// SPDX-License-Identifier: GPL-2.0 // // Freescale Generic ASoC Sound Card driver with ASRC // // Copyright (C) 2014 Freescale Semiconductor, Inc. // // Author: Nicolin Chen #include #include #include #include #if IS_ENABLED(CONFIG_SND_AC97_CODEC) #include #endif #include #include #include #include #include "fsl_esai.h" #include "fsl_sai.h" #include "imx-audmux.h" #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" #include "../codecs/tlv320aic31xx.h" #include "../codecs/nau8822.h" #include "../codecs/wm8904.h" #define DRIVER_NAME "fsl-asoc-card" #define CS427x_SYSCLK_MCLK 0 #define RX 0 #define TX 1 /* Default DAI format without Master and Slave flag */ #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) static const u32 cs42888_rates_48k[] = { 48000, 96000, 192000, }; static const u32 cs42888_rates_44k[] = { 44100, 88200, 176400, }; static const u32 cs42888_channels[] = { 1, 2, 4, 6, 8, }; static const struct snd_pcm_hw_constraint_list cs42888_rate_48k_constraints = { .list = cs42888_rates_48k, .count = ARRAY_SIZE(cs42888_rates_48k), }; static const struct snd_pcm_hw_constraint_list cs42888_rate_44k_constraints = { .list = cs42888_rates_44k, .count = ARRAY_SIZE(cs42888_rates_44k), }; static const struct snd_pcm_hw_constraint_list cs42888_channel_constraints = { .list = cs42888_channels, .count = ARRAY_SIZE(cs42888_channels), }; /** * struct codec_priv - CODEC private data * @mclk_freq: Clock rate of MCLK * @free_freq: Clock rate of MCLK for hw_free() * @mclk_id: MCLK (or main clock) id for set_sysclk() * @fll_id: FLL (or secordary clock) id for set_sysclk() * @pll_id: PLL id for set_pll() * @pll_ratio_s24: PLL output ratio for S24_LE format (PLL_freq = sample_rate × ratio) * Default is 384, but some codecs (e.g., WM8904) require lower values * to stay within PLL frequency limits */ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; int fll_id; int pll_id; int pll_ratio_s24; }; /** * struct cpu_priv - CPU private data * @sysclk_freq: SYSCLK rates for set_sysclk() * @sysclk_dir: SYSCLK directions for set_sysclk() * @sysclk_id: SYSCLK ids for set_sysclk() * @sysclk_ratio: SYSCLK ratio on sample rate * @slot_width: Slot width of each frame * @slot_num: Number of slots of each frame * * Note: [1] for tx and [0] for rx */ struct cpu_priv { unsigned long sysclk_freq[2]; u32 sysclk_dir[2]; u32 sysclk_id[2]; u32 sysclk_ratio[2]; u32 slot_width; u32 slot_num; }; struct fsl_asoc_card_priv; /* * struct fsl_asoc_card_pdata - per-compatible static card description * @sysclk_dir: initial CPU SYSCLK direction override (0 = leave default IN) * @sysclk_ratio: SYSCLK ratio on sample rate (0 = not used) * @slot_width: TDM slot width (0 = not TDM) * @codec_dai_name: name of the codec DAI * @codec_mclk_id: MCLK id passed to set_sysclk() for the codec * @codec_fll_id: FLL id; only valid when has_pll is true * @codec_pll_id: PLL id; only valid when has_pll is true * @codec_pll_ratio_s24: PLL output ratio for S24_LE * @has_pll: codec uses PLL/FLL; codec_fll_id and codec_pll_id are valid * @dai_fmt: DAI format flags * @playback_only: restrict card to playback direction * @capture_only: restrict card to capture direction * @dapm_routes: DAPM route table override * @num_dapm_routes: number of entries in dapm_routes * @exclude_format: PCM format bitmask excluded (for SAI + WM8960/WM8962) * @codec_init: codec-specific init run after mclk_freq is populated * @probe_init: optional DT-driven init run at end of probe() (e.g. SPDIF codec discovery) */ struct fsl_asoc_card_pdata { u32 sysclk_dir[2]; u32 sysclk_ratio[2]; u32 slot_width; const char *codec_dai_name; u32 codec_mclk_id; int codec_fll_id; int codec_pll_id; int codec_pll_ratio_s24; bool has_pll; bool playback_only; bool capture_only; u32 dai_fmt; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; u64 exclude_format; int (*codec_init)(struct fsl_asoc_card_priv *priv); int (*probe_init)(struct device_node *codec_np[], struct device_node *cpu_np, const char *codec_dai_name[], struct fsl_asoc_card_priv *priv); }; /** * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data * @dai_link: DAI link structure including normal one and DPCM link * @hp_jack: Headphone Jack structure * @mic_jack: Microphone Jack structure * @pdev: platform device pointer * @pdata: pointer to the per-compatible card platform data * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure * @constraint_rates: array of supported rates * @constraint_channels: array of supported channels * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends * @asrc_format: ASRC sample format used by Back-Ends * @dai_fmt: DAI format between CPU and CODEC * @exclude_format: excluded format; * @name: Card name */ struct fsl_asoc_card_priv { struct snd_soc_dai_link dai_link[3]; struct simple_util_jack hp_jack; struct simple_util_jack mic_jack; struct platform_device *pdev; const struct fsl_asoc_card_pdata *pdata; struct codec_priv codec_priv[2]; struct cpu_priv cpu_priv; struct snd_soc_card card; const struct snd_pcm_hw_constraint_list *constraint_rates; const struct snd_pcm_hw_constraint_list *constraint_channels; u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; snd_pcm_format_t asrc_format; u32 dai_fmt; u64 exclude_format; char name[32]; }; /* * This dapm route map exists for DPCM link only. * The other routes shall go through Device Tree. * * Note: keep all ASRC routes in the second half * to drop them easily for non-ASRC cases. */ static const struct snd_soc_dapm_route audio_map[] = { /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "CPU-Playback"}, {"CPU-Capture", NULL, "Capture"}, /* 2nd half -- ASRC DAPM routes */ {"CPU-Playback", NULL, "ASRC-Playback"}, {"ASRC-Capture", NULL, "CPU-Capture"}, }; static const struct snd_soc_dapm_route audio_map_ac97[] = { /* 1st half -- Normal DAPM routes */ {"AC97 Playback", NULL, "CPU AC97 Playback"}, {"CPU AC97 Capture", NULL, "AC97 Capture"}, /* 2nd half -- ASRC DAPM routes */ {"CPU AC97 Playback", NULL, "ASRC-Playback"}, {"ASRC-Capture", NULL, "CPU AC97 Capture"}, }; static const struct snd_soc_dapm_route audio_map_tx[] = { /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "CPU-Playback"}, /* 2nd half -- ASRC DAPM routes */ {"CPU-Playback", NULL, "ASRC-Playback"}, }; static const struct snd_soc_dapm_route audio_map_rx[] = { /* 1st half -- Normal DAPM routes */ {"CPU-Capture", NULL, "Capture"}, /* 2nd half -- ASRC DAPM routes */ {"ASRC-Capture", NULL, "CPU-Capture"}, }; /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), SND_SOC_DAPM_LINE("Line In Jack", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), }; static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) { return priv->dai_fmt == SND_SOC_DAIFMT_AC97; } static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct codec_priv *codec_priv; struct snd_soc_dai *codec_dai; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; unsigned int pll_out, sysclk_freq; int codec_idx; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); priv->streams |= BIT(substream->stream); if (fsl_asoc_card_is_ac97(priv)) return 0; if (!cpu_priv->sysclk_freq[tx] && cpu_priv->sysclk_ratio[tx]) sysclk_freq = priv->sample_rate * cpu_priv->sysclk_ratio[tx]; else sysclk_freq = cpu_priv->sysclk_freq[tx]; /* Specific configurations of DAIs starts from here */ ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx], sysclk_freq, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); goto fail; } if (cpu_priv->slot_width) { if (!cpu_priv->slot_num) cpu_priv->slot_num = 2; ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_cpu(rtd, 0), 0x3, 0x3, cpu_priv->slot_num, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); goto fail; } } /* Specific configuration for PLL */ for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { codec_priv = &priv->codec_priv[codec_idx]; if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * codec_priv->pll_ratio_s24; else pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, codec_priv->mclk_id, codec_priv->mclk_freq, pll_out); if (ret) { dev_err(dev, "failed to start FLL: %d\n", ret); goto fail; } ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, pll_out, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set SYSCLK: %d\n", ret); goto fail; } } } return 0; fail: priv->streams &= ~BIT(substream->stream); return ret; } static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct codec_priv *codec_priv; struct snd_soc_dai *codec_dai; struct device *dev = rtd->card->dev; int codec_idx; int ret; priv->streams &= ~BIT(substream->stream); for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { codec_priv = &priv->codec_priv[codec_idx]; if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->free_freq, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "failed to switch away from FLL: %d\n", ret); return ret; } ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } } return 0; } static int fsl_asoc_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_pcm_runtime *runtime = substream->runtime; int ret; if (priv->exclude_format && !rtd->dai_link->no_pcm) { ret = snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, ~priv->exclude_format); if (ret) return ret; } if (priv->constraint_channels) { ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, priv->constraint_channels); if (ret) return ret; } /* * Apply rate constraints only to frontend DAI links (no_pcm = 0). * Skip DPCM backend (no_pcm = 1) as rate is fixed by be_hw_params_fixup() * and ASRC frontend handles rate conversion. */ if (priv->constraint_rates && !rtd->dai_link->no_pcm) { ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->constraint_rates); if (ret) return ret; } return 0; } static const struct snd_soc_ops fsl_asoc_card_ops = { .startup = fsl_asoc_card_startup, .hw_params = fsl_asoc_card_hw_params, .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_interval *rate; struct snd_mask *mask; rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); rate->max = rate->min = priv->asrc_rate; mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); snd_mask_set_format(mask, priv->asrc_format); return 0; } static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", .stream_name = "HiFi", .ops = &fsl_asoc_card_ops, }, /* DPCM Link between Front-End and Back-End (Optional) */ { .name = "HiFi-ASRC-FE", .stream_name = "HiFi-ASRC-FE", .dynamic = 1, }, { .name = "HiFi-ASRC-BE", .stream_name = "HiFi-ASRC-BE", .be_hw_params_fixup = be_hw_params_fixup, .ops = &fsl_asoc_card_ops, .no_pcm = 1, }, }; static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { struct device *dev = &priv->pdev->dev; u32 int_ptcr = 0, ext_ptcr = 0; int int_port, ext_port; int ret; ret = of_property_read_u32(np, "mux-int-port", &int_port); if (ret) { dev_err(dev, "mux-int-port missing or invalid\n"); return ret; } ret = of_property_read_u32(np, "mux-ext-port", &ext_port); if (ret) { dev_err(dev, "mux-ext-port missing or invalid\n"); return ret; } /* * The port numbering in the hardware manual starts at 1, while * the AUDMUX API expects it starts at 0. */ int_port--; ext_port--; /* * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBP_CFP: int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_RFSDIR | IMX_AUDMUX_V2_PTCR_RCLKDIR | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; break; case SND_SOC_DAIFMT_CBP_CFC: int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_RCLKDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | IMX_AUDMUX_V2_PTCR_RFSDIR | IMX_AUDMUX_V2_PTCR_TFSDIR; break; case SND_SOC_DAIFMT_CBC_CFP: int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | IMX_AUDMUX_V2_PTCR_RFSDIR | IMX_AUDMUX_V2_PTCR_TFSDIR; ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | IMX_AUDMUX_V2_PTCR_RCLKDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; break; case SND_SOC_DAIFMT_CBC_CFC: ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | IMX_AUDMUX_V2_PTCR_RFSDIR | IMX_AUDMUX_V2_PTCR_RCLKDIR | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: if (!fsl_asoc_card_is_ac97(priv)) return -EINVAL; } if (fsl_asoc_card_is_ac97(priv)) { int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TCLKDIR; ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ if (!fsl_asoc_card_is_ac97(priv)) { unsigned int pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); ret = imx_audmux_v2_configure_port(int_port, 0, pdcr); if (ret) { dev_err(dev, "audmux internal port setup failed\n"); return ret; } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); if (ret) { dev_err(dev, "audmux internal port setup failed\n"); return ret; } if (!fsl_asoc_card_is_ac97(priv)) { unsigned int pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); ret = imx_audmux_v2_configure_port(ext_port, 0, pdcr); if (ret) { dev_err(dev, "audmux external port setup failed\n"); return ret; } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { dev_err(dev, "audmux external port setup failed\n"); return ret; } return 0; } static int fsl_asoc_card_spdif_init(struct device_node *codec_np[], struct device_node *cpu_np, const char *codec_dai_name[], struct fsl_asoc_card_priv *priv) { struct device *dev = &priv->pdev->dev; struct device_node *np = dev->of_node; if (!of_node_name_eq(cpu_np, "spdif")) { dev_err(dev, "CPU phandle invalid, should be an SPDIF device\n"); return -EINVAL; } priv->dai_link[0].playback_only = true; priv->dai_link[0].capture_only = true; for (int i = 0; i < 2; i++) { if (!codec_np[i]) break; if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) { priv->dai_link[0].capture_only = false; codec_dai_name[i] = "dit-hifi"; } else if (of_device_is_compatible(codec_np[i], "linux,spdif-dir")) { priv->dai_link[0].playback_only = false; codec_dai_name[i] = "dir-hifi"; } } // Old SPDIF DT binding if (!codec_np[0]) { codec_dai_name[0] = snd_soc_dummy_dlc.dai_name; if (of_property_read_bool(np, "spdif-out")) priv->dai_link[0].capture_only = false; if (of_property_read_bool(np, "spdif-in")) priv->dai_link[0].playback_only = false; } if (priv->dai_link[0].playback_only && priv->dai_link[0].capture_only) { dev_err(dev, "no enabled S/PDIF DAI link\n"); return -EINVAL; } if (priv->dai_link[0].playback_only) { priv->dai_link[1].playback_only = true; priv->dai_link[2].playback_only = true; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (priv->dai_link[0].capture_only) { priv->dai_link[1].capture_only = true; priv->dai_link[2].capture_only = true; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } // No DAPM routes with old bindings and dummy codec if (!codec_np[0]) { priv->card.dapm_routes = NULL; priv->card.num_dapm_routes = 0; } if (codec_np[0] && codec_np[1]) { priv->dai_link[0].num_codecs = 2; priv->dai_link[2].num_codecs = 2; } return 0; } static int fsl_asoc_card_cs42888_codec_init(struct fsl_asoc_card_priv *priv) { unsigned long mclk_freq = priv->codec_priv[0].mclk_freq; /* * Set CPU sysclk frequency from codec MCLK only if not already * set by the CPU DAI init (e.g. ESAI extal clock takes precedence). */ if (!priv->cpu_priv.sysclk_freq[TX]) priv->cpu_priv.sysclk_freq[TX] = mclk_freq; if (!priv->cpu_priv.sysclk_freq[RX]) priv->cpu_priv.sysclk_freq[RX] = mclk_freq; priv->constraint_channels = &cs42888_channel_constraints; if (mclk_freq % 12288000 == 0) priv->constraint_rates = &cs42888_rate_48k_constraints; else if (mclk_freq % 11289600 == 0) priv->constraint_rates = &cs42888_rate_44k_constraints; else dev_warn(&priv->pdev->dev, "Unknown MCLK frequency %lu, no rate constraints\n", mclk_freq); return 0; } static int fsl_asoc_card_wm8958_codec_init(struct fsl_asoc_card_priv *priv) { priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq; return 0; } static const struct fsl_asoc_card_pdata fsl_asoc_cs42888_pdata = { .codec_dai_name = "cs42888", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC, .sysclk_dir = { SND_SOC_CLOCK_OUT, SND_SOC_CLOCK_OUT }, .slot_width = 32, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .codec_init = fsl_asoc_card_cs42888_codec_init, }; static const struct fsl_asoc_card_pdata fsl_asoc_cs427x_pdata = { .codec_dai_name = "cs4271-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = CS427x_SYSCLK_MCLK, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_sgtl5000_pdata = { .codec_dai_name = "sgtl5000", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = SGTL5000_SYSCLK, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_tlv320aic32x4_pdata = { .codec_dai_name = "tlv320aic32x4-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_tlv320aic31xx_pdata = { .codec_dai_name = "tlv320dac31xx-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC, .sysclk_dir = { SND_SOC_CLOCK_OUT, SND_SOC_CLOCK_OUT }, .playback_only = true, .dapm_routes = audio_map_tx, .num_dapm_routes = ARRAY_SIZE(audio_map_tx), }; static const struct fsl_asoc_card_pdata fsl_asoc_wm8962_pdata = { .codec_dai_name = "wm8962", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = WM8962_SYSCLK_MCLK, .has_pll = true, .codec_fll_id = WM8962_SYSCLK_FLL, .codec_pll_id = WM8962_FLL, /* * WM8962 has same BCLK generation limitations as WM8960. * See WM8960 section for detailed explanation. */ .exclude_format = SNDRV_PCM_FMTBIT_S20_3LE, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_wm8960_pdata = { .codec_dai_name = "wm8960-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .has_pll = true, .codec_fll_id = WM8960_SYSCLK_AUTO, .codec_pll_id = WM8960_SYSCLK_AUTO, /* * WM8960 in master mode cannot generate exact 1.92 MHz BCLK * required for S20_3LE (48kHz x 2ch x 20bit). Closest available * is 2.048 MHz (SYSCLK/6), which causes right channel corruption. * * In SAI master mode, SAI derive BCLK from MCLK using integer * dividers only. S20_3LE requires non-integer divider ratios * with standard MCLK frequencies. For example, 48kHz stereo * needs 1.920 MHz BCLK, which requires a divider of 6.4 from * 12.288 MHz MCLK (not an integer). */ .exclude_format = SNDRV_PCM_FMTBIT_S20_3LE, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_ac97_pdata = { .codec_dai_name = "ac97-hifi", .dai_fmt = SND_SOC_DAIFMT_AC97, .dapm_routes = audio_map_ac97, .num_dapm_routes = ARRAY_SIZE(audio_map_ac97), }; static const struct fsl_asoc_card_pdata fsl_asoc_mqs_pdata = { .codec_dai_name = "fsl-mqs-dai", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC | SND_SOC_DAIFMT_NB_NF, .playback_only = true, .dapm_routes = audio_map_tx, .num_dapm_routes = ARRAY_SIZE(audio_map_tx), }; static const struct fsl_asoc_card_pdata fsl_asoc_wm8524_pdata = { .codec_dai_name = "wm8524-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC, /* RX=0, TX=1: set TX (index 1) to CLOCK_OUT, RX stays at default IN */ .sysclk_dir = { 0, SND_SOC_CLOCK_OUT }, .sysclk_ratio = { 0, 256 }, .slot_width = 32, .playback_only = true, .dapm_routes = audio_map_tx, .num_dapm_routes = ARRAY_SIZE(audio_map_tx), }; static const struct fsl_asoc_card_pdata fsl_asoc_si476x_pdata = { .codec_dai_name = "si476x-codec", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC, .dapm_routes = audio_map_rx, .num_dapm_routes = ARRAY_SIZE(audio_map_rx), }; static const struct fsl_asoc_card_pdata fsl_asoc_wm8958_pdata = { .codec_dai_name = "wm8994-aif1", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = WM8994_FLL_SRC_MCLK1, .has_pll = true, .codec_fll_id = WM8994_SYSCLK_FLL1, .codec_pll_id = WM8994_FLL1, .codec_init = fsl_asoc_card_wm8958_codec_init, }; static const struct fsl_asoc_card_pdata fsl_asoc_nau8822_pdata = { .codec_dai_name = "nau8822-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = NAU8822_CLK_MCLK, .has_pll = true, .codec_fll_id = NAU8822_CLK_PLL, .codec_pll_id = NAU8822_CLK_PLL, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_wm8904_pdata = { .codec_dai_name = "wm8904-hifi", .dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP, .codec_mclk_id = WM8904_FLL_MCLK, .has_pll = true, .codec_fll_id = WM8904_CLK_FLL, .codec_pll_id = WM8904_FLL_MCLK, .codec_pll_ratio_s24 = 192, .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static const struct fsl_asoc_card_pdata fsl_asoc_spdif_pdata = { .codec_dai_name = "spdif", .dai_fmt = DAI_FMT_BASE, .probe_init = fsl_asoc_card_spdif_init, }; static int hp_jack_event(struct notifier_block *nb, unsigned long event, void *data) { struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(jack->card); if (event & SND_JACK_HEADPHONE) /* Disable speaker if headphone is plugged in */ return snd_soc_dapm_disable_pin(dapm, "Ext Spk"); else return snd_soc_dapm_enable_pin(dapm, "Ext Spk"); } static struct notifier_block hp_jack_nb = { .notifier_call = hp_jack_event, }; static int mic_jack_event(struct notifier_block *nb, unsigned long event, void *data) { struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(jack->card); if (event & SND_JACK_MICROPHONE) /* Disable dmic if microphone is plugged in */ return snd_soc_dapm_disable_pin(dapm, "DMIC"); else return snd_soc_dapm_enable_pin(dapm, "DMIC"); } static struct notifier_block mic_jack_nb = { .notifier_call = mic_jack_event, }; /* * fsl_asoc_card_init_cpu - configure CPU DAI-specific settings. * * Called from late_probe() when the CPU DAI component is guaranteed bound. */ static int fsl_asoc_card_init_cpu(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct device_node *np = priv->pdev->dev.of_node; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); const char *comp_drv_name = cpu_dai->component->driver->name; struct device *dev = card->dev; int ret; if (!strcmp(comp_drv_name, "fsl-ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(dev, "failed to init audmux\n"); return ret; } } else if (!strcmp(comp_drv_name, "fsl-esai")) { struct clk *esai_clk = clk_get(cpu_dai->dev, "extal"); if (!IS_ERR(esai_clk)) { priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); clk_put(esai_clk); } else { dev_warn(dev, "failed to get ESAI extal clock: %ld\n", PTR_ERR(esai_clk)); } priv->cpu_priv.sysclk_id[TX] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[RX] = ESAI_HCKR_EXTAL; } else if (!strcmp(comp_drv_name, "fsl-sai")) { priv->cpu_priv.sysclk_id[TX] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[RX] = FSL_SAI_CLK_MAST1; if (priv->pdata->exclude_format) priv->exclude_format = priv->pdata->exclude_format; } return 0; } /* * fsl_asoc_card_init_codecs - read codec MCLK rates and set codec sysclk. * * Called from late_probe() after all components are bound. */ static int fsl_asoc_card_init_codecs(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); const struct fsl_asoc_card_pdata *pdata = priv->pdata; struct snd_soc_dai *codec_dai; struct codec_priv *codec_priv; struct device *dev = card->dev; int codec_idx; int ret; /* Read MCLK rate from each bound codec component */ for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { struct clk *codec_clk = clk_get(codec_dai->component->dev, NULL); codec_priv = &priv->codec_priv[codec_idx]; if (!IS_ERR(codec_clk)) { codec_priv->mclk_freq = clk_get_rate(codec_clk); clk_put(codec_clk); } } if (pdata->codec_init) { ret = pdata->codec_init(priv); if (ret) return ret; } for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { codec_priv = &priv->codec_priv[codec_idx]; ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk in %s\n", __func__); return ret; } } return 0; } static void fsl_asoc_card_free_jack(struct snd_soc_card *card) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); if (priv->hp_jack.gpio.desc) { snd_soc_jack_notifier_unregister(&priv->hp_jack.jack, &hp_jack_nb); snd_soc_jack_free_gpios(&priv->hp_jack.jack, 1, &priv->hp_jack.gpio); priv->hp_jack.gpio.desc = NULL; } if (priv->mic_jack.gpio.desc) { snd_soc_jack_notifier_unregister(&priv->mic_jack.jack, &mic_jack_nb); snd_soc_jack_free_gpios(&priv->mic_jack.jack, 1, &priv->mic_jack.gpio); priv->mic_jack.gpio.desc = NULL; } } /* * fsl_asoc_card_init_jack - register optional headphone and mic jacks. * * Called from late_probe() once per card bind cycle. */ static int fsl_asoc_card_init_jack(struct snd_soc_card *card) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct device_node *np = priv->pdev->dev.of_node; int ret; /* * Properties "hp-det-gpios" and "mic-det-gpios" are optional. * simple_util_init_jack() checks for the GPIO property and * does nothing if it is absent. */ if (of_property_present(np, "hp-det-gpios") || of_property_present(np, "hp-det-gpio") /* deprecated */) { ret = simple_util_init_jack(card, &priv->hp_jack, 1, NULL, "Headphone Jack"); if (ret) return ret; snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb); } if (of_property_present(np, "mic-det-gpios") || of_property_present(np, "mic-det-gpio") /* deprecated */) { ret = simple_util_init_jack(card, &priv->mic_jack, 0, NULL, "Mic Jack"); if (ret) return ret; snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb); } return 0; } static int fsl_asoc_card_late_probe(struct snd_soc_card *card) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct snd_soc_pcm_runtime *rtd; int ret; /* Use the first rtd which carries the CPU+codec DAIs */ rtd = list_first_entry(&card->rtd_list, struct snd_soc_pcm_runtime, list); ret = fsl_asoc_card_init_jack(card); if (ret) goto jack_fail; ret = fsl_asoc_card_init_cpu(card, rtd); if (ret) goto jack_fail; if (fsl_asoc_card_is_ac97(priv)) { #if IS_ENABLED(CONFIG_SND_AC97_CODEC) struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); /* * Use slots 3/4 for S/PDIF so SSI won't try to enable * other slots and send some samples there * due to SLOTREQ bits for S/PDIF received from codec */ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); #endif return 0; } ret = fsl_asoc_card_init_codecs(card, rtd); if (ret) goto jack_fail; return 0; jack_fail: fsl_asoc_card_free_jack(card); return ret; } static int fsl_asoc_card_card_remove(struct snd_soc_card *card) { fsl_asoc_card_free_jack(card); return 0; } static int fsl_asoc_card_probe(struct platform_device *pdev) { struct device_node *cpu_np, *asrc_np; struct snd_soc_dai_link_component *codec_comp; struct device_node *codec_np[2]; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct device_node *bitclkprovider = NULL; struct device_node *frameprovider = NULL; struct fsl_asoc_card_priv *priv; const struct fsl_asoc_card_pdata *pdata; struct snd_soc_dai_link_component *dlc; const char *codec_dai_name[2] = { NULL, NULL }; u32 asrc_fmt = 0; int codec_idx; u32 width; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; priv->pdev = pdev; pdata = of_device_get_match_data(&pdev->dev); if (!pdata) { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } priv->pdata = pdata; cpu_np = of_parse_phandle(np, "audio-cpu", 0); /* Give a chance to old DT bindings */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); if (!cpu_np) cpu_np = of_parse_phandle(np, "spdif-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } codec_np[0] = of_parse_phandle(np, "audio-codec", 0); codec_np[1] = of_parse_phandle(np, "audio-codec", 1); asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Default sample rate and format, will be updated in hw_params() */ priv->sample_rate = 44100; priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; /* Assign a default DAI format, and allow each card to overwrite it */ priv->dai_fmt = DAI_FMT_BASE; memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); /* * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms * "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms * "DPCM Link Back-End": 1 cpus, 2 codecs * totally 10 components */ dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; goto asrc_fail; } priv->dai_link[0].cpus = &dlc[0]; priv->dai_link[0].num_cpus = 1; priv->dai_link[0].codecs = &dlc[1]; priv->dai_link[0].num_codecs = 1; priv->dai_link[0].platforms = &dlc[3]; priv->dai_link[0].num_platforms = 1; priv->dai_link[1].cpus = &dlc[4]; priv->dai_link[1].num_cpus = 1; priv->dai_link[1].codecs = &dlc[5]; priv->dai_link[1].num_codecs = 0; /* dummy */ priv->dai_link[1].platforms = &dlc[6]; priv->dai_link[1].num_platforms = 1; priv->dai_link[2].cpus = &dlc[7]; priv->dai_link[2].num_cpus = 1; priv->dai_link[2].codecs = &dlc[8]; priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; for (codec_idx = 0; codec_idx < 2; codec_idx++) { priv->codec_priv[codec_idx].fll_id = -1; priv->codec_priv[codec_idx].pll_id = -1; priv->codec_priv[codec_idx].pll_ratio_s24 = 384; } /* Diversify the card configurations */ priv->cpu_priv.sysclk_dir[TX] = pdata->sysclk_dir[TX]; priv->cpu_priv.sysclk_dir[RX] = pdata->sysclk_dir[RX]; priv->cpu_priv.sysclk_ratio[TX] = pdata->sysclk_ratio[TX]; priv->cpu_priv.sysclk_ratio[RX] = pdata->sysclk_ratio[RX]; priv->cpu_priv.slot_width = pdata->slot_width; codec_dai_name[0] = pdata->codec_dai_name; priv->codec_priv[0].mclk_id = pdata->codec_mclk_id; if (pdata->has_pll) { priv->codec_priv[0].fll_id = pdata->codec_fll_id; priv->codec_priv[0].pll_id = pdata->codec_pll_id; } if (pdata->codec_pll_ratio_s24) priv->codec_priv[0].pll_ratio_s24 = pdata->codec_pll_ratio_s24; if (pdata->playback_only) { priv->dai_link[1].playback_only = 1; priv->dai_link[2].playback_only = 1; } if (pdata->capture_only) { priv->dai_link[1].capture_only = 1; priv->dai_link[2].capture_only = 1; } priv->dai_fmt = pdata->dai_fmt; priv->card.dapm_routes = pdata->dapm_routes; priv->card.num_dapm_routes = pdata->num_dapm_routes; if (pdata->probe_init) { ret = pdata->probe_init(codec_np, cpu_np, codec_dai_name, priv); if (ret) goto asrc_fail; } /* * Allow setting mclk-id from the device-tree node. Otherwise, the * default value for each card configuration is used. */ for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { of_property_read_u32_index(np, "mclk-id", codec_idx, &priv->codec_priv[codec_idx].mclk_id); } /* Format info from DT is optional. */ snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider); if (bitclkprovider || frameprovider) { unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL); bool codec_bitclkprovider = false; bool codec_frameprovider = false; for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { if (bitclkprovider && codec_np[codec_idx] == bitclkprovider) codec_bitclkprovider = true; if (frameprovider && codec_np[codec_idx] == frameprovider) codec_frameprovider = true; } if (codec_bitclkprovider) daifmt |= (codec_frameprovider) ? SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC; else daifmt |= (codec_frameprovider) ? SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC; /* Override dai_fmt with value from DT */ priv->dai_fmt = daifmt; } /* Change direction according to format */ if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) { priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; } of_node_put(bitclkprovider); of_node_put(frameprovider); /* Initialize sound card */ priv->card.dev = &pdev->dev; priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { /* * "model" is required by the DT binding. Enforce it here so * the driver fails with a clear message. */ dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); goto asrc_fail; } priv->card.dai_link = priv->dai_link; priv->card.late_probe = fsl_asoc_card_late_probe; priv->card.remove = fsl_asoc_card_card_remove; priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); /* Drop the second half of DAPM routes -- ASRC */ if (!asrc_pdev) priv->card.num_dapm_routes /= 2; if (of_property_present(np, "audio-routing")) { ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); if (ret) { dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); goto asrc_fail; } } /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { codec_comp->dai_name = codec_dai_name[codec_idx]; } // Old SPDIF DT binding support if (codec_dai_name[0] == snd_soc_dummy_dlc.dai_name) priv->dai_link[0].codecs[0].name = snd_soc_dummy_dlc.name; if (!fsl_asoc_card_is_ac97(priv)) { for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { codec_comp->of_node = codec_np[codec_idx]; } } else { u32 idx; ret = of_property_read_u32(cpu_np, "cell-index", &idx); if (ret) { dev_err(&pdev->dev, "cannot get CPU index property\n"); goto asrc_fail; } priv->dai_link[0].codecs[0].name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); if (!priv->dai_link[0].codecs[0].name) { ret = -ENOMEM; goto asrc_fail; } } priv->dai_link[0].platforms->of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; if (asrc_pdev) { /* DPCM DAI Links only if ASRC exists */ priv->dai_link[1].dpcm_merged_chan = 1; priv->dai_link[1].ignore_pmdown_time = 1; priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) { codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name; codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node; codec_comp->name = priv->dai_link[0].codecs[codec_idx].name; } priv->dai_link[2].cpus->of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->dai_link[2].ignore_pmdown_time = 1; priv->card.num_links = 3; ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", &priv->asrc_rate); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); ret = -EINVAL; goto asrc_fail; } ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt); priv->asrc_format = asrc_fmt; if (ret) { /* Fallback to old binding; translate to asrc_format */ ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); if (ret) { dev_err(&pdev->dev, "failed to decide output format\n"); goto asrc_fail; } if (width == 24) priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; else priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; } } /* Finish card registering */ platform_set_drvdata(pdev, priv); snd_soc_card_set_drvdata(&priv->card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); goto asrc_fail; } asrc_fail: of_node_put(asrc_np); of_node_put(codec_np[0]); of_node_put(codec_np[1]); fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", .data = &fsl_asoc_ac97_pdata }, { .compatible = "fsl,imx-audio-cs42888", .data = &fsl_asoc_cs42888_pdata }, { .compatible = "fsl,imx-audio-cs427x", .data = &fsl_asoc_cs427x_pdata }, { .compatible = "fsl,imx-audio-tlv320aic32x4", .data = &fsl_asoc_tlv320aic32x4_pdata }, { .compatible = "fsl,imx-audio-tlv320aic31xx", .data = &fsl_asoc_tlv320aic31xx_pdata }, { .compatible = "fsl,imx-audio-sgtl5000", .data = &fsl_asoc_sgtl5000_pdata }, { .compatible = "fsl,imx-audio-wm8962", .data = &fsl_asoc_wm8962_pdata }, { .compatible = "fsl,imx-audio-wm8960", .data = &fsl_asoc_wm8960_pdata }, { .compatible = "fsl,imx-audio-mqs", .data = &fsl_asoc_mqs_pdata }, { .compatible = "fsl,imx-audio-wm8524", .data = &fsl_asoc_wm8524_pdata }, { .compatible = "fsl,imx-audio-si476x", .data = &fsl_asoc_si476x_pdata }, { .compatible = "fsl,imx-audio-wm8958", .data = &fsl_asoc_wm8958_pdata }, { .compatible = "fsl,imx-audio-nau8822", .data = &fsl_asoc_nau8822_pdata }, { .compatible = "fsl,imx-audio-wm8904", .data = &fsl_asoc_wm8904_pdata }, { .compatible = "fsl,imx-audio-spdif", .data = &fsl_asoc_spdif_pdata }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); static struct platform_driver fsl_asoc_card_driver = { .probe = fsl_asoc_card_probe, .driver = { .name = DRIVER_NAME, .pm = &snd_soc_pm_ops, .of_match_table = fsl_asoc_card_dt_ids, }, }; module_platform_driver(fsl_asoc_card_driver); MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); MODULE_AUTHOR("Nicolin Chen "); MODULE_ALIAS("platform:" DRIVER_NAME); MODULE_LICENSE("GPL");