// 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" #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) /** * 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() */ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; u32 fll_id; u32 pll_id; }; /** * 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() * @slot_width: Slot width 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 slot_width; }; /** * 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 * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure * @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 * @name: Card name */ struct fsl_asoc_card_priv { struct snd_soc_dai_link dai_link[3]; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct platform_device *pdev; struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; snd_pcm_format_t asrc_format; u32 dai_fmt; 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 */ {"Playback", NULL, "AC97 Playback"}, {"AC97 Capture", NULL, "Capture"}, /* 2nd half -- ASRC DAPM routes */ {"AC97 Playback", NULL, "ASRC-Playback"}, {"ASRC-Capture", NULL, "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 = asoc_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 = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; unsigned int pll_out; 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; /* Specific configurations of DAIs starts from here */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx], cpu_priv->sysclk_freq[tx], 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) { ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 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 */ if (codec_priv->pll_id && codec_priv->fll_id) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 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(asoc_rtd_to_codec(rtd, 0), 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 = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = rtd->card->dev; int ret; priv->streams &= ~BIT(substream->stream); if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 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(asoc_rtd_to_codec(rtd, 0), 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 const struct snd_soc_ops fsl_asoc_card_ops = { .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; } SND_SOC_DAILINK_DEFS(hifi, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_DUMMY())); static struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", .stream_name = "HiFi", .ops = &fsl_asoc_card_ops, SND_SOC_DAILINK_REG(hifi), }, /* DPCM Link between Front-End and Back-End (Optional) */ { .name = "HiFi-ASRC-FE", .stream_name = "HiFi-ASRC-FE", .dpcm_playback = 1, .dpcm_capture = 1, .dynamic = 1, SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-BE", .stream_name = "HiFi-ASRC-BE", .be_hw_params_fixup = be_hw_params_fixup, .ops = &fsl_asoc_card_ops, .dpcm_playback = 1, .dpcm_capture = 1, .no_pcm = 1, SND_SOC_DAILINK_REG(hifi_be), }, }; 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_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: 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_CBM_CFS: 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_CBS_CFM: 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_CBS_CFS: 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 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 = &jack->card->dapm; if (event & SND_JACK_HEADPHONE) /* Disable speaker if headphone is plugged in */ snd_soc_dapm_disable_pin(dapm, "Ext Spk"); else snd_soc_dapm_enable_pin(dapm, "Ext Spk"); return 0; } 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 = &jack->card->dapm; if (event & SND_JACK_MICROPHONE) /* Disable dmic if microphone is plugged in */ snd_soc_dapm_disable_pin(dapm, "DMIC"); else snd_soc_dapm_enable_pin(dapm, "DMIC"); return 0; } static struct notifier_block mic_jack_nb = { .notifier_call = mic_jack_event, }; 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 = list_first_entry( &card->rtd_list, struct snd_soc_pcm_runtime, list); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; int ret; if (fsl_asoc_card_is_ac97(priv)) { #if IS_ENABLED(CONFIG_SND_AC97_CODEC) struct snd_soc_component *component = asoc_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 = 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 int fsl_asoc_card_probe(struct platform_device *pdev) { struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev = NULL; const char *codec_dai_name; const char *codec_dev_name; unsigned int daifmt; u32 width; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; cpu_np = of_parse_phandle(np, "audio-cpu", 0); /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } cpu_pdev = of_find_device_by_node(cpu_np); if (!cpu_pdev) { dev_err(&pdev->dev, "failed to find CPU DAI device\n"); ret = -EINVAL; goto fail; } codec_np = of_parse_phandle(np, "audio-codec", 0); if (codec_np) { struct platform_device *codec_pdev; struct i2c_client *codec_i2c; codec_i2c = of_find_i2c_device_by_node(codec_np); if (codec_i2c) { codec_dev = &codec_i2c->dev; codec_dev_name = codec_i2c->name; } if (!codec_dev) { codec_pdev = of_find_device_by_node(codec_np); if (codec_pdev) { codec_dev = &codec_pdev->dev; codec_dev_name = codec_pdev->name; } } } asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ if (codec_dev) { struct clk *codec_clk = clk_get(codec_dev, NULL); if (!IS_ERR(codec_clk)) { priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); clk_put(codec_clk); } } /* 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)); priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { codec_dai_name = "cs4271-hifi"; priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { codec_dai_name = "tlv320aic32x4-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; priv->cpu_priv.slot_width = 32; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { codec_dai_name = "si476x-codec"; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { codec_dai_name = "wm8994-aif1"; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; priv->codec_priv.pll_id = WM8994_FLL1; priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; priv->card.dapm_routes = NULL; priv->card.num_dapm_routes = 0; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } /* Format info from DT is optional. */ daifmt = snd_soc_of_parse_daifmt(np, NULL, &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (bitclkmaster || framemaster) { if (codec_np == bitclkmaster) daifmt |= (codec_np == framemaster) ? SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; else daifmt |= (codec_np == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; /* Override dai_fmt with value from DT */ priv->dai_fmt = daifmt; } /* Change direction according to format */ if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) { priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; } of_node_put(bitclkmaster); of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; goto asrc_fail; } /* Common settings for corresponding Freescale CPU DAI driver */ if (of_node_name_eq(cpu_np, "ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); goto asrc_fail; } } else if (of_node_name_eq(cpu_np, "esai")) { struct clk *esai_clk = clk_get(&cpu_pdev->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 if (PTR_ERR(esai_clk) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; goto asrc_fail; } priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; } else if (of_node_name_eq(cpu_np, "sai")) { priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name); priv->card.name = priv->name; } priv->card.dai_link = priv->dai_link; priv->card.late_probe = fsl_asoc_card_late_probe; 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_read_bool(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; priv->dai_link[0].codecs->dai_name = codec_dai_name; if (!fsl_asoc_card_is_ac97(priv)) priv->dai_link[0].codecs->of_node = codec_np; 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->name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); if (!priv->dai_link[0].codecs->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 exsits */ priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; priv->dai_link[2].codecs->dai_name = codec_dai_name; priv->dai_link[2].codecs->of_node = codec_np; priv->dai_link[2].codecs->name = priv->dai_link[0].codecs->name; priv->dai_link[2].cpus->of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; 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", &priv->asrc_format); 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) { if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto asrc_fail; } /* * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and * asoc_simple_init_jack uses these properties for creating * Headphone Jack and Microphone Jack. * * The notifier is initialized in snd_soc_card_jack_new(), then * snd_soc_jack_notifier_register can be called. */ if (of_property_read_bool(np, "hp-det-gpio")) { ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack, 1, NULL, "Headphone Jack"); if (ret) goto asrc_fail; snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb); } if (of_property_read_bool(np, "mic-det-gpio")) { ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack, 0, NULL, "Mic Jack"); if (ret) goto asrc_fail; snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb); } asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-cs427x", }, { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, { .compatible = "fsl,imx-audio-si476x", }, { .compatible = "fsl,imx-audio-wm8958", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); static struct platform_driver fsl_asoc_card_driver = { .probe = fsl_asoc_card_probe, .driver = { .name = "fsl-asoc-card", .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:fsl-asoc-card"); MODULE_LICENSE("GPL");