[PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

From: Grant Likely <grant.likely@secretlab.ca>

Simple utility layer for creating ASoC machine instances based on data
in the OpenFirmware device tree.  OF aware platform drivers and codec
drivers register themselves with this framework and the framework
automatically instantiates a machine driver.

This is most likely temporary glue code to work around limitations in
the ASoC v1 framework.  I expect ASoC v2 won't need this.
---

 sound/soc/Kconfig  |    6 ++
 sound/soc/Makefile |    1 
 sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 178 insertions(+), 0 deletions(-)

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac..c5736e5 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -23,6 +23,12 @@ config SND_SOC
 	  This ASoC audio support can also be built as a module.  If so, the module
 	  will be called snd-soc-core.
 
+config SND_SOC_OF
+	tristate "OF helpers for SoC audio support"
+	depends on SND_SOC
+	---help---
+	  Add support for OpenFirmware device tree descriptions of sound device
+
 # All the supported Soc's
 source "sound/soc/at91/Kconfig"
 source "sound/soc/pxa/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db21..191c2e5 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC_OF)	+= soc-of.o
diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
new file mode 100644
index 0000000..9694979
--- /dev/null
+++ b/sound/soc/soc-of.c
@@ -0,0 +1,171 @@
+/*
+ * OF helpers for ALSA SoC Layer
+ *
+ * Copyright (C) 2008, Secret Lab Technologies Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/of.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-of.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+
+DEFINE_MUTEX(of_snd_soc_mutex);
+LIST_HEAD(of_snd_soc_device_list);
+static int of_snd_soc_next_index;
+
+struct of_snd_soc_device {
+	int id;
+	struct list_head list;
+	struct snd_soc_device device;
+	struct snd_soc_machine machine;
+	struct snd_soc_dai_link dai_link;
+	struct platform_device *pdev;
+	struct device_node *platform_node;
+	struct device_node *codec_node;
+};
+
+static struct snd_soc_ops of_snd_soc_ops = {
+};
+
+static struct of_snd_soc_device *
+of_snd_soc_get_device(struct device_node *codec_node)
+{
+	struct of_snd_soc_device *of_soc;
+
+	list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
+		if (of_soc->codec_node == codec_node)
+			return of_soc;
+	}
+
+	of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
+	if (!of_soc)
+		return NULL;
+
+	/* Initialize the structure and add it to the global list */
+	of_soc->codec_node = codec_node;
+	of_soc->id = of_snd_soc_next_index++;
+	of_soc->machine.dai_link = &of_soc->dai_link;
+	of_soc->machine.num_links = 1;
+	of_soc->device.machine = &of_soc->machine;
+	of_soc->dai_link.ops = &of_snd_soc_ops;
+	list_add(&of_soc->list, &of_snd_soc_device_list);
+
+	return of_soc;
+}
+
+static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
+{
+	struct platform_device *pdev;
+	int rc;
+
+	/* Only register the device if both the codec and platform have
+	 * been registered */
+	if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
+		return;
+
+	pr_info("platform<-->codec match achieved; registering machine\n");
+
+	pdev = platform_device_alloc("soc-audio", of_soc->id);
+	if (!pdev) {
+		pr_err("of_soc: platform_device_alloc() failed\n");
+		return;
+	}
+
+	pdev->dev.platform_data = of_soc;
+	platform_set_drvdata(pdev, &of_soc->device);
+	of_soc->device.dev = &pdev->dev;
+
+	/* The ASoC device is complete; register it */
+	rc = platform_device_add(pdev);
+	if (rc) {
+		pr_err("of_soc: platform_device_add() failed\n");
+		return;
+	}
+
+}
+
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
+			      void *codec_data, struct snd_soc_codec_dai *dai,
+			      struct device_node *node)
+{
+	struct of_snd_soc_device *of_soc;
+	int rc = 0;
+
+	pr_info("registering ASoC codec driver: %s\n", node->full_name);
+
+	mutex_lock(&of_snd_soc_mutex);
+	of_soc = of_snd_soc_get_device(node);
+	if (!of_soc) {
+		rc = -ENOMEM;
+		goto out;
+	}
+
+	/* Store the codec data */
+	of_soc->device.codec_data = codec_data;
+	of_soc->device.codec_dev = codec_dev;
+	of_soc->dai_link.name = node->name;
+	of_soc->dai_link.stream_name = node->name;
+	of_soc->dai_link.codec_dai = dai;
+
+	/* Now try to register the SoC device */
+	of_snd_soc_register_device(of_soc);
+
+ out:
+	mutex_unlock(&of_snd_soc_mutex);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
+
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
+				 struct device_node *node,
+				 struct snd_soc_cpu_dai *cpu_dai)
+{
+	struct of_snd_soc_device *of_soc;
+	struct device_node *codec_node;
+	const phandle *handle;
+	int len, rc = 0;
+
+	pr_info("registering ASoC platform driver: %s\n", node->full_name);
+
+	handle = of_get_property(node, "codec-handle", &len);
+	if (!handle || len < sizeof(handle))
+		return -ENODEV;
+	codec_node = of_find_node_by_phandle(*handle);
+	if (!codec_node)
+		return -ENODEV;
+	pr_info("looking for codec: %s\n", codec_node->full_name);
+
+	mutex_lock(&of_snd_soc_mutex);
+	of_soc = of_snd_soc_get_device(codec_node);
+	if (!of_soc) {
+		rc = -ENOMEM;
+		goto out;
+	}
+
+	of_soc->platform_node = node;
+	of_soc->dai_link.cpu_dai = cpu_dai;
+	of_soc->device.platform = platform;
+	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+
+	/* Now try to register the SoC device */
+	of_snd_soc_register_device(of_soc);
+
+ out:
+	mutex_unlock(&of_snd_soc_mutex);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);

[PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

From: Grant Likely <grant.likely@secretlab.ca>

Simple utility layer for creating ASoC machine instances based on data
in the OpenFirmware device tree.  OF aware platform drivers and codec
drivers register themselves with this framework and the framework
automatically instantiates a machine driver.

This is most likely temporary glue code to work around limitations in
the ASoC v1 framework.  I expect ASoC v2 won't need this.
---

 sound/soc/Kconfig  |    6 ++
 sound/soc/Makefile |    1 
 sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 178 insertions(+), 0 deletions(-)

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac..c5736e5 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -23,6 +23,12 @@ config SND_SOC
 	  This ASoC audio support can also be built as a module.  If so, the module
 	  will be called snd-soc-core.
 
+config SND_SOC_OF
+	tristate "OF helpers for SoC audio support"
+	depends on SND_SOC
+	---help---
+	  Add support for OpenFirmware device tree descriptions of sound device
+
 # All the supported Soc's
 source "sound/soc/at91/Kconfig"
 source "sound/soc/pxa/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db21..191c2e5 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC_OF)	+= soc-of.o
diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
new file mode 100644
index 0000000..9694979
--- /dev/null
+++ b/sound/soc/soc-of.c
@@ -0,0 +1,171 @@
+/*
+ * OF helpers for ALSA SoC Layer
+ *
+ * Copyright (C) 2008, Secret Lab Technologies Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/of.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-of.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+
+DEFINE_MUTEX(of_snd_soc_mutex);
+LIST_HEAD(of_snd_soc_device_list);
+static int of_snd_soc_next_index;
+
+struct of_snd_soc_device {
+	int id;
+	struct list_head list;
+	struct snd_soc_device device;
+	struct snd_soc_machine machine;
+	struct snd_soc_dai_link dai_link;
+	struct platform_device *pdev;
+	struct device_node *platform_node;
+	struct device_node *codec_node;
+};
+
+static struct snd_soc_ops of_snd_soc_ops = {
+};
+
+static struct of_snd_soc_device *
+of_snd_soc_get_device(struct device_node *codec_node)
+{
+	struct of_snd_soc_device *of_soc;
+
+	list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
+		if (of_soc->codec_node == codec_node)
+			return of_soc;
+	}
+
+	of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
+	if (!of_soc)
+		return NULL;
+
+	/* Initialize the structure and add it to the global list */
+	of_soc->codec_node = codec_node;
+	of_soc->id = of_snd_soc_next_index++;
+	of_soc->machine.dai_link = &of_soc->dai_link;
+	of_soc->machine.num_links = 1;
+	of_soc->device.machine = &of_soc->machine;
+	of_soc->dai_link.ops = &of_snd_soc_ops;
+	list_add(&of_soc->list, &of_snd_soc_device_list);
+
+	return of_soc;
+}
+
+static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
+{
+	struct platform_device *pdev;
+	int rc;
+
+	/* Only register the device if both the codec and platform have
+	 * been registered */
+	if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
+		return;
+
+	pr_info("platform<-->codec match achieved; registering machine\n");
+
+	pdev = platform_device_alloc("soc-audio", of_soc->id);
+	if (!pdev) {
+		pr_err("of_soc: platform_device_alloc() failed\n");
+		return;
+	}
+
+	pdev->dev.platform_data = of_soc;
+	platform_set_drvdata(pdev, &of_soc->device);
+	of_soc->device.dev = &pdev->dev;
+
+	/* The ASoC device is complete; register it */
+	rc = platform_device_add(pdev);
+	if (rc) {
+		pr_err("of_soc: platform_device_add() failed\n");
+		return;
+	}
+
+}
+
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
+			      void *codec_data, struct snd_soc_codec_dai *dai,
+			      struct device_node *node)
+{
+	struct of_snd_soc_device *of_soc;
+	int rc = 0;
+
+	pr_info("registering ASoC codec driver: %s\n", node->full_name);
+
+	mutex_lock(&of_snd_soc_mutex);
+	of_soc = of_snd_soc_get_device(node);
+	if (!of_soc) {
+		rc = -ENOMEM;
+		goto out;
+	}
+
+	/* Store the codec data */
+	of_soc->device.codec_data = codec_data;
+	of_soc->device.codec_dev = codec_dev;
+	of_soc->dai_link.name = node->name;
+	of_soc->dai_link.stream_name = node->name;
+	of_soc->dai_link.codec_dai = dai;
+
+	/* Now try to register the SoC device */
+	of_snd_soc_register_device(of_soc);
+
+ out:
+	mutex_unlock(&of_snd_soc_mutex);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
+
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
+				 struct device_node *node,
+				 struct snd_soc_cpu_dai *cpu_dai)
+{
+	struct of_snd_soc_device *of_soc;
+	struct device_node *codec_node;
+	const phandle *handle;
+	int len, rc = 0;
+
+	pr_info("registering ASoC platform driver: %s\n", node->full_name);
+
+	handle = of_get_property(node, "codec-handle", &len);
+	if (!handle || len < sizeof(handle))
+		return -ENODEV;
+	codec_node = of_find_node_by_phandle(*handle);
+	if (!codec_node)
+		return -ENODEV;
+	pr_info("looking for codec: %s\n", codec_node->full_name);
+
+	mutex_lock(&of_snd_soc_mutex);
+	of_soc = of_snd_soc_get_device(codec_node);
+	if (!of_soc) {
+		rc = -ENOMEM;
+		goto out;
+	}
+
+	of_soc->platform_node = node;
+	of_soc->dai_link.cpu_dai = cpu_dai;
+	of_soc->device.platform = platform;
+	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+
+	/* Now try to register the SoC device */
+	of_snd_soc_register_device(of_soc);
+
+ out:
+	mutex_unlock(&of_snd_soc_mutex);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);

[PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

From: Grant Likely <grant.likely@secretlab.ca>

This is an I2S bus driver for the MPC5200 PSC device.  It is probably
will not be merged as-is because it uses v1 of the ASoC API, but I want
to get it out there for comments.
---

 sound/soc/fsl/Kconfig           |    6 
 sound/soc/fsl/Makefile          |    2 
 sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 907 insertions(+), 0 deletions(-)

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 257101f..5daa8d3 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
 	help
 	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
 
+config SND_SOC_MPC5200_I2S
+	bool "Freescale MPC5200 PSC in I2S mode driver"
+	depends on SND_SOC && PPC_MPC52xx
+	help
+	  Say Y here to support the MPC5200 PSCs in I2S mode.
+
 endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 62f680a..98729a1 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
 # MPC8610 Platform Support
 obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
 
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
new file mode 100644
index 0000000..81d0933
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -0,0 +1,899 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
+
+/**
+ * PSC_I2S_RATES: sample rates supported by the I2S
+ *
+ * This driver currently only supports the PSC running in I2S slave mode,
+ * which means the codec determines the sample rate.  Therefore, we tell
+ * ALSA that we support all rates and let the codec driver decide what rates
+ * are really supported.
+ */
+#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
+			SNDRV_PCM_RATE_CONTINUOUS)
+
+/**
+ * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
+ */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
+			 SNDRV_PCM_FMTBIT_S32_BE)
+
+/**
+ * psc_i2s_stream - Data specific to a single stream (playback or capture)
+ * @active:		flag indicating if the stream is active
+ * @psc_i2s:		pointer back to parent psc_i2s data structure
+ * @bcom_task:		bestcomm task structure
+ * @irq:		irq number for bestcomm task
+ * @period_start:	physical address of start of DMA region
+ * @period_end:		physical address of end of DMA region
+ * @period_next_pt:	physical address of next DMA buffer to enqueue
+ * @period_bytes:	size of DMA period in bytes
+ */
+struct psc_i2s_stream {
+	int active;
+	struct psc_i2s *psc_i2s;
+	struct bcom_task *bcom_task;
+	int irq;
+	struct snd_pcm_substream *stream;
+	dma_addr_t period_start;
+	dma_addr_t period_end;
+	dma_addr_t period_next_pt;
+	dma_addr_t period_current_pt;
+	int period_bytes;
+};
+
+/**
+ * psc_i2s - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @playback: the number of playback streams opened
+ * @capture: the number of capture streams opened
+ * @dai: the CPU DAI for this device
+ * @playback_stream: Playback stream context data
+ * @capture_stream: Capture stream context data
+ */
+struct psc_i2s {
+	char name[32];
+	struct mpc52xx_psc __iomem *psc_regs;
+	struct mpc52xx_psc_fifo __iomem *fifo_regs;
+	unsigned int irq;
+	struct device *dev;
+	struct snd_soc_cpu_dai dai;
+	spinlock_t lock;
+
+	/* per-stream data */
+	struct psc_i2s_stream playback_stream;
+	struct psc_i2s_stream capture_stream;
+
+	/* Statistics */
+	struct {
+		int overrun_count;
+		int underrun_count;
+	} stats;
+};
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
+{
+	struct psc_i2s *psc_i2s = _psc_i2s;
+	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	u16 imr;
+	u16 isr;
+
+	isr = in_be16(&regs->mpc52xx_psc_isr);
+	imr = in_be16(&regs->mpc52xx_psc_imr);
+
+	/* Playback underrun error */
+	if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
+		psc_i2s->stats.underrun_count++;
+
+	/* Capture overrun error */
+	if (isr & imr & MPC52xx_PSC_IMR_ORERR)
+		psc_i2s->stats.overrun_count++;
+
+	out_8(&regs->command, 4 << 4);	/* reset the error status */
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue.  Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
+{
+	struct bcom_bd *bd;
+
+	/* Prepare and enqueue the next buffer descriptor */
+	bd = bcom_prepare_next_buffer(s->bcom_task);
+	bd->status = s->period_bytes;
+	bd->data[0] = s->period_next_pt;
+	bcom_submit_next_buffer(s->bcom_task, NULL);
+
+	/* Update for next period */
+	s->period_next_pt += s->period_bytes;
+	if (s->period_next_pt >= s->period_end)
+		s->period_next_pt = s->period_start;
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
+{
+	struct psc_i2s_stream *s = _psc_i2s_stream;
+
+	//spin_lock(&s->psc_i2s->lock);
+
+	/* For each finished period, dequeue the completed period buffer
+	 * and enqueue a new one in it's place. */
+	while (bcom_buffer_done(s->bcom_task)) {
+		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+		s->period_current_pt += s->period_bytes;
+		if (s->period_current_pt >= s->period_end)
+			s->period_current_pt = s->period_start;
+		psc_i2s_bcom_enqueue_next_buffer(s);
+		bcom_enable(s->bcom_task);
+	}
+
+	//spin_unlock(&s->psc_i2s->lock);
+
+	/* If the stream is active, then also inform the PCM middle layer
+	 * of the period finished event. */
+	if (s->active)
+		snd_pcm_period_elapsed(s->stream);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_startup: create a new substream
+ *
+ * This is the first function called when a stream is opened.
+ *
+ * If this is the first stream open, then grab the IRQ and program most of
+ * the PSC registers.
+ */
+static int psc_i2s_startup(struct snd_pcm_substream *substream)
+{
+	int playback_irq, capture_irq, rc;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
+
+	/* Disable all interrupts and reset the PSC */
+	out_be16(&regs->mpc52xx_psc_imr, 0);
+	out_8(&regs->command, 3 << 4); /* reset transmitter */
+	out_8(&regs->command, 2 << 4); /* reset receiver */
+	out_8(&regs->command, 1 << 4); /* reset mode */
+	out_8(&regs->command, 4 << 4); /* reset error */
+
+	/* Default to CODEC8 mode */
+	out_be32(&regs->sicr,
+		 MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+		 MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
+
+	/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
+	/* Second write to mode: register Normal mode for non loopback */
+	out_8(&regs->mode, 0);
+	out_8(&regs->mode, 0);
+
+	/* Set the TX and RX fifo alarm thresholds */
+	out_be16(&fiforegs->rfalarm, 0x100);	/* set RFALARM level */
+	out_8(&fiforegs->rfcntl, 0x4);		/* set RFGRAN level (bytes) */
+	out_be16(&fiforegs->tfalarm, 0x100);	/* set TFALARM level */
+	out_8(&fiforegs->tfcntl, 0x7);		/* set TFGRAN level (bytes*4) */
+
+	/* Setup the IRQs */
+	playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
+	capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
+	rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
+			 "psc-i2s-status", psc_i2s);
+	rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
+			  "psc-i2s-capture", &psc_i2s->capture_stream);
+	rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
+			  "psc-i2s-playback", &psc_i2s->playback_stream);
+	if (rc) {
+		free_irq(psc_i2s->irq, psc_i2s);
+		free_irq(capture_irq, &psc_i2s->capture_stream);
+		free_irq(playback_irq, &psc_i2s->playback_stream);
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	u32 sicr;
+
+	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
+		__FUNCTION__, substream, params_period_size(params),
+		params_period_bytes(params), params_periods(params),
+		params_buffer_size(params), params_buffer_bytes(params));
+
+	sicr = MPC52xx_PSC_SICR_DTS1 |
+	       MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
+	switch (params_format(params)) {
+	 case SNDRV_PCM_FORMAT_S8:
+		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
+		break;
+	 case SNDRV_PCM_FORMAT_S16_BE:
+		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
+		break;
+	 case SNDRV_PCM_FORMAT_S24_BE:
+		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
+		break;
+	 case SNDRV_PCM_FORMAT_S32_BE:
+		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
+		break;
+	 default:
+		dev_dbg(psc_i2s->dev, "invalid format\n");
+		return -EINVAL;
+	}
+	out_be32(&psc_i2s->psc_regs->sicr, sicr);
+
+	//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	//if (rc) {
+	//	dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
+	//	return rc;
+	//}
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+{
+	//return snd_pcm_lib_free_pages(substream);
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+/**
+ * psc_i2s_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct psc_i2s_stream *s;
+	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	u16 imr;
+	u8 psc_cmd;
+	long flags;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture_stream;
+	else
+		s = &psc_i2s->playback_stream;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
+		" stream_id=%i\n",
+		substream, cmd, substream->pstr->stream);
+
+	switch (cmd) {
+	 case SNDRV_PCM_TRIGGER_START:
+		s->period_bytes = frames_to_bytes(runtime,
+						  runtime->period_size);
+		s->period_start = virt_to_phys(runtime->dma_area);
+		s->period_end = s->period_start +
+				(s->period_bytes * runtime->periods);
+		s->period_next_pt = s->period_start;
+		s->period_current_pt = s->period_start;
+		s->active = 1;
+
+		/* First; reset everything */
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			out_8(&regs->command, MPC52xx_PSC_RST_RX);
+			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+		} else {
+			out_8(&regs->command, MPC52xx_PSC_RST_TX);
+			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+		}
+
+		/* Next, fill up the bestcomm bd queue and enable DMA.
+		 * This will begin filling the PSC's fifo. */
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+			bcom_gen_bd_rx_reset(s->bcom_task);
+		else
+			bcom_gen_bd_tx_reset(s->bcom_task);
+		while (!bcom_queue_full(s->bcom_task))
+			psc_i2s_bcom_enqueue_next_buffer(s);
+		bcom_enable(s->bcom_task);
+
+		/* Update interrupt enable settings.  This must be done
+		 * before the PSC is enabled so that TX underrun events
+		 * are not missed. */
+		imr = 0;
+		if (psc_i2s->playback_stream.active)
+			imr |= MPC52xx_PSC_IMR_TXEMP;
+		if (psc_i2s->capture_stream.active)
+			imr |= MPC52xx_PSC_IMR_ORERR;
+		out_be16(&regs->isr_imr.imr, imr);
+
+		/* Due to errata in the i2s mode; need to line up enabling
+		 * the transmitter with a transition on the frame sync
+		 * line */
+
+		spin_lock_irqsave(&psc_i2s->lock, flags);
+		/* first make sure it is low */
+		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
+		/* then wait for the transition to high */
+		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
+		/* Finally, enable the PSC.
+		 * Receiver must always be enabled; even when we only want
+		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
+		psc_cmd = MPC52xx_PSC_RX_ENABLE;
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
+		out_8(&regs->command, psc_cmd);
+		spin_unlock_irqrestore(&psc_i2s->lock, flags);
+
+		break;
+
+	 case SNDRV_PCM_TRIGGER_STOP:
+		/* Turn off the PSC */
+		s->active = 0;
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			if (!psc_i2s->playback_stream.active) {
+				out_8(&regs->command, 2 << 4);	/* reset rx */
+				out_8(&regs->command, 3 << 4);	/* reset tx */
+				out_8(&regs->command, 4 << 4);	/* reset err */
+			}
+		} else {
+			out_8(&regs->command, 3 << 4);	/* reset tx */
+			out_8(&regs->command, 4 << 4);	/* reset err */
+			if (!psc_i2s->capture_stream.active)
+				out_8(&regs->command, 2 << 4);	/* reset rx */
+		}
+
+		bcom_disable(s->bcom_task);
+		while (!bcom_queue_empty(s->bcom_task))
+			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+		break;
+
+	 default:
+		dev_dbg(psc_i2s->dev, "invalid command\n");
+		return -EINVAL;
+	}
+
+	/* Update interrupt enable settings */
+	imr = 0;
+	if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
+	if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
+	out_be16(&regs->isr_imr.imr, imr);
+
+	return 0;
+}
+
+/**
+ * psc_i2s_shutdown: shutdown the data transfer on a stream
+ *
+ * Shutdown the PSC if there are no other substreams open.
+ */
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
+
+	/*
+	 * If this is the last active substream, disable the PSC and release
+	 * the IRQ.
+	 */
+	if (!psc_i2s->playback_stream.active &&
+	    !psc_i2s->capture_stream.active) {
+		/* TODO: shut off channels */
+		free_irq(psc_i2s->irq, psc_i2s);
+		free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
+			 &psc_i2s->capture_stream);
+		free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
+			 &psc_i2s->playback_stream);
+	}
+}
+
+/**
+ * psc_i2s_set_sysclk: set the clock frequency and direction
+ *
+ * This function is called by the machine driver to tell us what the clock
+ * frequency and direction are.
+ *
+ * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
+ * and we don't care about the frequency.  Return an error if the direction
+ * is not SND_SOC_CLOCK_IN.
+ *
+ * @clk_id: reserved, should be zero
+ * @freq: the frequency of the given clock ID, currently ignored
+ * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
+ */
+static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+			      int clk_id, unsigned int freq, int dir)
+{
+	struct psc_i2s *psc_i2s = cpu_dai->private_data;
+	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+				cpu_dai, dir);
+	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
+}
+
+/**
+ * psc_i2s_set_fmt: set the serial format.
+ *
+ * This function is called by the machine driver to tell us what serial
+ * format to use.
+ *
+ * This driver only supports I2S mode.  Return an error if the format is
+ * not SND_SOC_DAIFMT_I2S.
+ *
+ * @format: one of SND_SOC_DAIFMT_xxx
+ */
+static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
+{
+	struct psc_i2s *psc_i2s = cpu_dai->private_data;
+	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+				cpu_dai, format);
+	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_i2s_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_cpu_dai psc_i2s_dai_template = {
+	.type = SND_SOC_DAI_I2S,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PSC_I2S_RATES,
+		.formats = PSC_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PSC_I2S_RATES,
+		.formats = PSC_I2S_FORMATS,
+	},
+	.ops = {
+		.startup = psc_i2s_startup,
+		.hw_params = psc_i2s_hw_params,
+		.hw_free = psc_i2s_hw_free,
+		.shutdown = psc_i2s_shutdown,
+		.trigger = psc_i2s_trigger,
+	},
+	.dai_ops = {
+		.set_sysclk = psc_i2s_set_sysclk,
+		.set_fmt = psc_i2s_set_fmt,
+	},
+};
+
+/* ---------------------------------------------------------------------
+ * The PSC I2S 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
+	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+	.formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
+		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+	.rate_min = 8000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.period_bytes_max	= 1024 * 1024,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 1024 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 256,
+	.buffer_bytes_max	= 2 * 1024 * 1024,
+	.fifo_size		= 0,
+};
+
+static unsigned int psc_i2s_fixed_rates[] = {
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+};
+
+static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
+	.count = ARRAY_SIZE(psc_i2s_fixed_rates),
+	.list = psc_i2s_fixed_rates,
+	.mask = 0,
+};
+
+static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+	int rc;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture_stream;
+	else
+		s = &psc_i2s->playback_stream;
+
+	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
+
+	rc = snd_pcm_hw_constraint_integer(substream->runtime,
+					   SNDRV_PCM_HW_PARAM_PERIODS);
+	if (rc < 0) {
+		dev_err(psc_i2s->dev, "invalid buffer size\n");
+		return rc;
+	}
+	rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
+					SNDRV_PCM_HW_PARAM_RATE,
+					&psc_i2s_constraints_rates);
+	if (rc < 0) {
+		dev_err(psc_i2s->dev, "invalid rate\n");
+		return rc;
+	}
+
+	s->stream = substream;
+	return 0;
+}
+
+static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture_stream;
+	else
+		s = &psc_i2s->playback_stream;
+
+	s->stream = NULL;
+	return 0;
+}
+
+static snd_pcm_uframes_t
+psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+	dma_addr_t count;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture_stream;
+	else
+		s = &psc_i2s->playback_stream;
+
+	/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
+	count = s->period_current_pt - s->period_start;
+
+	return bytes_to_frames(substream->runtime, count);
+}
+
+static struct snd_pcm_ops psc_i2s_pcm_ops = {
+	.open		= psc_i2s_pcm_open,
+	.close		= psc_i2s_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.pointer	= psc_i2s_pcm_pointer,
+};
+
+static u64 psc_i2s_pcm_dmamask = 0xffffffff;
+static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+			   struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
+	int rc = 0;
+
+	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
+		card, dai, pcm);
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+				 &pcm->streams[0].substream->dma_buffer);
+	if (rc) {
+		dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
+		return -ENOMEM;
+	}
+
+	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+				 &pcm->streams[1].substream->dma_buffer);
+	if (rc) {
+		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+		dev_err(card->dev, "Can't allocate capture DMA buffer\n");
+		return -ENOMEM;
+	}
+
+	return 0;
+}
+
+static void psc_i2s_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_pcm_substream *substream;
+	int stream;
+
+	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (substream) {
+			snd_dma_free_pages(&substream->dma_buffer);
+			substream->dma_buffer.area = NULL;
+			substream->dma_buffer.addr = 0;
+		}
+	}
+}
+
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
+	.name		= "mpc5200-psc-audio",
+	.pcm_ops	= &psc_i2s_pcm_ops,
+	.pcm_new	= &psc_i2s_pcm_new,
+	.pcm_free	= &psc_i2s_pcm_free,
+};
+
+/* ---------------------------------------------------------------------
+ * Sysfs attributes for debugging
+ */
+
+static ssize_t psc_i2s_status_show(struct device *dev,
+			   struct device_attribute *attr, char *buf)
+{
+	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+
+	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
+			in_be16(&psc_i2s->psc_regs->sr_csr.status),
+			in_be32(&psc_i2s->psc_regs->sicr),
+			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
+			in_be16(&psc_i2s->fifo_regs->rfstat),
+			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
+			in_be16(&psc_i2s->fifo_regs->tfstat));
+}
+
+static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
+				   const char *name)
+{
+	if (strcmp(name, "playback_underrun") == 0)
+		return &psc_i2s->stats.underrun_count;
+	if (strcmp(name, "capture_overrun") == 0)
+		return &psc_i2s->stats.overrun_count;
+
+	return NULL;
+}
+
+static ssize_t psc_i2s_stat_show(struct device *dev,
+				 struct device_attribute *attr, char *buf)
+{
+	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+	int *attrib;
+
+	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+	if (!attrib)
+		return 0;
+
+	return sprintf(buf, "%i\n", *attrib);
+}
+
+static ssize_t psc_i2s_stat_store(struct device *dev,
+				  struct device_attribute *attr,
+				  const char *buf,
+				  size_t count)
+{
+	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+	int *attrib;
+
+	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+	if (!attrib)
+		return 0;
+
+	*attrib = simple_strtoul(buf, NULL, 0);
+	return count;
+}
+
+DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
+DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
+DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_i2s_of_probe(struct of_device *op,
+				      const struct of_device_id *match)
+{
+	phys_addr_t fifo;
+	struct psc_i2s *psc_i2s;
+	struct resource res;
+	int size, psc_id, irq, rc;
+	const __be32 *prop;
+	void __iomem *regs;
+
+	dev_dbg(&op->dev, "probing psc i2s device\n");
+
+	/* Get the PSC ID */
+	prop = of_get_property(op->node, "cell-index", &size);
+	if (!prop || size < sizeof *prop)
+		return -ENODEV;
+	psc_id = be32_to_cpu(*prop);
+
+	/* Fetch the registers and IRQ of the PSC */
+	irq = irq_of_parse_and_map(op->node, 0);
+	if (of_address_to_resource(op->node, 0, &res)) {
+		dev_err(&op->dev, "Missing reg property\n");
+		return -ENODEV;
+	}
+	regs = ioremap(res.start, 1 + res.end - res.start);
+	if (!regs) {
+		dev_err(&op->dev, "Could not map registers\n");
+		return -ENODEV;
+	}
+
+	/* Allocate and initialize the driver private data */
+	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
+	if (!psc_i2s) {
+		iounmap(regs);
+		return -ENOMEM;
+	}
+	spin_lock_init(&psc_i2s->lock);
+	psc_i2s->irq = irq;
+	psc_i2s->psc_regs = regs;
+	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
+	psc_i2s->dev = &op->dev;
+	psc_i2s->playback_stream.psc_i2s = psc_i2s;
+	psc_i2s->capture_stream.psc_i2s = psc_i2s;
+	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
+
+	/* Fill out the CPU DAI structure */
+	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
+	psc_i2s->dai.private_data = psc_i2s;
+	psc_i2s->dai.name = psc_i2s->name;
+	psc_i2s->dai.id = psc_id;
+
+	/* Find the address of the fifo data registers and setup the
+	 * DMA tasks */
+	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+	psc_i2s->capture_stream.bcom_task =
+		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
+	psc_i2s->playback_stream.bcom_task =
+		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
+	if (!psc_i2s->capture_stream.bcom_task ||
+	    !psc_i2s->playback_stream.bcom_task) {
+		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+		iounmap(regs);
+		kfree(psc_i2s);
+		return -ENODEV;
+	}
+
+	/* Save what we've done so it can be found again later */
+	dev_set_drvdata(&op->dev, psc_i2s);
+
+	/* Register the SYSFS files */
+	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
+	rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
+	rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
+	if (rc)
+		dev_info(psc_i2s->dev, "error creating sysfs files\n");
+
+	/* Tell the ASoC OF helpers about it */
+	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
+				     &psc_i2s->dai);
+
+	return 0;
+}
+
+static int __devexit psc_i2s_of_remove(struct of_device *op)
+{
+	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
+
+	dev_dbg(&op->dev, "psc_i2s_remove()\n");
+
+	bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
+	bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
+
+	iounmap(psc_i2s->psc_regs);
+	iounmap(psc_i2s->fifo_regs);
+	kfree(psc_i2s);
+	dev_set_drvdata(&op->dev, NULL);
+
+	return 0;
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_i2s_match[] __devinitdata = {
+	{ .compatible = "fsl,mpc5200-psc-i2s", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, psc_i2s_match);
+
+static struct of_platform_driver psc_i2s_driver = {
+	.match_table = psc_i2s_match,
+	.probe = psc_i2s_of_probe,
+	.remove = __devexit_p(psc_i2s_of_remove),
+	.driver = {
+		.name = "mpc5200-psc-i2s",
+		.owner = THIS_MODULE,
+	},
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in I2S mode.
+ */
+static int __init psc_i2s_init(void)
+{
+	return of_register_platform_driver(&psc_i2s_driver);
+}
+module_init(psc_i2s_init);
+
+static void __exit psc_i2s_exit(void)
+{
+	of_unregister_platform_driver(&psc_i2s_driver);
+}
+module_exit(psc_i2s_exit);
+
+

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Liam Girdwood]

On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
> 
> This is an I2S bus driver for the MPC5200 PSC device.  It is probably
> will not be merged as-is because it uses v1 of the ASoC API, but I want
> to get it out there for comments.

Looks good, just minor comments.

> ---
> 
>  sound/soc/fsl/Kconfig           |    6 
>  sound/soc/fsl/Makefile          |    2 
>  sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>  3 files changed, 907 insertions(+), 0 deletions(-)
> 
> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
> index 257101f..5daa8d3 100644
> --- a/sound/soc/fsl/Kconfig
> +++ b/sound/soc/fsl/Kconfig
> @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>  	help
>  	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>  
> +config SND_SOC_MPC5200_I2S
> +	bool "Freescale MPC5200 PSC in I2S mode driver"

Is this built-in only ?

> +	depends on SND_SOC && PPC_MPC52xx
> +	help
> +	  Say Y here to support the MPC5200 PSCs in I2S mode.
> +
>  endmenu
> diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
> index 62f680a..98729a1 100644
> --- a/sound/soc/fsl/Makefile
> +++ b/sound/soc/fsl/Makefile
> @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>  # MPC8610 Platform Support
>  obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>  
> +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
> +
> diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
> new file mode 100644
> index 0000000..81d0933
> --- /dev/null
> +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
> @@ -0,0 +1,899 @@
> +/*
> + * Freescale MPC5200 PSC in I2S mode
> + * ALSA SoC Digital Audio Interface (DAI) driver
> + *
> + * Copyright (C) 2008 Secret Lab Technologies Ltd.
> + */
> +
> +#include <linux/init.h>
> +#include <linux/module.h>
> +#include <linux/interrupt.h>
> +#include <linux/device.h>
> +#include <linux/delay.h>
> +#include <linux/of_device.h>
> +#include <linux/of_platform.h>
> +#include <linux/dma-mapping.h>
> +
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/initval.h>
> +#include <sound/soc.h>
> +#include <sound/soc-of.h>
> +
> +#include <sysdev/bestcomm/bestcomm.h>
> +#include <sysdev/bestcomm/gen_bd.h>
> +#include <asm/mpc52xx_psc.h>
> +
> +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
> +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
> +MODULE_LICENSE("GPL");
> +
> +/**
> + * PSC_I2S_RATES: sample rates supported by the I2S
> + *
> + * This driver currently only supports the PSC running in I2S slave mode,
> + * which means the codec determines the sample rate.  Therefore, we tell
> + * ALSA that we support all rates and let the codec driver decide what rates
> + * are really supported.
> + */
> +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
> +			SNDRV_PCM_RATE_CONTINUOUS)
> +
> +/**
> + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
> + */
> +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
> +			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
> +			 SNDRV_PCM_FMTBIT_S32_BE)
> +
> +/**
> + * psc_i2s_stream - Data specific to a single stream (playback or capture)
> + * @active:		flag indicating if the stream is active
> + * @psc_i2s:		pointer back to parent psc_i2s data structure
> + * @bcom_task:		bestcomm task structure
> + * @irq:		irq number for bestcomm task
> + * @period_start:	physical address of start of DMA region
> + * @period_end:		physical address of end of DMA region
> + * @period_next_pt:	physical address of next DMA buffer to enqueue
> + * @period_bytes:	size of DMA period in bytes
> + */
> +struct psc_i2s_stream {
> +	int active;
> +	struct psc_i2s *psc_i2s;
> +	struct bcom_task *bcom_task;
> +	int irq;
> +	struct snd_pcm_substream *stream;
> +	dma_addr_t period_start;
> +	dma_addr_t period_end;
> +	dma_addr_t period_next_pt;
> +	dma_addr_t period_current_pt;
> +	int period_bytes;
> +};
> +
> +/**
> + * psc_i2s - Private driver data
> + * @name: short name for this device ("PSC0", "PSC1", etc)
> + * @psc_regs: pointer to the PSC's registers
> + * @fifo_regs: pointer to the PSC's FIFO registers
> + * @irq: IRQ of this PSC
> + * @dev: struct device pointer
> + * @playback: the number of playback streams opened
> + * @capture: the number of capture streams opened
> + * @dai: the CPU DAI for this device
> + * @playback_stream: Playback stream context data
> + * @capture_stream: Capture stream context data
> + */
> +struct psc_i2s {
> +	char name[32];
> +	struct mpc52xx_psc __iomem *psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fifo_regs;
> +	unsigned int irq;
> +	struct device *dev;
> +	struct snd_soc_cpu_dai dai;
> +	spinlock_t lock;
> +
> +	/* per-stream data */
> +	struct psc_i2s_stream playback_stream;
> +	struct psc_i2s_stream capture_stream;
> +
> +	/* Statistics */
> +	struct {
> +		int overrun_count;
> +		int underrun_count;
> +	} stats;
> +};
> +
> +/*
> + * Interrupt handlers
> + */
> +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
> +{
> +	struct psc_i2s *psc_i2s = _psc_i2s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u16 isr;
> +
> +	isr = in_be16(&regs->mpc52xx_psc_isr);
> +	imr = in_be16(&regs->mpc52xx_psc_imr);
> +
> +	/* Playback underrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
> +		psc_i2s->stats.underrun_count++;
> +
> +	/* Capture overrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_ORERR)
> +		psc_i2s->stats.overrun_count++;
> +
> +	out_8(&regs->command, 4 << 4);	/* reset the error status */
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
> + * @s: pointer to stream private data structure
> + *
> + * Enqueues another audio period buffer into the bestcomm queue.
> + *
> + * Note: The routine must only be called when there is space available in
> + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
> + * will get out of sync
> + */
> +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
> +{
> +	struct bcom_bd *bd;
> +
> +	/* Prepare and enqueue the next buffer descriptor */
> +	bd = bcom_prepare_next_buffer(s->bcom_task);
> +	bd->status = s->period_bytes;
> +	bd->data[0] = s->period_next_pt;
> +	bcom_submit_next_buffer(s->bcom_task, NULL);
> +
> +	/* Update for next period */
> +	s->period_next_pt += s->period_bytes;
> +	if (s->period_next_pt >= s->period_end)
> +		s->period_next_pt = s->period_start;
> +}
> +
> +/* Bestcomm DMA irq handler */
> +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
> +{
> +	struct psc_i2s_stream *s = _psc_i2s_stream;
> +
> +	//spin_lock(&s->psc_i2s->lock);
> +
> +	/* For each finished period, dequeue the completed period buffer
> +	 * and enqueue a new one in it's place. */
> +	while (bcom_buffer_done(s->bcom_task)) {
> +		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +		s->period_current_pt += s->period_bytes;
> +		if (s->period_current_pt >= s->period_end)
> +			s->period_current_pt = s->period_start;
> +		psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +	}
> +
> +	//spin_unlock(&s->psc_i2s->lock);
> +
> +	/* If the stream is active, then also inform the PCM middle layer
> +	 * of the period finished event. */
> +	if (s->active)
> +		snd_pcm_period_elapsed(s->stream);
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_startup: create a new substream
> + *
> + * This is the first function called when a stream is opened.
> + *
> + * If this is the first stream open, then grab the IRQ and program most of
> + * the PSC registers.
> + */
> +static int psc_i2s_startup(struct snd_pcm_substream *substream)
> +{
> +	int playback_irq, capture_irq, rc;
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
> +
> +	/* Disable all interrupts and reset the PSC */
> +	out_be16(&regs->mpc52xx_psc_imr, 0);
> +	out_8(&regs->command, 3 << 4); /* reset transmitter */
> +	out_8(&regs->command, 2 << 4); /* reset receiver */
> +	out_8(&regs->command, 1 << 4); /* reset mode */
> +	out_8(&regs->command, 4 << 4); /* reset error */
> +
> +	/* Default to CODEC8 mode */
> +	out_be32(&regs->sicr,
> +		 MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
> +		 MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
> +
> +	/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
> +	/* Second write to mode: register Normal mode for non loopback */
> +	out_8(&regs->mode, 0);
> +	out_8(&regs->mode, 0);
> +
> +	/* Set the TX and RX fifo alarm thresholds */
> +	out_be16(&fiforegs->rfalarm, 0x100);	/* set RFALARM level */
> +	out_8(&fiforegs->rfcntl, 0x4);		/* set RFGRAN level (bytes) */
> +	out_be16(&fiforegs->tfalarm, 0x100);	/* set TFALARM level */
> +	out_8(&fiforegs->tfcntl, 0x7);		/* set TFGRAN level (bytes*4) */
> +
> +	/* Setup the IRQs */
> +	playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
> +	capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
> +	rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
> +			 "psc-i2s-status", psc_i2s);
> +	rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-capture", &psc_i2s->capture_stream);
> +	rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-playback", &psc_i2s->playback_stream);
> +	if (rc) {
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(capture_irq, &psc_i2s->capture_stream);
> +		free_irq(playback_irq, &psc_i2s->playback_stream);
> +		return -ENODEV;
> +	}
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
> +				 struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	u32 sicr;
> +
> +	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
> +		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
> +		__FUNCTION__, substream, params_period_size(params),
> +		params_period_bytes(params), params_periods(params),
> +		params_buffer_size(params), params_buffer_bytes(params));
> +
> +	sicr = MPC52xx_PSC_SICR_DTS1 |
> +	       MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
> +	switch (params_format(params)) {
> +	 case SNDRV_PCM_FORMAT_S8:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S16_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S24_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S32_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
> +		break;
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid format\n");
> +		return -EINVAL;
> +	}

case should have the same indent level as switch.

> +	out_be32(&psc_i2s->psc_regs->sicr, sicr);
> +
> +	//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
> +	//if (rc) {
> +	//	dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
> +	//	return rc;
> +	//}
> +
> +	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
> +{
> +	//return snd_pcm_lib_free_pages(substream);
> +	snd_pcm_set_runtime_buffer(substream, NULL);
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_trigger: start and stop the DMA transfer.
> + *
> + * This function is called by ALSA to start, stop, pause, and resume the DMA
> + * transfer of data.
> + */
> +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct snd_pcm_runtime *runtime = substream->runtime;
> +	struct psc_i2s_stream *s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u8 psc_cmd;
> +	long flags;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
> +		" stream_id=%i\n",
> +		substream, cmd, substream->pstr->stream);
> +
> +	switch (cmd) {
> +	 case SNDRV_PCM_TRIGGER_START:

ditto re case indent.

> +		s->period_bytes = frames_to_bytes(runtime,
> +						  runtime->period_size);
> +		s->period_start = virt_to_phys(runtime->dma_area);
> +		s->period_end = s->period_start +
> +				(s->period_bytes * runtime->periods);
> +		s->period_next_pt = s->period_start;
> +		s->period_current_pt = s->period_start;
> +		s->active = 1;
> +
> +		/* First; reset everything */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			out_8(&regs->command, MPC52xx_PSC_RST_RX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		} else {
> +			out_8(&regs->command, MPC52xx_PSC_RST_TX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		}
> +
> +		/* Next, fill up the bestcomm bd queue and enable DMA.
> +		 * This will begin filling the PSC's fifo. */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +			bcom_gen_bd_rx_reset(s->bcom_task);
> +		else
> +			bcom_gen_bd_tx_reset(s->bcom_task);
> +		while (!bcom_queue_full(s->bcom_task))
> +			psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +
> +		/* Update interrupt enable settings.  This must be done
> +		 * before the PSC is enabled so that TX underrun events
> +		 * are not missed. */
> +		imr = 0;
> +		if (psc_i2s->playback_stream.active)
> +			imr |= MPC52xx_PSC_IMR_TXEMP;
> +		if (psc_i2s->capture_stream.active)
> +			imr |= MPC52xx_PSC_IMR_ORERR;
> +		out_be16(&regs->isr_imr.imr, imr);
> +
> +		/* Due to errata in the i2s mode; need to line up enabling
> +		 * the transmitter with a transition on the frame sync
> +		 * line */
> +
> +		spin_lock_irqsave(&psc_i2s->lock, flags);
> +		/* first make sure it is low */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
> +		/* then wait for the transition to high */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);

We should be able to exit both while loops if the conditions are not met
within a certain time limit. 

> +		/* Finally, enable the PSC.
> +		 * Receiver must always be enabled; even when we only want
> +		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
> +		psc_cmd = MPC52xx_PSC_RX_ENABLE;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
> +			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
> +		out_8(&regs->command, psc_cmd);
> +		spin_unlock_irqrestore(&psc_i2s->lock, flags);
> +
> +		break;
> +
> +	 case SNDRV_PCM_TRIGGER_STOP:
> +		/* Turn off the PSC */
> +		s->active = 0;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			if (!psc_i2s->playback_stream.active) {
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +				out_8(&regs->command, 3 << 4);	/* reset tx */
> +				out_8(&regs->command, 4 << 4);	/* reset err */
> +			}
> +		} else {
> +			out_8(&regs->command, 3 << 4);	/* reset tx */
> +			out_8(&regs->command, 4 << 4);	/* reset err */
> +			if (!psc_i2s->capture_stream.active)
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +		}
> +
> +		bcom_disable(s->bcom_task);
> +		while (!bcom_queue_empty(s->bcom_task))
> +			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +
> +		break;
> +
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid command\n");
> +		return -EINVAL;
> +	}
> +
> +	/* Update interrupt enable settings */
> +	imr = 0;
> +	if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
> +	if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
> +	out_be16(&regs->isr_imr.imr, imr);
> +
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_shutdown: shutdown the data transfer on a stream
> + *
> + * Shutdown the PSC if there are no other substreams open.
> + */
> +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
> +
> +	/*
> +	 * If this is the last active substream, disable the PSC and release
> +	 * the IRQ.
> +	 */
> +	if (!psc_i2s->playback_stream.active &&
> +	    !psc_i2s->capture_stream.active) {
> +		/* TODO: shut off channels */
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
> +			 &psc_i2s->capture_stream);
> +		free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
> +			 &psc_i2s->playback_stream);
> +	}
> +}
> +
> +/**
> + * psc_i2s_set_sysclk: set the clock frequency and direction
> + *
> + * This function is called by the machine driver to tell us what the clock
> + * frequency and direction are.
> + *
> + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
> + * and we don't care about the frequency.  Return an error if the direction
> + * is not SND_SOC_CLOCK_IN.
> + *
> + * @clk_id: reserved, should be zero
> + * @freq: the frequency of the given clock ID, currently ignored
> + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
> + */
> +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
> +			      int clk_id, unsigned int freq, int dir)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
> +				cpu_dai, dir);
> +	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
> +}
> +
> +/**
> + * psc_i2s_set_fmt: set the serial format.
> + *
> + * This function is called by the machine driver to tell us what serial
> + * format to use.
> + *
> + * This driver only supports I2S mode.  Return an error if the format is
> + * not SND_SOC_DAIFMT_I2S.
> + *
> + * @format: one of SND_SOC_DAIFMT_xxx
> + */
> +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
> +				cpu_dai, format);
> +	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * ALSA SoC Bindings
> + *
> + * - Digital Audio Interface (DAI) template
> + * - create/destroy dai hooks
> + */
> +
> +/**
> + * psc_i2s_dai_template: template CPU Digital Audio Interface
> + */
> +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
> +	.type = SND_SOC_DAI_I2S,
> +	.playback = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.capture = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.ops = {
> +		.startup = psc_i2s_startup,
> +		.hw_params = psc_i2s_hw_params,
> +		.hw_free = psc_i2s_hw_free,
> +		.shutdown = psc_i2s_shutdown,
> +		.trigger = psc_i2s_trigger,
> +	},
> +	.dai_ops = {
> +		.set_sysclk = psc_i2s_set_sysclk,
> +		.set_fmt = psc_i2s_set_fmt,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * The PSC I2S 'ASoC platform' driver
> + *
> + * Can be referenced by an 'ASoC machine' driver
> + * This driver only deals with the audio bus; it doesn't have any
> + * interaction with the attached codec
> + */
> +
> +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
> +	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
> +		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
> +	.formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
> +		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
> +	.rate_min = 8000,
> +	.rate_max = 48000,
> +	.channels_min = 2,
> +	.channels_max = 2,
> +	.period_bytes_max	= 1024 * 1024,
> +	.period_bytes_min	= 32,
> +	.period_bytes_max	= 1024 * 1024,
> +	.periods_min		= 2,
> +	.periods_max		= 256,
> +	.buffer_bytes_max	= 2 * 1024 * 1024,
> +	.fifo_size		= 0,
> +};
> +

Fwiw, I usually separate out the DMA from I2S so it can be used by AC97,
PCM interfaces etc. If your hardware only has I2S the it's not required.

> +static unsigned int psc_i2s_fixed_rates[] = {
> +	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
> +};
> +
> +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
> +	.count = ARRAY_SIZE(psc_i2s_fixed_rates),
> +	.list = psc_i2s_fixed_rates,
> +	.mask = 0,
> +};
> +
> +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	int rc;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
> +
> +	rc = snd_pcm_hw_constraint_integer(substream->runtime,
> +					   SNDRV_PCM_HW_PARAM_PERIODS);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid buffer size\n");
> +		return rc;
> +	}
> +	rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
> +					SNDRV_PCM_HW_PARAM_RATE,
> +					&psc_i2s_constraints_rates);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid rate\n");
> +		return rc;
> +	}
> +
> +	s->stream = substream;
> +	return 0;
> +}
> +
> +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	s->stream = NULL;
> +	return 0;
> +}
> +
> +static snd_pcm_uframes_t
> +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	dma_addr_t count;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
> +	count = s->period_current_pt - s->period_start;
> +
> +	return bytes_to_frames(substream->runtime, count);
> +}
> +
> +static struct snd_pcm_ops psc_i2s_pcm_ops = {
> +	.open		= psc_i2s_pcm_open,
> +	.close		= psc_i2s_pcm_close,
> +	.ioctl		= snd_pcm_lib_ioctl,
> +	.pointer	= psc_i2s_pcm_pointer,
> +};
> +
> +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
> +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
> +			   struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
> +	int rc = 0;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
> +		card, dai, pcm);
> +
> +	if (!card->dev->dma_mask)
> +		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
> +	if (!card->dev->coherent_dma_mask)
> +		card->dev->coherent_dma_mask = 0xffffffff;
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[0].substream->dma_buffer);
> +	if (rc) {
> +		dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[1].substream->dma_buffer);
> +	if (rc) {
> +		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
> +		dev_err(card->dev, "Can't allocate capture DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	return 0;
> +}
> +
> +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	struct snd_pcm_substream *substream;
> +	int stream;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
> +
> +	for (stream = 0; stream < 2; stream++) {
> +		substream = pcm->streams[stream].substream;
> +		if (substream) {
> +			snd_dma_free_pages(&substream->dma_buffer);
> +			substream->dma_buffer.area = NULL;
> +			substream->dma_buffer.addr = 0;
> +		}
> +	}
> +}
> +
> +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
> +	.name		= "mpc5200-psc-audio",
> +	.pcm_ops	= &psc_i2s_pcm_ops,
> +	.pcm_new	= &psc_i2s_pcm_new,
> +	.pcm_free	= &psc_i2s_pcm_free,
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Sysfs attributes for debugging
> + */
> +
> +static ssize_t psc_i2s_status_show(struct device *dev,
> +			   struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +
> +	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
> +			in_be16(&psc_i2s->psc_regs->sr_csr.status),
> +			in_be32(&psc_i2s->psc_regs->sicr),
> +			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->rfstat),
> +			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->tfstat));
> +}
> +
> +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
> +				   const char *name)
> +{
> +	if (strcmp(name, "playback_underrun") == 0)
> +		return &psc_i2s->stats.underrun_count;
> +	if (strcmp(name, "capture_overrun") == 0)
> +		return &psc_i2s->stats.overrun_count;
> +
> +	return NULL;
> +}
> +
> +static ssize_t psc_i2s_stat_show(struct device *dev,
> +				 struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	return sprintf(buf, "%i\n", *attrib);
> +}
> +
> +static ssize_t psc_i2s_stat_store(struct device *dev,
> +				  struct device_attribute *attr,
> +				  const char *buf,
> +				  size_t count)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	*attrib = simple_strtoul(buf, NULL, 0);
> +	return count;
> +}
> +
> +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
> +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
> +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
> +
> +/* ---------------------------------------------------------------------
> + * OF platform bus binding code:
> + * - Probe/remove operations
> + * - OF device match table
> + */
> +static int __devinit psc_i2s_of_probe(struct of_device *op,
> +				      const struct of_device_id *match)
> +{
> +	phys_addr_t fifo;
> +	struct psc_i2s *psc_i2s;
> +	struct resource res;
> +	int size, psc_id, irq, rc;
> +	const __be32 *prop;
> +	void __iomem *regs;
> +
> +	dev_dbg(&op->dev, "probing psc i2s device\n");
> +
> +	/* Get the PSC ID */
> +	prop = of_get_property(op->node, "cell-index", &size);
> +	if (!prop || size < sizeof *prop)
> +		return -ENODEV;
> +	psc_id = be32_to_cpu(*prop);
> +
> +	/* Fetch the registers and IRQ of the PSC */
> +	irq = irq_of_parse_and_map(op->node, 0);
> +	if (of_address_to_resource(op->node, 0, &res)) {
> +		dev_err(&op->dev, "Missing reg property\n");
> +		return -ENODEV;
> +	}
> +	regs = ioremap(res.start, 1 + res.end - res.start);
> +	if (!regs) {
> +		dev_err(&op->dev, "Could not map registers\n");
> +		return -ENODEV;
> +	}
> +
> +	/* Allocate and initialize the driver private data */
> +	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
> +	if (!psc_i2s) {
> +		iounmap(regs);
> +		return -ENOMEM;
> +	}
> +	spin_lock_init(&psc_i2s->lock);
> +	psc_i2s->irq = irq;
> +	psc_i2s->psc_regs = regs;
> +	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
> +	psc_i2s->dev = &op->dev;
> +	psc_i2s->playback_stream.psc_i2s = psc_i2s;
> +	psc_i2s->capture_stream.psc_i2s = psc_i2s;
> +	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
> +
> +	/* Fill out the CPU DAI structure */
> +	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
> +	psc_i2s->dai.private_data = psc_i2s;
> +	psc_i2s->dai.name = psc_i2s->name;
> +	psc_i2s->dai.id = psc_id;
> +
> +	/* Find the address of the fifo data registers and setup the
> +	 * DMA tasks */
> +	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
> +	psc_i2s->capture_stream.bcom_task =
> +		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
> +	psc_i2s->playback_stream.bcom_task =
> +		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
> +	if (!psc_i2s->capture_stream.bcom_task ||
> +	    !psc_i2s->playback_stream.bcom_task) {
> +		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
> +		iounmap(regs);
> +		kfree(psc_i2s);
> +		return -ENODEV;
> +	}
> +
> +	/* Save what we've done so it can be found again later */
> +	dev_set_drvdata(&op->dev, psc_i2s);
> +
> +	/* Register the SYSFS files */
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
> +	if (rc)
> +		dev_info(psc_i2s->dev, "error creating sysfs files\n");
> +
> +	/* Tell the ASoC OF helpers about it */
> +	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
> +				     &psc_i2s->dai);
> +
> +	return 0;
> +}
> +
> +static int __devexit psc_i2s_of_remove(struct of_device *op)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
> +
> +	dev_dbg(&op->dev, "psc_i2s_remove()\n");
> +
> +	bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
> +	bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
> +
> +	iounmap(psc_i2s->psc_regs);
> +	iounmap(psc_i2s->fifo_regs);
> +	kfree(psc_i2s);
> +	dev_set_drvdata(&op->dev, NULL);
> +
> +	return 0;
> +}
> +
> +/* Match table for of_platform binding */
> +static struct of_device_id psc_i2s_match[] __devinitdata = {
> +	{ .compatible = "fsl,mpc5200-psc-i2s", },
> +	{}
> +};
> +MODULE_DEVICE_TABLE(of, psc_i2s_match);
> +
> +static struct of_platform_driver psc_i2s_driver = {
> +	.match_table = psc_i2s_match,
> +	.probe = psc_i2s_of_probe,
> +	.remove = __devexit_p(psc_i2s_of_remove),
> +	.driver = {
> +		.name = "mpc5200-psc-i2s",
> +		.owner = THIS_MODULE,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Module setup and teardown; simply register the of_platform driver
> + * for the PSC in I2S mode.
> + */
> +static int __init psc_i2s_init(void)
> +{
> +	return of_register_platform_driver(&psc_i2s_driver);
> +}
> +module_init(psc_i2s_init);
> +
> +static void __exit psc_i2s_exit(void)
> +{
> +	of_unregister_platform_driver(&psc_i2s_driver);
> +}
> +module_exit(psc_i2s_exit);
> +
> +
> 
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> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel


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Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Liam Girdwood]

On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
> 
> This is an I2S bus driver for the MPC5200 PSC device.  It is probably
> will not be merged as-is because it uses v1 of the ASoC API, but I want
> to get it out there for comments.

Looks good, just minor comments.

> ---
> 
>  sound/soc/fsl/Kconfig           |    6 
>  sound/soc/fsl/Makefile          |    2 
>  sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>  3 files changed, 907 insertions(+), 0 deletions(-)
> 
> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
> index 257101f..5daa8d3 100644
> --- a/sound/soc/fsl/Kconfig
> +++ b/sound/soc/fsl/Kconfig
> @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>  	help
>  	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>  
> +config SND_SOC_MPC5200_I2S
> +	bool "Freescale MPC5200 PSC in I2S mode driver"

Is this built-in only ?

> +	depends on SND_SOC && PPC_MPC52xx
> +	help
> +	  Say Y here to support the MPC5200 PSCs in I2S mode.
> +
>  endmenu
> diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
> index 62f680a..98729a1 100644
> --- a/sound/soc/fsl/Makefile
> +++ b/sound/soc/fsl/Makefile
> @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>  # MPC8610 Platform Support
>  obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>  
> +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
> +
> diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
> new file mode 100644
> index 0000000..81d0933
> --- /dev/null
> +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
> @@ -0,0 +1,899 @@
> +/*
> + * Freescale MPC5200 PSC in I2S mode
> + * ALSA SoC Digital Audio Interface (DAI) driver
> + *
> + * Copyright (C) 2008 Secret Lab Technologies Ltd.
> + */
> +
> +#include <linux/init.h>
> +#include <linux/module.h>
> +#include <linux/interrupt.h>
> +#include <linux/device.h>
> +#include <linux/delay.h>
> +#include <linux/of_device.h>
> +#include <linux/of_platform.h>
> +#include <linux/dma-mapping.h>
> +
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/initval.h>
> +#include <sound/soc.h>
> +#include <sound/soc-of.h>
> +
> +#include <sysdev/bestcomm/bestcomm.h>
> +#include <sysdev/bestcomm/gen_bd.h>
> +#include <asm/mpc52xx_psc.h>
> +
> +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
> +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
> +MODULE_LICENSE("GPL");
> +
> +/**
> + * PSC_I2S_RATES: sample rates supported by the I2S
> + *
> + * This driver currently only supports the PSC running in I2S slave mode,
> + * which means the codec determines the sample rate.  Therefore, we tell
> + * ALSA that we support all rates and let the codec driver decide what rates
> + * are really supported.
> + */
> +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
> +			SNDRV_PCM_RATE_CONTINUOUS)
> +
> +/**
> + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
> + */
> +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
> +			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
> +			 SNDRV_PCM_FMTBIT_S32_BE)
> +
> +/**
> + * psc_i2s_stream - Data specific to a single stream (playback or capture)
> + * @active:		flag indicating if the stream is active
> + * @psc_i2s:		pointer back to parent psc_i2s data structure
> + * @bcom_task:		bestcomm task structure
> + * @irq:		irq number for bestcomm task
> + * @period_start:	physical address of start of DMA region
> + * @period_end:		physical address of end of DMA region
> + * @period_next_pt:	physical address of next DMA buffer to enqueue
> + * @period_bytes:	size of DMA period in bytes
> + */
> +struct psc_i2s_stream {
> +	int active;
> +	struct psc_i2s *psc_i2s;
> +	struct bcom_task *bcom_task;
> +	int irq;
> +	struct snd_pcm_substream *stream;
> +	dma_addr_t period_start;
> +	dma_addr_t period_end;
> +	dma_addr_t period_next_pt;
> +	dma_addr_t period_current_pt;
> +	int period_bytes;
> +};
> +
> +/**
> + * psc_i2s - Private driver data
> + * @name: short name for this device ("PSC0", "PSC1", etc)
> + * @psc_regs: pointer to the PSC's registers
> + * @fifo_regs: pointer to the PSC's FIFO registers
> + * @irq: IRQ of this PSC
> + * @dev: struct device pointer
> + * @playback: the number of playback streams opened
> + * @capture: the number of capture streams opened
> + * @dai: the CPU DAI for this device
> + * @playback_stream: Playback stream context data
> + * @capture_stream: Capture stream context data
> + */
> +struct psc_i2s {
> +	char name[32];
> +	struct mpc52xx_psc __iomem *psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fifo_regs;
> +	unsigned int irq;
> +	struct device *dev;
> +	struct snd_soc_cpu_dai dai;
> +	spinlock_t lock;
> +
> +	/* per-stream data */
> +	struct psc_i2s_stream playback_stream;
> +	struct psc_i2s_stream capture_stream;
> +
> +	/* Statistics */
> +	struct {
> +		int overrun_count;
> +		int underrun_count;
> +	} stats;
> +};
> +
> +/*
> + * Interrupt handlers
> + */
> +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
> +{
> +	struct psc_i2s *psc_i2s = _psc_i2s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u16 isr;
> +
> +	isr = in_be16(&regs->mpc52xx_psc_isr);
> +	imr = in_be16(&regs->mpc52xx_psc_imr);
> +
> +	/* Playback underrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
> +		psc_i2s->stats.underrun_count++;
> +
> +	/* Capture overrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_ORERR)
> +		psc_i2s->stats.overrun_count++;
> +
> +	out_8(&regs->command, 4 << 4);	/* reset the error status */
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
> + * @s: pointer to stream private data structure
> + *
> + * Enqueues another audio period buffer into the bestcomm queue.
> + *
> + * Note: The routine must only be called when there is space available in
> + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
> + * will get out of sync
> + */
> +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
> +{
> +	struct bcom_bd *bd;
> +
> +	/* Prepare and enqueue the next buffer descriptor */
> +	bd = bcom_prepare_next_buffer(s->bcom_task);
> +	bd->status = s->period_bytes;
> +	bd->data[0] = s->period_next_pt;
> +	bcom_submit_next_buffer(s->bcom_task, NULL);
> +
> +	/* Update for next period */
> +	s->period_next_pt += s->period_bytes;
> +	if (s->period_next_pt >= s->period_end)
> +		s->period_next_pt = s->period_start;
> +}
> +
> +/* Bestcomm DMA irq handler */
> +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
> +{
> +	struct psc_i2s_stream *s = _psc_i2s_stream;
> +
> +	//spin_lock(&s->psc_i2s->lock);
> +
> +	/* For each finished period, dequeue the completed period buffer
> +	 * and enqueue a new one in it's place. */
> +	while (bcom_buffer_done(s->bcom_task)) {
> +		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +		s->period_current_pt += s->period_bytes;
> +		if (s->period_current_pt >= s->period_end)
> +			s->period_current_pt = s->period_start;
> +		psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +	}
> +
> +	//spin_unlock(&s->psc_i2s->lock);
> +
> +	/* If the stream is active, then also inform the PCM middle layer
> +	 * of the period finished event. */
> +	if (s->active)
> +		snd_pcm_period_elapsed(s->stream);
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_startup: create a new substream
> + *
> + * This is the first function called when a stream is opened.
> + *
> + * If this is the first stream open, then grab the IRQ and program most of
> + * the PSC registers.
> + */
> +static int psc_i2s_startup(struct snd_pcm_substream *substream)
> +{
> +	int playback_irq, capture_irq, rc;
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
> +
> +	/* Disable all interrupts and reset the PSC */
> +	out_be16(&regs->mpc52xx_psc_imr, 0);
> +	out_8(&regs->command, 3 << 4); /* reset transmitter */
> +	out_8(&regs->command, 2 << 4); /* reset receiver */
> +	out_8(&regs->command, 1 << 4); /* reset mode */
> +	out_8(&regs->command, 4 << 4); /* reset error */
> +
> +	/* Default to CODEC8 mode */
> +	out_be32(&regs->sicr,
> +		 MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
> +		 MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
> +
> +	/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
> +	/* Second write to mode: register Normal mode for non loopback */
> +	out_8(&regs->mode, 0);
> +	out_8(&regs->mode, 0);
> +
> +	/* Set the TX and RX fifo alarm thresholds */
> +	out_be16(&fiforegs->rfalarm, 0x100);	/* set RFALARM level */
> +	out_8(&fiforegs->rfcntl, 0x4);		/* set RFGRAN level (bytes) */
> +	out_be16(&fiforegs->tfalarm, 0x100);	/* set TFALARM level */
> +	out_8(&fiforegs->tfcntl, 0x7);		/* set TFGRAN level (bytes*4) */
> +
> +	/* Setup the IRQs */
> +	playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
> +	capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
> +	rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
> +			 "psc-i2s-status", psc_i2s);
> +	rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-capture", &psc_i2s->capture_stream);
> +	rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-playback", &psc_i2s->playback_stream);
> +	if (rc) {
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(capture_irq, &psc_i2s->capture_stream);
> +		free_irq(playback_irq, &psc_i2s->playback_stream);
> +		return -ENODEV;
> +	}
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
> +				 struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	u32 sicr;
> +
> +	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
> +		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
> +		__FUNCTION__, substream, params_period_size(params),
> +		params_period_bytes(params), params_periods(params),
> +		params_buffer_size(params), params_buffer_bytes(params));
> +
> +	sicr = MPC52xx_PSC_SICR_DTS1 |
> +	       MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
> +	switch (params_format(params)) {
> +	 case SNDRV_PCM_FORMAT_S8:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S16_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S24_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S32_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
> +		break;
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid format\n");
> +		return -EINVAL;
> +	}

case should have the same indent level as switch.

> +	out_be32(&psc_i2s->psc_regs->sicr, sicr);
> +
> +	//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
> +	//if (rc) {
> +	//	dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
> +	//	return rc;
> +	//}
> +
> +	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
> +{
> +	//return snd_pcm_lib_free_pages(substream);
> +	snd_pcm_set_runtime_buffer(substream, NULL);
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_trigger: start and stop the DMA transfer.
> + *
> + * This function is called by ALSA to start, stop, pause, and resume the DMA
> + * transfer of data.
> + */
> +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct snd_pcm_runtime *runtime = substream->runtime;
> +	struct psc_i2s_stream *s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u8 psc_cmd;
> +	long flags;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
> +		" stream_id=%i\n",
> +		substream, cmd, substream->pstr->stream);
> +
> +	switch (cmd) {
> +	 case SNDRV_PCM_TRIGGER_START:

ditto re case indent.

> +		s->period_bytes = frames_to_bytes(runtime,
> +						  runtime->period_size);
> +		s->period_start = virt_to_phys(runtime->dma_area);
> +		s->period_end = s->period_start +
> +				(s->period_bytes * runtime->periods);
> +		s->period_next_pt = s->period_start;
> +		s->period_current_pt = s->period_start;
> +		s->active = 1;
> +
> +		/* First; reset everything */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			out_8(&regs->command, MPC52xx_PSC_RST_RX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		} else {
> +			out_8(&regs->command, MPC52xx_PSC_RST_TX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		}
> +
> +		/* Next, fill up the bestcomm bd queue and enable DMA.
> +		 * This will begin filling the PSC's fifo. */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +			bcom_gen_bd_rx_reset(s->bcom_task);
> +		else
> +			bcom_gen_bd_tx_reset(s->bcom_task);
> +		while (!bcom_queue_full(s->bcom_task))
> +			psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +
> +		/* Update interrupt enable settings.  This must be done
> +		 * before the PSC is enabled so that TX underrun events
> +		 * are not missed. */
> +		imr = 0;
> +		if (psc_i2s->playback_stream.active)
> +			imr |= MPC52xx_PSC_IMR_TXEMP;
> +		if (psc_i2s->capture_stream.active)
> +			imr |= MPC52xx_PSC_IMR_ORERR;
> +		out_be16(&regs->isr_imr.imr, imr);
> +
> +		/* Due to errata in the i2s mode; need to line up enabling
> +		 * the transmitter with a transition on the frame sync
> +		 * line */
> +
> +		spin_lock_irqsave(&psc_i2s->lock, flags);
> +		/* first make sure it is low */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
> +		/* then wait for the transition to high */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);

We should be able to exit both while loops if the conditions are not met
within a certain time limit. 

> +		/* Finally, enable the PSC.
> +		 * Receiver must always be enabled; even when we only want
> +		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
> +		psc_cmd = MPC52xx_PSC_RX_ENABLE;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
> +			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
> +		out_8(&regs->command, psc_cmd);
> +		spin_unlock_irqrestore(&psc_i2s->lock, flags);
> +
> +		break;
> +
> +	 case SNDRV_PCM_TRIGGER_STOP:
> +		/* Turn off the PSC */
> +		s->active = 0;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			if (!psc_i2s->playback_stream.active) {
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +				out_8(&regs->command, 3 << 4);	/* reset tx */
> +				out_8(&regs->command, 4 << 4);	/* reset err */
> +			}
> +		} else {
> +			out_8(&regs->command, 3 << 4);	/* reset tx */
> +			out_8(&regs->command, 4 << 4);	/* reset err */
> +			if (!psc_i2s->capture_stream.active)
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +		}
> +
> +		bcom_disable(s->bcom_task);
> +		while (!bcom_queue_empty(s->bcom_task))
> +			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +
> +		break;
> +
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid command\n");
> +		return -EINVAL;
> +	}
> +
> +	/* Update interrupt enable settings */
> +	imr = 0;
> +	if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
> +	if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
> +	out_be16(&regs->isr_imr.imr, imr);
> +
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_shutdown: shutdown the data transfer on a stream
> + *
> + * Shutdown the PSC if there are no other substreams open.
> + */
> +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
> +
> +	/*
> +	 * If this is the last active substream, disable the PSC and release
> +	 * the IRQ.
> +	 */
> +	if (!psc_i2s->playback_stream.active &&
> +	    !psc_i2s->capture_stream.active) {
> +		/* TODO: shut off channels */
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
> +			 &psc_i2s->capture_stream);
> +		free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
> +			 &psc_i2s->playback_stream);
> +	}
> +}
> +
> +/**
> + * psc_i2s_set_sysclk: set the clock frequency and direction
> + *
> + * This function is called by the machine driver to tell us what the clock
> + * frequency and direction are.
> + *
> + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
> + * and we don't care about the frequency.  Return an error if the direction
> + * is not SND_SOC_CLOCK_IN.
> + *
> + * @clk_id: reserved, should be zero
> + * @freq: the frequency of the given clock ID, currently ignored
> + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
> + */
> +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
> +			      int clk_id, unsigned int freq, int dir)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
> +				cpu_dai, dir);
> +	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
> +}
> +
> +/**
> + * psc_i2s_set_fmt: set the serial format.
> + *
> + * This function is called by the machine driver to tell us what serial
> + * format to use.
> + *
> + * This driver only supports I2S mode.  Return an error if the format is
> + * not SND_SOC_DAIFMT_I2S.
> + *
> + * @format: one of SND_SOC_DAIFMT_xxx
> + */
> +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
> +				cpu_dai, format);
> +	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * ALSA SoC Bindings
> + *
> + * - Digital Audio Interface (DAI) template
> + * - create/destroy dai hooks
> + */
> +
> +/**
> + * psc_i2s_dai_template: template CPU Digital Audio Interface
> + */
> +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
> +	.type = SND_SOC_DAI_I2S,
> +	.playback = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.capture = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.ops = {
> +		.startup = psc_i2s_startup,
> +		.hw_params = psc_i2s_hw_params,
> +		.hw_free = psc_i2s_hw_free,
> +		.shutdown = psc_i2s_shutdown,
> +		.trigger = psc_i2s_trigger,
> +	},
> +	.dai_ops = {
> +		.set_sysclk = psc_i2s_set_sysclk,
> +		.set_fmt = psc_i2s_set_fmt,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * The PSC I2S 'ASoC platform' driver
> + *
> + * Can be referenced by an 'ASoC machine' driver
> + * This driver only deals with the audio bus; it doesn't have any
> + * interaction with the attached codec
> + */
> +
> +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
> +	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
> +		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
> +	.formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
> +		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
> +	.rate_min = 8000,
> +	.rate_max = 48000,
> +	.channels_min = 2,
> +	.channels_max = 2,
> +	.period_bytes_max	= 1024 * 1024,
> +	.period_bytes_min	= 32,
> +	.period_bytes_max	= 1024 * 1024,
> +	.periods_min		= 2,
> +	.periods_max		= 256,
> +	.buffer_bytes_max	= 2 * 1024 * 1024,
> +	.fifo_size		= 0,
> +};
> +

Fwiw, I usually separate out the DMA from I2S so it can be used by AC97,
PCM interfaces etc. If your hardware only has I2S the it's not required.

> +static unsigned int psc_i2s_fixed_rates[] = {
> +	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
> +};
> +
> +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
> +	.count = ARRAY_SIZE(psc_i2s_fixed_rates),
> +	.list = psc_i2s_fixed_rates,
> +	.mask = 0,
> +};
> +
> +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	int rc;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
> +
> +	rc = snd_pcm_hw_constraint_integer(substream->runtime,
> +					   SNDRV_PCM_HW_PARAM_PERIODS);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid buffer size\n");
> +		return rc;
> +	}
> +	rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
> +					SNDRV_PCM_HW_PARAM_RATE,
> +					&psc_i2s_constraints_rates);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid rate\n");
> +		return rc;
> +	}
> +
> +	s->stream = substream;
> +	return 0;
> +}
> +
> +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	s->stream = NULL;
> +	return 0;
> +}
> +
> +static snd_pcm_uframes_t
> +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	dma_addr_t count;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
> +	count = s->period_current_pt - s->period_start;
> +
> +	return bytes_to_frames(substream->runtime, count);
> +}
> +
> +static struct snd_pcm_ops psc_i2s_pcm_ops = {
> +	.open		= psc_i2s_pcm_open,
> +	.close		= psc_i2s_pcm_close,
> +	.ioctl		= snd_pcm_lib_ioctl,
> +	.pointer	= psc_i2s_pcm_pointer,
> +};
> +
> +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
> +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
> +			   struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
> +	int rc = 0;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
> +		card, dai, pcm);
> +
> +	if (!card->dev->dma_mask)
> +		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
> +	if (!card->dev->coherent_dma_mask)
> +		card->dev->coherent_dma_mask = 0xffffffff;
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[0].substream->dma_buffer);
> +	if (rc) {
> +		dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[1].substream->dma_buffer);
> +	if (rc) {
> +		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
> +		dev_err(card->dev, "Can't allocate capture DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	return 0;
> +}
> +
> +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	struct snd_pcm_substream *substream;
> +	int stream;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
> +
> +	for (stream = 0; stream < 2; stream++) {
> +		substream = pcm->streams[stream].substream;
> +		if (substream) {
> +			snd_dma_free_pages(&substream->dma_buffer);
> +			substream->dma_buffer.area = NULL;
> +			substream->dma_buffer.addr = 0;
> +		}
> +	}
> +}
> +
> +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
> +	.name		= "mpc5200-psc-audio",
> +	.pcm_ops	= &psc_i2s_pcm_ops,
> +	.pcm_new	= &psc_i2s_pcm_new,
> +	.pcm_free	= &psc_i2s_pcm_free,
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Sysfs attributes for debugging
> + */
> +
> +static ssize_t psc_i2s_status_show(struct device *dev,
> +			   struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +
> +	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
> +			in_be16(&psc_i2s->psc_regs->sr_csr.status),
> +			in_be32(&psc_i2s->psc_regs->sicr),
> +			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->rfstat),
> +			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->tfstat));
> +}
> +
> +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
> +				   const char *name)
> +{
> +	if (strcmp(name, "playback_underrun") == 0)
> +		return &psc_i2s->stats.underrun_count;
> +	if (strcmp(name, "capture_overrun") == 0)
> +		return &psc_i2s->stats.overrun_count;
> +
> +	return NULL;
> +}
> +
> +static ssize_t psc_i2s_stat_show(struct device *dev,
> +				 struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	return sprintf(buf, "%i\n", *attrib);
> +}
> +
> +static ssize_t psc_i2s_stat_store(struct device *dev,
> +				  struct device_attribute *attr,
> +				  const char *buf,
> +				  size_t count)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	*attrib = simple_strtoul(buf, NULL, 0);
> +	return count;
> +}
> +
> +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
> +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
> +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
> +
> +/* ---------------------------------------------------------------------
> + * OF platform bus binding code:
> + * - Probe/remove operations
> + * - OF device match table
> + */
> +static int __devinit psc_i2s_of_probe(struct of_device *op,
> +				      const struct of_device_id *match)
> +{
> +	phys_addr_t fifo;
> +	struct psc_i2s *psc_i2s;
> +	struct resource res;
> +	int size, psc_id, irq, rc;
> +	const __be32 *prop;
> +	void __iomem *regs;
> +
> +	dev_dbg(&op->dev, "probing psc i2s device\n");
> +
> +	/* Get the PSC ID */
> +	prop = of_get_property(op->node, "cell-index", &size);
> +	if (!prop || size < sizeof *prop)
> +		return -ENODEV;
> +	psc_id = be32_to_cpu(*prop);
> +
> +	/* Fetch the registers and IRQ of the PSC */
> +	irq = irq_of_parse_and_map(op->node, 0);
> +	if (of_address_to_resource(op->node, 0, &res)) {
> +		dev_err(&op->dev, "Missing reg property\n");
> +		return -ENODEV;
> +	}
> +	regs = ioremap(res.start, 1 + res.end - res.start);
> +	if (!regs) {
> +		dev_err(&op->dev, "Could not map registers\n");
> +		return -ENODEV;
> +	}
> +
> +	/* Allocate and initialize the driver private data */
> +	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
> +	if (!psc_i2s) {
> +		iounmap(regs);
> +		return -ENOMEM;
> +	}
> +	spin_lock_init(&psc_i2s->lock);
> +	psc_i2s->irq = irq;
> +	psc_i2s->psc_regs = regs;
> +	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
> +	psc_i2s->dev = &op->dev;
> +	psc_i2s->playback_stream.psc_i2s = psc_i2s;
> +	psc_i2s->capture_stream.psc_i2s = psc_i2s;
> +	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
> +
> +	/* Fill out the CPU DAI structure */
> +	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
> +	psc_i2s->dai.private_data = psc_i2s;
> +	psc_i2s->dai.name = psc_i2s->name;
> +	psc_i2s->dai.id = psc_id;
> +
> +	/* Find the address of the fifo data registers and setup the
> +	 * DMA tasks */
> +	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
> +	psc_i2s->capture_stream.bcom_task =
> +		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
> +	psc_i2s->playback_stream.bcom_task =
> +		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
> +	if (!psc_i2s->capture_stream.bcom_task ||
> +	    !psc_i2s->playback_stream.bcom_task) {
> +		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
> +		iounmap(regs);
> +		kfree(psc_i2s);
> +		return -ENODEV;
> +	}
> +
> +	/* Save what we've done so it can be found again later */
> +	dev_set_drvdata(&op->dev, psc_i2s);
> +
> +	/* Register the SYSFS files */
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
> +	if (rc)
> +		dev_info(psc_i2s->dev, "error creating sysfs files\n");
> +
> +	/* Tell the ASoC OF helpers about it */
> +	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
> +				     &psc_i2s->dai);
> +
> +	return 0;
> +}
> +
> +static int __devexit psc_i2s_of_remove(struct of_device *op)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
> +
> +	dev_dbg(&op->dev, "psc_i2s_remove()\n");
> +
> +	bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
> +	bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
> +
> +	iounmap(psc_i2s->psc_regs);
> +	iounmap(psc_i2s->fifo_regs);
> +	kfree(psc_i2s);
> +	dev_set_drvdata(&op->dev, NULL);
> +
> +	return 0;
> +}
> +
> +/* Match table for of_platform binding */
> +static struct of_device_id psc_i2s_match[] __devinitdata = {
> +	{ .compatible = "fsl,mpc5200-psc-i2s", },
> +	{}
> +};
> +MODULE_DEVICE_TABLE(of, psc_i2s_match);
> +
> +static struct of_platform_driver psc_i2s_driver = {
> +	.match_table = psc_i2s_match,
> +	.probe = psc_i2s_of_probe,
> +	.remove = __devexit_p(psc_i2s_of_remove),
> +	.driver = {
> +		.name = "mpc5200-psc-i2s",
> +		.owner = THIS_MODULE,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Module setup and teardown; simply register the of_platform driver
> + * for the PSC in I2S mode.
> + */
> +static int __init psc_i2s_init(void)
> +{
> +	return of_register_platform_driver(&psc_i2s_driver);
> +}
> +module_init(psc_i2s_init);
> +
> +static void __exit psc_i2s_exit(void)
> +{
> +	of_unregister_platform_driver(&psc_i2s_driver);
> +}
> +module_exit(psc_i2s_exit);
> +
> +
> 
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Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to
share DMA code. The new Phytec pcm030 baseboard is AC97 too.

What does the device tree look like?


On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---
>
>   sound/soc/fsl/Kconfig           |    6
>   sound/soc/fsl/Makefile          |    2
>   sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>   3 files changed, 907 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
>  index 257101f..5daa8d3 100644
>  --- a/sound/soc/fsl/Kconfig
>  +++ b/sound/soc/fsl/Kconfig
>  @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>         help
>           Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>
>  +config SND_SOC_MPC5200_I2S
>  +       bool "Freescale MPC5200 PSC in I2S mode driver"
>  +       depends on SND_SOC && PPC_MPC52xx
>  +       help
>  +         Say Y here to support the MPC5200 PSCs in I2S mode.
>  +
>   endmenu
>  diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
>  index 62f680a..98729a1 100644
>  --- a/sound/soc/fsl/Makefile
>  +++ b/sound/soc/fsl/Makefile
>  @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>   # MPC8610 Platform Support
>   obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>
>  +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
>  +
>  diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
>  new file mode 100644
>  index 0000000..81d0933
>  --- /dev/null
>  +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
>  @@ -0,0 +1,899 @@
>  +/*
>  + * Freescale MPC5200 PSC in I2S mode
>  + * ALSA SoC Digital Audio Interface (DAI) driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/init.h>
>  +#include <linux/module.h>
>  +#include <linux/interrupt.h>
>  +#include <linux/device.h>
>  +#include <linux/delay.h>
>  +#include <linux/of_device.h>
>  +#include <linux/of_platform.h>
>  +#include <linux/dma-mapping.h>
>  +
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/initval.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +
>  +#include <sysdev/bestcomm/bestcomm.h>
>  +#include <sysdev/bestcomm/gen_bd.h>
>  +#include <asm/mpc52xx_psc.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
>  +MODULE_LICENSE("GPL");
>  +
>  +/**
>  + * PSC_I2S_RATES: sample rates supported by the I2S
>  + *
>  + * This driver currently only supports the PSC running in I2S slave mode,
>  + * which means the codec determines the sample rate.  Therefore, we tell
>  + * ALSA that we support all rates and let the codec driver decide what rates
>  + * are really supported.
>  + */
>  +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
>  +                       SNDRV_PCM_RATE_CONTINUOUS)
>  +
>  +/**
>  + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
>  + */
>  +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
>  +                        SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/**
>  + * psc_i2s_stream - Data specific to a single stream (playback or capture)
>  + * @active:            flag indicating if the stream is active
>  + * @psc_i2s:           pointer back to parent psc_i2s data structure
>  + * @bcom_task:         bestcomm task structure
>  + * @irq:               irq number for bestcomm task
>  + * @period_start:      physical address of start of DMA region
>  + * @period_end:                physical address of end of DMA region
>  + * @period_next_pt:    physical address of next DMA buffer to enqueue
>  + * @period_bytes:      size of DMA period in bytes
>  + */
>  +struct psc_i2s_stream {
>  +       int active;
>  +       struct psc_i2s *psc_i2s;
>  +       struct bcom_task *bcom_task;
>  +       int irq;
>  +       struct snd_pcm_substream *stream;
>  +       dma_addr_t period_start;
>  +       dma_addr_t period_end;
>  +       dma_addr_t period_next_pt;
>  +       dma_addr_t period_current_pt;
>  +       int period_bytes;
>  +};
>  +
>  +/**
>  + * psc_i2s - Private driver data
>  + * @name: short name for this device ("PSC0", "PSC1", etc)
>  + * @psc_regs: pointer to the PSC's registers
>  + * @fifo_regs: pointer to the PSC's FIFO registers
>  + * @irq: IRQ of this PSC
>  + * @dev: struct device pointer
>  + * @playback: the number of playback streams opened
>  + * @capture: the number of capture streams opened
>  + * @dai: the CPU DAI for this device
>  + * @playback_stream: Playback stream context data
>  + * @capture_stream: Capture stream context data
>  + */
>  +struct psc_i2s {
>  +       char name[32];
>  +       struct mpc52xx_psc __iomem *psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fifo_regs;
>  +       unsigned int irq;
>  +       struct device *dev;
>  +       struct snd_soc_cpu_dai dai;
>  +       spinlock_t lock;
>  +
>  +       /* per-stream data */
>  +       struct psc_i2s_stream playback_stream;
>  +       struct psc_i2s_stream capture_stream;
>  +
>  +       /* Statistics */
>  +       struct {
>  +               int overrun_count;
>  +               int underrun_count;
>  +       } stats;
>  +};
>  +
>  +/*
>  + * Interrupt handlers
>  + */
>  +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
>  +{
>  +       struct psc_i2s *psc_i2s = _psc_i2s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u16 isr;
>  +
>  +       isr = in_be16(&regs->mpc52xx_psc_isr);
>  +       imr = in_be16(&regs->mpc52xx_psc_imr);
>  +
>  +       /* Playback underrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
>  +               psc_i2s->stats.underrun_count++;
>  +
>  +       /* Capture overrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_ORERR)
>  +               psc_i2s->stats.overrun_count++;
>  +
>  +       out_8(&regs->command, 4 << 4);  /* reset the error status */
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
>  + * @s: pointer to stream private data structure
>  + *
>  + * Enqueues another audio period buffer into the bestcomm queue.
>  + *
>  + * Note: The routine must only be called when there is space available in
>  + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
>  + * will get out of sync
>  + */
>  +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
>  +{
>  +       struct bcom_bd *bd;
>  +
>  +       /* Prepare and enqueue the next buffer descriptor */
>  +       bd = bcom_prepare_next_buffer(s->bcom_task);
>  +       bd->status = s->period_bytes;
>  +       bd->data[0] = s->period_next_pt;
>  +       bcom_submit_next_buffer(s->bcom_task, NULL);
>  +
>  +       /* Update for next period */
>  +       s->period_next_pt += s->period_bytes;
>  +       if (s->period_next_pt >= s->period_end)
>  +               s->period_next_pt = s->period_start;
>  +}
>  +
>  +/* Bestcomm DMA irq handler */
>  +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
>  +{
>  +       struct psc_i2s_stream *s = _psc_i2s_stream;
>  +
>  +       //spin_lock(&s->psc_i2s->lock);
>  +
>  +       /* For each finished period, dequeue the completed period buffer
>  +        * and enqueue a new one in it's place. */
>  +       while (bcom_buffer_done(s->bcom_task)) {
>  +               bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +               s->period_current_pt += s->period_bytes;
>  +               if (s->period_current_pt >= s->period_end)
>  +                       s->period_current_pt = s->period_start;
>  +               psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +       }
>  +
>  +       //spin_unlock(&s->psc_i2s->lock);
>  +
>  +       /* If the stream is active, then also inform the PCM middle layer
>  +        * of the period finished event. */
>  +       if (s->active)
>  +               snd_pcm_period_elapsed(s->stream);
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_startup: create a new substream
>  + *
>  + * This is the first function called when a stream is opened.
>  + *
>  + * If this is the first stream open, then grab the IRQ and program most of
>  + * the PSC registers.
>  + */
>  +static int psc_i2s_startup(struct snd_pcm_substream *substream)
>  +{
>  +       int playback_irq, capture_irq, rc;
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
>  +
>  +       /* Disable all interrupts and reset the PSC */
>  +       out_be16(&regs->mpc52xx_psc_imr, 0);
>  +       out_8(&regs->command, 3 << 4); /* reset transmitter */
>  +       out_8(&regs->command, 2 << 4); /* reset receiver */
>  +       out_8(&regs->command, 1 << 4); /* reset mode */
>  +       out_8(&regs->command, 4 << 4); /* reset error */
>  +
>  +       /* Default to CODEC8 mode */
>  +       out_be32(&regs->sicr,
>  +                MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
>  +                MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
>  +
>  +       /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
>  +       /* Second write to mode: register Normal mode for non loopback */
>  +       out_8(&regs->mode, 0);
>  +       out_8(&regs->mode, 0);
>  +
>  +       /* Set the TX and RX fifo alarm thresholds */
>  +       out_be16(&fiforegs->rfalarm, 0x100);    /* set RFALARM level */
>  +       out_8(&fiforegs->rfcntl, 0x4);          /* set RFGRAN level (bytes) */
>  +       out_be16(&fiforegs->tfalarm, 0x100);    /* set TFALARM level */
>  +       out_8(&fiforegs->tfcntl, 0x7);          /* set TFGRAN level (bytes*4) */
>  +
>  +       /* Setup the IRQs */
>  +       playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
>  +       capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
>  +       rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
>  +                        "psc-i2s-status", psc_i2s);
>  +       rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-capture", &psc_i2s->capture_stream);
>  +       rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-playback", &psc_i2s->playback_stream);
>  +       if (rc) {
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(capture_irq, &psc_i2s->capture_stream);
>  +               free_irq(playback_irq, &psc_i2s->playback_stream);
>  +               return -ENODEV;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
>  +                                struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       u32 sicr;
>  +
>  +       dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
>  +               " periods=%i buffer_size=%i  buffer_bytes=%i\n",
>  +               __FUNCTION__, substream, params_period_size(params),
>  +               params_period_bytes(params), params_periods(params),
>  +               params_buffer_size(params), params_buffer_bytes(params));
>  +
>  +       sicr = MPC52xx_PSC_SICR_DTS1 |
>  +              MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S16_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S24_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S32_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
>  +               break;
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid format\n");
>  +               return -EINVAL;
>  +       }
>  +       out_be32(&psc_i2s->psc_regs->sicr, sicr);
>  +
>  +       //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
>  +       //if (rc) {
>  +       //      dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
>  +       //      return rc;
>  +       //}
>  +
>  +       snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
>  +{
>  +       //return snd_pcm_lib_free_pages(substream);
>  +       snd_pcm_set_runtime_buffer(substream, NULL);
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_trigger: start and stop the DMA transfer.
>  + *
>  + * This function is called by ALSA to start, stop, pause, and resume the DMA
>  + * transfer of data.
>  + */
>  +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct snd_pcm_runtime *runtime = substream->runtime;
>  +       struct psc_i2s_stream *s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u8 psc_cmd;
>  +       long flags;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
>  +               " stream_id=%i\n",
>  +               substream, cmd, substream->pstr->stream);
>  +
>  +       switch (cmd) {
>  +        case SNDRV_PCM_TRIGGER_START:
>  +               s->period_bytes = frames_to_bytes(runtime,
>  +                                                 runtime->period_size);
>  +               s->period_start = virt_to_phys(runtime->dma_area);
>  +               s->period_end = s->period_start +
>  +                               (s->period_bytes * runtime->periods);
>  +               s->period_next_pt = s->period_start;
>  +               s->period_current_pt = s->period_start;
>  +               s->active = 1;
>  +
>  +               /* First; reset everything */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_RX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               } else {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_TX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               }
>  +
>  +               /* Next, fill up the bestcomm bd queue and enable DMA.
>  +                * This will begin filling the PSC's fifo. */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +                       bcom_gen_bd_rx_reset(s->bcom_task);
>  +               else
>  +                       bcom_gen_bd_tx_reset(s->bcom_task);
>  +               while (!bcom_queue_full(s->bcom_task))
>  +                       psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +
>  +               /* Update interrupt enable settings.  This must be done
>  +                * before the PSC is enabled so that TX underrun events
>  +                * are not missed. */
>  +               imr = 0;
>  +               if (psc_i2s->playback_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_TXEMP;
>  +               if (psc_i2s->capture_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_ORERR;
>  +               out_be16(&regs->isr_imr.imr, imr);
>  +
>  +               /* Due to errata in the i2s mode; need to line up enabling
>  +                * the transmitter with a transition on the frame sync
>  +                * line */
>  +
>  +               spin_lock_irqsave(&psc_i2s->lock, flags);
>  +               /* first make sure it is low */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
>  +               /* then wait for the transition to high */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
>  +               /* Finally, enable the PSC.
>  +                * Receiver must always be enabled; even when we only want
>  +                * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
>  +               psc_cmd = MPC52xx_PSC_RX_ENABLE;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
>  +                       psc_cmd |= MPC52xx_PSC_TX_ENABLE;
>  +               out_8(&regs->command, psc_cmd);
>  +               spin_unlock_irqrestore(&psc_i2s->lock, flags);
>  +
>  +               break;
>  +
>  +        case SNDRV_PCM_TRIGGER_STOP:
>  +               /* Turn off the PSC */
>  +               s->active = 0;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       if (!psc_i2s->playback_stream.active) {
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +                               out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                               out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       }
>  +               } else {
>  +                       out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                       out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       if (!psc_i2s->capture_stream.active)
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +               }
>  +
>  +               bcom_disable(s->bcom_task);
>  +               while (!bcom_queue_empty(s->bcom_task))
>  +                       bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +
>  +               break;
>  +
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid command\n");
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Update interrupt enable settings */
>  +       imr = 0;
>  +       if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
>  +       if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
>  +       out_be16(&regs->isr_imr.imr, imr);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_shutdown: shutdown the data transfer on a stream
>  + *
>  + * Shutdown the PSC if there are no other substreams open.
>  + */
>  +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
>  +
>  +       /*
>  +        * If this is the last active substream, disable the PSC and release
>  +        * the IRQ.
>  +        */
>  +       if (!psc_i2s->playback_stream.active &&
>  +           !psc_i2s->capture_stream.active) {
>  +               /* TODO: shut off channels */
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
>  +                        &psc_i2s->capture_stream);
>  +               free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
>  +                        &psc_i2s->playback_stream);
>  +       }
>  +}
>  +
>  +/**
>  + * psc_i2s_set_sysclk: set the clock frequency and direction
>  + *
>  + * This function is called by the machine driver to tell us what the clock
>  + * frequency and direction are.
>  + *
>  + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
>  + * and we don't care about the frequency.  Return an error if the direction
>  + * is not SND_SOC_CLOCK_IN.
>  + *
>  + * @clk_id: reserved, should be zero
>  + * @freq: the frequency of the given clock ID, currently ignored
>  + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
>  + */
>  +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
>  +                             int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
>  +                               cpu_dai, dir);
>  +       return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
>  +}
>  +
>  +/**
>  + * psc_i2s_set_fmt: set the serial format.
>  + *
>  + * This function is called by the machine driver to tell us what serial
>  + * format to use.
>  + *
>  + * This driver only supports I2S mode.  Return an error if the format is
>  + * not SND_SOC_DAIFMT_I2S.
>  + *
>  + * @format: one of SND_SOC_DAIFMT_xxx
>  + */
>  +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
>  +                               cpu_dai, format);
>  +       return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA SoC Bindings
>  + *
>  + * - Digital Audio Interface (DAI) template
>  + * - create/destroy dai hooks
>  + */
>  +
>  +/**
>  + * psc_i2s_dai_template: template CPU Digital Audio Interface
>  + */
>  +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
>  +       .type = SND_SOC_DAI_I2S,
>  +       .playback = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .capture = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .ops = {
>  +               .startup = psc_i2s_startup,
>  +               .hw_params = psc_i2s_hw_params,
>  +               .hw_free = psc_i2s_hw_free,
>  +               .shutdown = psc_i2s_shutdown,
>  +               .trigger = psc_i2s_trigger,
>  +       },
>  +       .dai_ops = {
>  +               .set_sysclk = psc_i2s_set_sysclk,
>  +               .set_fmt = psc_i2s_set_fmt,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * The PSC I2S 'ASoC platform' driver
>  + *
>  + * Can be referenced by an 'ASoC machine' driver
>  + * This driver only deals with the audio bus; it doesn't have any
>  + * interaction with the attached codec
>  + */
>  +
>  +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
>  +       .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
>  +               SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
>  +       .formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
>  +                  SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
>  +       .rate_min = 8000,
>  +       .rate_max = 48000,
>  +       .channels_min = 2,
>  +       .channels_max = 2,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .period_bytes_min       = 32,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .periods_min            = 2,
>  +       .periods_max            = 256,
>  +       .buffer_bytes_max       = 2 * 1024 * 1024,
>  +       .fifo_size              = 0,
>  +};
>  +
>  +static unsigned int psc_i2s_fixed_rates[] = {
>  +       8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
>  +};
>  +
>  +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
>  +       .count = ARRAY_SIZE(psc_i2s_fixed_rates),
>  +       .list = psc_i2s_fixed_rates,
>  +       .mask = 0,
>  +};
>  +
>  +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       int rc;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
>  +
>  +       rc = snd_pcm_hw_constraint_integer(substream->runtime,
>  +                                          SNDRV_PCM_HW_PARAM_PERIODS);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid buffer size\n");
>  +               return rc;
>  +       }
>  +       rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
>  +                                       SNDRV_PCM_HW_PARAM_RATE,
>  +                                       &psc_i2s_constraints_rates);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid rate\n");
>  +               return rc;
>  +       }
>  +
>  +       s->stream = substream;
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       s->stream = NULL;
>  +       return 0;
>  +}
>  +
>  +static snd_pcm_uframes_t
>  +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       dma_addr_t count;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
>  +       count = s->period_current_pt - s->period_start;
>  +
>  +       return bytes_to_frames(substream->runtime, count);
>  +}
>  +
>  +static struct snd_pcm_ops psc_i2s_pcm_ops = {
>  +       .open           = psc_i2s_pcm_open,
>  +       .close          = psc_i2s_pcm_close,
>  +       .ioctl          = snd_pcm_lib_ioctl,
>  +       .pointer        = psc_i2s_pcm_pointer,
>  +};
>  +
>  +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
>  +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
>  +                          struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
>  +       int rc = 0;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
>  +               card, dai, pcm);
>  +
>  +       if (!card->dev->dma_mask)
>  +               card->dev->dma_mask = &psc_i2s_pcm_dmamask;
>  +       if (!card->dev->coherent_dma_mask)
>  +               card->dev->coherent_dma_mask = 0xffffffff;
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[0].substream->dma_buffer);
>  +       if (rc) {
>  +               dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[1].substream->dma_buffer);
>  +       if (rc) {
>  +               snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
>  +               dev_err(card->dev, "Can't allocate capture DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       struct snd_pcm_substream *substream;
>  +       int stream;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
>  +
>  +       for (stream = 0; stream < 2; stream++) {
>  +               substream = pcm->streams[stream].substream;
>  +               if (substream) {
>  +                       snd_dma_free_pages(&substream->dma_buffer);
>  +                       substream->dma_buffer.area = NULL;
>  +                       substream->dma_buffer.addr = 0;
>  +               }
>  +       }
>  +}
>  +
>  +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
>  +       .name           = "mpc5200-psc-audio",
>  +       .pcm_ops        = &psc_i2s_pcm_ops,
>  +       .pcm_new        = &psc_i2s_pcm_new,
>  +       .pcm_free       = &psc_i2s_pcm_free,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Sysfs attributes for debugging
>  + */
>  +
>  +static ssize_t psc_i2s_status_show(struct device *dev,
>  +                          struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +
>  +       return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
>  +                       in_be16(&psc_i2s->psc_regs->sr_csr.status),
>  +                       in_be32(&psc_i2s->psc_regs->sicr),
>  +                       in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->rfstat),
>  +                       in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->tfstat));
>  +}
>  +
>  +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
>  +                                  const char *name)
>  +{
>  +       if (strcmp(name, "playback_underrun") == 0)
>  +               return &psc_i2s->stats.underrun_count;
>  +       if (strcmp(name, "capture_overrun") == 0)
>  +               return &psc_i2s->stats.overrun_count;
>  +
>  +       return NULL;
>  +}
>  +
>  +static ssize_t psc_i2s_stat_show(struct device *dev,
>  +                                struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       return sprintf(buf, "%i\n", *attrib);
>  +}
>  +
>  +static ssize_t psc_i2s_stat_store(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf,
>  +                                 size_t count)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       *attrib = simple_strtoul(buf, NULL, 0);
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
>  +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
>  +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
>  +
>  +/* ---------------------------------------------------------------------
>  + * OF platform bus binding code:
>  + * - Probe/remove operations
>  + * - OF device match table
>  + */
>  +static int __devinit psc_i2s_of_probe(struct of_device *op,
>  +                                     const struct of_device_id *match)
>  +{
>  +       phys_addr_t fifo;
>  +       struct psc_i2s *psc_i2s;
>  +       struct resource res;
>  +       int size, psc_id, irq, rc;
>  +       const __be32 *prop;
>  +       void __iomem *regs;
>  +
>  +       dev_dbg(&op->dev, "probing psc i2s device\n");
>  +
>  +       /* Get the PSC ID */
>  +       prop = of_get_property(op->node, "cell-index", &size);
>  +       if (!prop || size < sizeof *prop)
>  +               return -ENODEV;
>  +       psc_id = be32_to_cpu(*prop);
>  +
>  +       /* Fetch the registers and IRQ of the PSC */
>  +       irq = irq_of_parse_and_map(op->node, 0);
>  +       if (of_address_to_resource(op->node, 0, &res)) {
>  +               dev_err(&op->dev, "Missing reg property\n");
>  +               return -ENODEV;
>  +       }
>  +       regs = ioremap(res.start, 1 + res.end - res.start);
>  +       if (!regs) {
>  +               dev_err(&op->dev, "Could not map registers\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Allocate and initialize the driver private data */
>  +       psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
>  +       if (!psc_i2s) {
>  +               iounmap(regs);
>  +               return -ENOMEM;
>  +       }
>  +       spin_lock_init(&psc_i2s->lock);
>  +       psc_i2s->irq = irq;
>  +       psc_i2s->psc_regs = regs;
>  +       psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
>  +       psc_i2s->dev = &op->dev;
>  +       psc_i2s->playback_stream.psc_i2s = psc_i2s;
>  +       psc_i2s->capture_stream.psc_i2s = psc_i2s;
>  +       snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
>  +
>  +       /* Fill out the CPU DAI structure */
>  +       memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
>  +       psc_i2s->dai.private_data = psc_i2s;
>  +       psc_i2s->dai.name = psc_i2s->name;
>  +       psc_i2s->dai.id = psc_id;
>  +
>  +       /* Find the address of the fifo data registers and setup the
>  +        * DMA tasks */
>  +       fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
>  +       psc_i2s->capture_stream.bcom_task =
>  +               bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
>  +       psc_i2s->playback_stream.bcom_task =
>  +               bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
>  +       if (!psc_i2s->capture_stream.bcom_task ||
>  +           !psc_i2s->playback_stream.bcom_task) {
>  +               dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
>  +               iounmap(regs);
>  +               kfree(psc_i2s);
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Save what we've done so it can be found again later */
>  +       dev_set_drvdata(&op->dev, psc_i2s);
>  +
>  +       /* Register the SYSFS files */
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_status);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
>  +       if (rc)
>  +               dev_info(psc_i2s->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the ASoC OF helpers about it */
>  +       of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
>  +                                    &psc_i2s->dai);
>  +
>  +       return 0;
>  +}
>  +
>  +static int __devexit psc_i2s_of_remove(struct of_device *op)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
>  +
>  +       dev_dbg(&op->dev, "psc_i2s_remove()\n");
>  +
>  +       bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
>  +       bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
>  +
>  +       iounmap(psc_i2s->psc_regs);
>  +       iounmap(psc_i2s->fifo_regs);
>  +       kfree(psc_i2s);
>  +       dev_set_drvdata(&op->dev, NULL);
>  +
>  +       return 0;
>  +}
>  +
>  +/* Match table for of_platform binding */
>  +static struct of_device_id psc_i2s_match[] __devinitdata = {
>  +       { .compatible = "fsl,mpc5200-psc-i2s", },
>  +       {}
>  +};
>  +MODULE_DEVICE_TABLE(of, psc_i2s_match);
>  +
>  +static struct of_platform_driver psc_i2s_driver = {
>  +       .match_table = psc_i2s_match,
>  +       .probe = psc_i2s_of_probe,
>  +       .remove = __devexit_p(psc_i2s_of_remove),
>  +       .driver = {
>  +               .name = "mpc5200-psc-i2s",
>  +               .owner = THIS_MODULE,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Module setup and teardown; simply register the of_platform driver
>  + * for the PSC in I2S mode.
>  + */
>  +static int __init psc_i2s_init(void)
>  +{
>  +       return of_register_platform_driver(&psc_i2s_driver);
>  +}
>  +module_init(psc_i2s_init);
>  +
>  +static void __exit psc_i2s_exit(void)
>  +{
>  +       of_unregister_platform_driver(&psc_i2s_driver);
>  +}
>  +module_exit(psc_i2s_exit);
>  +
>  +
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to
share DMA code. The new Phytec pcm030 baseboard is AC97 too.

What does the device tree look like?


On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---
>
>   sound/soc/fsl/Kconfig           |    6
>   sound/soc/fsl/Makefile          |    2
>   sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>   3 files changed, 907 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
>  index 257101f..5daa8d3 100644
>  --- a/sound/soc/fsl/Kconfig
>  +++ b/sound/soc/fsl/Kconfig
>  @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>         help
>           Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>
>  +config SND_SOC_MPC5200_I2S
>  +       bool "Freescale MPC5200 PSC in I2S mode driver"
>  +       depends on SND_SOC && PPC_MPC52xx
>  +       help
>  +         Say Y here to support the MPC5200 PSCs in I2S mode.
>  +
>   endmenu
>  diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
>  index 62f680a..98729a1 100644
>  --- a/sound/soc/fsl/Makefile
>  +++ b/sound/soc/fsl/Makefile
>  @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>   # MPC8610 Platform Support
>   obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>
>  +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
>  +
>  diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
>  new file mode 100644
>  index 0000000..81d0933
>  --- /dev/null
>  +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
>  @@ -0,0 +1,899 @@
>  +/*
>  + * Freescale MPC5200 PSC in I2S mode
>  + * ALSA SoC Digital Audio Interface (DAI) driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/init.h>
>  +#include <linux/module.h>
>  +#include <linux/interrupt.h>
>  +#include <linux/device.h>
>  +#include <linux/delay.h>
>  +#include <linux/of_device.h>
>  +#include <linux/of_platform.h>
>  +#include <linux/dma-mapping.h>
>  +
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/initval.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +
>  +#include <sysdev/bestcomm/bestcomm.h>
>  +#include <sysdev/bestcomm/gen_bd.h>
>  +#include <asm/mpc52xx_psc.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
>  +MODULE_LICENSE("GPL");
>  +
>  +/**
>  + * PSC_I2S_RATES: sample rates supported by the I2S
>  + *
>  + * This driver currently only supports the PSC running in I2S slave mode,
>  + * which means the codec determines the sample rate.  Therefore, we tell
>  + * ALSA that we support all rates and let the codec driver decide what rates
>  + * are really supported.
>  + */
>  +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
>  +                       SNDRV_PCM_RATE_CONTINUOUS)
>  +
>  +/**
>  + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
>  + */
>  +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
>  +                        SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/**
>  + * psc_i2s_stream - Data specific to a single stream (playback or capture)
>  + * @active:            flag indicating if the stream is active
>  + * @psc_i2s:           pointer back to parent psc_i2s data structure
>  + * @bcom_task:         bestcomm task structure
>  + * @irq:               irq number for bestcomm task
>  + * @period_start:      physical address of start of DMA region
>  + * @period_end:                physical address of end of DMA region
>  + * @period_next_pt:    physical address of next DMA buffer to enqueue
>  + * @period_bytes:      size of DMA period in bytes
>  + */
>  +struct psc_i2s_stream {
>  +       int active;
>  +       struct psc_i2s *psc_i2s;
>  +       struct bcom_task *bcom_task;
>  +       int irq;
>  +       struct snd_pcm_substream *stream;
>  +       dma_addr_t period_start;
>  +       dma_addr_t period_end;
>  +       dma_addr_t period_next_pt;
>  +       dma_addr_t period_current_pt;
>  +       int period_bytes;
>  +};
>  +
>  +/**
>  + * psc_i2s - Private driver data
>  + * @name: short name for this device ("PSC0", "PSC1", etc)
>  + * @psc_regs: pointer to the PSC's registers
>  + * @fifo_regs: pointer to the PSC's FIFO registers
>  + * @irq: IRQ of this PSC
>  + * @dev: struct device pointer
>  + * @playback: the number of playback streams opened
>  + * @capture: the number of capture streams opened
>  + * @dai: the CPU DAI for this device
>  + * @playback_stream: Playback stream context data
>  + * @capture_stream: Capture stream context data
>  + */
>  +struct psc_i2s {
>  +       char name[32];
>  +       struct mpc52xx_psc __iomem *psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fifo_regs;
>  +       unsigned int irq;
>  +       struct device *dev;
>  +       struct snd_soc_cpu_dai dai;
>  +       spinlock_t lock;
>  +
>  +       /* per-stream data */
>  +       struct psc_i2s_stream playback_stream;
>  +       struct psc_i2s_stream capture_stream;
>  +
>  +       /* Statistics */
>  +       struct {
>  +               int overrun_count;
>  +               int underrun_count;
>  +       } stats;
>  +};
>  +
>  +/*
>  + * Interrupt handlers
>  + */
>  +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
>  +{
>  +       struct psc_i2s *psc_i2s = _psc_i2s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u16 isr;
>  +
>  +       isr = in_be16(&regs->mpc52xx_psc_isr);
>  +       imr = in_be16(&regs->mpc52xx_psc_imr);
>  +
>  +       /* Playback underrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
>  +               psc_i2s->stats.underrun_count++;
>  +
>  +       /* Capture overrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_ORERR)
>  +               psc_i2s->stats.overrun_count++;
>  +
>  +       out_8(&regs->command, 4 << 4);  /* reset the error status */
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
>  + * @s: pointer to stream private data structure
>  + *
>  + * Enqueues another audio period buffer into the bestcomm queue.
>  + *
>  + * Note: The routine must only be called when there is space available in
>  + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
>  + * will get out of sync
>  + */
>  +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
>  +{
>  +       struct bcom_bd *bd;
>  +
>  +       /* Prepare and enqueue the next buffer descriptor */
>  +       bd = bcom_prepare_next_buffer(s->bcom_task);
>  +       bd->status = s->period_bytes;
>  +       bd->data[0] = s->period_next_pt;
>  +       bcom_submit_next_buffer(s->bcom_task, NULL);
>  +
>  +       /* Update for next period */
>  +       s->period_next_pt += s->period_bytes;
>  +       if (s->period_next_pt >= s->period_end)
>  +               s->period_next_pt = s->period_start;
>  +}
>  +
>  +/* Bestcomm DMA irq handler */
>  +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
>  +{
>  +       struct psc_i2s_stream *s = _psc_i2s_stream;
>  +
>  +       //spin_lock(&s->psc_i2s->lock);
>  +
>  +       /* For each finished period, dequeue the completed period buffer
>  +        * and enqueue a new one in it's place. */
>  +       while (bcom_buffer_done(s->bcom_task)) {
>  +               bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +               s->period_current_pt += s->period_bytes;
>  +               if (s->period_current_pt >= s->period_end)
>  +                       s->period_current_pt = s->period_start;
>  +               psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +       }
>  +
>  +       //spin_unlock(&s->psc_i2s->lock);
>  +
>  +       /* If the stream is active, then also inform the PCM middle layer
>  +        * of the period finished event. */
>  +       if (s->active)
>  +               snd_pcm_period_elapsed(s->stream);
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_startup: create a new substream
>  + *
>  + * This is the first function called when a stream is opened.
>  + *
>  + * If this is the first stream open, then grab the IRQ and program most of
>  + * the PSC registers.
>  + */
>  +static int psc_i2s_startup(struct snd_pcm_substream *substream)
>  +{
>  +       int playback_irq, capture_irq, rc;
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
>  +
>  +       /* Disable all interrupts and reset the PSC */
>  +       out_be16(&regs->mpc52xx_psc_imr, 0);
>  +       out_8(&regs->command, 3 << 4); /* reset transmitter */
>  +       out_8(&regs->command, 2 << 4); /* reset receiver */
>  +       out_8(&regs->command, 1 << 4); /* reset mode */
>  +       out_8(&regs->command, 4 << 4); /* reset error */
>  +
>  +       /* Default to CODEC8 mode */
>  +       out_be32(&regs->sicr,
>  +                MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
>  +                MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
>  +
>  +       /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
>  +       /* Second write to mode: register Normal mode for non loopback */
>  +       out_8(&regs->mode, 0);
>  +       out_8(&regs->mode, 0);
>  +
>  +       /* Set the TX and RX fifo alarm thresholds */
>  +       out_be16(&fiforegs->rfalarm, 0x100);    /* set RFALARM level */
>  +       out_8(&fiforegs->rfcntl, 0x4);          /* set RFGRAN level (bytes) */
>  +       out_be16(&fiforegs->tfalarm, 0x100);    /* set TFALARM level */
>  +       out_8(&fiforegs->tfcntl, 0x7);          /* set TFGRAN level (bytes*4) */
>  +
>  +       /* Setup the IRQs */
>  +       playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
>  +       capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
>  +       rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
>  +                        "psc-i2s-status", psc_i2s);
>  +       rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-capture", &psc_i2s->capture_stream);
>  +       rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-playback", &psc_i2s->playback_stream);
>  +       if (rc) {
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(capture_irq, &psc_i2s->capture_stream);
>  +               free_irq(playback_irq, &psc_i2s->playback_stream);
>  +               return -ENODEV;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
>  +                                struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       u32 sicr;
>  +
>  +       dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
>  +               " periods=%i buffer_size=%i  buffer_bytes=%i\n",
>  +               __FUNCTION__, substream, params_period_size(params),
>  +               params_period_bytes(params), params_periods(params),
>  +               params_buffer_size(params), params_buffer_bytes(params));
>  +
>  +       sicr = MPC52xx_PSC_SICR_DTS1 |
>  +              MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S16_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S24_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S32_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
>  +               break;
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid format\n");
>  +               return -EINVAL;
>  +       }
>  +       out_be32(&psc_i2s->psc_regs->sicr, sicr);
>  +
>  +       //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
>  +       //if (rc) {
>  +       //      dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
>  +       //      return rc;
>  +       //}
>  +
>  +       snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
>  +{
>  +       //return snd_pcm_lib_free_pages(substream);
>  +       snd_pcm_set_runtime_buffer(substream, NULL);
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_trigger: start and stop the DMA transfer.
>  + *
>  + * This function is called by ALSA to start, stop, pause, and resume the DMA
>  + * transfer of data.
>  + */
>  +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct snd_pcm_runtime *runtime = substream->runtime;
>  +       struct psc_i2s_stream *s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u8 psc_cmd;
>  +       long flags;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
>  +               " stream_id=%i\n",
>  +               substream, cmd, substream->pstr->stream);
>  +
>  +       switch (cmd) {
>  +        case SNDRV_PCM_TRIGGER_START:
>  +               s->period_bytes = frames_to_bytes(runtime,
>  +                                                 runtime->period_size);
>  +               s->period_start = virt_to_phys(runtime->dma_area);
>  +               s->period_end = s->period_start +
>  +                               (s->period_bytes * runtime->periods);
>  +               s->period_next_pt = s->period_start;
>  +               s->period_current_pt = s->period_start;
>  +               s->active = 1;
>  +
>  +               /* First; reset everything */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_RX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               } else {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_TX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               }
>  +
>  +               /* Next, fill up the bestcomm bd queue and enable DMA.
>  +                * This will begin filling the PSC's fifo. */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +                       bcom_gen_bd_rx_reset(s->bcom_task);
>  +               else
>  +                       bcom_gen_bd_tx_reset(s->bcom_task);
>  +               while (!bcom_queue_full(s->bcom_task))
>  +                       psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +
>  +               /* Update interrupt enable settings.  This must be done
>  +                * before the PSC is enabled so that TX underrun events
>  +                * are not missed. */
>  +               imr = 0;
>  +               if (psc_i2s->playback_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_TXEMP;
>  +               if (psc_i2s->capture_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_ORERR;
>  +               out_be16(&regs->isr_imr.imr, imr);
>  +
>  +               /* Due to errata in the i2s mode; need to line up enabling
>  +                * the transmitter with a transition on the frame sync
>  +                * line */
>  +
>  +               spin_lock_irqsave(&psc_i2s->lock, flags);
>  +               /* first make sure it is low */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
>  +               /* then wait for the transition to high */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
>  +               /* Finally, enable the PSC.
>  +                * Receiver must always be enabled; even when we only want
>  +                * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
>  +               psc_cmd = MPC52xx_PSC_RX_ENABLE;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
>  +                       psc_cmd |= MPC52xx_PSC_TX_ENABLE;
>  +               out_8(&regs->command, psc_cmd);
>  +               spin_unlock_irqrestore(&psc_i2s->lock, flags);
>  +
>  +               break;
>  +
>  +        case SNDRV_PCM_TRIGGER_STOP:
>  +               /* Turn off the PSC */
>  +               s->active = 0;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       if (!psc_i2s->playback_stream.active) {
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +                               out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                               out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       }
>  +               } else {
>  +                       out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                       out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       if (!psc_i2s->capture_stream.active)
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +               }
>  +
>  +               bcom_disable(s->bcom_task);
>  +               while (!bcom_queue_empty(s->bcom_task))
>  +                       bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +
>  +               break;
>  +
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid command\n");
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Update interrupt enable settings */
>  +       imr = 0;
>  +       if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
>  +       if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
>  +       out_be16(&regs->isr_imr.imr, imr);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_shutdown: shutdown the data transfer on a stream
>  + *
>  + * Shutdown the PSC if there are no other substreams open.
>  + */
>  +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
>  +
>  +       /*
>  +        * If this is the last active substream, disable the PSC and release
>  +        * the IRQ.
>  +        */
>  +       if (!psc_i2s->playback_stream.active &&
>  +           !psc_i2s->capture_stream.active) {
>  +               /* TODO: shut off channels */
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
>  +                        &psc_i2s->capture_stream);
>  +               free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
>  +                        &psc_i2s->playback_stream);
>  +       }
>  +}
>  +
>  +/**
>  + * psc_i2s_set_sysclk: set the clock frequency and direction
>  + *
>  + * This function is called by the machine driver to tell us what the clock
>  + * frequency and direction are.
>  + *
>  + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
>  + * and we don't care about the frequency.  Return an error if the direction
>  + * is not SND_SOC_CLOCK_IN.
>  + *
>  + * @clk_id: reserved, should be zero
>  + * @freq: the frequency of the given clock ID, currently ignored
>  + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
>  + */
>  +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
>  +                             int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
>  +                               cpu_dai, dir);
>  +       return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
>  +}
>  +
>  +/**
>  + * psc_i2s_set_fmt: set the serial format.
>  + *
>  + * This function is called by the machine driver to tell us what serial
>  + * format to use.
>  + *
>  + * This driver only supports I2S mode.  Return an error if the format is
>  + * not SND_SOC_DAIFMT_I2S.
>  + *
>  + * @format: one of SND_SOC_DAIFMT_xxx
>  + */
>  +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
>  +                               cpu_dai, format);
>  +       return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA SoC Bindings
>  + *
>  + * - Digital Audio Interface (DAI) template
>  + * - create/destroy dai hooks
>  + */
>  +
>  +/**
>  + * psc_i2s_dai_template: template CPU Digital Audio Interface
>  + */
>  +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
>  +       .type = SND_SOC_DAI_I2S,
>  +       .playback = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .capture = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .ops = {
>  +               .startup = psc_i2s_startup,
>  +               .hw_params = psc_i2s_hw_params,
>  +               .hw_free = psc_i2s_hw_free,
>  +               .shutdown = psc_i2s_shutdown,
>  +               .trigger = psc_i2s_trigger,
>  +       },
>  +       .dai_ops = {
>  +               .set_sysclk = psc_i2s_set_sysclk,
>  +               .set_fmt = psc_i2s_set_fmt,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * The PSC I2S 'ASoC platform' driver
>  + *
>  + * Can be referenced by an 'ASoC machine' driver
>  + * This driver only deals with the audio bus; it doesn't have any
>  + * interaction with the attached codec
>  + */
>  +
>  +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
>  +       .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
>  +               SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
>  +       .formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
>  +                  SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
>  +       .rate_min = 8000,
>  +       .rate_max = 48000,
>  +       .channels_min = 2,
>  +       .channels_max = 2,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .period_bytes_min       = 32,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .periods_min            = 2,
>  +       .periods_max            = 256,
>  +       .buffer_bytes_max       = 2 * 1024 * 1024,
>  +       .fifo_size              = 0,
>  +};
>  +
>  +static unsigned int psc_i2s_fixed_rates[] = {
>  +       8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
>  +};
>  +
>  +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
>  +       .count = ARRAY_SIZE(psc_i2s_fixed_rates),
>  +       .list = psc_i2s_fixed_rates,
>  +       .mask = 0,
>  +};
>  +
>  +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       int rc;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
>  +
>  +       rc = snd_pcm_hw_constraint_integer(substream->runtime,
>  +                                          SNDRV_PCM_HW_PARAM_PERIODS);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid buffer size\n");
>  +               return rc;
>  +       }
>  +       rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
>  +                                       SNDRV_PCM_HW_PARAM_RATE,
>  +                                       &psc_i2s_constraints_rates);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid rate\n");
>  +               return rc;
>  +       }
>  +
>  +       s->stream = substream;
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       s->stream = NULL;
>  +       return 0;
>  +}
>  +
>  +static snd_pcm_uframes_t
>  +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       dma_addr_t count;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
>  +       count = s->period_current_pt - s->period_start;
>  +
>  +       return bytes_to_frames(substream->runtime, count);
>  +}
>  +
>  +static struct snd_pcm_ops psc_i2s_pcm_ops = {
>  +       .open           = psc_i2s_pcm_open,
>  +       .close          = psc_i2s_pcm_close,
>  +       .ioctl          = snd_pcm_lib_ioctl,
>  +       .pointer        = psc_i2s_pcm_pointer,
>  +};
>  +
>  +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
>  +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
>  +                          struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
>  +       int rc = 0;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
>  +               card, dai, pcm);
>  +
>  +       if (!card->dev->dma_mask)
>  +               card->dev->dma_mask = &psc_i2s_pcm_dmamask;
>  +       if (!card->dev->coherent_dma_mask)
>  +               card->dev->coherent_dma_mask = 0xffffffff;
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[0].substream->dma_buffer);
>  +       if (rc) {
>  +               dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[1].substream->dma_buffer);
>  +       if (rc) {
>  +               snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
>  +               dev_err(card->dev, "Can't allocate capture DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       struct snd_pcm_substream *substream;
>  +       int stream;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
>  +
>  +       for (stream = 0; stream < 2; stream++) {
>  +               substream = pcm->streams[stream].substream;
>  +               if (substream) {
>  +                       snd_dma_free_pages(&substream->dma_buffer);
>  +                       substream->dma_buffer.area = NULL;
>  +                       substream->dma_buffer.addr = 0;
>  +               }
>  +       }
>  +}
>  +
>  +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
>  +       .name           = "mpc5200-psc-audio",
>  +       .pcm_ops        = &psc_i2s_pcm_ops,
>  +       .pcm_new        = &psc_i2s_pcm_new,
>  +       .pcm_free       = &psc_i2s_pcm_free,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Sysfs attributes for debugging
>  + */
>  +
>  +static ssize_t psc_i2s_status_show(struct device *dev,
>  +                          struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +
>  +       return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
>  +                       in_be16(&psc_i2s->psc_regs->sr_csr.status),
>  +                       in_be32(&psc_i2s->psc_regs->sicr),
>  +                       in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->rfstat),
>  +                       in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->tfstat));
>  +}
>  +
>  +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
>  +                                  const char *name)
>  +{
>  +       if (strcmp(name, "playback_underrun") == 0)
>  +               return &psc_i2s->stats.underrun_count;
>  +       if (strcmp(name, "capture_overrun") == 0)
>  +               return &psc_i2s->stats.overrun_count;
>  +
>  +       return NULL;
>  +}
>  +
>  +static ssize_t psc_i2s_stat_show(struct device *dev,
>  +                                struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       return sprintf(buf, "%i\n", *attrib);
>  +}
>  +
>  +static ssize_t psc_i2s_stat_store(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf,
>  +                                 size_t count)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       *attrib = simple_strtoul(buf, NULL, 0);
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
>  +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
>  +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
>  +
>  +/* ---------------------------------------------------------------------
>  + * OF platform bus binding code:
>  + * - Probe/remove operations
>  + * - OF device match table
>  + */
>  +static int __devinit psc_i2s_of_probe(struct of_device *op,
>  +                                     const struct of_device_id *match)
>  +{
>  +       phys_addr_t fifo;
>  +       struct psc_i2s *psc_i2s;
>  +       struct resource res;
>  +       int size, psc_id, irq, rc;
>  +       const __be32 *prop;
>  +       void __iomem *regs;
>  +
>  +       dev_dbg(&op->dev, "probing psc i2s device\n");
>  +
>  +       /* Get the PSC ID */
>  +       prop = of_get_property(op->node, "cell-index", &size);
>  +       if (!prop || size < sizeof *prop)
>  +               return -ENODEV;
>  +       psc_id = be32_to_cpu(*prop);
>  +
>  +       /* Fetch the registers and IRQ of the PSC */
>  +       irq = irq_of_parse_and_map(op->node, 0);
>  +       if (of_address_to_resource(op->node, 0, &res)) {
>  +               dev_err(&op->dev, "Missing reg property\n");
>  +               return -ENODEV;
>  +       }
>  +       regs = ioremap(res.start, 1 + res.end - res.start);
>  +       if (!regs) {
>  +               dev_err(&op->dev, "Could not map registers\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Allocate and initialize the driver private data */
>  +       psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
>  +       if (!psc_i2s) {
>  +               iounmap(regs);
>  +               return -ENOMEM;
>  +       }
>  +       spin_lock_init(&psc_i2s->lock);
>  +       psc_i2s->irq = irq;
>  +       psc_i2s->psc_regs = regs;
>  +       psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
>  +       psc_i2s->dev = &op->dev;
>  +       psc_i2s->playback_stream.psc_i2s = psc_i2s;
>  +       psc_i2s->capture_stream.psc_i2s = psc_i2s;
>  +       snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
>  +
>  +       /* Fill out the CPU DAI structure */
>  +       memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
>  +       psc_i2s->dai.private_data = psc_i2s;
>  +       psc_i2s->dai.name = psc_i2s->name;
>  +       psc_i2s->dai.id = psc_id;
>  +
>  +       /* Find the address of the fifo data registers and setup the
>  +        * DMA tasks */
>  +       fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
>  +       psc_i2s->capture_stream.bcom_task =
>  +               bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
>  +       psc_i2s->playback_stream.bcom_task =
>  +               bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
>  +       if (!psc_i2s->capture_stream.bcom_task ||
>  +           !psc_i2s->playback_stream.bcom_task) {
>  +               dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
>  +               iounmap(regs);
>  +               kfree(psc_i2s);
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Save what we've done so it can be found again later */
>  +       dev_set_drvdata(&op->dev, psc_i2s);
>  +
>  +       /* Register the SYSFS files */
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_status);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
>  +       if (rc)
>  +               dev_info(psc_i2s->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the ASoC OF helpers about it */
>  +       of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
>  +                                    &psc_i2s->dai);
>  +
>  +       return 0;
>  +}
>  +
>  +static int __devexit psc_i2s_of_remove(struct of_device *op)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
>  +
>  +       dev_dbg(&op->dev, "psc_i2s_remove()\n");
>  +
>  +       bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
>  +       bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
>  +
>  +       iounmap(psc_i2s->psc_regs);
>  +       iounmap(psc_i2s->fifo_regs);
>  +       kfree(psc_i2s);
>  +       dev_set_drvdata(&op->dev, NULL);
>  +
>  +       return 0;
>  +}
>  +
>  +/* Match table for of_platform binding */
>  +static struct of_device_id psc_i2s_match[] __devinitdata = {
>  +       { .compatible = "fsl,mpc5200-psc-i2s", },
>  +       {}
>  +};
>  +MODULE_DEVICE_TABLE(of, psc_i2s_match);
>  +
>  +static struct of_platform_driver psc_i2s_driver = {
>  +       .match_table = psc_i2s_match,
>  +       .probe = psc_i2s_of_probe,
>  +       .remove = __devexit_p(psc_i2s_of_remove),
>  +       .driver = {
>  +               .name = "mpc5200-psc-i2s",
>  +               .owner = THIS_MODULE,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Module setup and teardown; simply register the of_platform driver
>  + * for the PSC in I2S mode.
>  + */
>  +static int __init psc_i2s_init(void)
>  +{
>  +       return of_register_platform_driver(&psc_i2s_driver);
>  +}
>  +module_init(psc_i2s_init);
>  +
>  +static void __exit psc_i2s_exit(void)
>  +{
>  +       of_unregister_platform_driver(&psc_i2s_driver);
>  +}
>  +module_exit(psc_i2s_exit);
>  +
>  +
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Wed, Jul 02, 2008 at 09:51:59AM -0400, Jon Smirl wrote:
> DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to
> share DMA code. The new Phytec pcm030 baseboard is AC97 too.

I agree, but I'm not sure the best way to organize a split.  I'm not
doing anything with the Efika at the moment and I haven't dug into the
details of what needs to be done for AC97.

> What does the device tree look like?

Here is the relevant excerpt, but it is not documented yet.  I haven't
made any attempt encode the layout into the device tree (clocking, etc.)

	spi@f00 {
		#address-cells = <1>;
		#size-cells = <0>;
		compatible = "fsl,mpc5200b-spi","fsl,mpc5200-spi";
		reg = <0xf00 0x20>;
		num-slaves = <32>;
		interrupts = <2 13 0 2 14 0>;
		interrupt-parent = <&mpc5200_pic>;

		codec1: codec@2 {
			compatible = "ti,tlv320aic26";
			linux,modalias = "tlv320aic26";
			max-speed = <1000000>;
			reg = <2>;
		};

		codec2: codec@3 {
			compatible = "ti,tlv320aic26";
			linux,modalias = "tlv320aic26";
			max-speed = <1000000>;
			reg = <3>;
		};
	};

	i2s@2200 {		// PSC2
		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
		cell-index = <1>;
		reg = <0x2200 0x100>;
		interrupts = <2 2 0>;
		interrupt-parent = <&mpc5200_pic>;
		codec-handle = <&codec1>;
	};
	i2s@2400 {		// PSC3
		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
		cell-index = <2>;
		reg = <0x2400 0x100>;
		interrupts = <2 3 0>;
		interrupt-parent = <&mpc5200_pic>;
		codec-handle = <&codec2>;
	};

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Wed, Jul 02, 2008 at 09:51:59AM -0400, Jon Smirl wrote:
> DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to
> share DMA code. The new Phytec pcm030 baseboard is AC97 too.

I agree, but I'm not sure the best way to organize a split.  I'm not
doing anything with the Efika at the moment and I haven't dug into the
details of what needs to be done for AC97.

> What does the device tree look like?

Here is the relevant excerpt, but it is not documented yet.  I haven't
made any attempt encode the layout into the device tree (clocking, etc.)

	spi@f00 {
		#address-cells = <1>;
		#size-cells = <0>;
		compatible = "fsl,mpc5200b-spi","fsl,mpc5200-spi";
		reg = <0xf00 0x20>;
		num-slaves = <32>;
		interrupts = <2 13 0 2 14 0>;
		interrupt-parent = <&mpc5200_pic>;

		codec1: codec@2 {
			compatible = "ti,tlv320aic26";
			linux,modalias = "tlv320aic26";
			max-speed = <1000000>;
			reg = <2>;
		};

		codec2: codec@3 {
			compatible = "ti,tlv320aic26";
			linux,modalias = "tlv320aic26";
			max-speed = <1000000>;
			reg = <3>;
		};
	};

	i2s@2200 {		// PSC2
		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
		cell-index = <1>;
		reg = <0x2200 0x100>;
		interrupts = <2 2 0>;
		interrupt-parent = <&mpc5200_pic>;
		codec-handle = <&codec1>;
	};
	i2s@2400 {		// PSC3
		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
		cell-index = <2>;
		reg = <0x2400 0x100>;
		interrupts = <2 3 0>;
		interrupt-parent = <&mpc5200_pic>;
		codec-handle = <&codec2>;
	};

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Timur Tabi]

> 	i2s@2200 {		// PSC2
> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
> 		cell-index = <1>;

cell-index should be zero-based, not one-based.

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Timur Tabi]

> 	i2s@2200 {		// PSC2
> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
> 		cell-index = <1>;

cell-index should be zero-based, not one-based.

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
>> 	i2s@2200 {		// PSC2
>> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
>> 		cell-index = <1>;
>
> cell-index should be zero-based, not one-based.
>

Umm... this is for PSC #2... cell index is set to '1'.  It is zero
based.

g.

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
>> 	i2s@2200 {		// PSC2
>> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
>> 		cell-index = <1>;
>
> cell-index should be zero-based, not one-based.
>

Umm... this is for PSC #2... cell index is set to '1'.  It is zero
based.

g.

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[David Gibson]

On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
>> 	i2s@2200 {		// PSC2
>> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
>> 		cell-index = <1>;
>
> cell-index should be zero-based, not one-based.

Well...since cell-index is for indexing shared resources, the
cell-index values should be whatever the convention is for that shared
resource, which can be defined as whatever is convenient for that
resource.  I think that's been zero-based in every user of cell-index
so far, but there's no reason it *has* to be if a different
enumeration is convenient for the shared resource in question.

-- 
David Gibson			| I'll have my music baroque, and my code
david AT gibson.dropbear.id.au	| minimalist, thank you.  NOT _the_ _other_
				| _way_ _around_!
http://www.ozlabs.org/~dgibson

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[David Gibson]

On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
>> 	i2s@2200 {		// PSC2
>> 		compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
>> 		cell-index = <1>;
>
> cell-index should be zero-based, not one-based.

Well...since cell-index is for indexing shared resources, the
cell-index values should be whatever the convention is for that shared
resource, which can be defined as whatever is convenient for that
resource.  I think that's been zero-based in every user of cell-index
so far, but there's no reason it *has* to be if a different
enumeration is convenient for the shared resource in question.

-- 
David Gibson			| I'll have my music baroque, and my code
david AT gibson.dropbear.id.au	| minimalist, thank you.  NOT _the_ _other_
				| _way_ _around_!
http://www.ozlabs.org/~dgibson

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---
>
>   sound/soc/fsl/Kconfig           |    6
>   sound/soc/fsl/Makefile          |    2
>   sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>   3 files changed, 907 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
>  index 257101f..5daa8d3 100644
>  --- a/sound/soc/fsl/Kconfig
>  +++ b/sound/soc/fsl/Kconfig
>  @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>         help
>           Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>
>  +config SND_SOC_MPC5200_I2S
>  +       bool "Freescale MPC5200 PSC in I2S mode driver"
>  +       depends on SND_SOC && PPC_MPC52xx
>  +       help
>  +         Say Y here to support the MPC5200 PSCs in I2S mode.
>  +
>   endmenu
>  diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
>  index 62f680a..98729a1 100644
>  --- a/sound/soc/fsl/Makefile
>  +++ b/sound/soc/fsl/Makefile
>  @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>   # MPC8610 Platform Support
>   obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>
>  +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
>  +
>  diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
>  new file mode 100644
>  index 0000000..81d0933
>  --- /dev/null
>  +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
>  @@ -0,0 +1,899 @@
>  +/*
>  + * Freescale MPC5200 PSC in I2S mode
>  + * ALSA SoC Digital Audio Interface (DAI) driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/init.h>
>  +#include <linux/module.h>
>  +#include <linux/interrupt.h>
>  +#include <linux/device.h>
>  +#include <linux/delay.h>
>  +#include <linux/of_device.h>
>  +#include <linux/of_platform.h>
>  +#include <linux/dma-mapping.h>
>  +
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/initval.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +
>  +#include <sysdev/bestcomm/bestcomm.h>
>  +#include <sysdev/bestcomm/gen_bd.h>
>  +#include <asm/mpc52xx_psc.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
>  +MODULE_LICENSE("GPL");
>  +
>  +/**
>  + * PSC_I2S_RATES: sample rates supported by the I2S
>  + *
>  + * This driver currently only supports the PSC running in I2S slave mode,
>  + * which means the codec determines the sample rate.  Therefore, we tell
>  + * ALSA that we support all rates and let the codec driver decide what rates
>  + * are really supported.
>  + */
>  +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
>  +                       SNDRV_PCM_RATE_CONTINUOUS)
>  +
>  +/**
>  + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
>  + */
>  +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
>  +                        SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/**
>  + * psc_i2s_stream - Data specific to a single stream (playback or capture)
>  + * @active:            flag indicating if the stream is active
>  + * @psc_i2s:           pointer back to parent psc_i2s data structure
>  + * @bcom_task:         bestcomm task structure
>  + * @irq:               irq number for bestcomm task
>  + * @period_start:      physical address of start of DMA region
>  + * @period_end:                physical address of end of DMA region
>  + * @period_next_pt:    physical address of next DMA buffer to enqueue
>  + * @period_bytes:      size of DMA period in bytes
>  + */
>  +struct psc_i2s_stream {
>  +       int active;
>  +       struct psc_i2s *psc_i2s;
>  +       struct bcom_task *bcom_task;
>  +       int irq;
>  +       struct snd_pcm_substream *stream;
>  +       dma_addr_t period_start;
>  +       dma_addr_t period_end;
>  +       dma_addr_t period_next_pt;
>  +       dma_addr_t period_current_pt;
>  +       int period_bytes;
>  +};
>  +
>  +/**
>  + * psc_i2s - Private driver data
>  + * @name: short name for this device ("PSC0", "PSC1", etc)
>  + * @psc_regs: pointer to the PSC's registers
>  + * @fifo_regs: pointer to the PSC's FIFO registers
>  + * @irq: IRQ of this PSC
>  + * @dev: struct device pointer
>  + * @playback: the number of playback streams opened
>  + * @capture: the number of capture streams opened
>  + * @dai: the CPU DAI for this device
>  + * @playback_stream: Playback stream context data
>  + * @capture_stream: Capture stream context data
>  + */
>  +struct psc_i2s {
>  +       char name[32];
>  +       struct mpc52xx_psc __iomem *psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fifo_regs;
>  +       unsigned int irq;
>  +       struct device *dev;
>  +       struct snd_soc_cpu_dai dai;
>  +       spinlock_t lock;
>  +
>  +       /* per-stream data */
>  +       struct psc_i2s_stream playback_stream;
>  +       struct psc_i2s_stream capture_stream;
>  +
>  +       /* Statistics */
>  +       struct {
>  +               int overrun_count;
>  +               int underrun_count;
>  +       } stats;
>  +};
>  +
>  +/*
>  + * Interrupt handlers
>  + */
>  +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
>  +{
>  +       struct psc_i2s *psc_i2s = _psc_i2s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u16 isr;
>  +
>  +       isr = in_be16(&regs->mpc52xx_psc_isr);
>  +       imr = in_be16(&regs->mpc52xx_psc_imr);
>  +
>  +       /* Playback underrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
>  +               psc_i2s->stats.underrun_count++;
>  +
>  +       /* Capture overrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_ORERR)
>  +               psc_i2s->stats.overrun_count++;
>  +
>  +       out_8(&regs->command, 4 << 4);  /* reset the error status */
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
>  + * @s: pointer to stream private data structure
>  + *
>  + * Enqueues another audio period buffer into the bestcomm queue.
>  + *
>  + * Note: The routine must only be called when there is space available in
>  + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
>  + * will get out of sync
>  + */
>  +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
>  +{
>  +       struct bcom_bd *bd;
>  +
>  +       /* Prepare and enqueue the next buffer descriptor */
>  +       bd = bcom_prepare_next_buffer(s->bcom_task);
>  +       bd->status = s->period_bytes;
>  +       bd->data[0] = s->period_next_pt;
>  +       bcom_submit_next_buffer(s->bcom_task, NULL);
>  +
>  +       /* Update for next period */
>  +       s->period_next_pt += s->period_bytes;
>  +       if (s->period_next_pt >= s->period_end)
>  +               s->period_next_pt = s->period_start;
>  +}
>  +
>  +/* Bestcomm DMA irq handler */
>  +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
>  +{
>  +       struct psc_i2s_stream *s = _psc_i2s_stream;
>  +
>  +       //spin_lock(&s->psc_i2s->lock);
>  +
>  +       /* For each finished period, dequeue the completed period buffer
>  +        * and enqueue a new one in it's place. */
>  +       while (bcom_buffer_done(s->bcom_task)) {
>  +               bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +               s->period_current_pt += s->period_bytes;
>  +               if (s->period_current_pt >= s->period_end)
>  +                       s->period_current_pt = s->period_start;
>  +               psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +       }
>  +
>  +       //spin_unlock(&s->psc_i2s->lock);
>  +
>  +       /* If the stream is active, then also inform the PCM middle layer
>  +        * of the period finished event. */
>  +       if (s->active)
>  +               snd_pcm_period_elapsed(s->stream);
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_startup: create a new substream
>  + *
>  + * This is the first function called when a stream is opened.
>  + *
>  + * If this is the first stream open, then grab the IRQ and program most of
>  + * the PSC registers.
>  + */
>  +static int psc_i2s_startup(struct snd_pcm_substream *substream)
>  +{
>  +       int playback_irq, capture_irq, rc;
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
>  +
>  +       /* Disable all interrupts and reset the PSC */
>  +       out_be16(&regs->mpc52xx_psc_imr, 0);
>  +       out_8(&regs->command, 3 << 4); /* reset transmitter */
>  +       out_8(&regs->command, 2 << 4); /* reset receiver */
>  +       out_8(&regs->command, 1 << 4); /* reset mode */
>  +       out_8(&regs->command, 4 << 4); /* reset error */
>  +
>  +       /* Default to CODEC8 mode */
>  +       out_be32(&regs->sicr,
>  +                MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
>  +                MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
>  +
>  +       /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
>  +       /* Second write to mode: register Normal mode for non loopback */
>  +       out_8(&regs->mode, 0);
>  +       out_8(&regs->mode, 0);
>  +
>  +       /* Set the TX and RX fifo alarm thresholds */
>  +       out_be16(&fiforegs->rfalarm, 0x100);    /* set RFALARM level */
>  +       out_8(&fiforegs->rfcntl, 0x4);          /* set RFGRAN level (bytes) */
>  +       out_be16(&fiforegs->tfalarm, 0x100);    /* set TFALARM level */
>  +       out_8(&fiforegs->tfcntl, 0x7);          /* set TFGRAN level (bytes*4) */
>  +
>  +       /* Setup the IRQs */
>  +       playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
>  +       capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
>  +       rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
>  +                        "psc-i2s-status", psc_i2s);
>  +       rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-capture", &psc_i2s->capture_stream);
>  +       rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-playback", &psc_i2s->playback_stream);
>  +       if (rc) {
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(capture_irq, &psc_i2s->capture_stream);
>  +               free_irq(playback_irq, &psc_i2s->playback_stream);
>  +               return -ENODEV;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
>  +                                struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       u32 sicr;
>  +
>  +       dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
>  +               " periods=%i buffer_size=%i  buffer_bytes=%i\n",
>  +               __FUNCTION__, substream, params_period_size(params),
>  +               params_period_bytes(params), params_periods(params),
>  +               params_buffer_size(params), params_buffer_bytes(params));
>  +
>  +       sicr = MPC52xx_PSC_SICR_DTS1 |
>  +              MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S16_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S24_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S32_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
>  +               break;
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid format\n");
>  +               return -EINVAL;
>  +       }
>  +       out_be32(&psc_i2s->psc_regs->sicr, sicr);
>  +
>  +       //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
>  +       //if (rc) {
>  +       //      dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
>  +       //      return rc;
>  +       //}
>  +
>  +       snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
>  +{
>  +       //return snd_pcm_lib_free_pages(substream);
>  +       snd_pcm_set_runtime_buffer(substream, NULL);
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_trigger: start and stop the DMA transfer.
>  + *
>  + * This function is called by ALSA to start, stop, pause, and resume the DMA
>  + * transfer of data.
>  + */
>  +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct snd_pcm_runtime *runtime = substream->runtime;
>  +       struct psc_i2s_stream *s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u8 psc_cmd;
>  +       long flags;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
>  +               " stream_id=%i\n",
>  +               substream, cmd, substream->pstr->stream);
>  +
>  +       switch (cmd) {
>  +        case SNDRV_PCM_TRIGGER_START:
>  +               s->period_bytes = frames_to_bytes(runtime,
>  +                                                 runtime->period_size);
>  +               s->period_start = virt_to_phys(runtime->dma_area);
>  +               s->period_end = s->period_start +
>  +                               (s->period_bytes * runtime->periods);
>  +               s->period_next_pt = s->period_start;
>  +               s->period_current_pt = s->period_start;
>  +               s->active = 1;
>  +
>  +               /* First; reset everything */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_RX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               } else {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_TX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               }
>  +
>  +               /* Next, fill up the bestcomm bd queue and enable DMA.
>  +                * This will begin filling the PSC's fifo. */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +                       bcom_gen_bd_rx_reset(s->bcom_task);
>  +               else
>  +                       bcom_gen_bd_tx_reset(s->bcom_task);
>  +               while (!bcom_queue_full(s->bcom_task))
>  +                       psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +
>  +               /* Update interrupt enable settings.  This must be done
>  +                * before the PSC is enabled so that TX underrun events
>  +                * are not missed. */
>  +               imr = 0;
>  +               if (psc_i2s->playback_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_TXEMP;
>  +               if (psc_i2s->capture_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_ORERR;
>  +               out_be16(&regs->isr_imr.imr, imr);
>  +
>  +               /* Due to errata in the i2s mode; need to line up enabling
>  +                * the transmitter with a transition on the frame sync
>  +                * line */
>  +
>  +               spin_lock_irqsave(&psc_i2s->lock, flags);
>  +               /* first make sure it is low */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);

Could this be moved to the front of the routine, to increase parallelism?

Once you detect the 0, it will be a fixed interval before the 1
happens. Might as well overlap the computations.

>  +               /* then wait for the transition to high */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
>  +               /* Finally, enable the PSC.
>  +                * Receiver must always be enabled; even when we only want
>  +                * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
>  +               psc_cmd = MPC52xx_PSC_RX_ENABLE;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
>  +                       psc_cmd |= MPC52xx_PSC_TX_ENABLE;
>  +               out_8(&regs->command, psc_cmd);
>  +               spin_unlock_irqrestore(&psc_i2s->lock, flags);
>  +
>  +               break;
>  +
>  +        case SNDRV_PCM_TRIGGER_STOP:
>  +               /* Turn off the PSC */
>  +               s->active = 0;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       if (!psc_i2s->playback_stream.active) {
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +                               out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                               out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       }
>  +               } else {
>  +                       out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                       out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       if (!psc_i2s->capture_stream.active)
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +               }
>  +
>  +               bcom_disable(s->bcom_task);
>  +               while (!bcom_queue_empty(s->bcom_task))
>  +                       bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +
>  +               break;
>  +
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid command\n");
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Update interrupt enable settings */
>  +       imr = 0;
>  +       if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
>  +       if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
>  +       out_be16(&regs->isr_imr.imr, imr);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_shutdown: shutdown the data transfer on a stream
>  + *
>  + * Shutdown the PSC if there are no other substreams open.
>  + */
>  +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
>  +
>  +       /*
>  +        * If this is the last active substream, disable the PSC and release
>  +        * the IRQ.
>  +        */
>  +       if (!psc_i2s->playback_stream.active &&
>  +           !psc_i2s->capture_stream.active) {
>  +               /* TODO: shut off channels */
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
>  +                        &psc_i2s->capture_stream);
>  +               free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
>  +                        &psc_i2s->playback_stream);
>  +       }
>  +}
>  +
>  +/**
>  + * psc_i2s_set_sysclk: set the clock frequency and direction
>  + *
>  + * This function is called by the machine driver to tell us what the clock
>  + * frequency and direction are.
>  + *
>  + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
>  + * and we don't care about the frequency.  Return an error if the direction
>  + * is not SND_SOC_CLOCK_IN.
>  + *
>  + * @clk_id: reserved, should be zero
>  + * @freq: the frequency of the given clock ID, currently ignored
>  + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
>  + */
>  +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
>  +                             int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
>  +                               cpu_dai, dir);
>  +       return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
>  +}
>  +
>  +/**
>  + * psc_i2s_set_fmt: set the serial format.
>  + *
>  + * This function is called by the machine driver to tell us what serial
>  + * format to use.
>  + *
>  + * This driver only supports I2S mode.  Return an error if the format is
>  + * not SND_SOC_DAIFMT_I2S.
>  + *
>  + * @format: one of SND_SOC_DAIFMT_xxx
>  + */
>  +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
>  +                               cpu_dai, format);
>  +       return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA SoC Bindings
>  + *
>  + * - Digital Audio Interface (DAI) template
>  + * - create/destroy dai hooks
>  + */
>  +
>  +/**
>  + * psc_i2s_dai_template: template CPU Digital Audio Interface
>  + */
>  +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
>  +       .type = SND_SOC_DAI_I2S,
>  +       .playback = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .capture = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .ops = {
>  +               .startup = psc_i2s_startup,
>  +               .hw_params = psc_i2s_hw_params,
>  +               .hw_free = psc_i2s_hw_free,
>  +               .shutdown = psc_i2s_shutdown,
>  +               .trigger = psc_i2s_trigger,
>  +       },
>  +       .dai_ops = {
>  +               .set_sysclk = psc_i2s_set_sysclk,
>  +               .set_fmt = psc_i2s_set_fmt,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * The PSC I2S 'ASoC platform' driver
>  + *
>  + * Can be referenced by an 'ASoC machine' driver
>  + * This driver only deals with the audio bus; it doesn't have any
>  + * interaction with the attached codec
>  + */
>  +
>  +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
>  +       .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
>  +               SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
>  +       .formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
>  +                  SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
>  +       .rate_min = 8000,
>  +       .rate_max = 48000,
>  +       .channels_min = 2,
>  +       .channels_max = 2,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .period_bytes_min       = 32,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .periods_min            = 2,
>  +       .periods_max            = 256,
>  +       .buffer_bytes_max       = 2 * 1024 * 1024,
>  +       .fifo_size              = 0,
>  +};
>  +
>  +static unsigned int psc_i2s_fixed_rates[] = {
>  +       8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
>  +};
>  +
>  +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
>  +       .count = ARRAY_SIZE(psc_i2s_fixed_rates),
>  +       .list = psc_i2s_fixed_rates,
>  +       .mask = 0,
>  +};
>  +
>  +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       int rc;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
>  +
>  +       rc = snd_pcm_hw_constraint_integer(substream->runtime,
>  +                                          SNDRV_PCM_HW_PARAM_PERIODS);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid buffer size\n");
>  +               return rc;
>  +       }
>  +       rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
>  +                                       SNDRV_PCM_HW_PARAM_RATE,
>  +                                       &psc_i2s_constraints_rates);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid rate\n");
>  +               return rc;
>  +       }
>  +
>  +       s->stream = substream;
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       s->stream = NULL;
>  +       return 0;
>  +}
>  +
>  +static snd_pcm_uframes_t
>  +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       dma_addr_t count;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
>  +       count = s->period_current_pt - s->period_start;
>  +
>  +       return bytes_to_frames(substream->runtime, count);
>  +}
>  +
>  +static struct snd_pcm_ops psc_i2s_pcm_ops = {
>  +       .open           = psc_i2s_pcm_open,
>  +       .close          = psc_i2s_pcm_close,
>  +       .ioctl          = snd_pcm_lib_ioctl,
>  +       .pointer        = psc_i2s_pcm_pointer,
>  +};
>  +
>  +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
>  +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
>  +                          struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
>  +       int rc = 0;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
>  +               card, dai, pcm);
>  +
>  +       if (!card->dev->dma_mask)
>  +               card->dev->dma_mask = &psc_i2s_pcm_dmamask;
>  +       if (!card->dev->coherent_dma_mask)
>  +               card->dev->coherent_dma_mask = 0xffffffff;
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[0].substream->dma_buffer);
>  +       if (rc) {
>  +               dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[1].substream->dma_buffer);
>  +       if (rc) {
>  +               snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
>  +               dev_err(card->dev, "Can't allocate capture DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       struct snd_pcm_substream *substream;
>  +       int stream;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
>  +
>  +       for (stream = 0; stream < 2; stream++) {
>  +               substream = pcm->streams[stream].substream;
>  +               if (substream) {
>  +                       snd_dma_free_pages(&substream->dma_buffer);
>  +                       substream->dma_buffer.area = NULL;
>  +                       substream->dma_buffer.addr = 0;
>  +               }
>  +       }
>  +}
>  +
>  +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
>  +       .name           = "mpc5200-psc-audio",
>  +       .pcm_ops        = &psc_i2s_pcm_ops,
>  +       .pcm_new        = &psc_i2s_pcm_new,
>  +       .pcm_free       = &psc_i2s_pcm_free,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Sysfs attributes for debugging
>  + */
>  +
>  +static ssize_t psc_i2s_status_show(struct device *dev,
>  +                          struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +
>  +       return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
>  +                       in_be16(&psc_i2s->psc_regs->sr_csr.status),
>  +                       in_be32(&psc_i2s->psc_regs->sicr),
>  +                       in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->rfstat),
>  +                       in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->tfstat));
>  +}
>  +
>  +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
>  +                                  const char *name)
>  +{
>  +       if (strcmp(name, "playback_underrun") == 0)
>  +               return &psc_i2s->stats.underrun_count;
>  +       if (strcmp(name, "capture_overrun") == 0)
>  +               return &psc_i2s->stats.overrun_count;
>  +
>  +       return NULL;
>  +}
>  +
>  +static ssize_t psc_i2s_stat_show(struct device *dev,
>  +                                struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       return sprintf(buf, "%i\n", *attrib);
>  +}
>  +
>  +static ssize_t psc_i2s_stat_store(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf,
>  +                                 size_t count)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       *attrib = simple_strtoul(buf, NULL, 0);
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
>  +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
>  +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
>  +
>  +/* ---------------------------------------------------------------------
>  + * OF platform bus binding code:
>  + * - Probe/remove operations
>  + * - OF device match table
>  + */
>  +static int __devinit psc_i2s_of_probe(struct of_device *op,
>  +                                     const struct of_device_id *match)
>  +{
>  +       phys_addr_t fifo;
>  +       struct psc_i2s *psc_i2s;
>  +       struct resource res;
>  +       int size, psc_id, irq, rc;
>  +       const __be32 *prop;
>  +       void __iomem *regs;
>  +
>  +       dev_dbg(&op->dev, "probing psc i2s device\n");
>  +
>  +       /* Get the PSC ID */
>  +       prop = of_get_property(op->node, "cell-index", &size);
>  +       if (!prop || size < sizeof *prop)
>  +               return -ENODEV;
>  +       psc_id = be32_to_cpu(*prop);
>  +
>  +       /* Fetch the registers and IRQ of the PSC */
>  +       irq = irq_of_parse_and_map(op->node, 0);
>  +       if (of_address_to_resource(op->node, 0, &res)) {
>  +               dev_err(&op->dev, "Missing reg property\n");
>  +               return -ENODEV;
>  +       }
>  +       regs = ioremap(res.start, 1 + res.end - res.start);
>  +       if (!regs) {
>  +               dev_err(&op->dev, "Could not map registers\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Allocate and initialize the driver private data */
>  +       psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
>  +       if (!psc_i2s) {
>  +               iounmap(regs);
>  +               return -ENOMEM;
>  +       }
>  +       spin_lock_init(&psc_i2s->lock);
>  +       psc_i2s->irq = irq;
>  +       psc_i2s->psc_regs = regs;
>  +       psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
>  +       psc_i2s->dev = &op->dev;
>  +       psc_i2s->playback_stream.psc_i2s = psc_i2s;
>  +       psc_i2s->capture_stream.psc_i2s = psc_i2s;
>  +       snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
>  +
>  +       /* Fill out the CPU DAI structure */
>  +       memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
>  +       psc_i2s->dai.private_data = psc_i2s;
>  +       psc_i2s->dai.name = psc_i2s->name;
>  +       psc_i2s->dai.id = psc_id;
>  +
>  +       /* Find the address of the fifo data registers and setup the
>  +        * DMA tasks */
>  +       fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
>  +       psc_i2s->capture_stream.bcom_task =
>  +               bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
>  +       psc_i2s->playback_stream.bcom_task =
>  +               bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
>  +       if (!psc_i2s->capture_stream.bcom_task ||
>  +           !psc_i2s->playback_stream.bcom_task) {
>  +               dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
>  +               iounmap(regs);
>  +               kfree(psc_i2s);
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Save what we've done so it can be found again later */
>  +       dev_set_drvdata(&op->dev, psc_i2s);
>  +
>  +       /* Register the SYSFS files */
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_status);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
>  +       if (rc)
>  +               dev_info(psc_i2s->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the ASoC OF helpers about it */
>  +       of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
>  +                                    &psc_i2s->dai);
>  +
>  +       return 0;
>  +}
>  +
>  +static int __devexit psc_i2s_of_remove(struct of_device *op)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
>  +
>  +       dev_dbg(&op->dev, "psc_i2s_remove()\n");
>  +
>  +       bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
>  +       bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
>  +
>  +       iounmap(psc_i2s->psc_regs);
>  +       iounmap(psc_i2s->fifo_regs);
>  +       kfree(psc_i2s);
>  +       dev_set_drvdata(&op->dev, NULL);
>  +
>  +       return 0;
>  +}
>  +
>  +/* Match table for of_platform binding */
>  +static struct of_device_id psc_i2s_match[] __devinitdata = {
>  +       { .compatible = "fsl,mpc5200-psc-i2s", },
>  +       {}
>  +};
>  +MODULE_DEVICE_TABLE(of, psc_i2s_match);
>  +
>  +static struct of_platform_driver psc_i2s_driver = {
>  +       .match_table = psc_i2s_match,
>  +       .probe = psc_i2s_of_probe,
>  +       .remove = __devexit_p(psc_i2s_of_remove),
>  +       .driver = {
>  +               .name = "mpc5200-psc-i2s",
>  +               .owner = THIS_MODULE,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Module setup and teardown; simply register the of_platform driver
>  + * for the PSC in I2S mode.
>  + */
>  +static int __init psc_i2s_init(void)
>  +{
>  +       return of_register_platform_driver(&psc_i2s_driver);
>  +}
>  +module_init(psc_i2s_init);
>  +
>  +static void __exit psc_i2s_exit(void)
>  +{
>  +       of_unregister_platform_driver(&psc_i2s_driver);
>  +}
>  +module_exit(psc_i2s_exit);
>  +
>  +
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---
>
>   sound/soc/fsl/Kconfig           |    6
>   sound/soc/fsl/Makefile          |    2
>   sound/soc/fsl/mpc5200_psc_i2s.c |  899 +++++++++++++++++++++++++++++++++++++++
>   3 files changed, 907 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
>  index 257101f..5daa8d3 100644
>  --- a/sound/soc/fsl/Kconfig
>  +++ b/sound/soc/fsl/Kconfig
>  @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>         help
>           Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>
>  +config SND_SOC_MPC5200_I2S
>  +       bool "Freescale MPC5200 PSC in I2S mode driver"
>  +       depends on SND_SOC && PPC_MPC52xx
>  +       help
>  +         Say Y here to support the MPC5200 PSCs in I2S mode.
>  +
>   endmenu
>  diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
>  index 62f680a..98729a1 100644
>  --- a/sound/soc/fsl/Makefile
>  +++ b/sound/soc/fsl/Makefile
>  @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>   # MPC8610 Platform Support
>   obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>
>  +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
>  +
>  diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
>  new file mode 100644
>  index 0000000..81d0933
>  --- /dev/null
>  +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
>  @@ -0,0 +1,899 @@
>  +/*
>  + * Freescale MPC5200 PSC in I2S mode
>  + * ALSA SoC Digital Audio Interface (DAI) driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/init.h>
>  +#include <linux/module.h>
>  +#include <linux/interrupt.h>
>  +#include <linux/device.h>
>  +#include <linux/delay.h>
>  +#include <linux/of_device.h>
>  +#include <linux/of_platform.h>
>  +#include <linux/dma-mapping.h>
>  +
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/initval.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +
>  +#include <sysdev/bestcomm/bestcomm.h>
>  +#include <sysdev/bestcomm/gen_bd.h>
>  +#include <asm/mpc52xx_psc.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
>  +MODULE_LICENSE("GPL");
>  +
>  +/**
>  + * PSC_I2S_RATES: sample rates supported by the I2S
>  + *
>  + * This driver currently only supports the PSC running in I2S slave mode,
>  + * which means the codec determines the sample rate.  Therefore, we tell
>  + * ALSA that we support all rates and let the codec driver decide what rates
>  + * are really supported.
>  + */
>  +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
>  +                       SNDRV_PCM_RATE_CONTINUOUS)
>  +
>  +/**
>  + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
>  + */
>  +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
>  +                        SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/**
>  + * psc_i2s_stream - Data specific to a single stream (playback or capture)
>  + * @active:            flag indicating if the stream is active
>  + * @psc_i2s:           pointer back to parent psc_i2s data structure
>  + * @bcom_task:         bestcomm task structure
>  + * @irq:               irq number for bestcomm task
>  + * @period_start:      physical address of start of DMA region
>  + * @period_end:                physical address of end of DMA region
>  + * @period_next_pt:    physical address of next DMA buffer to enqueue
>  + * @period_bytes:      size of DMA period in bytes
>  + */
>  +struct psc_i2s_stream {
>  +       int active;
>  +       struct psc_i2s *psc_i2s;
>  +       struct bcom_task *bcom_task;
>  +       int irq;
>  +       struct snd_pcm_substream *stream;
>  +       dma_addr_t period_start;
>  +       dma_addr_t period_end;
>  +       dma_addr_t period_next_pt;
>  +       dma_addr_t period_current_pt;
>  +       int period_bytes;
>  +};
>  +
>  +/**
>  + * psc_i2s - Private driver data
>  + * @name: short name for this device ("PSC0", "PSC1", etc)
>  + * @psc_regs: pointer to the PSC's registers
>  + * @fifo_regs: pointer to the PSC's FIFO registers
>  + * @irq: IRQ of this PSC
>  + * @dev: struct device pointer
>  + * @playback: the number of playback streams opened
>  + * @capture: the number of capture streams opened
>  + * @dai: the CPU DAI for this device
>  + * @playback_stream: Playback stream context data
>  + * @capture_stream: Capture stream context data
>  + */
>  +struct psc_i2s {
>  +       char name[32];
>  +       struct mpc52xx_psc __iomem *psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fifo_regs;
>  +       unsigned int irq;
>  +       struct device *dev;
>  +       struct snd_soc_cpu_dai dai;
>  +       spinlock_t lock;
>  +
>  +       /* per-stream data */
>  +       struct psc_i2s_stream playback_stream;
>  +       struct psc_i2s_stream capture_stream;
>  +
>  +       /* Statistics */
>  +       struct {
>  +               int overrun_count;
>  +               int underrun_count;
>  +       } stats;
>  +};
>  +
>  +/*
>  + * Interrupt handlers
>  + */
>  +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
>  +{
>  +       struct psc_i2s *psc_i2s = _psc_i2s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u16 isr;
>  +
>  +       isr = in_be16(&regs->mpc52xx_psc_isr);
>  +       imr = in_be16(&regs->mpc52xx_psc_imr);
>  +
>  +       /* Playback underrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
>  +               psc_i2s->stats.underrun_count++;
>  +
>  +       /* Capture overrun error */
>  +       if (isr & imr & MPC52xx_PSC_IMR_ORERR)
>  +               psc_i2s->stats.overrun_count++;
>  +
>  +       out_8(&regs->command, 4 << 4);  /* reset the error status */
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
>  + * @s: pointer to stream private data structure
>  + *
>  + * Enqueues another audio period buffer into the bestcomm queue.
>  + *
>  + * Note: The routine must only be called when there is space available in
>  + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
>  + * will get out of sync
>  + */
>  +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
>  +{
>  +       struct bcom_bd *bd;
>  +
>  +       /* Prepare and enqueue the next buffer descriptor */
>  +       bd = bcom_prepare_next_buffer(s->bcom_task);
>  +       bd->status = s->period_bytes;
>  +       bd->data[0] = s->period_next_pt;
>  +       bcom_submit_next_buffer(s->bcom_task, NULL);
>  +
>  +       /* Update for next period */
>  +       s->period_next_pt += s->period_bytes;
>  +       if (s->period_next_pt >= s->period_end)
>  +               s->period_next_pt = s->period_start;
>  +}
>  +
>  +/* Bestcomm DMA irq handler */
>  +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
>  +{
>  +       struct psc_i2s_stream *s = _psc_i2s_stream;
>  +
>  +       //spin_lock(&s->psc_i2s->lock);
>  +
>  +       /* For each finished period, dequeue the completed period buffer
>  +        * and enqueue a new one in it's place. */
>  +       while (bcom_buffer_done(s->bcom_task)) {
>  +               bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +               s->period_current_pt += s->period_bytes;
>  +               if (s->period_current_pt >= s->period_end)
>  +                       s->period_current_pt = s->period_start;
>  +               psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +       }
>  +
>  +       //spin_unlock(&s->psc_i2s->lock);
>  +
>  +       /* If the stream is active, then also inform the PCM middle layer
>  +        * of the period finished event. */
>  +       if (s->active)
>  +               snd_pcm_period_elapsed(s->stream);
>  +
>  +       return IRQ_HANDLED;
>  +}
>  +
>  +/**
>  + * psc_i2s_startup: create a new substream
>  + *
>  + * This is the first function called when a stream is opened.
>  + *
>  + * If this is the first stream open, then grab the IRQ and program most of
>  + * the PSC registers.
>  + */
>  +static int psc_i2s_startup(struct snd_pcm_substream *substream)
>  +{
>  +       int playback_irq, capture_irq, rc;
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
>  +
>  +       /* Disable all interrupts and reset the PSC */
>  +       out_be16(&regs->mpc52xx_psc_imr, 0);
>  +       out_8(&regs->command, 3 << 4); /* reset transmitter */
>  +       out_8(&regs->command, 2 << 4); /* reset receiver */
>  +       out_8(&regs->command, 1 << 4); /* reset mode */
>  +       out_8(&regs->command, 4 << 4); /* reset error */
>  +
>  +       /* Default to CODEC8 mode */
>  +       out_be32(&regs->sicr,
>  +                MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
>  +                MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
>  +
>  +       /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
>  +       /* Second write to mode: register Normal mode for non loopback */
>  +       out_8(&regs->mode, 0);
>  +       out_8(&regs->mode, 0);
>  +
>  +       /* Set the TX and RX fifo alarm thresholds */
>  +       out_be16(&fiforegs->rfalarm, 0x100);    /* set RFALARM level */
>  +       out_8(&fiforegs->rfcntl, 0x4);          /* set RFGRAN level (bytes) */
>  +       out_be16(&fiforegs->tfalarm, 0x100);    /* set TFALARM level */
>  +       out_8(&fiforegs->tfcntl, 0x7);          /* set TFGRAN level (bytes*4) */
>  +
>  +       /* Setup the IRQs */
>  +       playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
>  +       capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
>  +       rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
>  +                        "psc-i2s-status", psc_i2s);
>  +       rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-capture", &psc_i2s->capture_stream);
>  +       rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
>  +                         "psc-i2s-playback", &psc_i2s->playback_stream);
>  +       if (rc) {
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(capture_irq, &psc_i2s->capture_stream);
>  +               free_irq(playback_irq, &psc_i2s->playback_stream);
>  +               return -ENODEV;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
>  +                                struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       u32 sicr;
>  +
>  +       dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
>  +               " periods=%i buffer_size=%i  buffer_bytes=%i\n",
>  +               __FUNCTION__, substream, params_period_size(params),
>  +               params_period_bytes(params), params_periods(params),
>  +               params_buffer_size(params), params_buffer_bytes(params));
>  +
>  +       sicr = MPC52xx_PSC_SICR_DTS1 |
>  +              MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S16_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S24_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
>  +               break;
>  +        case SNDRV_PCM_FORMAT_S32_BE:
>  +               sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
>  +               break;
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid format\n");
>  +               return -EINVAL;
>  +       }
>  +       out_be32(&psc_i2s->psc_regs->sicr, sicr);
>  +
>  +       //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
>  +       //if (rc) {
>  +       //      dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
>  +       //      return rc;
>  +       //}
>  +
>  +       snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
>  +
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
>  +{
>  +       //return snd_pcm_lib_free_pages(substream);
>  +       snd_pcm_set_runtime_buffer(substream, NULL);
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_trigger: start and stop the DMA transfer.
>  + *
>  + * This function is called by ALSA to start, stop, pause, and resume the DMA
>  + * transfer of data.
>  + */
>  +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct snd_pcm_runtime *runtime = substream->runtime;
>  +       struct psc_i2s_stream *s;
>  +       struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
>  +       u16 imr;
>  +       u8 psc_cmd;
>  +       long flags;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
>  +               " stream_id=%i\n",
>  +               substream, cmd, substream->pstr->stream);
>  +
>  +       switch (cmd) {
>  +        case SNDRV_PCM_TRIGGER_START:
>  +               s->period_bytes = frames_to_bytes(runtime,
>  +                                                 runtime->period_size);
>  +               s->period_start = virt_to_phys(runtime->dma_area);
>  +               s->period_end = s->period_start +
>  +                               (s->period_bytes * runtime->periods);
>  +               s->period_next_pt = s->period_start;
>  +               s->period_current_pt = s->period_start;
>  +               s->active = 1;
>  +
>  +               /* First; reset everything */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_RX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               } else {
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_TX);
>  +                       out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
>  +               }
>  +
>  +               /* Next, fill up the bestcomm bd queue and enable DMA.
>  +                * This will begin filling the PSC's fifo. */
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +                       bcom_gen_bd_rx_reset(s->bcom_task);
>  +               else
>  +                       bcom_gen_bd_tx_reset(s->bcom_task);
>  +               while (!bcom_queue_full(s->bcom_task))
>  +                       psc_i2s_bcom_enqueue_next_buffer(s);
>  +               bcom_enable(s->bcom_task);
>  +
>  +               /* Update interrupt enable settings.  This must be done
>  +                * before the PSC is enabled so that TX underrun events
>  +                * are not missed. */
>  +               imr = 0;
>  +               if (psc_i2s->playback_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_TXEMP;
>  +               if (psc_i2s->capture_stream.active)
>  +                       imr |= MPC52xx_PSC_IMR_ORERR;
>  +               out_be16(&regs->isr_imr.imr, imr);
>  +
>  +               /* Due to errata in the i2s mode; need to line up enabling
>  +                * the transmitter with a transition on the frame sync
>  +                * line */
>  +
>  +               spin_lock_irqsave(&psc_i2s->lock, flags);
>  +               /* first make sure it is low */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);

Could this be moved to the front of the routine, to increase parallelism?

Once you detect the 0, it will be a fixed interval before the 1
happens. Might as well overlap the computations.

>  +               /* then wait for the transition to high */
>  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
>  +               /* Finally, enable the PSC.
>  +                * Receiver must always be enabled; even when we only want
>  +                * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
>  +               psc_cmd = MPC52xx_PSC_RX_ENABLE;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
>  +                       psc_cmd |= MPC52xx_PSC_TX_ENABLE;
>  +               out_8(&regs->command, psc_cmd);
>  +               spin_unlock_irqrestore(&psc_i2s->lock, flags);
>  +
>  +               break;
>  +
>  +        case SNDRV_PCM_TRIGGER_STOP:
>  +               /* Turn off the PSC */
>  +               s->active = 0;
>  +               if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
>  +                       if (!psc_i2s->playback_stream.active) {
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +                               out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                               out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       }
>  +               } else {
>  +                       out_8(&regs->command, 3 << 4);  /* reset tx */
>  +                       out_8(&regs->command, 4 << 4);  /* reset err */
>  +                       if (!psc_i2s->capture_stream.active)
>  +                               out_8(&regs->command, 2 << 4);  /* reset rx */
>  +               }
>  +
>  +               bcom_disable(s->bcom_task);
>  +               while (!bcom_queue_empty(s->bcom_task))
>  +                       bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
>  +
>  +               break;
>  +
>  +        default:
>  +               dev_dbg(psc_i2s->dev, "invalid command\n");
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Update interrupt enable settings */
>  +       imr = 0;
>  +       if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
>  +       if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
>  +       out_be16(&regs->isr_imr.imr, imr);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * psc_i2s_shutdown: shutdown the data transfer on a stream
>  + *
>  + * Shutdown the PSC if there are no other substreams open.
>  + */
>  +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
>  +
>  +       /*
>  +        * If this is the last active substream, disable the PSC and release
>  +        * the IRQ.
>  +        */
>  +       if (!psc_i2s->playback_stream.active &&
>  +           !psc_i2s->capture_stream.active) {
>  +               /* TODO: shut off channels */
>  +               free_irq(psc_i2s->irq, psc_i2s);
>  +               free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
>  +                        &psc_i2s->capture_stream);
>  +               free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
>  +                        &psc_i2s->playback_stream);
>  +       }
>  +}
>  +
>  +/**
>  + * psc_i2s_set_sysclk: set the clock frequency and direction
>  + *
>  + * This function is called by the machine driver to tell us what the clock
>  + * frequency and direction are.
>  + *
>  + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
>  + * and we don't care about the frequency.  Return an error if the direction
>  + * is not SND_SOC_CLOCK_IN.
>  + *
>  + * @clk_id: reserved, should be zero
>  + * @freq: the frequency of the given clock ID, currently ignored
>  + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
>  + */
>  +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
>  +                             int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
>  +                               cpu_dai, dir);
>  +       return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
>  +}
>  +
>  +/**
>  + * psc_i2s_set_fmt: set the serial format.
>  + *
>  + * This function is called by the machine driver to tell us what serial
>  + * format to use.
>  + *
>  + * This driver only supports I2S mode.  Return an error if the format is
>  + * not SND_SOC_DAIFMT_I2S.
>  + *
>  + * @format: one of SND_SOC_DAIFMT_xxx
>  + */
>  +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
>  +{
>  +       struct psc_i2s *psc_i2s = cpu_dai->private_data;
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
>  +                               cpu_dai, format);
>  +       return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA SoC Bindings
>  + *
>  + * - Digital Audio Interface (DAI) template
>  + * - create/destroy dai hooks
>  + */
>  +
>  +/**
>  + * psc_i2s_dai_template: template CPU Digital Audio Interface
>  + */
>  +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
>  +       .type = SND_SOC_DAI_I2S,
>  +       .playback = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .capture = {
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = PSC_I2S_RATES,
>  +               .formats = PSC_I2S_FORMATS,
>  +       },
>  +       .ops = {
>  +               .startup = psc_i2s_startup,
>  +               .hw_params = psc_i2s_hw_params,
>  +               .hw_free = psc_i2s_hw_free,
>  +               .shutdown = psc_i2s_shutdown,
>  +               .trigger = psc_i2s_trigger,
>  +       },
>  +       .dai_ops = {
>  +               .set_sysclk = psc_i2s_set_sysclk,
>  +               .set_fmt = psc_i2s_set_fmt,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * The PSC I2S 'ASoC platform' driver
>  + *
>  + * Can be referenced by an 'ASoC machine' driver
>  + * This driver only deals with the audio bus; it doesn't have any
>  + * interaction with the attached codec
>  + */
>  +
>  +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
>  +       .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
>  +               SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
>  +       .formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
>  +                  SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
>  +       .rate_min = 8000,
>  +       .rate_max = 48000,
>  +       .channels_min = 2,
>  +       .channels_max = 2,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .period_bytes_min       = 32,
>  +       .period_bytes_max       = 1024 * 1024,
>  +       .periods_min            = 2,
>  +       .periods_max            = 256,
>  +       .buffer_bytes_max       = 2 * 1024 * 1024,
>  +       .fifo_size              = 0,
>  +};
>  +
>  +static unsigned int psc_i2s_fixed_rates[] = {
>  +       8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
>  +};
>  +
>  +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
>  +       .count = ARRAY_SIZE(psc_i2s_fixed_rates),
>  +       .list = psc_i2s_fixed_rates,
>  +       .mask = 0,
>  +};
>  +
>  +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       int rc;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
>  +
>  +       rc = snd_pcm_hw_constraint_integer(substream->runtime,
>  +                                          SNDRV_PCM_HW_PARAM_PERIODS);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid buffer size\n");
>  +               return rc;
>  +       }
>  +       rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
>  +                                       SNDRV_PCM_HW_PARAM_RATE,
>  +                                       &psc_i2s_constraints_rates);
>  +       if (rc < 0) {
>  +               dev_err(psc_i2s->dev, "invalid rate\n");
>  +               return rc;
>  +       }
>  +
>  +       s->stream = substream;
>  +       return 0;
>  +}
>  +
>  +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +
>  +       dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       s->stream = NULL;
>  +       return 0;
>  +}
>  +
>  +static snd_pcm_uframes_t
>  +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
>  +       struct psc_i2s_stream *s;
>  +       dma_addr_t count;
>  +
>  +       if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
>  +               s = &psc_i2s->capture_stream;
>  +       else
>  +               s = &psc_i2s->playback_stream;
>  +
>  +       /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
>  +       count = s->period_current_pt - s->period_start;
>  +
>  +       return bytes_to_frames(substream->runtime, count);
>  +}
>  +
>  +static struct snd_pcm_ops psc_i2s_pcm_ops = {
>  +       .open           = psc_i2s_pcm_open,
>  +       .close          = psc_i2s_pcm_close,
>  +       .ioctl          = snd_pcm_lib_ioctl,
>  +       .pointer        = psc_i2s_pcm_pointer,
>  +};
>  +
>  +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
>  +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
>  +                          struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
>  +       int rc = 0;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
>  +               card, dai, pcm);
>  +
>  +       if (!card->dev->dma_mask)
>  +               card->dev->dma_mask = &psc_i2s_pcm_dmamask;
>  +       if (!card->dev->coherent_dma_mask)
>  +               card->dev->coherent_dma_mask = 0xffffffff;
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[0].substream->dma_buffer);
>  +       if (rc) {
>  +               dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
>  +                                &pcm->streams[1].substream->dma_buffer);
>  +       if (rc) {
>  +               snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
>  +               dev_err(card->dev, "Can't allocate capture DMA buffer\n");
>  +               return -ENOMEM;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = pcm->private_data;
>  +       struct snd_pcm_substream *substream;
>  +       int stream;
>  +
>  +       dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
>  +
>  +       for (stream = 0; stream < 2; stream++) {
>  +               substream = pcm->streams[stream].substream;
>  +               if (substream) {
>  +                       snd_dma_free_pages(&substream->dma_buffer);
>  +                       substream->dma_buffer.area = NULL;
>  +                       substream->dma_buffer.addr = 0;
>  +               }
>  +       }
>  +}
>  +
>  +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
>  +       .name           = "mpc5200-psc-audio",
>  +       .pcm_ops        = &psc_i2s_pcm_ops,
>  +       .pcm_new        = &psc_i2s_pcm_new,
>  +       .pcm_free       = &psc_i2s_pcm_free,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Sysfs attributes for debugging
>  + */
>  +
>  +static ssize_t psc_i2s_status_show(struct device *dev,
>  +                          struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +
>  +       return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
>  +                       in_be16(&psc_i2s->psc_regs->sr_csr.status),
>  +                       in_be32(&psc_i2s->psc_regs->sicr),
>  +                       in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->rfstat),
>  +                       in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
>  +                       in_be16(&psc_i2s->fifo_regs->tfstat));
>  +}
>  +
>  +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
>  +                                  const char *name)
>  +{
>  +       if (strcmp(name, "playback_underrun") == 0)
>  +               return &psc_i2s->stats.underrun_count;
>  +       if (strcmp(name, "capture_overrun") == 0)
>  +               return &psc_i2s->stats.overrun_count;
>  +
>  +       return NULL;
>  +}
>  +
>  +static ssize_t psc_i2s_stat_show(struct device *dev,
>  +                                struct device_attribute *attr, char *buf)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       return sprintf(buf, "%i\n", *attrib);
>  +}
>  +
>  +static ssize_t psc_i2s_stat_store(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf,
>  +                                 size_t count)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
>  +       int *attrib;
>  +
>  +       attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
>  +       if (!attrib)
>  +               return 0;
>  +
>  +       *attrib = simple_strtoul(buf, NULL, 0);
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
>  +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store);
>  +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
>  +
>  +/* ---------------------------------------------------------------------
>  + * OF platform bus binding code:
>  + * - Probe/remove operations
>  + * - OF device match table
>  + */
>  +static int __devinit psc_i2s_of_probe(struct of_device *op,
>  +                                     const struct of_device_id *match)
>  +{
>  +       phys_addr_t fifo;
>  +       struct psc_i2s *psc_i2s;
>  +       struct resource res;
>  +       int size, psc_id, irq, rc;
>  +       const __be32 *prop;
>  +       void __iomem *regs;
>  +
>  +       dev_dbg(&op->dev, "probing psc i2s device\n");
>  +
>  +       /* Get the PSC ID */
>  +       prop = of_get_property(op->node, "cell-index", &size);
>  +       if (!prop || size < sizeof *prop)
>  +               return -ENODEV;
>  +       psc_id = be32_to_cpu(*prop);
>  +
>  +       /* Fetch the registers and IRQ of the PSC */
>  +       irq = irq_of_parse_and_map(op->node, 0);
>  +       if (of_address_to_resource(op->node, 0, &res)) {
>  +               dev_err(&op->dev, "Missing reg property\n");
>  +               return -ENODEV;
>  +       }
>  +       regs = ioremap(res.start, 1 + res.end - res.start);
>  +       if (!regs) {
>  +               dev_err(&op->dev, "Could not map registers\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Allocate and initialize the driver private data */
>  +       psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
>  +       if (!psc_i2s) {
>  +               iounmap(regs);
>  +               return -ENOMEM;
>  +       }
>  +       spin_lock_init(&psc_i2s->lock);
>  +       psc_i2s->irq = irq;
>  +       psc_i2s->psc_regs = regs;
>  +       psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
>  +       psc_i2s->dev = &op->dev;
>  +       psc_i2s->playback_stream.psc_i2s = psc_i2s;
>  +       psc_i2s->capture_stream.psc_i2s = psc_i2s;
>  +       snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
>  +
>  +       /* Fill out the CPU DAI structure */
>  +       memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
>  +       psc_i2s->dai.private_data = psc_i2s;
>  +       psc_i2s->dai.name = psc_i2s->name;
>  +       psc_i2s->dai.id = psc_id;
>  +
>  +       /* Find the address of the fifo data registers and setup the
>  +        * DMA tasks */
>  +       fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
>  +       psc_i2s->capture_stream.bcom_task =
>  +               bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
>  +       psc_i2s->playback_stream.bcom_task =
>  +               bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
>  +       if (!psc_i2s->capture_stream.bcom_task ||
>  +           !psc_i2s->playback_stream.bcom_task) {
>  +               dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
>  +               iounmap(regs);
>  +               kfree(psc_i2s);
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* Save what we've done so it can be found again later */
>  +       dev_set_drvdata(&op->dev, psc_i2s);
>  +
>  +       /* Register the SYSFS files */
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_status);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
>  +       rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
>  +       if (rc)
>  +               dev_info(psc_i2s->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the ASoC OF helpers about it */
>  +       of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
>  +                                    &psc_i2s->dai);
>  +
>  +       return 0;
>  +}
>  +
>  +static int __devexit psc_i2s_of_remove(struct of_device *op)
>  +{
>  +       struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
>  +
>  +       dev_dbg(&op->dev, "psc_i2s_remove()\n");
>  +
>  +       bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
>  +       bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
>  +
>  +       iounmap(psc_i2s->psc_regs);
>  +       iounmap(psc_i2s->fifo_regs);
>  +       kfree(psc_i2s);
>  +       dev_set_drvdata(&op->dev, NULL);
>  +
>  +       return 0;
>  +}
>  +
>  +/* Match table for of_platform binding */
>  +static struct of_device_id psc_i2s_match[] __devinitdata = {
>  +       { .compatible = "fsl,mpc5200-psc-i2s", },
>  +       {}
>  +};
>  +MODULE_DEVICE_TABLE(of, psc_i2s_match);
>  +
>  +static struct of_platform_driver psc_i2s_driver = {
>  +       .match_table = psc_i2s_match,
>  +       .probe = psc_i2s_of_probe,
>  +       .remove = __devexit_p(psc_i2s_of_remove),
>  +       .driver = {
>  +               .name = "mpc5200-psc-i2s",
>  +               .owner = THIS_MODULE,
>  +       },
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Module setup and teardown; simply register the of_platform driver
>  + * for the PSC in I2S mode.
>  + */
>  +static int __init psc_i2s_init(void)
>  +{
>  +       return of_register_platform_driver(&psc_i2s_driver);
>  +}
>  +module_init(psc_i2s_init);
>  +
>  +static void __exit psc_i2s_exit(void)
>  +{
>  +       of_unregister_platform_driver(&psc_i2s_driver);
>  +}
>  +module_exit(psc_i2s_exit);
>  +
>  +
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Wed, Jul 02, 2008 at 11:19:18AM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> >  +               /* Due to errata in the i2s mode; need to line up enabling
> >  +                * the transmitter with a transition on the frame sync
> >  +                * line */
> >  +
> >  +               spin_lock_irqsave(&psc_i2s->lock, flags);
> >  +               /* first make sure it is low */
> >  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
> 
> Could this be moved to the front of the routine, to increase parallelism?
> 
> Once you detect the 0, it will be a fixed interval before the 1
> happens. Might as well overlap the computations.
> 

Good point.  I'll try this out and see if it remains stable.  If this
gets done wrong, then left and right channels will often get swapped.

g.

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Wed, Jul 02, 2008 at 11:19:18AM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> >  +               /* Due to errata in the i2s mode; need to line up enabling
> >  +                * the transmitter with a transition on the frame sync
> >  +                * line */
> >  +
> >  +               spin_lock_irqsave(&psc_i2s->lock, flags);
> >  +               /* first make sure it is low */
> >  +               while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
> 
> Could this be moved to the front of the routine, to increase parallelism?
> 
> Once you detect the 0, it will be a fixed interval before the 1
> happens. Might as well overlap the computations.
> 

Good point.  I'll try this out and see if it remains stable.  If this
gets done wrong, then left and right channels will often get swapped.

g.

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---

The driver is assuming a capture stream exists. My codec is output only.

I'm using external clocking, but the driver should support using the
mpc5200 for clocking. That's a little complicated since you have to
compute the divisors. For example the Phytec pcm030 board has a
33.3333Mhz xtal and runs at 400Mhz.

In order to reduce options, can the psc-i2s driver always try to use
mpc5200 clocking, then let the codec or fabric driver override it?

-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---

The driver is assuming a capture stream exists. My codec is output only.

I'm using external clocking, but the driver should support using the
mpc5200 for clocking. That's a little complicated since you have to
compute the divisors. For example the Phytec pcm030 board has a
33.3333Mhz xtal and runs at 400Mhz.

In order to reduce options, can the psc-i2s driver always try to use
mpc5200 clocking, then let the codec or fabric driver override it?

-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Mark Brown]

On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:

> The driver is assuming a capture stream exists. My codec is output only.

While the driver declares a capture stream the core doesn't require that
both capture and playback be available - it will cope with a capture
only or a playback only DAI (this is fairly common due to DAC only and
ADC only parts).  Unless there's some other issue specific to this
driver?

> I'm using external clocking, but the driver should support using the
> mpc5200 for clocking. That's a little complicated since you have to
> compute the divisors. For example the Phytec pcm030 board has a
> 33.3333Mhz xtal and runs at 400Mhz.

This is desirable, though it shouldn't be an obstacle for merging if the
driver only supports running in slave mode.

> In order to reduce options, can the psc-i2s driver always try to use
> mpc5200 clocking, then let the codec or fabric driver override it?

The clocking should always be under the control of the machine driver
with the codec and platform drivers exporting the required dividers and
PLLs/FLLs.  Neither the platform driver nor the codec driver are really
in a position to know how a given board is wired up and what
interdependencies or external requirements there are.

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Mark Brown]

On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:

> The driver is assuming a capture stream exists. My codec is output only.

While the driver declares a capture stream the core doesn't require that
both capture and playback be available - it will cope with a capture
only or a playback only DAI (this is fairly common due to DAC only and
ADC only parts).  Unless there's some other issue specific to this
driver?

> I'm using external clocking, but the driver should support using the
> mpc5200 for clocking. That's a little complicated since you have to
> compute the divisors. For example the Phytec pcm030 board has a
> 33.3333Mhz xtal and runs at 400Mhz.

This is desirable, though it shouldn't be an obstacle for merging if the
driver only supports running in slave mode.

> In order to reduce options, can the psc-i2s driver always try to use
> mpc5200 clocking, then let the codec or fabric driver override it?

The clocking should always be under the control of the machine driver
with the codec and platform drivers exporting the required dividers and
PLLs/FLLs.  Neither the platform driver nor the codec driver are really
in a position to know how a given board is wired up and what
interdependencies or external requirements there are.

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/7/08, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
> On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
>
>  > The driver is assuming a capture stream exists. My codec is output only.
>
>
> While the driver declares a capture stream the core doesn't require that
>  both capture and playback be available - it will cope with a capture
>  only or a playback only DAI (this is fairly common due to DAC only and
>  ADC only parts).  Unless there's some other issue specific to this
>  driver?

Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.

>
>  > I'm using external clocking, but the driver should support using the
>  > mpc5200 for clocking. That's a little complicated since you have to
>  > compute the divisors. For example the Phytec pcm030 board has a
>  > 33.3333Mhz xtal and runs at 400Mhz.
>
>
> This is desirable, though it shouldn't be an obstacle for merging if the
>  driver only supports running in slave mode.
>
>
>  > In order to reduce options, can the psc-i2s driver always try to use
>  > mpc5200 clocking, then let the codec or fabric driver override it?
>
>
> The clocking should always be under the control of the machine driver
>  with the codec and platform drivers exporting the required dividers and
>  PLLs/FLLs.  Neither the platform driver nor the codec driver are really
>  in a position to know how a given board is wired up and what
>  interdependencies or external requirements there are.
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/7/08, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
> On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
>
>  > The driver is assuming a capture stream exists. My codec is output only.
>
>
> While the driver declares a capture stream the core doesn't require that
>  both capture and playback be available - it will cope with a capture
>  only or a playback only DAI (this is fairly common due to DAC only and
>  ADC only parts).  Unless there's some other issue specific to this
>  driver?

Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.

>
>  > I'm using external clocking, but the driver should support using the
>  > mpc5200 for clocking. That's a little complicated since you have to
>  > compute the divisors. For example the Phytec pcm030 board has a
>  > 33.3333Mhz xtal and runs at 400Mhz.
>
>
> This is desirable, though it shouldn't be an obstacle for merging if the
>  driver only supports running in slave mode.
>
>
>  > In order to reduce options, can the psc-i2s driver always try to use
>  > mpc5200 clocking, then let the codec or fabric driver override it?
>
>
> The clocking should always be under the control of the machine driver
>  with the codec and platform drivers exporting the required dividers and
>  PLLs/FLLs.  Neither the platform driver nor the codec driver are really
>  in a position to know how a given board is wired up and what
>  interdependencies or external requirements there are.
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Mon, Jul 07, 2008 at 09:23:24AM -0400, Jon Smirl wrote:
> On 7/7/08, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
> > On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
> >
> >  > The driver is assuming a capture stream exists. My codec is output only.
> >
> >
> > While the driver declares a capture stream the core doesn't require that
> >  both capture and playback be available - it will cope with a capture
> >  only or a playback only DAI (this is fairly common due to DAC only and
> >  ADC only parts).  Unless there's some other issue specific to this
> >  driver?
> 
> Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.

Where does it GPF?  When dereferencing pcm->streams[x].substream in
psc_i2s_pcm_new?

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Mon, Jul 07, 2008 at 09:23:24AM -0400, Jon Smirl wrote:
> On 7/7/08, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
> > On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
> >
> >  > The driver is assuming a capture stream exists. My codec is output only.
> >
> >
> > While the driver declares a capture stream the core doesn't require that
> >  both capture and playback be available - it will cope with a capture
> >  only or a playback only DAI (this is fairly common due to DAC only and
> >  ADC only parts).  Unless there's some other issue specific to this
> >  driver?
> 
> Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.

Where does it GPF?  When dereferencing pcm->streams[x].substream in
psc_i2s_pcm_new?

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---

I need some slight tweaks since we are using PSC1 in cellphone mode to
distribute the audio clock.

		i2s@2000 { /* PSC1 in i2s mode */
			compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
			cell-index = <0>;
			reg = <0x2000 0x100>;
			interrupts = <0x2 0x1 0x0>;
			interrupt-parent = <&mpc5200_pic>;
		};

		i2s@2200 { /* PSC2 in i2s mode */
			compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
			cell-index = <1>;
			reg = <0x2200 0x100>;
			interrupts = <0x2 0x2 0x0>;
			interrupt-parent = <&mpc5200_pic>;
			codec-handle = <&tas0>;
			fsl5200-cellslave;
		};

Our PSC1 is in master mode, but it doesn't have a codec hooked to it.
I needed to modify the driver to initialize the PSC to i2s master mode
but then not start all of the ALSA support. You can detect this state
since there is no codec node. Putting PSC1 into master mode lets us
get our external audio clock inside the mpc5200.

PSC2 is a cellphone slave. It gets its clock from PSC1. Everything is
the same as what you are doing except I need to set
MPC52xx_PSC_SICR_CELLSLAVE and MPC52xx_PSC_SICR_GENCLK when the
fsl5200-cellslave attribute is present.

We need to tie the two PSCs up like this to get the audio clock in via
PSC1 and then have PSC2 generate the frame clock when the i2s data is
transmitted.

Do you want a diff, or do you have a new version with DMA broken out?


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
>  will not be merged as-is because it uses v1 of the ASoC API, but I want
>  to get it out there for comments.
>  ---

I need some slight tweaks since we are using PSC1 in cellphone mode to
distribute the audio clock.

		i2s@2000 { /* PSC1 in i2s mode */
			compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
			cell-index = <0>;
			reg = <0x2000 0x100>;
			interrupts = <0x2 0x1 0x0>;
			interrupt-parent = <&mpc5200_pic>;
		};

		i2s@2200 { /* PSC2 in i2s mode */
			compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s";
			cell-index = <1>;
			reg = <0x2200 0x100>;
			interrupts = <0x2 0x2 0x0>;
			interrupt-parent = <&mpc5200_pic>;
			codec-handle = <&tas0>;
			fsl5200-cellslave;
		};

Our PSC1 is in master mode, but it doesn't have a codec hooked to it.
I needed to modify the driver to initialize the PSC to i2s master mode
but then not start all of the ALSA support. You can detect this state
since there is no codec node. Putting PSC1 into master mode lets us
get our external audio clock inside the mpc5200.

PSC2 is a cellphone slave. It gets its clock from PSC1. Everything is
the same as what you are doing except I need to set
MPC52xx_PSC_SICR_CELLSLAVE and MPC52xx_PSC_SICR_GENCLK when the
fsl5200-cellslave attribute is present.

We need to tie the two PSCs up like this to get the audio clock in via
PSC1 and then have PSC2 generate the frame clock when the i2s data is
transmitted.

Do you want a diff, or do you have a new version with DMA broken out?


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Mon, Jul 07, 2008 at 12:32:29PM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> > From: Grant Likely <grant.likely@secretlab.ca>
> >
> >  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
> >  will not be merged as-is because it uses v1 of the ASoC API, but I want
> >  to get it out there for comments.
> >  ---
> We need to tie the two PSCs up like this to get the audio clock in via
> PSC1 and then have PSC2 generate the frame clock when the i2s data is
> transmitted.
> 
> Do you want a diff, or do you have a new version with DMA broken out?

You may as well send me the diff.  I can't promise that I'll actually
get it merged in, but I'll try.

g.

Re: [alsa-devel] [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[Grant Likely]

On Mon, Jul 07, 2008 at 12:32:29PM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> > From: Grant Likely <grant.likely@secretlab.ca>
> >
> >  This is an I2S bus driver for the MPC5200 PSC device.  It is probably
> >  will not be merged as-is because it uses v1 of the ASoC API, but I want
> >  to get it out there for comments.
> >  ---
> We need to tie the two PSCs up like this to get the audio clock in via
> PSC1 and then have PSC2 generate the frame clock when the i2s data is
> transmitted.
> 
> Do you want a diff, or do you have a new version with DMA broken out?

You may as well send me the diff.  I can't promise that I'll actually
get it merged in, but I'll try.

g.

Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver

[new to sound world]

[-- Attachment #1: Type: text/plain, Size: 31126 bytes --]


i am using your mpc5200 psc soc sound driver as a model for my task but for
that what i am missing is "soc-of.h" and "dts" file for the same. please
provide me with these files and if possible mail me at
dinesh.dua@coraltele.com.
thanks for your help. 

Grant Likely-2 wrote:
> 
> From: Grant Likely 
> 
> This is an I2S bus driver for the MPC5200 PSC device.  It is probably
> will not be merged as-is because it uses v1 of the ASoC API, but I want
> to get it out there for comments.
> ---
> 
>  sound/soc/fsl/Kconfig           |    6 
>  sound/soc/fsl/Makefile          |    2 
>  sound/soc/fsl/mpc5200_psc_i2s.c |  899
> +++++++++++++++++++++++++++++++++++++++
>  3 files changed, 907 insertions(+), 0 deletions(-)
> 
> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
> index 257101f..5daa8d3 100644
> --- a/sound/soc/fsl/Kconfig
> +++ b/sound/soc/fsl/Kconfig
> @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD
>  	help
>  	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
>  
> +config SND_SOC_MPC5200_I2S
> +	bool "Freescale MPC5200 PSC in I2S mode driver"
> +	depends on SND_SOC && PPC_MPC52xx
> +	help
> +	  Say Y here to support the MPC5200 PSCs in I2S mode.
> +
>  endmenu
> diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
> index 62f680a..98729a1 100644
> --- a/sound/soc/fsl/Makefile
> +++ b/sound/soc/fsl/Makefile
> @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
>  # MPC8610 Platform Support
>  obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
>  
> +obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
> +
> diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c
> b/sound/soc/fsl/mpc5200_psc_i2s.c
> new file mode 100644
> index 0000000..81d0933
> --- /dev/null
> +++ b/sound/soc/fsl/mpc5200_psc_i2s.c
> @@ -0,0 +1,899 @@
> +/*
> + * Freescale MPC5200 PSC in I2S mode
> + * ALSA SoC Digital Audio Interface (DAI) driver
> + *
> + * Copyright (C) 2008 Secret Lab Technologies Ltd.
> + */
> +
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +#include 
> +
> +#include 
> +#include 
> +#include 
> +
> +MODULE_AUTHOR("Grant Likely ");
> +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
> +MODULE_LICENSE("GPL");
> +
> +/**
> + * PSC_I2S_RATES: sample rates supported by the I2S
> + *
> + * This driver currently only supports the PSC running in I2S slave mode,
> + * which means the codec determines the sample rate.  Therefore, we tell
> + * ALSA that we support all rates and let the codec driver decide what
> rates
> + * are really supported.
> + */
> +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 |
> \
> +			SNDRV_PCM_RATE_CONTINUOUS)
> +
> +/**
> + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
> + */
> +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
> \
> +			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
> +			 SNDRV_PCM_FMTBIT_S32_BE)
> +
> +/**
> + * psc_i2s_stream - Data specific to a single stream (playback or
> capture)
> + * @active:		flag indicating if the stream is active
> + * @psc_i2s:		pointer back to parent psc_i2s data structure
> + * @bcom_task:		bestcomm task structure
> + * @irq:		irq number for bestcomm task
> + * @period_start:	physical address of start of DMA region
> + * @period_end:		physical address of end of DMA region
> + * @period_next_pt:	physical address of next DMA buffer to enqueue
> + * @period_bytes:	size of DMA period in bytes
> + */
> +struct psc_i2s_stream {
> +	int active;
> +	struct psc_i2s *psc_i2s;
> +	struct bcom_task *bcom_task;
> +	int irq;
> +	struct snd_pcm_substream *stream;
> +	dma_addr_t period_start;
> +	dma_addr_t period_end;
> +	dma_addr_t period_next_pt;
> +	dma_addr_t period_current_pt;
> +	int period_bytes;
> +};
> +
> +/**
> + * psc_i2s - Private driver data
> + * @name: short name for this device ("PSC0", "PSC1", etc)
> + * @psc_regs: pointer to the PSC's registers
> + * @fifo_regs: pointer to the PSC's FIFO registers
> + * @irq: IRQ of this PSC
> + * @dev: struct device pointer
> + * @playback: the number of playback streams opened
> + * @capture: the number of capture streams opened
> + * @dai: the CPU DAI for this device
> + * @playback_stream: Playback stream context data
> + * @capture_stream: Capture stream context data
> + */
> +struct psc_i2s {
> +	char name[32];
> +	struct mpc52xx_psc __iomem *psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fifo_regs;
> +	unsigned int irq;
> +	struct device *dev;
> +	struct snd_soc_cpu_dai dai;
> +	spinlock_t lock;
> +
> +	/* per-stream data */
> +	struct psc_i2s_stream playback_stream;
> +	struct psc_i2s_stream capture_stream;
> +
> +	/* Statistics */
> +	struct {
> +		int overrun_count;
> +		int underrun_count;
> +	} stats;
> +};
> +
> +/*
> + * Interrupt handlers
> + */
> +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
> +{
> +	struct psc_i2s *psc_i2s = _psc_i2s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u16 isr;
> +
> +	isr = in_be16(&regs->mpc52xx_psc_isr);
> +	imr = in_be16(&regs->mpc52xx_psc_imr);
> +
> +	/* Playback underrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
> +		psc_i2s->stats.underrun_count++;
> +
> +	/* Capture overrun error */
> +	if (isr & imr & MPC52xx_PSC_IMR_ORERR)
> +		psc_i2s->stats.overrun_count++;
> +
> +	out_8(&regs->command, 4 << 4);	/* reset the error status */
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
> + * @s: pointer to stream private data structure
> + *
> + * Enqueues another audio period buffer into the bestcomm queue.
> + *
> + * Note: The routine must only be called when there is space available in
> + * the queue.  Otherwise the enqueue will fail and the audio ring buffer
> + * will get out of sync
> + */
> +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
> +{
> +	struct bcom_bd *bd;
> +
> +	/* Prepare and enqueue the next buffer descriptor */
> +	bd = bcom_prepare_next_buffer(s->bcom_task);
> +	bd->status = s->period_bytes;
> +	bd->data[0] = s->period_next_pt;
> +	bcom_submit_next_buffer(s->bcom_task, NULL);
> +
> +	/* Update for next period */
> +	s->period_next_pt += s->period_bytes;
> +	if (s->period_next_pt >= s->period_end)
> +		s->period_next_pt = s->period_start;
> +}
> +
> +/* Bestcomm DMA irq handler */
> +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
> +{
> +	struct psc_i2s_stream *s = _psc_i2s_stream;
> +
> +	//spin_lock(&s->psc_i2s->lock);
> +
> +	/* For each finished period, dequeue the completed period buffer
> +	 * and enqueue a new one in it's place. */
> +	while (bcom_buffer_done(s->bcom_task)) {
> +		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +		s->period_current_pt += s->period_bytes;
> +		if (s->period_current_pt >= s->period_end)
> +			s->period_current_pt = s->period_start;
> +		psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +	}
> +
> +	//spin_unlock(&s->psc_i2s->lock);
> +
> +	/* If the stream is active, then also inform the PCM middle layer
> +	 * of the period finished event. */
> +	if (s->active)
> +		snd_pcm_period_elapsed(s->stream);
> +
> +	return IRQ_HANDLED;
> +}
> +
> +/**
> + * psc_i2s_startup: create a new substream
> + *
> + * This is the first function called when a stream is opened.
> + *
> + * If this is the first stream open, then grab the IRQ and program most
> of
> + * the PSC registers.
> + */
> +static int psc_i2s_startup(struct snd_pcm_substream *substream)
> +{
> +	int playback_irq, capture_irq, rc;
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
> +
> +	/* Disable all interrupts and reset the PSC */
> +	out_be16(&regs->mpc52xx_psc_imr, 0);
> +	out_8(&regs->command, 3 << 4); /* reset transmitter */
> +	out_8(&regs->command, 2 << 4); /* reset receiver */
> +	out_8(&regs->command, 1 << 4); /* reset mode */
> +	out_8(&regs->command, 4 << 4); /* reset error */
> +
> +	/* Default to CODEC8 mode */
> +	out_be32(&regs->sicr,
> +		 MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
> +		 MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);
> +
> +	/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
> +	/* Second write to mode: register Normal mode for non loopback */
> +	out_8(&regs->mode, 0);
> +	out_8(&regs->mode, 0);
> +
> +	/* Set the TX and RX fifo alarm thresholds */
> +	out_be16(&fiforegs->rfalarm, 0x100);	/* set RFALARM level */
> +	out_8(&fiforegs->rfcntl, 0x4);		/* set RFGRAN level (bytes) */
> +	out_be16(&fiforegs->tfalarm, 0x100);	/* set TFALARM level */
> +	out_8(&fiforegs->tfcntl, 0x7);		/* set TFGRAN level (bytes*4) */
> +
> +	/* Setup the IRQs */
> +	playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
> +	capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
> +	rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
> +			 "psc-i2s-status", psc_i2s);
> +	rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-capture", &psc_i2s->capture_stream);
> +	rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
> +			  "psc-i2s-playback", &psc_i2s->playback_stream);
> +	if (rc) {
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(capture_irq, &psc_i2s->capture_stream);
> +		free_irq(playback_irq, &psc_i2s->playback_stream);
> +		return -ENODEV;
> +	}
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
> +				 struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	u32 sicr;
> +
> +	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
> +		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
> +		__FUNCTION__, substream, params_period_size(params),
> +		params_period_bytes(params), params_periods(params),
> +		params_buffer_size(params), params_buffer_bytes(params));
> +
> +	sicr = MPC52xx_PSC_SICR_DTS1 |
> +	       MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;
> +	switch (params_format(params)) {
> +	 case SNDRV_PCM_FORMAT_S8:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S16_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S24_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;
> +		break;
> +	 case SNDRV_PCM_FORMAT_S32_BE:
> +		sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;
> +		break;
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid format\n");
> +		return -EINVAL;
> +	}
> +	out_be32(&psc_i2s->psc_regs->sicr, sicr);
> +
> +	//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
> +	//if (rc) {
> +	//	dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
> +	//	return rc;
> +	//}
> +
> +	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
> +
> +	return 0;
> +}
> +
> +static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
> +{
> +	//return snd_pcm_lib_free_pages(substream);
> +	snd_pcm_set_runtime_buffer(substream, NULL);
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_trigger: start and stop the DMA transfer.
> + *
> + * This function is called by ALSA to start, stop, pause, and resume the
> DMA
> + * transfer of data.
> + */
> +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct snd_pcm_runtime *runtime = substream->runtime;
> +	struct psc_i2s_stream *s;
> +	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
> +	u16 imr;
> +	u8 psc_cmd;
> +	long flags;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
> +		" stream_id=%i\n",
> +		substream, cmd, substream->pstr->stream);
> +
> +	switch (cmd) {
> +	 case SNDRV_PCM_TRIGGER_START:
> +		s->period_bytes = frames_to_bytes(runtime,
> +						  runtime->period_size);
> +		s->period_start = virt_to_phys(runtime->dma_area);
> +		s->period_end = s->period_start +
> +				(s->period_bytes * runtime->periods);
> +		s->period_next_pt = s->period_start;
> +		s->period_current_pt = s->period_start;
> +		s->active = 1;
> +
> +		/* First; reset everything */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			out_8(&regs->command, MPC52xx_PSC_RST_RX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		} else {
> +			out_8(&regs->command, MPC52xx_PSC_RST_TX);
> +			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
> +		}
> +
> +		/* Next, fill up the bestcomm bd queue and enable DMA.
> +		 * This will begin filling the PSC's fifo. */
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +			bcom_gen_bd_rx_reset(s->bcom_task);
> +		else
> +			bcom_gen_bd_tx_reset(s->bcom_task);
> +		while (!bcom_queue_full(s->bcom_task))
> +			psc_i2s_bcom_enqueue_next_buffer(s);
> +		bcom_enable(s->bcom_task);
> +
> +		/* Update interrupt enable settings.  This must be done
> +		 * before the PSC is enabled so that TX underrun events
> +		 * are not missed. */
> +		imr = 0;
> +		if (psc_i2s->playback_stream.active)
> +			imr |= MPC52xx_PSC_IMR_TXEMP;
> +		if (psc_i2s->capture_stream.active)
> +			imr |= MPC52xx_PSC_IMR_ORERR;
> +		out_be16(&regs->isr_imr.imr, imr);
> +
> +		/* Due to errata in the i2s mode; need to line up enabling
> +		 * the transmitter with a transition on the frame sync
> +		 * line */
> +
> +		spin_lock_irqsave(&psc_i2s->lock, flags);
> +		/* first make sure it is low */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0);
> +		/* then wait for the transition to high */
> +		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0);
> +		/* Finally, enable the PSC.
> +		 * Receiver must always be enabled; even when we only want
> +		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
> +		psc_cmd = MPC52xx_PSC_RX_ENABLE;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
> +			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
> +		out_8(&regs->command, psc_cmd);
> +		spin_unlock_irqrestore(&psc_i2s->lock, flags);
> +
> +		break;
> +
> +	 case SNDRV_PCM_TRIGGER_STOP:
> +		/* Turn off the PSC */
> +		s->active = 0;
> +		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
> +			if (!psc_i2s->playback_stream.active) {
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +				out_8(&regs->command, 3 << 4);	/* reset tx */
> +				out_8(&regs->command, 4 << 4);	/* reset err */
> +			}
> +		} else {
> +			out_8(&regs->command, 3 << 4);	/* reset tx */
> +			out_8(&regs->command, 4 << 4);	/* reset err */
> +			if (!psc_i2s->capture_stream.active)
> +				out_8(&regs->command, 2 << 4);	/* reset rx */
> +		}
> +
> +		bcom_disable(s->bcom_task);
> +		while (!bcom_queue_empty(s->bcom_task))
> +			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
> +
> +		break;
> +
> +	 default:
> +		dev_dbg(psc_i2s->dev, "invalid command\n");
> +		return -EINVAL;
> +	}
> +
> +	/* Update interrupt enable settings */
> +	imr = 0;
> +	if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
> +	if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
> +	out_be16(&regs->isr_imr.imr, imr);
> +
> +	return 0;
> +}
> +
> +/**
> + * psc_i2s_shutdown: shutdown the data transfer on a stream
> + *
> + * Shutdown the PSC if there are no other substreams open.
> + */
> +static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
> +
> +	/*
> +	 * If this is the last active substream, disable the PSC and release
> +	 * the IRQ.
> +	 */
> +	if (!psc_i2s->playback_stream.active &&
> +	    !psc_i2s->capture_stream.active) {
> +		/* TODO: shut off channels */
> +		free_irq(psc_i2s->irq, psc_i2s);
> +		free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),
> +			 &psc_i2s->capture_stream);
> +		free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),
> +			 &psc_i2s->playback_stream);
> +	}
> +}
> +
> +/**
> + * psc_i2s_set_sysclk: set the clock frequency and direction
> + *
> + * This function is called by the machine driver to tell us what the
> clock
> + * frequency and direction are.
> + *
> + * Currently, we only support operating as a clock slave
> (SND_SOC_CLOCK_IN),
> + * and we don't care about the frequency.  Return an error if the
> direction
> + * is not SND_SOC_CLOCK_IN.
> + *
> + * @clk_id: reserved, should be zero
> + * @freq: the frequency of the given clock ID, currently ignored
> + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock
> master)
> + */
> +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
> +			      int clk_id, unsigned int freq, int dir)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
> +				cpu_dai, dir);
> +	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
> +}
> +
> +/**
> + * psc_i2s_set_fmt: set the serial format.
> + *
> + * This function is called by the machine driver to tell us what serial
> + * format to use.
> + *
> + * This driver only supports I2S mode.  Return an error if the format is
> + * not SND_SOC_DAIFMT_I2S.
> + *
> + * @format: one of SND_SOC_DAIFMT_xxx
> + */
> +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int
> format)
> +{
> +	struct psc_i2s *psc_i2s = cpu_dai->private_data;
> +	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
> +				cpu_dai, format);
> +	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * ALSA SoC Bindings
> + *
> + * - Digital Audio Interface (DAI) template
> + * - create/destroy dai hooks
> + */
> +
> +/**
> + * psc_i2s_dai_template: template CPU Digital Audio Interface
> + */
> +static struct snd_soc_cpu_dai psc_i2s_dai_template = {
> +	.type = SND_SOC_DAI_I2S,
> +	.playback = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.capture = {
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = PSC_I2S_RATES,
> +		.formats = PSC_I2S_FORMATS,
> +	},
> +	.ops = {
> +		.startup = psc_i2s_startup,
> +		.hw_params = psc_i2s_hw_params,
> +		.hw_free = psc_i2s_hw_free,
> +		.shutdown = psc_i2s_shutdown,
> +		.trigger = psc_i2s_trigger,
> +	},
> +	.dai_ops = {
> +		.set_sysclk = psc_i2s_set_sysclk,
> +		.set_fmt = psc_i2s_set_fmt,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * The PSC I2S 'ASoC platform' driver
> + *
> + * Can be referenced by an 'ASoC machine' driver
> + * This driver only deals with the audio bus; it doesn't have any
> + * interaction with the attached codec
> + */
> +
> +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
> +	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
> +		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
> +	.formats = SNDRV_PCM_FMTBIT_S8 |SNDRV_PCM_FMTBIT_S16_BE |
> +		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
> +	.rate_min = 8000,
> +	.rate_max = 48000,
> +	.channels_min = 2,
> +	.channels_max = 2,
> +	.period_bytes_max	= 1024 * 1024,
> +	.period_bytes_min	= 32,
> +	.period_bytes_max	= 1024 * 1024,
> +	.periods_min		= 2,
> +	.periods_max		= 256,
> +	.buffer_bytes_max	= 2 * 1024 * 1024,
> +	.fifo_size		= 0,
> +};
> +
> +static unsigned int psc_i2s_fixed_rates[] = {
> +	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
> +};
> +
> +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
> +	.count = ARRAY_SIZE(psc_i2s_fixed_rates),
> +	.list = psc_i2s_fixed_rates,
> +	.mask = 0,
> +};
> +
> +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	int rc;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
> +
> +	rc = snd_pcm_hw_constraint_integer(substream->runtime,
> +					   SNDRV_PCM_HW_PARAM_PERIODS);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid buffer size\n");
> +		return rc;
> +	}
> +	rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
> +					SNDRV_PCM_HW_PARAM_RATE,
> +					&psc_i2s_constraints_rates);
> +	if (rc < 0) {
> +		dev_err(psc_i2s->dev, "invalid rate\n");
> +		return rc;
> +	}
> +
> +	s->stream = substream;
> +	return 0;
> +}
> +
> +static int psc_i2s_pcm_close(struct snd_pcm_substream * substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +
> +	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	s->stream = NULL;
> +	return 0;
> +}
> +
> +static snd_pcm_uframes_t
> +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
> +	struct psc_i2s_stream *s;
> +	dma_addr_t count;
> +
> +	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
> +		s = &psc_i2s->capture_stream;
> +	else
> +		s = &psc_i2s->playback_stream;
> +
> +	/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
> +	count = s->period_current_pt - s->period_start;
> +
> +	return bytes_to_frames(substream->runtime, count);
> +}
> +
> +static struct snd_pcm_ops psc_i2s_pcm_ops = {
> +	.open		= psc_i2s_pcm_open,
> +	.close		= psc_i2s_pcm_close,
> +	.ioctl		= snd_pcm_lib_ioctl,
> +	.pointer	= psc_i2s_pcm_pointer,
> +};
> +
> +static u64 psc_i2s_pcm_dmamask = 0xffffffff;
> +static int psc_i2s_pcm_new(struct snd_card *card, struct
> snd_soc_codec_dai *dai,
> +			   struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
> +	int rc = 0;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
> +		card, dai, pcm);
> +
> +	if (!card->dev->dma_mask)
> +		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
> +	if (!card->dev->coherent_dma_mask)
> +		card->dev->coherent_dma_mask = 0xffffffff;
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[0].substream->dma_buffer);
> +	if (rc) {
> +		dev_err(card->dev, "Cannot alloc playback DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
> +				 &pcm->streams[1].substream->dma_buffer);
> +	if (rc) {
> +		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
> +		dev_err(card->dev, "Can't allocate capture DMA buffer\n");
> +		return -ENOMEM;
> +	}
> +
> +	return 0;
> +}
> +
> +static void psc_i2s_pcm_free(struct snd_pcm *pcm)
> +{
> +	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
> +	struct snd_pcm_substream *substream;
> +	int stream;
> +
> +	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
> +
> +	for (stream = 0; stream < 2; stream++) {
> +		substream = pcm->streams[stream].substream;
> +		if (substream) {
> +			snd_dma_free_pages(&substream->dma_buffer);
> +			substream->dma_buffer.area = NULL;
> +			substream->dma_buffer.addr = 0;
> +		}
> +	}
> +}
> +
> +struct snd_soc_platform psc_i2s_pcm_soc_platform = {
> +	.name		= "mpc5200-psc-audio",
> +	.pcm_ops	= &psc_i2s_pcm_ops,
> +	.pcm_new	= &psc_i2s_pcm_new,
> +	.pcm_free	= &psc_i2s_pcm_free,
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Sysfs attributes for debugging
> + */
> +
> +static ssize_t psc_i2s_status_show(struct device *dev,
> +			   struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +
> +	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x
> tfnum=%i tfstat=0x%.4x\n",
> +			in_be16(&psc_i2s->psc_regs->sr_csr.status),
> +			in_be32(&psc_i2s->psc_regs->sicr),
> +			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->rfstat),
> +			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
> +			in_be16(&psc_i2s->fifo_regs->tfstat));
> +}
> +
> +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
> +				   const char *name)
> +{
> +	if (strcmp(name, "playback_underrun") == 0)
> +		return &psc_i2s->stats.underrun_count;
> +	if (strcmp(name, "capture_overrun") == 0)
> +		return &psc_i2s->stats.overrun_count;
> +
> +	return NULL;
> +}
> +
> +static ssize_t psc_i2s_stat_show(struct device *dev,
> +				 struct device_attribute *attr, char *buf)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	return sprintf(buf, "%i\n", *attrib);
> +}
> +
> +static ssize_t psc_i2s_stat_store(struct device *dev,
> +				  struct device_attribute *attr,
> +				  const char *buf,
> +				  size_t count)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
> +	int *attrib;
> +
> +	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
> +	if (!attrib)
> +		return 0;
> +
> +	*attrib = simple_strtoul(buf, NULL, 0);
> +	return count;
> +}
> +
> +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
> +DEVICE_ATTR(playback_underrun, 0644,
> psc_i2s_stat_show,psc_i2s_stat_store);
> +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
> psc_i2s_stat_store);
> +
> +/* ---------------------------------------------------------------------
> + * OF platform bus binding code:
> + * - Probe/remove operations
> + * - OF device match table
> + */
> +static int __devinit psc_i2s_of_probe(struct of_device *op,
> +				      const struct of_device_id *match)
> +{
> +	phys_addr_t fifo;
> +	struct psc_i2s *psc_i2s;
> +	struct resource res;
> +	int size, psc_id, irq, rc;
> +	const __be32 *prop;
> +	void __iomem *regs;
> +
> +	dev_dbg(&op->dev, "probing psc i2s device\n");
> +
> +	/* Get the PSC ID */
> +	prop = of_get_property(op->node, "cell-index", &size);
> +	if (!prop || size < sizeof *prop)
> +		return -ENODEV;
> +	psc_id = be32_to_cpu(*prop);
> +
> +	/* Fetch the registers and IRQ of the PSC */
> +	irq = irq_of_parse_and_map(op->node, 0);
> +	if (of_address_to_resource(op->node, 0, &res)) {
> +		dev_err(&op->dev, "Missing reg property\n");
> +		return -ENODEV;
> +	}
> +	regs = ioremap(res.start, 1 + res.end - res.start);
> +	if (!regs) {
> +		dev_err(&op->dev, "Could not map registers\n");
> +		return -ENODEV;
> +	}
> +
> +	/* Allocate and initialize the driver private data */
> +	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
> +	if (!psc_i2s) {
> +		iounmap(regs);
> +		return -ENOMEM;
> +	}
> +	spin_lock_init(&psc_i2s->lock);
> +	psc_i2s->irq = irq;
> +	psc_i2s->psc_regs = regs;
> +	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
> +	psc_i2s->dev = &op->dev;
> +	psc_i2s->playback_stream.psc_i2s = psc_i2s;
> +	psc_i2s->capture_stream.psc_i2s = psc_i2s;
> +	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
> +
> +	/* Fill out the CPU DAI structure */
> +	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
> +	psc_i2s->dai.private_data = psc_i2s;
> +	psc_i2s->dai.name = psc_i2s->name;
> +	psc_i2s->dai.id = psc_id;
> +
> +	/* Find the address of the fifo data registers and setup the
> +	 * DMA tasks */
> +	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
> +	psc_i2s->capture_stream.bcom_task =
> +		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
> +	psc_i2s->playback_stream.bcom_task =
> +		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
> +	if (!psc_i2s->capture_stream.bcom_task ||
> +	    !psc_i2s->playback_stream.bcom_task) {
> +		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
> +		iounmap(regs);
> +		kfree(psc_i2s);
> +		return -ENODEV;
> +	}
> +
> +	/* Save what we've done so it can be found again later */
> +	dev_set_drvdata(&op->dev, psc_i2s);
> +
> +	/* Register the SYSFS files */
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
> +	rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
> +	if (rc)
> +		dev_info(psc_i2s->dev, "error creating sysfs files\n");
> +
> +	/* Tell the ASoC OF helpers about it */
> +	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
> +				     &psc_i2s->dai);
> +
> +	return 0;
> +}
> +
> +static int __devexit psc_i2s_of_remove(struct of_device *op)
> +{
> +	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
> +
> +	dev_dbg(&op->dev, "psc_i2s_remove()\n");
> +
> +	bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
> +	bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
> +
> +	iounmap(psc_i2s->psc_regs);
> +	iounmap(psc_i2s->fifo_regs);
> +	kfree(psc_i2s);
> +	dev_set_drvdata(&op->dev, NULL);
> +
> +	return 0;
> +}
> +
> +/* Match table for of_platform binding */
> +static struct of_device_id psc_i2s_match[] __devinitdata = {
> +	{ .compatible = "fsl,mpc5200-psc-i2s", },
> +	{}
> +};
> +MODULE_DEVICE_TABLE(of, psc_i2s_match);
> +
> +static struct of_platform_driver psc_i2s_driver = {
> +	.match_table = psc_i2s_match,
> +	.probe = psc_i2s_of_probe,
> +	.remove = __devexit_p(psc_i2s_of_remove),
> +	.driver = {
> +		.name = "mpc5200-psc-i2s",
> +		.owner = THIS_MODULE,
> +	},
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Module setup and teardown; simply register the of_platform driver
> + * for the PSC in I2S mode.
> + */
> +static int __init psc_i2s_init(void)
> +{
> +	return of_register_platform_driver(&psc_i2s_driver);
> +}
> +module_init(psc_i2s_init);
> +
> +static void __exit psc_i2s_exit(void)
> +{
> +	of_unregister_platform_driver(&psc_i2s_driver);
> +}
> +module_exit(psc_i2s_exit);
> +
> +
> 
> _______________________________________________
> Linuxppc-dev mailing list
> Linuxppc-dev@ozlabs.org
> https://ozlabs.org/mailman/listinfo/linuxppc-dev
> 
> 

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[PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

From: Grant Likely <grant.likely@secretlab.ca>

ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
v1 API, so I don't expect it to get merged as-is, but I want to get it
out there for review.
---

 sound/soc/codecs/Kconfig       |    4 
 sound/soc/codecs/Makefile      |    2 
 sound/soc/codecs/tlv320aic26.c |  630 ++++++++++++++++++++++++++++++++++++++++
 3 files changed, 636 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3903ab7..96c7bfe 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA
 	bool
 	depends on SND_SOC_CS4270
 
+config SND_SOC_TLV320AIC26
+	tristate "TI TLB320AIC26 Codec support"
+	depends on SND_SOC && SPI
+
 config SND_SOC_TLV320AIC3X
 	tristate
 	depends on SND_SOC && I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c..ec0cd93 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 snd-soc-cs4270-objs := cs4270.o
+snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
@@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
new file mode 100644
index 0000000..aee1dbc
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -0,0 +1,630 @@
+/*
+ * Texas Instruments TLV320AIC26 low power audio CODEC
+ * ALSA SoC CODEC driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/device.h>
+#include <linux/sysfs.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc-of.h>
+#include <sound/initval.h>
+
+MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_LICENSE("GPL");
+
+/* AIC26 Registers */
+#define AIC26_READ_COMMAND_WORD(addr)	((1 << 15) | (addr << 5))
+#define AIC26_WRITE_COMMAND_WORD(addr)	((0 << 15) | (addr << 5))
+#define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
+#define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
+#define AIC26_REG_CACHE_SIZE		(0x20) /* only page 2 cached */
+#define AIC26_REG_IS_CACHED(addr)	((addr & ~0x1f) == (2 << 6))
+#define AIC26_REG_CACHE_ADDR(addr)	(addr & 0x1f)
+
+/* Page 0: Auxillary data registers */
+#define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
+#define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
+#define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
+#define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
+#define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
+
+/* Page 1: Auxillary control registers */
+#define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
+#define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
+#define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
+#define AIC26_REG_RESET			AIC26_PAGE_ADDR(1, 0x04)
+
+/* Page 2: Audio control registers */
+#define AIC26_REG_AUDIO_CTRL1		AIC26_PAGE_ADDR(2, 0x00)
+#define AIC26_REG_ADC_GAIN		AIC26_PAGE_ADDR(2, 0x01)
+#define AIC26_REG_DAC_GAIN		AIC26_PAGE_ADDR(2, 0x02)
+#define AIC26_REG_SIDETONE		AIC26_PAGE_ADDR(2, 0x03)
+#define AIC26_REG_AUDIO_CTRL2		AIC26_PAGE_ADDR(2, 0x04)
+#define AIC26_REG_POWER_CTRL		AIC26_PAGE_ADDR(2, 0x05)
+#define AIC26_REG_AUDIO_CTRL3		AIC26_PAGE_ADDR(2, 0x06)
+
+#define AIC26_REG_FILTER_COEFF_L_N0	AIC26_PAGE_ADDR(2, 0x07)
+#define AIC26_REG_FILTER_COEFF_L_N1	AIC26_PAGE_ADDR(2, 0x08)
+#define AIC26_REG_FILTER_COEFF_L_N2	AIC26_PAGE_ADDR(2, 0x09)
+#define AIC26_REG_FILTER_COEFF_L_N3	AIC26_PAGE_ADDR(2, 0x0A)
+#define AIC26_REG_FILTER_COEFF_L_N4	AIC26_PAGE_ADDR(2, 0x0B)
+#define AIC26_REG_FILTER_COEFF_L_N5	AIC26_PAGE_ADDR(2, 0x0C)
+#define AIC26_REG_FILTER_COEFF_L_D1	AIC26_PAGE_ADDR(2, 0x0D)
+#define AIC26_REG_FILTER_COEFF_L_D2	AIC26_PAGE_ADDR(2, 0x0E)
+#define AIC26_REG_FILTER_COEFF_L_D4	AIC26_PAGE_ADDR(2, 0x0F)
+#define AIC26_REG_FILTER_COEFF_L_D5	AIC26_PAGE_ADDR(2, 0x10)
+#define AIC26_REG_FILTER_COEFF_R_N0	AIC26_PAGE_ADDR(2, 0x11)
+#define AIC26_REG_FILTER_COEFF_R_N1	AIC26_PAGE_ADDR(2, 0x12)
+#define AIC26_REG_FILTER_COEFF_R_N2	AIC26_PAGE_ADDR(2, 0x13)
+#define AIC26_REG_FILTER_COEFF_R_N3	AIC26_PAGE_ADDR(2, 0x14)
+#define AIC26_REG_FILTER_COEFF_R_N4	AIC26_PAGE_ADDR(2, 0x15)
+#define AIC26_REG_FILTER_COEFF_R_N5	AIC26_PAGE_ADDR(2, 0x16)
+#define AIC26_REG_FILTER_COEFF_R_D1	AIC26_PAGE_ADDR(2, 0x17)
+#define AIC26_REG_FILTER_COEFF_R_D2	AIC26_PAGE_ADDR(2, 0x18)
+#define AIC26_REG_FILTER_COEFF_R_D4	AIC26_PAGE_ADDR(2, 0x19)
+#define AIC26_REG_FILTER_COEFF_R_D5	AIC26_PAGE_ADDR(2, 0x1A)
+
+#define AIC26_REG_PLL_PROG1		AIC26_PAGE_ADDR(2, 0x1B)
+#define AIC26_REG_PLL_PROG2		AIC26_PAGE_ADDR(2, 0x1C)
+#define AIC26_REG_AUDIO_CTRL4		AIC26_PAGE_ADDR(2, 0x1D)
+#define AIC26_REG_AUDIO_CTRL5		AIC26_PAGE_ADDR(2, 0x1E)
+
+#define AIC26_RATES	(SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
+			 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+			 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+			 SNDRV_PCM_RATE_48000)
+#define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
+			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+/* fsref dividers; used in register 'Audio Control 1' */
+enum aic26_divisors {
+	AIC26_DIV_1	= 0,
+	AIC26_DIV_1_5	= 1,
+	AIC26_DIV_2	= 2,
+	AIC26_DIV_3	= 3,
+	AIC26_DIV_4	= 4,
+	AIC26_DIV_5	= 5,
+	AIC26_DIV_5_5	= 6,
+	AIC26_DIV_6	= 7,
+};
+
+/* Digital data format */
+enum aic26_datfm {
+	AIC26_DATFM_I2S		= 0 << 8,
+	AIC26_DATFM_DSP		= 1 << 8,
+	AIC26_DATFM_RIGHTJ	= 2 << 8, /* right justified */
+	AIC26_DATFM_LEFTJ	= 3 << 8, /* left justified */
+};
+
+/* Sample word length in bits; used in register 'Audio Control 1' */
+enum aic26_wlen {
+	AIC26_WLEN_16	= 0 << 10,
+	AIC26_WLEN_20	= 1 << 10,
+	AIC26_WLEN_24	= 2 << 10,
+	AIC26_WLEN_32	= 3 << 10,
+};
+
+/* AIC26 driver private data */
+struct aic26 {
+	struct spi_device *spi;
+	struct snd_soc_codec codec;
+	u16 reg_cache[AIC26_REG_CACHE_SIZE];	/* shadow registers */
+	int master;
+	int datfm;
+	int mclk;
+
+	/* Keyclick parameters */
+	int keyclick_amplitude;
+	int keyclick_freq;
+	int keyclick_len;
+};
+
+/* ---------------------------------------------------------------------
+ * Register access routines
+ */
+static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
+				   unsigned int reg)
+{
+	struct aic26 *aic26 = codec->private_data;
+	u16 *cache = codec->reg_cache;
+	u16 cmd, value;
+	u8 buffer[2];
+	int rc;
+
+	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
+		WARN_ON_ONCE(1);
+		return 0;
+	}
+
+	/* Do SPI transfer; first 16bits are command; remaining is
+	 * register contents */
+	cmd = AIC26_READ_COMMAND_WORD(reg);
+	buffer[0] = (cmd >> 8) & 0xff;
+	buffer[1] = cmd & 0xff;
+	rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
+	if (rc) {
+		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
+		return -EIO;
+	}
+	value = (buffer[0] << 8) | buffer[1];
+
+	/* Update the cache before returning with the value */
+	if (AIC26_REG_IS_CACHED(reg))
+		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
+	return value;
+}
+
+static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
+					 unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
+		WARN_ON_ONCE(1);
+		return 0;
+	}
+
+	if (AIC26_REG_IS_CACHED(reg))
+		return cache[AIC26_REG_CACHE_ADDR(reg)];
+
+	return aic26_reg_read(codec, reg);
+}
+
+static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
+		           unsigned int value)
+{
+	struct aic26 *aic26 = codec->private_data;
+	u16 *cache = codec->reg_cache;
+	u16 cmd;
+	u8 buffer[4];
+	int rc;
+
+	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
+		WARN_ON_ONCE(1);
+		return -EINVAL;
+	}
+
+	/* Do SPI transfer; first 16bits are command; remaining is data
+	 * to write into register */
+	cmd = AIC26_WRITE_COMMAND_WORD(reg);
+	buffer[0] = (cmd >> 8) & 0xff;
+	buffer[1] = cmd & 0xff;
+	buffer[2] = value >> 8;
+	buffer[3] = value;
+	rc = spi_write(aic26->spi, buffer, 4);
+	if (rc) {
+		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
+		return -EIO;
+	}
+
+	/* update cache before returning */
+	if (AIC26_REG_IS_CACHED(reg))
+		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
+	return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * Digital Audio Interface Operations
+ */
+static int aic26_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct aic26 *aic26 = codec->private_data;
+	int fsref, divisor, wlen, pval, jval, dval, qval;
+	u16 reg;
+
+	dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
+		substream, params);
+	dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
+		params_format(params));
+
+	switch (params_rate(params)) {
+	 case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
+	 case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
+	 case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
+	 case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
+	 case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
+	 case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
+	 case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
+	 case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
+	 case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
+	 default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
+	}
+
+	/* select data word length */
+	switch (params_format(params)) {
+	 case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
+	 case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
+	 case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
+	 case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
+	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
+	}
+
+	/* Configure PLL */
+	pval = 1;
+	jval = (fsref == 44100) ? 7 : 8;
+	dval = (fsref == 44100) ? 5264 : 1920;
+	qval = 0;
+	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
+	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
+	reg = dval << 2;
+	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
+
+	/* Power up CODEC */
+	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
+
+	/* Audio Control 3 (master mode, fsref rate) */
+	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
+	reg &= ~0xf800;
+	if (aic26->master)
+		reg |= 0x0800;
+	if (fsref == 48000)
+		reg |= 0x2000;
+	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+
+	/* Audio Control 1 (FSref divisor) */
+	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
+	reg &= ~0x0fff;
+	reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
+	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
+
+	return 0;
+}
+
+/**
+ * aic26_mute - Mute control to reduce noise when changing audio format
+ */
+static int aic26_mute(struct snd_soc_codec_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct aic26 *aic26 = codec->private_data;
+	u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
+
+	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
+		dai, mute);
+
+	if (mute)
+		reg |= 0x8080;
+	else
+		reg &= ~0x8080;
+	aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
+
+	return 0;
+}
+
+static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
+			    int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic26 *aic26 = codec->private_data;
+
+	dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
+		" freq=%i, dir=%i)\n",
+		codec_dai, clk_id, freq, dir);
+
+	/* MCLK needs to fall between 2MHz and 50 MHz */
+	if ((freq < 2000000) || (freq > 50000000))
+		return -EINVAL;
+
+	aic26->mclk = freq;
+	return 0;
+}
+
+static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic26 *aic26 = codec->private_data;
+
+	dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
+		codec_dai, fmt);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	 case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
+	 //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
+	 default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	 case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
+	 case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
+	 case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
+	 case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
+	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
+	}
+
+	return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * Digital Audio Interface Definition
+ */
+struct snd_soc_codec_dai aic26_dai = {
+	.name = "tlv320aic26",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = AIC26_RATES,
+		.formats = AIC26_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = AIC26_RATES,
+		.formats = AIC26_FORMATS,
+	},
+	.ops = {
+		.hw_params = aic26_hw_params,
+	},
+	.dai_ops = {
+		.digital_mute = aic26_mute,
+		.set_sysclk = aic26_set_sysclk,
+		.set_fmt = aic26_set_fmt,
+	},
+};
+EXPORT_SYMBOL_GPL(aic26_dai);
+
+/* ---------------------------------------------------------------------
+ * ALSA controls
+ */
+static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
+static const struct soc_enum aic26_capture_src_enum =
+	SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
+
+static const struct snd_kcontrol_new aic26_snd_controls[] = {
+	/* Output */
+	SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
+	SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
+	SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
+	SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
+	SOC_ENUM("Capture Source", aic26_capture_src_enum),
+};
+
+/* ---------------------------------------------------------------------
+ * SoC CODEC portion of driver: probe and release routines
+ */
+static int aic26_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct snd_kcontrol *kcontrol;
+	struct aic26 *aic26;
+	int i, ret, err;
+
+	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
+	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
+	dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
+
+	/* Fetch the relevant aic26 private data here (it's already been
+	 * stored in the .codec pointer) */
+	aic26 = socdev->codec_data;
+	if (aic26 == NULL) {
+		dev_err(&pdev->dev, "aic26: missing codec pointer\n");
+		return -ENODEV;
+	}
+	codec = &aic26->codec;
+	socdev->codec = codec;
+
+	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
+		&pdev->dev, socdev->dev);
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "aic26: failed to create pcms\n");
+		return -ENODEV;
+	}
+
+	/* register controls */
+	dev_dbg(&pdev->dev, "Registering controls\n");
+	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
+		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
+		err = snd_ctl_add(codec->card, kcontrol);
+		WARN_ON(err < 0);
+	}
+
+	/* CODEC is setup, we can register the card now */
+	dev_dbg(&pdev->dev, "Registering card\n");
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "aic26: failed to register card\n");
+		goto card_err;
+	}
+	return 0;
+
+ card_err:
+	snd_soc_free_pcms(socdev);
+	return ret;
+}
+
+static int aic26_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	snd_soc_free_pcms(socdev);
+	return 0;
+}
+
+struct snd_soc_codec_device aic26_soc_codec_dev = {
+	.probe = aic26_probe,
+	.remove = aic26_remove,
+};
+
+/* ---------------------------------------------------------------------
+ * SPI device portion of driver: sysfs files for debugging
+ */
+
+static ssize_t aic26_regs_show(struct device *dev,
+				struct device_attribute *attr, char *buf)
+{
+	struct aic26 *aic26 = dev_get_drvdata(dev);
+	char *idx = buf;
+	int cache_flag, addr, page, i, reg;
+
+	cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
+
+	for (page = 0; page < 3; page++) {
+		for (i = 0; i < 0x20; i++) {
+			addr = AIC26_PAGE_ADDR(page, i);
+			if (i % 8 == 0)
+				idx += sprintf(idx, "%i:%.2i:", page,i);
+			if (cache_flag)
+				reg = aic26_reg_read_cache(&aic26->codec, addr);
+			else
+				reg = aic26_reg_read(&aic26->codec, addr);
+			idx += sprintf(idx, " %.4x", reg);
+			if (i % 8 == 7)
+				idx += sprintf(idx, "\n");
+		}
+	}
+	return idx - buf;
+}
+
+static ssize_t aic26_keyclick_show(struct device *dev,
+				   struct device_attribute *attr, char *buf)
+{
+	struct aic26 *aic26 = dev_get_drvdata(dev);
+	int val, amp, freq, len;
+
+	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+	amp = (val >> 12) & 0x7;
+	freq = (125 << ((val >> 8) & 0x7)) >> 1;
+	len = 2 * (1 +((val >> 8) & 0xf));
+
+	return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
+}
+
+/* Any write to the keyclick attribute will trigger the keyclick */
+static ssize_t aic26_keyclick_set(struct device *dev,
+				  struct device_attribute *attr,
+				  const char *buf, size_t count)
+{
+	struct aic26 *aic26 = dev_get_drvdata(dev);
+	int val;
+
+	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+	val |= 0x8000;
+	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
+
+	return count;
+}
+
+DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL);
+DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL);
+DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
+
+/* ---------------------------------------------------------------------
+ * SPI device portion of driver: probe and release routines and SPI
+ * 				 driver registration.
+ */
+static int aic26_spi_probe(struct spi_device *spi)
+{
+	struct aic26 *aic26;
+	int rc, i, reg;
+
+	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
+
+	/* Allocate driver data */
+	aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
+	if (!aic26)
+		return -ENOMEM;
+
+	/* Initialize the driver data */
+	aic26->spi = spi;
+	dev_set_drvdata(&spi->dev, aic26);
+
+	/* Setup what we can in the codec structure so that the register
+	 * access functions will work as expected.  More will be filled
+	 * out when it is probed by the SoC CODEC part of this driver */
+	aic26->codec.private_data = aic26;
+	aic26->codec.name = "aic26";
+	aic26->codec.owner = THIS_MODULE;
+	aic26->codec.dai = &aic26_dai;
+	aic26->codec.num_dai = 1;
+	aic26->codec.read = aic26_reg_read;
+	aic26->codec.write = aic26_reg_write;
+	aic26->master = 1;
+	mutex_init(&aic26->codec.mutex);
+	INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
+	INIT_LIST_HEAD(&aic26->codec.dapm_paths);
+	aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
+	aic26->codec.reg_cache = aic26->reg_cache;
+
+	/* Reset the codec to power on defaults */
+	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
+
+	/* Power up CODEC */
+	aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
+
+	/* Audio Control 3 (master mode, fsref rate) */
+	reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
+	reg &= ~0xf800;
+	reg |= 0x0800; /* set master mode */
+	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
+
+	/* Fill page 2 register cache */
+	for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
+		aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));
+
+	/* Register the sysfs files for debugging */
+	/* Create SysFS files */
+	rc = device_create_file(&spi->dev, &dev_attr_regs);
+	rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
+	rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
+	if (rc)
+		dev_info(&spi->dev, "error creating sysfs files\n");
+
+	/* Tell the of_soc helper about this codec */
+	of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
+				  spi->dev.archdata.of_node);
+
+	dev_dbg(&spi->dev, "SPI device initialized\n");
+	return 0;
+}
+
+static int aic26_spi_remove(struct spi_device *spi)
+{
+	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
+
+	kfree(aic26);
+
+	return 0;
+}
+
+static struct spi_driver aic26_spi = {
+	.driver = {
+		.name = "tlv320aic26",
+		.owner = THIS_MODULE,
+	},
+	.probe = aic26_spi_probe,
+	.remove = aic26_spi_remove,
+};
+
+static int __init aic26_init(void)
+{
+	return spi_register_driver(&aic26_spi);
+}
+module_init(aic26_init);
+
+static void __exit aic26_exit(void)
+{
+	spi_unregister_driver(&aic26_spi);
+}
+module_exit(aic26_exit);

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Liam Girdwood]

On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
> 
> ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
> v1 API, so I don't expect it to get merged as-is, but I want to get it
> out there for review.
> ---
> 
>  sound/soc/codecs/Kconfig       |    4 
>  sound/soc/codecs/Makefile      |    2 
>  sound/soc/codecs/tlv320aic26.c |  630 ++++++++++++++++++++++++++++++++++++++++
>  3 files changed, 636 insertions(+), 0 deletions(-)
> 
> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
> index 3903ab7..96c7bfe 100644
> --- a/sound/soc/codecs/Kconfig
> +++ b/sound/soc/codecs/Kconfig
> @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA
>  	bool
>  	depends on SND_SOC_CS4270
>  
> +config SND_SOC_TLV320AIC26
> +	tristate "TI TLB320AIC26 Codec support"
> +	depends on SND_SOC && SPI
> +
>  config SND_SOC_TLV320AIC3X
>  	tristate
>  	depends on SND_SOC && I2C
> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
> index 4e1314c..ec0cd93 100644
> --- a/sound/soc/codecs/Makefile
> +++ b/sound/soc/codecs/Makefile
> @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o
>  snd-soc-wm9712-objs := wm9712.o
>  snd-soc-wm9713-objs := wm9713.o
>  snd-soc-cs4270-objs := cs4270.o
> +snd-soc-tlv320aic26-objs := tlv320aic26.o
>  snd-soc-tlv320aic3x-objs := tlv320aic3x.o
>  
>  obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
> @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
>  obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
>  obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
>  obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
> +obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
>  obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
> diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
> new file mode 100644
> index 0000000..aee1dbc
> --- /dev/null
> +++ b/sound/soc/codecs/tlv320aic26.c
> @@ -0,0 +1,630 @@
> +/*
> + * Texas Instruments TLV320AIC26 low power audio CODEC
> + * ALSA SoC CODEC driver
> + *
> + * Copyright (C) 2008 Secret Lab Technologies Ltd.
> + */
> +
> +#include <linux/module.h>
> +#include <linux/moduleparam.h>
> +#include <linux/init.h>
> +#include <linux/delay.h>
> +#include <linux/pm.h>
> +#include <linux/device.h>
> +#include <linux/sysfs.h>
> +#include <linux/spi/spi.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <sound/soc-dapm.h>
> +#include <sound/soc-of.h>
> +#include <sound/initval.h>
> +
> +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
> +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
> +MODULE_LICENSE("GPL");
> +
> +/* AIC26 Registers */
> +#define AIC26_READ_COMMAND_WORD(addr)	((1 << 15) | (addr << 5))
> +#define AIC26_WRITE_COMMAND_WORD(addr)	((0 << 15) | (addr << 5))
> +#define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
> +#define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
> +#define AIC26_REG_CACHE_SIZE		(0x20) /* only page 2 cached */
> +#define AIC26_REG_IS_CACHED(addr)	((addr & ~0x1f) == (2 << 6))
> +#define AIC26_REG_CACHE_ADDR(addr)	(addr & 0x1f)
> +
> +/* Page 0: Auxillary data registers */
> +#define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
> +#define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
> +#define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
> +#define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
> +#define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
> +
> +/* Page 1: Auxillary control registers */
> +#define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
> +#define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
> +#define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
> +#define AIC26_REG_RESET			AIC26_PAGE_ADDR(1, 0x04)
> +
> +/* Page 2: Audio control registers */
> +#define AIC26_REG_AUDIO_CTRL1		AIC26_PAGE_ADDR(2, 0x00)
> +#define AIC26_REG_ADC_GAIN		AIC26_PAGE_ADDR(2, 0x01)
> +#define AIC26_REG_DAC_GAIN		AIC26_PAGE_ADDR(2, 0x02)
> +#define AIC26_REG_SIDETONE		AIC26_PAGE_ADDR(2, 0x03)
> +#define AIC26_REG_AUDIO_CTRL2		AIC26_PAGE_ADDR(2, 0x04)
> +#define AIC26_REG_POWER_CTRL		AIC26_PAGE_ADDR(2, 0x05)
> +#define AIC26_REG_AUDIO_CTRL3		AIC26_PAGE_ADDR(2, 0x06)
> +
> +#define AIC26_REG_FILTER_COEFF_L_N0	AIC26_PAGE_ADDR(2, 0x07)
> +#define AIC26_REG_FILTER_COEFF_L_N1	AIC26_PAGE_ADDR(2, 0x08)
> +#define AIC26_REG_FILTER_COEFF_L_N2	AIC26_PAGE_ADDR(2, 0x09)
> +#define AIC26_REG_FILTER_COEFF_L_N3	AIC26_PAGE_ADDR(2, 0x0A)
> +#define AIC26_REG_FILTER_COEFF_L_N4	AIC26_PAGE_ADDR(2, 0x0B)
> +#define AIC26_REG_FILTER_COEFF_L_N5	AIC26_PAGE_ADDR(2, 0x0C)
> +#define AIC26_REG_FILTER_COEFF_L_D1	AIC26_PAGE_ADDR(2, 0x0D)
> +#define AIC26_REG_FILTER_COEFF_L_D2	AIC26_PAGE_ADDR(2, 0x0E)
> +#define AIC26_REG_FILTER_COEFF_L_D4	AIC26_PAGE_ADDR(2, 0x0F)
> +#define AIC26_REG_FILTER_COEFF_L_D5	AIC26_PAGE_ADDR(2, 0x10)
> +#define AIC26_REG_FILTER_COEFF_R_N0	AIC26_PAGE_ADDR(2, 0x11)
> +#define AIC26_REG_FILTER_COEFF_R_N1	AIC26_PAGE_ADDR(2, 0x12)
> +#define AIC26_REG_FILTER_COEFF_R_N2	AIC26_PAGE_ADDR(2, 0x13)
> +#define AIC26_REG_FILTER_COEFF_R_N3	AIC26_PAGE_ADDR(2, 0x14)
> +#define AIC26_REG_FILTER_COEFF_R_N4	AIC26_PAGE_ADDR(2, 0x15)
> +#define AIC26_REG_FILTER_COEFF_R_N5	AIC26_PAGE_ADDR(2, 0x16)
> +#define AIC26_REG_FILTER_COEFF_R_D1	AIC26_PAGE_ADDR(2, 0x17)
> +#define AIC26_REG_FILTER_COEFF_R_D2	AIC26_PAGE_ADDR(2, 0x18)
> +#define AIC26_REG_FILTER_COEFF_R_D4	AIC26_PAGE_ADDR(2, 0x19)
> +#define AIC26_REG_FILTER_COEFF_R_D5	AIC26_PAGE_ADDR(2, 0x1A)
> +
> +#define AIC26_REG_PLL_PROG1		AIC26_PAGE_ADDR(2, 0x1B)
> +#define AIC26_REG_PLL_PROG2		AIC26_PAGE_ADDR(2, 0x1C)
> +#define AIC26_REG_AUDIO_CTRL4		AIC26_PAGE_ADDR(2, 0x1D)
> +#define AIC26_REG_AUDIO_CTRL5		AIC26_PAGE_ADDR(2, 0x1E)
> +
> +#define AIC26_RATES	(SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
> +			 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
> +			 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
> +			 SNDRV_PCM_RATE_48000)
> +#define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
> +			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
> +

Fwiw, I usually put all the codec registers defs in a separate header
just in case we need to do any codec stuff in the machine driver.

> +/* fsref dividers; used in register 'Audio Control 1' */
> +enum aic26_divisors {
> +	AIC26_DIV_1	= 0,
> +	AIC26_DIV_1_5	= 1,
> +	AIC26_DIV_2	= 2,
> +	AIC26_DIV_3	= 3,
> +	AIC26_DIV_4	= 4,
> +	AIC26_DIV_5	= 5,
> +	AIC26_DIV_5_5	= 6,
> +	AIC26_DIV_6	= 7,
> +};
> +
> +/* Digital data format */
> +enum aic26_datfm {
> +	AIC26_DATFM_I2S		= 0 << 8,
> +	AIC26_DATFM_DSP		= 1 << 8,
> +	AIC26_DATFM_RIGHTJ	= 2 << 8, /* right justified */
> +	AIC26_DATFM_LEFTJ	= 3 << 8, /* left justified */
> +};
> +
> +/* Sample word length in bits; used in register 'Audio Control 1' */
> +enum aic26_wlen {
> +	AIC26_WLEN_16	= 0 << 10,
> +	AIC26_WLEN_20	= 1 << 10,
> +	AIC26_WLEN_24	= 2 << 10,
> +	AIC26_WLEN_32	= 3 << 10,
> +};
> +
> +/* AIC26 driver private data */
> +struct aic26 {
> +	struct spi_device *spi;
> +	struct snd_soc_codec codec;
> +	u16 reg_cache[AIC26_REG_CACHE_SIZE];	/* shadow registers */
> +	int master;
> +	int datfm;
> +	int mclk;
> +
> +	/* Keyclick parameters */
> +	int keyclick_amplitude;
> +	int keyclick_freq;
> +	int keyclick_len;
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Register access routines
> + */
> +static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
> +				   unsigned int reg)
> +{
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 *cache = codec->reg_cache;
> +	u16 cmd, value;
> +	u8 buffer[2];
> +	int rc;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return 0;
> +	}
> +
> +	/* Do SPI transfer; first 16bits are command; remaining is
> +	 * register contents */
> +	cmd = AIC26_READ_COMMAND_WORD(reg);
> +	buffer[0] = (cmd >> 8) & 0xff;
> +	buffer[1] = cmd & 0xff;
> +	rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
> +	if (rc) {
> +		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
> +		return -EIO;
> +	}
> +	value = (buffer[0] << 8) | buffer[1];
> +
> +	/* Update the cache before returning with the value */
> +	if (AIC26_REG_IS_CACHED(reg))
> +		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
> +	return value;
> +}
> +
> +static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
> +					 unsigned int reg)
> +{
> +	u16 *cache = codec->reg_cache;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return 0;
> +	}
> +
> +	if (AIC26_REG_IS_CACHED(reg))
> +		return cache[AIC26_REG_CACHE_ADDR(reg)];
> +
> +	return aic26_reg_read(codec, reg);
> +}
> +
> +static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
> +		           unsigned int value)
> +{
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 *cache = codec->reg_cache;
> +	u16 cmd;
> +	u8 buffer[4];
> +	int rc;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return -EINVAL;
> +	}
> +
> +	/* Do SPI transfer; first 16bits are command; remaining is data
> +	 * to write into register */
> +	cmd = AIC26_WRITE_COMMAND_WORD(reg);
> +	buffer[0] = (cmd >> 8) & 0xff;
> +	buffer[1] = cmd & 0xff;
> +	buffer[2] = value >> 8;
> +	buffer[3] = value;
> +	rc = spi_write(aic26->spi, buffer, 4);
> +	if (rc) {
> +		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
> +		return -EIO;
> +	}
> +
> +	/* update cache before returning */
> +	if (AIC26_REG_IS_CACHED(reg))
> +		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
> +	return 0;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * Digital Audio Interface Operations
> + */
> +static int aic26_hw_params(struct snd_pcm_substream *substream,
> +			   struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct snd_soc_device *socdev = rtd->socdev;
> +	struct snd_soc_codec *codec = socdev->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +	int fsref, divisor, wlen, pval, jval, dval, qval;
> +	u16 reg;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
> +		substream, params);
> +	dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
> +		params_format(params));
> +
> +	switch (params_rate(params)) {
> +	 case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
> +	 case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
> +	 case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
> +	 case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
> +	 case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
> +	 case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
> +	 case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
> +	 case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
> +	 case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
> +	}
> +

Indentation.

> +	/* select data word length */
> +	switch (params_format(params)) {
> +	 case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
> +	 case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
> +	 case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
> +	 case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
> +	}
> +

ditto.

> +	/* Configure PLL */
> +	pval = 1;
> +	jval = (fsref == 44100) ? 7 : 8;
> +	dval = (fsref == 44100) ? 5264 : 1920;
> +	qval = 0;
> +	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
> +	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
> +	reg = dval << 2;
> +	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
> +

PLL/FLL/clock config is usually done in a separate function
(codec_set_pll(), callable by machine driver) so that we can change
clocks depending on the available machine clocks and srate. 

> +	/* Power up CODEC */
> +	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
> +

Codec domain (i.e Bias power) PM stuff should be done in
codec_dapm_event(). This allows us to power the codec on when we do
things like sidetone (with no active playback or capture stream).

> +	/* Audio Control 3 (master mode, fsref rate) */
> +	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
> +	reg &= ~0xf800;
> +	if (aic26->master)
> +		reg |= 0x0800;
> +	if (fsref == 48000)
> +		reg |= 0x2000;
> +	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
> +
> +	/* Audio Control 1 (FSref divisor) */
> +	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
> +	reg &= ~0x0fff;
> +	reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
> +	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
> +
> +	return 0;
> +}
> +
> +/**
> + * aic26_mute - Mute control to reduce noise when changing audio format
> + */
> +static int aic26_mute(struct snd_soc_codec_dai *dai, int mute)
> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
> +		dai, mute);
> +
> +	if (mute)
> +		reg |= 0x8080;
> +	else
> +		reg &= ~0x8080;
> +	aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
> +
> +	return 0;
> +}
> +
> +static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
> +			    int clk_id, unsigned int freq, int dir)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
> +		" freq=%i, dir=%i)\n",
> +		codec_dai, clk_id, freq, dir);
> +
> +	/* MCLK needs to fall between 2MHz and 50 MHz */
> +	if ((freq < 2000000) || (freq > 50000000))
> +		return -EINVAL;
> +
> +	aic26->mclk = freq;
> +	return 0;
> +}
> +
> +static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
> +		codec_dai, fmt);
> +
> +	/* set master/slave audio interface */
> +	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> +	 case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
> +	 //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
> +	}
> +
> +	/* interface format */
> +	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> +	 case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
> +	 case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
> +	 case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
> +	 case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
> +	}
> +
> +	return 0;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * Digital Audio Interface Definition
> + */
> +struct snd_soc_codec_dai aic26_dai = {
> +	.name = "tlv320aic26",
> +	.playback = {
> +		.stream_name = "Playback",
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = AIC26_RATES,
> +		.formats = AIC26_FORMATS,
> +	},
> +	.capture = {
> +		.stream_name = "Capture",
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = AIC26_RATES,
> +		.formats = AIC26_FORMATS,
> +	},
> +	.ops = {
> +		.hw_params = aic26_hw_params,
> +	},
> +	.dai_ops = {
> +		.digital_mute = aic26_mute,
> +		.set_sysclk = aic26_set_sysclk,
> +		.set_fmt = aic26_set_fmt,
> +	},
> +};
> +EXPORT_SYMBOL_GPL(aic26_dai);
> +
> +/* ---------------------------------------------------------------------
> + * ALSA controls
> + */
> +static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
> +static const struct soc_enum aic26_capture_src_enum =
> +	SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
> +
> +static const struct snd_kcontrol_new aic26_snd_controls[] = {
> +	/* Output */
> +	SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
> +	SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
> +	SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
> +	SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
> +	SOC_ENUM("Capture Source", aic26_capture_src_enum),
> +};
> +
> +/* ---------------------------------------------------------------------
> + * SoC CODEC portion of driver: probe and release routines
> + */
> +static int aic26_probe(struct platform_device *pdev)
> +{
> +	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
> +	struct snd_soc_codec *codec;
> +	struct snd_kcontrol *kcontrol;
> +	struct aic26 *aic26;
> +	int i, ret, err;
> +
> +	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
> +	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
> +	dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
> +
> +	/* Fetch the relevant aic26 private data here (it's already been
> +	 * stored in the .codec pointer) */
> +	aic26 = socdev->codec_data;
> +	if (aic26 == NULL) {
> +		dev_err(&pdev->dev, "aic26: missing codec pointer\n");
> +		return -ENODEV;
> +	}
> +	codec = &aic26->codec;
> +	socdev->codec = codec;
> +
> +	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
> +		&pdev->dev, socdev->dev);
> +	/* register pcms */
> +	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
> +	if (ret < 0) {
> +		dev_err(&pdev->dev, "aic26: failed to create pcms\n");
> +		return -ENODEV;
> +	}
> +
> +	/* register controls */
> +	dev_dbg(&pdev->dev, "Registering controls\n");
> +	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
> +		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
> +		err = snd_ctl_add(codec->card, kcontrol);
> +		WARN_ON(err < 0);
> +	}
> +
> +	/* CODEC is setup, we can register the card now */
> +	dev_dbg(&pdev->dev, "Registering card\n");
> +	ret = snd_soc_register_card(socdev);
> +	if (ret < 0) {
> +		dev_err(&pdev->dev, "aic26: failed to register card\n");
> +		goto card_err;
> +	}
> +	return 0;
> +
> + card_err:
> +	snd_soc_free_pcms(socdev);
> +	return ret;
> +}
> +
> +static int aic26_remove(struct platform_device *pdev)
> +{
> +	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
> +	snd_soc_free_pcms(socdev);
> +	return 0;
> +}
> +
> +struct snd_soc_codec_device aic26_soc_codec_dev = {
> +	.probe = aic26_probe,
> +	.remove = aic26_remove,
> +};
> +
> +/* ---------------------------------------------------------------------
> + * SPI device portion of driver: sysfs files for debugging
> + */
> +
> +static ssize_t aic26_regs_show(struct device *dev,
> +				struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	char *idx = buf;
> +	int cache_flag, addr, page, i, reg;
> +
> +	cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
> +
> +	for (page = 0; page < 3; page++) {
> +		for (i = 0; i < 0x20; i++) {
> +			addr = AIC26_PAGE_ADDR(page, i);
> +			if (i % 8 == 0)
> +				idx += sprintf(idx, "%i:%.2i:", page,i);
> +			if (cache_flag)
> +				reg = aic26_reg_read_cache(&aic26->codec, addr);
> +			else
> +				reg = aic26_reg_read(&aic26->codec, addr);
> +			idx += sprintf(idx, " %.4x", reg);
> +			if (i % 8 == 7)
> +				idx += sprintf(idx, "\n");
> +		}
> +	}
> +	return idx - buf;
> +}
> +

The soc_core already has a codec reg dump sysfs file, so we don't need
this.

> +static ssize_t aic26_keyclick_show(struct device *dev,
> +				   struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val, amp, freq, len;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	amp = (val >> 12) & 0x7;
> +	freq = (125 << ((val >> 8) & 0x7)) >> 1;
> +	len = 2 * (1 +((val >> 8) & 0xf));
> +
> +	return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
> +}
> +
> +/* Any write to the keyclick attribute will trigger the keyclick */
> +static ssize_t aic26_keyclick_set(struct device *dev,
> +				  struct device_attribute *attr,
> +				  const char *buf, size_t count)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	val |= 0x8000;
> +	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
> +
> +	return count;
> +}
> +
> +DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL);
> +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL);
> +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
> +
> +/* ---------------------------------------------------------------------
> + * SPI device portion of driver: probe and release routines and SPI
> + * 				 driver registration.
> + */
> +static int aic26_spi_probe(struct spi_device *spi)
> +{
> +	struct aic26 *aic26;
> +	int rc, i, reg;
> +
> +	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
> +
> +	/* Allocate driver data */
> +	aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
> +	if (!aic26)
> +		return -ENOMEM;
> +
> +	/* Initialize the driver data */
> +	aic26->spi = spi;
> +	dev_set_drvdata(&spi->dev, aic26);
> +
> +	/* Setup what we can in the codec structure so that the register
> +	 * access functions will work as expected.  More will be filled
> +	 * out when it is probed by the SoC CODEC part of this driver */
> +	aic26->codec.private_data = aic26;
> +	aic26->codec.name = "aic26";
> +	aic26->codec.owner = THIS_MODULE;
> +	aic26->codec.dai = &aic26_dai;
> +	aic26->codec.num_dai = 1;
> +	aic26->codec.read = aic26_reg_read;
> +	aic26->codec.write = aic26_reg_write;
> +	aic26->master = 1;
> +	mutex_init(&aic26->codec.mutex);
> +	INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
> +	INIT_LIST_HEAD(&aic26->codec.dapm_paths);
> +	aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
> +	aic26->codec.reg_cache = aic26->reg_cache;
> +
> +	/* Reset the codec to power on defaults */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
> +
> +	/* Power up CODEC */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
> +
> +	/* Audio Control 3 (master mode, fsref rate) */
> +	reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
> +	reg &= ~0xf800;
> +	reg |= 0x0800; /* set master mode */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
> +
> +	/* Fill page 2 register cache */
> +	for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
> +		aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));
> +
> +	/* Register the sysfs files for debugging */
> +	/* Create SysFS files */
> +	rc = device_create_file(&spi->dev, &dev_attr_regs);
> +	rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
> +	rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
> +	if (rc)
> +		dev_info(&spi->dev, "error creating sysfs files\n");
> +
> +	/* Tell the of_soc helper about this codec */
> +	of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
> +				  spi->dev.archdata.of_node);
> +
> +	dev_dbg(&spi->dev, "SPI device initialized\n");
> +	return 0;
> +}
> +
> +static int aic26_spi_remove(struct spi_device *spi)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
> +
> +	kfree(aic26);
> +
> +	return 0;
> +}
> +
> +static struct spi_driver aic26_spi = {
> +	.driver = {
> +		.name = "tlv320aic26",
> +		.owner = THIS_MODULE,
> +	},
> +	.probe = aic26_spi_probe,
> +	.remove = aic26_spi_remove,
> +};
> +
> +static int __init aic26_init(void)
> +{
> +	return spi_register_driver(&aic26_spi);
> +}
> +module_init(aic26_init);
> +
> +static void __exit aic26_exit(void)
> +{
> +	spi_unregister_driver(&aic26_spi);
> +}
> +module_exit(aic26_exit);
> 
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel


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Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Liam Girdwood]

On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
> 
> ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
> v1 API, so I don't expect it to get merged as-is, but I want to get it
> out there for review.
> ---
> 
>  sound/soc/codecs/Kconfig       |    4 
>  sound/soc/codecs/Makefile      |    2 
>  sound/soc/codecs/tlv320aic26.c |  630 ++++++++++++++++++++++++++++++++++++++++
>  3 files changed, 636 insertions(+), 0 deletions(-)
> 
> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
> index 3903ab7..96c7bfe 100644
> --- a/sound/soc/codecs/Kconfig
> +++ b/sound/soc/codecs/Kconfig
> @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA
>  	bool
>  	depends on SND_SOC_CS4270
>  
> +config SND_SOC_TLV320AIC26
> +	tristate "TI TLB320AIC26 Codec support"
> +	depends on SND_SOC && SPI
> +
>  config SND_SOC_TLV320AIC3X
>  	tristate
>  	depends on SND_SOC && I2C
> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
> index 4e1314c..ec0cd93 100644
> --- a/sound/soc/codecs/Makefile
> +++ b/sound/soc/codecs/Makefile
> @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o
>  snd-soc-wm9712-objs := wm9712.o
>  snd-soc-wm9713-objs := wm9713.o
>  snd-soc-cs4270-objs := cs4270.o
> +snd-soc-tlv320aic26-objs := tlv320aic26.o
>  snd-soc-tlv320aic3x-objs := tlv320aic3x.o
>  
>  obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
> @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
>  obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
>  obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
>  obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
> +obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
>  obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
> diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
> new file mode 100644
> index 0000000..aee1dbc
> --- /dev/null
> +++ b/sound/soc/codecs/tlv320aic26.c
> @@ -0,0 +1,630 @@
> +/*
> + * Texas Instruments TLV320AIC26 low power audio CODEC
> + * ALSA SoC CODEC driver
> + *
> + * Copyright (C) 2008 Secret Lab Technologies Ltd.
> + */
> +
> +#include <linux/module.h>
> +#include <linux/moduleparam.h>
> +#include <linux/init.h>
> +#include <linux/delay.h>
> +#include <linux/pm.h>
> +#include <linux/device.h>
> +#include <linux/sysfs.h>
> +#include <linux/spi/spi.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <sound/soc-dapm.h>
> +#include <sound/soc-of.h>
> +#include <sound/initval.h>
> +
> +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
> +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
> +MODULE_LICENSE("GPL");
> +
> +/* AIC26 Registers */
> +#define AIC26_READ_COMMAND_WORD(addr)	((1 << 15) | (addr << 5))
> +#define AIC26_WRITE_COMMAND_WORD(addr)	((0 << 15) | (addr << 5))
> +#define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
> +#define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
> +#define AIC26_REG_CACHE_SIZE		(0x20) /* only page 2 cached */
> +#define AIC26_REG_IS_CACHED(addr)	((addr & ~0x1f) == (2 << 6))
> +#define AIC26_REG_CACHE_ADDR(addr)	(addr & 0x1f)
> +
> +/* Page 0: Auxillary data registers */
> +#define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
> +#define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
> +#define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
> +#define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
> +#define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
> +
> +/* Page 1: Auxillary control registers */
> +#define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
> +#define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
> +#define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
> +#define AIC26_REG_RESET			AIC26_PAGE_ADDR(1, 0x04)
> +
> +/* Page 2: Audio control registers */
> +#define AIC26_REG_AUDIO_CTRL1		AIC26_PAGE_ADDR(2, 0x00)
> +#define AIC26_REG_ADC_GAIN		AIC26_PAGE_ADDR(2, 0x01)
> +#define AIC26_REG_DAC_GAIN		AIC26_PAGE_ADDR(2, 0x02)
> +#define AIC26_REG_SIDETONE		AIC26_PAGE_ADDR(2, 0x03)
> +#define AIC26_REG_AUDIO_CTRL2		AIC26_PAGE_ADDR(2, 0x04)
> +#define AIC26_REG_POWER_CTRL		AIC26_PAGE_ADDR(2, 0x05)
> +#define AIC26_REG_AUDIO_CTRL3		AIC26_PAGE_ADDR(2, 0x06)
> +
> +#define AIC26_REG_FILTER_COEFF_L_N0	AIC26_PAGE_ADDR(2, 0x07)
> +#define AIC26_REG_FILTER_COEFF_L_N1	AIC26_PAGE_ADDR(2, 0x08)
> +#define AIC26_REG_FILTER_COEFF_L_N2	AIC26_PAGE_ADDR(2, 0x09)
> +#define AIC26_REG_FILTER_COEFF_L_N3	AIC26_PAGE_ADDR(2, 0x0A)
> +#define AIC26_REG_FILTER_COEFF_L_N4	AIC26_PAGE_ADDR(2, 0x0B)
> +#define AIC26_REG_FILTER_COEFF_L_N5	AIC26_PAGE_ADDR(2, 0x0C)
> +#define AIC26_REG_FILTER_COEFF_L_D1	AIC26_PAGE_ADDR(2, 0x0D)
> +#define AIC26_REG_FILTER_COEFF_L_D2	AIC26_PAGE_ADDR(2, 0x0E)
> +#define AIC26_REG_FILTER_COEFF_L_D4	AIC26_PAGE_ADDR(2, 0x0F)
> +#define AIC26_REG_FILTER_COEFF_L_D5	AIC26_PAGE_ADDR(2, 0x10)
> +#define AIC26_REG_FILTER_COEFF_R_N0	AIC26_PAGE_ADDR(2, 0x11)
> +#define AIC26_REG_FILTER_COEFF_R_N1	AIC26_PAGE_ADDR(2, 0x12)
> +#define AIC26_REG_FILTER_COEFF_R_N2	AIC26_PAGE_ADDR(2, 0x13)
> +#define AIC26_REG_FILTER_COEFF_R_N3	AIC26_PAGE_ADDR(2, 0x14)
> +#define AIC26_REG_FILTER_COEFF_R_N4	AIC26_PAGE_ADDR(2, 0x15)
> +#define AIC26_REG_FILTER_COEFF_R_N5	AIC26_PAGE_ADDR(2, 0x16)
> +#define AIC26_REG_FILTER_COEFF_R_D1	AIC26_PAGE_ADDR(2, 0x17)
> +#define AIC26_REG_FILTER_COEFF_R_D2	AIC26_PAGE_ADDR(2, 0x18)
> +#define AIC26_REG_FILTER_COEFF_R_D4	AIC26_PAGE_ADDR(2, 0x19)
> +#define AIC26_REG_FILTER_COEFF_R_D5	AIC26_PAGE_ADDR(2, 0x1A)
> +
> +#define AIC26_REG_PLL_PROG1		AIC26_PAGE_ADDR(2, 0x1B)
> +#define AIC26_REG_PLL_PROG2		AIC26_PAGE_ADDR(2, 0x1C)
> +#define AIC26_REG_AUDIO_CTRL4		AIC26_PAGE_ADDR(2, 0x1D)
> +#define AIC26_REG_AUDIO_CTRL5		AIC26_PAGE_ADDR(2, 0x1E)
> +
> +#define AIC26_RATES	(SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
> +			 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
> +			 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
> +			 SNDRV_PCM_RATE_48000)
> +#define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
> +			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
> +

Fwiw, I usually put all the codec registers defs in a separate header
just in case we need to do any codec stuff in the machine driver.

> +/* fsref dividers; used in register 'Audio Control 1' */
> +enum aic26_divisors {
> +	AIC26_DIV_1	= 0,
> +	AIC26_DIV_1_5	= 1,
> +	AIC26_DIV_2	= 2,
> +	AIC26_DIV_3	= 3,
> +	AIC26_DIV_4	= 4,
> +	AIC26_DIV_5	= 5,
> +	AIC26_DIV_5_5	= 6,
> +	AIC26_DIV_6	= 7,
> +};
> +
> +/* Digital data format */
> +enum aic26_datfm {
> +	AIC26_DATFM_I2S		= 0 << 8,
> +	AIC26_DATFM_DSP		= 1 << 8,
> +	AIC26_DATFM_RIGHTJ	= 2 << 8, /* right justified */
> +	AIC26_DATFM_LEFTJ	= 3 << 8, /* left justified */
> +};
> +
> +/* Sample word length in bits; used in register 'Audio Control 1' */
> +enum aic26_wlen {
> +	AIC26_WLEN_16	= 0 << 10,
> +	AIC26_WLEN_20	= 1 << 10,
> +	AIC26_WLEN_24	= 2 << 10,
> +	AIC26_WLEN_32	= 3 << 10,
> +};
> +
> +/* AIC26 driver private data */
> +struct aic26 {
> +	struct spi_device *spi;
> +	struct snd_soc_codec codec;
> +	u16 reg_cache[AIC26_REG_CACHE_SIZE];	/* shadow registers */
> +	int master;
> +	int datfm;
> +	int mclk;
> +
> +	/* Keyclick parameters */
> +	int keyclick_amplitude;
> +	int keyclick_freq;
> +	int keyclick_len;
> +};
> +
> +/* ---------------------------------------------------------------------
> + * Register access routines
> + */
> +static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
> +				   unsigned int reg)
> +{
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 *cache = codec->reg_cache;
> +	u16 cmd, value;
> +	u8 buffer[2];
> +	int rc;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return 0;
> +	}
> +
> +	/* Do SPI transfer; first 16bits are command; remaining is
> +	 * register contents */
> +	cmd = AIC26_READ_COMMAND_WORD(reg);
> +	buffer[0] = (cmd >> 8) & 0xff;
> +	buffer[1] = cmd & 0xff;
> +	rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
> +	if (rc) {
> +		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
> +		return -EIO;
> +	}
> +	value = (buffer[0] << 8) | buffer[1];
> +
> +	/* Update the cache before returning with the value */
> +	if (AIC26_REG_IS_CACHED(reg))
> +		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
> +	return value;
> +}
> +
> +static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
> +					 unsigned int reg)
> +{
> +	u16 *cache = codec->reg_cache;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return 0;
> +	}
> +
> +	if (AIC26_REG_IS_CACHED(reg))
> +		return cache[AIC26_REG_CACHE_ADDR(reg)];
> +
> +	return aic26_reg_read(codec, reg);
> +}
> +
> +static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
> +		           unsigned int value)
> +{
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 *cache = codec->reg_cache;
> +	u16 cmd;
> +	u8 buffer[4];
> +	int rc;
> +
> +	if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
> +		WARN_ON_ONCE(1);
> +		return -EINVAL;
> +	}
> +
> +	/* Do SPI transfer; first 16bits are command; remaining is data
> +	 * to write into register */
> +	cmd = AIC26_WRITE_COMMAND_WORD(reg);
> +	buffer[0] = (cmd >> 8) & 0xff;
> +	buffer[1] = cmd & 0xff;
> +	buffer[2] = value >> 8;
> +	buffer[3] = value;
> +	rc = spi_write(aic26->spi, buffer, 4);
> +	if (rc) {
> +		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
> +		return -EIO;
> +	}
> +
> +	/* update cache before returning */
> +	if (AIC26_REG_IS_CACHED(reg))
> +		cache[AIC26_REG_CACHE_ADDR(reg)] = value;
> +	return 0;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * Digital Audio Interface Operations
> + */
> +static int aic26_hw_params(struct snd_pcm_substream *substream,
> +			   struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct snd_soc_device *socdev = rtd->socdev;
> +	struct snd_soc_codec *codec = socdev->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +	int fsref, divisor, wlen, pval, jval, dval, qval;
> +	u16 reg;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
> +		substream, params);
> +	dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
> +		params_format(params));
> +
> +	switch (params_rate(params)) {
> +	 case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
> +	 case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
> +	 case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
> +	 case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
> +	 case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
> +	 case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
> +	 case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
> +	 case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
> +	 case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
> +	}
> +

Indentation.

> +	/* select data word length */
> +	switch (params_format(params)) {
> +	 case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
> +	 case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
> +	 case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
> +	 case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
> +	}
> +

ditto.

> +	/* Configure PLL */
> +	pval = 1;
> +	jval = (fsref == 44100) ? 7 : 8;
> +	dval = (fsref == 44100) ? 5264 : 1920;
> +	qval = 0;
> +	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
> +	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
> +	reg = dval << 2;
> +	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
> +

PLL/FLL/clock config is usually done in a separate function
(codec_set_pll(), callable by machine driver) so that we can change
clocks depending on the available machine clocks and srate. 

> +	/* Power up CODEC */
> +	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
> +

Codec domain (i.e Bias power) PM stuff should be done in
codec_dapm_event(). This allows us to power the codec on when we do
things like sidetone (with no active playback or capture stream).

> +	/* Audio Control 3 (master mode, fsref rate) */
> +	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
> +	reg &= ~0xf800;
> +	if (aic26->master)
> +		reg |= 0x0800;
> +	if (fsref == 48000)
> +		reg |= 0x2000;
> +	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
> +
> +	/* Audio Control 1 (FSref divisor) */
> +	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
> +	reg &= ~0x0fff;
> +	reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
> +	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
> +
> +	return 0;
> +}
> +
> +/**
> + * aic26_mute - Mute control to reduce noise when changing audio format
> + */
> +static int aic26_mute(struct snd_soc_codec_dai *dai, int mute)
> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +	u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
> +		dai, mute);
> +
> +	if (mute)
> +		reg |= 0x8080;
> +	else
> +		reg &= ~0x8080;
> +	aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
> +
> +	return 0;
> +}
> +
> +static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
> +			    int clk_id, unsigned int freq, int dir)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
> +		" freq=%i, dir=%i)\n",
> +		codec_dai, clk_id, freq, dir);
> +
> +	/* MCLK needs to fall between 2MHz and 50 MHz */
> +	if ((freq < 2000000) || (freq > 50000000))
> +		return -EINVAL;
> +
> +	aic26->mclk = freq;
> +	return 0;
> +}
> +
> +static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	struct aic26 *aic26 = codec->private_data;
> +
> +	dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
> +		codec_dai, fmt);
> +
> +	/* set master/slave audio interface */
> +	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> +	 case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
> +	 //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
> +	}
> +
> +	/* interface format */
> +	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> +	 case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
> +	 case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
> +	 case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
> +	 case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
> +	 default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
> +	}
> +
> +	return 0;
> +}
> +
> +/* ---------------------------------------------------------------------
> + * Digital Audio Interface Definition
> + */
> +struct snd_soc_codec_dai aic26_dai = {
> +	.name = "tlv320aic26",
> +	.playback = {
> +		.stream_name = "Playback",
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = AIC26_RATES,
> +		.formats = AIC26_FORMATS,
> +	},
> +	.capture = {
> +		.stream_name = "Capture",
> +		.channels_min = 2,
> +		.channels_max = 2,
> +		.rates = AIC26_RATES,
> +		.formats = AIC26_FORMATS,
> +	},
> +	.ops = {
> +		.hw_params = aic26_hw_params,
> +	},
> +	.dai_ops = {
> +		.digital_mute = aic26_mute,
> +		.set_sysclk = aic26_set_sysclk,
> +		.set_fmt = aic26_set_fmt,
> +	},
> +};
> +EXPORT_SYMBOL_GPL(aic26_dai);
> +
> +/* ---------------------------------------------------------------------
> + * ALSA controls
> + */
> +static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
> +static const struct soc_enum aic26_capture_src_enum =
> +	SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
> +
> +static const struct snd_kcontrol_new aic26_snd_controls[] = {
> +	/* Output */
> +	SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
> +	SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
> +	SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
> +	SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
> +	SOC_ENUM("Capture Source", aic26_capture_src_enum),
> +};
> +
> +/* ---------------------------------------------------------------------
> + * SoC CODEC portion of driver: probe and release routines
> + */
> +static int aic26_probe(struct platform_device *pdev)
> +{
> +	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
> +	struct snd_soc_codec *codec;
> +	struct snd_kcontrol *kcontrol;
> +	struct aic26 *aic26;
> +	int i, ret, err;
> +
> +	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
> +	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
> +	dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
> +
> +	/* Fetch the relevant aic26 private data here (it's already been
> +	 * stored in the .codec pointer) */
> +	aic26 = socdev->codec_data;
> +	if (aic26 == NULL) {
> +		dev_err(&pdev->dev, "aic26: missing codec pointer\n");
> +		return -ENODEV;
> +	}
> +	codec = &aic26->codec;
> +	socdev->codec = codec;
> +
> +	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
> +		&pdev->dev, socdev->dev);
> +	/* register pcms */
> +	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
> +	if (ret < 0) {
> +		dev_err(&pdev->dev, "aic26: failed to create pcms\n");
> +		return -ENODEV;
> +	}
> +
> +	/* register controls */
> +	dev_dbg(&pdev->dev, "Registering controls\n");
> +	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
> +		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
> +		err = snd_ctl_add(codec->card, kcontrol);
> +		WARN_ON(err < 0);
> +	}
> +
> +	/* CODEC is setup, we can register the card now */
> +	dev_dbg(&pdev->dev, "Registering card\n");
> +	ret = snd_soc_register_card(socdev);
> +	if (ret < 0) {
> +		dev_err(&pdev->dev, "aic26: failed to register card\n");
> +		goto card_err;
> +	}
> +	return 0;
> +
> + card_err:
> +	snd_soc_free_pcms(socdev);
> +	return ret;
> +}
> +
> +static int aic26_remove(struct platform_device *pdev)
> +{
> +	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
> +	snd_soc_free_pcms(socdev);
> +	return 0;
> +}
> +
> +struct snd_soc_codec_device aic26_soc_codec_dev = {
> +	.probe = aic26_probe,
> +	.remove = aic26_remove,
> +};
> +
> +/* ---------------------------------------------------------------------
> + * SPI device portion of driver: sysfs files for debugging
> + */
> +
> +static ssize_t aic26_regs_show(struct device *dev,
> +				struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	char *idx = buf;
> +	int cache_flag, addr, page, i, reg;
> +
> +	cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
> +
> +	for (page = 0; page < 3; page++) {
> +		for (i = 0; i < 0x20; i++) {
> +			addr = AIC26_PAGE_ADDR(page, i);
> +			if (i % 8 == 0)
> +				idx += sprintf(idx, "%i:%.2i:", page,i);
> +			if (cache_flag)
> +				reg = aic26_reg_read_cache(&aic26->codec, addr);
> +			else
> +				reg = aic26_reg_read(&aic26->codec, addr);
> +			idx += sprintf(idx, " %.4x", reg);
> +			if (i % 8 == 7)
> +				idx += sprintf(idx, "\n");
> +		}
> +	}
> +	return idx - buf;
> +}
> +

The soc_core already has a codec reg dump sysfs file, so we don't need
this.

> +static ssize_t aic26_keyclick_show(struct device *dev,
> +				   struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val, amp, freq, len;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	amp = (val >> 12) & 0x7;
> +	freq = (125 << ((val >> 8) & 0x7)) >> 1;
> +	len = 2 * (1 +((val >> 8) & 0xf));
> +
> +	return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
> +}
> +
> +/* Any write to the keyclick attribute will trigger the keyclick */
> +static ssize_t aic26_keyclick_set(struct device *dev,
> +				  struct device_attribute *attr,
> +				  const char *buf, size_t count)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	val |= 0x8000;
> +	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
> +
> +	return count;
> +}
> +
> +DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL);
> +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL);
> +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
> +
> +/* ---------------------------------------------------------------------
> + * SPI device portion of driver: probe and release routines and SPI
> + * 				 driver registration.
> + */
> +static int aic26_spi_probe(struct spi_device *spi)
> +{
> +	struct aic26 *aic26;
> +	int rc, i, reg;
> +
> +	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
> +
> +	/* Allocate driver data */
> +	aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
> +	if (!aic26)
> +		return -ENOMEM;
> +
> +	/* Initialize the driver data */
> +	aic26->spi = spi;
> +	dev_set_drvdata(&spi->dev, aic26);
> +
> +	/* Setup what we can in the codec structure so that the register
> +	 * access functions will work as expected.  More will be filled
> +	 * out when it is probed by the SoC CODEC part of this driver */
> +	aic26->codec.private_data = aic26;
> +	aic26->codec.name = "aic26";
> +	aic26->codec.owner = THIS_MODULE;
> +	aic26->codec.dai = &aic26_dai;
> +	aic26->codec.num_dai = 1;
> +	aic26->codec.read = aic26_reg_read;
> +	aic26->codec.write = aic26_reg_write;
> +	aic26->master = 1;
> +	mutex_init(&aic26->codec.mutex);
> +	INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
> +	INIT_LIST_HEAD(&aic26->codec.dapm_paths);
> +	aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
> +	aic26->codec.reg_cache = aic26->reg_cache;
> +
> +	/* Reset the codec to power on defaults */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
> +
> +	/* Power up CODEC */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
> +
> +	/* Audio Control 3 (master mode, fsref rate) */
> +	reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
> +	reg &= ~0xf800;
> +	reg |= 0x0800; /* set master mode */
> +	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
> +
> +	/* Fill page 2 register cache */
> +	for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
> +		aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));
> +
> +	/* Register the sysfs files for debugging */
> +	/* Create SysFS files */
> +	rc = device_create_file(&spi->dev, &dev_attr_regs);
> +	rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
> +	rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
> +	if (rc)
> +		dev_info(&spi->dev, "error creating sysfs files\n");
> +
> +	/* Tell the of_soc helper about this codec */
> +	of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
> +				  spi->dev.archdata.of_node);
> +
> +	dev_dbg(&spi->dev, "SPI device initialized\n");
> +	return 0;
> +}
> +
> +static int aic26_spi_remove(struct spi_device *spi)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
> +
> +	kfree(aic26);
> +
> +	return 0;
> +}
> +
> +static struct spi_driver aic26_spi = {
> +	.driver = {
> +		.name = "tlv320aic26",
> +		.owner = THIS_MODULE,
> +	},
> +	.probe = aic26_spi_probe,
> +	.remove = aic26_spi_remove,
> +};
> +
> +static int __init aic26_init(void)
> +{
> +	return spi_register_driver(&aic26_spi);
> +}
> +module_init(aic26_init);
> +
> +static void __exit aic26_exit(void)
> +{
> +	spi_unregister_driver(&aic26_spi);
> +}
> +module_exit(aic26_exit);
> 
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Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:
> On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> > From: Grant Likely <grant.likely@secretlab.ca>
> > 
> > ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
> > v1 API, so I don't expect it to get merged as-is, but I want to get it
> > out there for review.
> > ---
> 
> Fwiw, I usually put all the codec registers defs in a separate header
> just in case we need to do any codec stuff in the machine driver.

okay, done

> > +	switch (params_rate(params)) {
> > +	 case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
> > +	 case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
> > +	 case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
> > +	 case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
> > +	 case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
> > +	 case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
> > +	 case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
> > +	 case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
> > +	 case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
> > +	 default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
> > +	}
> > +
> 
> Indentation.

done.

> > +	/* Configure PLL */
> > +	pval = 1;
> > +	jval = (fsref == 44100) ? 7 : 8;
> > +	dval = (fsref == 44100) ? 5264 : 1920;
> > +	qval = 0;
> > +	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
> > +	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
> > +	reg = dval << 2;
> > +	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
> > +
> 
> PLL/FLL/clock config is usually done in a separate function
> (codec_set_pll(), callable by machine driver) so that we can change
> clocks depending on the available machine clocks and srate. 
> 
> > +	/* Power up CODEC */
> > +	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
> > +
> 
> Codec domain (i.e Bias power) PM stuff should be done in
> codec_dapm_event(). This allows us to power the codec on when we do
> things like sidetone (with no active playback or capture stream).
> 

Ugh, I'm going to have to leave these two for now.  I don't understand
enough about the ASoC structure yet to understand what it should look
like.  I'll probably need help, but I don't think I can get it sorted
out before the merge window.

Do these two comments need to be addressed before the driver is merged?

> > +static ssize_t aic26_regs_show(struct device *dev,
> > +				struct device_attribute *attr, char *buf)
> > +{
> > +	struct aic26 *aic26 = dev_get_drvdata(dev);
> > +	char *idx = buf;
> > +	int cache_flag, addr, page, i, reg;
> > +
> > +	cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
> > +
> > +	for (page = 0; page < 3; page++) {
> > +		for (i = 0; i < 0x20; i++) {
> > +			addr = AIC26_PAGE_ADDR(page, i);
> > +			if (i % 8 == 0)
> > +				idx += sprintf(idx, "%i:%.2i:", page,i);
> > +			if (cache_flag)
> > +				reg = aic26_reg_read_cache(&aic26->codec, addr);
> > +			else
> > +				reg = aic26_reg_read(&aic26->codec, addr);
> > +			idx += sprintf(idx, " %.4x", reg);
> > +			if (i % 8 == 7)
> > +				idx += sprintf(idx, "\n");
> > +		}
> > +	}
> > +	return idx - buf;
> > +}
> > +
> 
> The soc_core already has a codec reg dump sysfs file, so we don't need
> this.

Hmmm, I haven't been able to find this; either in the code or on a live
running system.  Where is the common reg dump implemented.

Thanks for the comments.
g.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:
> On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
> > From: Grant Likely <grant.likely@secretlab.ca>
> > 
> > ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
> > v1 API, so I don't expect it to get merged as-is, but I want to get it
> > out there for review.
> > ---
> 
> Fwiw, I usually put all the codec registers defs in a separate header
> just in case we need to do any codec stuff in the machine driver.

okay, done

> > +	switch (params_rate(params)) {
> > +	 case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
> > +	 case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
> > +	 case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
> > +	 case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
> > +	 case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
> > +	 case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
> > +	 case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
> > +	 case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
> > +	 case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
> > +	 default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
> > +	}
> > +
> 
> Indentation.

done.

> > +	/* Configure PLL */
> > +	pval = 1;
> > +	jval = (fsref == 44100) ? 7 : 8;
> > +	dval = (fsref == 44100) ? 5264 : 1920;
> > +	qval = 0;
> > +	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
> > +	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
> > +	reg = dval << 2;
> > +	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
> > +
> 
> PLL/FLL/clock config is usually done in a separate function
> (codec_set_pll(), callable by machine driver) so that we can change
> clocks depending on the available machine clocks and srate. 
> 
> > +	/* Power up CODEC */
> > +	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
> > +
> 
> Codec domain (i.e Bias power) PM stuff should be done in
> codec_dapm_event(). This allows us to power the codec on when we do
> things like sidetone (with no active playback or capture stream).
> 

Ugh, I'm going to have to leave these two for now.  I don't understand
enough about the ASoC structure yet to understand what it should look
like.  I'll probably need help, but I don't think I can get it sorted
out before the merge window.

Do these two comments need to be addressed before the driver is merged?

> > +static ssize_t aic26_regs_show(struct device *dev,
> > +				struct device_attribute *attr, char *buf)
> > +{
> > +	struct aic26 *aic26 = dev_get_drvdata(dev);
> > +	char *idx = buf;
> > +	int cache_flag, addr, page, i, reg;
> > +
> > +	cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
> > +
> > +	for (page = 0; page < 3; page++) {
> > +		for (i = 0; i < 0x20; i++) {
> > +			addr = AIC26_PAGE_ADDR(page, i);
> > +			if (i % 8 == 0)
> > +				idx += sprintf(idx, "%i:%.2i:", page,i);
> > +			if (cache_flag)
> > +				reg = aic26_reg_read_cache(&aic26->codec, addr);
> > +			else
> > +				reg = aic26_reg_read(&aic26->codec, addr);
> > +			idx += sprintf(idx, " %.4x", reg);
> > +			if (i % 8 == 7)
> > +				idx += sprintf(idx, "\n");
> > +		}
> > +	}
> > +	return idx - buf;
> > +}
> > +
> 
> The soc_core already has a codec reg dump sysfs file, so we don't need
> this.

Hmmm, I haven't been able to find this; either in the code or on a live
running system.  Where is the common reg dump implemented.

Thanks for the comments.
g.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:
> On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:

> > PLL/FLL/clock config is usually done in a separate function
> > (codec_set_pll(), callable by machine driver) so that we can change
> > clocks depending on the available machine clocks and srate. 

...

> > Codec domain (i.e Bias power) PM stuff should be done in
> > codec_dapm_event(). This allows us to power the codec on when we do
> > things like sidetone (with no active playback or capture stream).

> Ugh, I'm going to have to leave these two for now.  I don't understand
> enough about the ASoC structure yet to understand what it should look
> like.  I'll probably need help, but I don't think I can get it sorted
> out before the merge window.

> Do these two comments need to be addressed before the driver is merged?

It wouldn't be the only driver not to implement PLL configuration in
this way so that's probably be OK for an initial merge.  What's expected
for PLL configuration is that you implement the DAI set_pll() operation
in the codec driver, allowing machine drivers to configure the PLL when
they wish.

The power configuration should be fixed, though.  Normally drivers
either fully implement DAPM (including set_bias_level()) or power
everything in the codec up when the driver is loaded.  At the minute
what the driver is doing appears to be powering the codec up in both
_hw_params() and _probe() but never powering anything down - if that is
the case then probably all you need to do is remove the extra power up
from hw_params(), giving you the simple option.

> Hmmm, I haven't been able to find this; either in the code or on a live
> running system.  Where is the common reg dump implemented.

/sys/bus/platform/devices/soc-audio/codec_reg

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:
> On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:

> > PLL/FLL/clock config is usually done in a separate function
> > (codec_set_pll(), callable by machine driver) so that we can change
> > clocks depending on the available machine clocks and srate. 

...

> > Codec domain (i.e Bias power) PM stuff should be done in
> > codec_dapm_event(). This allows us to power the codec on when we do
> > things like sidetone (with no active playback or capture stream).

> Ugh, I'm going to have to leave these two for now.  I don't understand
> enough about the ASoC structure yet to understand what it should look
> like.  I'll probably need help, but I don't think I can get it sorted
> out before the merge window.

> Do these two comments need to be addressed before the driver is merged?

It wouldn't be the only driver not to implement PLL configuration in
this way so that's probably be OK for an initial merge.  What's expected
for PLL configuration is that you implement the DAI set_pll() operation
in the codec driver, allowing machine drivers to configure the PLL when
they wish.

The power configuration should be fixed, though.  Normally drivers
either fully implement DAPM (including set_bias_level()) or power
everything in the codec up when the driver is loaded.  At the minute
what the driver is doing appears to be powering the codec up in both
_hw_params() and _probe() but never powering anything down - if that is
the case then probably all you need to do is remove the extra power up
from hw_params(), giving you the simple option.

> Hmmm, I haven't been able to find this; either in the code or on a live
> running system.  Where is the common reg dump implemented.

/sys/bus/platform/devices/soc-audio/codec_reg

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Sat, Jul 12, 2008 at 11:36 AM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> The power configuration should be fixed, though.  Normally drivers
> either fully implement DAPM (including set_bias_level()) or power
> everything in the codec up when the driver is loaded.  At the minute
> what the driver is doing appears to be powering the codec up in both
> _hw_params() and _probe() but never powering anything down - if that is
> the case then probably all you need to do is remove the extra power up
> from hw_params(), giving you the simple option.

Okay, cool.  I'll do this for the time being then.

Thanks,
g.

-- 
Grant Likely, B.Sc., P.Eng.
Secret Lab Technologies Ltd.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Sat, Jul 12, 2008 at 11:36 AM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> The power configuration should be fixed, though.  Normally drivers
> either fully implement DAPM (including set_bias_level()) or power
> everything in the codec up when the driver is loaded.  At the minute
> what the driver is doing appears to be powering the codec up in both
> _hw_params() and _probe() but never powering anything down - if that is
> the case then probably all you need to do is remove the extra power up
> from hw_params(), giving you the simple option.

Okay, cool.  I'll do this for the time being then.

Thanks,
g.

-- 
Grant Likely, B.Sc., P.Eng.
Secret Lab Technologies Ltd.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:
> On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:

> It wouldn't be the only driver not to implement PLL configuration in
> this way so that's probably be OK for an initial merge.  What's expected
> for PLL configuration is that you implement the DAI set_pll() operation
> in the codec driver, allowing machine drivers to configure the PLL when
> they wish.

okay

> The power configuration should be fixed, though.  Normally drivers
> either fully implement DAPM (including set_bias_level()) or power
> everything in the codec up when the driver is loaded.  At the minute
> what the driver is doing appears to be powering the codec up in both
> _hw_params() and _probe() but never powering anything down - if that is
> the case then probably all you need to do is remove the extra power up
> from hw_params(), giving you the simple option.

done

> > Hmmm, I haven't been able to find this; either in the code or on a live
> > running system.  Where is the common reg dump implemented.
> 
> /sys/bus/platform/devices/soc-audio/codec_reg

Yikes.  The AIC26 has registers all over the place and most of them are
empty.  The codec_reg attribute handling means I need to maintain a
cache of the entire register file; not just the part that is actually
used.  Oh well; I can work around it.

Thanks,
g.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:
> On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:

> It wouldn't be the only driver not to implement PLL configuration in
> this way so that's probably be OK for an initial merge.  What's expected
> for PLL configuration is that you implement the DAI set_pll() operation
> in the codec driver, allowing machine drivers to configure the PLL when
> they wish.

okay

> The power configuration should be fixed, though.  Normally drivers
> either fully implement DAPM (including set_bias_level()) or power
> everything in the codec up when the driver is loaded.  At the minute
> what the driver is doing appears to be powering the codec up in both
> _hw_params() and _probe() but never powering anything down - if that is
> the case then probably all you need to do is remove the extra power up
> from hw_params(), giving you the simple option.

done

> > Hmmm, I haven't been able to find this; either in the code or on a live
> > running system.  Where is the common reg dump implemented.
> 
> /sys/bus/platform/devices/soc-audio/codec_reg

Yikes.  The AIC26 has registers all over the place and most of them are
empty.  The codec_reg attribute handling means I need to maintain a
cache of the entire register file; not just the part that is actually
used.  Oh well; I can work around it.

Thanks,
g.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Thu, Jul 17, 2008 at 05:31:53PM -0600, Grant Likely wrote:
> On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:

> > /sys/bus/platform/devices/soc-audio/codec_reg

> Yikes.  The AIC26 has registers all over the place and most of them are
> empty.  The codec_reg attribute handling means I need to maintain a
> cache of the entire register file; not just the part that is actually
> used.  Oh well; I can work around it.

Sparse (or sparsely used) register maps are pretty common - in practice
it's not been a problem to just have a cache that covers everything and
only gets read, the amount of data involved is rarely that large in the
context of what you need to store the audio you're working with.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Thu, Jul 17, 2008 at 05:31:53PM -0600, Grant Likely wrote:
> On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:

> > /sys/bus/platform/devices/soc-audio/codec_reg

> Yikes.  The AIC26 has registers all over the place and most of them are
> empty.  The codec_reg attribute handling means I need to maintain a
> cache of the entire register file; not just the part that is actually
> used.  Oh well; I can work around it.

Sparse (or sparsely used) register maps are pretty common - in practice
it's not been a problem to just have a cache that covers everything and
only gets read, the amount of data involved is rarely that large in the
context of what you need to store the audio you're working with.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Fri, Jul 18, 2008 at 10:58:25AM +0100, Mark Brown wrote:
> Sparse (or sparsely used) register maps are pretty common - in practice
> it's not been a problem to just have a cache that covers everything and
> only gets read, the amount of data involved is rarely that large in the
> context of what you need to store the audio you're working with.

okay.

g.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Fri, Jul 18, 2008 at 10:58:25AM +0100, Mark Brown wrote:
> Sparse (or sparsely used) register maps are pretty common - in practice
> it's not been a problem to just have a cache that covers everything and
> only gets read, the amount of data involved is rarely that large in the
> context of what you need to store the audio you're working with.

okay.

g.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
>  v1 API, so I don't expect it to get merged as-is, but I want to get it
>  out there for review.
>  ---
>
>   sound/soc/codecs/Kconfig       |    4
>   sound/soc/codecs/Makefile      |    2
>   sound/soc/codecs/tlv320aic26.c |  630 ++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 636 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
>  index 3903ab7..96c7bfe 100644
>  --- a/sound/soc/codecs/Kconfig
>  +++ b/sound/soc/codecs/Kconfig
>  @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA
>         bool
>         depends on SND_SOC_CS4270
>
>  +config SND_SOC_TLV320AIC26
>  +       tristate "TI TLB320AIC26 Codec support"
>  +       depends on SND_SOC && SPI
>  +
>   config SND_SOC_TLV320AIC3X
>         tristate
>         depends on SND_SOC && I2C
>  diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
>  index 4e1314c..ec0cd93 100644
>  --- a/sound/soc/codecs/Makefile
>  +++ b/sound/soc/codecs/Makefile
>  @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o
>   snd-soc-wm9712-objs := wm9712.o
>   snd-soc-wm9713-objs := wm9713.o
>   snd-soc-cs4270-objs := cs4270.o
>  +snd-soc-tlv320aic26-objs := tlv320aic26.o
>   snd-soc-tlv320aic3x-objs := tlv320aic3x.o
>
>   obj-$(CONFIG_SND_SOC_AC97_CODEC)       += snd-soc-ac97.o
>  @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753)  += snd-soc-wm8753.o
>   obj-$(CONFIG_SND_SOC_WM9712)   += snd-soc-wm9712.o
>   obj-$(CONFIG_SND_SOC_WM9713)   += snd-soc-wm9713.o
>   obj-$(CONFIG_SND_SOC_CS4270)   += snd-soc-cs4270.o
>  +obj-$(CONFIG_SND_SOC_TLV320AIC26)      += snd-soc-tlv320aic26.o
>   obj-$(CONFIG_SND_SOC_TLV320AIC3X)      += snd-soc-tlv320aic3x.o
>  diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
>  new file mode 100644
>  index 0000000..aee1dbc
>  --- /dev/null
>  +++ b/sound/soc/codecs/tlv320aic26.c
>  @@ -0,0 +1,630 @@
>  +/*
>  + * Texas Instruments TLV320AIC26 low power audio CODEC
>  + * ALSA SoC CODEC driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/device.h>
>  +#include <linux/sysfs.h>
>  +#include <linux/spi/spi.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-dapm.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +
>  +/* AIC26 Registers */
>  +#define AIC26_READ_COMMAND_WORD(addr)  ((1 << 15) | (addr << 5))
>  +#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5))
>  +#define AIC26_PAGE_ADDR(page, offset)  ((page << 6) | offset)
>  +#define AIC26_NUM_REGS                 AIC26_PAGE_ADDR(3, 0)
>  +#define AIC26_REG_CACHE_SIZE           (0x20) /* only page 2 cached */
>  +#define AIC26_REG_IS_CACHED(addr)      ((addr & ~0x1f) == (2 << 6))
>  +#define AIC26_REG_CACHE_ADDR(addr)     (addr & 0x1f)
>  +
>  +/* Page 0: Auxillary data registers */
>  +#define AIC26_REG_BAT1                 AIC26_PAGE_ADDR(0, 0x05)
>  +#define AIC26_REG_BAT2                 AIC26_PAGE_ADDR(0, 0x06)
>  +#define AIC26_REG_AUX                  AIC26_PAGE_ADDR(0, 0x07)
>  +#define AIC26_REG_TEMP1                        AIC26_PAGE_ADDR(0, 0x09)
>  +#define AIC26_REG_TEMP2                        AIC26_PAGE_ADDR(0, 0x0A)
>  +
>  +/* Page 1: Auxillary control registers */
>  +#define AIC26_REG_AUX_ADC              AIC26_PAGE_ADDR(1, 0x00)
>  +#define AIC26_REG_STATUS               AIC26_PAGE_ADDR(1, 0x01)
>  +#define AIC26_REG_REFERENCE            AIC26_PAGE_ADDR(1, 0x03)
>  +#define AIC26_REG_RESET                        AIC26_PAGE_ADDR(1, 0x04)
>  +
>  +/* Page 2: Audio control registers */
>  +#define AIC26_REG_AUDIO_CTRL1          AIC26_PAGE_ADDR(2, 0x00)
>  +#define AIC26_REG_ADC_GAIN             AIC26_PAGE_ADDR(2, 0x01)
>  +#define AIC26_REG_DAC_GAIN             AIC26_PAGE_ADDR(2, 0x02)
>  +#define AIC26_REG_SIDETONE             AIC26_PAGE_ADDR(2, 0x03)
>  +#define AIC26_REG_AUDIO_CTRL2          AIC26_PAGE_ADDR(2, 0x04)
>  +#define AIC26_REG_POWER_CTRL           AIC26_PAGE_ADDR(2, 0x05)
>  +#define AIC26_REG_AUDIO_CTRL3          AIC26_PAGE_ADDR(2, 0x06)
>  +
>  +#define AIC26_REG_FILTER_COEFF_L_N0    AIC26_PAGE_ADDR(2, 0x07)
>  +#define AIC26_REG_FILTER_COEFF_L_N1    AIC26_PAGE_ADDR(2, 0x08)
>  +#define AIC26_REG_FILTER_COEFF_L_N2    AIC26_PAGE_ADDR(2, 0x09)
>  +#define AIC26_REG_FILTER_COEFF_L_N3    AIC26_PAGE_ADDR(2, 0x0A)
>  +#define AIC26_REG_FILTER_COEFF_L_N4    AIC26_PAGE_ADDR(2, 0x0B)
>  +#define AIC26_REG_FILTER_COEFF_L_N5    AIC26_PAGE_ADDR(2, 0x0C)
>  +#define AIC26_REG_FILTER_COEFF_L_D1    AIC26_PAGE_ADDR(2, 0x0D)
>  +#define AIC26_REG_FILTER_COEFF_L_D2    AIC26_PAGE_ADDR(2, 0x0E)
>  +#define AIC26_REG_FILTER_COEFF_L_D4    AIC26_PAGE_ADDR(2, 0x0F)
>  +#define AIC26_REG_FILTER_COEFF_L_D5    AIC26_PAGE_ADDR(2, 0x10)
>  +#define AIC26_REG_FILTER_COEFF_R_N0    AIC26_PAGE_ADDR(2, 0x11)
>  +#define AIC26_REG_FILTER_COEFF_R_N1    AIC26_PAGE_ADDR(2, 0x12)
>  +#define AIC26_REG_FILTER_COEFF_R_N2    AIC26_PAGE_ADDR(2, 0x13)
>  +#define AIC26_REG_FILTER_COEFF_R_N3    AIC26_PAGE_ADDR(2, 0x14)
>  +#define AIC26_REG_FILTER_COEFF_R_N4    AIC26_PAGE_ADDR(2, 0x15)
>  +#define AIC26_REG_FILTER_COEFF_R_N5    AIC26_PAGE_ADDR(2, 0x16)
>  +#define AIC26_REG_FILTER_COEFF_R_D1    AIC26_PAGE_ADDR(2, 0x17)
>  +#define AIC26_REG_FILTER_COEFF_R_D2    AIC26_PAGE_ADDR(2, 0x18)
>  +#define AIC26_REG_FILTER_COEFF_R_D4    AIC26_PAGE_ADDR(2, 0x19)
>  +#define AIC26_REG_FILTER_COEFF_R_D5    AIC26_PAGE_ADDR(2, 0x1A)
>  +
>  +#define AIC26_REG_PLL_PROG1            AIC26_PAGE_ADDR(2, 0x1B)
>  +#define AIC26_REG_PLL_PROG2            AIC26_PAGE_ADDR(2, 0x1C)
>  +#define AIC26_REG_AUDIO_CTRL4          AIC26_PAGE_ADDR(2, 0x1D)
>  +#define AIC26_REG_AUDIO_CTRL5          AIC26_PAGE_ADDR(2, 0x1E)
>  +
>  +#define AIC26_RATES    (SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
>  +                        SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
>  +                        SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
>  +                        SNDRV_PCM_RATE_48000)
>  +#define AIC26_FORMATS  (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/* fsref dividers; used in register 'Audio Control 1' */
>  +enum aic26_divisors {
>  +       AIC26_DIV_1     = 0,
>  +       AIC26_DIV_1_5   = 1,
>  +       AIC26_DIV_2     = 2,
>  +       AIC26_DIV_3     = 3,
>  +       AIC26_DIV_4     = 4,
>  +       AIC26_DIV_5     = 5,
>  +       AIC26_DIV_5_5   = 6,
>  +       AIC26_DIV_6     = 7,
>  +};
>  +
>  +/* Digital data format */
>  +enum aic26_datfm {
>  +       AIC26_DATFM_I2S         = 0 << 8,
>  +       AIC26_DATFM_DSP         = 1 << 8,
>  +       AIC26_DATFM_RIGHTJ      = 2 << 8, /* right justified */
>  +       AIC26_DATFM_LEFTJ       = 3 << 8, /* left justified */
>  +};
>  +
>  +/* Sample word length in bits; used in register 'Audio Control 1' */
>  +enum aic26_wlen {
>  +       AIC26_WLEN_16   = 0 << 10,
>  +       AIC26_WLEN_20   = 1 << 10,
>  +       AIC26_WLEN_24   = 2 << 10,
>  +       AIC26_WLEN_32   = 3 << 10,
>  +};
>  +
>  +/* AIC26 driver private data */
>  +struct aic26 {
>  +       struct spi_device *spi;
>  +       struct snd_soc_codec codec;
>  +       u16 reg_cache[AIC26_REG_CACHE_SIZE];    /* shadow registers */
>  +       int master;
>  +       int datfm;
>  +       int mclk;
>  +
>  +       /* Keyclick parameters */
>  +       int keyclick_amplitude;
>  +       int keyclick_freq;
>  +       int keyclick_len;
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Register access routines
>  + */
>  +static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
>  +                                  unsigned int reg)
>  +{
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 *cache = codec->reg_cache;
>  +       u16 cmd, value;
>  +       u8 buffer[2];
>  +       int rc;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return 0;
>  +       }
>  +
>  +       /* Do SPI transfer; first 16bits are command; remaining is
>  +        * register contents */
>  +       cmd = AIC26_READ_COMMAND_WORD(reg);
>  +       buffer[0] = (cmd >> 8) & 0xff;
>  +       buffer[1] = cmd & 0xff;
>  +       rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
>  +       if (rc) {
>  +               dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
>  +               return -EIO;
>  +       }
>  +       value = (buffer[0] << 8) | buffer[1];
>  +
>  +       /* Update the cache before returning with the value */
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               cache[AIC26_REG_CACHE_ADDR(reg)] = value;
>  +       return value;
>  +}
>  +
>  +static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
>  +                                        unsigned int reg)
>  +{
>  +       u16 *cache = codec->reg_cache;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return 0;
>  +       }
>  +
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               return cache[AIC26_REG_CACHE_ADDR(reg)];
>  +
>  +       return aic26_reg_read(codec, reg);
>  +}
>  +
>  +static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
>  +                          unsigned int value)
>  +{
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 *cache = codec->reg_cache;
>  +       u16 cmd;
>  +       u8 buffer[4];
>  +       int rc;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Do SPI transfer; first 16bits are command; remaining is data
>  +        * to write into register */
>  +       cmd = AIC26_WRITE_COMMAND_WORD(reg);
>  +       buffer[0] = (cmd >> 8) & 0xff;
>  +       buffer[1] = cmd & 0xff;
>  +       buffer[2] = value >> 8;
>  +       buffer[3] = value;
>  +       rc = spi_write(aic26->spi, buffer, 4);
>  +       if (rc) {
>  +               dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
>  +               return -EIO;
>  +       }
>  +
>  +       /* update cache before returning */
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               cache[AIC26_REG_CACHE_ADDR(reg)] = value;
>  +       return 0;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * Digital Audio Interface Operations
>  + */
>  +static int aic26_hw_params(struct snd_pcm_substream *substream,
>  +                          struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct snd_soc_device *socdev = rtd->socdev;
>  +       struct snd_soc_codec *codec = socdev->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +       int fsref, divisor, wlen, pval, jval, dval, qval;
>  +       u16 reg;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
>  +               substream, params);
>  +       dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
>  +               params_format(params));
>  +
>  +       switch (params_rate(params)) {
>  +        case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
>  +        case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
>  +        case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
>  +        case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
>  +        case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
>  +        case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
>  +        case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
>  +        case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
>  +        case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
>  +       }
>  +
>  +       /* select data word length */
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
>  +        case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
>  +        case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
>  +        case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
>  +       }
>  +
>  +       /* Configure PLL */
>  +       pval = 1;
>  +       jval = (fsref == 44100) ? 7 : 8;
>  +       dval = (fsref == 44100) ? 5264 : 1920;
>  +       qval = 0;
>  +       reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
>  +       aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
>  +       reg = dval << 2;
>  +       aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
>  +
>  +       /* Power up CODEC */
>  +       aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
>  +
>  +       /* Audio Control 3 (master mode, fsref rate) */
>  +       reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
>  +       reg &= ~0xf800;
>  +       if (aic26->master)
>  +               reg |= 0x0800;
>  +       if (fsref == 48000)
>  +               reg |= 0x2000;
>  +       aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
>  +
>  +       /* Audio Control 1 (FSref divisor) */
>  +       reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
>  +       reg &= ~0x0fff;
>  +       reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
>  +       aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * aic26_mute - Mute control to reduce noise when changing audio format
>  + */
>  +static int aic26_mute(struct snd_soc_codec_dai *dai, int mute)
>  +{
>  +       struct snd_soc_codec *codec = dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
>  +               dai, mute);
>  +
>  +       if (mute)
>  +               reg |= 0x8080;
>  +       else
>  +               reg &= ~0x8080;
>  +       aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
>  +
>  +       return 0;
>  +}
>  +
>  +static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
>  +                           int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct snd_soc_codec *codec = codec_dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
>  +               " freq=%i, dir=%i)\n",
>  +               codec_dai, clk_id, freq, dir);
>  +
>  +       /* MCLK needs to fall between 2MHz and 50 MHz */
>  +       if ((freq < 2000000) || (freq > 50000000))
>  +               return -EINVAL;
>  +
>  +       aic26->mclk = freq;
>  +       return 0;
>  +}
>  +
>  +static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt)
>  +{
>  +       struct snd_soc_codec *codec = codec_dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
>  +               codec_dai, fmt);
>  +
>  +       /* set master/slave audio interface */
>  +       switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
>  +        case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
>  +        //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
>  +       }
>  +
>  +       /* interface format */
>  +       switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
>  +        case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
>  +        case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
>  +        case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
>  +        case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * Digital Audio Interface Definition
>  + */
>  +struct snd_soc_codec_dai aic26_dai = {
>  +       .name = "tlv320aic26",
>  +       .playback = {
>  +               .stream_name = "Playback",
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = AIC26_RATES,
>  +               .formats = AIC26_FORMATS,
>  +       },
>  +       .capture = {
>  +               .stream_name = "Capture",
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = AIC26_RATES,
>  +               .formats = AIC26_FORMATS,
>  +       },
>  +       .ops = {
>  +               .hw_params = aic26_hw_params,
>  +       },
>  +       .dai_ops = {
>  +               .digital_mute = aic26_mute,
>  +               .set_sysclk = aic26_set_sysclk,
>  +               .set_fmt = aic26_set_fmt,
>  +       },
>  +};
>  +EXPORT_SYMBOL_GPL(aic26_dai);
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA controls
>  + */
>  +static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
>  +static const struct soc_enum aic26_capture_src_enum =
>  +       SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
>  +
>  +static const struct snd_kcontrol_new aic26_snd_controls[] = {
>  +       /* Output */
>  +       SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
>  +       SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
>  +       SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
>  +       SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
>  +       SOC_ENUM("Capture Source", aic26_capture_src_enum),
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * SoC CODEC portion of driver: probe and release routines
>  + */
>  +static int aic26_probe(struct platform_device *pdev)
>  +{
>  +       struct snd_soc_device *socdev = platform_get_drvdata(pdev);
>  +       struct snd_soc_codec *codec;
>  +       struct snd_kcontrol *kcontrol;
>  +       struct aic26 *aic26;
>  +       int i, ret, err;
>  +
>  +       dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
>  +       dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
>  +       dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
>  +
>  +       /* Fetch the relevant aic26 private data here (it's already been
>  +        * stored in the .codec pointer) */
>  +       aic26 = socdev->codec_data;
>  +       if (aic26 == NULL) {
>  +               dev_err(&pdev->dev, "aic26: missing codec pointer\n");
>  +               return -ENODEV;
>  +       }
>  +       codec = &aic26->codec;
>  +       socdev->codec = codec;
>  +
>  +       dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
>  +               &pdev->dev, socdev->dev);
>  +       /* register pcms */
>  +       ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
>  +       if (ret < 0) {
>  +               dev_err(&pdev->dev, "aic26: failed to create pcms\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* register controls */
>  +       dev_dbg(&pdev->dev, "Registering controls\n");
>  +       for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
>  +               kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
>  +               err = snd_ctl_add(codec->card, kcontrol);
>  +               WARN_ON(err < 0);
>  +       }
>  +
>  +       /* CODEC is setup, we can register the card now */
>  +       dev_dbg(&pdev->dev, "Registering card\n");
>  +       ret = snd_soc_register_card(socdev);
>  +       if (ret < 0) {
>  +               dev_err(&pdev->dev, "aic26: failed to register card\n");
>  +               goto card_err;
>  +       }
>  +       return 0;
>  +
>  + card_err:
>  +       snd_soc_free_pcms(socdev);
>  +       return ret;
>  +}
>  +
>  +static int aic26_remove(struct platform_device *pdev)
>  +{
>  +       struct snd_soc_device *socdev = platform_get_drvdata(pdev);
>  +       snd_soc_free_pcms(socdev);
>  +       return 0;
>  +}
>  +
>  +struct snd_soc_codec_device aic26_soc_codec_dev = {
>  +       .probe = aic26_probe,
>  +       .remove = aic26_remove,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * SPI device portion of driver: sysfs files for debugging
>  + */
>  +
>  +static ssize_t aic26_regs_show(struct device *dev,
>  +                               struct device_attribute *attr, char *buf)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       char *idx = buf;
>  +       int cache_flag, addr, page, i, reg;
>  +
>  +       cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
>  +
>  +       for (page = 0; page < 3; page++) {
>  +               for (i = 0; i < 0x20; i++) {
>  +                       addr = AIC26_PAGE_ADDR(page, i);
>  +                       if (i % 8 == 0)
>  +                               idx += sprintf(idx, "%i:%.2i:", page,i);
>  +                       if (cache_flag)
>  +                               reg = aic26_reg_read_cache(&aic26->codec, addr);
>  +                       else
>  +                               reg = aic26_reg_read(&aic26->codec, addr);
>  +                       idx += sprintf(idx, " %.4x", reg);
>  +                       if (i % 8 == 7)
>  +                               idx += sprintf(idx, "\n");
>  +               }
>  +       }
>  +       return idx - buf;
>  +}
>  +
>  +static ssize_t aic26_keyclick_show(struct device *dev,
>  +                                  struct device_attribute *attr, char *buf)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       int val, amp, freq, len;
>  +
>  +       val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
>  +       amp = (val >> 12) & 0x7;
>  +       freq = (125 << ((val >> 8) & 0x7)) >> 1;
>  +       len = 2 * (1 +((val >> 8) & 0xf));
>  +
>  +       return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
>  +}
>  +
>  +/* Any write to the keyclick attribute will trigger the keyclick */
>  +static ssize_t aic26_keyclick_set(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf, size_t count)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       int val;
>  +
>  +       val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
>  +       val |= 0x8000;
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
>  +
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL);
>  +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL);
>  +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
>  +
>  +/* ---------------------------------------------------------------------
>  + * SPI device portion of driver: probe and release routines and SPI
>  + *                              driver registration.
>  + */
>  +static int aic26_spi_probe(struct spi_device *spi)
>  +{
>  +       struct aic26 *aic26;
>  +       int rc, i, reg;
>  +
>  +       dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
>  +
>  +       /* Allocate driver data */
>  +       aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
>  +       if (!aic26)
>  +               return -ENOMEM;
>  +
>  +       /* Initialize the driver data */
>  +       aic26->spi = spi;
>  +       dev_set_drvdata(&spi->dev, aic26);
>  +
>  +       /* Setup what we can in the codec structure so that the register
>  +        * access functions will work as expected.  More will be filled
>  +        * out when it is probed by the SoC CODEC part of this driver */
>  +       aic26->codec.private_data = aic26;
>  +       aic26->codec.name = "aic26";
>  +       aic26->codec.owner = THIS_MODULE;
>  +       aic26->codec.dai = &aic26_dai;
>  +       aic26->codec.num_dai = 1;
>  +       aic26->codec.read = aic26_reg_read;
>  +       aic26->codec.write = aic26_reg_write;
>  +       aic26->master = 1;
>  +       mutex_init(&aic26->codec.mutex);
>  +       INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
>  +       INIT_LIST_HEAD(&aic26->codec.dapm_paths);
>  +       aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
>  +       aic26->codec.reg_cache = aic26->reg_cache;
>  +
>  +       /* Reset the codec to power on defaults */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
>  +
>  +       /* Power up CODEC */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
>  +
>  +       /* Audio Control 3 (master mode, fsref rate) */
>  +       reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
>  +       reg &= ~0xf800;
>  +       reg |= 0x0800; /* set master mode */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
>  +
>  +       /* Fill page 2 register cache */
>  +       for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
>  +               aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));
>  +
>  +       /* Register the sysfs files for debugging */
>  +       /* Create SysFS files */
>  +       rc = device_create_file(&spi->dev, &dev_attr_regs);
>  +       rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
>  +       rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
>  +       if (rc)
>  +               dev_info(&spi->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the of_soc helper about this codec */
>  +       of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
>  +                                 spi->dev.archdata.of_node);

I've been using asoc-v2 so it may not be possible to make this arch
independent in asoc-v1.  Maybe try porting this to asoc-v2 and see if
a bunch of the registration complexity disappears. Most of soc-of.c
should be unnecessary.

Liam, when is asoc-v2 going into mainline?


>  +
>  +       dev_dbg(&spi->dev, "SPI device initialized\n");
>  +       return 0;
>  +}
>  +
>  +static int aic26_spi_remove(struct spi_device *spi)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
>  +
>  +       kfree(aic26);
>  +
>  +       return 0;
>  +}
>  +
>  +static struct spi_driver aic26_spi = {
>  +       .driver = {
>  +               .name = "tlv320aic26",
>  +               .owner = THIS_MODULE,
>  +       },
>  +       .probe = aic26_spi_probe,
>  +       .remove = aic26_spi_remove,
>  +};
>  +
>  +static int __init aic26_init(void)
>  +{
>  +       return spi_register_driver(&aic26_spi);
>  +}
>  +module_init(aic26_init);
>  +
>  +static void __exit aic26_exit(void)
>  +{
>  +       spi_unregister_driver(&aic26_spi);
>  +}
>  +module_exit(aic26_exit);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  ASoC Codec driver for the TLV320AIC26 device.  This driver uses the ASoC
>  v1 API, so I don't expect it to get merged as-is, but I want to get it
>  out there for review.
>  ---
>
>   sound/soc/codecs/Kconfig       |    4
>   sound/soc/codecs/Makefile      |    2
>   sound/soc/codecs/tlv320aic26.c |  630 ++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 636 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
>  index 3903ab7..96c7bfe 100644
>  --- a/sound/soc/codecs/Kconfig
>  +++ b/sound/soc/codecs/Kconfig
>  @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA
>         bool
>         depends on SND_SOC_CS4270
>
>  +config SND_SOC_TLV320AIC26
>  +       tristate "TI TLB320AIC26 Codec support"
>  +       depends on SND_SOC && SPI
>  +
>   config SND_SOC_TLV320AIC3X
>         tristate
>         depends on SND_SOC && I2C
>  diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
>  index 4e1314c..ec0cd93 100644
>  --- a/sound/soc/codecs/Makefile
>  +++ b/sound/soc/codecs/Makefile
>  @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o
>   snd-soc-wm9712-objs := wm9712.o
>   snd-soc-wm9713-objs := wm9713.o
>   snd-soc-cs4270-objs := cs4270.o
>  +snd-soc-tlv320aic26-objs := tlv320aic26.o
>   snd-soc-tlv320aic3x-objs := tlv320aic3x.o
>
>   obj-$(CONFIG_SND_SOC_AC97_CODEC)       += snd-soc-ac97.o
>  @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753)  += snd-soc-wm8753.o
>   obj-$(CONFIG_SND_SOC_WM9712)   += snd-soc-wm9712.o
>   obj-$(CONFIG_SND_SOC_WM9713)   += snd-soc-wm9713.o
>   obj-$(CONFIG_SND_SOC_CS4270)   += snd-soc-cs4270.o
>  +obj-$(CONFIG_SND_SOC_TLV320AIC26)      += snd-soc-tlv320aic26.o
>   obj-$(CONFIG_SND_SOC_TLV320AIC3X)      += snd-soc-tlv320aic3x.o
>  diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
>  new file mode 100644
>  index 0000000..aee1dbc
>  --- /dev/null
>  +++ b/sound/soc/codecs/tlv320aic26.c
>  @@ -0,0 +1,630 @@
>  +/*
>  + * Texas Instruments TLV320AIC26 low power audio CODEC
>  + * ALSA SoC CODEC driver
>  + *
>  + * Copyright (C) 2008 Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/device.h>
>  +#include <linux/sysfs.h>
>  +#include <linux/spi/spi.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-dapm.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +
>  +/* AIC26 Registers */
>  +#define AIC26_READ_COMMAND_WORD(addr)  ((1 << 15) | (addr << 5))
>  +#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5))
>  +#define AIC26_PAGE_ADDR(page, offset)  ((page << 6) | offset)
>  +#define AIC26_NUM_REGS                 AIC26_PAGE_ADDR(3, 0)
>  +#define AIC26_REG_CACHE_SIZE           (0x20) /* only page 2 cached */
>  +#define AIC26_REG_IS_CACHED(addr)      ((addr & ~0x1f) == (2 << 6))
>  +#define AIC26_REG_CACHE_ADDR(addr)     (addr & 0x1f)
>  +
>  +/* Page 0: Auxillary data registers */
>  +#define AIC26_REG_BAT1                 AIC26_PAGE_ADDR(0, 0x05)
>  +#define AIC26_REG_BAT2                 AIC26_PAGE_ADDR(0, 0x06)
>  +#define AIC26_REG_AUX                  AIC26_PAGE_ADDR(0, 0x07)
>  +#define AIC26_REG_TEMP1                        AIC26_PAGE_ADDR(0, 0x09)
>  +#define AIC26_REG_TEMP2                        AIC26_PAGE_ADDR(0, 0x0A)
>  +
>  +/* Page 1: Auxillary control registers */
>  +#define AIC26_REG_AUX_ADC              AIC26_PAGE_ADDR(1, 0x00)
>  +#define AIC26_REG_STATUS               AIC26_PAGE_ADDR(1, 0x01)
>  +#define AIC26_REG_REFERENCE            AIC26_PAGE_ADDR(1, 0x03)
>  +#define AIC26_REG_RESET                        AIC26_PAGE_ADDR(1, 0x04)
>  +
>  +/* Page 2: Audio control registers */
>  +#define AIC26_REG_AUDIO_CTRL1          AIC26_PAGE_ADDR(2, 0x00)
>  +#define AIC26_REG_ADC_GAIN             AIC26_PAGE_ADDR(2, 0x01)
>  +#define AIC26_REG_DAC_GAIN             AIC26_PAGE_ADDR(2, 0x02)
>  +#define AIC26_REG_SIDETONE             AIC26_PAGE_ADDR(2, 0x03)
>  +#define AIC26_REG_AUDIO_CTRL2          AIC26_PAGE_ADDR(2, 0x04)
>  +#define AIC26_REG_POWER_CTRL           AIC26_PAGE_ADDR(2, 0x05)
>  +#define AIC26_REG_AUDIO_CTRL3          AIC26_PAGE_ADDR(2, 0x06)
>  +
>  +#define AIC26_REG_FILTER_COEFF_L_N0    AIC26_PAGE_ADDR(2, 0x07)
>  +#define AIC26_REG_FILTER_COEFF_L_N1    AIC26_PAGE_ADDR(2, 0x08)
>  +#define AIC26_REG_FILTER_COEFF_L_N2    AIC26_PAGE_ADDR(2, 0x09)
>  +#define AIC26_REG_FILTER_COEFF_L_N3    AIC26_PAGE_ADDR(2, 0x0A)
>  +#define AIC26_REG_FILTER_COEFF_L_N4    AIC26_PAGE_ADDR(2, 0x0B)
>  +#define AIC26_REG_FILTER_COEFF_L_N5    AIC26_PAGE_ADDR(2, 0x0C)
>  +#define AIC26_REG_FILTER_COEFF_L_D1    AIC26_PAGE_ADDR(2, 0x0D)
>  +#define AIC26_REG_FILTER_COEFF_L_D2    AIC26_PAGE_ADDR(2, 0x0E)
>  +#define AIC26_REG_FILTER_COEFF_L_D4    AIC26_PAGE_ADDR(2, 0x0F)
>  +#define AIC26_REG_FILTER_COEFF_L_D5    AIC26_PAGE_ADDR(2, 0x10)
>  +#define AIC26_REG_FILTER_COEFF_R_N0    AIC26_PAGE_ADDR(2, 0x11)
>  +#define AIC26_REG_FILTER_COEFF_R_N1    AIC26_PAGE_ADDR(2, 0x12)
>  +#define AIC26_REG_FILTER_COEFF_R_N2    AIC26_PAGE_ADDR(2, 0x13)
>  +#define AIC26_REG_FILTER_COEFF_R_N3    AIC26_PAGE_ADDR(2, 0x14)
>  +#define AIC26_REG_FILTER_COEFF_R_N4    AIC26_PAGE_ADDR(2, 0x15)
>  +#define AIC26_REG_FILTER_COEFF_R_N5    AIC26_PAGE_ADDR(2, 0x16)
>  +#define AIC26_REG_FILTER_COEFF_R_D1    AIC26_PAGE_ADDR(2, 0x17)
>  +#define AIC26_REG_FILTER_COEFF_R_D2    AIC26_PAGE_ADDR(2, 0x18)
>  +#define AIC26_REG_FILTER_COEFF_R_D4    AIC26_PAGE_ADDR(2, 0x19)
>  +#define AIC26_REG_FILTER_COEFF_R_D5    AIC26_PAGE_ADDR(2, 0x1A)
>  +
>  +#define AIC26_REG_PLL_PROG1            AIC26_PAGE_ADDR(2, 0x1B)
>  +#define AIC26_REG_PLL_PROG2            AIC26_PAGE_ADDR(2, 0x1C)
>  +#define AIC26_REG_AUDIO_CTRL4          AIC26_PAGE_ADDR(2, 0x1D)
>  +#define AIC26_REG_AUDIO_CTRL5          AIC26_PAGE_ADDR(2, 0x1E)
>  +
>  +#define AIC26_RATES    (SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
>  +                        SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
>  +                        SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
>  +                        SNDRV_PCM_RATE_48000)
>  +#define AIC26_FORMATS  (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
>  +                        SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
>  +
>  +/* fsref dividers; used in register 'Audio Control 1' */
>  +enum aic26_divisors {
>  +       AIC26_DIV_1     = 0,
>  +       AIC26_DIV_1_5   = 1,
>  +       AIC26_DIV_2     = 2,
>  +       AIC26_DIV_3     = 3,
>  +       AIC26_DIV_4     = 4,
>  +       AIC26_DIV_5     = 5,
>  +       AIC26_DIV_5_5   = 6,
>  +       AIC26_DIV_6     = 7,
>  +};
>  +
>  +/* Digital data format */
>  +enum aic26_datfm {
>  +       AIC26_DATFM_I2S         = 0 << 8,
>  +       AIC26_DATFM_DSP         = 1 << 8,
>  +       AIC26_DATFM_RIGHTJ      = 2 << 8, /* right justified */
>  +       AIC26_DATFM_LEFTJ       = 3 << 8, /* left justified */
>  +};
>  +
>  +/* Sample word length in bits; used in register 'Audio Control 1' */
>  +enum aic26_wlen {
>  +       AIC26_WLEN_16   = 0 << 10,
>  +       AIC26_WLEN_20   = 1 << 10,
>  +       AIC26_WLEN_24   = 2 << 10,
>  +       AIC26_WLEN_32   = 3 << 10,
>  +};
>  +
>  +/* AIC26 driver private data */
>  +struct aic26 {
>  +       struct spi_device *spi;
>  +       struct snd_soc_codec codec;
>  +       u16 reg_cache[AIC26_REG_CACHE_SIZE];    /* shadow registers */
>  +       int master;
>  +       int datfm;
>  +       int mclk;
>  +
>  +       /* Keyclick parameters */
>  +       int keyclick_amplitude;
>  +       int keyclick_freq;
>  +       int keyclick_len;
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * Register access routines
>  + */
>  +static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
>  +                                  unsigned int reg)
>  +{
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 *cache = codec->reg_cache;
>  +       u16 cmd, value;
>  +       u8 buffer[2];
>  +       int rc;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return 0;
>  +       }
>  +
>  +       /* Do SPI transfer; first 16bits are command; remaining is
>  +        * register contents */
>  +       cmd = AIC26_READ_COMMAND_WORD(reg);
>  +       buffer[0] = (cmd >> 8) & 0xff;
>  +       buffer[1] = cmd & 0xff;
>  +       rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
>  +       if (rc) {
>  +               dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
>  +               return -EIO;
>  +       }
>  +       value = (buffer[0] << 8) | buffer[1];
>  +
>  +       /* Update the cache before returning with the value */
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               cache[AIC26_REG_CACHE_ADDR(reg)] = value;
>  +       return value;
>  +}
>  +
>  +static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
>  +                                        unsigned int reg)
>  +{
>  +       u16 *cache = codec->reg_cache;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return 0;
>  +       }
>  +
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               return cache[AIC26_REG_CACHE_ADDR(reg)];
>  +
>  +       return aic26_reg_read(codec, reg);
>  +}
>  +
>  +static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
>  +                          unsigned int value)
>  +{
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 *cache = codec->reg_cache;
>  +       u16 cmd;
>  +       u8 buffer[4];
>  +       int rc;
>  +
>  +       if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
>  +               WARN_ON_ONCE(1);
>  +               return -EINVAL;
>  +       }
>  +
>  +       /* Do SPI transfer; first 16bits are command; remaining is data
>  +        * to write into register */
>  +       cmd = AIC26_WRITE_COMMAND_WORD(reg);
>  +       buffer[0] = (cmd >> 8) & 0xff;
>  +       buffer[1] = cmd & 0xff;
>  +       buffer[2] = value >> 8;
>  +       buffer[3] = value;
>  +       rc = spi_write(aic26->spi, buffer, 4);
>  +       if (rc) {
>  +               dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
>  +               return -EIO;
>  +       }
>  +
>  +       /* update cache before returning */
>  +       if (AIC26_REG_IS_CACHED(reg))
>  +               cache[AIC26_REG_CACHE_ADDR(reg)] = value;
>  +       return 0;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * Digital Audio Interface Operations
>  + */
>  +static int aic26_hw_params(struct snd_pcm_substream *substream,
>  +                          struct snd_pcm_hw_params *params)
>  +{
>  +       struct snd_soc_pcm_runtime *rtd = substream->private_data;
>  +       struct snd_soc_device *socdev = rtd->socdev;
>  +       struct snd_soc_codec *codec = socdev->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +       int fsref, divisor, wlen, pval, jval, dval, qval;
>  +       u16 reg;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
>  +               substream, params);
>  +       dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
>  +               params_format(params));
>  +
>  +       switch (params_rate(params)) {
>  +        case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
>  +        case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
>  +        case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
>  +        case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
>  +        case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
>  +        case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
>  +        case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
>  +        case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
>  +        case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
>  +       }
>  +
>  +       /* select data word length */
>  +       switch (params_format(params)) {
>  +        case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
>  +        case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
>  +        case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
>  +        case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
>  +       }
>  +
>  +       /* Configure PLL */
>  +       pval = 1;
>  +       jval = (fsref == 44100) ? 7 : 8;
>  +       dval = (fsref == 44100) ? 5264 : 1920;
>  +       qval = 0;
>  +       reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
>  +       aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
>  +       reg = dval << 2;
>  +       aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
>  +
>  +       /* Power up CODEC */
>  +       aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
>  +
>  +       /* Audio Control 3 (master mode, fsref rate) */
>  +       reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
>  +       reg &= ~0xf800;
>  +       if (aic26->master)
>  +               reg |= 0x0800;
>  +       if (fsref == 48000)
>  +               reg |= 0x2000;
>  +       aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
>  +
>  +       /* Audio Control 1 (FSref divisor) */
>  +       reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
>  +       reg &= ~0x0fff;
>  +       reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
>  +       aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
>  +
>  +       return 0;
>  +}
>  +
>  +/**
>  + * aic26_mute - Mute control to reduce noise when changing audio format
>  + */
>  +static int aic26_mute(struct snd_soc_codec_dai *dai, int mute)
>  +{
>  +       struct snd_soc_codec *codec = dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +       u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
>  +               dai, mute);
>  +
>  +       if (mute)
>  +               reg |= 0x8080;
>  +       else
>  +               reg &= ~0x8080;
>  +       aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
>  +
>  +       return 0;
>  +}
>  +
>  +static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
>  +                           int clk_id, unsigned int freq, int dir)
>  +{
>  +       struct snd_soc_codec *codec = codec_dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
>  +               " freq=%i, dir=%i)\n",
>  +               codec_dai, clk_id, freq, dir);
>  +
>  +       /* MCLK needs to fall between 2MHz and 50 MHz */
>  +       if ((freq < 2000000) || (freq > 50000000))
>  +               return -EINVAL;
>  +
>  +       aic26->mclk = freq;
>  +       return 0;
>  +}
>  +
>  +static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt)
>  +{
>  +       struct snd_soc_codec *codec = codec_dai->codec;
>  +       struct aic26 *aic26 = codec->private_data;
>  +
>  +       dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
>  +               codec_dai, fmt);
>  +
>  +       /* set master/slave audio interface */
>  +       switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
>  +        case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
>  +        //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
>  +       }
>  +
>  +       /* interface format */
>  +       switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
>  +        case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
>  +        case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
>  +        case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
>  +        case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
>  +        default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
>  +       }
>  +
>  +       return 0;
>  +}
>  +
>  +/* ---------------------------------------------------------------------
>  + * Digital Audio Interface Definition
>  + */
>  +struct snd_soc_codec_dai aic26_dai = {
>  +       .name = "tlv320aic26",
>  +       .playback = {
>  +               .stream_name = "Playback",
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = AIC26_RATES,
>  +               .formats = AIC26_FORMATS,
>  +       },
>  +       .capture = {
>  +               .stream_name = "Capture",
>  +               .channels_min = 2,
>  +               .channels_max = 2,
>  +               .rates = AIC26_RATES,
>  +               .formats = AIC26_FORMATS,
>  +       },
>  +       .ops = {
>  +               .hw_params = aic26_hw_params,
>  +       },
>  +       .dai_ops = {
>  +               .digital_mute = aic26_mute,
>  +               .set_sysclk = aic26_set_sysclk,
>  +               .set_fmt = aic26_set_fmt,
>  +       },
>  +};
>  +EXPORT_SYMBOL_GPL(aic26_dai);
>  +
>  +/* ---------------------------------------------------------------------
>  + * ALSA controls
>  + */
>  +static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
>  +static const struct soc_enum aic26_capture_src_enum =
>  +       SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
>  +
>  +static const struct snd_kcontrol_new aic26_snd_controls[] = {
>  +       /* Output */
>  +       SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
>  +       SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
>  +       SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
>  +       SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
>  +       SOC_ENUM("Capture Source", aic26_capture_src_enum),
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * SoC CODEC portion of driver: probe and release routines
>  + */
>  +static int aic26_probe(struct platform_device *pdev)
>  +{
>  +       struct snd_soc_device *socdev = platform_get_drvdata(pdev);
>  +       struct snd_soc_codec *codec;
>  +       struct snd_kcontrol *kcontrol;
>  +       struct aic26 *aic26;
>  +       int i, ret, err;
>  +
>  +       dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
>  +       dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
>  +       dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
>  +
>  +       /* Fetch the relevant aic26 private data here (it's already been
>  +        * stored in the .codec pointer) */
>  +       aic26 = socdev->codec_data;
>  +       if (aic26 == NULL) {
>  +               dev_err(&pdev->dev, "aic26: missing codec pointer\n");
>  +               return -ENODEV;
>  +       }
>  +       codec = &aic26->codec;
>  +       socdev->codec = codec;
>  +
>  +       dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
>  +               &pdev->dev, socdev->dev);
>  +       /* register pcms */
>  +       ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
>  +       if (ret < 0) {
>  +               dev_err(&pdev->dev, "aic26: failed to create pcms\n");
>  +               return -ENODEV;
>  +       }
>  +
>  +       /* register controls */
>  +       dev_dbg(&pdev->dev, "Registering controls\n");
>  +       for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
>  +               kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
>  +               err = snd_ctl_add(codec->card, kcontrol);
>  +               WARN_ON(err < 0);
>  +       }
>  +
>  +       /* CODEC is setup, we can register the card now */
>  +       dev_dbg(&pdev->dev, "Registering card\n");
>  +       ret = snd_soc_register_card(socdev);
>  +       if (ret < 0) {
>  +               dev_err(&pdev->dev, "aic26: failed to register card\n");
>  +               goto card_err;
>  +       }
>  +       return 0;
>  +
>  + card_err:
>  +       snd_soc_free_pcms(socdev);
>  +       return ret;
>  +}
>  +
>  +static int aic26_remove(struct platform_device *pdev)
>  +{
>  +       struct snd_soc_device *socdev = platform_get_drvdata(pdev);
>  +       snd_soc_free_pcms(socdev);
>  +       return 0;
>  +}
>  +
>  +struct snd_soc_codec_device aic26_soc_codec_dev = {
>  +       .probe = aic26_probe,
>  +       .remove = aic26_remove,
>  +};
>  +
>  +/* ---------------------------------------------------------------------
>  + * SPI device portion of driver: sysfs files for debugging
>  + */
>  +
>  +static ssize_t aic26_regs_show(struct device *dev,
>  +                               struct device_attribute *attr, char *buf)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       char *idx = buf;
>  +       int cache_flag, addr, page, i, reg;
>  +
>  +       cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
>  +
>  +       for (page = 0; page < 3; page++) {
>  +               for (i = 0; i < 0x20; i++) {
>  +                       addr = AIC26_PAGE_ADDR(page, i);
>  +                       if (i % 8 == 0)
>  +                               idx += sprintf(idx, "%i:%.2i:", page,i);
>  +                       if (cache_flag)
>  +                               reg = aic26_reg_read_cache(&aic26->codec, addr);
>  +                       else
>  +                               reg = aic26_reg_read(&aic26->codec, addr);
>  +                       idx += sprintf(idx, " %.4x", reg);
>  +                       if (i % 8 == 7)
>  +                               idx += sprintf(idx, "\n");
>  +               }
>  +       }
>  +       return idx - buf;
>  +}
>  +
>  +static ssize_t aic26_keyclick_show(struct device *dev,
>  +                                  struct device_attribute *attr, char *buf)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       int val, amp, freq, len;
>  +
>  +       val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
>  +       amp = (val >> 12) & 0x7;
>  +       freq = (125 << ((val >> 8) & 0x7)) >> 1;
>  +       len = 2 * (1 +((val >> 8) & 0xf));
>  +
>  +       return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
>  +}
>  +
>  +/* Any write to the keyclick attribute will trigger the keyclick */
>  +static ssize_t aic26_keyclick_set(struct device *dev,
>  +                                 struct device_attribute *attr,
>  +                                 const char *buf, size_t count)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(dev);
>  +       int val;
>  +
>  +       val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
>  +       val |= 0x8000;
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
>  +
>  +       return count;
>  +}
>  +
>  +DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL);
>  +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL);
>  +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
>  +
>  +/* ---------------------------------------------------------------------
>  + * SPI device portion of driver: probe and release routines and SPI
>  + *                              driver registration.
>  + */
>  +static int aic26_spi_probe(struct spi_device *spi)
>  +{
>  +       struct aic26 *aic26;
>  +       int rc, i, reg;
>  +
>  +       dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
>  +
>  +       /* Allocate driver data */
>  +       aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
>  +       if (!aic26)
>  +               return -ENOMEM;
>  +
>  +       /* Initialize the driver data */
>  +       aic26->spi = spi;
>  +       dev_set_drvdata(&spi->dev, aic26);
>  +
>  +       /* Setup what we can in the codec structure so that the register
>  +        * access functions will work as expected.  More will be filled
>  +        * out when it is probed by the SoC CODEC part of this driver */
>  +       aic26->codec.private_data = aic26;
>  +       aic26->codec.name = "aic26";
>  +       aic26->codec.owner = THIS_MODULE;
>  +       aic26->codec.dai = &aic26_dai;
>  +       aic26->codec.num_dai = 1;
>  +       aic26->codec.read = aic26_reg_read;
>  +       aic26->codec.write = aic26_reg_write;
>  +       aic26->master = 1;
>  +       mutex_init(&aic26->codec.mutex);
>  +       INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
>  +       INIT_LIST_HEAD(&aic26->codec.dapm_paths);
>  +       aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
>  +       aic26->codec.reg_cache = aic26->reg_cache;
>  +
>  +       /* Reset the codec to power on defaults */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
>  +
>  +       /* Power up CODEC */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
>  +
>  +       /* Audio Control 3 (master mode, fsref rate) */
>  +       reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
>  +       reg &= ~0xf800;
>  +       reg |= 0x0800; /* set master mode */
>  +       aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
>  +
>  +       /* Fill page 2 register cache */
>  +       for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
>  +               aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));
>  +
>  +       /* Register the sysfs files for debugging */
>  +       /* Create SysFS files */
>  +       rc = device_create_file(&spi->dev, &dev_attr_regs);
>  +       rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
>  +       rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
>  +       if (rc)
>  +               dev_info(&spi->dev, "error creating sysfs files\n");
>  +
>  +       /* Tell the of_soc helper about this codec */
>  +       of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
>  +                                 spi->dev.archdata.of_node);

I've been using asoc-v2 so it may not be possible to make this arch
independent in asoc-v1.  Maybe try porting this to asoc-v2 and see if
a bunch of the registration complexity disappears. Most of soc-of.c
should be unnecessary.

Liam, when is asoc-v2 going into mainline?


>  +
>  +       dev_dbg(&spi->dev, "SPI device initialized\n");
>  +       return 0;
>  +}
>  +
>  +static int aic26_spi_remove(struct spi_device *spi)
>  +{
>  +       struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
>  +
>  +       kfree(aic26);
>  +
>  +       return 0;
>  +}
>  +
>  +static struct spi_driver aic26_spi = {
>  +       .driver = {
>  +               .name = "tlv320aic26",
>  +               .owner = THIS_MODULE,
>  +       },
>  +       .probe = aic26_spi_probe,
>  +       .remove = aic26_spi_remove,
>  +};
>  +
>  +static int __init aic26_init(void)
>  +{
>  +       return spi_register_driver(&aic26_spi);
>  +}
>  +module_init(aic26_init);
>  +
>  +static void __exit aic26_exit(void)
>  +{
>  +       spi_unregister_driver(&aic26_spi);
>  +}
>  +module_exit(aic26_exit);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Liam Girdwood]

On Wed, 2008-07-02 at 09:52 -0400, Jon Smirl wrote:
> 
> I've been using asoc-v2 so it may not be possible to make this arch
> independent in asoc-v1.  Maybe try porting this to asoc-v2 and see if
> a bunch of the registration complexity disappears. Most of soc-of.c
> should be unnecessary.
> 
> Liam, when is asoc-v2 going into mainline?

Hopefully over the next 2 merge windows. I'm about to start queueing
patches....

This also assumes Mark and I remain at our current work load levels.

Liam

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Liam Girdwood]

On Wed, 2008-07-02 at 09:52 -0400, Jon Smirl wrote:
> 
> I've been using asoc-v2 so it may not be possible to make this arch
> independent in asoc-v1.  Maybe try porting this to asoc-v2 and see if
> a bunch of the registration complexity disappears. Most of soc-of.c
> should be unnecessary.
> 
> Liam, when is asoc-v2 going into mainline?

Hopefully over the next 2 merge windows. I'm about to start queueing
patches....

This also assumes Mark and I remain at our current work load levels.

Liam

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:

> +static ssize_t aic26_keyclick_show(struct device *dev,
> +				   struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val, amp, freq, len;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	amp = (val >> 12) & 0x7;
> +	freq = (125 << ((val >> 8) & 0x7)) >> 1;
> +	len = 2 * (1 +((val >> 8) & 0xf));
> +
> +	return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
> +}

It might be nice if these parameters were exposed as controls if they
can be configured, though obviously that's not essential.

> +	/* Tell the of_soc helper about this codec */
> +	of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
> +				  spi->dev.archdata.of_node);

Does this need to be ifdefed out if of_snd_soc_register_codec() is not
in use or is the function stubbed out if it's not in use?  Stubbing it
out would be cleaner.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Mark Brown]

On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:

> +static ssize_t aic26_keyclick_show(struct device *dev,
> +				   struct device_attribute *attr, char *buf)
> +{
> +	struct aic26 *aic26 = dev_get_drvdata(dev);
> +	int val, amp, freq, len;
> +
> +	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
> +	amp = (val >> 12) & 0x7;
> +	freq = (125 << ((val >> 8) & 0x7)) >> 1;
> +	len = 2 * (1 +((val >> 8) & 0xf));
> +
> +	return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
> +}

It might be nice if these parameters were exposed as controls if they
can be configured, though obviously that's not essential.

> +	/* Tell the of_soc helper about this codec */
> +	of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
> +				  spi->dev.archdata.of_node);

Does this need to be ifdefed out if of_snd_soc_register_codec() is not
in use or is the function stubbed out if it's not in use?  Stubbing it
out would be cleaner.

Re: [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Fri, Jul 4, 2008 at 2:49 PM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:
>> +     /* Tell the of_soc helper about this codec */
>> +     of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
>> +                               spi->dev.archdata.of_node);
>
> Does this need to be ifdefed out if of_snd_soc_register_codec() is not
> in use or is the function stubbed out if it's not in use?  Stubbing it
> out would be cleaner.

yeah, probably.  This is a stop-gap measure anyway until ASoC v2 is merged.

g.

-- 
Grant Likely, B.Sc., P.Eng.
Secret Lab Technologies Ltd.

Re: [alsa-devel] [PATCH 3/3] ALSA SoC: Add Texas Instruments TLV320AIC26 codec driver

[Grant Likely]

On Fri, Jul 4, 2008 at 2:49 PM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:
>> +     /* Tell the of_soc helper about this codec */
>> +     of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
>> +                               spi->dev.archdata.of_node);
>
> Does this need to be ifdefed out if of_snd_soc_register_codec() is not
> in use or is the function stubbed out if it's not in use?  Stubbing it
> out would be cleaner.

yeah, probably.  This is a stop-gap measure anyway until ASoC v2 is merged.

g.

-- 
Grant Likely, B.Sc., P.Eng.
Secret Lab Technologies Ltd.

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Takashi Iwai]

At Tue, 01 Jul 2008 17:53:30 -0600,
Grant Likely wrote:
> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
> index 18f28ac..c5736e5 100644
> --- a/sound/soc/Kconfig
> +++ b/sound/soc/Kconfig
> @@ -23,6 +23,12 @@ config SND_SOC
>  	  This ASoC audio support can also be built as a module.  If so, the module
>  	  will be called snd-soc-core.
>  
> +config SND_SOC_OF
> +	tristate "OF helpers for SoC audio support"
> +	depends on SND_SOC
> +	---help---
> +	  Add support for OpenFirmware device tree descriptions of sound device
> +

This is a helper module and not necessarily manually selectable.
Better to make the other driver selecting this.


> diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
> new file mode 100644
> index 0000000..9694979
> --- /dev/null
> +++ b/sound/soc/soc-of.c
(snip)
> +DEFINE_MUTEX(of_snd_soc_mutex);
> +LIST_HEAD(of_snd_soc_device_list);

Missing static.


Takashi

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Takashi Iwai]

At Tue, 01 Jul 2008 17:53:30 -0600,
Grant Likely wrote:
> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
> index 18f28ac..c5736e5 100644
> --- a/sound/soc/Kconfig
> +++ b/sound/soc/Kconfig
> @@ -23,6 +23,12 @@ config SND_SOC
>  	  This ASoC audio support can also be built as a module.  If so, the module
>  	  will be called snd-soc-core.
>  
> +config SND_SOC_OF
> +	tristate "OF helpers for SoC audio support"
> +	depends on SND_SOC
> +	---help---
> +	  Add support for OpenFirmware device tree descriptions of sound device
> +

This is a helper module and not necessarily manually selectable.
Better to make the other driver selecting this.


> diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
> new file mode 100644
> index 0000000..9694979
> --- /dev/null
> +++ b/sound/soc/soc-of.c
(snip)
> +DEFINE_MUTEX(of_snd_soc_mutex);
> +LIST_HEAD(of_snd_soc_device_list);

Missing static.


Takashi

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
> At Tue, 01 Jul 2008 17:53:30 -0600,
> Grant Likely wrote:
> > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
> > index 18f28ac..c5736e5 100644
> > --- a/sound/soc/Kconfig
> > +++ b/sound/soc/Kconfig
> > @@ -23,6 +23,12 @@ config SND_SOC
> >  	  This ASoC audio support can also be built as a module.  If so, the module
> >  	  will be called snd-soc-core.
> >  
> > +config SND_SOC_OF
> > +	tristate "OF helpers for SoC audio support"
> > +	depends on SND_SOC
> > +	---help---
> > +	  Add support for OpenFirmware device tree descriptions of sound device
> > +
> 
> This is a helper module and not necessarily manually selectable.
> Better to make the other driver selecting this.

Yes, you're right.  I hadn't put too much thought into the Kconfig stuff
for this patch because, at the moment, I'm viewing this as a temporary
solution until ASoCv2 is merged.

Which raises another question: Liam and Mark, what is your opinion on
merging this driver?  Is it something that you would merge with the v1
API and then rework it when v2 is merged?  Or would you rather it be
kept out until v2 is ready?  I haven't even looked at the v2 API yet, so
I don't know how much rework is involved.

Cheers,
g.

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
> At Tue, 01 Jul 2008 17:53:30 -0600,
> Grant Likely wrote:
> > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
> > index 18f28ac..c5736e5 100644
> > --- a/sound/soc/Kconfig
> > +++ b/sound/soc/Kconfig
> > @@ -23,6 +23,12 @@ config SND_SOC
> >  	  This ASoC audio support can also be built as a module.  If so, the module
> >  	  will be called snd-soc-core.
> >  
> > +config SND_SOC_OF
> > +	tristate "OF helpers for SoC audio support"
> > +	depends on SND_SOC
> > +	---help---
> > +	  Add support for OpenFirmware device tree descriptions of sound device
> > +
> 
> This is a helper module and not necessarily manually selectable.
> Better to make the other driver selecting this.

Yes, you're right.  I hadn't put too much thought into the Kconfig stuff
for this patch because, at the moment, I'm viewing this as a temporary
solution until ASoCv2 is merged.

Which raises another question: Liam and Mark, what is your opinion on
merging this driver?  Is it something that you would merge with the v1
API and then rework it when v2 is merged?  Or would you rather it be
kept out until v2 is ready?  I haven't even looked at the v2 API yet, so
I don't know how much rework is involved.

Cheers,
g.

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Liam Girdwood]

On Wed, 2008-07-02 at 09:48 -0600, Grant Likely wrote:
> On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
> > 
> > This is a helper module and not necessarily manually selectable.
> > Better to make the other driver selecting this.
> 
> Yes, you're right.  I hadn't put too much thought into the Kconfig stuff
> for this patch because, at the moment, I'm viewing this as a temporary
> solution until ASoCv2 is merged.
> 
> Which raises another question: Liam and Mark, what is your opinion on
> merging this driver?  Is it something that you would merge with the v1
> API and then rework it when v2 is merged?  Or would you rather it be
> kept out until v2 is ready?  I haven't even looked at the v2 API yet, so
> I don't know how much rework is involved.
> 

I'm happy with this atm for v1. We can then rework for V2.

Liam

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Liam Girdwood]

On Wed, 2008-07-02 at 09:48 -0600, Grant Likely wrote:
> On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
> > 
> > This is a helper module and not necessarily manually selectable.
> > Better to make the other driver selecting this.
> 
> Yes, you're right.  I hadn't put too much thought into the Kconfig stuff
> for this patch because, at the moment, I'm viewing this as a temporary
> solution until ASoCv2 is merged.
> 
> Which raises another question: Liam and Mark, what is your opinion on
> merging this driver?  Is it something that you would merge with the v1
> API and then rework it when v2 is merged?  Or would you rather it be
> kept out until v2 is ready?  I haven't even looked at the v2 API yet, so
> I don't know how much rework is involved.
> 

I'm happy with this atm for v1. We can then rework for V2.

Liam

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  Simple utility layer for creating ASoC machine instances based on data
>  in the OpenFirmware device tree.  OF aware platform drivers and codec
>  drivers register themselves with this framework and the framework
>  automatically instantiates a machine driver.
>
>  This is most likely temporary glue code to work around limitations in
>  the ASoC v1 framework.  I expect ASoC v2 won't need this.
>  ---
>
>   sound/soc/Kconfig  |    6 ++
>   sound/soc/Makefile |    1
>   sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 178 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
>  index 18f28ac..c5736e5 100644
>  --- a/sound/soc/Kconfig
>  +++ b/sound/soc/Kconfig
>  @@ -23,6 +23,12 @@ config SND_SOC
>           This ASoC audio support can also be built as a module.  If so, the module
>           will be called snd-soc-core.
>
>  +config SND_SOC_OF
>  +       tristate "OF helpers for SoC audio support"
>  +       depends on SND_SOC
>  +       ---help---
>  +         Add support for OpenFirmware device tree descriptions of sound device
>  +
>   # All the supported Soc's
>   source "sound/soc/at91/Kconfig"
>   source "sound/soc/pxa/Kconfig"
>  diff --git a/sound/soc/Makefile b/sound/soc/Makefile
>  index 782db21..191c2e5 100644
>  --- a/sound/soc/Makefile
>  +++ b/sound/soc/Makefile
>  @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
>
>   obj-$(CONFIG_SND_SOC)  += snd-soc-core.o
>   obj-$(CONFIG_SND_SOC)  += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
>  +obj-$(CONFIG_SND_SOC_OF)       += soc-of.o
>  diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
>  new file mode 100644
>  index 0000000..9694979
>  --- /dev/null
>  +++ b/sound/soc/soc-of.c
>  @@ -0,0 +1,171 @@
>  +/*
>  + * OF helpers for ALSA SoC Layer
>  + *
>  + * Copyright (C) 2008, Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/bitops.h>
>  +#include <linux/platform_device.h>
>  +#include <linux/of.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
>  +
>  +DEFINE_MUTEX(of_snd_soc_mutex);
>  +LIST_HEAD(of_snd_soc_device_list);
>  +static int of_snd_soc_next_index;
>  +
>  +struct of_snd_soc_device {
>  +       int id;
>  +       struct list_head list;
>  +       struct snd_soc_device device;
>  +       struct snd_soc_machine machine;
>  +       struct snd_soc_dai_link dai_link;
>  +       struct platform_device *pdev;
>  +       struct device_node *platform_node;
>  +       struct device_node *codec_node;
>  +};
>  +
>  +static struct snd_soc_ops of_snd_soc_ops = {
>  +};
>  +
>  +static struct of_snd_soc_device *
>  +of_snd_soc_get_device(struct device_node *codec_node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +
>  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
>  +               if (of_soc->codec_node == codec_node)
>  +                       return of_soc;
>  +       }
>  +
>  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
>  +       if (!of_soc)
>  +               return NULL;
>  +
>  +       /* Initialize the structure and add it to the global list */
>  +       of_soc->codec_node = codec_node;
>  +       of_soc->id = of_snd_soc_next_index++;
>  +       of_soc->machine.dai_link = &of_soc->dai_link;
>  +       of_soc->machine.num_links = 1;
>  +       of_soc->device.machine = &of_soc->machine;
>  +       of_soc->dai_link.ops = &of_snd_soc_ops;
>  +       list_add(&of_soc->list, &of_snd_soc_device_list);
>  +
>  +       return of_soc;
>  +}
>  +
>  +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
>  +{
>  +       struct platform_device *pdev;
>  +       int rc;
>  +
>  +       /* Only register the device if both the codec and platform have
>  +        * been registered */
>  +       if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
>  +               return;
>  +
>  +       pr_info("platform<-->codec match achieved; registering machine\n");
>  +
>  +       pdev = platform_device_alloc("soc-audio", of_soc->id);
>  +       if (!pdev) {
>  +               pr_err("of_soc: platform_device_alloc() failed\n");
>  +               return;
>  +       }
>  +
>  +       pdev->dev.platform_data = of_soc;
>  +       platform_set_drvdata(pdev, &of_soc->device);
>  +       of_soc->device.dev = &pdev->dev;
>  +
>  +       /* The ASoC device is complete; register it */
>  +       rc = platform_device_add(pdev);
>  +       if (rc) {
>  +               pr_err("of_soc: platform_device_add() failed\n");
>  +               return;
>  +       }

Is there a driver for this device?  Is another file missing?
sof_of.h is missing too.

>  +
>  +}
>  +
>  +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
>  +                             void *codec_data, struct snd_soc_codec_dai *dai,
>  +                             struct device_node *node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       int rc = 0;
>  +
>  +       pr_info("registering ASoC codec driver: %s\n", node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       /* Store the codec data */
>  +       of_soc->device.codec_data = codec_data;
>  +       of_soc->device.codec_dev = codec_dev;
>  +       of_soc->dai_link.name = node->name;
>  +       of_soc->dai_link.stream_name = node->name;
>  +       of_soc->dai_link.codec_dai = dai;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
>  +
>  +int of_snd_soc_register_platform(struct snd_soc_platform *platform,
>  +                                struct device_node *node,
>  +                                struct snd_soc_cpu_dai *cpu_dai)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       struct device_node *codec_node;
>  +       const phandle *handle;
>  +       int len, rc = 0;
>  +
>  +       pr_info("registering ASoC platform driver: %s\n", node->full_name);
>  +
>  +       handle = of_get_property(node, "codec-handle", &len);
>  +       if (!handle || len < sizeof(handle))
>  +               return -ENODEV;
>  +       codec_node = of_find_node_by_phandle(*handle);
>  +       if (!codec_node)
>  +               return -ENODEV;
>  +       pr_info("looking for codec: %s\n", codec_node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(codec_node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       of_soc->platform_node = node;
>  +       of_soc->dai_link.cpu_dai = cpu_dai;
>  +       of_soc->device.platform = platform;
>  +       of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  Simple utility layer for creating ASoC machine instances based on data
>  in the OpenFirmware device tree.  OF aware platform drivers and codec
>  drivers register themselves with this framework and the framework
>  automatically instantiates a machine driver.
>
>  This is most likely temporary glue code to work around limitations in
>  the ASoC v1 framework.  I expect ASoC v2 won't need this.
>  ---
>
>   sound/soc/Kconfig  |    6 ++
>   sound/soc/Makefile |    1
>   sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 178 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
>  index 18f28ac..c5736e5 100644
>  --- a/sound/soc/Kconfig
>  +++ b/sound/soc/Kconfig
>  @@ -23,6 +23,12 @@ config SND_SOC
>           This ASoC audio support can also be built as a module.  If so, the module
>           will be called snd-soc-core.
>
>  +config SND_SOC_OF
>  +       tristate "OF helpers for SoC audio support"
>  +       depends on SND_SOC
>  +       ---help---
>  +         Add support for OpenFirmware device tree descriptions of sound device
>  +
>   # All the supported Soc's
>   source "sound/soc/at91/Kconfig"
>   source "sound/soc/pxa/Kconfig"
>  diff --git a/sound/soc/Makefile b/sound/soc/Makefile
>  index 782db21..191c2e5 100644
>  --- a/sound/soc/Makefile
>  +++ b/sound/soc/Makefile
>  @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
>
>   obj-$(CONFIG_SND_SOC)  += snd-soc-core.o
>   obj-$(CONFIG_SND_SOC)  += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
>  +obj-$(CONFIG_SND_SOC_OF)       += soc-of.o
>  diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
>  new file mode 100644
>  index 0000000..9694979
>  --- /dev/null
>  +++ b/sound/soc/soc-of.c
>  @@ -0,0 +1,171 @@
>  +/*
>  + * OF helpers for ALSA SoC Layer
>  + *
>  + * Copyright (C) 2008, Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/bitops.h>
>  +#include <linux/platform_device.h>
>  +#include <linux/of.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
>  +
>  +DEFINE_MUTEX(of_snd_soc_mutex);
>  +LIST_HEAD(of_snd_soc_device_list);
>  +static int of_snd_soc_next_index;
>  +
>  +struct of_snd_soc_device {
>  +       int id;
>  +       struct list_head list;
>  +       struct snd_soc_device device;
>  +       struct snd_soc_machine machine;
>  +       struct snd_soc_dai_link dai_link;
>  +       struct platform_device *pdev;
>  +       struct device_node *platform_node;
>  +       struct device_node *codec_node;
>  +};
>  +
>  +static struct snd_soc_ops of_snd_soc_ops = {
>  +};
>  +
>  +static struct of_snd_soc_device *
>  +of_snd_soc_get_device(struct device_node *codec_node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +
>  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
>  +               if (of_soc->codec_node == codec_node)
>  +                       return of_soc;
>  +       }
>  +
>  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
>  +       if (!of_soc)
>  +               return NULL;
>  +
>  +       /* Initialize the structure and add it to the global list */
>  +       of_soc->codec_node = codec_node;
>  +       of_soc->id = of_snd_soc_next_index++;
>  +       of_soc->machine.dai_link = &of_soc->dai_link;
>  +       of_soc->machine.num_links = 1;
>  +       of_soc->device.machine = &of_soc->machine;
>  +       of_soc->dai_link.ops = &of_snd_soc_ops;
>  +       list_add(&of_soc->list, &of_snd_soc_device_list);
>  +
>  +       return of_soc;
>  +}
>  +
>  +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
>  +{
>  +       struct platform_device *pdev;
>  +       int rc;
>  +
>  +       /* Only register the device if both the codec and platform have
>  +        * been registered */
>  +       if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
>  +               return;
>  +
>  +       pr_info("platform<-->codec match achieved; registering machine\n");
>  +
>  +       pdev = platform_device_alloc("soc-audio", of_soc->id);
>  +       if (!pdev) {
>  +               pr_err("of_soc: platform_device_alloc() failed\n");
>  +               return;
>  +       }
>  +
>  +       pdev->dev.platform_data = of_soc;
>  +       platform_set_drvdata(pdev, &of_soc->device);
>  +       of_soc->device.dev = &pdev->dev;
>  +
>  +       /* The ASoC device is complete; register it */
>  +       rc = platform_device_add(pdev);
>  +       if (rc) {
>  +               pr_err("of_soc: platform_device_add() failed\n");
>  +               return;
>  +       }

Is there a driver for this device?  Is another file missing?
sof_of.h is missing too.

>  +
>  +}
>  +
>  +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
>  +                             void *codec_data, struct snd_soc_codec_dai *dai,
>  +                             struct device_node *node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       int rc = 0;
>  +
>  +       pr_info("registering ASoC codec driver: %s\n", node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       /* Store the codec data */
>  +       of_soc->device.codec_data = codec_data;
>  +       of_soc->device.codec_dev = codec_dev;
>  +       of_soc->dai_link.name = node->name;
>  +       of_soc->dai_link.stream_name = node->name;
>  +       of_soc->dai_link.codec_dai = dai;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
>  +
>  +int of_snd_soc_register_platform(struct snd_soc_platform *platform,
>  +                                struct device_node *node,
>  +                                struct snd_soc_cpu_dai *cpu_dai)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       struct device_node *codec_node;
>  +       const phandle *handle;
>  +       int len, rc = 0;
>  +
>  +       pr_info("registering ASoC platform driver: %s\n", node->full_name);
>  +
>  +       handle = of_get_property(node, "codec-handle", &len);
>  +       if (!handle || len < sizeof(handle))
>  +               return -ENODEV;
>  +       codec_node = of_find_node_by_phandle(*handle);
>  +       if (!codec_node)
>  +               return -ENODEV;
>  +       pr_info("looking for codec: %s\n", codec_node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(codec_node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       of_soc->platform_node = node;
>  +       of_soc->dai_link.cpu_dai = cpu_dai;
>  +       of_soc->device.platform = platform;
>  +       of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  Simple utility layer for creating ASoC machine instances based on data
>  in the OpenFirmware device tree.  OF aware platform drivers and codec
>  drivers register themselves with this framework and the framework
>  automatically instantiates a machine driver.
>
>  This is most likely temporary glue code to work around limitations in
>  the ASoC v1 framework.  I expect ASoC v2 won't need this.
>  ---
>
>   sound/soc/Kconfig  |    6 ++
>   sound/soc/Makefile |    1
>   sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 178 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
>  index 18f28ac..c5736e5 100644
>  --- a/sound/soc/Kconfig
>  +++ b/sound/soc/Kconfig
>  @@ -23,6 +23,12 @@ config SND_SOC
>           This ASoC audio support can also be built as a module.  If so, the module
>           will be called snd-soc-core.
>
>  +config SND_SOC_OF
>  +       tristate "OF helpers for SoC audio support"
>  +       depends on SND_SOC
>  +       ---help---
>  +         Add support for OpenFirmware device tree descriptions of sound device
>  +
>   # All the supported Soc's
>   source "sound/soc/at91/Kconfig"
>   source "sound/soc/pxa/Kconfig"
>  diff --git a/sound/soc/Makefile b/sound/soc/Makefile
>  index 782db21..191c2e5 100644
>  --- a/sound/soc/Makefile
>  +++ b/sound/soc/Makefile
>  @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
>
>   obj-$(CONFIG_SND_SOC)  += snd-soc-core.o
>   obj-$(CONFIG_SND_SOC)  += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
>  +obj-$(CONFIG_SND_SOC_OF)       += soc-of.o
>  diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
>  new file mode 100644
>  index 0000000..9694979
>  --- /dev/null
>  +++ b/sound/soc/soc-of.c
>  @@ -0,0 +1,171 @@
>  +/*
>  + * OF helpers for ALSA SoC Layer
>  + *
>  + * Copyright (C) 2008, Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/bitops.h>
>  +#include <linux/platform_device.h>
>  +#include <linux/of.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
>  +
>  +DEFINE_MUTEX(of_snd_soc_mutex);
>  +LIST_HEAD(of_snd_soc_device_list);
>  +static int of_snd_soc_next_index;
>  +
>  +struct of_snd_soc_device {
>  +       int id;
>  +       struct list_head list;
>  +       struct snd_soc_device device;
>  +       struct snd_soc_machine machine;
>  +       struct snd_soc_dai_link dai_link;
>  +       struct platform_device *pdev;
>  +       struct device_node *platform_node;
>  +       struct device_node *codec_node;
>  +};
>  +
>  +static struct snd_soc_ops of_snd_soc_ops = {
>  +};
>  +
>  +static struct of_snd_soc_device *
>  +of_snd_soc_get_device(struct device_node *codec_node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +
>  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
>  +               if (of_soc->codec_node == codec_node)
>  +                       return of_soc;
>  +       }
>  +
>  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
>  +       if (!of_soc)
>  +               return NULL;
>  +
>  +       /* Initialize the structure and add it to the global list */
>  +       of_soc->codec_node = codec_node;
>  +       of_soc->id = of_snd_soc_next_index++;
>  +       of_soc->machine.dai_link = &of_soc->dai_link;
>  +       of_soc->machine.num_links = 1;
>  +       of_soc->device.machine = &of_soc->machine;
>  +       of_soc->dai_link.ops = &of_snd_soc_ops;
>  +       list_add(&of_soc->list, &of_snd_soc_device_list);
>  +
>  +       return of_soc;
>  +}

Isn't this performing the same basic function as this code (except for
spi)? Should this list be maintained in alsa or should there be an
equivalent for searching the spi bus? If you follow the link to the
codec node and get it's parent you know which bus to search. But it
might be simpler to just search the buses sequentially for the node.

static int of_dev_node_match(struct device *dev, void *data)
{
        return dev->archdata.of_node == data;
}

struct i2c_client *of_find_i2c_device_by_node(struct device_node *node)
{
	struct device *dev;
	
	dev = bus_find_device(&i2c_bus_type, NULL, node,
					 of_dev_node_match);
	if (!dev)
		return NULL;
		
	return to_i2c_client(dev);
}
EXPORT_SYMBOL(of_find_i2c_device_by_node);



>  +
>  +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
>  +{
>  +       struct platform_device *pdev;
>  +       int rc;
>  +
>  +       /* Only register the device if both the codec and platform have
>  +        * been registered */
>  +       if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
>  +               return;
>  +
>  +       pr_info("platform<-->codec match achieved; registering machine\n");
>  +
>  +       pdev = platform_device_alloc("soc-audio", of_soc->id);
>  +       if (!pdev) {
>  +               pr_err("of_soc: platform_device_alloc() failed\n");
>  +               return;
>  +       }
>  +
>  +       pdev->dev.platform_data = of_soc;
>  +       platform_set_drvdata(pdev, &of_soc->device);
>  +       of_soc->device.dev = &pdev->dev;
>  +
>  +       /* The ASoC device is complete; register it */
>  +       rc = platform_device_add(pdev);
>  +       if (rc) {
>  +               pr_err("of_soc: platform_device_add() failed\n");
>  +               return;
>  +       }
>  +
>  +}
>  +
>  +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
>  +                             void *codec_data, struct snd_soc_codec_dai *dai,
>  +                             struct device_node *node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       int rc = 0;
>  +
>  +       pr_info("registering ASoC codec driver: %s\n", node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       /* Store the codec data */
>  +       of_soc->device.codec_data = codec_data;
>  +       of_soc->device.codec_dev = codec_dev;
>  +       of_soc->dai_link.name = node->name;
>  +       of_soc->dai_link.stream_name = node->name;
>  +       of_soc->dai_link.codec_dai = dai;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
>  +
>  +int of_snd_soc_register_platform(struct snd_soc_platform *platform,
>  +                                struct device_node *node,
>  +                                struct snd_soc_cpu_dai *cpu_dai)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       struct device_node *codec_node;
>  +       const phandle *handle;
>  +       int len, rc = 0;
>  +
>  +       pr_info("registering ASoC platform driver: %s\n", node->full_name);
>  +
>  +       handle = of_get_property(node, "codec-handle", &len);
>  +       if (!handle || len < sizeof(handle))
>  +               return -ENODEV;
>  +       codec_node = of_find_node_by_phandle(*handle);
>  +       if (!codec_node)
>  +               return -ENODEV;
>  +       pr_info("looking for codec: %s\n", codec_node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(codec_node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       of_soc->platform_node = node;
>  +       of_soc->dai_link.cpu_dai = cpu_dai;
>  +       of_soc->device.platform = platform;
>  +       of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Jon Smirl]

On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> From: Grant Likely <grant.likely@secretlab.ca>
>
>  Simple utility layer for creating ASoC machine instances based on data
>  in the OpenFirmware device tree.  OF aware platform drivers and codec
>  drivers register themselves with this framework and the framework
>  automatically instantiates a machine driver.
>
>  This is most likely temporary glue code to work around limitations in
>  the ASoC v1 framework.  I expect ASoC v2 won't need this.
>  ---
>
>   sound/soc/Kconfig  |    6 ++
>   sound/soc/Makefile |    1
>   sound/soc/soc-of.c |  171 ++++++++++++++++++++++++++++++++++++++++++++++++++++
>   3 files changed, 178 insertions(+), 0 deletions(-)
>
>  diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
>  index 18f28ac..c5736e5 100644
>  --- a/sound/soc/Kconfig
>  +++ b/sound/soc/Kconfig
>  @@ -23,6 +23,12 @@ config SND_SOC
>           This ASoC audio support can also be built as a module.  If so, the module
>           will be called snd-soc-core.
>
>  +config SND_SOC_OF
>  +       tristate "OF helpers for SoC audio support"
>  +       depends on SND_SOC
>  +       ---help---
>  +         Add support for OpenFirmware device tree descriptions of sound device
>  +
>   # All the supported Soc's
>   source "sound/soc/at91/Kconfig"
>   source "sound/soc/pxa/Kconfig"
>  diff --git a/sound/soc/Makefile b/sound/soc/Makefile
>  index 782db21..191c2e5 100644
>  --- a/sound/soc/Makefile
>  +++ b/sound/soc/Makefile
>  @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
>
>   obj-$(CONFIG_SND_SOC)  += snd-soc-core.o
>   obj-$(CONFIG_SND_SOC)  += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
>  +obj-$(CONFIG_SND_SOC_OF)       += soc-of.o
>  diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c
>  new file mode 100644
>  index 0000000..9694979
>  --- /dev/null
>  +++ b/sound/soc/soc-of.c
>  @@ -0,0 +1,171 @@
>  +/*
>  + * OF helpers for ALSA SoC Layer
>  + *
>  + * Copyright (C) 2008, Secret Lab Technologies Ltd.
>  + */
>  +
>  +#include <linux/module.h>
>  +#include <linux/moduleparam.h>
>  +#include <linux/init.h>
>  +#include <linux/delay.h>
>  +#include <linux/pm.h>
>  +#include <linux/bitops.h>
>  +#include <linux/platform_device.h>
>  +#include <linux/of.h>
>  +#include <sound/core.h>
>  +#include <sound/pcm.h>
>  +#include <sound/pcm_params.h>
>  +#include <sound/soc.h>
>  +#include <sound/soc-of.h>
>  +#include <sound/initval.h>
>  +
>  +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
>  +MODULE_LICENSE("GPL");
>  +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
>  +
>  +DEFINE_MUTEX(of_snd_soc_mutex);
>  +LIST_HEAD(of_snd_soc_device_list);
>  +static int of_snd_soc_next_index;
>  +
>  +struct of_snd_soc_device {
>  +       int id;
>  +       struct list_head list;
>  +       struct snd_soc_device device;
>  +       struct snd_soc_machine machine;
>  +       struct snd_soc_dai_link dai_link;
>  +       struct platform_device *pdev;
>  +       struct device_node *platform_node;
>  +       struct device_node *codec_node;
>  +};
>  +
>  +static struct snd_soc_ops of_snd_soc_ops = {
>  +};
>  +
>  +static struct of_snd_soc_device *
>  +of_snd_soc_get_device(struct device_node *codec_node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +
>  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
>  +               if (of_soc->codec_node == codec_node)
>  +                       return of_soc;
>  +       }
>  +
>  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
>  +       if (!of_soc)
>  +               return NULL;
>  +
>  +       /* Initialize the structure and add it to the global list */
>  +       of_soc->codec_node = codec_node;
>  +       of_soc->id = of_snd_soc_next_index++;
>  +       of_soc->machine.dai_link = &of_soc->dai_link;
>  +       of_soc->machine.num_links = 1;
>  +       of_soc->device.machine = &of_soc->machine;
>  +       of_soc->dai_link.ops = &of_snd_soc_ops;
>  +       list_add(&of_soc->list, &of_snd_soc_device_list);
>  +
>  +       return of_soc;
>  +}

Isn't this performing the same basic function as this code (except for
spi)? Should this list be maintained in alsa or should there be an
equivalent for searching the spi bus? If you follow the link to the
codec node and get it's parent you know which bus to search. But it
might be simpler to just search the buses sequentially for the node.

static int of_dev_node_match(struct device *dev, void *data)
{
        return dev->archdata.of_node == data;
}

struct i2c_client *of_find_i2c_device_by_node(struct device_node *node)
{
	struct device *dev;
	
	dev = bus_find_device(&i2c_bus_type, NULL, node,
					 of_dev_node_match);
	if (!dev)
		return NULL;
		
	return to_i2c_client(dev);
}
EXPORT_SYMBOL(of_find_i2c_device_by_node);



>  +
>  +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
>  +{
>  +       struct platform_device *pdev;
>  +       int rc;
>  +
>  +       /* Only register the device if both the codec and platform have
>  +        * been registered */
>  +       if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
>  +               return;
>  +
>  +       pr_info("platform<-->codec match achieved; registering machine\n");
>  +
>  +       pdev = platform_device_alloc("soc-audio", of_soc->id);
>  +       if (!pdev) {
>  +               pr_err("of_soc: platform_device_alloc() failed\n");
>  +               return;
>  +       }
>  +
>  +       pdev->dev.platform_data = of_soc;
>  +       platform_set_drvdata(pdev, &of_soc->device);
>  +       of_soc->device.dev = &pdev->dev;
>  +
>  +       /* The ASoC device is complete; register it */
>  +       rc = platform_device_add(pdev);
>  +       if (rc) {
>  +               pr_err("of_soc: platform_device_add() failed\n");
>  +               return;
>  +       }
>  +
>  +}
>  +
>  +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
>  +                             void *codec_data, struct snd_soc_codec_dai *dai,
>  +                             struct device_node *node)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       int rc = 0;
>  +
>  +       pr_info("registering ASoC codec driver: %s\n", node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       /* Store the codec data */
>  +       of_soc->device.codec_data = codec_data;
>  +       of_soc->device.codec_dev = codec_dev;
>  +       of_soc->dai_link.name = node->name;
>  +       of_soc->dai_link.stream_name = node->name;
>  +       of_soc->dai_link.codec_dai = dai;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
>  +
>  +int of_snd_soc_register_platform(struct snd_soc_platform *platform,
>  +                                struct device_node *node,
>  +                                struct snd_soc_cpu_dai *cpu_dai)
>  +{
>  +       struct of_snd_soc_device *of_soc;
>  +       struct device_node *codec_node;
>  +       const phandle *handle;
>  +       int len, rc = 0;
>  +
>  +       pr_info("registering ASoC platform driver: %s\n", node->full_name);
>  +
>  +       handle = of_get_property(node, "codec-handle", &len);
>  +       if (!handle || len < sizeof(handle))
>  +               return -ENODEV;
>  +       codec_node = of_find_node_by_phandle(*handle);
>  +       if (!codec_node)
>  +               return -ENODEV;
>  +       pr_info("looking for codec: %s\n", codec_node->full_name);
>  +
>  +       mutex_lock(&of_snd_soc_mutex);
>  +       of_soc = of_snd_soc_get_device(codec_node);
>  +       if (!of_soc) {
>  +               rc = -ENOMEM;
>  +               goto out;
>  +       }
>  +
>  +       of_soc->platform_node = node;
>  +       of_soc->dai_link.cpu_dai = cpu_dai;
>  +       of_soc->device.platform = platform;
>  +       of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
>  +
>  +       /* Now try to register the SoC device */
>  +       of_snd_soc_register_device(of_soc);
>  +
>  + out:
>  +       mutex_unlock(&of_snd_soc_mutex);
>  +       return rc;
>  +}
>  +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
>
>  _______________________________________________
>  Alsa-devel mailing list
>  Alsa-devel@alsa-project.org
>  http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>


-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

On Wed, Jul 02, 2008 at 11:27:17AM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> >  +static struct of_snd_soc_device *
> >  +of_snd_soc_get_device(struct device_node *codec_node)
> >  +{
> >  +       struct of_snd_soc_device *of_soc;
> >  +
> >  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
> >  +               if (of_soc->codec_node == codec_node)
> >  +                       return of_soc;
> >  +       }
> >  +
> >  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
> >  +       if (!of_soc)
> >  +               return NULL;
> >  +
> >  +       /* Initialize the structure and add it to the global list */
> >  +       of_soc->codec_node = codec_node;
> >  +       of_soc->id = of_snd_soc_next_index++;
> >  +       of_soc->machine.dai_link = &of_soc->dai_link;
> >  +       of_soc->machine.num_links = 1;
> >  +       of_soc->device.machine = &of_soc->machine;
> >  +       of_soc->dai_link.ops = &of_snd_soc_ops;
> >  +       list_add(&of_soc->list, &of_snd_soc_device_list);
> >  +
> >  +       return of_soc;
> >  +}
> 
> Isn't this performing the same basic function as this code (except for
> spi)? Should this list be maintained in alsa or should there be an
> equivalent for searching the spi bus? If you follow the link to the
> codec node and get it's parent you know which bus to search. But it
> might be simpler to just search the buses sequentially for the node.
> 
> static int of_dev_node_match(struct device *dev, void *data)
> {
>         return dev->archdata.of_node == data;
> }
> 
> struct i2c_client *of_find_i2c_device_by_node(struct device_node *node)
> {
> 	struct device *dev;
> 	
> 	dev = bus_find_device(&i2c_bus_type, NULL, node,
> 					 of_dev_node_match);
> 	if (!dev)
> 		return NULL;
> 		
> 	return to_i2c_client(dev);
> }
> EXPORT_SYMBOL(of_find_i2c_device_by_node);

Yes, but you hadn't written these functions when I wrote this helper.
:-P.  I believe ASoC v2 makes all this stuff unnecessary, but I'm
waiting for v2 to hit mainline before I port forward.

g.

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Grant Likely]

On Wed, Jul 02, 2008 at 11:27:17AM -0400, Jon Smirl wrote:
> On 7/1/08, Grant Likely <grant.likely@secretlab.ca> wrote:
> >  +static struct of_snd_soc_device *
> >  +of_snd_soc_get_device(struct device_node *codec_node)
> >  +{
> >  +       struct of_snd_soc_device *of_soc;
> >  +
> >  +       list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
> >  +               if (of_soc->codec_node == codec_node)
> >  +                       return of_soc;
> >  +       }
> >  +
> >  +       of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
> >  +       if (!of_soc)
> >  +               return NULL;
> >  +
> >  +       /* Initialize the structure and add it to the global list */
> >  +       of_soc->codec_node = codec_node;
> >  +       of_soc->id = of_snd_soc_next_index++;
> >  +       of_soc->machine.dai_link = &of_soc->dai_link;
> >  +       of_soc->machine.num_links = 1;
> >  +       of_soc->device.machine = &of_soc->machine;
> >  +       of_soc->dai_link.ops = &of_snd_soc_ops;
> >  +       list_add(&of_soc->list, &of_snd_soc_device_list);
> >  +
> >  +       return of_soc;
> >  +}
> 
> Isn't this performing the same basic function as this code (except for
> spi)? Should this list be maintained in alsa or should there be an
> equivalent for searching the spi bus? If you follow the link to the
> codec node and get it's parent you know which bus to search. But it
> might be simpler to just search the buses sequentially for the node.
> 
> static int of_dev_node_match(struct device *dev, void *data)
> {
>         return dev->archdata.of_node == data;
> }
> 
> struct i2c_client *of_find_i2c_device_by_node(struct device_node *node)
> {
> 	struct device *dev;
> 	
> 	dev = bus_find_device(&i2c_bus_type, NULL, node,
> 					 of_dev_node_match);
> 	if (!dev)
> 		return NULL;
> 		
> 	return to_i2c_client(dev);
> }
> EXPORT_SYMBOL(of_find_i2c_device_by_node);

Yes, but you hadn't written these functions when I wrote this helper.
:-P.  I believe ASoC v2 makes all this stuff unnecessary, but I'm
waiting for v2 to hit mainline before I port forward.

g.

Re: [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Timur Tabi]

On Jul 3, 2008, at 12:33 PM, Grant Likely wrote:

> Yes, but you hadn't written these functions when I wrote this helper.
> :-P.  I believe ASoC v2 makes all this stuff unnecessary, but I'm
> waiting for v2 to hit mainline before I port forward.

FYI, it took me about a month to port my drivers to ASoC V2.  I had  
more work than most people because I addressed issues that others  
drivers didn't.  For instance, my CS4270 driver is the only codec  
driver in V2 that supports multiple instances of itself.

Re: [alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers

[Timur Tabi]

On Jul 3, 2008, at 12:33 PM, Grant Likely wrote:

> Yes, but you hadn't written these functions when I wrote this helper.
> :-P.  I believe ASoC v2 makes all this stuff unnecessary, but I'm
> waiting for v2 to hit mainline before I port forward.

FYI, it took me about a month to port my drivers to ASoC V2.  I had  
more work than most people because I addressed issues that others  
drivers didn't.  For instance, my CS4270 driver is the only codec  
driver in V2 that supports multiple instances of itself.