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@xxxxxxx> 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@xxxxxxx> > --- > .../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 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
