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-rw-r--r--sound/soc/fsl/Kconfig26
-rw-r--r--sound/soc/fsl/Makefile4
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c574
-rw-r--r--sound/soc/fsl/fsl_asrc.c6
-rw-r--r--sound/soc/fsl/fsl_esai.c19
-rw-r--r--sound/soc/fsl/fsl_esai.h8
-rw-r--r--sound/soc/fsl/fsl_sai.c58
-rw-r--r--sound/soc/fsl/fsl_sai.h8
-rw-r--r--sound/soc/fsl/fsl_spdif.c6
-rw-r--r--sound/soc/fsl/fsl_ssi.c101
-rw-r--r--sound/soc/fsl/imx-es8328.c232
11 files changed, 983 insertions, 59 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f3012b645b51..081e406b3713 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -240,6 +240,18 @@ config SND_SOC_IMX_WM8962
Say Y if you want to add support for SoC audio on an i.MX board with
a wm8962 codec.
+config SND_SOC_IMX_ES8328
+ tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+ depends on OF && (I2C || SPI)
+ select SND_SOC_ES8328_I2C if I2C
+ select SND_SOC_ES8328_SPI if SPI_MASTER
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ help
+ Say Y if you want to add support for the ES8328 audio codec connected
+ via SSI/I2S over either SPI or I2C.
+
config SND_SOC_IMX_SGTL5000
tristate "SoC Audio support for i.MX boards with sgtl5000"
depends on OF && I2C
@@ -268,6 +280,20 @@ config SND_SOC_IMX_MC13783
select SND_SOC_MC13783
select SND_SOC_IMX_PCM_DMA
+config SND_SOC_FSL_ASOC_CARD
+ tristate "Generic ASoC Sound Card with ASRC support"
+ depends on OF && I2C
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_ESAI
+ select SND_SOC_FSL_SAI
+ select SND_SOC_FSL_SSI
+ 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.
+
endif # SND_IMX_SOC
endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 9ff59267eac9..d28dc25c9375 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-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
@@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
snd-soc-phycore-ac97-objs := phycore-ac97.o
snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-wm8962-objs := imx-wm8962.o
snd-soc-imx-spdif-objs := imx-spdif.o
@@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
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-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.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..007c772f3cef
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,574 @@
+/*
+ * 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 asrc_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 asrc_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 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(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+asrc_fail:
+ of_node_put(asrc_np);
+fail:
+ of_node_put(codec_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");
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 822110420b71..3b145313f93e 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static struct regmap_config fsl_asrc_regmap_config = {
+static const struct regmap_config fsl_asrc_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
asrc_priv->paddr = res->start;
- /* Register regmap and let it prepare core clock */
- if (of_property_read_bool(np, "big-endian"))
- fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
&fsl_asrc_regmap_config);
if (IS_ERR(asrc_priv->regmap)) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a3b29ed84963..8bcdfda09d7a 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -37,6 +37,7 @@
* @fsysclk: system clock source to derive HCK, SCK and FS
* @fifo_depth: depth of tx/rx FIFO
* @slot_width: width of each DAI slot
+ * @slots: number of slots
* @hck_rate: clock rate of desired HCKx clock
* @sck_rate: clock rate of desired SCKx clock
* @hck_dir: the direction of HCKx pads
@@ -55,6 +56,7 @@ struct fsl_esai {
struct clk *fsysclk;
u32 fifo_depth;
u32 slot_width;
+ u32 slots;
u32 hck_rate[2];
u32 sck_rate[2];
bool hck_dir[2];
@@ -362,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
esai_priv->slot_width = slot_width;
+ esai_priv->slots = slots;
return 0;
}
@@ -509,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
u32 width = snd_pcm_format_width(params_format(params));
u32 channels = params_channels(params);
+ u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
u32 bclk, mask, val;
int ret;
- bclk = params_rate(params) * esai_priv->slot_width * 2;
+ bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
ret = fsl_esai_set_bclk(dai, tx, bclk);
if (ret)
@@ -529,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
(tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
- (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+ (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
@@ -564,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
u8 i, channels = substream->runtime->channels;
+ u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -578,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
- tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+ tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
@@ -705,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static struct regmap_config fsl_esai_regmap_config = {
+static const struct regmap_config fsl_esai_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -731,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
esai_priv->pdev = pdev;
strcpy(esai_priv->name, np->name);
- if (of_property_read_bool(np, "big-endian"))
- fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
@@ -781,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
/* Set a default slot size */
esai_priv->slot_width = 32;
+ /* Set a default slot number */
+ esai_priv->slots = 2;
+
/* Set a default master/slave state */
esai_priv->slave_mode = true;
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 75e14033e8d8..91a550f4a10d 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -130,8 +130,8 @@
#define ESAI_xFCR_RE_WIDTH 4
#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
-#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
-#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
#define ESAI_xFCR_xFR_SHIFT 1
#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT)
#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT)
@@ -272,8 +272,8 @@
#define ESAI_xCR_RE_WIDTH 4
#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
-#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
-#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
/*
* Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index faa049797897..7eeb1dd8ce27 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
u32 val_cr2 = 0, val_cr4 = 0;
- if (!sai->big_endian_data)
+ if (!sai->is_lsb_first)
val_cr4 |= FSL_SAI_CR4_MF;
/* DAI mode */
@@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
val_cr5 |= FSL_SAI_CR5_WNW(word_width);
val_cr5 |= FSL_SAI_CR5_W0W(word_width);
- if (sai->big_endian_data)
+ if (sai->is_lsb_first)
val_cr5 |= FSL_SAI_CR5_FBT(0);
else
val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
@@ -330,13 +330,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
u32 xcsr, count = 100;
/*
- * The transmitter bit clock and frame sync are to be
- * used by both the transmitter and receiver.
+ * Asynchronous mode: Clear SYNC for both Tx and Rx.
+ * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
+ * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
*/
- regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
- ~FSL_SAI_CR2_SYNC);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
- FSL_SAI_CR2_SYNC);
+ sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
/*
* It is recommended that the transmitter is the last enabled
@@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
- regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0);
- regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0);
+ /* Software Reset for both Tx and Rx */
+ regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+ /* Clear SR bit to finish the reset */
+ regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+
regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
FSL_SAI_MAXBURST_TX * 2);
regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
@@ -539,7 +544,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static struct regmap_config fsl_sai_regmap_config = {
+static const struct regmap_config fsl_sai_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -568,11 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
sai->sai_on_imx = true;
- sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
- if (sai->big_endian_regs)
- fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
- sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+ sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
base = devm_ioremap_resource(&pdev->dev, res);
@@ -621,6 +622,33 @@ static int fsl_sai_probe(struct platform_device *pdev)
return ret;
}
+ /* Sync Tx with Rx as default by following old DT binding */
+ sai->synchronous[RX] = true;
+ sai->synchronous[TX] = false;
+ fsl_sai_dai.symmetric_rates = 1;
+ fsl_sai_dai.symmetric_channels = 1;
+ fsl_sai_dai.symmetric_samplebits = 1;
+
+ if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) &&
+ of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+ /* error out if both synchronous and asynchronous are present */
+ dev_err(&pdev->dev, "invalid binding for synchronous mode\n");
+ return -EINVAL;
+ }
+
+ if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) {
+ /* Sync Rx with Tx */
+ sai->synchronous[RX] = false;
+ sai->synchronous[TX] = true;
+ } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+ /* Discard all settings for asynchronous mode */
+ sai->synchronous[RX] = false;
+ sai->synchronous[TX] = false;
+ fsl_sai_dai.symmetric_rates = 0;
+ fsl_sai_dai.symmetric_channels = 0;
+ fsl_sai_dai.symmetric_samplebits = 0;
+ }
+
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0e6c9f595d75..34667209b607 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -48,6 +48,7 @@
/* SAI Transmit/Recieve Control Register */
#define FSL_SAI_CSR_TERE BIT(31)
#define FSL_SAI_CSR_FR BIT(25)
+#define FSL_SAI_CSR_SR BIT(24)
#define FSL_SAI_CSR_xF_SHIFT 16
#define FSL_SAI_CSR_xF_W_SHIFT 18
#define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT)
@@ -131,13 +132,16 @@ struct fsl_sai {
struct clk *bus_clk;
struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
- bool big_endian_regs;
- bool big_endian_data;
+ bool is_lsb_first;
bool is_dsp_mode;
bool sai_on_imx;
+ bool synchronous[2];
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct snd_dmaengine_dai_dma_data dma_params_tx;
};
+#define TX 1
+#define RX 0
+
#endif /* __FSL_SAI_H */
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 70acfe4a9bd5..9b791621294c 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -15,7 +15,6 @@
#include <linux/bitrev.h>
#include <linux/clk.h>
-#include <linux/clk-private.h>
#include <linux/module.h>
#include <linux/of_address.h>
#include <linux/of_device.h>
@@ -1040,7 +1039,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static struct regmap_config fsl_spdif_regmap_config = {
+static const struct regmap_config fsl_spdif_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -1184,9 +1183,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
spdif_priv->cpu_dai_drv.name = spdif_priv->name;
- if (of_property_read_bool(np, "big-endian"))
- fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 87eb5776a39b..e6955170dc42 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -169,6 +169,7 @@ struct fsl_ssi_private {
u8 i2s_mode;
bool use_dma;
bool use_dual_fifo;
+ bool has_ipg_clk_name;
unsigned int fifo_depth;
struct fsl_ssi_rxtx_reg_val rxtx_reg_val;
@@ -259,6 +260,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
SND_SOC_DAIFMT_CBS_CFS;
}
+static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private)
+{
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+ SND_SOC_DAIFMT_CBM_CFS;
+}
/**
* fsl_ssi_isr: SSI interrupt handler
*
@@ -525,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private =
snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ int ret;
+
+ ret = clk_prepare_enable(ssi_private->clk);
+ if (ret)
+ return ret;
/* When using dual fifo mode, it is safer to ensure an even period
* size. If appearing to an odd number while DMA always starts its
@@ -539,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
}
/**
+ * fsl_ssi_shutdown: shutdown the SSI
+ *
+ */
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private =
+ snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+ clk_disable_unprepare(ssi_private->clk);
+
+}
+
+/**
* fsl_ssi_set_bclk - configure Digital Audio Interface bit clock
*
* Note: This function can be only called when using SSI as DAI master
@@ -705,6 +731,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
}
}
+ if (!fsl_ssi_is_ac97(ssi_private)) {
+ u8 i2smode;
+ /*
+ * Switch to normal net mode in order to have a frame sync
+ * signal every 32 bits instead of 16 bits
+ */
+ if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16)
+ i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL |
+ CCSR_SSI_SCR_NET;
+ else
+ i2smode = ssi_private->i2s_mode;
+
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
+ channels == 1 ? 0 : i2smode);
+ }
+
/*
* FIXME: The documentation says that SxCCR[WL] should not be
* modified while the SSI is enabled. The only time this can
@@ -724,11 +767,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK,
wl);
- if (!fsl_ssi_is_ac97(ssi_private))
- regmap_update_bits(regs, CCSR_SSI_SCR,
- CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
- channels == 1 ? 0 : ssi_private->i2s_mode);
-
return 0;
}
@@ -748,8 +786,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
return 0;
}
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
- unsigned int fmt)
+static int _fsl_ssi_set_dai_fmt(struct device *dev,
+ struct fsl_ssi_private *ssi_private,
+ unsigned int fmt)
{
struct regmap *regs = ssi_private->regs;
u32 strcr = 0, stcr, srcr, scr, mask;
@@ -758,7 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
- dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
+ dev_err(dev, "baudclk is missing which is necessary for master mode\n");
return -EINVAL;
}
@@ -780,6 +819,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFS:
ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
regmap_update_bits(regs, CCSR_SSI_STCCR,
@@ -853,6 +893,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
case SND_SOC_DAIFMT_CBM_CFM:
scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ strcr &= ~CCSR_SSI_STCR_TXDIR;
+ strcr |= CCSR_SSI_STCR_TFDIR;
+ scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+ break;
default:
return -EINVAL;
}
@@ -913,7 +958,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
- return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
+ return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
}
/**
@@ -1020,6 +1065,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
.startup = fsl_ssi_startup,
+ .shutdown = fsl_ssi_shutdown,
.hw_params = fsl_ssi_hw_params,
.hw_free = fsl_ssi_hw_free,
.set_fmt = fsl_ssi_set_dai_fmt,
@@ -1145,17 +1191,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
u32 dmas[4];
int ret;
- ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
+ if (ssi_private->has_ipg_clk_name)
+ ssi_private->clk = devm_clk_get(&pdev->dev, "ipg");
+ else
+ ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
if (IS_ERR(ssi_private->clk)) {
ret = PTR_ERR(ssi_private->clk);
dev_err(&pdev->dev, "could not get clock: %d\n", ret);
return ret;
}
- ret = clk_prepare_enable(ssi_private->clk);
- if (ret) {
- dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
- return ret;
+ if (!ssi_private->has_ipg_clk_name) {
+ ret = clk_prepare_enable(ssi_private->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
+ return ret;
+ }
}
/* For those SLAVE implementations, we ingore non-baudclk cases
@@ -1213,8 +1264,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
return 0;
error_pcm:
- clk_disable_unprepare(ssi_private->clk);
+ if (!ssi_private->has_ipg_clk_name)
+ clk_disable_unprepare(ssi_private->clk);
return ret;
}
@@ -1223,7 +1275,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev,
{
if (!ssi_private->use_dma)
imx_pcm_fiq_exit(pdev);
- clk_disable_unprepare(ssi_private->clk);
+ if (!ssi_private->has_ipg_clk_name)
+ clk_disable_unprepare(ssi_private->clk);
}
static int fsl_ssi_probe(struct platform_device *pdev)
@@ -1262,9 +1315,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
if (sprop) {
if (!strcmp(sprop, "ac97-slave"))
ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97;
- else if (!strcmp(sprop, "i2s-slave"))
- ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_CBM_CFM;
}
ssi_private->use_dma = !of_property_read_bool(np,
@@ -1298,8 +1348,16 @@ static int fsl_ssi_probe(struct platform_device *pdev)
return -ENOMEM;
}
- ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
+ ret = of_property_match_string(np, "clock-names", "ipg");
+ if (ret < 0) {
+ ssi_private->has_ipg_clk_name = false;
+ ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
&fsl_ssi_regconfig);
+ } else {
+ ssi_private->has_ipg_clk_name = true;
+ ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
+ "ipg", iomem, &fsl_ssi_regconfig);
+ }
if (IS_ERR(ssi_private->regs)) {
dev_err(&pdev->dev, "Failed to init register map\n");
return PTR_ERR(ssi_private->regs);
@@ -1387,7 +1445,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done:
if (ssi_private->dai_fmt)
- _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
+ _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,
+ ssi_private->dai_fmt);
return 0;
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
new file mode 100644
index 000000000000..653e66d150c8
--- /dev/null
+++ b/sound/soc/fsl/imx-es8328.c
@@ -0,0 +1,232 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * 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/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+#define MUX_PORT_MAX 7
+
+struct imx_es8328_data {
+ struct device *dev;
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+ {
+ .gpio = -1,
+ .name = "headset-gpio",
+ .report = SND_JACK_HEADSET,
+ .invert = 0,
+ .debounce_time = 200,
+ },
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct imx_es8328_data *data = container_of(rtd->card,
+ struct imx_es8328_data, card);
+ int ret = 0;
+
+ /* Headphone jack detection */
+ if (gpio_is_valid(data->jack_gpio)) {
+ ret = snd_soc_jack_new(rtd->codec, "Headphone",
+ SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+ &headset_jack);
+ if (ret)
+ return ret;
+
+ headset_jack_gpios[0].gpio = data->jack_gpio;
+ ret = snd_soc_jack_add_gpios(&headset_jack,
+ ARRAY_SIZE(headset_jack_gpios),
+ headset_jack_gpios);
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
+ struct platform_device *ssi_pdev;
+ struct imx_es8328_data *data;
+ u32 int_port, ext_port;
+ int ret;
+ struct device *dev = &pdev->dev;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(dev, "mux-int-port missing or invalid\n");
+ goto fail;
+ }
+ if (int_port > MUX_PORT_MAX || int_port == 0) {
+ dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+ MUX_PORT_MAX);
+ goto fail;
+ }
+
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(dev, "mux-ext-port missing or invalid\n");
+ goto fail;
+ }
+ if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+ dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+ MUX_PORT_MAX);
+ goto fail;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ 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,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(dev, "failed to find SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->dev = dev;
+
+ data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+ data->dai.name = "hifi";
+ data->dai.stream_name = "hifi";
+ data->dai.codec_dai_name = "es8328-hifi-analog";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_of_node = ssi_np;
+ data->dai.platform_of_node = ssi_np;
+ data->dai.init = &imx_es8328_dai_init;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = dev;
+ data->card.dapm_widgets = imx_es8328_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret) {
+ dev_err(dev, "Unable to parse card name\n");
+ goto fail;
+ }
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret) {
+ dev_err(dev, "Unable to parse routing: %d\n", ret);
+ goto fail;
+ }
+ data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
+ data->card.dai_link = &data->dai;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(dev, "Unable to register: %d\n", ret);
+ goto fail;
+ }
+
+ platform_set_drvdata(pdev, data);
+fail:
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+ struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+ snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+ headset_jack_gpios);
+
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-es8328", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+ .driver = {
+ .name = "imx-es8328",
+ .of_match_table = imx_es8328_dt_ids,
+ },
+ .probe = imx_es8328_probe,
+ .remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");