +
> +	return 0;
> +}
> +#else
> +#define ak4671_suspend		NULL
> +#define ak4671_resume		NULL
> +#endif

May as well just remove these completely if they're not implemented.