[PATCHv3 0/8] ALSA: Add SAI driver and enable SGT15000 codec

[PATCHv3 0/8] ALSA: Add SAI driver and enable SGT15000 codec

[Xiubo Li]

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 v3:
- Add big-endian mode support.
- Revise some commit comments, coding styles,etc.
- Keep header files ordered by alphabet,etc.
- Add MODULE_ALIAS("platform:fsl-sai") for SAI driver.
- Rename the files to be in a same pattern with the existed ones.
- Remove some not useful functions or code.
- Only enable the clock while the device is in active use.
- Revise one error message according to Bill.

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.

[PATCHv3 0/8] ALSA: Add SAI driver and enable SGT15000 codec

[Xiubo Li]

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 v3:
- Add big-endian mode support.
- Revise some commit comments, coding styles,etc.
- Keep header files ordered by alphabet,etc.
- Add MODULE_ALIAS("platform:fsl-sai") for SAI driver.
- Rename the files to be in a same pattern with the existed ones.
- Remove some not useful functions or code.
- Only enable the clock while the device is in active use.
- Revise one error message according to Bill.

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.

[PATCHv3 0/8] ALSA: Add SAI driver and enable SGT15000 codec

[Xiubo Li]

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 v3:
- Add big-endian mode support.
- Revise some commit comments, coding styles,etc.
- Keep header files ordered by alphabet,etc.
- Add MODULE_ALIAS("platform:fsl-sai") for SAI driver.
- Rename the files to be in a same pattern with the existed ones.
- Remove some not useful functions or code.
- Only enable the clock while the device is in active use.
- Revise one error message according to Bill.

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.

[PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Xiubo Li]

Fix the description of one comment.

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 master and slave mode.

Signed-off-by: Alison Wang <b18965@freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 sound/soc/fsl/Kconfig   |   4 +
 sound/soc/fsl/Makefile  |   4 +-
 sound/soc/fsl/fsl_sai.c | 492 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_sai.h | 114 +++++++++++
 4 files changed, 613 insertions(+), 1 deletion(-)
 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..ac4fe4e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,3 +1,7 @@
+config SND_SOC_FSL_SAI
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
 config SND_SOC_FSL_SSI
 	tristate
 
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8db705b..aaccbee 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
 snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
-# Freescale PowerPC SSI/DMA Platform Support
+# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
new file mode 100644
index 0000000..50a797e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.c
@@ -0,0 +1,492 @@
+/*
+ * Freescale ALSA SoC Digital Audio Interface (SAI) 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/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_sai.h"
+
+static inline u32 sai_readl(struct fsl_sai *sai,
+		const void __iomem *addr)
+{
+	u32 val;
+
+	val = __raw_readl(addr);
+
+	if (likely(sai->big_endian_regs))
+		val = be32_to_cpu(val);
+	else
+		val = le32_to_cpu(val);
+	rmb();
+
+	return val;
+}
+
+static inline void sai_writel(struct fsl_sai *sai,
+		u32 val, void __iomem *addr)
+{
+	wmb();
+	if (likely(sai->big_endian_regs))
+		val = cpu_to_be32(val);
+	else
+		val = cpu_to_le32(val);
+
+	__raw_writel(val, addr);
+}
+
+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 = sai_readl(sai, 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;
+	}
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
+	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
+	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
+
+	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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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 = sai_readl(sai, sai->base + reg_cr2);
+	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+
+	if (sai->big_endian_data)
+		val_cr4 |= FSL_SAI_CR4_MF;
+	else
+		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;
+
+	sai_writel(sai, val_cr2, sai->base + reg_cr2);
+	sai_writel(sai, val_cr3, sai->base + reg_cr3);
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return 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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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, val_mr, reg_cr4, reg_cr5, reg_mr, word_width;
+	unsigned int channels = params_channels(params);
+	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;
+		reg_mr = FSL_SAI_TMR;
+	} else {
+		reg_cr4 = FSL_SAI_RCR4;
+		reg_cr5 = FSL_SAI_RCR5;
+		reg_mr = FSL_SAI_RMR;
+	}
+
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
+	val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+
+	val_cr5 = sai_readl(sai, 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->big_endian_data)
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+	else
+		val_cr5 |= FSL_SAI_CR5_FBT(0);
+
+	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+	if (channels == 2 || channels == 1)
+		val_mr = ~0UL - ((1 << channels) - 1);
+	else
+		return -EINVAL;
+
+	sai_writel(sai, val_cr4, sai->base + reg_cr4);
+	sai_writel(sai, val_cr5, sai->base + reg_cr5);
+	sai_writel(sai, val_mr, sai->base + reg_mr);
+
+	return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int tcsr, rcsr;
+
+	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
+	rcsr = sai_readl(sai, 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;
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!(cpu_dai->playback_active || cpu_dai->capture_active)) {
+			tcsr &= ~FSL_SAI_CSR_TERE;
+			rcsr &= ~FSL_SAI_CSR_TERE;
+		}
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+
+	return ret;
+}
+
+static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	clk_disable_unprepare(sai->clk);
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_sysclk	= fsl_sai_set_dai_sysclk,
+	.set_fmt	= fsl_sai_set_dai_fmt,
+	.hw_params	= fsl_sai_hw_params,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+	.shutdown	= fsl_sai_shutdown,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+
+	cpu_dai->playback_dma_data = &sai->dma_params_tx;
+	cpu_dai->capture_dma_data = &sai->dma_params_rx;
+
+	snd_soc_dai_set_drvdata(cpu_dai, sai);
+
+	return 0;
+}
+
+static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
+{
+	cpu_dai->playback_dma_data = NULL;
+	cpu_dai->capture_dma_data = NULL;
+
+	snd_soc_dai_set_drvdata(cpu_dai, NULL);
+
+	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)
+{
+	int ret;
+	struct fsl_sai *sai;
+	struct resource *res;
+	struct device_node *np = pdev->dev.of_node;
+
+	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+	if (!sai)
+		return -ENOMEM;
+
+	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\n");
+		return PTR_ERR(sai->clk);
+	}
+
+	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;
+	sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
+
+	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
+	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_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_ALIAS("platform:fsl-sai");
+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..41bb62e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.h
@@ -0,0 +1,114 @@
+/*
+ * 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)
+
+/* 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 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
+
+/* SAI data transfer numbers per DMA request */
+#define FSL_SAI_MAXBURST_TX 6
+#define FSL_SAI_MAXBURST_RX 6
+
+struct fsl_sai {
+	struct clk *clk;
+
+	void __iomem *base;
+
+	bool big_endian_regs;
+	bool big_endian_data;
+
+	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

[PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Xiubo Li]

Fix the description of one comment.

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 master and slave mode.

Signed-off-by: Alison Wang <b18965@freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 sound/soc/fsl/Kconfig   |   4 +
 sound/soc/fsl/Makefile  |   4 +-
 sound/soc/fsl/fsl_sai.c | 492 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_sai.h | 114 +++++++++++
 4 files changed, 613 insertions(+), 1 deletion(-)
 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..ac4fe4e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,3 +1,7 @@
+config SND_SOC_FSL_SAI
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
 config SND_SOC_FSL_SSI
 	tristate
 
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8db705b..aaccbee 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
 snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
-# Freescale PowerPC SSI/DMA Platform Support
+# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
new file mode 100644
index 0000000..50a797e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.c
@@ -0,0 +1,492 @@
+/*
+ * Freescale ALSA SoC Digital Audio Interface (SAI) 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/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_sai.h"
+
+static inline u32 sai_readl(struct fsl_sai *sai,
+		const void __iomem *addr)
+{
+	u32 val;
+
+	val = __raw_readl(addr);
+
+	if (likely(sai->big_endian_regs))
+		val = be32_to_cpu(val);
+	else
+		val = le32_to_cpu(val);
+	rmb();
+
+	return val;
+}
+
+static inline void sai_writel(struct fsl_sai *sai,
+		u32 val, void __iomem *addr)
+{
+	wmb();
+	if (likely(sai->big_endian_regs))
+		val = cpu_to_be32(val);
+	else
+		val = cpu_to_le32(val);
+
+	__raw_writel(val, addr);
+}
+
+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 = sai_readl(sai, 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;
+	}
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
+	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
+	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
+
+	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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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 = sai_readl(sai, sai->base + reg_cr2);
+	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+
+	if (sai->big_endian_data)
+		val_cr4 |= FSL_SAI_CR4_MF;
+	else
+		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;
+
+	sai_writel(sai, val_cr2, sai->base + reg_cr2);
+	sai_writel(sai, val_cr3, sai->base + reg_cr3);
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return 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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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, val_mr, reg_cr4, reg_cr5, reg_mr, word_width;
+	unsigned int channels = params_channels(params);
+	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;
+		reg_mr = FSL_SAI_TMR;
+	} else {
+		reg_cr4 = FSL_SAI_RCR4;
+		reg_cr5 = FSL_SAI_RCR5;
+		reg_mr = FSL_SAI_RMR;
+	}
+
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
+	val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+
+	val_cr5 = sai_readl(sai, 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->big_endian_data)
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+	else
+		val_cr5 |= FSL_SAI_CR5_FBT(0);
+
+	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+	if (channels == 2 || channels == 1)
+		val_mr = ~0UL - ((1 << channels) - 1);
+	else
+		return -EINVAL;
+
+	sai_writel(sai, val_cr4, sai->base + reg_cr4);
+	sai_writel(sai, val_cr5, sai->base + reg_cr5);
+	sai_writel(sai, val_mr, sai->base + reg_mr);
+
+	return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int tcsr, rcsr;
+
+	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
+	rcsr = sai_readl(sai, 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;
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!(cpu_dai->playback_active || cpu_dai->capture_active)) {
+			tcsr &= ~FSL_SAI_CSR_TERE;
+			rcsr &= ~FSL_SAI_CSR_TERE;
+		}
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+
+	return ret;
+}
+
+static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	clk_disable_unprepare(sai->clk);
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_sysclk	= fsl_sai_set_dai_sysclk,
+	.set_fmt	= fsl_sai_set_dai_fmt,
+	.hw_params	= fsl_sai_hw_params,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+	.shutdown	= fsl_sai_shutdown,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+
+	cpu_dai->playback_dma_data = &sai->dma_params_tx;
+	cpu_dai->capture_dma_data = &sai->dma_params_rx;
+
+	snd_soc_dai_set_drvdata(cpu_dai, sai);
+
+	return 0;
+}
+
+static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
+{
+	cpu_dai->playback_dma_data = NULL;
+	cpu_dai->capture_dma_data = NULL;
+
+	snd_soc_dai_set_drvdata(cpu_dai, NULL);
+
+	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)
+{
+	int ret;
+	struct fsl_sai *sai;
+	struct resource *res;
+	struct device_node *np = pdev->dev.of_node;
+
+	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+	if (!sai)
+		return -ENOMEM;
+
+	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\n");
+		return PTR_ERR(sai->clk);
+	}
+
+	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;
+	sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
+
+	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
+	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_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_ALIAS("platform:fsl-sai");
+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..41bb62e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.h
@@ -0,0 +1,114 @@
+/*
+ * 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)
+
+/* 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 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
+
+/* SAI data transfer numbers per DMA request */
+#define FSL_SAI_MAXBURST_TX 6
+#define FSL_SAI_MAXBURST_RX 6
+
+struct fsl_sai {
+	struct clk *clk;
+
+	void __iomem *base;
+
+	bool big_endian_regs;
+	bool big_endian_data;
+
+	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

[PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Xiubo Li]

Fix the description of one comment.

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 master and slave mode.

Signed-off-by: Alison Wang <b18965 at freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 sound/soc/fsl/Kconfig   |   4 +
 sound/soc/fsl/Makefile  |   4 +-
 sound/soc/fsl/fsl_sai.c | 492 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_sai.h | 114 +++++++++++
 4 files changed, 613 insertions(+), 1 deletion(-)
 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..ac4fe4e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,3 +1,7 @@
+config SND_SOC_FSL_SAI
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
 config SND_SOC_FSL_SSI
 	tristate
 
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8db705b..aaccbee 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
 snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
-# Freescale PowerPC SSI/DMA Platform Support
+# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
new file mode 100644
index 0000000..50a797e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.c
@@ -0,0 +1,492 @@
+/*
+ * Freescale ALSA SoC Digital Audio Interface (SAI) 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/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_sai.h"
+
+static inline u32 sai_readl(struct fsl_sai *sai,
+		const void __iomem *addr)
+{
+	u32 val;
+
+	val = __raw_readl(addr);
+
+	if (likely(sai->big_endian_regs))
+		val = be32_to_cpu(val);
+	else
+		val = le32_to_cpu(val);
+	rmb();
+
+	return val;
+}
+
+static inline void sai_writel(struct fsl_sai *sai,
+		u32 val, void __iomem *addr)
+{
+	wmb();
+	if (likely(sai->big_endian_regs))
+		val = cpu_to_be32(val);
+	else
+		val = cpu_to_le32(val);
+
+	__raw_writel(val, addr);
+}
+
+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 = sai_readl(sai, 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;
+	}
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
+	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
+	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
+
+	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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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 = sai_readl(sai, sai->base + reg_cr2);
+	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+
+	if (sai->big_endian_data)
+		val_cr4 |= FSL_SAI_CR4_MF;
+	else
+		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;
+
+	sai_writel(sai, val_cr2, sai->base + reg_cr2);
+	sai_writel(sai, val_cr3, sai->base + reg_cr3);
+	sai_writel(sai, 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;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return 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;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	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, val_mr, reg_cr4, reg_cr5, reg_mr, word_width;
+	unsigned int channels = params_channels(params);
+	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;
+		reg_mr = FSL_SAI_TMR;
+	} else {
+		reg_cr4 = FSL_SAI_RCR4;
+		reg_cr5 = FSL_SAI_RCR5;
+		reg_mr = FSL_SAI_RMR;
+	}
+
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
+	val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+
+	val_cr5 = sai_readl(sai, 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->big_endian_data)
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+	else
+		val_cr5 |= FSL_SAI_CR5_FBT(0);
+
+	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+	if (channels == 2 || channels == 1)
+		val_mr = ~0UL - ((1 << channels) - 1);
+	else
+		return -EINVAL;
+
+	sai_writel(sai, val_cr4, sai->base + reg_cr4);
+	sai_writel(sai, val_cr5, sai->base + reg_cr5);
+	sai_writel(sai, val_mr, sai->base + reg_mr);
+
+	return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int tcsr, rcsr;
+
+	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
+	rcsr = sai_readl(sai, 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;
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!(cpu_dai->playback_active || cpu_dai->capture_active)) {
+			tcsr &= ~FSL_SAI_CSR_TERE;
+			rcsr &= ~FSL_SAI_CSR_TERE;
+		}
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+
+	return ret;
+}
+
+static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	clk_disable_unprepare(sai->clk);
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_sysclk	= fsl_sai_set_dai_sysclk,
+	.set_fmt	= fsl_sai_set_dai_fmt,
+	.hw_params	= fsl_sai_hw_params,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+	.shutdown	= fsl_sai_shutdown,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+
+	cpu_dai->playback_dma_data = &sai->dma_params_tx;
+	cpu_dai->capture_dma_data = &sai->dma_params_rx;
+
+	snd_soc_dai_set_drvdata(cpu_dai, sai);
+
+	return 0;
+}
+
+static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
+{
+	cpu_dai->playback_dma_data = NULL;
+	cpu_dai->capture_dma_data = NULL;
+
+	snd_soc_dai_set_drvdata(cpu_dai, NULL);
+
+	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)
+{
+	int ret;
+	struct fsl_sai *sai;
+	struct resource *res;
+	struct device_node *np = pdev->dev.of_node;
+
+	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+	if (!sai)
+		return -ENOMEM;
+
+	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\n");
+		return PTR_ERR(sai->clk);
+	}
+
+	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;
+	sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
+
+	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
+	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_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_ALIAS("platform:fsl-sai");
+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..41bb62e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.h
@@ -0,0 +1,114 @@
+/*
+ * 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)
+
+/* 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 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
+
+/* SAI data transfer numbers per DMA request */
+#define FSL_SAI_MAXBURST_TX 6
+#define FSL_SAI_MAXBURST_RX 6
+
+struct fsl_sai {
+	struct clk *clk;
+
+	void __iomem *base;
+
+	bool big_endian_regs;
+	bool big_endian_data;
+
+	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

[PATCHv3 2/7] ARM: dts: Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610.

[Xiubo Li]

This patch adds 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 | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi
index e84d84c..2d6ddd0 100644
--- a/arch/arm/boot/dts/vf610.dtsi
+++ b/arch/arm/boot/dts/vf610.dtsi
@@ -166,6 +166,9 @@
 				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

[PATCHv3 2/7] ARM: dts: Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610.

[Xiubo Li]

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

Signed-off-by: Jingchang Lu <b35083-KZfg59tc24xl57MIdRCFDg@public.gmane.org>
Signed-off-by: Xiubo Li <Li.Xiubo-KZfg59tc24xl57MIdRCFDg@public.gmane.org>
---
 arch/arm/boot/dts/vf610.dtsi | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi
index e84d84c..2d6ddd0 100644
--- a/arch/arm/boot/dts/vf610.dtsi
+++ b/arch/arm/boot/dts/vf610.dtsi
@@ -166,6 +166,9 @@
 				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


--
To unsubscribe from this list: send the line "unsubscribe devicetree" in
the body of a message to majordomo-u79uwXL29TY76Z2rM5mHXA@public.gmane.org
More majordomo info at  http://vger.kernel.org/majordomo-info.html

[PATCHv3 2/7] ARM: dts: Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610.

[Xiubo Li]

This patch adds 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 | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi
index e84d84c..2d6ddd0 100644
--- a/arch/arm/boot/dts/vf610.dtsi
+++ b/arch/arm/boot/dts/vf610.dtsi
@@ -166,6 +166,9 @@
 				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

[PATCHv3 3/7] ARM: dts: Enable SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.

[Xiubo Li]

This patch adds and enables the 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 c8047ca..e60c20c 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -74,6 +74,12 @@
 	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

[PATCHv3 3/7] ARM: dts: Enable SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.

[Xiubo Li]

This patch adds and enables the SAI device.

Signed-off-by: Xiubo Li <Li.Xiubo-KZfg59tc24xl57MIdRCFDg@public.gmane.org>
---
 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 c8047ca..e60c20c 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -74,6 +74,12 @@
 	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


--
To unsubscribe from this list: send the line "unsubscribe devicetree" in
the body of a message to majordomo-u79uwXL29TY76Z2rM5mHXA@public.gmane.org
More majordomo info at  http://vger.kernel.org/majordomo-info.html

[PATCHv3 3/7] ARM: dts: Enable SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.

[Xiubo Li]

This patch adds and enables the 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 c8047ca..e60c20c 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -74,6 +74,12 @@
 	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

[PATCHv3 4/7] 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          | 40 ++++++++++++++++++++++
 1 file changed, 40 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..ca49ad1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,40 @@
+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.
+- big-endian-regs: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  device registers.
+- big-endian-data: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  fifo data.
+
+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>;
+	      big-endian_regs;
+	      big-endian_data;
+};
-- 
1.8.4

[PATCHv3 4/7] 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          | 40 ++++++++++++++++++++++
 1 file changed, 40 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..ca49ad1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,40 @@
+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.
+- big-endian-regs: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  device registers.
+- big-endian-data: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  fifo data.
+
+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>;
+	      big-endian_regs;
+	      big-endian_data;
+};
-- 
1.8.4

[PATCHv3 4/7] 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          | 40 ++++++++++++++++++++++
 1 file changed, 40 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..ca49ad1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,40 @@
+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.
+- big-endian-regs: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  device registers.
+- big-endian-data: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  fifo data.
+
+Example:
+sai2: sai at 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>;
+	      big-endian_regs;
+	      big-endian_data;
+};
-- 
1.8.4

[PATCHv3 5/7] 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: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Alison Wang <b18965@freescale.com
---
 sound/soc/fsl/Kconfig          |  22 +++++
 sound/soc/fsl/Makefile         |   4 +
 sound/soc/fsl/vf610-sgtl5000.c | 178 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 204 insertions(+)
 create mode 100644 sound/soc/fsl/vf610-sgtl5000.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index ac4fe4e..bb94c3c 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -217,3 +217,25 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
 endif # SND_IMX_SOC
+
+menuconfig SND_FSL_SOC
+	tristate "SoC Audio for Freescale VF610 CPUs"
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the VF610 CPUs.
+
+	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
+	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.
+
+if SND_FSL_SOC
+
+config SND_SOC_VF610_SGTL5000
+	tristate "SoC Audio support for VF610 boards with SGTL5000"
+	depends on OF && I2C
+	select SND_SOC_FSL_SAI
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on an VF610 board with
+	  a SGTL5000 codec and a SAI.
+
+endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index aaccbee..96110d1 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -22,6 +22,10 @@ obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
 obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
+# VF610 Platform Support
+snd-soc-vf610-sgtl5000-objs := vf610-sgtl5000.o
+obj-$(CONFIG_SND_SOC_VF610_SGTL5000) += snd-soc-vf610-sgtl5000.o
+
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/vf610-sgtl5000.c b/sound/soc/fsl/vf610-sgtl5000.c
new file mode 100644
index 0000000..75337bf
--- /dev/null
+++ b/sound/soc/fsl/vf610-sgtl5000.c
@@ -0,0 +1,178 @@
+/*
+ * Freescale ALSA SoC Audio using SGTL5000 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/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl_sai.h"
+
+static unsigned int sysclk_rate;
+
+static int vf610_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 struct snd_soc_dai_link vf610_sgtl5000_dai = {
+	.name = "HiFi",
+	.stream_name = "HiFi",
+	.codec_dai_name = "sgtl5000",
+	.init = &vf610_sgtl5000_dai_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static const struct snd_soc_dapm_widget vf610_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 vf610_sgtl5000_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+	.dai_link = &vf610_sgtl5000_dai,
+	.dapm_widgets = vf610_sgtl5000_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(vf610_sgtl5000_dapm_widgets),
+};
+
+static int vf610_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(&vf610_sgtl5000_card, "model");
+	if (ret)
+		return ret;
+
+	ret = snd_soc_of_parse_audio_routing(&vf610_sgtl5000_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;
+	}
+	vf610_sgtl5000_dai.cpu_of_node = sai_np;
+	vf610_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;
+	}
+	vf610_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 vf610_sgtl5000_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	vf610_sgtl5000_card.dev = &pdev->dev;
+
+	ret = vf610_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, &vf610_sgtl5000_card);
+	if (ret) {
+		dev_err(&pdev->dev, "TWR-AUDIO-SGTL board required :%d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int vf610_sgtl5000_remove(struct platform_device *pdev)
+{
+	return 0;
+}
+
+static const struct of_device_id vf610_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,vf610-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, vf610_sgtl5000_dt_ids);
+
+static struct platform_driver vf610_sgtl5000_driver = {
+	.driver = {
+		.name = "vf610-sgtl5000",
+		.owner = THIS_MODULE,
+		.of_match_table = vf610_sgtl5000_dt_ids,
+	},
+	.probe = vf610_sgtl5000_probe,
+	.remove = vf610_sgtl5000_remove,
+};
+module_platform_driver(vf610_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");
-- 
1.8.4

[PATCHv3 5/7] 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: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Alison Wang <b18965@freescale.com
---
 sound/soc/fsl/Kconfig          |  22 +++++
 sound/soc/fsl/Makefile         |   4 +
 sound/soc/fsl/vf610-sgtl5000.c | 178 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 204 insertions(+)
 create mode 100644 sound/soc/fsl/vf610-sgtl5000.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index ac4fe4e..bb94c3c 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -217,3 +217,25 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
 endif # SND_IMX_SOC
+
+menuconfig SND_FSL_SOC
+	tristate "SoC Audio for Freescale VF610 CPUs"
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the VF610 CPUs.
+
+	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
+	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.
+
+if SND_FSL_SOC
+
+config SND_SOC_VF610_SGTL5000
+	tristate "SoC Audio support for VF610 boards with SGTL5000"
+	depends on OF && I2C
+	select SND_SOC_FSL_SAI
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on an VF610 board with
+	  a SGTL5000 codec and a SAI.
+
+endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index aaccbee..96110d1 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -22,6 +22,10 @@ obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
 obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
+# VF610 Platform Support
+snd-soc-vf610-sgtl5000-objs := vf610-sgtl5000.o
+obj-$(CONFIG_SND_SOC_VF610_SGTL5000) += snd-soc-vf610-sgtl5000.o
+
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/vf610-sgtl5000.c b/sound/soc/fsl/vf610-sgtl5000.c
new file mode 100644
index 0000000..75337bf
--- /dev/null
+++ b/sound/soc/fsl/vf610-sgtl5000.c
@@ -0,0 +1,178 @@
+/*
+ * Freescale ALSA SoC Audio using SGTL5000 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/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl_sai.h"
+
+static unsigned int sysclk_rate;
+
+static int vf610_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 struct snd_soc_dai_link vf610_sgtl5000_dai = {
+	.name = "HiFi",
+	.stream_name = "HiFi",
+	.codec_dai_name = "sgtl5000",
+	.init = &vf610_sgtl5000_dai_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static const struct snd_soc_dapm_widget vf610_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 vf610_sgtl5000_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+	.dai_link = &vf610_sgtl5000_dai,
+	.dapm_widgets = vf610_sgtl5000_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(vf610_sgtl5000_dapm_widgets),
+};
+
+static int vf610_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(&vf610_sgtl5000_card, "model");
+	if (ret)
+		return ret;
+
+	ret = snd_soc_of_parse_audio_routing(&vf610_sgtl5000_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;
+	}
+	vf610_sgtl5000_dai.cpu_of_node = sai_np;
+	vf610_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;
+	}
+	vf610_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 vf610_sgtl5000_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	vf610_sgtl5000_card.dev = &pdev->dev;
+
+	ret = vf610_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, &vf610_sgtl5000_card);
+	if (ret) {
+		dev_err(&pdev->dev, "TWR-AUDIO-SGTL board required :%d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int vf610_sgtl5000_remove(struct platform_device *pdev)
+{
+	return 0;
+}
+
+static const struct of_device_id vf610_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,vf610-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, vf610_sgtl5000_dt_ids);
+
+static struct platform_driver vf610_sgtl5000_driver = {
+	.driver = {
+		.name = "vf610-sgtl5000",
+		.owner = THIS_MODULE,
+		.of_match_table = vf610_sgtl5000_dt_ids,
+	},
+	.probe = vf610_sgtl5000_probe,
+	.remove = vf610_sgtl5000_remove,
+};
+module_platform_driver(vf610_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");
-- 
1.8.4

[PATCHv3 5/7] 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: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Alison Wang <b18965 at freescale.com
---
 sound/soc/fsl/Kconfig          |  22 +++++
 sound/soc/fsl/Makefile         |   4 +
 sound/soc/fsl/vf610-sgtl5000.c | 178 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 204 insertions(+)
 create mode 100644 sound/soc/fsl/vf610-sgtl5000.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index ac4fe4e..bb94c3c 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -217,3 +217,25 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
 endif # SND_IMX_SOC
+
+menuconfig SND_FSL_SOC
+	tristate "SoC Audio for Freescale VF610 CPUs"
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the VF610 CPUs.
+
+	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
+	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.
+
+if SND_FSL_SOC
+
+config SND_SOC_VF610_SGTL5000
+	tristate "SoC Audio support for VF610 boards with SGTL5000"
+	depends on OF && I2C
+	select SND_SOC_FSL_SAI
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on an VF610 board with
+	  a SGTL5000 codec and a SAI.
+
+endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index aaccbee..96110d1 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -22,6 +22,10 @@ obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
 obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
+# VF610 Platform Support
+snd-soc-vf610-sgtl5000-objs := vf610-sgtl5000.o
+obj-$(CONFIG_SND_SOC_VF610_SGTL5000) += snd-soc-vf610-sgtl5000.o
+
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/vf610-sgtl5000.c b/sound/soc/fsl/vf610-sgtl5000.c
new file mode 100644
index 0000000..75337bf
--- /dev/null
+++ b/sound/soc/fsl/vf610-sgtl5000.c
@@ -0,0 +1,178 @@
+/*
+ * Freescale ALSA SoC Audio using SGTL5000 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@the following locations:
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl_sai.h"
+
+static unsigned int sysclk_rate;
+
+static int vf610_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 struct snd_soc_dai_link vf610_sgtl5000_dai = {
+	.name = "HiFi",
+	.stream_name = "HiFi",
+	.codec_dai_name = "sgtl5000",
+	.init = &vf610_sgtl5000_dai_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static const struct snd_soc_dapm_widget vf610_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 vf610_sgtl5000_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+	.dai_link = &vf610_sgtl5000_dai,
+	.dapm_widgets = vf610_sgtl5000_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(vf610_sgtl5000_dapm_widgets),
+};
+
+static int vf610_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(&vf610_sgtl5000_card, "model");
+	if (ret)
+		return ret;
+
+	ret = snd_soc_of_parse_audio_routing(&vf610_sgtl5000_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;
+	}
+	vf610_sgtl5000_dai.cpu_of_node = sai_np;
+	vf610_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;
+	}
+	vf610_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 vf610_sgtl5000_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	vf610_sgtl5000_card.dev = &pdev->dev;
+
+	ret = vf610_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, &vf610_sgtl5000_card);
+	if (ret) {
+		dev_err(&pdev->dev, "TWR-AUDIO-SGTL board required :%d\n",
+				ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int vf610_sgtl5000_remove(struct platform_device *pdev)
+{
+	return 0;
+}
+
+static const struct of_device_id vf610_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,vf610-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, vf610_sgtl5000_dt_ids);
+
+static struct platform_driver vf610_sgtl5000_driver = {
+	.driver = {
+		.name = "vf610-sgtl5000",
+		.owner = THIS_MODULE,
+		.of_match_table = vf610_sgtl5000_dt_ids,
+	},
+	.probe = vf610_sgtl5000_probe,
+	.remove = vf610_sgtl5000_remove,
+};
+module_platform_driver(vf610_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");
-- 
1.8.4

[PATCHv3 6/7] ARM: dts: Enable SGTL5000 codec based audio driver node for VF610 TOWER.

[Xiubo Li]

This patch adds and enables SGTL5000 codec support, and also specified
the corresponding SAI node.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 arch/arm/boot/dts/vf610-twr.dts | 33 +++++++++++++++++++++++++++++++++
 1 file changed, 33 insertions(+)

diff --git a/arch/arm/boot/dts/vf610-twr.dts b/arch/arm/boot/dts/vf610-twr.dts
index e60c20c..6a8ee0a 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -34,6 +34,31 @@
 		};
 	};
 
+	regulators {
+		compatible = "simple-bus";
+
+		reg_3p3v: 3p3v {
+			compatible = "regulator-fixed";
+			regulator-name = "3P3V";
+			regulator-min-microvolt = <3300000>;
+			regulator-max-microvolt = <3300000>;
+			regulator-always-on;
+		};
+	};
+
+	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";
+	};
+
 };
 
 &dspi0 {
@@ -72,6 +97,14 @@
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_i2c0_1>;
 	status = "okay";
+
+	codec: sgtl5000@0a {
+		compatible = "fsl,sgtl5000";
+		reg = <0x0a>;
+		VDDA-supply = <&reg_3p3v>;
+		VDDIO-supply = <&reg_3p3v>;
+		clocks = <&clks VF610_CLK_SAI2>;
+	};
 };
 
 &sai2 {
-- 
1.8.4

[PATCHv3 6/7] ARM: dts: Enable SGTL5000 codec based audio driver node for VF610 TOWER.

[Xiubo Li]

This patch adds and enables SGTL5000 codec support, and also specified
the corresponding SAI node.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 arch/arm/boot/dts/vf610-twr.dts | 33 +++++++++++++++++++++++++++++++++
 1 file changed, 33 insertions(+)

diff --git a/arch/arm/boot/dts/vf610-twr.dts b/arch/arm/boot/dts/vf610-twr.dts
index e60c20c..6a8ee0a 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -34,6 +34,31 @@
 		};
 	};
 
+	regulators {
+		compatible = "simple-bus";
+
+		reg_3p3v: 3p3v {
+			compatible = "regulator-fixed";
+			regulator-name = "3P3V";
+			regulator-min-microvolt = <3300000>;
+			regulator-max-microvolt = <3300000>;
+			regulator-always-on;
+		};
+	};
+
+	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";
+	};
+
 };
 
 &dspi0 {
@@ -72,6 +97,14 @@
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_i2c0_1>;
 	status = "okay";
+
+	codec: sgtl5000@0a {
+		compatible = "fsl,sgtl5000";
+		reg = <0x0a>;
+		VDDA-supply = <&reg_3p3v>;
+		VDDIO-supply = <&reg_3p3v>;
+		clocks = <&clks VF610_CLK_SAI2>;
+	};
 };
 
 &sai2 {
-- 
1.8.4

[PATCHv3 6/7] ARM: dts: Enable SGTL5000 codec based audio driver node for VF610 TOWER.

[Xiubo Li]

This patch adds and enables SGTL5000 codec support, and also specified
the corresponding SAI node.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
 arch/arm/boot/dts/vf610-twr.dts | 33 +++++++++++++++++++++++++++++++++
 1 file changed, 33 insertions(+)

diff --git a/arch/arm/boot/dts/vf610-twr.dts b/arch/arm/boot/dts/vf610-twr.dts
index e60c20c..6a8ee0a 100644
--- a/arch/arm/boot/dts/vf610-twr.dts
+++ b/arch/arm/boot/dts/vf610-twr.dts
@@ -34,6 +34,31 @@
 		};
 	};
 
+	regulators {
+		compatible = "simple-bus";
+
+		reg_3p3v: 3p3v {
+			compatible = "regulator-fixed";
+			regulator-name = "3P3V";
+			regulator-min-microvolt = <3300000>;
+			regulator-max-microvolt = <3300000>;
+			regulator-always-on;
+		};
+	};
+
+	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";
+	};
+
 };
 
 &dspi0 {
@@ -72,6 +97,14 @@
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_i2c0_1>;
 	status = "okay";
+
+	codec: sgtl5000 at 0a {
+		compatible = "fsl,sgtl5000";
+		reg = <0x0a>;
+		VDDA-supply = <&reg_3p3v>;
+		VDDIO-supply = <&reg_3p3v>;
+		clocks = <&clks VF610_CLK_SAI2>;
+	};
 };
 
 &sai2 {
-- 
1.8.4

[PATCHv3 7/7] 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/vf610-audio-sgt15000.txt        | 46 ++++++++++++++++++++++
 1 file changed, 46 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt

diff --git a/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt
new file mode 100644
index 0000000..76ff838
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.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

[PATCHv3 7/7] 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/vf610-audio-sgt15000.txt        | 46 ++++++++++++++++++++++
 1 file changed, 46 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt

diff --git a/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt
new file mode 100644
index 0000000..76ff838
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.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

[PATCHv3 7/7] 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/vf610-audio-sgt15000.txt        | 46 ++++++++++++++++++++++
 1 file changed, 46 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt

diff --git a/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.txt
new file mode 100644
index 0000000..76ff838
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/vf610-audio-sgt15000.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 at 0a {
+	       compatible = "fsl,sgtl5000";
+	       reg = <0x0a>;
+	       clocks = <&clks VF610_CLK_SAI2>;
+       };
+};
-- 
1.8.4

Re: [PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 932 bytes --]

On Tue, Dec 17, 2013 at 11:24:38AM +0800, Xiubo Li wrote:

> This adds Freescale SAI ASoC Audio support.
> This implementation is only compatible with device tree definition.

This is mostly good, there are two small issues below but I've applied it
since they don't really affect the driver as a whole.  Please send
followup patches to fix the issues below.

> +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
> +{
> +	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
> +
> +	cpu_dai->playback_dma_data = &sai->dma_params_tx;
> +	cpu_dai->capture_dma_data = &sai->dma_params_rx;

You should use snd_soc_dai_init_dma_data() for this.

> +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
> +{
> +	cpu_dai->playback_dma_data = NULL;
> +	cpu_dai->capture_dma_data = NULL;
> +
> +	snd_soc_dai_set_drvdata(cpu_dai, NULL);
> +
> +	return 0;
> +}

This is all unneeded and this function should be removed entirely.

[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Mark Brown]

On Tue, Dec 17, 2013 at 11:24:38AM +0800, Xiubo Li wrote:

> This adds Freescale SAI ASoC Audio support.
> This implementation is only compatible with device tree definition.

This is mostly good, there are two small issues below but I've applied it
since they don't really affect the driver as a whole.  Please send
followup patches to fix the issues below.

> +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
> +{
> +	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
> +
> +	cpu_dai->playback_dma_data = &sai->dma_params_tx;
> +	cpu_dai->capture_dma_data = &sai->dma_params_rx;

You should use snd_soc_dai_init_dma_data() for this.

> +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
> +{
> +	cpu_dai->playback_dma_data = NULL;
> +	cpu_dai->capture_dma_data = NULL;
> +
> +	snd_soc_dai_set_drvdata(cpu_dai, NULL);
> +
> +	return 0;
> +}

This is all unneeded and this function should be removed entirely.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: Digital signature
URL: <http://lists.infradead.org/pipermail/linux-arm-kernel/attachments/20131218/0ba92578/attachment-0001.sig>

Re: [PATCHv3 4/7] Documentation: Add device tree bindings for Freescale SAI.

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 167 bytes --]

On Tue, Dec 17, 2013 at 11:24:41AM +0800, Xiubo Li wrote:

> +	      big-endian_regs;
> +	      big-endian_data;

I've applied this and fixed these two typos, thanks.

[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[PATCHv3 4/7] Documentation: Add device tree bindings for Freescale SAI.

[Mark Brown]

On Tue, Dec 17, 2013 at 11:24:41AM +0800, Xiubo Li wrote:

> +	      big-endian_regs;
> +	      big-endian_data;

I've applied this and fixed these two typos, thanks.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: Digital signature
URL: <http://lists.infradead.org/pipermail/linux-arm-kernel/attachments/20131218/89c2a4cf/attachment.sig>

Re: [PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 1002 bytes --]

On Tue, Dec 17, 2013 at 11:24:42AM +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.

It looks like these cards should be supportable using the generic simple
card bindings that Morimoto-san implemented, you can see these in -next
as "ASoC: simple-card: add Device Tree support".  Can you check to see
if that's the case?  If it is you'll still need Kconfig to enable
building the SoC and CODEC drivers.
 
> +menuconfig SND_FSL_SOC
> +	tristate "SoC Audio for Freescale VF610 CPUs"
> +	help
> +	  Say Y or M if you want to add support for codecs attached to
> +	  the VF610 CPUs.
> +
> +	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
> +	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.

This should select the dmaengine helpers (and should really have been
split out as part of the first patch).

[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Mark Brown]

On Tue, Dec 17, 2013 at 11:24:42AM +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.

It looks like these cards should be supportable using the generic simple
card bindings that Morimoto-san implemented, you can see these in -next
as "ASoC: simple-card: add Device Tree support".  Can you check to see
if that's the case?  If it is you'll still need Kconfig to enable
building the SoC and CODEC drivers.
 
> +menuconfig SND_FSL_SOC
> +	tristate "SoC Audio for Freescale VF610 CPUs"
> +	help
> +	  Say Y or M if you want to add support for codecs attached to
> +	  the VF610 CPUs.
> +
> +	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
> +	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.

This should select the dmaengine helpers (and should really have been
split out as part of the first patch).
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: Digital signature
URL: <http://lists.infradead.org/pipermail/linux-arm-kernel/attachments/20131218/7b6e6675/attachment-0001.sig>

Re: [PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Li.Xiubo]

> On Tue, Dec 17, 2013 at 11:24:38AM +0800, Xiubo Li wrote:
> 
> > This adds Freescale SAI ASoC Audio support.
> > This implementation is only compatible with device tree definition.
> 
> This is mostly good, there are two small issues below but I've applied it
> since they don't really affect the driver as a whole.  Please send followup
> patches to fix the issues below.
> 
> > +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) {
> > +	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
> > +
> > +	cpu_dai->playback_dma_data = &sai->dma_params_tx;
> > +	cpu_dai->capture_dma_data = &sai->dma_params_rx;
> 
> You should use snd_soc_dai_init_dma_data() for this.
>

Yes, I will use it.
 
> > +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) {
> > +	cpu_dai->playback_dma_data = NULL;
> > +	cpu_dai->capture_dma_data = NULL;
> > +
> > +	snd_soc_dai_set_drvdata(cpu_dai, NULL);
> > +
> > +	return 0;
> > +}
> 
> This is all unneeded and this function should be removed entirely.

I will remove this and other unneeded functions like this.

Best Regards,
Xiubo

[PATCHv3 1/7] ALSA: Add SAI SoC Digital Audio Interface driver.

[Li.Xiubo at freescale.com]

> On Tue, Dec 17, 2013 at 11:24:38AM +0800, Xiubo Li wrote:
> 
> > This adds Freescale SAI ASoC Audio support.
> > This implementation is only compatible with device tree definition.
> 
> This is mostly good, there are two small issues below but I've applied it
> since they don't really affect the driver as a whole.  Please send followup
> patches to fix the issues below.
> 
> > +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) {
> > +	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
> > +
> > +	cpu_dai->playback_dma_data = &sai->dma_params_tx;
> > +	cpu_dai->capture_dma_data = &sai->dma_params_rx;
> 
> You should use snd_soc_dai_init_dma_data() for this.
>

Yes, I will use it.
 
> > +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) {
> > +	cpu_dai->playback_dma_data = NULL;
> > +	cpu_dai->capture_dma_data = NULL;
> > +
> > +	snd_soc_dai_set_drvdata(cpu_dai, NULL);
> > +
> > +	return 0;
> > +}
> 
> This is all unneeded and this function should be removed entirely.

I will remove this and other unneeded functions like this.

Best Regards,
Xiubo

Re: [PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Li.Xiubo]

> On Tue, Dec 17, 2013 at 11:24:42AM +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.
> 
> It looks like these cards should be supportable using the generic simple card
> bindings that Morimoto-san implemented, you can see these in -next as "ASoC:
> simple-card: add Device Tree support".  Can you check to see if that's the
> case?  If it is you'll still need Kconfig to enable building the SoC and CODEC
> drivers.
> 

Yes, almost, but maybe still need to do some adjust about it.

In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
sound driver separately. Could you apply the SAI's first if they are all fine?

I will use the simpe-card driver for the VF610-twr later.

> > +menuconfig SND_FSL_SOC
> > +	tristate "SoC Audio for Freescale VF610 CPUs"
> > +	help
> > +	  Say Y or M if you want to add support for codecs attached to
> > +	  the VF610 CPUs.
> > +
> > +	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
> > +	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.
> 
> This should select the dmaengine helpers (and should really have been split
> out as part of the first patch).

I will revise this.

Best Regards,
Xiubo

[PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Li.Xiubo at freescale.com]

> On Tue, Dec 17, 2013 at 11:24:42AM +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.
> 
> It looks like these cards should be supportable using the generic simple card
> bindings that Morimoto-san implemented, you can see these in -next as "ASoC:
> simple-card: add Device Tree support".  Can you check to see if that's the
> case?  If it is you'll still need Kconfig to enable building the SoC and CODEC
> drivers.
> 

Yes, almost, but maybe still need to do some adjust about it.

In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
sound driver separately. Could you apply the SAI's first if they are all fine?

I will use the simpe-card driver for the VF610-twr later.

> > +menuconfig SND_FSL_SOC
> > +	tristate "SoC Audio for Freescale VF610 CPUs"
> > +	help
> > +	  Say Y or M if you want to add support for codecs attached to
> > +	  the VF610 CPUs.
> > +
> > +	  This will enable Freeacale SAI and SGTL5000 codec, and an extra
> > +	  TWR-AUDIO-SGTL sub-board is needed for SGTL5000.
> 
> This should select the dmaengine helpers (and should really have been split
> out as part of the first patch).

I will revise this.

Best Regards,
Xiubo

Re: [PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 420 bytes --]

On Thu, Dec 19, 2013 at 05:19:26AM +0000, Li.Xiubo@freescale.com wrote:

> In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
> apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
> sound driver separately. Could you apply the SAI's first if they are all fine?

What are you asking me to apply here?  I already applied your first
patch as I said, is that not everything?

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 176 bytes --]

_______________________________________________
linux-arm-kernel mailing list
linux-arm-kernel@lists.infradead.org
http://lists.infradead.org/mailman/listinfo/linux-arm-kernel

[PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Mark Brown]

On Thu, Dec 19, 2013 at 05:19:26AM +0000, Li.Xiubo at freescale.com wrote:

> In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
> apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
> sound driver separately. Could you apply the SAI's first if they are all fine?

What are you asking me to apply here?  I already applied your first
patch as I said, is that not everything?
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: Digital signature
URL: <http://lists.infradead.org/pipermail/linux-arm-kernel/attachments/20131219/45b1a6fd/attachment.sig>

Re: [PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Li.Xiubo]

Hi Mark,

> On Thu, Dec 19, 2013 at 05:19:26AM +0000, Li.Xiubo@freescale.com wrote:
> 
> > In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
> > apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
> > sound driver separately. Could you apply the SAI's first if they are all
> fine?
> 
> What are you asking me to apply here?  I already applied your first
> patch as I said, is that not everything?

Sorry for understanding.

Please ignore the series of patches for v4.

I will send the followed patch to fix that issues you have pointed.

Thanks,

Best Regards,
Xiubo

[PATCHv3 5/7] ASoC: fsl: Add SGTL5000 based audio machine driver.

[Li.Xiubo at freescale.com]

Hi Mark,

> On Thu, Dec 19, 2013 at 05:19:26AM +0000, Li.Xiubo at freescale.com wrote:
> 
> > In this series of patches, the PATCH1~PATCH4 are about SAI only, and can be
> > apart from VF610-twr sound driver, I will sent the SAI driver and VF610-twr
> > sound driver separately. Could you apply the SAI's first if they are all
> fine?
> 
> What are you asking me to apply here?  I already applied your first
> patch as I said, is that not everything?

Sorry for understanding.

Please ignore the series of patches for v4.

I will send the followed patch to fix that issues you have pointed.

Thanks,

Best Regards,
Xiubo

Re: [PATCHv3 4/7] Documentation: Add device tree bindings for Freescale SAI.

[Li.Xiubo]

Hi Mark,

> On Tue, Dec 17, 2013 at 11:24:41AM +0800, Xiubo Li wrote:
> 
> > +	      big-endian_regs;
> > +	      big-endian_data;
> 
> I've applied this and fixed these two typos, thanks.

There is something wrong with my mails these days.

I will revise this then.

Thanks very much,


Best Regards,
Xiubo

[PATCHv3 4/7] Documentation: Add device tree bindings for Freescale SAI.

[Li.Xiubo at freescale.com]

Hi Mark,

> On Tue, Dec 17, 2013 at 11:24:41AM +0800, Xiubo Li wrote:
> 
> > +	      big-endian_regs;
> > +	      big-endian_data;
> 
> I've applied this and fixed these two typos, thanks.

There is something wrong with my mails these days.

I will revise this then.

Thanks very much,


Best Regards,
Xiubo