(no subject)

[Xiubo Li]

Hello,

This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition.

This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree.

Changed in v2:
- Use default settings for the generic dmaengine PCM driver.
- Separate receive and transmit setting in most functions, but some couldn't for the HW limitation.
- Drop some not reduntant code.
- Use devm_snd_soc_register_component() instead of snd_soc_register_component().
- Use devm_snd_soc_register_card() instead of devm_snd_soc_register_card().
- Adjust the code sentences sequence.
- Make the namespacing consistent.
- Rename CONFIG_SND_SOC_FSL_SGTL5000 to CONFIG_SND_SOC_FSL_SGTL5000_VF610.
- Drop some meaningless lines.
- Rename the binding document file.

Added in v1:
- Add SAI SoC Digital Audio Interface driver.
- Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610.
- Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.
- Add device tree bindings for Freescale SAI.
- Revise the bugs about the sgt15000 codec.
- Add SGT15000 based audio machine driver.
- Enable SGT15000 codec based audio driver node for VF610.
- Add device tree bindings for Freescale VF610 sound.

[PATCHv2 1/8] ALSA: Add SAI SoC Digital Audio Interface driver.

[Xiubo Li]

This adds Freescale SAI ASoC Audio support.
This implementation is only compatible with device tree definition.
Features:
o Supports playback/capture
o Supports 16/20/24 bit PCM
o Supports 8k - 96k sample rates
o Supports slave mode only.

Signed-off-by: Alison Wang <b18965@freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 sound/soc/fsl/Kconfig   |  16 ++
 sound/soc/fsl/Makefile  |   5 +
 sound/soc/fsl/fsl-sai.c | 472 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl-sai.h | 120 ++++++++++++
 4 files changed, 613 insertions(+)
 create mode 100644 sound/soc/fsl/fsl-sai.c
 create mode 100644 sound/soc/fsl/fsl-sai.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index b7ab71f..9a8851e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -213,3 +213,19 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
 endif # SND_IMX_SOC
+
+menuconfig SND_FSL_SOC
+	tristate "SoC Audio for Freescale FSL CPUs"
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the FSL CPUs.
+
+	  This will enable Freeacale SAI, SGT15000 codec.
+
+if SND_FSL_SOC
+
+config SND_SOC_FSL_SAI
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8db705b..e5acc03 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -56,3 +56,8 @@ obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
+
+# FSL ARM SAI/SGT15000 Platform Support
+snd-soc-fsl-sai-objs := fsl-sai.o
+
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
diff --git a/sound/soc/fsl/fsl-sai.c b/sound/soc/fsl/fsl-sai.c
new file mode 100644
index 0000000..bb57e67
--- /dev/null
+++ b/sound/soc/fsl/fsl-sai.c
@@ -0,0 +1,472 @@
+/*
+ * Freescale SAI ALSA SoC Digital Audio Interface driver.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/of_address.h>
+#include <sound/core.h>
+#include <sound/pcm_params.h>
+#include <linux/delay.h>
+#include <linux/dmaengine.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "fsl-sai.h"
+
+static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
+		int clk_id, unsigned int freq, int fsl_dir)
+{
+	u32 val_cr2, reg_cr2;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (fsl_dir == FSL_FMT_TRANSMITTER)
+		reg_cr2 = FSL_SAI_TCR2;
+	else
+		reg_cr2 = FSL_SAI_RCR2;
+
+	val_cr2 = readl(sai->base + reg_cr2);
+	switch (clk_id) {
+	case FSL_SAI_CLK_BUS:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_BUS;
+		break;
+	case FSL_SAI_CLK_MAST1:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1;
+		break;
+	case FSL_SAI_CLK_MAST2:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2;
+		break;
+	case FSL_SAI_CLK_MAST3:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3;
+		break;
+	default:
+		return -EINVAL;
+	}
+	writel(val_cr2, sai->base + reg_cr2);
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	int ret;
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+					FSL_FMT_TRANSMITTER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set sai's transmitter sysclk: %d\n",
+				ret);
+		return ret;
+	}
+
+	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+					FSL_FMT_RECEIVER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set sai's receiver sysclk: %d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+		int div_id, int div)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 tcr2, rcr2;
+
+	if (div_id == FSL_SAI_TX_DIV) {
+		tcr2 = readl(sai->base + FSL_SAI_TCR2);
+		tcr2 &= ~FSL_SAI_CR2_DIV_MASK;
+		tcr2 |= FSL_SAI_CR2_DIV(div);
+		writel(tcr2, sai->base + FSL_SAI_TCR2);
+
+	} else if (div_id == FSL_SAI_RX_DIV) {
+		rcr2 = readl(sai->base + FSL_SAI_RCR2);
+		rcr2 &= ~FSL_SAI_CR2_DIV_MASK;
+		rcr2 |= FSL_SAI_CR2_DIV(div);
+		writel(rcr2, sai->base + FSL_SAI_RCR2);
+
+	} else
+		return -EINVAL;
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
+				unsigned int fmt, int fsl_dir)
+{
+	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (fsl_dir == FSL_FMT_TRANSMITTER) {
+		reg_cr2 = FSL_SAI_TCR2;
+		reg_cr3 = FSL_SAI_TCR3;
+		reg_cr4 = FSL_SAI_TCR4;
+	} else {
+		reg_cr2 = FSL_SAI_RCR2;
+		reg_cr3 = FSL_SAI_RCR3;
+		reg_cr4 = FSL_SAI_RCR4;
+	}
+
+	val_cr2 = readl(sai->base + reg_cr2);
+	val_cr3 = readl(sai->base + reg_cr3);
+	val_cr4 = readl(sai->base + reg_cr4);
+
+	if (sai->fbt == FSL_SAI_FBT_MSB)
+		val_cr4 |= FSL_SAI_CR4_MF;
+	else if (sai->fbt == FSL_SAI_FBT_LSB)
+		val_cr4 &= ~FSL_SAI_CR4_MF;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val_cr4 |= FSL_SAI_CR4_FSE;
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		val_cr2 &= ~FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		val_cr4 &= ~FSL_SAI_CR4_FSP;
+		val_cr2 &= ~FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		val_cr2 |= FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		val_cr4 &= ~FSL_SAI_CR4_FSP;
+		val_cr2 |= FSL_SAI_CR2_BCP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
+		val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val_cr3 |= FSL_SAI_CR3_TRCE;
+
+	if (fsl_dir == FSL_FMT_RECEIVER)
+		val_cr2 |= FSL_SAI_CR2_SYNC;
+
+	writel(val_cr2, sai->base + reg_cr2);
+	writel(val_cr3, sai->base + reg_cr3);
+	writel(val_cr4, sai->base + reg_cr4);
+
+	return 0;
+
+}
+
+static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	int ret;
+
+	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set sai's transmitter format: %d\n",
+				ret);
+		return ret;
+	}
+
+	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set sai's receiver format: %d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 tcr4, rcr4;
+
+	tcr4 = readl(sai->base + FSL_SAI_TCR4);
+	tcr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+	tcr4 |= FSL_SAI_CR4_FRSZ(2);
+	writel(tcr4, sai->base + FSL_SAI_TCR4);
+	writel(tx_mask, sai->base + FSL_SAI_TMR);
+
+	rcr4 = readl(sai->base + FSL_SAI_RCR4);
+	rcr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+	rcr4 |= FSL_SAI_CR4_FRSZ(2);
+	writel(rcr4, sai->base + FSL_SAI_RCR4);
+	writel(rx_mask, sai->base + FSL_SAI_RMR);
+
+	return 0;
+}
+
+static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *cpu_dai)
+{
+	u32 val_cr4, val_cr5, reg_cr4, reg_cr5, word_width;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		reg_cr4 = FSL_SAI_TCR4;
+		reg_cr5 = FSL_SAI_TCR5;
+	} else {
+		reg_cr4 = FSL_SAI_RCR4;
+		reg_cr5 = FSL_SAI_RCR5;
+	}
+
+	val_cr4 = readl(sai->base + reg_cr4);
+	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
+
+	val_cr5 = readl(sai->base + reg_cr5);
+	val_cr5 &= ~FSL_SAI_CR5_WNW_MASK;
+	val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
+	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_width = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		word_width = 20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		word_width = 24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
+	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
+
+	if (sai->fbt == FSL_SAI_FBT_MSB)
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+	else if (sai->fbt == FSL_SAI_FBT_LSB)
+		val_cr5 |= FSL_SAI_CR5_FBT(0);
+
+	writel(val_cr4, sai->base + reg_cr4);
+	writel(val_cr5, sai->base + reg_cr5);
+
+	return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
+	unsigned int tcsr, rcsr;
+
+	tcsr = readl(sai->base + FSL_SAI_TCSR);
+	rcsr = readl(sai->base + FSL_SAI_RCSR);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		tcsr |= FSL_SAI_CSR_FRDE;
+		rcsr &= ~FSL_SAI_CSR_FRDE;
+	} else {
+		rcsr |= FSL_SAI_CSR_FRDE;
+		tcsr &= ~FSL_SAI_CSR_FRDE;
+	}
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		tcsr |= FSL_SAI_CSR_TERE;
+		rcsr |= FSL_SAI_CSR_TERE;
+		writel(rcsr, sai->base + FSL_SAI_RCSR);
+		udelay(10);
+		writel(tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!(dai->playback_active || dai->capture_active)) {
+			tcsr &= ~FSL_SAI_CSR_TERE;
+			rcsr &= ~FSL_SAI_CSR_TERE;
+		}
+		writel(rcsr, sai->base + FSL_SAI_RCSR);
+		udelay(10);
+		writel(tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_sysclk	= fsl_sai_set_dai_sysclk,
+	.set_clkdiv	= fsl_sai_set_dai_clkdiv,
+	.set_fmt	= fsl_sai_set_dai_fmt,
+	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
+	.hw_params	= fsl_sai_hw_params,
+	.trigger	= fsl_sai_trigger,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *dai)
+{
+	int ret;
+	struct fsl_sai *sai = dev_get_drvdata(dai->dev);
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	writel(0x0, sai->base + FSL_SAI_RCSR);
+	writel(0x0, sai->base + FSL_SAI_TCSR);
+	writel(sai->dma_params_tx.maxburst, sai->base + FSL_SAI_TCR1);
+	writel(sai->dma_params_rx.maxburst, sai->base + FSL_SAI_RCR1);
+
+	dai->playback_dma_data = &sai->dma_params_tx;
+	dai->capture_dma_data = &sai->dma_params_rx;
+
+	snd_soc_dai_set_drvdata(dai, sai);
+
+	return 0;
+}
+
+int fsl_sai_dai_remove(struct snd_soc_dai *dai)
+{
+	struct fsl_sai *sai = dev_get_drvdata(dai->dev);
+
+	clk_disable_unprepare(sai->clk);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver fsl_sai_dai = {
+	.probe = fsl_sai_dai_probe,
+	.remove = fsl_sai_dai_remove,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = FSL_SAI_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = FSL_SAI_FORMATS,
+	},
+	.ops = &fsl_sai_pcm_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_component = {
+	.name           = "fsl-sai",
+};
+
+static int fsl_sai_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct fsl_sai *sai;
+	int ret = 0;
+
+	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+	if (!sai)
+		return -ENOMEM;
+
+	sai->fbt = FSL_SAI_FBT_MSB;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	sai->base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(sai->base))
+		return PTR_ERR(sai->base);
+
+	sai->clk = devm_clk_get(&pdev->dev, "sai");
+	if (IS_ERR(sai->clk)) {
+		dev_err(&pdev->dev, "Cannot get sai's clock: %d\n", ret);
+		return PTR_ERR(sai->clk);
+	}
+
+	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
+	sai->dma_params_rx.maxburst = 6;
+
+	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
+	sai->dma_params_tx.maxburst = 6;
+
+	platform_set_drvdata(pdev, sai);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+			&fsl_sai_dai, 1);
+	if (ret)
+		return ret;
+
+	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static int fsl_sai_remove(struct platform_device *pdev)
+{
+	snd_dmaengine_pcm_unregister(&pdev->dev);
+
+	return 0;
+}
+
+static const struct of_device_id fsl_sai_ids[] = {
+	{ .compatible = "fsl,vf610-sai", },
+	{ /*sentinel*/ }
+};
+
+static struct platform_driver fsl_sai_driver = {
+	.probe = fsl_sai_probe,
+	.remove = fsl_sai_remove,
+
+	.driver = {
+		.name = "fsl-sai",
+		.owner = THIS_MODULE,
+		.of_match_table = fsl_sai_ids,
+	},
+};
+module_platform_driver(fsl_sai_driver);
+
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_AUTHOR("Alison Wang, <b18965@freescale.com>");
+MODULE_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl-sai.h b/sound/soc/fsl/fsl-sai.h
new file mode 100644
index 0000000..1637679
--- /dev/null
+++ b/sound/soc/fsl/fsl-sai.h
@@ -0,0 +1,120 @@
+/*
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __FSL_SAI_H
+#define __FSL_SAI_H
+
+#include <sound/dmaengine_pcm.h>
+
+#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE)
+
+/* SAI Transmit/Recieve Control Register */
+#define FSL_SAI_TCSR		0x00
+#define FSL_SAI_RCSR		0x80
+#define FSL_SAI_CSR_TERE	BIT(31)
+#define FSL_SAI_CSR_FWF		BIT(17)
+#define FSL_SAI_CSR_FRIE	BIT(8)
+#define FSL_SAI_CSR_FRDE	BIT(0)
+
+/* SAI Transmit Data/FIFO/MASK Register */
+#define FSL_SAI_TDR		0x20
+#define FSL_SAI_TFR		0x40
+#define FSL_SAI_TMR		0x60
+
+/* SAI Recieve Data/FIFO/MASK Register */
+#define FSL_SAI_RDR		0xa0
+#define FSL_SAI_RFR		0xc0
+#define FSL_SAI_RMR		0xe0
+
+/* SAI Transmit and Recieve Configuration 1 Register */
+#define FSL_SAI_TCR1		0x04
+#define FSL_SAI_RCR1		0x84
+
+/* SAI Transmit and Recieve Configuration 2 Register */
+#define FSL_SAI_TCR2		0x08
+#define FSL_SAI_RCR2		0x88
+#define FSL_SAI_CR2_SYNC	BIT(30)
+#define FSL_SAI_CR2_MSEL_MASK	(0xff << 26)
+#define FSL_SAI_CR2_MSEL_BUS	0
+#define FSL_SAI_CR2_MSEL_MCLK1	BIT(26)
+#define FSL_SAI_CR2_MSEL_MCLK2	BIT(27)
+#define FSL_SAI_CR2_MSEL_MCLK3	(BIT(26) | BIT(27))
+#define FSL_SAI_CR2_BCP		BIT(25)
+#define FSL_SAI_CR2_BCD_MSTR	BIT(24)
+#define FSL_SAI_CR2_DIV(x)	(x)
+#define FSL_SAI_CR2_DIV_MASK	0xff
+
+/* SAI Transmit and Recieve Configuration 3 Register */
+#define FSL_SAI_TCR3		0x0c
+#define FSL_SAI_RCR3		0x8c
+#define FSL_SAI_CR3_TRCE	BIT(16)
+#define FSL_SAI_CR3_WDFL(x)	(x)
+#define FSL_SAI_CR3_WDFL_MASK	0x1f
+
+/* SAI Transmit and Recieve Configuration 4 Register */
+#define FSL_SAI_TCR4		0x10
+#define FSL_SAI_RCR4		0x90
+#define FSL_SAI_CR4_FRSZ(x)	(((x) - 1) << 16)
+#define FSL_SAI_CR4_FRSZ_MASK	(0x1f << 16)
+#define FSL_SAI_CR4_SYWD(x)	(((x) - 1) << 8)
+#define FSL_SAI_CR4_SYWD_MASK	(0x1f << 8)
+#define FSL_SAI_CR4_MF		BIT(4)
+#define FSL_SAI_CR4_FSE		BIT(3)
+#define FSL_SAI_CR4_FSP		BIT(1)
+#define FSL_SAI_CR4_FSD_MSTR	BIT(0)
+
+/* SAI Transmit and Recieve Configuration 5 Register */
+#define FSL_SAI_TCR5		0x14
+#define FSL_SAI_RCR5		0x94
+#define FSL_SAI_CR5_WNW(x)	(((x) - 1) << 24)
+#define FSL_SAI_CR5_WNW_MASK	(0x1f << 24)
+#define FSL_SAI_CR5_W0W(x)	(((x) - 1) << 16)
+#define FSL_SAI_CR5_W0W_MASK	(0x1f << 16)
+#define FSL_SAI_CR5_FBT(x)	((x) << 8)
+#define FSL_SAI_CR5_FBT_MASK	(0x1f << 8)
+
+/* SAI audio dividers */
+#define FSL_SAI_TX_DIV		0
+#define FSL_SAI_RX_DIV		1
+
+/* SAI type */
+#define FSL_SAI_DMA		BIT(0)
+#define FSL_SAI_USE_AC97	BIT(1)
+#define FSL_SAI_NET		BIT(2)
+#define FSL_SAI_TRA_SYN		BIT(3)
+#define FSL_SAI_REC_SYN		BIT(4)
+#define FSL_SAI_USE_I2S_SLAVE	BIT(5)
+
+#define FSL_FMT_TRANSMITTER	0
+#define FSL_FMT_RECEIVER	1
+
+/* SAI clock sources */
+#define FSL_SAI_CLK_BUS		0
+#define FSL_SAI_CLK_MAST1	1
+#define FSL_SAI_CLK_MAST2	2
+#define FSL_SAI_CLK_MAST3	3
+
+enum fsl_sai_fbt {
+	FSL_SAI_FBT_MSB,
+	FSL_SAI_FBT_LSB,
+};
+
+struct fsl_sai {
+	struct clk *clk;
+
+	void __iomem *base;
+
+	enum fsl_sai_fbt fbt;
+
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+};
+
+#endif /* __FSL_SAI_H */
-- 
1.8.4

[PATCHv2 2/8] ARM: dts: Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610.

[Xiubo Li]

This patch add the SAI's edma mux Tx and Rx support.

Signed-off-by: Jingchang Lu <b35083@freescale.com>
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 arch/arm/boot/dts/vf610.dtsi | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi
index 18e3a4c..391f180 100644
--- a/arch/arm/boot/dts/vf610.dtsi
+++ b/arch/arm/boot/dts/vf610.dtsi
@@ -151,9 +151,11 @@
 			sai2: sai@40031000 {
 				compatible = "fsl,vf610-sai";
 				reg = <0x40031000 0x1000>;
-				interrupts = <0 86 0x04>;
 				clocks = <&clks VF610_CLK_SAI2>;
 				clock-names = "sai";
+				dma-names = "tx", "rx";
+				dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
+					<&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
 				status = "disabled";
 			};
 
-- 
1.8.4

[PATCHv2 3/8] ARM: dts: Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.

[Xiubo Li]

This patch add and enable SAI device.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 arch/arm/boot/dts/vf610-twr.dts | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/arch/arm/boot/dts/vf610-twr.dts b/arch/arm/boot/dts/vf610-twr.dts
index 1a58678..e4106dd 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -57,6 +57,12 @@
 	status = "okay";
 };
 
+&sai2 {
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_sai2_1>;
+	status = "okay";
+};
+
 &uart1 {
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_uart1_1>;
-- 
1.8.4

[PATCHv2 4/8] Documentation: Add device tree bindings for Freescale SAI.

[Xiubo Li]

This adds the Document for Freescale SAI driver under
Documentation/devicetree/bindings/sound/.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 .../devicetree/bindings/sound/fsl-sai.txt          | 32 ++++++++++++++++++++++
 1 file changed, 32 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt

diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
new file mode 100644
index 0000000..267afbd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,32 @@
+Freescale Synchronous Audio Interface (SAI).
+
+The SAI is based on I2S module that used communicating with audio codecs,
+which provides a synchronous audio interface that supports fullduplex
+serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+codec/DSP interfaces.
+
+
+Required properties:
+- compatible: Compatible list, contains "fsl,vf610-sai".
+- reg: Offset and length of the register set for the device.
+- clocks: Must contain an entry for each entry in clock-names.
+- clock-names : Must include the "sai" entry.
+- dmas : Generic dma devicetree binding as described in
+  Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names : Two dmas have to be defined, "tx" and "rx".
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
+  See ../pinctrl/pinctrl-bindings.txt for details of the property values.
+
+Example:
+sai2: sai@40031000 {
+	      compatible = "fsl,vf610-sai";
+	      reg = <0x40031000 0x1000>;
+	      pinctrl-names = "default";
+	      pinctrl-0 = <&pinctrl_sai2_1>;
+	      clocks = <&clks VF610_CLK_SAI2>;
+	      clock-names = "sai";
+	      dma-names = "tx", "rx";
+	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
+		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
+};
-- 
1.8.4

[PATCHv2 5/8] ASoC: SGTL5000: Enhance the SGTL5000 codec driver about regulator.

[Xiubo Li]

On VF610 series there are no regulators used, and now whether the
CONFIG_REGULATOR mirco is enabled or not, for the VF610 audio
patch series, the board cannot be probe successfully.
And this patch will solve this issue.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 sound/soc/codecs/sgtl5000.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 1f4093f..c2f6d86 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -61,6 +61,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
 };
 
+#ifdef CONFIG_REGULATOR
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
 enum sgtl5000_regulator_supplies {
 	VDDA,
@@ -93,6 +94,9 @@ static struct regulator_init_data ldo_init_data = {
 	.num_consumer_supplies = 1,
 	.consumer_supplies = &ldo_consumer[0],
 };
+#else
+#define SGTL5000_SUPPLY_NUM 0
+#endif
 
 /*
  * sgtl5000 internal ldo regulator,
@@ -112,7 +116,9 @@ struct sgtl5000_priv {
 	int master;	/* i2s master or not */
 	int fmt;	/* i2s data format */
 	struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM];
+#ifdef CONFIG_REGULATOR
 	struct ldo_regulator *ldo;
+#endif
 	struct regmap *regmap;
 	struct clk *mclk;
 };
@@ -879,6 +885,7 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 #else
+#ifndef CONFIG_SND_SOC_FSL_SGTL5000_VF610
 static int ldo_regulator_register(struct snd_soc_codec *codec,
 				struct regulator_init_data *init_data,
 				int voltage)
@@ -886,6 +893,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
 	dev_err(codec->dev, "this setup needs regulator support in the kernel\n");
 	return -EINVAL;
 }
+#endif
 
 static int ldo_regulator_remove(struct snd_soc_codec *codec)
 {
@@ -1137,6 +1145,7 @@ static int sgtl5000_resume(struct snd_soc_codec *codec)
 #define sgtl5000_resume  NULL
 #endif	/* CONFIG_SUSPEND */
 
+#ifdef CONFIG_REGULATOR
 /*
  * sgtl5000 has 3 internal power supplies:
  * 1. VAG, normally set to vdda/2
@@ -1373,6 +1382,7 @@ err_regulator_free:
 	return ret;
 
 }
+#endif
 
 static int sgtl5000_probe(struct snd_soc_codec *codec)
 {
@@ -1387,6 +1397,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
+#ifdef CONFIG_REGULATOR
 	ret = sgtl5000_enable_regulators(codec);
 	if (ret)
 		return ret;
@@ -1395,6 +1406,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	ret = sgtl5000_set_power_regs(codec);
 	if (ret)
 		goto err;
+#endif
 
 	/* enable small pop, introduce 400ms delay in turning off */
 	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
-- 
1.8.4

[PATCHv2 6/8] ASoC: fsl: add SGTL5000 based audio machine driver.

[Xiubo Li]

This is the SGTL5000 codec based audio driver supported with both
playback and capture dai link implemention.

This implementation is only compatible with device tree definition.

Signed-off-by: Alison Wang <b18965@freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>

Conflicts:
	sound/soc/fsl/Makefile
---
 sound/soc/fsl/Kconfig              |  10 ++
 sound/soc/fsl/Makefile             |   2 +
 sound/soc/fsl/fsl-sgtl5000-vf610.c | 208 +++++++++++++++++++++++++++++++++++++
 3 files changed, 220 insertions(+)
 create mode 100644 sound/soc/fsl/fsl-sgtl5000-vf610.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9a8851e..1b835ba 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -228,4 +228,14 @@ config SND_SOC_FSL_SAI
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
+config SND_SOC_FSL_SGTL5000_VF610
+	tristate "SoC Audio support for FSL boards with sgtl5000"
+	depends on OF && I2C
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_PCM
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on an FSL board with
+	  a sgtl5000 codec.
+
 endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index e5acc03..26fc551 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -59,5 +59,7 @@ obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
 
 # FSL ARM SAI/SGT15000 Platform Support
 snd-soc-fsl-sai-objs := fsl-sai.o
+snd-soc-fsl-sgtl5000-vf610-objs := fsl-sgtl5000-vf610.o
 
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
+obj-$(CONFIG_SND_SOC_FSL_SGTL5000_VF610) += snd-soc-fsl-sgtl5000-vf610.o
diff --git a/sound/soc/fsl/fsl-sgtl5000-vf610.c b/sound/soc/fsl/fsl-sgtl5000-vf610.c
new file mode 100644
index 0000000..f535b42
--- /dev/null
+++ b/sound/soc/fsl/fsl-sgtl5000-vf610.c
@@ -0,0 +1,208 @@
+/*
+ * Freeacale ALSA SoC Audio using SGT1500 as codec.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * 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:
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl-sai.h"
+
+static unsigned int sysclk_rate;
+
+static int fsl_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret;
+	struct device *dev = rtd->card->dev;
+
+	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+				     sysclk_rate, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "could not set codec driver clock params :%d\n",
+				ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_BUS,
+				     sysclk_rate, SND_SOC_CLOCK_OUT);
+	if (ret) {
+		dev_err(dev, "could not set cpu dai driver clock params :%d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int sgtl5000_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;
+	unsigned int channels = params_channels(params);
+
+	/* TODO: The SAI driver should figure this out for us */
+	switch (channels) {
+	case 2:
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffc, 0xfffffffc, 2, 0);
+		break;
+	case 1:
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffe, 0xfffffffe, 1, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops fsl_sgtl5000_hifi_ops = {
+	.hw_params = sgtl5000_params,
+};
+
+static struct snd_soc_dai_link fsl_sgtl5000_dai = {
+	.name = "HiFi",
+	.stream_name = "HiFi",
+	.codec_dai_name = "sgtl5000",
+	.init = &fsl_sgtl5000_dai_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &fsl_sgtl5000_hifi_ops,
+};
+
+static const struct snd_soc_dapm_widget fsl_sgtl5000_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static struct snd_soc_card fsl_sgt1500_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+	.dai_link = &fsl_sgtl5000_dai,
+	.dapm_widgets = fsl_sgtl5000_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(fsl_sgtl5000_dapm_widgets),
+};
+
+static int fsl_sgtl5000_parse_dt(struct platform_device *pdev)
+{
+	int ret;
+	struct device_node *sai_np, *codec_np;
+	struct clk *codec_clk;
+	struct i2c_client *codec_dev;
+	struct device_node *np = pdev->dev.of_node;
+
+	ret = snd_soc_of_parse_card_name(&fsl_sgt1500_card, "model");
+	if (ret)
+		return ret;
+
+	ret = snd_soc_of_parse_audio_routing(&fsl_sgt1500_card,
+			"audio-routing");
+	if (ret)
+		return ret;
+
+	sai_np = of_parse_phandle(np, "saif-controller", 0);
+	if (!sai_np) {
+		dev_err(&pdev->dev, "\"saif-controller\" phandle missing or "
+				"invalid\n");
+		return -EINVAL;
+	}
+	fsl_sgtl5000_dai.cpu_of_node = sai_np;
+	fsl_sgtl5000_dai.platform_of_node = sai_np;
+
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "\"audio-codec\" phandle missing or "
+				"invalid\n");
+		ret = -EINVAL;
+		goto sai_np_fail;
+	}
+	fsl_sgtl5000_dai.codec_of_node = codec_np;
+
+	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 = PTR_ERR(codec_dev);
+		goto codec_np_fail;
+	}
+
+	codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+	if (IS_ERR(codec_clk)) {
+		dev_err(&pdev->dev, "failed to get codec clock\n");
+		ret = PTR_ERR(codec_clk);
+		goto codec_np_fail;
+	}
+
+	sysclk_rate = clk_get_rate(codec_clk);
+
+codec_np_fail:
+	of_node_put(codec_np);
+sai_np_fail:
+	of_node_put(sai_np);
+
+	return ret;
+}
+
+static int fsl_sgtl5000_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	fsl_sgt1500_card.dev = &pdev->dev;
+
+	ret = fsl_sgtl5000_parse_dt(pdev);
+	if (ret) {
+		dev_err(&pdev->dev,
+				"parse sgtl5000 device tree failed :%d\n",
+				ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &fsl_sgt1500_card);
+	if (ret) {
+		dev_err(&pdev->dev, "register soc sound card failed :%d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_sgtl5000_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&fsl_sgt1500_card);
+
+	return 0;
+}
+
+static const struct of_device_id fsl_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,vf610-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_sgtl5000_dt_ids);
+
+static struct platform_driver fsl_sgtl5000_driver = {
+	.driver = {
+		.name = "fsl-sgtl5000",
+		.owner = THIS_MODULE,
+		.of_match_table = fsl_sgtl5000_dt_ids,
+	},
+	.probe = fsl_sgtl5000_probe,
+	.remove = fsl_sgtl5000_remove,
+};
+module_platform_driver(fsl_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");
-- 
1.8.4

[PATCHv2 7/8] ARM: dts: Enable SGTL5000 codec based audio driver node for VF610.

[Xiubo Li]

This patch add and enable SGTL5000 codec support, and also specified
the corresponding SAI node.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Alison Wang <b18965@freescale.com
---
 arch/arm/boot/dts/vf610-twr.dts | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

diff --git a/arch/arm/boot/dts/vf610-twr.dts b/arch/arm/boot/dts/vf610-twr.dts
index e4106dd..a2d9214 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -34,6 +34,19 @@
 		};
 	};
 
+	sound {
+		compatible = "fsl,vf610-sgtl5000";
+		model = "vf610-sgtl5000";
+		saif-controller = <&sai2>;
+		audio-codec = <&codec>;
+		audio-routing =
+			"MIC_IN", "Mic Jack",
+			"Mic Jack", "Mic Bias",
+			"LINE_IN", "Line In Jack",
+			"Headphone Jack", "HP_OUT",
+			"Ext Spk", "LINE_OUT";
+	};
+
 };
 
 &fec0 {
@@ -55,6 +68,12 @@
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_i2c0_1>;
 	status = "okay";
+
+	codec: sgtl5000@0a {
+		compatible = "fsl,sgtl5000";
+		reg = <0x0a>;
+		clocks = <&clks VF610_CLK_SAI2>;
+	};
 };
 
 &sai2 {
-- 
1.8.4

[PATCHv2 8/8] Documentation: Add device tree bindings for Freescale VF610 sound.

[Xiubo Li]

This adds the Document for Freescale VF610 sound driver under
Documentation/devicetree/bindings/sound/.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 .../bindings/sound/fsl_audio_sgt15000_vf610.txt    | 46 ++++++++++++++++++++++
 1 file changed, 46 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/fsl_audio_sgt15000_vf610.txt

diff --git a/Documentation/devicetree/bindings/sound/fsl_audio_sgt15000_vf610.txt b/Documentation/devicetree/bindings/sound/fsl_audio_sgt15000_vf610.txt
new file mode 100644
index 0000000..76ff838
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl_audio_sgt15000_vf610.txt
@@ -0,0 +1,46 @@
+Freescale VF610 audio complex with SGTL5000 codec
+
+Required properties:
+- compatible: "fsl,vf610-sgtl5000"
+- model: The user-visible name of this sound complex.
+- saif-controllers: The phandle list of the SAI controller.
+- audio-codec: The phandle of the SGTL5000 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. Valid names could be power
+  supplies, SGTL5000 pins, and the jacks on the board:
+
+  -- Power supplies:
+     * Mic Bias
+
+  -- Board connectors:
+     * Mic Jack
+     * Line In Jack
+     * Headphone Jack
+     * Line Out Jack
+     * Ext Spk
+
+Example:
+
+sound {
+	compatible = "fsl,vf610-sgtl5000";
+	model = "vf610-sgtl5000";
+	saif-controller = <&sai2>;
+	audio-codec = <&codec>;
+	audio-routing =
+		"MIC_IN", "Mic Jack",
+		"Mic Jack", "Mic Bias",
+		"LINE_IN", "Line In Jack",
+		"Headphone Jack", "HP_OUT",
+		"Ext Spk", "LINE_OUT";
+};
+
+&i2c0 {
+	...
+
+	codec: sgtl5000@0a {
+	       compatible = "fsl,sgtl5000";
+	       reg = <0x0a>;
+	       clocks = <&clks VF610_CLK_SAI2>;
+       };
+};
-- 
1.8.4

Re: [PATCHv2 1/8] ALSA: Add SAI SoC Digital Audio Interface driver.

[Nicolin Chen]

Hi Xiubo,

On Fri, Nov 01, 2013 at 03:04:48PM +0800, Xiubo Li wrote:
> This adds Freescale SAI ASoC Audio support.
> This implementation is only compatible with device tree definition.
> Features:
> o Supports playback/capture
> o Supports 16/20/24 bit PCM
> o Supports 8k - 96k sample rates
> o Supports slave mode only.
> 

Just for curiosity, I found there're quite a bit code actually support
I2S master mode such as set_sysclk() and register configuration to FMT
SND_SOC_DAIFMT_CBS_CFS.

Is that really "supports slave mode only"? Or just you haven't make
it properly work?

> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
> index b7ab71f..9a8851e 100644
> --- a/sound/soc/fsl/Kconfig
> +++ b/sound/soc/fsl/Kconfig
> @@ -213,3 +213,19 @@ config SND_SOC_IMX_MC13783
>  	select SND_SOC_IMX_PCM_DMA
>  
>  endif # SND_IMX_SOC
> +
> +menuconfig SND_FSL_SOC
> +	tristate "SoC Audio for Freescale FSL CPUs"
> +	help
> +	  Say Y or M if you want to add support for codecs attached to
> +	  the FSL CPUs.
> +
> +	  This will enable Freeacale SAI, SGT15000 codec.
> +
> +if SND_FSL_SOC

Why check this here? SAI should be a regular IP module which can be used by
other SoC as well. We would never know if next i.MX SoC won't contain SAI.

> +
> +config SND_SOC_FSL_SAI
> +	tristate
> +	select SND_SOC_GENERIC_DMAENGINE_PCM

I think you should keep SAI beside SSI/SPDIF directly.

> +
> +endif # SND_FSL_SOC
> diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
> index 8db705b..e5acc03 100644
> --- a/sound/soc/fsl/Makefile
> +++ b/sound/soc/fsl/Makefile
> @@ -56,3 +56,8 @@ obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
>  obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
>  obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
>  obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
> +
> +# FSL ARM SAI/SGT15000 Platform Support

Should be SGTL5000, not SGT1. And SAI should not be used with SGTL5000 only,
it's a bit confusing to mention SGTL5000 here.

> +snd-soc-fsl-sai-objs := fsl-sai.o

And I think it should be better to put it along with fsl-ssi.o and fsl-spdif.o

> +
> +obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o

Ditto

> diff --git a/sound/soc/fsl/fsl-sai.c b/sound/soc/fsl/fsl-sai.c
> new file mode 100644
> index 0000000..bb57e67
> --- /dev/null
> +++ b/sound/soc/fsl/fsl-sai.c
> @@ -0,0 +1,472 @@
> +/*
> + * Freescale SAI ALSA SoC Digital Audio Interface driver.
> + *
> + * Copyright 2012-2013 Freescale Semiconductor, Inc.
> + *
> + * This program is free software; you can redistribute  it and/or modify it
> + * under  the terms of  the GNU General  Public License as published by the
> + * Free Software Foundation;  either version 2 of the  License, or (at your
> + * option) any later version.

There're too many double space inside. Could you pls revise it?

> + *
> + */
> +
> +#include <linux/clk.h>
> +#include <linux/module.h>
> +#include <linux/slab.h>
> +#include <linux/of_address.h>
> +#include <sound/core.h>
> +#include <sound/pcm_params.h>
> +#include <linux/delay.h>
> +#include <linux/dmaengine.h>
> +#include <sound/dmaengine_pcm.h>

I think it's better to keep <sound/xxx.h> together. And basically
we can keep header files ordered by alphabet.

> +
> +#include "fsl-sai.h"
> +
> +static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
> +		int clk_id, unsigned int freq, int fsl_dir)
> +{
> +	u32 val_cr2, reg_cr2;
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +
> +	if (fsl_dir == FSL_FMT_TRANSMITTER)
> +		reg_cr2 = FSL_SAI_TCR2;
> +	else
> +		reg_cr2 = FSL_SAI_RCR2;
> +
> +	val_cr2 = readl(sai->base + reg_cr2);
> +	switch (clk_id) {
> +	case FSL_SAI_CLK_BUS:
> +		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
> +		val_cr2 |= FSL_SAI_CR2_MSEL_BUS;
> +		break;
> +	case FSL_SAI_CLK_MAST1:
> +		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
> +		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1;
> +		break;
> +	case FSL_SAI_CLK_MAST2:
> +		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
> +		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2;
> +		break;
> +	case FSL_SAI_CLK_MAST3:
> +		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
> +		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +	writel(val_cr2, sai->base + reg_cr2);
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
> +		int clk_id, unsigned int freq, int dir)
> +{
> +	int ret;
> +
> +	if (dir == SND_SOC_CLOCK_IN)
> +		return 0;
> +
> +	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
> +					FSL_FMT_TRANSMITTER);
> +	if (ret) {
> +		dev_err(cpu_dai->dev,
> +				"Cannot set sai's transmitter sysclk: %d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
> +					FSL_FMT_RECEIVER);
> +	if (ret) {
> +		dev_err(cpu_dai->dev,
> +				"Cannot set sai's receiver sysclk: %d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
> +		int div_id, int div)
> +{
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +	u32 tcr2, rcr2;
> +
> +	if (div_id == FSL_SAI_TX_DIV) {
> +		tcr2 = readl(sai->base + FSL_SAI_TCR2);
> +		tcr2 &= ~FSL_SAI_CR2_DIV_MASK;
> +		tcr2 |= FSL_SAI_CR2_DIV(div);
> +		writel(tcr2, sai->base + FSL_SAI_TCR2);
> +
> +	} else if (div_id == FSL_SAI_RX_DIV) {
> +		rcr2 = readl(sai->base + FSL_SAI_RCR2);
> +		rcr2 &= ~FSL_SAI_CR2_DIV_MASK;
> +		rcr2 |= FSL_SAI_CR2_DIV(div);
> +		writel(rcr2, sai->base + FSL_SAI_RCR2);
> +
> +	} else
> +		return -EINVAL;

It would look better if using switch-case. And the last 'else'
could be 'default:'.

> +
> +	return 0;
> +}
> +
> +static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
> +				unsigned int fmt, int fsl_dir)
> +{
> +	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +
> +	if (fsl_dir == FSL_FMT_TRANSMITTER) {
> +		reg_cr2 = FSL_SAI_TCR2;
> +		reg_cr3 = FSL_SAI_TCR3;
> +		reg_cr4 = FSL_SAI_TCR4;
> +	} else {
> +		reg_cr2 = FSL_SAI_RCR2;
> +		reg_cr3 = FSL_SAI_RCR3;
> +		reg_cr4 = FSL_SAI_RCR4;
> +	}
> +
> +	val_cr2 = readl(sai->base + reg_cr2);
> +	val_cr3 = readl(sai->base + reg_cr3);
> +	val_cr4 = readl(sai->base + reg_cr4);
> +
> +	if (sai->fbt == FSL_SAI_FBT_MSB)
> +		val_cr4 |= FSL_SAI_CR4_MF;
> +	else if (sai->fbt == FSL_SAI_FBT_LSB)
> +		val_cr4 &= ~FSL_SAI_CR4_MF;
> +
> +	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> +	case SND_SOC_DAIFMT_I2S:
> +		val_cr4 |= FSL_SAI_CR4_FSE;
> +		val_cr4 |= FSL_SAI_CR4_FSP;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
> +	case SND_SOC_DAIFMT_IB_IF:
> +		val_cr4 |= FSL_SAI_CR4_FSP;
> +		val_cr2 &= ~FSL_SAI_CR2_BCP;
> +		break;
> +	case SND_SOC_DAIFMT_IB_NF:
> +		val_cr4 &= ~FSL_SAI_CR4_FSP;
> +		val_cr2 &= ~FSL_SAI_CR2_BCP;
> +		break;
> +	case SND_SOC_DAIFMT_NB_IF:
> +		val_cr4 |= FSL_SAI_CR4_FSP;
> +		val_cr2 |= FSL_SAI_CR2_BCP;
> +		break;
> +	case SND_SOC_DAIFMT_NB_NF:
> +		val_cr4 &= ~FSL_SAI_CR4_FSP;
> +		val_cr2 |= FSL_SAI_CR2_BCP;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> +	case SND_SOC_DAIFMT_CBS_CFS:
> +		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
> +		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
> +		break;
> +	case SND_SOC_DAIFMT_CBM_CFM:
> +		val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
> +		val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	val_cr3 |= FSL_SAI_CR3_TRCE;
> +
> +	if (fsl_dir == FSL_FMT_RECEIVER)
> +		val_cr2 |= FSL_SAI_CR2_SYNC;
> +
> +	writel(val_cr2, sai->base + reg_cr2);
> +	writel(val_cr3, sai->base + reg_cr3);
> +	writel(val_cr4, sai->base + reg_cr4);
> +
> +	return 0;

> +

Pls drop this extra line.

> +}
> +
> +static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
> +{
> +	int ret;
> +
> +	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
> +	if (ret) {
> +		dev_err(cpu_dai->dev,
> +				"Cannot set sai's transmitter format: %d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
> +	if (ret) {
> +		dev_err(cpu_dai->dev,
> +				"Cannot set sai's receiver format: %d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
> +		unsigned int tx_mask, unsigned int rx_mask,
> +		int slots, int slot_width)
> +{
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +	u32 tcr4, rcr4;
> +
> +	tcr4 = readl(sai->base + FSL_SAI_TCR4);
> +	tcr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
> +	tcr4 |= FSL_SAI_CR4_FRSZ(2);
> +	writel(tcr4, sai->base + FSL_SAI_TCR4);
> +	writel(tx_mask, sai->base + FSL_SAI_TMR);
> +
> +	rcr4 = readl(sai->base + FSL_SAI_RCR4);
> +	rcr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
> +	rcr4 |= FSL_SAI_CR4_FRSZ(2);
> +	writel(rcr4, sai->base + FSL_SAI_RCR4);
> +	writel(rx_mask, sai->base + FSL_SAI_RMR);
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
> +		struct snd_pcm_hw_params *params,
> +		struct snd_soc_dai *cpu_dai)
> +{
> +	u32 val_cr4, val_cr5, reg_cr4, reg_cr5, word_width;
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +
> +	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
> +		reg_cr4 = FSL_SAI_TCR4;
> +		reg_cr5 = FSL_SAI_TCR5;
> +	} else {
> +		reg_cr4 = FSL_SAI_RCR4;
> +		reg_cr5 = FSL_SAI_RCR5;
> +	}
> +
> +	val_cr4 = readl(sai->base + reg_cr4);
> +	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
> +
> +	val_cr5 = readl(sai->base + reg_cr5);
> +	val_cr5 &= ~FSL_SAI_CR5_WNW_MASK;
> +	val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
> +	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
> +
> +	switch (params_format(params)) {
> +	case SNDRV_PCM_FORMAT_S16_LE:
> +		word_width = 16;
> +		break;
> +	case SNDRV_PCM_FORMAT_S20_3LE:
> +		word_width = 20;
> +		break;
> +	case SNDRV_PCM_FORMAT_S24_LE:
> +		word_width = 24;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
> +	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
> +	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
> +
> +	if (sai->fbt == FSL_SAI_FBT_MSB)
> +		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
> +	else if (sai->fbt == FSL_SAI_FBT_LSB)
> +		val_cr5 |= FSL_SAI_CR5_FBT(0);
> +
> +	writel(val_cr4, sai->base + reg_cr4);
> +	writel(val_cr5, sai->base + reg_cr5);
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
> +		struct snd_soc_dai *dai)
> +{
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
> +	unsigned int tcsr, rcsr;
> +
> +	tcsr = readl(sai->base + FSL_SAI_TCSR);
> +	rcsr = readl(sai->base + FSL_SAI_RCSR);
> +
> +	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
> +		tcsr |= FSL_SAI_CSR_FRDE;
> +		rcsr &= ~FSL_SAI_CSR_FRDE;
> +	} else {
> +		rcsr |= FSL_SAI_CSR_FRDE;
> +		tcsr &= ~FSL_SAI_CSR_FRDE;
> +	}
> +
> +	switch (cmd) {
> +	case SNDRV_PCM_TRIGGER_START:
> +	case SNDRV_PCM_TRIGGER_RESUME:
> +	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
> +		tcsr |= FSL_SAI_CSR_TERE;
> +		rcsr |= FSL_SAI_CSR_TERE;
> +		writel(rcsr, sai->base + FSL_SAI_RCSR);
> +		udelay(10);

Does SAI really needs this udelay() here? Required by IP's operation flow?
If so, I think it's better to add comments here to explain it.

> +		writel(tcsr, sai->base + FSL_SAI_TCSR);
> +		break;
> +
> +	case SNDRV_PCM_TRIGGER_STOP:
> +	case SNDRV_PCM_TRIGGER_SUSPEND:
> +	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
> +		if (!(dai->playback_active || dai->capture_active)) {
> +			tcsr &= ~FSL_SAI_CSR_TERE;
> +			rcsr &= ~FSL_SAI_CSR_TERE;
> +		}
> +		writel(rcsr, sai->base + FSL_SAI_RCSR);
> +		udelay(10);
> +		writel(tcsr, sai->base + FSL_SAI_TCSR);
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	return 0;
> +}
> +
> +static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
> +	.set_sysclk	= fsl_sai_set_dai_sysclk,
> +	.set_clkdiv	= fsl_sai_set_dai_clkdiv,
> +	.set_fmt	= fsl_sai_set_dai_fmt,
> +	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
> +	.hw_params	= fsl_sai_hw_params,
> +	.trigger	= fsl_sai_trigger,
> +};
> +
> +static int fsl_sai_dai_probe(struct snd_soc_dai *dai)
> +{
> +	int ret;
> +	struct fsl_sai *sai = dev_get_drvdata(dai->dev);
> +
> +	ret = clk_prepare_enable(sai->clk);
> +	if (ret)
> +		return ret;
> +
> +	writel(0x0, sai->base + FSL_SAI_RCSR);
> +	writel(0x0, sai->base + FSL_SAI_TCSR);
> +	writel(sai->dma_params_tx.maxburst, sai->base + FSL_SAI_TCR1);
> +	writel(sai->dma_params_rx.maxburst, sai->base + FSL_SAI_RCR1);
> +
> +	dai->playback_dma_data = &sai->dma_params_tx;
> +	dai->capture_dma_data = &sai->dma_params_rx;
> +
> +	snd_soc_dai_set_drvdata(dai, sai);
> +
> +	return 0;
> +}
> +
> +int fsl_sai_dai_remove(struct snd_soc_dai *dai)

static

> +{
> +	struct fsl_sai *sai = dev_get_drvdata(dai->dev);
> +
> +	clk_disable_unprepare(sai->clk);
> +
> +	return 0;
> +}
> +
> +static struct snd_soc_dai_driver fsl_sai_dai = {
> +	.probe = fsl_sai_dai_probe,
> +	.remove = fsl_sai_dai_remove,
> +	.playback = {
> +		.channels_min = 1,
> +		.channels_max = 2,
> +		.rates = SNDRV_PCM_RATE_8000_96000,
> +		.formats = FSL_SAI_FORMATS,
> +	},
> +	.capture = {
> +		.channels_min = 1,
> +		.channels_max = 2,
> +		.rates = SNDRV_PCM_RATE_8000_96000,
> +		.formats = FSL_SAI_FORMATS,
> +	},
> +	.ops = &fsl_sai_pcm_dai_ops,
> +};
> +
> +static const struct snd_soc_component_driver fsl_component = {
> +	.name           = "fsl-sai",
> +};
> +
> +static int fsl_sai_probe(struct platform_device *pdev)
> +{
> +	struct resource *res;
> +	struct fsl_sai *sai;
> +	int ret = 0;
> +
> +	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
> +	if (!sai)
> +		return -ENOMEM;
> +
> +	sai->fbt = FSL_SAI_FBT_MSB;
> +
> +	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
> +	sai->base = devm_ioremap_resource(&pdev->dev, res);
> +	if (IS_ERR(sai->base))
> +		return PTR_ERR(sai->base);
> +
> +	sai->clk = devm_clk_get(&pdev->dev, "sai");
> +	if (IS_ERR(sai->clk)) {
> +		dev_err(&pdev->dev, "Cannot get sai's clock: %d\n", ret);
> +		return PTR_ERR(sai->clk);
> +	}
> +
> +	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
> +	sai->dma_params_rx.maxburst = 6;
> +
> +	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
> +	sai->dma_params_tx.maxburst = 6;
> +
> +	platform_set_drvdata(pdev, sai);
> +
> +	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
> +			&fsl_sai_dai, 1);
> +	if (ret)
> +		return ret;
> +
> +	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL,
> +			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
> +	if (ret)
> +		return ret;
> +
> +	return 0;
> +}
> +
> +static int fsl_sai_remove(struct platform_device *pdev)
> +{
> +	snd_dmaengine_pcm_unregister(&pdev->dev);
> +
> +	return 0;
> +}
> +
> +static const struct of_device_id fsl_sai_ids[] = {
> +	{ .compatible = "fsl,vf610-sai", },

> +	{ /*sentinel*/ }

I think this could be left in blank without comments inside.
And if you really want to add it, pls add like: /* sentinel */
						  ^        ^
> +};
> +
> +static struct platform_driver fsl_sai_driver = {
> +	.probe = fsl_sai_probe,
> +	.remove = fsl_sai_remove,
> +
> +	.driver = {
> +		.name = "fsl-sai",
> +		.owner = THIS_MODULE,
> +		.of_match_table = fsl_sai_ids,
> +	},
> +};
> +module_platform_driver(fsl_sai_driver);
> +
> +MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
> +MODULE_AUTHOR("Alison Wang, <b18965@freescale.com>");
> +MODULE_DESCRIPTION("Freescale Soc SAI Interface");
> +MODULE_LICENSE("GPL");

Should be better if added:
MODULE_ALIAS("platform:fsl-sai");

This would support module auto-load feature in some Linux-OS.

Best regards,
Nicolin Chen

> diff --git a/sound/soc/fsl/fsl-sai.h b/sound/soc/fsl/fsl-sai.h
> new file mode 100644
> index 0000000..1637679
> --- /dev/null
> +++ b/sound/soc/fsl/fsl-sai.h
> @@ -0,0 +1,120 @@
> +/*
> + * Copyright 2012-2013 Freescale Semiconductor, Inc.
> + *
> + * This program is free software; you can redistribute it and/or modify
> + * it under the terms of the GNU General Public License version 2 as
> + * published by the Free Software Foundation.
> + */
> +
> +#ifndef __FSL_SAI_H
> +#define __FSL_SAI_H
> +
> +#include <sound/dmaengine_pcm.h>
> +
> +#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
> +			 SNDRV_PCM_FMTBIT_S20_3LE |\
> +			 SNDRV_PCM_FMTBIT_S24_LE)
> +
> +/* SAI Transmit/Recieve Control Register */
> +#define FSL_SAI_TCSR		0x00
> +#define FSL_SAI_RCSR		0x80
> +#define FSL_SAI_CSR_TERE	BIT(31)
> +#define FSL_SAI_CSR_FWF		BIT(17)
> +#define FSL_SAI_CSR_FRIE	BIT(8)
> +#define FSL_SAI_CSR_FRDE	BIT(0)
> +
> +/* SAI Transmit Data/FIFO/MASK Register */
> +#define FSL_SAI_TDR		0x20
> +#define FSL_SAI_TFR		0x40
> +#define FSL_SAI_TMR		0x60
> +
> +/* SAI Recieve Data/FIFO/MASK Register */
> +#define FSL_SAI_RDR		0xa0
> +#define FSL_SAI_RFR		0xc0
> +#define FSL_SAI_RMR		0xe0
> +
> +/* SAI Transmit and Recieve Configuration 1 Register */
> +#define FSL_SAI_TCR1		0x04
> +#define FSL_SAI_RCR1		0x84
> +
> +/* SAI Transmit and Recieve Configuration 2 Register */
> +#define FSL_SAI_TCR2		0x08
> +#define FSL_SAI_RCR2		0x88
> +#define FSL_SAI_CR2_SYNC	BIT(30)
> +#define FSL_SAI_CR2_MSEL_MASK	(0xff << 26)
> +#define FSL_SAI_CR2_MSEL_BUS	0
> +#define FSL_SAI_CR2_MSEL_MCLK1	BIT(26)
> +#define FSL_SAI_CR2_MSEL_MCLK2	BIT(27)
> +#define FSL_SAI_CR2_MSEL_MCLK3	(BIT(26) | BIT(27))
> +#define FSL_SAI_CR2_BCP		BIT(25)
> +#define FSL_SAI_CR2_BCD_MSTR	BIT(24)
> +#define FSL_SAI_CR2_DIV(x)	(x)
> +#define FSL_SAI_CR2_DIV_MASK	0xff
> +
> +/* SAI Transmit and Recieve Configuration 3 Register */
> +#define FSL_SAI_TCR3		0x0c
> +#define FSL_SAI_RCR3		0x8c
> +#define FSL_SAI_CR3_TRCE	BIT(16)
> +#define FSL_SAI_CR3_WDFL(x)	(x)
> +#define FSL_SAI_CR3_WDFL_MASK	0x1f
> +
> +/* SAI Transmit and Recieve Configuration 4 Register */
> +#define FSL_SAI_TCR4		0x10
> +#define FSL_SAI_RCR4		0x90
> +#define FSL_SAI_CR4_FRSZ(x)	(((x) - 1) << 16)
> +#define FSL_SAI_CR4_FRSZ_MASK	(0x1f << 16)
> +#define FSL_SAI_CR4_SYWD(x)	(((x) - 1) << 8)
> +#define FSL_SAI_CR4_SYWD_MASK	(0x1f << 8)
> +#define FSL_SAI_CR4_MF		BIT(4)
> +#define FSL_SAI_CR4_FSE		BIT(3)
> +#define FSL_SAI_CR4_FSP		BIT(1)
> +#define FSL_SAI_CR4_FSD_MSTR	BIT(0)
> +
> +/* SAI Transmit and Recieve Configuration 5 Register */
> +#define FSL_SAI_TCR5		0x14
> +#define FSL_SAI_RCR5		0x94
> +#define FSL_SAI_CR5_WNW(x)	(((x) - 1) << 24)
> +#define FSL_SAI_CR5_WNW_MASK	(0x1f << 24)
> +#define FSL_SAI_CR5_W0W(x)	(((x) - 1) << 16)
> +#define FSL_SAI_CR5_W0W_MASK	(0x1f << 16)
> +#define FSL_SAI_CR5_FBT(x)	((x) << 8)
> +#define FSL_SAI_CR5_FBT_MASK	(0x1f << 8)
> +
> +/* SAI audio dividers */
> +#define FSL_SAI_TX_DIV		0
> +#define FSL_SAI_RX_DIV		1
> +
> +/* SAI type */
> +#define FSL_SAI_DMA		BIT(0)
> +#define FSL_SAI_USE_AC97	BIT(1)
> +#define FSL_SAI_NET		BIT(2)
> +#define FSL_SAI_TRA_SYN		BIT(3)
> +#define FSL_SAI_REC_SYN		BIT(4)
> +#define FSL_SAI_USE_I2S_SLAVE	BIT(5)
> +
> +#define FSL_FMT_TRANSMITTER	0
> +#define FSL_FMT_RECEIVER	1
> +
> +/* SAI clock sources */
> +#define FSL_SAI_CLK_BUS		0
> +#define FSL_SAI_CLK_MAST1	1
> +#define FSL_SAI_CLK_MAST2	2
> +#define FSL_SAI_CLK_MAST3	3
> +
> +enum fsl_sai_fbt {
> +	FSL_SAI_FBT_MSB,
> +	FSL_SAI_FBT_LSB,
> +};
> +
> +struct fsl_sai {
> +	struct clk *clk;
> +
> +	void __iomem *base;
> +
> +	enum fsl_sai_fbt fbt;
> +
> +	struct snd_dmaengine_dai_dma_data dma_params_rx;
> +	struct snd_dmaengine_dai_dma_data dma_params_tx;
> +};
> +
> +#endif /* __FSL_SAI_H */
> -- 
> 1.8.4
> 

Re: [PATCHv2 5/8] ASoC: SGTL5000: Enhance the SGTL5000 codec driver about regulator.

[Nicolin Chen]

On Fri, Nov 01, 2013 at 03:04:52PM +0800, Xiubo Li wrote:
> On VF610 series there are no regulators used, and now whether the
> CONFIG_REGULATOR mirco is enabled or not, for the VF610 audio

micro? or macro?

> patch series, the board cannot be probe successfully.
> And this patch will solve this issue.
> 

I finally got your idea about what this patch does. But it seems to
be more likely a work around. Maybe I here can't think it through
comprehensively, but I think you can try to add a fixed regulator to
sgtl5000 in the devicetree, rather than disabling common functions
even if not breaking others.

> Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
> ---
>  sound/soc/codecs/sgtl5000.c | 12 ++++++++++++
>  1 file changed, 12 insertions(+)
> 
> diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
> index 1f4093f..c2f6d86 100644
> --- a/sound/soc/codecs/sgtl5000.c
> +++ b/sound/soc/codecs/sgtl5000.c
> @@ -61,6 +61,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
>  	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
>  };
>  
> +#ifdef CONFIG_REGULATOR
>  /* regulator supplies for sgtl5000, VDDD is an optional external supply */
>  enum sgtl5000_regulator_supplies {
>  	VDDA,
> @@ -93,6 +94,9 @@ static struct regulator_init_data ldo_init_data = {
>  	.num_consumer_supplies = 1,
>  	.consumer_supplies = &ldo_consumer[0],
>  };
> +#else
> +#define SGTL5000_SUPPLY_NUM 0
> +#endif
>  
>  /*
>   * sgtl5000 internal ldo regulator,
> @@ -112,7 +116,9 @@ struct sgtl5000_priv {
>  	int master;	/* i2s master or not */
>  	int fmt;	/* i2s data format */
>  	struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM];
> +#ifdef CONFIG_REGULATOR
>  	struct ldo_regulator *ldo;
> +#endif
>  	struct regmap *regmap;
>  	struct clk *mclk;
>  };
> @@ -879,6 +885,7 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec)
>  	return 0;
>  }
>  #else
> +#ifndef CONFIG_SND_SOC_FSL_SGTL5000_VF610

Checking a platform or SoC specific define here is just a bit....

Could you pls find a better solution? Like I said at first, use fixed regulator
or figure out why your System would hang with current sgtl5000 driver. If it
really has a critical bug in its regulator-related code, I think you can then
make a greater contribution by fixing the bug rather than bypassing it.

Best regards,
Nicolin Chen

>  static int ldo_regulator_register(struct snd_soc_codec *codec,
>  				struct regulator_init_data *init_data,
>  				int voltage)
> @@ -886,6 +893,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
>  	dev_err(codec->dev, "this setup needs regulator support in the kernel\n");
>  	return -EINVAL;
>  }
> +#endif
>  
>  static int ldo_regulator_remove(struct snd_soc_codec *codec)
>  {
> @@ -1137,6 +1145,7 @@ static int sgtl5000_resume(struct snd_soc_codec *codec)
>  #define sgtl5000_resume  NULL
>  #endif	/* CONFIG_SUSPEND */
>  
> +#ifdef CONFIG_REGULATOR
>  /*
>   * sgtl5000 has 3 internal power supplies:
>   * 1. VAG, normally set to vdda/2
> @@ -1373,6 +1382,7 @@ err_regulator_free:
>  	return ret;
>  
>  }
> +#endif
>  
>  static int sgtl5000_probe(struct snd_soc_codec *codec)
>  {
> @@ -1387,6 +1397,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
>  		return ret;
>  	}
>  
> +#ifdef CONFIG_REGULATOR
>  	ret = sgtl5000_enable_regulators(codec);
>  	if (ret)
>  		return ret;
> @@ -1395,6 +1406,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
>  	ret = sgtl5000_set_power_regs(codec);
>  	if (ret)
>  		goto err;
> +#endif
>  
>  	/* enable small pop, introduce 400ms delay in turning off */
>  	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
> -- 
> 1.8.4
> 

Re: [PATCHv2 6/8] ASoC: fsl: add SGTL5000 based audio machine driver.

[Oskar Schirmer]

Not that it would improve functionality, but:

On Fri, Nov 01, 2013 at 15:04:53 +0800, Xiubo Li wrote:
[...]
> diff --git a/sound/soc/fsl/fsl-sgtl5000-vf610.c b/sound/soc/fsl/fsl-sgtl5000-vf610.c
> new file mode 100644
> index 0000000..f535b42
> --- /dev/null
> +++ b/sound/soc/fsl/fsl-sgtl5000-vf610.c
> @@ -0,0 +1,208 @@
> +/*
> + * Freeacale ALSA SoC Audio using SGT1500 as codec.
          ^                             ^   ^
     "Freescale"                    "SGTL5000"

regards,
  Oskar

Re: [PATCHv2 6/8] ASoC: fsl: add SGTL5000 based audio machine driver.

[Nicolin Chen]

Hi Xiubo,

On Fri, Nov 01, 2013 at 03:04:53PM +0800, Xiubo Li wrote:
> This is the SGTL5000 codec based audio driver supported with both
> playback and capture dai link implemention.
> 
> This implementation is only compatible with device tree definition.
> 
> Signed-off-by: Alison Wang <b18965@freescale.com
> Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
> 
> Conflicts:
> 	sound/soc/fsl/Makefile
> ---
>  sound/soc/fsl/Kconfig              |  10 ++
>  sound/soc/fsl/Makefile             |   2 +

>  sound/soc/fsl/fsl-sgtl5000-vf610.c | 208 +++++++++++++++++++++++++++++++++++++

I just doubt if this file naming is appropriate. Even if we might not have
rigor rule for the file names, according to existing ones, they are all in
a same pattern: [SoC name]-[codec name].c

"imx-sgtl5000.c" for example

I think it would make user less confused about what this file exactly is if
this machine driver also follow the pattern: vf610-sgtl5000.c


@Shawn

What do you think about the file name?

>  3 files changed, 220 insertions(+)
>  create mode 100644 sound/soc/fsl/fsl-sgtl5000-vf610.c
> 
> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
> index 9a8851e..1b835ba 100644
> --- a/sound/soc/fsl/Kconfig
> +++ b/sound/soc/fsl/Kconfig
> @@ -228,4 +228,14 @@ config SND_SOC_FSL_SAI
>  	tristate
>  	select SND_SOC_GENERIC_DMAENGINE_PCM
>  
> +config SND_SOC_FSL_SGTL5000_VF610

Same problem with the this define.

> +	tristate "SoC Audio support for FSL boards with sgtl5000"

And 'FSL' here confuses me a lot. Because those boards based on i.MX series
also could be called FSL boards.

> +	depends on OF && I2C
> +	select SND_SOC_FSL_SAI
> +	select SND_SOC_FSL_PCM
> +	select SND_SOC_SGTL5000
> +	help
> +	  Say Y if you want to add support for SoC audio on an FSL board with
> +	  a sgtl5000 codec.
> +
>  endif # SND_FSL_SOC
> diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
> index e5acc03..26fc551 100644
> --- a/sound/soc/fsl/Makefile
> +++ b/sound/soc/fsl/Makefile
> @@ -59,5 +59,7 @@ obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
>  
>  # FSL ARM SAI/SGT15000 Platform Support
>  snd-soc-fsl-sai-objs := fsl-sai.o
> +snd-soc-fsl-sgtl5000-vf610-objs := fsl-sgtl5000-vf610.o
>  
>  obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
> +obj-$(CONFIG_SND_SOC_FSL_SGTL5000_VF610) += snd-soc-fsl-sgtl5000-vf610.o
> diff --git a/sound/soc/fsl/fsl-sgtl5000-vf610.c b/sound/soc/fsl/fsl-sgtl5000-vf610.c
> new file mode 100644
> index 0000000..f535b42
> --- /dev/null
> +++ b/sound/soc/fsl/fsl-sgtl5000-vf610.c
> @@ -0,0 +1,208 @@
> +/*
> + * Freeacale ALSA SoC Audio using SGT1500 as codec.
> + *
> + * Copyright 2012-2013 Freescale Semiconductor, Inc.
> + *
> + * 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:
> + *
> + */
> +
> +#include <linux/module.h>
> +#include <linux/of.h>
> +#include <linux/of_platform.h>
> +#include <linux/i2c.h>
> +#include <linux/clk.h>
> +
> +#include "../codecs/sgtl5000.h"
> +#include "fsl-sai.h"
> +
> +static unsigned int sysclk_rate;
> +
> +static int fsl_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)

Naming issue here again.

At least from my point of view, if you actually merged imx-sgtl5000 with
vf610-sgtl5000 and made it also compatible to other freescale SoCs, you
could then fairly call it fsl_sgtl5000_xxxx.

Well, I might be a little picky here because it's a static function and
won't conflict others. Just the name here doesn't look so explicit to me.

Please reconsider about this whole file's naming.

Best regards,
Nicolin Chen

> +{
> +	int ret;
> +	struct device *dev = rtd->card->dev;
> +
> +	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
> +				     sysclk_rate, SND_SOC_CLOCK_IN);
> +	if (ret) {
> +		dev_err(dev, "could not set codec driver clock params :%d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_BUS,
> +				     sysclk_rate, SND_SOC_CLOCK_OUT);
> +	if (ret) {
> +		dev_err(dev, "could not set cpu dai driver clock params :%d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	return 0;
> +}
> +
> +static int sgtl5000_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;
> +	unsigned int channels = params_channels(params);
> +
> +	/* TODO: The SAI driver should figure this out for us */
> +	switch (channels) {
> +	case 2:
> +		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffc, 0xfffffffc, 2, 0);
> +		break;
> +	case 1:
> +		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffe, 0xfffffffe, 1, 0);
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	return 0;
> +}
> +
> +static struct snd_soc_ops fsl_sgtl5000_hifi_ops = {
> +	.hw_params = sgtl5000_params,
> +};
> +
> +static struct snd_soc_dai_link fsl_sgtl5000_dai = {
> +	.name = "HiFi",
> +	.stream_name = "HiFi",
> +	.codec_dai_name = "sgtl5000",
> +	.init = &fsl_sgtl5000_dai_init,
> +	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
> +		SND_SOC_DAIFMT_CBM_CFM,
> +	.ops = &fsl_sgtl5000_hifi_ops,
> +};
> +
> +static const struct snd_soc_dapm_widget fsl_sgtl5000_dapm_widgets[] = {
> +	SND_SOC_DAPM_MIC("Mic Jack", NULL),
> +	SND_SOC_DAPM_LINE("Line In Jack", NULL),
> +	SND_SOC_DAPM_HP("Headphone Jack", NULL),
> +	SND_SOC_DAPM_SPK("Line Out Jack", NULL),
> +	SND_SOC_DAPM_SPK("Ext Spk", NULL),
> +};
> +
> +static struct snd_soc_card fsl_sgt1500_card = {
> +	.owner = THIS_MODULE,
> +	.num_links = 1,
> +	.dai_link = &fsl_sgtl5000_dai,
> +	.dapm_widgets = fsl_sgtl5000_dapm_widgets,
> +	.num_dapm_widgets = ARRAY_SIZE(fsl_sgtl5000_dapm_widgets),
> +};
> +
> +static int fsl_sgtl5000_parse_dt(struct platform_device *pdev)
> +{
> +	int ret;
> +	struct device_node *sai_np, *codec_np;
> +	struct clk *codec_clk;
> +	struct i2c_client *codec_dev;
> +	struct device_node *np = pdev->dev.of_node;
> +
> +	ret = snd_soc_of_parse_card_name(&fsl_sgt1500_card, "model");
> +	if (ret)
> +		return ret;
> +
> +	ret = snd_soc_of_parse_audio_routing(&fsl_sgt1500_card,
> +			"audio-routing");
> +	if (ret)
> +		return ret;
> +
> +	sai_np = of_parse_phandle(np, "saif-controller", 0);
> +	if (!sai_np) {
> +		dev_err(&pdev->dev, "\"saif-controller\" phandle missing or "
> +				"invalid\n");
> +		return -EINVAL;
> +	}
> +	fsl_sgtl5000_dai.cpu_of_node = sai_np;
> +	fsl_sgtl5000_dai.platform_of_node = sai_np;
> +
> +	codec_np = of_parse_phandle(np, "audio-codec", 0);
> +	if (!codec_np) {
> +		dev_err(&pdev->dev, "\"audio-codec\" phandle missing or "
> +				"invalid\n");
> +		ret = -EINVAL;
> +		goto sai_np_fail;
> +	}
> +	fsl_sgtl5000_dai.codec_of_node = codec_np;
> +
> +	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 = PTR_ERR(codec_dev);
> +		goto codec_np_fail;
> +	}
> +
> +	codec_clk = devm_clk_get(&codec_dev->dev, NULL);
> +	if (IS_ERR(codec_clk)) {
> +		dev_err(&pdev->dev, "failed to get codec clock\n");
> +		ret = PTR_ERR(codec_clk);
> +		goto codec_np_fail;
> +	}
> +
> +	sysclk_rate = clk_get_rate(codec_clk);
> +
> +codec_np_fail:
> +	of_node_put(codec_np);
> +sai_np_fail:
> +	of_node_put(sai_np);
> +
> +	return ret;
> +}
> +
> +static int fsl_sgtl5000_probe(struct platform_device *pdev)
> +{
> +	int ret;
> +
> +	fsl_sgt1500_card.dev = &pdev->dev;
> +
> +	ret = fsl_sgtl5000_parse_dt(pdev);
> +	if (ret) {
> +		dev_err(&pdev->dev,
> +				"parse sgtl5000 device tree failed :%d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	ret = devm_snd_soc_register_card(&pdev->dev, &fsl_sgt1500_card);
> +	if (ret) {
> +		dev_err(&pdev->dev, "register soc sound card failed :%d\n",
> +				ret);
> +		return ret;
> +	}
> +
> +	return 0;
> +}
> +
> +static int fsl_sgtl5000_remove(struct platform_device *pdev)
> +{
> +	snd_soc_unregister_card(&fsl_sgt1500_card);
> +
> +	return 0;
> +}
> +
> +static const struct of_device_id fsl_sgtl5000_dt_ids[] = {
> +	{ .compatible = "fsl,vf610-sgtl5000", },
> +	{ /* sentinel */ }
> +};
> +MODULE_DEVICE_TABLE(of, fsl_sgtl5000_dt_ids);
> +
> +static struct platform_driver fsl_sgtl5000_driver = {
> +	.driver = {
> +		.name = "fsl-sgtl5000",
> +		.owner = THIS_MODULE,
> +		.of_match_table = fsl_sgtl5000_dt_ids,
> +	},
> +	.probe = fsl_sgtl5000_probe,
> +	.remove = fsl_sgtl5000_remove,
> +};
> +module_platform_driver(fsl_sgtl5000_driver);
> +
> +MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
> +MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
> +MODULE_LICENSE("GPL");
> -- 
> 1.8.4
> 

Re: [PATCHv2 6/8] ASoC: fsl: add SGTL5000 based audio machine driver.

[Shawn Guo]

On Fri, Nov 01, 2013 at 06:28:05PM +0800, Nicolin Chen wrote:
> >  sound/soc/fsl/fsl-sgtl5000-vf610.c | 208 +++++++++++++++++++++++++++++++++++++
> 
> I just doubt if this file naming is appropriate. Even if we might not have
> rigor rule for the file names, according to existing ones, they are all in
> a same pattern: [SoC name]-[codec name].c
> 
> "imx-sgtl5000.c" for example
> 
> I think it would make user less confused about what this file exactly is if
> this machine driver also follow the pattern: vf610-sgtl5000.c
> 
> 
> @Shawn
> 
> What do you think about the file name?

Yeah, it would be better to name the file following the existing the
pattern.

Shawn

Re: [PATCHv2 1/8] ALSA: Add SAI SoC Digital Audio Interface driver.

[Mark Brown]

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On Fri, Nov 01, 2013 at 03:04:48PM +0800, Xiubo Li wrote:

> +static int fsl_sai_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
> +		int div_id, int div)
> +{
> +	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
> +	u32 tcr2, rcr2;
> +
> +	if (div_id == FSL_SAI_TX_DIV) {
> +		tcr2 = readl(sai->base + FSL_SAI_TCR2);
> +		tcr2 &= ~FSL_SAI_CR2_DIV_MASK;
> +		tcr2 |= FSL_SAI_CR2_DIV(div);
> +		writel(tcr2, sai->base + FSL_SAI_TCR2);

What is this divider and why does the user have to set it manually?

> +	} else
> +		return -EINVAL;
> +

Coding style?

> +static int fsl_sai_dai_probe(struct snd_soc_dai *dai)
> +{
> +	int ret;
> +	struct fsl_sai *sai = dev_get_drvdata(dai->dev);
> +
> +	ret = clk_prepare_enable(sai->clk);
> +	if (ret)
> +		return ret;

It'd be nicer to only enable the clock while the device is in active
use.

> +	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL,
> +			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
> +	if (ret)
> +		return ret;

We should have a devm_ version of this.

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Re: [PATCHv2 6/8] ASoC: fsl: add SGTL5000 based audio machine driver.

[Mark Brown]

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On Fri, Nov 01, 2013 at 03:04:53PM +0800, Xiubo Li wrote:

> Conflicts:
> 	sound/soc/fsl/Makefile

Ahem.

> +	/* TODO: The SAI driver should figure this out for us */
> +	switch (channels) {
> +	case 2:
> +		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffc, 0xfffffffc, 2, 0);
> +		break;
> +	case 1:
> +		snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffe, 0xfffffffe, 1, 0);
> +		break;
> +	default:
> +		return -EINVAL;
> +	}

Yes, it should - this code should probably just be copied straight into
the SAI driver.  If we need to support other configurations we can do
that later.

> +static int fsl_sgtl5000_remove(struct platform_device *pdev)
> +{
> +	snd_soc_unregister_card(&fsl_sgt1500_card);
> +
> +	return 0;
> +}

You're using snd_soc_unregister_card() so you don't need to do this.

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Re: [PATCHv2 5/8] ASoC: SGTL5000: Enhance the SGTL5000 codec driver about regulator.

[Mark Brown]

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On Fri, Nov 01, 2013 at 03:04:52PM +0800, Xiubo Li wrote:
> On VF610 series there are no regulators used, and now whether the
> CONFIG_REGULATOR mirco is enabled or not, for the VF610 audio
> patch series, the board cannot be probe successfully.
> And this patch will solve this issue.

I don't understand what this is for at all, you're just saying there is
a problem you're trying to solve but you don't explain anything about
what the problem is or how your changes address it.

> +#ifndef CONFIG_SND_SOC_FSL_SGTL5000_VF610
>  static int ldo_regulator_register(struct snd_soc_codec *codec,

This is definitely broken, it won't work with multi-platform kernels,
and I don't understand what this is supposed to do - what is the reason
for making this change?

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