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 --- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++++++ 4 files changed, 688 insertions(+) create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7b1d997..0af2e056 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically.

+config SND_SOC_FSL_AUDMIX + tristate "Audio Mixer (AUDMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AUDMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 3c0ff31..4172d5a 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o

# Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-audmix-objs := fsl_audmix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -22,6 +23,8 @@ snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-micfil-objs := fsl_micfil.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c new file mode 100644 index 0000000..3356cb6 --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.c @@ -0,0 +1,576 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "fsl_audmix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_audmix_enum[] = { +/* FSL_AUDMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AUDMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 1, updn_sel), +/* FSL_AUDMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_audmix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_audmix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_audmix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_audmix_state prm) +{ + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK) + >> FSL_AUDMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_audmix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AUDMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AUDMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { + /* FSL_AUDMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], + snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], + snd_soc_get_enum_double, fsl_audmix_put_out_src), + SOC_ENUM("Output Width", fsl_audmix_enum[2]), + SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), + SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), + SOC_ENUM("Sync Mode Config", fsl_audmix_enum[5]), + SOC_ENUM("Sync Mode Clk Source", fsl_audmix_enum[6]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_audmix_enum[7]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_audmix_enum[8]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AUDMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AUDMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_audmix_enum[9]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_audmix_enum[10]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AUDMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AUDMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AUDMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AUDMIX 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: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AUDMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_audmix_dai_ops = { + .set_fmt = fsl_audmix_dai_set_fmt, + .trigger = fsl_audmix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_audmix_dai[] = { + { + .id = 0, + .name = "audmix-0", + .playback = { + .stream_name = "AUDMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, + { + .id = 1, + .name = "audmix-1", + .playback = { + .stream_name = "AUDMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_audmix_component = { + .name = "fsl-audmix-dai", + .controls = fsl_audmix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_audmix_snd_controls), +}; + +static bool fsl_audmix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_STR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATTNVAL0: + case FSL_AUDMIX_ATSTP0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + case FSL_AUDMIX_ATTNVAL1: + case FSL_AUDMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_audmix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_audmix_reg[] = { + { FSL_AUDMIX_CTR, 0x00060 }, + { FSL_AUDMIX_STR, 0x00003 }, + { FSL_AUDMIX_ATCR0, 0x00000 }, + { FSL_AUDMIX_ATIVAL0, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN0, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT0, 0x00010 }, + { FSL_AUDMIX_ATTNVAL0, 0x00000 }, + { FSL_AUDMIX_ATSTP0, 0x00000 }, + { FSL_AUDMIX_ATCR1, 0x00000 }, + { FSL_AUDMIX_ATIVAL1, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN1, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT1, 0x00010 }, + { FSL_AUDMIX_ATTNVAL1, 0x00000 }, + { FSL_AUDMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_audmix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AUDMIX_ATSTP1, + .reg_defaults = fsl_audmix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_audmix_reg), + .readable_reg = fsl_audmix_readable_reg, + .writeable_reg = fsl_audmix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int fsl_audmix_probe(struct platform_device *pdev) +{ + struct fsl_audmix *priv; + struct resource *res; + void __iomem *regs; + int ret; + const char *sprop; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + &fsl_audmix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_audmix_component, + fsl_audmix_dai, + ARRAY_SIZE(fsl_audmix_dai)); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + sprop = of_get_property(pdev->dev.of_node, "model", NULL); + if (sprop) { + priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0, + NULL, 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(&pdev->dev, + "failed to register platform %s: %d\n", sprop, + ret); + } + } else { + dev_err(&pdev->dev, "[model] attribute missing.\n"); + ret = -EINVAL; + } + + return ret; +} + +static int fsl_audmix_remove(struct platform_device *pdev) +{ + struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev); + + if (priv->pdev) + platform_device_unregister(priv->pdev); + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_audmix_runtime_resume(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_audmix_runtime_suspend(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_audmix_pm = { + SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, + fsl_audmix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_audmix_ids[] = { + { .compatible = "fsl,imx8qm-audmix", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + +static struct platform_driver fsl_audmix_driver = { + .probe = fsl_audmix_probe, + .remove = fsl_audmix_remove, + .driver = { + .name = "fsl-audmix", + .of_match_table = fsl_audmix_ids, + .pm = &fsl_audmix_pm, + }, +}; +module_platform_driver(fsl_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman viorel.suman@nxp.com"); +MODULE_ALIAS("platform:fsl-audmix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h new file mode 100644 index 0000000..7812ffe --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.h @@ -0,0 +1,102 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AUDMIX_H +#define __FSL_AUDMIX_H + +#define FSL_AUDMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AUDMIX Registers */ +#define FSL_AUDMIX_CTR 0x200 /* Control */ +#define FSL_AUDMIX_STR 0x204 /* Status */ + +#define FSL_AUDMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AUDMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AUDMIX Control Register */ +#define FSL_AUDMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AUDMIX_CTR_MIXCLK_MASK BIT(FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_MIXCLK(i) ((i) << FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AUDMIX_CTR_OUTSRC_MASK (0x3 << FSL_AUDMIX_CTR_OUTSRC_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC(i) (((i) << FSL_AUDMIX_CTR_OUTSRC_SHIFT)\ + & FSL_AUDMIX_CTR_OUTSRC_MASK) +#define FSL_AUDMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AUDMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AUDMIX_CTR_OUTWIDTH(i) (((i) << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)\ + & FSL_AUDMIX_CTR_OUTWIDTH_MASK) +#define FSL_AUDMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AUDMIX_CTR_OUTCKPOL_MASK BIT(FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_OUTCKPOL(i) ((i) << FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AUDMIX_CTR_MASKRTDF_MASK BIT(FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF(i) ((i) << FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AUDMIX_CTR_MASKCKDF_MASK BIT(FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF(i) ((i) << FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AUDMIX_CTR_SYNCMODE_MASK BIT(FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE(i) ((i) << FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AUDMIX_CTR_SYNCSRC_MASK BIT(FSL_AUDMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC(i) ((i) << FSL_AUDMIX_CTR_SYNCSRC_SHIFT) + +/* AUDMIX Status Register */ +#define FSL_AUDMIX_STR_RATEDIFF BIT(0) +#define FSL_AUDMIX_STR_CLKDIFF BIT(1) +#define FSL_AUDMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AUDMIX_STR_MIXSTAT_MASK (0x3 << FSL_AUDMIX_STR_MIXSTAT_SHIFT) +#define FSL_AUDMIX_STR_MIXSTAT(i) (((i) & FSL_AUDMIX_STR_MIXSTAT_MASK) \ + >> FSL_AUDMIX_STR_MIXSTAT_SHIFT) +/* AUDMIX Attenuation Control Register */ +#define FSL_AUDMIX_ATCR_AT_EN BIT(0) +#define FSL_AUDMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AUDMIX_ATCR_ATSTPDFI_MASK \ + (0xfff << FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT) + +/* AUDMIX Attenuation Initial Value Register */ +#define FSL_AUDMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Up Factor Register */ +#define FSL_AUDMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Down Factor Register */ +#define FSL_AUDMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Target Register */ +#define FSL_AUDMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AUDMIX Attenuation Value Register */ +#define FSL_AUDMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Number Register */ +#define FSL_AUDMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AUDMIX_MAX_DAIS 2 +struct fsl_audmix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AUDMIX_H */

Add the DT binding documentation for NXP Audio Mixer CPU DAI driver.

Signed-off-by: Viorel Suman viorel.suman@nxp.com --- .../devicetree/bindings/sound/fsl,audmix.txt | 54 ++++++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt

diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt new file mode 100644 index 0000000..45f807e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt @@ -0,0 +1,54 @@ +NXP Audio Mixer (AUDMIX). + +The Audio Mixer is a on-chip functional module that allows mixing of two +audio streams into a single audio stream. Audio Mixer has two input serial +audio interfaces. These are driven by two Synchronous Audio interface +modules (SAI). Each input serial interface carries 8 audio channels in its +frame in TDM manner. Mixer mixes audio samples of corresponding channels +from two interfaces into a single sample. Before mixing, audio samples of +two inputs can be attenuated based on configuration. The output of the +Audio Mixer is also a serial audio interface. Like input interfaces it has +the same TDM frame format. This output is used to drive the serial DAC TDM +interface of audio codec and also sent to the external pins along with the +receive path of normal audio SAI module for readback by the CPU. + +The output of Audio Mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - mixed audio + +Mixing operation is independent of audio sample rate but the two audio +input streams must have same audio sample rate with same number of channels +in TDM frame to be eligible for mixing. + +Device driver required properties: +================================= + - compatible : Compatible list, contains "fsl,imx8qm-audmix" + + - 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 AUDMIX power domain node + + - dais : Must contain a list of phandles to AUDMIX connected + DAIs. The current implementation requires two phandles + to SAI interfaces to be provided, the first SAI in the + list being used to route the AUDMIX output. + + - model : Must contain machine driver name which will configure + and instantiate the appropriate audio card. + +Device driver configuration example: +====================================== + audmix: audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x0 0x59840000 0x0 0x10000>; + clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + model = "imx-audmix"; + };

This patch implements Audio Mixer machine driver for NXP iMX8 SOCs. It connects together Audio Mixer and related SAI instances.

Signed-off-by: Viorel Suman viorel.suman@nxp.com --- sound/soc/fsl/Kconfig | 9 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 338 insertions(+) create mode 100644 sound/soc/fsl/imx-audmix.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0af2e056..d87c842 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -303,6 +303,15 @@ config SND_SOC_FSL_ASOC_CARD CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card.

+config SND_SOC_IMX_AUDMIX + tristate "SoC Audio support for i.MX boards with AUDMIX" + select SND_SOC_FSL_AUDMIX + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with Audio Mixer + Say Y if you want to add support for SoC audio on an i.MX board with + an Audio Mixer. + endif # SND_IMX_SOC

endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4172d5a..c0dd044 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -62,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o +snd-soc-imx-audmix-objs := imx-audmix.o

obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o +obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c new file mode 100644 index 0000000..72e37ca --- /dev/null +++ b/sound/soc/fsl/imx-audmix.c @@ -0,0 +1,327 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright 2017 NXP + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/pm_runtime.h> +#include "fsl_sai.h" +#include "fsl_audmix.h" + +struct imx_audmix { + struct platform_device *pdev; + struct snd_soc_card card; + struct platform_device *audmix_pdev; + struct platform_device *out_pdev; + struct clk *cpu_mclk; + int num_dai; + struct snd_soc_dai_link *dai; + int num_dai_conf; + struct snd_soc_codec_conf *dai_conf; + int num_dapm_routes; + struct snd_soc_dapm_route *dapm_routes; +}; + +static const u32 imx_audmix_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { + .count = ARRAY_SIZE(imx_audmix_rates), + .list = imx_audmix_rates, +}; + +static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->card->dev; + unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); + int ret; + + if (clk_rate % 24576000 == 0) { + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_audmix_rate_constraints); + if (ret < 0) + return ret; + } else { + dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); + } + + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 8); + if (ret < 0) + return ret; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + FSL_AUDMIX_FORMATS); +} + +static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + u32 channels = params_channels(params); + int ret, dir; + + /* For playback the AUDMIX is slave, and for record is master */ + fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + + /* set DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + /* + * Per datasheet, AUDMIX expects 8 slots and 32 bits + * for every slot in TDM mode. + */ + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + BIT(channels) - 1, 8, 32); + if (ret) + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + + return ret; +} + +static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + int ret; + + if (!tx) + return 0; + + /* For playback the AUDMIX is slave */ + fmt |= SND_SOC_DAIFMT_CBM_CFM; + + /* set AUDMIX DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) + dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops imx_audmix_fe_ops = { + .startup = imx_audmix_fe_startup, + .hw_params = imx_audmix_fe_hw_params, +}; + +static struct snd_soc_ops imx_audmix_be_ops = { + .hw_params = imx_audmix_be_hw_params, +}; + +static int imx_audmix_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *audmix_np = NULL, *out_cpu_np = NULL; + struct platform_device *audmix_pdev = NULL; + struct platform_device *cpu_pdev; + struct of_phandle_args args; + struct imx_audmix *priv; + int i, num_dai, ret; + const char *fe_name_pref = "HiFi-AUDMIX-FE-"; + char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name; + + if (pdev->dev.parent) { + audmix_np = pdev->dev.parent->of_node; + } else { + dev_err(&pdev->dev, "Missing parent device.\n"); + return -EINVAL; + } + + if (!audmix_np) { + dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + return -EINVAL; + } + + audmix_pdev = of_find_device_by_node(audmix_np); + if (!audmix_pdev) { + dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n", + np->full_name); + return -EINVAL; + } + + num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); + if (num_dai != FSL_AUDMIX_MAX_DAIS) { + dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n", + audmix_np->full_name); + return -EINVAL; + } + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->num_dai = 2 * num_dai; + priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai * + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + if (!priv->dai) + return -ENOMEM; + + priv->num_dai_conf = num_dai; + priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf * + sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!priv->dai_conf) + return -ENOMEM; + + priv->num_dapm_routes = 3 * num_dai; + priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes * + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!priv->dapm_routes) + return -ENOMEM; + + for (i = 0; i < num_dai; i++) { + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, + &args); + if (ret < 0) { + dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); + return ret; + } + + cpu_pdev = of_find_device_by_node(args.np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + + dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", + fe_name_pref, args.np->full_name + 1); + + dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); + + if (i == 0) { + out_cpu_np = args.np; + capture_dai_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Capture"); + } + + priv->dai[i].name = dai_name; + priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; + priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codec_name = "snd-soc-dummy"; + priv->dai[i].cpu_of_node = args.np; + priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platform_of_node = args.np; + priv->dai[i].dynamic = 1; + priv->dai[i].dpcm_playback = 1; + priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); + priv->dai[i].ignore_pmdown_time = 1; + priv->dai[i].ops = &imx_audmix_fe_ops; + + /* Add AUDMIX Backend */ + be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "audmix-%d", i); + be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Playback-%d", i); + be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Capture-%d", i); + + priv->dai[num_dai + i].name = be_name; + priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpu_of_node = audmix_np; + priv->dai[num_dai + i].cpu_dai_name = be_name; + priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].no_pcm = 1; + priv->dai[num_dai + i].dpcm_playback = 1; + priv->dai[num_dai + i].dpcm_capture = 1; + priv->dai[num_dai + i].ignore_pmdown_time = 1; + priv->dai[num_dai + i].ops = &imx_audmix_be_ops; + + priv->dai_conf[i].of_node = args.np; + priv->dai_conf[i].name_prefix = dai_name; + + priv->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Playback"); + priv->dapm_routes[i].sink = be_pb; + priv->dapm_routes[num_dai + i].source = be_pb; + priv->dapm_routes[num_dai + i].sink = be_cp; + priv->dapm_routes[2 * num_dai + i].source = be_cp; + priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name; + } + + cpu_pdev = of_find_device_by_node(out_cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); + if (IS_ERR(priv->cpu_mclk)) { + ret = PTR_ERR(priv->cpu_mclk); + dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); + return -EINVAL; + } + + priv->audmix_pdev = audmix_pdev; + priv->out_pdev = cpu_pdev; + + priv->card.dai_link = priv->dai; + priv->card.num_links = priv->num_dai; + priv->card.codec_conf = priv->dai_conf; + priv->card.num_configs = priv->num_dai_conf; + priv->card.dapm_routes = priv->dapm_routes; + priv->card.num_dapm_routes = priv->num_dapm_routes; + priv->card.dev = pdev->dev.parent; + priv->card.owner = THIS_MODULE; + priv->card.name = "imx-audmix"; + + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + return ret; + } + + return ret; +} + +static struct platform_driver imx_audmix_driver = { + .probe = imx_audmix_probe, + .driver = { + .name = "imx-audmix", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(imx_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver"); +MODULE_AUTHOR("Viorel Suman viorel.suman@nxp.com"); +MODULE_ALIAS("platform:imx-audmix"); +MODULE_LICENSE("GPL v2");