diff options
author | Shengjiu Wang <shengjiu.wang@freescale.com> | 2014-08-04 16:46:01 +0800 |
---|---|---|
committer | Nitin Garg <nitin.garg@nxp.com> | 2016-01-14 11:00:04 -0600 |
commit | 2c77e09d5e986bd256a4102e32bf5085245e6546 (patch) | |
tree | f4e816f6780431843e50ff4930343f11ecc64625 | |
parent | 3986758c3c12d3eb044a5fe4514bc7a23040bddd (diff) |
MLK-11429-21: ASoC: imx-cs42888: port cs42888 machine driver from imx_3.10.y
cherry-pick below patch from imx_3.14.y
ENGR00330403-1: ASoC: imx-cs42888: port cs42888 machine driver from imx_3.10.y
Port the cs42888 machine driver from imx_3.10.y and do update according to
new esai driver and asrc driver.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
(cherry picked from commit 7ed3aac83630a38eb397ed92f815a28e07198748)
-rw-r--r-- | Documentation/devicetree/bindings/sound/imx-audio-cs42888.txt | 25 | ||||
-rw-r--r-- | sound/soc/fsl/Kconfig | 13 | ||||
-rw-r--r-- | sound/soc/fsl/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-cs42888.c | 332 |
4 files changed, 372 insertions, 0 deletions
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-cs42888.txt b/Documentation/devicetree/bindings/sound/imx-audio-cs42888.txt new file mode 100644 index 000000000000..af746c4c81df --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-cs42888.txt @@ -0,0 +1,25 @@ +Freescale i.MX audio complex with CS42888 codec + +Required properties: +- compatible : "fsl,imx-audio-cs42888" +- model : The user-visible name of this sound complex +- esai-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the CS42888 audio codec + +Optional properties: +- asrc-controller : The phandle of the i.MX ASRC controller +- 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. Valid names could be power + supplies, CS42888 pins, and the jacks on the board: + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "cs42888-audio"; + esai-controller = <&esai>; + asrc-controller = <&asrc_p2p>; + audio-codec = <&codec>; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 99e737835d2a..de1a97fe99e6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -256,6 +256,19 @@ config SND_SOC_IMX_WM8958 Say Y if you want to add support for SoC audio on an i.MX board with a wm8958 codec. +config SND_SOC_IMX_CS42888 + tristate "SoC Audio support for i.MX boards with cs42888" + depends on OF && I2C + select SND_SOC_CS42XX8_I2C + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_ASRC + select SND_SOC_FSL_UTILS + help + SoC Audio support for i.MX boards with cs42888 + Say Y if you want to add support for SoC audio on an i.MX board with + a cs42888 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 93a4ac787949..a6f65a36cf29 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -53,6 +53,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o +snd-soc-imx-cs42888-objs := imx-cs42888.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8960-objs := imx-wm8960.o @@ -66,6 +67,7 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o +obj-$(CONFIG_SND_SOC_IMX_CS42888) += snd-soc-imx-cs42888.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-${CONFIG_SND_SOC_IMX_WM8958} += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8960) += snd-soc-imx-wm8960.o diff --git a/sound/soc/fsl/imx-cs42888.c b/sound/soc/fsl/imx-cs42888.c new file mode 100644 index 000000000000..68eaaaa9cfcd --- /dev/null +++ b/sound/soc/fsl/imx-cs42888.c @@ -0,0 +1,332 @@ +/* + * Copyright (C) 2010-2015 Freescale Semiconductor, Inc. All Rights Reserved. + */ + +/* + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> + +#include "fsl_esai.h" + +#define CODEC_CLK_EXTER_OSC 1 +#define CODEC_CLK_ESAI_HCKT 2 +#define SUPPORT_RATE_NUM 10 + +struct imx_priv { + unsigned int mclk_freq; + struct platform_device *pdev; + struct platform_device *asrc_pdev; + u32 asrc_rate; + u32 asrc_format; +}; + +static struct imx_priv card_priv; + +static int imx_cs42888_surround_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + u32 dai_format = 0; + + dai_format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL, + priv->mclk_freq, SND_SOC_CLOCK_OUT); + else + snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL, + priv->mclk_freq, SND_SOC_CLOCK_OUT); + snd_soc_dai_set_sysclk(codec_dai, 0, priv->mclk_freq, SND_SOC_CLOCK_IN); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + /* set i.MX active slot mask */ + snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + return 0; +} + +static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + static struct snd_pcm_hw_constraint_list constraint_rates; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + static u32 support_rates[SUPPORT_RATE_NUM]; + int ret; + + if (priv->mclk_freq == 24576000) { + support_rates[0] = 48000; + support_rates[1] = 96000; + support_rates[2] = 192000; + constraint_rates.list = support_rates; + constraint_rates.count = 3; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraint_rates); + if (ret) + return ret; + } else + dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq); + + return 0; +} + +static struct snd_soc_ops imx_cs42888_surround_ops = { + .startup = imx_cs42888_surround_startup, + .hw_params = imx_cs42888_surround_hw_params, +}; + +/** + * imx_cs42888_surround_startup() is to set constrain for hw parameter, but + * backend use same runtime as frontend, for p2p backend need to use different + * parameter, so backend can't use the startup. + */ +static struct snd_soc_ops imx_cs42888_surround_ops_be = { + .hw_params = imx_cs42888_surround_hw_params, +}; + + +static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Line out jack */ + {"Line Out Jack", NULL, "AOUT1L"}, + {"Line Out Jack", NULL, "AOUT1R"}, + {"Line Out Jack", NULL, "AOUT2L"}, + {"Line Out Jack", NULL, "AOUT2R"}, + {"Line Out Jack", NULL, "AOUT3L"}, + {"Line Out Jack", NULL, "AOUT3R"}, + {"Line Out Jack", NULL, "AOUT4L"}, + {"Line Out Jack", NULL, "AOUT4R"}, + {"AIN1L", NULL, "Line In Jack"}, + {"AIN1R", NULL, "Line In Jack"}, + {"AIN2L", NULL, "Line In Jack"}, + {"AIN2R", NULL, "Line In Jack"}, + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */ + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "Capture"}, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { + + struct imx_priv *priv = &card_priv; + struct snd_interval *rate; + struct snd_mask *mask; + + if (!priv->asrc_pdev) + return -EINVAL; + + 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 imx_cs42888_dai[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai_name = "cs42888", + .ops = &imx_cs42888_surround_ops, + .ignore_pmdown_time = 1, + }, + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .codec_dai_name = "cs42888", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &imx_cs42888_surround_ops_be, + .be_hw_params_fixup = be_hw_params_fixup, + }, +}; + +static struct snd_soc_card snd_soc_card_imx_cs42888 = { + .name = "cs42888-audio", + .dai_link = imx_cs42888_dai, + .dapm_widgets = imx_cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + +/* + * This function will register the snd_soc_pcm_link drivers. + */ +static int imx_cs42888_probe(struct platform_device *pdev) +{ + struct device_node *esai_np, *codec_np; + struct device_node *asrc_np; + struct platform_device *esai_pdev; + struct platform_device *asrc_pdev = NULL; + struct i2c_client *codec_dev; + struct imx_priv *priv = &card_priv; + struct clk *codec_clk = NULL; + int ret; + u32 width; + + priv->pdev = pdev; + priv->asrc_pdev = NULL; + + esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!esai_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0); + if (asrc_np) { + asrc_pdev = of_find_device_by_node(asrc_np); + priv->asrc_pdev = asrc_pdev; + } + + esai_pdev = of_find_device_by_node(esai_np); + if (!esai_pdev) { + dev_err(&pdev->dev, "failed to find ESAI platform 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; + } + + /*if there is no asrc controller, we only enable one device*/ + if (!asrc_pdev) { + imx_cs42888_dai[0].codec_of_node = codec_np; + imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev); + imx_cs42888_dai[0].platform_of_node = esai_np; + snd_soc_card_imx_cs42888.num_links = 1; + } else { + imx_cs42888_dai[0].codec_of_node = codec_np; + imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev); + imx_cs42888_dai[0].platform_of_node = esai_np; + imx_cs42888_dai[1].cpu_of_node = asrc_np; + imx_cs42888_dai[1].platform_of_node = asrc_np; + imx_cs42888_dai[2].codec_of_node = codec_np; + imx_cs42888_dai[2].cpu_dai_name = dev_name(&esai_pdev->dev); + snd_soc_card_imx_cs42888.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; + } + + codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(codec_clk)) { + ret = PTR_ERR(codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + priv->mclk_freq = clk_get_rate(codec_clk); + + snd_soc_card_imx_cs42888.dev = &pdev->dev; + + platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888); + + ret = snd_soc_register_card(&snd_soc_card_imx_cs42888); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); +fail: + if (asrc_np) + of_node_put(asrc_np); + if (esai_np) + of_node_put(esai_np); + if (codec_np) + of_node_put(codec_np); + return ret; +} + +static int imx_cs42888_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&snd_soc_card_imx_cs42888); + return 0; +} + +static const struct of_device_id imx_cs42888_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { /* sentinel */ } +}; + +static struct platform_driver imx_cs42888_driver = { + .probe = imx_cs42888_probe, + .remove = imx_cs42888_remove, + .driver = { + .name = "imx-cs42888", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_cs42888_dt_ids, + }, +}; +module_platform_driver(imx_cs42888_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver"); +MODULE_ALIAS("platform:imx-cs42888"); +MODULE_LICENSE("GPL"); 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