[PATCH 00/10 v3] ASoC: Renesas adds DPCM based sampling rate convert

[PATCH 00/10 v3] ASoC: Renesas adds DPCM based sampling rate convert

[Kuninori Morimoto]

Hi Mark

These are v3 of DPCM based sampling rate convert support patch-set for Rensas
sound driver. Then, it doesn't use simple-card. it uses new Renesas specific
sound card driver, basically it was copied from simple-card.
v2 patch-set had ak4642 patch, but, it is already applied. v3 doesn't include it.

 1      : prepare patch for Renesas new sound card driver
 2 -  3 : new Renesas sound card driver
 4 -  8 : fixup/tidyup patches for rsnd
 9 - 10 : DPCM support patch for rsnd

Kuninori Morimoto (10):
       1) ASoC: core: add snd_soc_get_dai_drv() to get DAI driver
       2) ASoC: rsrc-card: add Renesas sampling rate convert sound card support
       3) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate
       4) ASoC: rsnd: no more SSI restart when unusual situation
       5) ASoC: rsnd: no more SRC restart when unusual situation
       6) ASoC: rsnd: tidyup error message format
       7) ASoC: rsnd: show debug info for sampling rate convert
       8) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock
       9) ASoC: rsnd: add dai_link stream name
      10) ASoC: rsnd: add DPCM based sampling rate convert

 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt |  67 +++++++++++++
 include/sound/soc.h                                           |   1 +
 sound/soc/sh/Kconfig                                          |   5 +
 sound/soc/sh/rcar/Makefile                                    |   5 +-
 sound/soc/sh/rcar/adg.c                                       |   2 +
 sound/soc/sh/rcar/core.c                                      |  48 +++++++++-
 sound/soc/sh/rcar/dvc.c                                       |  12 +--
 sound/soc/sh/rcar/rsnd.h                                      |  16 +++-
 sound/soc/sh/rcar/rsrc-card.c                                 | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/sh/rcar/src.c                                       |  63 ++++++++++--
 sound/soc/sh/rcar/ssi.c                                       |  31 ++++--
 sound/soc/soc-core.c                                          |  69 ++++++++++----
 12 files changed, 777 insertions(+), 54 deletions(-)

Best regards
---
Kuninori Morimoto
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[PATCH 01/10 v3] ASoC: core: add snd_soc_get_dai_drv() to get DAI driver

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

Sometimes sound card driver want to get cpu/codec driver.
This patch adds snd_soc_get_dai_drv() and enable it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - keep .of_xlate_dai_name

 include/sound/soc.h  |  1 +
 sound/soc/soc-core.c | 69 ++++++++++++++++++++++++++++++++++++----------------
 2 files changed, 49 insertions(+), 21 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index bc6cfab..35f7aa5c 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1508,6 +1508,7 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
 				     const char *prefix,
 				     struct device_node **bitclkmaster,
 				     struct device_node **framemaster);
+struct snd_soc_dai_driver *snd_soc_get_dai_drv(struct of_phandle_args *args);
 int snd_soc_of_get_dai_name(struct device_node *of_node,
 			    const char **dai_name);
 int snd_soc_of_get_dai_link_codecs(struct device *dev,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a85bd42..89eac2b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3454,6 +3454,47 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt);
 
+static struct snd_soc_dai_driver
+*_snd_soc_get_dai_drv(struct of_phandle_args *args,
+		      struct snd_soc_component *pos)
+{
+	int id = -1;
+
+	switch (args->args_count) {
+	case 0:
+		id = 0; /* same as dai_drv[0] */
+		break;
+	case 1:
+		id = args->args[0];
+		break;
+	}
+
+	if (id < 0 || id >= pos->num_dai)
+		return NULL;
+
+	return  pos->dai_drv + id;
+}
+
+struct snd_soc_dai_driver
+*snd_soc_get_dai_drv(struct of_phandle_args *args)
+{
+	struct snd_soc_dai_driver *drv = NULL;
+	struct snd_soc_component *pos;
+
+	mutex_lock(&client_mutex);
+	list_for_each_entry(pos, &component_list, list) {
+		if (pos->dev->of_node != args->np)
+			continue;
+
+		drv = _snd_soc_get_dai_drv(args, pos);
+		break;
+	}
+	mutex_unlock(&client_mutex);
+
+	return drv;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_dai_drv);
+
 static int snd_soc_get_dai_name(struct of_phandle_args *args,
 				const char **dai_name)
 {
@@ -3470,30 +3511,16 @@ static int snd_soc_get_dai_name(struct of_phandle_args *args,
 							     args,
 							     dai_name);
 		} else {
-			int id = -1;
+			struct snd_soc_dai_driver *drv;
 
-			switch (args->args_count) {
-			case 0:
-				id = 0; /* same as dai_drv[0] */
-				break;
-			case 1:
-				id = args->args[0];
-				break;
-			default:
-				/* not supported */
-				break;
-			}
+			drv = _snd_soc_get_dai_drv(args, pos);
+			if (drv) {
+				ret = 0;
 
-			if (id < 0 || id >= pos->num_dai) {
-				ret = -EINVAL;
-				continue;
+				*dai_name = drv->name;
+				if (!*dai_name)
+					*dai_name = pos->name;
 			}
-
-			ret = 0;
-
-			*dai_name = pos->dai_drv[id].name;
-			if (!*dai_name)
-				*dai_name = pos->name;
 		}
 
 		break;
-- 
1.9.1

[PATCH 02/10 v3] ASoC: rsrc-card: add Renesas sampling rate convert sound card support

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

Renesas sound card has "sampling rate convert" feature which
should be implemented via DPCM.
But, sound card driver point of view, it is difficult to add
this DPCM feature on simple-card driver. Especially, DT binding
support is very difficult.

This patch implements DPCM feature on DT as Renesas specific sound card.
This new driver is copied from current simple-card driver.
Main difference between simple-card and this driver are...
 1. removed unused feature from simple-card
 2. removed driver named prefix from DT property
 3. CPU will be FE, CODEC will be BE with snd-soc-dummy
 4. it supports sampling rate convert via .be_hw_params_fixup
 5. board specific routing is implemented in driver

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - rename function "rdpcm_xxx" -> "rsrc_card_xxx"
 - rename compatible "renesas-dpcm-audio-card" -> "renesas,rsrc-card,<board>"
 - rename file "rsnd-dpcm-card.c" -> "rsrc-card.c"

 .../bindings/sound/renesas,rsrc-card.txt           |  66 +++
 sound/soc/sh/Kconfig                               |   5 +
 sound/soc/sh/rcar/Makefile                         |   5 +-
 sound/soc/sh/rcar/rsrc-card.c                      | 489 +++++++++++++++++++++
 4 files changed, 564 insertions(+), 1 deletion(-)
 create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
 create mode 100644 sound/soc/sh/rcar/rsrc-card.c

diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
new file mode 100644
index 0000000..12e287e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
@@ -0,0 +1,66 @@
+Renesas Sampling Rate Convert Sound Card:
+
+Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC <-> codec.
+
+Required properties:
+
+- compatible				: "renesas,rsrc-card,<board>"
+					  Examples with soctypes are:
+					    - "renesas,rsrc-card,lager"
+					    - "renesas,rsrc-card,koelsch"
+Optional properties:
+
+- card_name				: User specified audio sound card name, one string
+					  property.
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
+
+Optional subnode properties:
+
+- format				: CPU/CODEC common audio format.
+					  "i2s", "right_j", "left_j" , "dsp_a"
+					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- frame-master				: Indicates dai-link frame master.
+					  phandle to a cpu or codec subnode.
+- bitclock-master			: Indicates dai-link bit clock master.
+					  phandle to a cpu or codec subnode.
+- bitclock-inversion			: bool property. Add this if the
+					  dai-link uses bit clock inversion.
+- frame-inversion			: bool property. Add this if the
+					  dai-link uses frame clock inversion.
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai				: phandle and port of CPU/CODEC
+
+Optional CPU/CODEC subnodes properties:
+
+- clocks / system-clock-frequency	: specify subnode's clock if needed.
+					  it can be specified via "clocks" if system has
+					  clock node (= common clock), or "system-clock-frequency"
+					  (if system doens't support common clock)
+					  If a clock is specified, it is
+					  enabled with clk_prepare_enable()
+					  in dai startup() and disabled with
+					  clk_disable_unprepare() in dai
+					  shutdown().
+
+Example
+
+sound {
+	compatible = "renesas,rsrc-card,lager";
+
+	card-name = "rsnd-ak4643";
+	format = "left_j";
+	bitclock-master = <&sndcodec>;
+	frame-master = <&sndcodec>;
+
+	sndcpu: cpu {
+		sound-dai = <&rcar_sound>;
+	};
+
+	sndcodec: codec {
+		sound-dai = <&ak4643>;
+		system-clock-frequency = <11289600>;
+	};
+};
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 80245b6..2b30304 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -41,6 +41,11 @@ config SND_SOC_RCAR
 	help
 	  This option enables R-Car SUR/SCU/SSIU/SSI sound support
 
+config SND_SOC_RSRC_CARD
+	tristate "Renesas Sampling Rate Convert Sound Card"
+	help
+	  This option enables simple sound if you need sampling rate convert
+
 ##
 ## Boards
 ##
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 7b20492..f1b4451 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,5 @@
 snd-soc-rcar-objs	:= core.o gen.o dma.o src.o adg.o ssi.o dvc.o
-obj-$(CONFIG_SND_SOC_RCAR)	+= snd-soc-rcar.o
\ No newline at end of file
+obj-$(CONFIG_SND_SOC_RCAR)	+= snd-soc-rcar.o
+
+snd-soc-rsrc-card-objs	:= rsrc-card.o
+obj-$(CONFIG_SND_SOC_RSRC_CARD)	+= snd-soc-rsrc-card.o
diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c
new file mode 100644
index 0000000..f209419
--- /dev/null
+++ b/sound/soc/sh/rcar/rsrc-card.c
@@ -0,0 +1,489 @@
+/*
+ * Renesas Sampling Rate Convert Sound Card for DPCM
+ *
+ * Copyright (C) 2015 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * based on ${LINUX}/sound/soc/generic/simple-card.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/string.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+struct rsrc_card_of_data {
+	const struct snd_soc_dapm_route *routes;
+	int num_routes;
+};
+
+static const struct snd_soc_dapm_route routes_ssi0_ak4642[] = {
+	{"ak4642-hifi Playback", NULL, "DAI0 Playback"},
+	{"DAI0 Capture", NULL, "ak4642-hifi Capture"},
+};
+
+static const struct rsrc_card_of_data routes_of_ssi0_ak4642 = {
+	.routes		= routes_ssi0_ak4642,
+	.num_routes	= ARRAY_SIZE(routes_ssi0_ak4642),
+};
+
+static const struct of_device_id rsrc_card_of_match[] = {
+	{ .compatible = "renesas,rsrc-card,lager",	.data = &routes_of_ssi0_ak4642 },
+	{ .compatible = "renesas,rsrc-card,koelsch",	.data = &routes_of_ssi0_ak4642 },
+	{},
+};
+MODULE_DEVICE_TABLE(of, rsrc_card_of_match);
+
+struct rsrc_card_dai {
+	const char *name;
+	unsigned int fmt;
+	unsigned int sysclk;
+	struct clk *clk;
+};
+
+#define RSRC_FB_NUM	2 /* FE/BE */
+#define IDX_CPU		0
+#define IDX_CODEC	1
+struct rsrc_card_priv {
+	struct snd_soc_card snd_card;
+	struct rsrc_card_dai_props {
+		struct rsrc_card_dai cpu_dai;
+		struct rsrc_card_dai codec_dai;
+	} dai_props[RSRC_FB_NUM];
+	struct snd_soc_codec_conf codec_conf;
+	struct snd_soc_dai_link dai_link[RSRC_FB_NUM];
+};
+
+#define rsrc_priv_to_dev(priv) ((priv)->snd_card.dev)
+#define rsrc_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i)
+#define rsrc_priv_to_props(priv, i) ((priv)->dai_props + i)
+
+static int rsrc_card_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct rsrc_card_priv *priv =	snd_soc_card_get_drvdata(rtd->card);
+	struct rsrc_card_dai_props *dai_props =
+		&priv->dai_props[rtd - rtd->card->rtd];
+	int ret;
+
+	ret = clk_prepare_enable(dai_props->cpu_dai.clk);
+	if (ret)
+		return ret;
+
+	ret = clk_prepare_enable(dai_props->codec_dai.clk);
+	if (ret)
+		clk_disable_unprepare(dai_props->cpu_dai.clk);
+
+	return ret;
+}
+
+static void rsrc_card_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct rsrc_card_priv *priv =	snd_soc_card_get_drvdata(rtd->card);
+	struct rsrc_card_dai_props *dai_props =
+		&priv->dai_props[rtd - rtd->card->rtd];
+
+	clk_disable_unprepare(dai_props->cpu_dai.clk);
+
+	clk_disable_unprepare(dai_props->codec_dai.clk);
+}
+
+static struct snd_soc_ops rsrc_card_ops = {
+	.startup = rsrc_card_startup,
+	.shutdown = rsrc_card_shutdown,
+};
+
+static int __rsrc_card_dai_init(struct snd_soc_dai *dai,
+				struct rsrc_card_dai *set)
+{
+	int ret;
+
+	if (set->fmt) {
+		ret = snd_soc_dai_set_fmt(dai, set->fmt);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(dai->dev, "set_fmt error\n");
+			goto err;
+		}
+	}
+
+	if (set->sysclk) {
+		ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(dai->dev, "set_sysclk error\n");
+			goto err;
+		}
+	}
+
+	ret = 0;
+
+err:
+	return ret;
+}
+
+static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct rsrc_card_priv *priv =	snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *codec = rtd->codec_dai;
+	struct snd_soc_dai *cpu = rtd->cpu_dai;
+	struct rsrc_card_dai_props *dai_props;
+	int num, ret;
+
+	num = rtd - rtd->card->rtd;
+	dai_props = &priv->dai_props[num];
+	ret = __rsrc_card_dai_init(codec, &dai_props->codec_dai);
+	if (ret < 0)
+		return ret;
+
+	ret = __rsrc_card_dai_init(cpu, &dai_props->cpu_dai);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int
+rsrc_card_sub_parse_of(struct rsrc_card_priv *priv,
+		       struct device_node *np,
+		       struct rsrc_card_dai *dai,
+		       struct snd_soc_dai_link *dai_link,
+		       int *args_count)
+{
+	struct of_phandle_args args;
+	struct device_node **p_node;
+	struct snd_soc_dai_driver *dai_drv;
+	struct clk *clk;
+	const char **dai_name;
+	const char **name;
+	u32 val;
+	int ret;
+
+	if (args_count) {
+		p_node		= &dai_link->cpu_of_node;
+		dai_name	= &dai_link->cpu_dai_name;
+		name		= &dai_link->cpu_name;
+	} else {
+		p_node		= &dai_link->codec_of_node;
+		dai_name	= &dai_link->codec_dai_name;
+		name		= &dai_link->codec_name;
+	}
+
+	if (!np) {
+		/* use snd-soc-dummy */
+		*p_node		= NULL;
+		*dai_name	= "snd-soc-dummy-dai";
+		*name		= "snd-soc-dummy";
+		return 0;
+	}
+
+	/*
+	 * Get node via "sound-dai = <&phandle port>"
+	 * it will be used as xxx_of_node on soc_bind_dai_link()
+	 */
+	ret = of_parse_phandle_with_args(np, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret)
+		return ret;
+
+	*p_node = args.np;
+
+	/* Get dai->name */
+	ret = snd_soc_of_get_dai_name(np, dai_name);
+	if (ret < 0)
+		return ret;
+
+	/* Get DAI driver */
+	dai_drv = snd_soc_get_dai_drv(&args);
+	if (!dai_drv)
+		return -EINVAL;
+
+	/* for DPCM */
+	if (dai_drv->playback.stream_name)
+		dai_link->dpcm_playback = 1;
+	if (dai_drv->capture.stream_name)
+		dai_link->dpcm_capture = 1;
+
+	if (args_count) {
+		*args_count = args.args_count;
+		dai_link->dynamic = 1;
+	} else {
+		dai_link->no_pcm = 1;
+		priv->codec_conf.of_node = (*p_node);
+		priv->codec_conf.name_prefix = dai_drv->name;
+	}
+
+	/*
+	 * Parse dai->sysclk come from "clocks = <&xxx>"
+	 * (if system has common clock)
+	 *  or "system-clock-frequency = <xxx>"
+	 *  or device's module clock.
+	 */
+	if (of_property_read_bool(np, "clocks")) {
+		clk = of_clk_get(np, 0);
+		if (IS_ERR(clk)) {
+			ret = PTR_ERR(clk);
+			return ret;
+		}
+
+		dai->sysclk = clk_get_rate(clk);
+		dai->clk = clk;
+	} else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
+		dai->sysclk = val;
+	} else {
+		clk = of_clk_get(args.np, 0);
+		if (!IS_ERR(clk))
+			dai->sysclk = clk_get_rate(clk);
+	}
+
+	return 0;
+}
+
+static int rsrc_card_parse_daifmt(struct device_node *node,
+				  struct rsrc_card_priv *priv,
+				  struct device_node *codec,
+				  int idx)
+{
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	struct rsrc_card_dai_props *dai_props = rsrc_priv_to_props(priv, idx);
+	struct rsrc_card_dai *cpu_dai = &dai_props->cpu_dai;
+	struct rsrc_card_dai *codec_dai = &dai_props->codec_dai;
+	unsigned int daifmt;
+
+	daifmt = snd_soc_of_parse_daifmt(node, NULL,
+					 &bitclkmaster, &framemaster);
+	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+	if (!bitclkmaster && !framemaster)
+		return -EINVAL;
+
+	if (codec == bitclkmaster)
+		daifmt |= (codec == framemaster) ?
+			SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
+	else
+		daifmt |= (codec == framemaster) ?
+			SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
+
+	cpu_dai->fmt	= daifmt;
+	codec_dai->fmt	= daifmt;
+
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+
+	return 0;
+}
+
+static int rsrc_card_dai_link_of(struct device_node *node,
+				 struct rsrc_card_priv *priv,
+				 int idx)
+{
+	struct device *dev = rsrc_priv_to_dev(priv);
+	struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx);
+	struct rsrc_card_dai_props *dai_props = rsrc_priv_to_props(priv, idx);
+	struct device_node *cpu = NULL;
+	struct device_node *codec = NULL;
+	char *name;
+	char prop[128];
+	int ret, cpu_args;
+
+	cpu = of_get_child_by_name(node, "cpu");
+	codec = of_get_child_by_name(node, "codec");
+
+	if (!cpu || !codec) {
+		ret = -EINVAL;
+		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
+		goto dai_link_of_err;
+	}
+
+	ret = rsrc_card_parse_daifmt(node, priv, codec, idx);
+	if (ret < 0)
+		goto dai_link_of_err;
+
+	ret = rsrc_card_sub_parse_of(priv, (idx == IDX_CPU) ? cpu : NULL,
+				     &dai_props->cpu_dai,
+				     dai_link,
+				     &cpu_args);
+	if (ret < 0)
+		goto dai_link_of_err;
+
+	ret = rsrc_card_sub_parse_of(priv, (idx == IDX_CODEC) ? codec : NULL,
+				     &dai_props->codec_dai,
+				     dai_link,
+				     NULL);
+	if (ret < 0)
+		goto dai_link_of_err;
+
+	if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) {
+		ret = -EINVAL;
+		goto dai_link_of_err;
+	}
+
+	/* Simple Card assumes platform == cpu */
+	dai_link->platform_of_node = dai_link->cpu_of_node;
+
+	/* DAI link name is created from CPU/CODEC dai name */
+	name = devm_kzalloc(dev,
+			    strlen(dai_link->cpu_dai_name)   +
+			    strlen(dai_link->codec_dai_name) + 2,
+			    GFP_KERNEL);
+	if (!name) {
+		ret = -ENOMEM;
+		goto dai_link_of_err;
+	}
+
+	sprintf(name, "%s-%s", dai_link->cpu_dai_name,
+		dai_link->codec_dai_name);
+	dai_link->name = dai_link->stream_name = name;
+	dai_link->ops = &rsrc_card_ops;
+	dai_link->init = rsrc_card_dai_init;
+
+	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
+	dev_dbg(dev, "\tcpu : %s / %04x / %d\n",
+		dai_link->cpu_dai_name,
+		dai_props->cpu_dai.fmt,
+		dai_props->cpu_dai.sysclk);
+	dev_dbg(dev, "\tcodec : %s / %04x / %d\n",
+		dai_link->codec_dai_name,
+		dai_props->codec_dai.fmt,
+		dai_props->codec_dai.sysclk);
+
+	/*
+	 * In soc_bind_dai_link() will check cpu name after
+	 * of_node matching if dai_link has cpu_dai_name.
+	 * but, it will never match if name was created by
+	 * fmt_single_name() remove cpu_dai_name if cpu_args
+	 * was 0. See:
+	 *	fmt_single_name()
+	 *	fmt_multiple_name()
+	 */
+	if (!cpu_args)
+		dai_link->cpu_dai_name = NULL;
+
+dai_link_of_err:
+	of_node_put(cpu);
+	of_node_put(codec);
+
+	return ret;
+}
+
+static int rsrc_card_parse_of(struct device_node *node,
+			      struct rsrc_card_priv *priv)
+{
+	struct device *dev = rsrc_priv_to_dev(priv);
+	const struct of_device_id *of_id = of_match_device(rsrc_card_of_match, dev);
+	const struct rsrc_card_of_data *of_data = of_id->data;
+	int ret;
+	int i;
+
+	if (!node)
+		return -EINVAL;
+
+	/* Parse the card name from DT */
+	snd_soc_of_parse_card_name(&priv->snd_card, "card-name");
+
+	/* DAPM routes */
+	priv->snd_card.of_dapm_routes		= of_data->routes;
+	priv->snd_card.num_of_dapm_routes	= of_data->num_routes;
+
+	dev_dbg(dev, "New rsrc-audio-card: %s\n", priv->snd_card.name ?
+		priv->snd_card.name : "");
+
+	/* FE/BE */
+	for (i = 0; i < RSRC_FB_NUM; i++) {
+		ret = rsrc_card_dai_link_of(node, priv, i);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (!priv->snd_card.name)
+		priv->snd_card.name = priv->snd_card.dai_link->name;
+
+	return 0;
+}
+
+/* Decrease the reference count of the device nodes */
+static int rsrc_card_unref(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *dai_link;
+	int num_links;
+
+	for (num_links = 0, dai_link = card->dai_link;
+	     num_links < card->num_links;
+	     num_links++, dai_link++) {
+		of_node_put(dai_link->cpu_of_node);
+		of_node_put(dai_link->codec_of_node);
+	}
+	return 0;
+}
+
+static int rsrc_card_probe(struct platform_device *pdev)
+{
+	struct rsrc_card_priv *priv;
+	struct snd_soc_dai_link *dai_link;
+	struct device_node *np = pdev->dev.of_node;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	/* Allocate the private data */
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	/* Init snd_soc_card */
+	priv->snd_card.owner = THIS_MODULE;
+	priv->snd_card.dev = dev;
+	dai_link = priv->dai_link;
+	priv->snd_card.dai_link = dai_link;
+	priv->snd_card.num_links = RSRC_FB_NUM;
+	priv->snd_card.codec_conf = &priv->codec_conf;
+	priv->snd_card.num_configs = 1;
+
+	ret = rsrc_card_parse_of(np, priv);
+	if (ret < 0) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "parse error %d\n", ret);
+		goto err;
+	}
+
+	snd_soc_card_set_drvdata(&priv->snd_card, priv);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+	if (ret >= 0)
+		return ret;
+err:
+	rsrc_card_unref(&priv->snd_card);
+
+	return ret;
+}
+
+static int rsrc_card_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	return rsrc_card_unref(card);
+}
+
+static struct platform_driver rsrc_card = {
+	.driver = {
+		.name = "renesas-src-audio-card",
+		.of_match_table = rsrc_card_of_match,
+	},
+	.probe = rsrc_card_probe,
+	.remove = rsrc_card_remove,
+};
+
+module_platform_driver(rsrc_card);
+
+MODULE_ALIAS("platform:renesas-src-audio-card");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Renesas Sampling Rate Convert Sound Card");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
-- 
1.9.1

[PATCH 03/10 v3] ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

Current rsnd-dpcm-card is supporting DPCM FE/BE sound card.
This patch adds .be_hw_params_fixup and enabled sampling convert rate.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change, but based on new driver

 .../bindings/sound/renesas,rsrc-card.txt           |  1 +
 sound/soc/sh/rcar/rsrc-card.c                      | 27 ++++++++++++++++++++--
 2 files changed, 26 insertions(+), 2 deletions(-)

diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
index 12e287e..c641550 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
@@ -28,6 +28,7 @@ Optional subnode properties:
 					  dai-link uses bit clock inversion.
 - frame-inversion			: bool property. Add this if the
 					  dai-link uses frame clock inversion.
+- convert-rate				: platform specified sampling rate convert
 
 Required CPU/CODEC subnodes properties:
 
diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c
index f209419..765d04d 100644
--- a/sound/soc/sh/rcar/rsrc-card.c
+++ b/sound/soc/sh/rcar/rsrc-card.c
@@ -61,6 +61,7 @@ struct rsrc_card_priv {
 	} dai_props[RSRC_FB_NUM];
 	struct snd_soc_codec_conf codec_conf;
 	struct snd_soc_dai_link dai_link[RSRC_FB_NUM];
+	u32 convert_rate;
 };
 
 #define rsrc_priv_to_dev(priv) ((priv)->snd_card.dev)
@@ -151,6 +152,21 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int rsrc_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+					struct snd_pcm_hw_params *params)
+{
+	struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_interval *rate = hw_param_interval(params,
+						      SNDRV_PCM_HW_PARAM_RATE);
+
+	if (!priv->convert_rate)
+		return 0;
+
+	rate->min = rate->max = priv->convert_rate;
+
+	return 0;
+}
+
 static int
 rsrc_card_sub_parse_of(struct rsrc_card_priv *priv,
 		       struct device_node *np,
@@ -346,6 +362,9 @@ static int rsrc_card_dai_link_of(struct device_node *node,
 	dai_link->ops = &rsrc_card_ops;
 	dai_link->init = rsrc_card_dai_init;
 
+	if (idx == IDX_CODEC)
+		dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup;
+
 	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
 	dev_dbg(dev, "\tcpu : %s / %04x / %d\n",
 		dai_link->cpu_dai_name,
@@ -394,8 +413,12 @@ static int rsrc_card_parse_of(struct device_node *node,
 	priv->snd_card.of_dapm_routes		= of_data->routes;
 	priv->snd_card.num_of_dapm_routes	= of_data->num_routes;
 
-	dev_dbg(dev, "New rsrc-audio-card: %s\n", priv->snd_card.name ?
-		priv->snd_card.name : "");
+	/* sampling rate convert */
+	of_property_read_u32(node, "convert-rate", &priv->convert_rate);
+
+	dev_dbg(dev, "New rsrc-audio-card: %s (%d)\n",
+		priv->snd_card.name ? priv->snd_card.name : "",
+		priv->convert_rate);
 
 	/* FE/BE */
 	for (i = 0; i < RSRC_FB_NUM; i++) {
-- 
1.9.1

[PATCH 04/10 v3] ASoC: rsnd: no more SSI restart when unusual situation

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

It will be SSI interrupt endless loop f unusual situation happen.
This patch adds restart limit for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/ssi.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fea4aa5..060d3d2 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -416,11 +416,14 @@ static irqreturn_t rsnd_ssi_interrupt(int irq, void *data)
 		/*
 		 * restart SSI
 		 */
-		rsnd_ssi_stop(mod, priv);
-		rsnd_ssi_start(mod, priv);
-
 		dev_dbg(dev, "%s[%d] restart\n",
 			rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+		rsnd_ssi_stop(mod, priv);
+		if (ssi->err < 1024)
+			rsnd_ssi_start(mod, priv);
+		else
+			dev_warn(dev, "no more SSI restart\n");
 	}
 
 	rsnd_ssi_record_error(ssi, status);
-- 
1.9.1

[PATCH 05/10 v3] ASoC: rsnd: no more SRC restart when unusual situation

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

It will be SRC interrupt endless loop f unusual situation happen.
This patch adds restart limit for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/src.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6ce8985..cc93f32 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -620,13 +620,17 @@ static irqreturn_t rsnd_src_interrupt_gen2(int irq, void *data)
 
 	if (rsnd_src_error_record_gen2(mod)) {
 		struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+		struct rsnd_src *src = rsnd_mod_to_src(mod);
 		struct device *dev = rsnd_priv_to_dev(priv);
 
-		_rsnd_src_stop_gen2(mod);
-		_rsnd_src_start_gen2(mod);
-
 		dev_dbg(dev, "%s[%d] restart\n",
 			rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+		_rsnd_src_stop_gen2(mod);
+		if (src->err < 1024)
+			_rsnd_src_start_gen2(mod);
+		else
+			dev_warn(dev, "no more SRC restart\n");
 	}
 
 	return IRQ_HANDLED;
-- 
1.9.1

[PATCH 06/10 v3] ASoC: rsnd: tidyup error message format

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

This driver sometimes fixups debug/error message format
30cc4faf703955cd5cd07da489bd817ae43e3fec
(ASoC: rsnd: tidyup debug message format and timing)
337b0b4c5f415705f1b97df57cecfac45903449a
(ASoC: rsnd: error meesage indicates its port)
But, it still exist un-fomated error message. This patch fixup it

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index be03474..54042d9 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -232,7 +232,7 @@ static int rsnd_dai_connect(struct rsnd_mod *mod,
 		struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
 		struct device *dev = rsnd_priv_to_dev(priv);
 
-		dev_err(dev, "%s%d is not empty\n",
+		dev_err(dev, "%s[%d] is not empty\n",
 			rsnd_mod_name(mod),
 			rsnd_mod_id(mod));
 		return -EIO;
-- 
1.9.1

[PATCH 07/10 v3] ASoC: rsnd: show debug info for sampling rate convert

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/adg.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 7ac35c9..7af374b 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -183,6 +183,8 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
 
 	rsnd_mod_bset(mod, DIV_EN, en, en);
 
+	dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate);
+
 	return 0;
 }
 
-- 
1.9.1

[PATCH 08/10 v3] ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

clk_prepare_enable/unprepare() uses mutex inside, and it uses __schedule().
Then, raw_spin_lock/unlock_irq() is called, and it breaks Renesas
sound driver's spin lock irq.
This patch moves clk_prepare_enable/unprepare to out of spin lock area.
Special thanks to Das Biju.

Reported-by: Das Biju <biju.das@bp.renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/core.c | 29 ++++++++++++++++++++++++++++-
 sound/soc/sh/rcar/dvc.c  | 12 +-----------
 sound/soc/sh/rcar/rsnd.h |  9 +++++++--
 sound/soc/sh/rcar/src.c  |  6 ++----
 sound/soc/sh/rcar/ssi.c  | 22 ++++++++++++++++++----
 5 files changed, 56 insertions(+), 22 deletions(-)

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 54042d9..e6c1c7a4 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -190,7 +190,7 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod)
 ({								\
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);		\
 	struct device *dev = rsnd_priv_to_dev(priv);		\
-	u32 mask = 1 << __rsnd_mod_shift_##func;			\
+	u32 mask = (1 << __rsnd_mod_shift_##func) & ~(1 << 31);		\
 	u32 call = __rsnd_mod_call_##func << __rsnd_mod_shift_##func;	\
 	int ret = 0;							\
 	if ((mod->status & mask) == call) {				\
@@ -222,6 +222,27 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod)
 	ret;							\
 })
 
+int rsnd_mod_clk(struct rsnd_mod *mod, int enable)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	/*
+	 * clk_prepare_enable/unprepare() should not be called
+	 * from inside spin lock
+	 */
+	if (enable)
+		clk_prepare_enable(mod->clk);
+	else
+		clk_disable_unprepare(mod->clk);
+
+	dev_dbg(dev, "%s[%d] clk %s\n",
+		rsnd_mod_name(mod), rsnd_mod_id(mod),
+		enable ? "ON" : "OFF");
+
+	return 0;
+}
+
 static int rsnd_dai_connect(struct rsnd_mod *mod,
 			    struct rsnd_dai_stream *io)
 {
@@ -345,6 +366,9 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	int ret;
 	unsigned long flags;
 
+	if (cmd == SNDRV_PCM_TRIGGER_START)
+		rsnd_dai_call(clk, io, 1);
+
 	rsnd_lock(priv, flags);
 
 	switch (cmd) {
@@ -385,6 +409,9 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 dai_trigger_end:
 	rsnd_unlock(priv, flags);
 
+	if (cmd == SNDRV_PCM_TRIGGER_STOP)
+		rsnd_dai_call(clk, io, 0);
+
 	return ret;
 }
 
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index aeeef13..9420349 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -166,8 +166,6 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod,
 		return -EINVAL;
 	}
 
-	rsnd_mod_hw_start(dvc_mod);
-
 	/*
 	 * fixme
 	 * it doesn't support CTU/MIX
@@ -191,14 +189,6 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod,
 	return 0;
 }
 
-static int rsnd_dvc_quit(struct rsnd_mod *mod,
-			 struct rsnd_priv *priv)
-{
-	rsnd_mod_hw_stop(mod);
-
-	return 0;
-}
-
 static int rsnd_dvc_start(struct rsnd_mod *mod,
 			  struct rsnd_priv *priv)
 {
@@ -286,9 +276,9 @@ static struct rsnd_mod_ops rsnd_dvc_ops = {
 	.probe		= rsnd_dvc_probe_gen2,
 	.remove		= rsnd_dvc_remove_gen2,
 	.init		= rsnd_dvc_init,
-	.quit		= rsnd_dvc_quit,
 	.start		= rsnd_dvc_start,
 	.stop		= rsnd_dvc_stop,
+	.clk		= rsnd_mod_clk,
 	.pcm_new	= rsnd_dvc_pcm_new,
 };
 
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 52c401c..40a12f4 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -237,6 +237,7 @@ struct rsnd_mod_ops {
 		     struct rsnd_priv *priv);
 	int (*stop)(struct rsnd_mod *mod,
 		    struct rsnd_priv *priv);
+	int (*clk)(struct rsnd_mod *mod, int enable);
 	int (*pcm_new)(struct rsnd_mod *mod,
 		       struct snd_soc_pcm_runtime *rtd);
 	int (*fallback)(struct rsnd_mod *mod,
@@ -262,6 +263,9 @@ struct rsnd_mod {
  * 2	0: start	1: stop
  * 3	0: pcm_new
  * 4	0: fallback
+ *
+ * 31 bit is always called (see __rsnd_mod_call)
+ * 31	0: clk
  */
 #define __rsnd_mod_shift_probe		0
 #define __rsnd_mod_shift_remove		0
@@ -271,6 +275,7 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_stop		2
 #define __rsnd_mod_shift_pcm_new	3
 #define __rsnd_mod_shift_fallback	4
+#define __rsnd_mod_shift_clk		31 /* always called */
 
 #define __rsnd_mod_call_probe		0
 #define __rsnd_mod_call_remove		1
@@ -280,13 +285,12 @@ struct rsnd_mod {
 #define __rsnd_mod_call_stop		1
 #define __rsnd_mod_call_pcm_new		0
 #define __rsnd_mod_call_fallback	0
+#define __rsnd_mod_call_clk		0
 
 #define rsnd_mod_to_priv(mod) (rsnd_io_to_priv(rsnd_mod_to_io(mod)))
 #define rsnd_mod_to_dma(mod) (&(mod)->dma)
 #define rsnd_mod_to_io(mod) ((mod)->io)
 #define rsnd_mod_id(mod) ((mod)->id)
-#define rsnd_mod_hw_start(mod)	clk_prepare_enable((mod)->clk)
-#define rsnd_mod_hw_stop(mod)	clk_disable_unprepare((mod)->clk)
 
 void rsnd_mod_init(struct rsnd_mod *mod,
 		   struct rsnd_mod_ops *ops,
@@ -294,6 +298,7 @@ void rsnd_mod_init(struct rsnd_mod *mod,
 		   enum rsnd_mod_type type,
 		   int id);
 char *rsnd_mod_name(struct rsnd_mod *mod);
+int rsnd_mod_clk(struct rsnd_mod *mod, int enable);
 struct dma_chan *rsnd_mod_dma_req(struct rsnd_mod *mod);
 
 /*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index cc93f32..83032ee 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -292,8 +292,6 @@ static int rsnd_src_init(struct rsnd_mod *mod)
 {
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 
-	rsnd_mod_hw_start(mod);
-
 	src->err = 0;
 
 	/*
@@ -311,8 +309,6 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	rsnd_mod_hw_stop(mod);
-
 	if (src->err)
 		dev_warn(dev, "%s[%d] under/over flow err = %d\n",
 			 rsnd_mod_name(mod), rsnd_mod_id(mod), src->err);
@@ -523,6 +519,7 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = {
 	.quit	= rsnd_src_quit,
 	.start	= rsnd_src_start_gen1,
 	.stop	= rsnd_src_stop_gen1,
+	.clk	= rsnd_mod_clk,
 };
 
 /*
@@ -803,6 +800,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = {
 	.quit	= rsnd_src_quit,
 	.start	= rsnd_src_start_gen2,
 	.stop	= rsnd_src_stop_gen2,
+	.clk	= rsnd_mod_clk,
 };
 
 struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 060d3d2..583ca97 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -176,6 +176,22 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
 	rsnd_adg_ssi_clk_stop(&ssi->mod);
 }
 
+static int rsnd_ssi_clk(struct rsnd_mod *mod, int enable)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+
+	rsnd_mod_clk(mod, enable);
+
+	if (rsnd_rdai_is_clk_master(rdai)) {
+		if (rsnd_ssi_clk_from_parent(ssi))
+			rsnd_mod_clk(&ssi->parent->mod, enable);
+	}
+
+	return 0;
+}
+
 static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 			      struct rsnd_dai_stream *io)
 {
@@ -186,8 +202,6 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 	u32 cr;
 
 	if (0 == ssi->usrcnt) {
-		rsnd_mod_hw_start(&ssi->mod);
-
 		if (rsnd_rdai_is_clk_master(rdai)) {
 			if (rsnd_ssi_clk_from_parent(ssi))
 				rsnd_ssi_hw_start(ssi->parent, io);
@@ -258,8 +272,6 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi)
 			else
 				rsnd_ssi_master_clk_stop(ssi);
 		}
-
-		rsnd_mod_hw_stop(&ssi->mod);
 	}
 
 	dev_dbg(dev, "%s[%d] hw stopped\n",
@@ -462,6 +474,7 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
 	.quit	= rsnd_ssi_quit,
 	.start	= rsnd_ssi_start,
 	.stop	= rsnd_ssi_stop,
+	.clk	= rsnd_ssi_clk,
 };
 
 static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
@@ -581,6 +594,7 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
 	.quit	= rsnd_ssi_quit,
 	.start	= rsnd_ssi_dma_start,
 	.stop	= rsnd_ssi_dma_stop,
+	.clk	= rsnd_ssi_clk,
 	.fallback = rsnd_ssi_fallback,
 };
 
-- 
1.9.1

[PATCH 09/10 v3] ASoC: rsnd: add dai_link stream name

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

This patch adds missing dai_link stream_name
which is used when DPCM

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/core.c | 8 ++++++++
 sound/soc/sh/rcar/rsnd.h | 1 +
 2 files changed, 9 insertions(+)

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e6c1c7a4..929c282 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -675,20 +675,28 @@ static int rsnd_dai_probe(struct platform_device *pdev,
 		drv[i].name	= rdai[i].name;
 		drv[i].ops	= &rsnd_soc_dai_ops;
 		if (pmod) {
+			snprintf(rdai[i].playback.name, RSND_DAI_NAME_SIZE,
+				 "DAI%d Playback", i);
+
 			drv[i].playback.rates		= RSND_RATES;
 			drv[i].playback.formats		= RSND_FMTS;
 			drv[i].playback.channels_min	= 2;
 			drv[i].playback.channels_max	= 2;
+			drv[i].playback.stream_name	= rdai[i].playback.name;
 
 			rdai[i].playback.info = &info->dai_info[i].playback;
 			rdai[i].playback.rdai = rdai + i;
 			rsnd_path_init(priv, &rdai[i], &rdai[i].playback);
 		}
 		if (cmod) {
+			snprintf(rdai[i].capture.name, RSND_DAI_NAME_SIZE,
+				 "DAI%d Capture", i);
+
 			drv[i].capture.rates		= RSND_RATES;
 			drv[i].capture.formats		= RSND_FMTS;
 			drv[i].capture.channels_min	= 2;
 			drv[i].capture.channels_max	= 2;
+			drv[i].capture.stream_name	= rdai[i].capture.name;
 
 			rdai[i].capture.info = &info->dai_info[i].capture;
 			rdai[i].capture.rdai = rdai + i;
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 40a12f4..be1ca7a 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -306,6 +306,7 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_mod *mod);
  */
 #define RSND_DAI_NAME_SIZE	16
 struct rsnd_dai_stream {
+	char name[RSND_DAI_NAME_SIZE];
 	struct snd_pcm_substream *substream;
 	struct rsnd_mod *mod[RSND_MOD_MAX];
 	struct rsnd_dai_path_info *info; /* rcar_snd.h */
-- 
1.9.1

[PATCH 10/10 v3] ASoC: rsnd: add DPCM based sampling rate convert

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

This patch supports DPCM based sampling rate convert on Renesas sound
driver. It assumes...
 1. SRC is implemented as FE
 2. BE dai_link supports .be_hw_params_fixup

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v2 -> v3

 - no change

 sound/soc/sh/rcar/core.c |  9 +++++++++
 sound/soc/sh/rcar/rsnd.h |  6 ++++++
 sound/soc/sh/rcar/src.c  | 49 ++++++++++++++++++++++++++++++++++++++++++++----
 3 files changed, 60 insertions(+), 4 deletions(-)

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 929c282..b3851cf 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -742,6 +742,15 @@ static int rsnd_pcm_open(struct snd_pcm_substream *substream)
 static int rsnd_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *hw_params)
 {
+	struct snd_soc_dai *dai = rsnd_substream_to_dai(substream);
+	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+	int ret;
+
+	ret = rsnd_dai_call(hw_params, io, substream, hw_params);
+	if (ret)
+		return ret;
+
 	return snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index be1ca7a..4ada02b 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -240,6 +240,9 @@ struct rsnd_mod_ops {
 	int (*clk)(struct rsnd_mod *mod, int enable);
 	int (*pcm_new)(struct rsnd_mod *mod,
 		       struct snd_soc_pcm_runtime *rtd);
+	int (*hw_params)(struct rsnd_mod *mod,
+			 struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params);
 	int (*fallback)(struct rsnd_mod *mod,
 			struct rsnd_priv *priv);
 };
@@ -266,6 +269,7 @@ struct rsnd_mod {
  *
  * 31 bit is always called (see __rsnd_mod_call)
  * 31	0: clk
+ * 31	0: hw_params
  */
 #define __rsnd_mod_shift_probe		0
 #define __rsnd_mod_shift_remove		0
@@ -276,6 +280,7 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_pcm_new	3
 #define __rsnd_mod_shift_fallback	4
 #define __rsnd_mod_shift_clk		31 /* always called */
+#define __rsnd_mod_shift_hw_params	31 /* always called */
 
 #define __rsnd_mod_call_probe		0
 #define __rsnd_mod_call_remove		1
@@ -286,6 +291,7 @@ struct rsnd_mod {
 #define __rsnd_mod_call_pcm_new		0
 #define __rsnd_mod_call_fallback	0
 #define __rsnd_mod_call_clk		0
+#define __rsnd_mod_call_hw_params	0
 
 #define rsnd_mod_to_priv(mod) (rsnd_io_to_priv(rsnd_mod_to_io(mod)))
 #define rsnd_mod_to_dma(mod) (&(mod)->dma)
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 83032ee..f53f625 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -22,12 +22,13 @@
 struct rsnd_src {
 	struct rsnd_src_platform_info *info; /* rcar_snd.h */
 	struct rsnd_mod mod;
+	u32 convert_rate; /* sampling rate convert */
 	int err;
 };
 
 #define RSND_SRC_NAME_SIZE 16
 
-#define rsnd_src_convert_rate(p) ((p)->info->convert_rate)
+#define rsnd_src_convert_rate(s) ((s)->convert_rate)
 #define rsnd_src_of_node(priv) \
 	of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src")
 
@@ -288,7 +289,43 @@ static int rsnd_src_set_convert_rate(struct rsnd_mod *mod)
 	return 0;
 }
 
-static int rsnd_src_init(struct rsnd_mod *mod)
+static int rsnd_src_hw_params(struct rsnd_mod *mod,
+			      struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *fe_params)
+{
+	struct rsnd_src *src = rsnd_mod_to_src(mod);
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+
+	/* default value (mainly for non-DT) */
+	src->convert_rate = src->info->convert_rate;
+
+	/*
+	 * SRC assumes that it is used under DPCM if user want to use
+	 * sampling rate convert. Then, SRC should be FE.
+	 * And then, this function will be called *after* BE settings.
+	 * this means, each BE already has fixuped hw_params.
+	 * see
+	 *	dpcm_fe_dai_hw_params()
+	 *	dpcm_be_dai_hw_params()
+	 */
+	if (fe->dai_link->dynamic) {
+		int stream = substream->stream;
+		struct snd_soc_dpcm *dpcm;
+		struct snd_pcm_hw_params *be_params;
+
+		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+			be_params = &dpcm->hw_params;
+
+			if (params_rate(fe_params) != params_rate(be_params))
+				src->convert_rate = params_rate(be_params);
+		}
+	}
+
+	return 0;
+}
+
+static int rsnd_src_init(struct rsnd_mod *mod,
+			 struct rsnd_priv *priv)
 {
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 
@@ -313,6 +350,8 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
 		dev_warn(dev, "%s[%d] under/over flow err = %d\n",
 			 rsnd_mod_name(mod), rsnd_mod_id(mod), src->err);
 
+	src->convert_rate = 0;
+
 	return 0;
 }
 
@@ -472,7 +511,7 @@ static int rsnd_src_init_gen1(struct rsnd_mod *mod,
 {
 	int ret;
 
-	ret = rsnd_src_init(mod);
+	ret = rsnd_src_init(mod, priv);
 	if (ret < 0)
 		return ret;
 
@@ -520,6 +559,7 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = {
 	.start	= rsnd_src_start_gen1,
 	.stop	= rsnd_src_stop_gen1,
 	.clk	= rsnd_mod_clk,
+	.hw_params = rsnd_src_hw_params,
 };
 
 /*
@@ -756,7 +796,7 @@ static int rsnd_src_init_gen2(struct rsnd_mod *mod,
 {
 	int ret;
 
-	ret = rsnd_src_init(mod);
+	ret = rsnd_src_init(mod, priv);
 	if (ret < 0)
 		return ret;
 
@@ -801,6 +841,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = {
 	.start	= rsnd_src_start_gen2,
 	.stop	= rsnd_src_stop_gen2,
 	.clk	= rsnd_mod_clk,
+	.hw_params = rsnd_src_hw_params,
 };
 
 struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
-- 
1.9.1

Re: [PATCH 01/10 v3] ASoC: core: add snd_soc_get_dai_drv() to get DAI driver

[Mark Brown]

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On Thu, Mar 19, 2015 at 04:12:50AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> Sometimes sound card driver want to get cpu/codec driver.
> This patch adds snd_soc_get_dai_drv() and enable it.

Why would a sound card driver need to get the DAI driver?  The changelog
should explain this.

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Re: [PATCH 04/10 v3] ASoC: rsnd: no more SSI restart when unusual situation

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 264 bytes --]

On Thu, Mar 19, 2015 at 04:13:47AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> It will be SSI interrupt endless loop f unusual situation happen.
> This patch adds restart limit for it.

Applied, thanks.

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Re: [PATCH 05/10 v3] ASoC: rsnd: no more SRC restart when unusual situation

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 264 bytes --]

On Thu, Mar 19, 2015 at 04:14:04AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> It will be SRC interrupt endless loop f unusual situation happen.
> This patch adds restart limit for it.

Applied, thanks.

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Re: [PATCH 06/10 v3] ASoC: rsnd: tidyup error message format

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 313 bytes --]

On Thu, Mar 19, 2015 at 04:14:20AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> This driver sometimes fixups debug/error message format
> 30cc4faf703955cd5cd07da489bd817ae43e3fec
> (ASoC: rsnd: tidyup debug message format and timing)

Applied, thanks.

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Re: [PATCH 07/10 v3] ASoC: rsnd: show debug info for sampling rate convert

[Mark Brown]

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On Thu, Mar 19, 2015 at 04:14:45AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

Applied, thanks.

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Re: [PATCH 08/10 v3] ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock

[Mark Brown]

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On Thu, Mar 19, 2015 at 04:15:01AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> clk_prepare_enable/unprepare() uses mutex inside, and it uses __schedule().
> Then, raw_spin_lock/unlock_irq() is called, and it breaks Renesas
> sound driver's spin lock irq.
> This patch moves clk_prepare_enable/unprepare to out of spin lock area.
> Special thanks to Das Biju.

This is definitely a bug which should be fixed but this means we should
send a fix to Linus rather than having something that depends on -next.
Also...

> +int rsnd_mod_clk(struct rsnd_mod *mod, int enable)
> +{
> +	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
> +	struct device *dev = rsnd_priv_to_dev(priv);
> +
> +	/*
> +	 * clk_prepare_enable/unprepare() should not be called
> +	 * from inside spin lock
> +	 */
> +	if (enable)
> +		clk_prepare_enable(mod->clk);
> +	else
> +		clk_disable_unprepare(mod->clk);

It's not entirely clear what this wrapper function is doing (and note
that we don't check the error code from clk_prepare_enable()).

> @@ -345,6 +366,9 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
>  	int ret;
>  	unsigned long flags;
>  
> +	if (cmd == SNDRV_PCM_TRIGGER_START)
> +		rsnd_dai_call(clk, io, 1);
> +
>  	rsnd_lock(priv, flags);

The trigger function is called from atomic context so this seems to
leave us with the same problem this was supposed to be fixing?

> +static int rsnd_ssi_clk(struct rsnd_mod *mod, int enable)
> +{
> +	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
> +	struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
> +	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
> +
> +	rsnd_mod_clk(mod, enable);
> +
> +	if (rsnd_rdai_is_clk_master(rdai)) {
> +		if (rsnd_ssi_clk_from_parent(ssi))
> +			rsnd_mod_clk(&ssi->parent->mod, enable);
> +	}

This would be clearer with a single if statement with an &&, however it
seems like something that the clock API should be doing itself - it's
already got support for handling enabling of parents.  If we need to
manually handle dependencies (which is what this looks like) that seems
like a problem.

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Re: [PATCH 09/10 v3] ASoC: rsnd: add dai_link stream name

[Mark Brown]

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On Thu, Mar 19, 2015 at 04:15:18AM +0000, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> This patch adds missing dai_link stream_name
> which is used when DPCM

Applied, thanks.

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Re: [PATCH 01/10 v3] ASoC: core: add snd_soc_get_dai_drv() to get DAI driver

[Kuninori Morimoto]

Hi Mark

> > Sometimes sound card driver want to get cpu/codec driver.
> > This patch adds snd_soc_get_dai_drv() and enable it.
> 
> Why would a sound card driver need to get the DAI driver?  The changelog
> should explain this.

Because, sound card needs to set dai_link->dpcm_playback / dai_link->dpcm_capture
on soc_new_pcm() if it was DPCM. but, it is depends on CPU/Codec driver.
So, sound card driver need to get DAI driver, and consider it.
I will add this explain in v4

Re: [PATCH 08/10 v3] ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock

[Kuninori Morimoto]

Hi Mark

> > +static int rsnd_ssi_clk(struct rsnd_mod *mod, int enable)
> > +{
> > +	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
> > +	struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
> > +	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
> > +
> > +	rsnd_mod_clk(mod, enable);
> > +
> > +	if (rsnd_rdai_is_clk_master(rdai)) {
> > +		if (rsnd_ssi_clk_from_parent(ssi))
> > +			rsnd_mod_clk(&ssi->parent->mod, enable);
> > +	}
> 
> This would be clearer with a single if statement with an &&, however it
> seems like something that the clock API should be doing itself - it's
> already got support for handling enabling of parents.  If we need to
> manually handle dependencies (which is what this looks like) that seems
> like a problem.

Hmm... indeed.
Thank you for your feedback, I will reconsider about this

Re: [PATCH 08/10 v3] ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock

[Kuninori Morimoto]

Hi Mark again

> > +static int rsnd_ssi_clk(struct rsnd_mod *mod, int enable)
> > +{
> > +	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
> > +	struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
> > +	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
> > +
> > +	rsnd_mod_clk(mod, enable);
> > +
> > +	if (rsnd_rdai_is_clk_master(rdai)) {
> > +		if (rsnd_ssi_clk_from_parent(ssi))
> > +			rsnd_mod_clk(&ssi->parent->mod, enable);
> > +	}
> 
> This would be clearer with a single if statement with an &&, however it
> seems like something that the clock API should be doing itself - it's
> already got support for handling enabling of parents.  If we need to
> manually handle dependencies (which is what this looks like) that seems
> like a problem.

I rechecked this.
In our sound device, sound LR and clock pin is controled by PFC (= pin function controller)
and, these pins are shared with other device. it depends on platform/board.
If other device uses this pin, we can't use it in sound device.
Then, this sound device share LR/clock pin in sound device.
And then, device clock has no relation to this

The image is like this

If no other devices use pin, sound device can use all pins

SSI0 deivce clock
 SSI0 LR  ----------------|------
 SSI0 clk ----------------|------
                          |
SSI1 deivce clock         |
 SSI1 LR  ----------------|------
 SSI1 clk ----------------|------
 
If other device uses pin, sound device share same pins

SSI0 deivce clock
 SSI0 LR  ---+------------|------
 SSI0 clk ---|+-----------|------
             ||           |
SSI1 deivce clock         |
 SSI1 LR  ---+|   Dev A --|------
 SSI1 clk ----+   Dev B --|------

Here, SSI1 needs SSI0's LR/clk pin, but it doesn't need SSI0's device clock
because SSI1 has SSI1 device clock.
SSI0 is called as parent for SSI1 in my driver, but it is not "clock parent".
And here, if you want to use SSI1, you need to start SSI0 too.
So Unfortunately, manually handle dependencies is understandable IMO.

Re: [PATCH 01/10 v3] ASoC: core: add snd_soc_get_dai_drv() to get DAI driver

[Mark Brown]

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On Mon, Mar 23, 2015 at 07:14:51AM +0000, Kuninori Morimoto wrote:
> > > Sometimes sound card driver want to get cpu/codec driver.
> > > This patch adds snd_soc_get_dai_drv() and enable it.

> > Why would a sound card driver need to get the DAI driver?  The changelog
> > should explain this.

> Because, sound card needs to set dai_link->dpcm_playback / dai_link->dpcm_capture
> on soc_new_pcm() if it was DPCM. but, it is depends on CPU/Codec driver.
> So, sound card driver need to get DAI driver, and consider it.
> I will add this explain in v4

Hrm, I see.  This doesn't feel quite elegant - can we do something like
factor this check out into the core here?  It feels like something the
card driver ought not to need to worry about.

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Re: [PATCH 01/10 v3] ASoC: core: add snd_soc_get_dai_drv() to get DAI driver

[Kuninori Morimoto]

Hi Mark

> > Because, sound card needs to set dai_link->dpcm_playback / dai_link->dpcm_capture
> > on soc_new_pcm() if it was DPCM. but, it is depends on CPU/Codec driver.
> > So, sound card driver need to get DAI driver, and consider it.
> > I will add this explain in v4
> 
> Hrm, I see.  This doesn't feel quite elegant - can we do something like
> factor this check out into the core here?  It feels like something the
> card driver ought not to need to worry about.

OK, I will reconsider it.