From: Mark Brown <broonie@opensource.wolfsonmicro.com>
To: johnnyhsu <johnnyhsu@realtek.com>
Cc: alsa-devel@alsa-project.org, flove <flove@realtek.com>,
Liam Girdwood <lrg@slimlogic.co.uk>
Subject: Re: Please review rt5631 driver on alsa 1.0.24 again
Date: Mon, 22 Aug 2011 23:47:58 +0100 [thread overview]
Message-ID: <20110822224757.GA31272@opensource.wolfsonmicro.com> (raw)
In-Reply-To: <348C3C33CEB74E6681F0BC4EF5B616D1@realtek.com.tw>
On Mon, Aug 22, 2011 at 11:31:30AM +0800, johnnyhsu wrote:
> +/* MIC Input Type */
> +static const char *rt5631_input_mode[] = {
> + "Single-end", "Differential"};
Single ended.
> +/* MIC Boost Volume */
> +static const char *rt5631_mic_boost[] = {"Bypass", "+20db", "+24db", "+30db",
> + "+35db", "+40db", "+44db", "+50db", "+52db"};
TLV please. I'm fairly sure I mentioned this last time.
> +/* Record Data Select */
> +static const char *rt5631_rec_sel[] = {"Disable", "MIC1", "MIC2", "Stereo"};
> +
> +static const SOC_ENUM_SINGLE_DECL(
> + rt5631_rec_sel_enum, RT5631_INT_ST_IRQ_CTRL_2,
> + RT5631_ADC_DATA_SEL_SHIFT, rt5631_rec_sel);
This looks like DAPM...
> +/* SPK Channel Volume Input Select */
> +static const char *rt5631_spkvol_sel[] = {"VMID", "SPKMIX"};
> +static const SOC_ENUM_DOUBLE_DECL(
> + rt5631_spkvol_enum, RT5631_SPK_OUT_VOL,
> + RT5631_L_EN_SHIFT, RT5631_R_EN_SHIFT, rt5631_spkvol_sel);
This too, and similarly for the other mixers.
> +static void rt5631_set_dmic_params(struct snd_soc_codec *codec,
> + struct snd_pcm_hw_params *params)
> +{
> + int rate;
> +
> + snd_soc_update_bits(codec, RT5631_GPIO_CTRL,
> + RT5631_GPIO_PIN_FUN_SEL_MASK |
> + RT5631_GPIO_DMIC_FUN_SEL_MASK,
> + RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC |
> + RT5631_GPIO_DMIC_FUN_SEL_DIMC);
Shouldn't this be done once in th eprobe function? DMICs aren't usually
hotplugged...
> + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL,
> + RT5631_DMIC_ENA_MASK, RT5631_DMIC_ENA);
This looks like DAPM.
> + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL,
> + RT5631_DMIC_L_CH_LATCH_MASK |
> + RT5631_DMIC_R_CH_LATCH_MASK,
> + RT5631_DMIC_L_CH_LATCH_FALLING |
> + RT5631_DMIC_R_CH_LATCH_RISING);
This also looks like probe work.
> + default:
> + break;
> + }
Should return an error if you don't know how to configure.
> + snd_soc_update_bits(codec, RT5631_SDP_CTRL,
> + RT5631_SDP_I2S_DL_MASK, iface);
> + if (coeff >= 0)
> + snd_soc_write(codec, RT5631_STEREO_AD_DA_CLK_CTRL,
> + coeff_div[coeff].reg_val);
What if you didn't find coefficients?
> +static int rt5631_hifi_codec_set_dai_sysclk(struct snd_soc_dai *codec_dai,
> + int clk_id, unsigned int freq, int dir)
> +{
> + struct snd_soc_codec *codec = codec_dai->codec;
> + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
> +
> + pr_info("enter %s, syclk=%d\n", __func__, freq);
> + if ((freq >= (256 * 8000)) && (freq <= (512 * 96000))) {
> + rt5631->sysclk = freq;
> + return 0;
> + }
> +
> + pr_info("unsupported sysclk freq %u for audio i2s\n", freq);
> + pr_info("set sysclk to 24.576Mhz by default\n");
> +
> + rt5631->sysclk = 24576000;
> + return 0;
No, return an error if the user specified something you don't
understand. If that's the sysclk the user can always specify it
directly.
> +static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
> + int source, unsigned int freq_in, unsigned int freq_out)
> +{
> + struct snd_soc_codec *codec = codec_dai->codec;
> + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
> + int i, ret = -EINVAL;
> +
> + printk(KERN_DEBUG "enter %s\n", __func__);
> +
> + if (!freq_in || !freq_out)
> + return 0;
Should stop the PLL if a zero output is requested.
> +/**
> + * rt5631_index_reg_show - sysfs file for dumping index registers of 2nd layer
> + */
> +#define IDX_REG_FMT "%02x: %04x\n"
> +#define IDX_REG_LEN 9
> +static ssize_t rt5631_index_reg_show(struct device *dev,
> + struct device_attribute *attr, char *buf)
You didn't answer any of my queries about this stuff on the prior
posting...
> + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
> + if (ret != 0) {
> + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
> + return ret;
> + }
> + codec->cache_bypass = 1;
Why cache_bypass?
> + /* speaker ratio is 1.44x */
> + snd_soc_write(codec, RT5631_GEN_PUR_CTRL_REG, 0x4e00);
Hrm?
parent reply other threads:[~2011-08-22 22:48 UTC|newest]
Thread overview: expand[flat|nested] mbox.gz Atom feed
[parent not found: <348C3C33CEB74E6681F0BC4EF5B616D1@realtek.com.tw>]
Reply instructions:
You may reply publicly to this message via plain-text email
using any one of the following methods:
* Save the following mbox file, import it into your mail client,
and reply-to-all from there: mbox
Avoid top-posting and favor interleaved quoting:
https://en.wikipedia.org/wiki/Posting_style#Interleaved_style
* Reply using the --to, --cc, and --in-reply-to
switches of git-send-email(1):
git send-email \
--in-reply-to=20110822224757.GA31272@opensource.wolfsonmicro.com \
--to=broonie@opensource.wolfsonmicro.com \
--cc=alsa-devel@alsa-project.org \
--cc=flove@realtek.com \
--cc=johnnyhsu@realtek.com \
--cc=lrg@slimlogic.co.uk \
/path/to/YOUR_REPLY
https://kernel.org/pub/software/scm/git/docs/git-send-email.html
* If your mail client supports setting the In-Reply-To header
via mailto: links, try the mailto: link
Be sure your reply has a Subject: header at the top and a blank line
before the message body.
This is a public inbox, see mirroring instructions
for how to clone and mirror all data and code used for this inbox