| // 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" |
| |
| #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 */ |
| {"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 = 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_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 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 */ |
| 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 = &jack->card->dapm; |
| |
| 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 = 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 *bitclkprovider = NULL; |
| struct device_node *frameprovider = 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; |
| u32 asrc_fmt = 0; |
| 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_CBC_CFC; |
| } 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_CBP_CFP; |
| } 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_CBP_CFP; |
| } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { |
| codec_dai_name = "tlv320aic32x4-hifi"; |
| priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; |
| } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) { |
| codec_dai_name = "tlv320dac31xx-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.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 = "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_CBP_CFP; |
| } 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_CBP_CFP; |
| } 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_CBC_CFC | |
| 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_CBC_CFC; |
| 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_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 = "wm8994-aif1"; |
| priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; |
| 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; |
| } |
| |
| /* |
| * Allow setting mclk-id from the device-tree node. Otherwise, the |
| * default value for each card configuration is used. |
| */ |
| of_property_read_u32(np, "mclk-id", &priv->codec_priv.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); |
| |
| if (codec_np == bitclkprovider) |
| daifmt |= (codec_np == frameprovider) ? |
| SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC; |
| else |
| daifmt |= (codec_np == 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) { |
| 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->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", &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-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-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", }, |
| {} |
| }; |
| 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 <nicoleotsuka@gmail.com>"); |
| MODULE_ALIAS("platform:fsl-asoc-card"); |
| MODULE_LICENSE("GPL"); |