diff options
| -rw-r--r-- | Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 82 | ||||
| -rw-r--r-- | sound/soc/fsl/Kconfig | 16 | ||||
| -rw-r--r-- | sound/soc/fsl/Makefile | 2 | ||||
| -rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 573 |
4 files changed, 673 insertions, 0 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt new file mode 100644 index 000000000000..a96774c194c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -0,0 +1,82 @@ +Freescale Generic ASoC Sound Card with ASRC support + +The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale +SoCs connecting with external CODECs. + +The idea of this generic sound card is a bit like ASoC Simple Card. However, +for Freescale SoCs (especially those released in recent years), most of them +have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And +this is a specific feature that might be painstakingly controlled and merged +into the Simple Card. + +So having this generic sound card allows all Freescale SoC users to benefit +from the simplification of a new card support and the capability of the wide +sample rates support through ASRC. + +Note: The card is initially designed for those sound cards who use I2S and + PCM DAI formats. However, it'll be also possible to support those non + I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long + as the driver has been properly upgraded. + + +The compatible list for this generic sound card currently: + "fsl,imx-audio-cs42888" + + "fsl,imx-audio-wm8962" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) + + "fsl,imx-audio-sgtl5000" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) + +Required properties: + + - compatible : Contains one of entries in the compatible list. + + - model : The user-visible name of this sound complex + + - audio-cpu : The phandle of an CPU DAI controller + + - audio-codec : The phandle of an audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. There're a few pre-designed board connectors: + * Line Out Jack + * Line In Jack + * Headphone Jack + * Mic Jack + * Ext Spk + * AMIC (stands for Analog Microphone Jack) + * DMIC (stands for Digital Microphone Jack) + + Note: The "Mic Jack" and "AMIC" are redundant while + coexsiting in order to support the old bindings + of wm8962 and sgtl5000. + +Optional properties: + + - audio-asrc : The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + +Example: +sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc99291..2b99a9e86899 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -59,6 +59,22 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + select SND_SOC_CS42XX8_I2C + select SND_SOC_SGTL5000 + select SND_SOC_WM8962 + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9ff59267eac9..8f6d84efa973 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o @@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 000000000000..cf3f1f47f1e8 --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,573 @@ +/* + * Freescale Generic ASoC Sound Card driver with ASRC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <nicoleotsuka@gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "fsl_esai.h" +#include "fsl_sai.h" +#include "imx-audmux.h" + +#include "../codecs/sgtl5000.h" +#include "../codecs/wm8962.h" + +#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) + +/** + * CODEC private data + * + * @mclk_freq: Clock rate of MCLK + * @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; + u32 mclk_id; + u32 fll_id; + u32 pll_id; +}; + +/** + * CPU private data + * + * @sysclk_freq[2]: SYSCLK rates for set_sysclk() + * @sysclk_dir[2]: SYSCLK directions for set_sysclk() + * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * + * Note: [1] for tx and [0] for rx + */ +struct cpu_priv { + unsigned long sysclk_freq[2]; + u32 sysclk_dir[2]; + u32 sysclk_id[2]; +}; + +/** + * Freescale Generic ASOC card private data + * + * @dai_link[3]: DAI link structure including normal one and DPCM link + * @pdev: platform device pointer + * @codec_priv: CODEC private data + * @cpu_priv: CPU private data + * @card: ASoC card structure + * @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 platform_device *pdev; + struct codec_priv codec_priv; + struct cpu_priv cpu_priv; + struct snd_soc_card card; + u32 sample_rate; + u32 sample_format; + u32 asrc_rate; + u32 asrc_format; + u32 dai_fmt; + char name[32]; +}; + +/** + * This dapm route map exsits for DPCM link only. + * The other routes shall go through Device Tree. + */ +static const struct snd_soc_dapm_route audio_map[] = { + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "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 int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct cpu_priv *cpu_priv = &priv->cpu_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); + + if (priv->card.set_bias_level) + return 0; + + /* Specific configurations of DAIs starts from here */ + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + cpu_priv->sysclk_freq[tx], + cpu_priv->sysclk_dir[tx]); + if (ret) { + dev_err(dev, "failed to set sysclk for cpu dai\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops fsl_asoc_card_ops = { + .hw_params = fsl_asoc_card_hw_params, +}; + +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(mask, priv->asrc_format); + + return 0; +} + +static 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", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .dpcm_capture = 1, + .dynamic = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .platform_name = "snd-soc-dummy", + .be_hw_params_fixup = be_hw_params_fixup, + .ops = &fsl_asoc_card_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level != SND_SOC_BIAS_STANDBY) + break; + + 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); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_PREPARE) + break; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_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) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + return 0; +} + +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. + * 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: + return -EINVAL; + } + + /* Asynchronous mode can not be set along with RCLKDIR */ + ret = imx_audmux_v2_configure_port(int_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + 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; + } + + ret = imx_audmux_v2_configure_port(ext_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + 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_late_probe(struct snd_soc_card *card) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret) { + 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 platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct i2c_client *codec_dev; + struct clk *codec_clk; + 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); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!cpu_np || !codec_np) { + dev_err(&pdev->dev, "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_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + 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 */ + codec_clk = clk_get(&codec_dev->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; + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + priv->card.set_bias_level = NULL; + 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->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-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-wm8962")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + 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 { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + return -EINVAL; + } + + /* Common settings for corresponding Freescale CPU DAI driver */ + if (strstr(cpu_np->name, "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 fail; + } + } else if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; + priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; + } else if (strstr(cpu_np->name, "sai")) { + priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; + priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; + } + + sprintf(priv->name, "%s-audio", codec_dev->name); + + /* Initialize sound card */ + priv->pdev = pdev; + priv->card.dev = &pdev->dev; + priv->card.name = priv->name; + priv->card.dai_link = priv->dai_link; + priv->card.dapm_routes = audio_map; + priv->card.late_probe = fsl_asoc_card_late_probe; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); + priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + + memcpy(priv->dai_link, fsl_asoc_card_dai, + sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + + /* Normal DAI Link */ + priv->dai_link[0].cpu_of_node = cpu_np; + priv->dai_link[0].codec_of_node = codec_np; + priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].platform_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].cpu_of_node = asrc_np; + priv->dai_link[1].platform_of_node = asrc_np; + priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].cpu_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 fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto 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(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + +fail: + of_node_put(codec_np); + of_node_put(asrc_np); + of_node_put(cpu_np); + + return ret; +} + +static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8962", }, + {} +}; + +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"); |