Re: [RFC PATCH] ASoC: fsl: Add Audio Mixer CPU DAI driver

Nicolin Chen
2018-12-26T17:24:22Z

Hi Viorel,

Sorry for the late response, having been on a long vacation.

The code looks pretty clean. Just some small concerns/questions below.

On Wed, Dec 19, 2018 at 12:30 AM Viorel Suman <viorel.suman@nxp.com> wrote:
>
> This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
> The Audio Mixer is a on-chip functional module that allows mixing of
> two audio streams into a single audio stream.
>
> Audio Mixer datasheet is available here:
> https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf
>
> Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
> ---
> .../devicetree/bindings/sound/fsl,amix.txt | 45 ++
> sound/soc/fsl/Kconfig | 7 +
> sound/soc/fsl/Makefile | 3 +
> sound/soc/fsl/fsl_amix.c | 554 +++++++++++++++++++++
> sound/soc/fsl/fsl_amix.h | 101 ++++

I aimn't against the naming here, but it seems to be AUDMIX in RM?

Would it be better to align with that? It's your decision though.

> diff --git a/Documentation/devicetree/bindings/sound/fsl,amix.txt b/Documentation/devicetree/bindings/sound/fsl,amix.txt
> +=================================
> + - compatible : Compatible list, contains "fsl,imx8qm-amix"
> +
> + - reg : Offset and length of the register set for the device.
> +
> + - clocks : Must contain an entry for each entry in clock-names.
> +
> + - clock-names : Must include the "ipg" for register access.
> +
> + - power-domains : Must contain the phandle to the AMIX power domain node
> +
> +Device driver configuration example:
> +======================================
> + amix: amix@59840000 {
> + compatible = "fsl,imx8qm-amix";
> + reg = <0x0 0x59840000 0x0 0x10000>;
> + clocks = <&clk IMX8QXP_AUD_AMIX_IPG>;
> + clock-names = "ipg";
> + power-domains = <&pd_amix>;
> + };

>From the description of DT and RM, I don't see how it connects to SAIs.

Are they fixed to SAI0 and SAI1 in imx8qm? Wondering if it'd be better
to have such information in the doc.

> diff --git a/sound/soc/fsl/fsl_amix.c b/sound/soc/fsl/fsl_amix.c

+static const char
+ *width_sel[] = { "16b", "18b", "20b", "24b", "32b", },
+ *pol_sel[] = { "Positive edge", "Negative edge", },
[...]
+static const struct soc_enum fsl_amix_enum[] = {
+/* FSL_AMIX_CTR enums */
[...]
+SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTWIDTH_SHIFT, width_sel),
+SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTCKPOL_SHIFT, pol_sel),

Should we handle the width in hw_param()?

Why do we change pol here? It feels like against set_fmt().

> +static int fsl_amix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
> +{
> + /* For playback the AMIX is slave, and for record is master */
> + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> + case SND_SOC_DAIFMT_CBM_CFM:
> + case SND_SOC_DAIFMT_CBS_CFS:

So it's used either for playback or capture only, not both at same time?

Thanks
Nicolin