diff options
Diffstat (limited to 'sound/soc/fsl/fsl-asoc-card.c')
| -rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 1112 |
1 files changed, 1112 insertions, 0 deletions
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 000000000000..2c7eb0baa0f3 --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,1112 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/simple_card_utils.h> + +#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) + +/** + * struct codec_priv - CODEC private data + * @mclk: Main clock of the CODEC + * @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 { + struct clk *mclk; + unsigned long mclk_freq; + unsigned long free_freq; + u32 mclk_id; + int fll_id; + int 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() + * @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 - 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 simple_util_jack hp_jack; + struct simple_util_jack mic_jack; + struct platform_device *pdev; + struct codec_priv codec_priv[2]; + 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 */ + {"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 * 384; + 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 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; +} + +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 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, +}; + +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; + struct codec_priv *codec_priv; + struct device *dev = card->dev; + int codec_idx; + int ret; + + 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; + } + + 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; + } + + if (!IS_ERR_OR_NULL(codec_priv->mclk)) + clk_prepare_enable(codec_priv->mclk); + } + + 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 platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct device *codec_dev[2] = { NULL, NULL }; + struct snd_soc_dai_link_component *dlc; + const char *codec_dai_name[2]; + const char *codec_dev_name[2]; + 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; + + 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; + } + + 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[0] = of_parse_phandle(np, "audio-codec", 0); + codec_np[1] = of_parse_phandle(np, "audio-codec", 1); + + for (codec_idx = 0; codec_idx < 2; codec_idx++) { + if (codec_np[codec_idx]) { + struct platform_device *codec_pdev; + struct i2c_client *codec_i2c; + + codec_i2c = of_find_i2c_device_by_node(codec_np[codec_idx]); + if (codec_i2c) { + codec_dev[codec_idx] = &codec_i2c->dev; + codec_dev_name[codec_idx] = codec_i2c->name; + } + if (!codec_dev[codec_idx]) { + codec_pdev = of_find_device_by_node(codec_np[codec_idx]); + if (codec_pdev) { + codec_dev[codec_idx] = &codec_pdev->dev; + codec_dev_name[codec_idx] = 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 */ + for (codec_idx = 0; codec_idx < 2; codec_idx++) { + if (codec_dev[codec_idx]) { + struct clk *codec_clk = clk_get(codec_dev[codec_idx], NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv[codec_idx].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)); + /* + * "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; + } + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name[0] = "cs42888"; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv[0].mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv[0].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_CBC_CFC; + } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { + codec_dai_name[0] = "cs4271-hifi"; + priv->codec_priv[0].mclk_id = CS427x_SYSCLK_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name[0] = "sgtl5000"; + priv->codec_priv[0].mclk_id = SGTL5000_SYSCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { + codec_dai_name[0] = "tlv320aic32x4-hifi"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) { + codec_dai_name[0] = "tlv320dac31xx-hifi"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; + priv->dai_link[1].playback_only = 1; + priv->dai_link[2].playback_only = 1; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + 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-wm8962")) { + codec_dai_name[0] = "wm8962"; + priv->codec_priv[0].mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv[0].fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv[0].pll_id = WM8962_FLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + codec_dai_name[0] = "wm8960-hifi"; + priv->codec_priv[0].fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv[0].pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name[0] = "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[0] = "fsl-mqs-dai"; + priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBC_CFC | + SND_SOC_DAIFMT_NB_NF; + priv->dai_link[1].playback_only = 1; + priv->dai_link[2].playback_only = 1; + 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[0] = "wm8524-hifi"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; + priv->dai_link[1].playback_only = 1; + priv->dai_link[2].playback_only = 1; + priv->cpu_priv.slot_width = 32; + priv->card.dapm_routes = audio_map_tx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_ratio[TX] = 256; + } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { + codec_dai_name[0] = "si476x-codec"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; + 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[0] = "wm8994-aif1"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + priv->codec_priv[0].mclk_id = WM8994_FLL_SRC_MCLK1; + priv->codec_priv[0].fll_id = WM8994_SYSCLK_FLL1; + priv->codec_priv[0].pll_id = WM8994_FLL1; + priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq; + priv->card.dapm_routes = NULL; + priv->card.num_dapm_routes = 0; + } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) { + codec_dai_name[0] = "nau8822-hifi"; + priv->codec_priv[0].mclk_id = NAU8822_CLK_MCLK; + priv->codec_priv[0].fll_id = NAU8822_CLK_PLL; + priv->codec_priv[0].pll_id = NAU8822_CLK_PLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + if (codec_dev[0]) + priv->codec_priv[0].mclk = devm_clk_get(codec_dev[0], NULL); + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name[0] = "wm8904-hifi"; + priv->codec_priv[0].mclk_id = WM8904_FLL_MCLK; + priv->codec_priv[0].fll_id = WM8904_CLK_FLL; + priv->codec_priv[0].pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-spdif")) { + ret = fsl_asoc_card_spdif_init(codec_np, cpu_np, codec_dai_name, priv); + if (ret) + goto asrc_fail; + } else { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + ret = -EINVAL; + 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); + + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0] + && codec_dai_name[0] != snd_soc_dummy_dlc.dai_name) { + dev_dbg(&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->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[0]); + 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_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].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->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 = (__force snd_pcm_format_t)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; + } + + /* + * Properties "hp-det-gpios" and "mic-det-gpios" are optional, and + * simple_util_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-gpios") || + of_property_read_bool(np, "hp-det-gpio") /* deprecated */) { + ret = simple_util_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-gpios") || + of_property_read_bool(np, "mic-det-gpio") /* deprecated */) { + ret = simple_util_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[0]); + of_node_put(codec_np[1]); + 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-tlv320aic31xx", }, + { .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", }, + { .compatible = "fsl,imx-audio-nau8822", }, + { .compatible = "fsl,imx-audio-wm8904", }, + { .compatible = "fsl,imx-audio-spdif", }, + {} +}; +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 <nicoleotsuka@gmail.com>"); +MODULE_ALIAS("platform:" DRIVER_NAME); +MODULE_LICENSE("GPL"); |