Re: [PATCH 07/13] ASoC: codecs: add AD24xx codec driver

Mark Brown
2024-05-17T15:03:58Z

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On Fri, May 17, 2024 at 02:58:05PM +0200, Alvin Šipraga wrote:

> +++ b/sound/soc/codecs/ad24xx-codec.c
> @@ -0,0 +1,665 @@
> +// SPDX-License-Identifier: GPL-2.0-only
> +/*
> + * AD24xx codec driver

Please make the whole comment a C++ comment.

> +static const char *const ad24xx_codec_slot_size_text[] = {
> + "8 bits", "12 bits", "16 bits", "20 bits",
> + "24 bits", "28 bits", "32 bits",
> +};

Why is this configured by the user rather than via set_tdm_slot(), and
how would one usefully use this at runtime?

> +static int ad24xx_codec_slot_config_put(struct snd_kcontrol *kcontrol,
> + struct snd_ctl_elem_value *ucontrol)
> +{

> + } else if (priv == &ad24xx_codec_up_slot_format_enum ||
> + priv == &ad24xx_codec_dn_slot_format_enum) {
> + if (val >= ARRAY_SIZE(ad24xx_codec_slot_format_text))
> + return -EINVAL;
> + slot_config->format[direction] = val;
> + } else
> + return -ENOENT;

If one side has {} both sides should, see coding-style.rst.

> +
> + return 0;
> +}

This won't flag changes by returning 1 which will mean no events are
generated and break some UIs. Please show the output of the mixer-test
selftest on new submissions, it will check for this and other issues.

> + /* Main node must be BCLK/FSYNC consumer, subordinate node provider */
> + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) !=
> + (is_a2b_main(adc->node) ? SND_SOC_DAIFMT_CBC_CFC :
> + SND_SOC_DAIFMT_CBP_CFP))
> + return -EINVAL;

Please don't use the ternery operator like this, it just makes things
harder to read.

> + val = bclk_invert ? A2B_I2SCFG_RXBCLKINV_MASK :
> + A2B_I2SCFG_TXBCLKINV_MASK;

Similarly, please use normal conditional statements.

> +static int ad24xx_codec_hw_params(struct snd_pcm_substream *substream,
> + struct snd_pcm_hw_params *params,
> + struct snd_soc_dai *dai)

> +
> + /* Finally, request slots */
> + ret = a2b_node_request_slots(adc->node, &slot_req);
> + if (ret)
> + return ret;

Note that hw_params() can be called multiple times before starting the
audio stream, will this leak?

> + struct snd_soc_dai *dai)
> +{
> + struct snd_soc_component *component = dai->component;
> + struct ad24xx_codec *adc = snd_soc_component_get_drvdata(component);
> + int ret;
> +
> + ret = a2b_node_free_slots(adc->node);
> + if (ret)
> + return ret;

What if we close without having called hw_params()?

> +static const struct snd_soc_dai_driver ad24xx_codec_dai_drv[] = {
> + [AD24XX_DAI_I2S] = {
> + .name = "ad24xx-i2s",
> + .playback = {
> + .stream_name = "I2S Playback",
> + .channels_min = 1,
> + .channels_max = 32,
> + },
> + .capture = {
> + .stream_name = "I2S Capture",
> + .channels_min = 1,
> + .channels_max = 32,
> + },
> + .ops = &ad24xx_codec_dai_ops,
> + .symmetric_rate = 1,
> + },
> +};

Why is this an array?

> +static const struct regmap_config ad24xx_codec_regmap_config = {
> + .reg_bits = 8,
> + .val_bits = 8,
> + .cache_type = REGCACHE_RBTREE,
> +};

New code should use _MAPLE unless there's a strong reason to use
something else.

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