audio: add loopback

PD#147538: audio: loopback for pdm/spdif/tdm

1. pdm/spdif/tdm as loopback datain soruce, tdmin_lb as datalb source
2. add mixer kcontrols for loopback

Change-Id: I579db913080b3bb02bc99885eb330e24af8a0edb
Signed-off-by: Xing Wang <xing.wang@amlogic.com>

MAINTAINERS

@@ -13994,3 +13994,11 @@ F:	driver/amlogic/ddr_window/Kconfig
 AMLOGIC AUDIO INFO
 M:	wang xing <xing.wang@amlogic.com>
 F: drivers/amlogic/audioinfo/
+
+AMLOGIC AXG ADD LOOPBACK INTERFACE
+M: wang xing <xing.wang@amlogic.com>
+F: arch/arm64/boot/dts/amlogic/axg_s400.dts
+F: sound/soc/amlogic/auge/*
+F: sound/soc/codecs/amlogic/pdm_dummy.c
+F: sound/soc/codecs/amlogic/tlv320adc3101.c
+

arch/arm64/boot/dts/amlogic/axg_s400.dts

@@ -446,6 +446,8 @@
 		aml-audio-card,name = "AML-AXGSOUND";
 		//aml-audio-card,mclk-fs = <256>;
 
+		aml-audio-card,loopback = <&aml_loopback>;
+
 		aml-audio-card,dai-link@0 {
 			format = "dsp_a";
 			mclk-fs = <512>;
@@ -984,6 +986,40 @@
 		filter_mode = <1>; /* mode 0~4, defalut:1 */
 		status = "okay";
 	};
+
+	aml_loopback: loopback {
+		compatible = "amlogic, snd-loopback";
+		/*
+		 * 0: out rate = in data rate;
+		 * 1: out rate = loopback data rate;
+		 */
+		lb_mode = <0>;
+
+		/* datain src
+		 * 0: tdmin_a;
+		 * 1: tdmin_b;
+		 * 2: tdmin_c;
+		 * 3: spdifin;
+		 * 4: pdmin;
+		 */
+		datain_src = <4>;
+		datain_chnum = <8>;
+		datain_chmask = <0xfc>;
+
+		/* tdmin_lb src
+		 * 0: tdmoutA
+		 * 1: tdmoutB
+		 * 2: tdmoutC
+		 * 3: PAD_tdminA
+		 * 4: PAD_tdminB
+		 * 5: PAD_tdminC
+		 */
+		datalb_src = <2>;
+		datalb_chnum = <8>;
+		datalb_chmask = <0x3>;
+
+		status = "okay";
+	};
 }; /* end of audiobus */
 
 &pinctrl_periphs {

sound/soc/amlogic/auge/Makefile

@@ -11,4 +11,6 @@ obj-$(CONFIG_AMLOGIC_SND_SOC_AUGE)	+= audio_controller.o \
 						pdm_hw.o \
 						pdm_hw_coeff.o \
 						iomap.o \
-						ddr_mngr.o
+						ddr_mngr.o \
+						loopback_hw.o \
+						audio_utils.o

sound/soc/amlogic/auge/audio_utils.c

@@ -0,0 +1,331 @@
+/*
+ * sound/soc/amlogic/auge/audio_utils.c
+ *
+ * Copyright (C) 2017 Amlogic, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ */
+
+#include "audio_utils.h"
+#include "regs.h"
+#include "iomap.h"
+#include "loopback_hw.h"
+
+static unsigned int loopback_enable;
+
+static const char *const loopback_enable_texts[] = {
+	"Disable",
+	"Enable",
+};
+
+static const struct soc_enum loopback_enable_enum =
+	SOC_ENUM_SINGLE(EE_AUDIO_LB_CTRL0,
+			31,
+			ARRAY_SIZE(loopback_enable_texts),
+			loopback_enable_texts);
+
+
+static int loopback_enable_get_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = loopback_enable;
+
+	return 0;
+}
+
+static int loopback_enable_set_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	loopback_enable = ucontrol->value.enumerated.item[0];
+
+	return 0;
+}
+
+
+static unsigned int loopback_datain;
+
+static const char *const loopback_datain_texts[] = {
+	"Tdmin A",
+	"Tdmin B",
+	"Tdmin C",
+	"Spdif In",
+	"Pdm In",
+};
+
+static const struct soc_enum loopback_datain_enum =
+	SOC_ENUM_SINGLE(EE_AUDIO_LB_CTRL0, 0, ARRAY_SIZE(loopback_datain_texts),
+			loopback_datain_texts);
+
+
+static int loopback_datain_get_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = loopback_datain;
+
+	return 0;
+}
+
+static int loopback_datain_set_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	loopback_datain = ucontrol->value.enumerated.item[0];
+	audiobus_update_bits(EE_AUDIO_LB_CTRL0, 0, loopback_datain);
+
+	return 0;
+}
+
+static unsigned int loopback_tdminlb;
+
+static const char *const loopback_tdminlb_texts[] = {
+	"tdmoutA",
+	"tdmoutB",
+	"tdmoutC",
+	"tdminA",
+	"tdminB",
+	"tdminC",
+};
+
+static const struct soc_enum loopback_tdminlb_enum =
+	SOC_ENUM_SINGLE(EE_AUDIO_TDMIN_LB_CTRL,
+			20,
+			ARRAY_SIZE(loopback_tdminlb_texts),
+			loopback_tdminlb_texts);
+
+
+static int loopback_tdminlb_get_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = loopback_tdminlb;
+
+	return 0;
+}
+
+static int loopback_tdminlb_set_enum(
+	struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	loopback_tdminlb = ucontrol->value.enumerated.item[0];
+	audiobus_update_bits(EE_AUDIO_TDMIN_LB_CTRL,
+		0xf << 20,
+		loopback_datain);
+
+	return 0;
+}
+
+
+static const struct snd_kcontrol_new snd_auge_controls[] = {
+	/* loopback enable */
+	SOC_ENUM_EXT("Loopback Enable",
+		     loopback_enable_enum,
+		     loopback_enable_get_enum,
+		     loopback_enable_set_enum),
+
+	/* loopback data in source */
+	SOC_ENUM_EXT("Loopback datain source",
+		     loopback_datain_enum,
+		     loopback_datain_get_enum,
+		     loopback_datain_set_enum),
+
+	/* loopback data tdmin lb */
+	SOC_ENUM_EXT("Loopback tmdin lb source",
+		     loopback_tdminlb_enum,
+		     loopback_tdminlb_get_enum,
+		     loopback_tdminlb_set_enum),
+
+};
+
+
+int snd_card_add_kcontrols(struct snd_soc_card *card)
+{
+	pr_info("%s card:%p\n", __func__, card);
+	return snd_soc_add_card_controls(card,
+		snd_auge_controls, ARRAY_SIZE(snd_auge_controls));
+
+}
+
+int loopback_parse_of(struct device_node *node,
+	struct loopback_cfg *lb_cfg)
+{
+	int ret;
+
+	ret = of_property_read_u32(node, "lb_mode",
+		&lb_cfg->lb_mode);
+	if (ret) {
+		pr_err("failed to get lb_mode, set it default\n");
+		lb_cfg->lb_mode = 0;
+		ret = 0;
+	}
+
+	ret = of_property_read_u32(node, "datain_src",
+		&lb_cfg->datain_src);
+	if (ret) {
+		pr_err("failed to get datain_src\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	ret = of_property_read_u32(node, "datain_chnum",
+		&lb_cfg->datain_chnum);
+	if (ret) {
+		pr_err("failed to get datain_chnum\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	ret = of_property_read_u32(node, "datain_chmask",
+		&lb_cfg->datain_chmask);
+	if (ret) {
+		pr_err("failed to get datain_chmask\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ret = of_property_read_u32(node, "datalb_src",
+		&lb_cfg->datalb_src);
+	if (ret) {
+		pr_err("failed to get datalb_src\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	ret = of_property_read_u32(node, "datalb_chnum",
+		&lb_cfg->datalb_chnum);
+	if (ret) {
+		pr_err("failed to get datalb_chnum\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	ret = of_property_read_u32(node, "datalb_chmask",
+		&lb_cfg->datalb_chmask);
+	if (ret) {
+		pr_err("failed to get datalb_chmask\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+
+	loopback_datain = lb_cfg->datain_src;
+	loopback_tdminlb = lb_cfg->datalb_src;
+
+	pr_info("parse loopback:, \tlb mode:%d\n",
+		lb_cfg->lb_mode);
+	pr_info("\tdatain_src:%d, datain_chnum:%d, datain_chumask:%x\n",
+		lb_cfg->datain_src,
+		lb_cfg->datain_chnum,
+		lb_cfg->datain_chmask);
+	pr_info("\tdatalb_src:%d, datalb_chnum:%d, datalb_chumask:%x\n",
+		lb_cfg->datalb_src,
+		lb_cfg->datalb_chnum,
+		lb_cfg->datalb_chmask);
+
+fail:
+	return ret;
+}
+
+int loopback_prepare(
+	struct snd_pcm_substream *substream,
+	struct loopback_cfg *lb_cfg)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int bitwidth;
+	unsigned int datain_toddr_type, datalb_toddr_type;
+	unsigned int datain_msb, datain_lsb, datalb_msb, datalb_lsb;
+	struct lb_cfg datain_cfg;
+	struct lb_cfg datalb_cfg;
+	struct audio_data ddrdata;
+	struct data_in datain;
+	struct data_lb datalb;
+
+	if (!lb_cfg)
+		return -EINVAL;
+
+	bitwidth = snd_pcm_format_width(runtime->format);
+	switch (lb_cfg->datain_src) {
+	case DATAIN_TDMA:
+	case DATAIN_TDMB:
+	case DATAIN_TDMC:
+	case DATAIN_PDM:
+		datain_toddr_type = 0;
+		datain_msb = 32 - 1;
+		datain_lsb = 0;
+		break;
+	case DATAIN_SPDIF:
+		datain_toddr_type = 3;
+		datain_msb = 27;
+		datain_lsb = 4;
+		if (bitwidth <= 24)
+			datain_lsb = 28 - bitwidth;
+		else
+			datain_lsb = 4;
+		break;
+	default:
+		pr_err("unsupport data in source:%d\n",
+			lb_cfg->datain_src);
+		return -EINVAL;
+	}
+
+	switch (lb_cfg->datalb_src) {
+	case TDMINLB_TDMOUTA:
+	case TDMINLB_TDMOUTB:
+	case TDMINLB_TDMOUTC:
+	case TDMINLB_PAD_TDMINA:
+	case TDMINLB_PAD_TDMINB:
+	case TDMINLB_PAD_TDMINC:
+		if (bitwidth == 24) {
+			datalb_toddr_type = 4;
+			datalb_msb = 32 - 1;
+			datalb_lsb = 32 - bitwidth;
+		} else {
+			datalb_toddr_type = 0;
+			datalb_msb = 32 - 1;
+			datalb_lsb = 0;
+		}
+		break;
+	default:
+		pr_err("unsupport data lb source:%d\n",
+			lb_cfg->datalb_src);
+		return -EINVAL;
+	}
+
+	datain_cfg.ext_signed    = 0;
+	datain_cfg.chnum         = runtime->channels;
+	datain_cfg.chmask        = lb_cfg->datain_chmask;
+	ddrdata.combined_type    = datain_toddr_type;
+	ddrdata.msb              = datain_msb;
+	ddrdata.lsb              = datain_lsb;
+	ddrdata.src              = lb_cfg->datain_src;
+	datain.config            = &datain_cfg;
+	datain.ddrdata           = &ddrdata;
+
+	datalb_cfg.ext_signed    = 0;
+	datalb_cfg.chnum         = lb_cfg->datalb_chnum;
+	datalb_cfg.chmask        = lb_cfg->datalb_chmask;
+	datalb.config            = &datalb_cfg;
+	datalb.ddr_type          = datalb_toddr_type;
+	datalb.msb               = datalb_msb;
+	datalb.lsb               = datalb_lsb;
+
+	datain_config(&datain);
+	datalb_config(&datalb);
+
+	datalb_ctrl(lb_cfg->datalb_src, bitwidth);
+	lb_enable_ex(lb_cfg->lb_mode, true);
+
+	return 0;
+}
+
+int loopback_is_enable(void)
+{
+	return (loopback_enable == 1);
+}

sound/soc/amlogic/auge/audio_utils.h

@@ -0,0 +1,137 @@
+/*
+ * sound/soc/amlogic/auge/audio_utils.h
+ *
+ * Copyright (C) 2017 Amlogic, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ */
+
+#ifndef __AML_AUDIO_UTILS_H__
+#define __AML_AUDIO_UTILS_H__
+
+#include <linux/types.h>
+#include <sound/soc.h>
+
+/* datain src
+ * [4]: pdmin;
+ * [3]: spdifin;
+ * [2]: tdmin_c;
+ * [1]: tdmin_b;
+ * [0]: tdmin_a;
+ */
+enum datain_src {
+	DATAIN_TDMA = 0,
+	DATAIN_TDMB,
+	DATAIN_TDMC,
+	DATAIN_SPDIF,
+	DATAIN_PDM,
+};
+
+/* datalb src/tdmlb src
+ * [0]: tdmoutA
+ * [1]: tdmoutB
+ * [2]: tdmoutC
+ * [3]: PAD_tdminA
+ * [4]: PAD_tdminB
+ * [5]: PAD_tdminC
+ */
+enum tdmin_lb_src {
+	TDMINLB_TDMOUTA = 0,
+	TDMINLB_TDMOUTB,
+	TDMINLB_TDMOUTC,
+	TDMINLB_PAD_TDMINA,
+	TDMINLB_PAD_TDMINB,
+	TDMINLB_PAD_TDMINC,
+};
+
+/* toddr_src_sel
+ *		[7]: loopback;
+ *		[6]: tdmin_lb;
+ *		[5]: reserved;
+ *		[4]: pdmin;
+ *		[3]: spdifin;
+ *		[2]: tdmin_c;
+ *		[1]: tdmin_b;
+ *		[0]: tdmin_a;
+ */
+enum fifoin_src {
+	FIFOIN_TDMINA   = 0,
+	FIFOIN_TDMINB   = 1,
+	FIFOIN_TDMINC   = 2,
+	FIFOIN_SPDIF    = 3,
+	FIFOIN_PDM      = 4,
+	FIFOIN_TDMINLB  = 6,
+	FIFOIN_LOOPBACK = 7
+};
+
+/* audio data selected to ddr */
+struct audio_data {
+	unsigned int resample;
+	/* reg_dat_sel
+	 *      0: combined data[m:n] without gap;
+	 *              like S0[m:n],S1[m:n],S2[m:n],
+	 *      1: combined data[m:n] as 16bits;
+	 *              like {S0[11:0],4'd0},{S1[11:0],4'd0}
+	 *      2: combined data[m:n] as 16bits;
+	 *              like {4'd0,S0[11:0]},{4'd0,{S1[11:0]}
+	 *      3: combined data[m:n] as 32bits;
+	 *              like {S0[27:4],8'd0},{S1[27:4],8'd0}
+	 *      4: combined data[m:n] as 32bits;
+	 *              like {8'd0,S0[27:4]},{8'd0,{S1[27:4]}
+	 */
+	unsigned int combined_type;
+	/* the msb positioin in data */
+	unsigned int msb;
+	/* the lsb position in data */
+	unsigned int lsb;
+	/* toddr_src_sel
+	 *      [7]: loopback;
+	 *      [6]: tdmin_lb;
+	 *      [5]: reserved;
+	 *      [4]: pdmin;
+	 *      [3]: spdifin;
+	 *      [2]: tdmin_c;
+	 *      [1]: tdmin_b;
+	 *      [0]: tdmin_a;
+	 */
+	unsigned int src;
+};
+
+/**/
+struct loopback_cfg {
+	/* lb_mode
+	 * 0: out rate = in data rate;
+	 * 1: out rate = loopback data rate;
+	 */
+	unsigned int lb_mode;
+
+	enum datain_src datain_src;
+	unsigned int datain_chnum;
+	unsigned int datain_chmask;
+
+	enum tdmin_lb_src datalb_src;
+	unsigned int datalb_chnum;
+	unsigned int datalb_chmask;
+};
+
+extern int loopback_is_enable(void);
+
+extern int snd_card_add_kcontrols(struct snd_soc_card *card);
+
+extern int loopback_parse_of(struct device_node *node,
+					   struct loopback_cfg *lb_cfg);
+
+extern int loopback_prepare(
+	struct snd_pcm_substream *substream,
+	struct loopback_cfg *lb_cfg);
+
+#endif

sound/soc/amlogic/auge/card.c

@@ -47,6 +47,7 @@ struct aml_card_data {
 	struct snd_soc_dai_link *dai_link;
 	int spk_mute_gpio;
 	bool spk_mute_active_low;
+	struct loopback_cfg lb_cfg;
 };
 
 #define aml_priv_to_dev(priv) ((priv)->snd_card.dev)
@@ -217,10 +218,21 @@ err:
 	return ret;
 }
 
+int aml_card_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct aml_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+
+	loopback_prepare(substream, &priv->lb_cfg);
+
+	return 0;
+}
+
 static struct snd_soc_ops aml_card_ops = {
-	.startup = aml_card_startup,
-	.shutdown = aml_card_shutdown,
-	.hw_params = aml_card_hw_params,
+	.startup    = aml_card_startup,
+	.shutdown   = aml_card_shutdown,
+	.hw_params  = aml_card_hw_params,
+	.prepare    = aml_card_prepare,
 };
 
 static int aml_card_dai_init(struct snd_soc_pcm_runtime *rtd)
@@ -466,6 +478,7 @@ static int aml_card_parse_of(struct device_node *node,
 {
 	struct device *dev = aml_priv_to_dev(priv);
 	struct device_node *dai_link;
+	struct device_node *lb_link;
 	int ret;
 
 	if (!node)
@@ -492,6 +505,14 @@ static int aml_card_parse_of(struct device_node *node,
 	/* Factor to mclk, used in hw_params() */
 	of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs);
 
+	/* Loopback */
+	lb_link = of_parse_phandle(node, PREFIX "loopback", 0);
+	if (lb_link) {
+		ret = loopback_parse_of(lb_link, &priv->lb_cfg);
+		if (ret < 0)
+			pr_err("failed parse loopback, ignore it\n");
+	}
+
 	/* Single/Muti DAI link(s) & New style of DT node */
 	if (dai_link) {
 		struct device_node *np = NULL;
@@ -522,6 +543,7 @@ static int aml_card_parse_of(struct device_node *node,
 
 card_parse_end:
 	of_node_put(dai_link);
+	of_node_put(lb_link);
 
 	return ret;
 }
@@ -608,6 +630,14 @@ static int aml_card_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
 	if (ret < 0) {
+		dev_err(dev, "failed to register sound card\n");
+		goto err;
+	}
+
+	/* Add controls */
+	ret = aml_card_add_controls(&priv->snd_card);
+	if (ret < 0) {
+		dev_err(dev, "failed to register mixer kcontrols\n");
 		goto err;
 	}
 

sound/soc/amlogic/auge/card_utils.c

@@ -18,6 +18,8 @@
 #include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/kernel.h>
+
 #include "card_utils.h"
 
 int aml_card_parse_daifmt(struct device *dev,
@@ -288,3 +290,16 @@ int aml_card_clean_reference(struct snd_soc_card *card)
 
 	return 0;
 }
+
+int aml_card_add_controls(struct snd_soc_card *card)
+{
+	int ret;
+
+	ret = snd_card_add_kcontrols(card);
+	if (ret < 0) {
+		pr_info("failed register kcontrols\n");
+		return ret;
+	}
+
+	return 0;
+}

sound/soc/amlogic/auge/card_utils.h

@@ -19,6 +19,7 @@
 #define __AML_CARD_CORE_H
 
 #include <sound/soc.h>
+#include "audio_utils.h"
 
 struct aml_dai {
 	const char *name;
@@ -78,4 +79,6 @@ void aml_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
 
 int aml_card_clean_reference(struct snd_soc_card *card);
 
+extern int aml_card_add_controls(struct snd_soc_card *card);
+
 #endif /* __AML_CARD_CORE_H */

sound/soc/amlogic/auge/ddr_mngr.c

@@ -19,6 +19,7 @@
 #include <linux/slab.h>
 #include "regs.h"
 #include "ddr_mngr.h"
+#include "audio_utils.h"
 
 static DEFINE_MUTEX(ddr_mutex);
 
@@ -218,6 +219,10 @@ void aml_toddr_select_src(struct toddr *to, enum toddr_src src)
 	unsigned int reg_base = to->reg_base;
 	unsigned int reg;
 
+	/* check whether loopback enable */
+	if (loopback_is_enable())
+		src = LOOPBACK;
+
 	reg = calc_toddr_address(EE_AUDIO_TODDR_A_CTRL0, reg_base);
 	aml_audiobus_update_bits(actrl,	reg, 0x7, src & 0x7);
 }

sound/soc/amlogic/auge/loopback_hw.c

@@ -0,0 +1,125 @@
+/*
+ * sound/soc/amlogic/auge/loopback_hw.c
+ *
+ * Copyright (C) 2017 Amlogic, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ */
+#include <sound/soc.h>
+
+#include "regs.h"
+#include "loopback_hw.h"
+#include "iomap.h"
+
+void datain_config(struct data_in *datain)
+{
+	audiobus_write(
+		EE_AUDIO_LB_CTRL0,
+		datain->config->ext_signed  << 29 |
+		(datain->config->chnum - 1) << 24 |
+		datain->config->chmask      << 16 |
+		datain->ddrdata->combined_type << 13 |
+		datain->ddrdata->msb        << 8  |
+		datain->ddrdata->lsb        << 3  |
+		datain->ddrdata->src        << 0
+		);
+}
+
+void datalb_config(struct data_lb *datalb)
+{
+	audiobus_write(
+		EE_AUDIO_LB_CTRL1,
+		datalb->config->ext_signed  << 29 |
+		(datalb->config->chnum - 1) << 24 |
+		datalb->config->chmask      << 16 |
+		datalb->ddr_type            << 13 |
+		datalb->msb                 << 8  |
+		datalb->lsb                 << 3
+		);
+}
+
+void datalb_ctrl(int lb_src, int bitdepth)
+{
+	//tdmin lb,  same as tdm out
+	audiobus_update_bits(
+		EE_AUDIO_CLK_TDMIN_LB_CTRL,
+		0x3 << 30 | 0 << 29 | 0xf << 24 | 0xf << 20,
+		0x3 << 30 | 0 << 29 | 2 << 24 | 2 << 20
+		);
+	//tdmin ctrl, from tdmout
+	//reset
+	audiobus_update_bits(EE_AUDIO_TDMIN_LB_CTRL, 3<<28, 0);
+	audiobus_update_bits(EE_AUDIO_TDMIN_LB_CTRL, 1<<29, 1<<29);
+	audiobus_update_bits(EE_AUDIO_TDMIN_LB_CTRL, 1<<28, 1<<28);
+
+	audiobus_write(
+		EE_AUDIO_TDMIN_LB_CTRL,
+		1 << 31 |
+		0 << 30 | /*0:tdm mode; 1: i2s mode;*/
+		1 << 29 |
+		1 << 28 |
+		lb_src << 20|
+		4 << 16|
+		(bitdepth - 1) << 0
+		);
+	audiobus_write(
+		EE_AUDIO_TDMIN_LB_MASK0,
+		0xff);
+
+}
+
+void lb_enable_ex(int mode, bool is_enable)
+{
+	audiobus_update_bits(
+		EE_AUDIO_LB_CTRL0,
+		0x3 << 30,
+		is_enable << 31 |
+		mode << 30
+		);
+}
+
+
+void lb_set_datain_src(int src)
+{
+}
+
+void lb_set_tdminlb_src(int src)
+{
+}
+
+void lb_set_tdminlb_enable(bool is_enable)
+{
+}
+
+void lb_enable(bool is_enable)
+{
+	if (is_enable)
+		audiobus_update_bits(
+			EE_AUDIO_LB_CTRL0,
+			0x1           << 31,
+			is_enable     << 31
+			);
+	else
+		audiobus_write(
+			EE_AUDIO_LB_CTRL0,
+			0);
+}
+
+void lb_set_mode(int mode)
+{
+	audiobus_update_bits(
+		EE_AUDIO_LB_CTRL0,
+		0x1           << 30,
+		mode          << 30
+		);
+}
+

sound/soc/amlogic/auge/loopback_hw.h

@@ -0,0 +1,64 @@
+/*
+ * sound/soc/amlogic/auge/loopback_hw.h
+ *
+ * Copyright (C) 2017 Amlogic, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ */
+
+#ifndef __AML_LOOPBACK_HW_H__
+#define __AML_LOOPBACK_HW_H__
+
+#include "audio_utils.h"
+
+struct lb_cfg {
+	/*
+	 * 0: extend bits as "0"
+	 * 1: extend bits as "msb"
+	 */
+	unsigned int ext_signed;
+	/* total channel number */
+	unsigned int chnum;
+	/* which channel is selected for loopback */
+	unsigned int chmask;
+};
+
+struct data_in {
+	struct lb_cfg *config;
+	struct audio_data *ddrdata;
+};
+
+struct data_lb {
+	struct lb_cfg *config;
+	unsigned int ddr_type;
+	unsigned int msb;
+	unsigned int lsb;
+};
+
+struct loopback {
+	struct device *dev;
+	unsigned int lb_mode;
+
+	struct data_in *datain;
+	struct data_lb *datalb;
+};
+
+
+extern void datain_config(struct data_in *datain);
+
+extern void datalb_config(struct data_lb *datalb);
+
+extern void datalb_ctrl(int lb_src, int bitdepth);
+
+extern void lb_enable_ex(int mode, bool is_enable);
+
+#endif

sound/soc/amlogic/auge/pdm.c

@@ -47,11 +47,11 @@ static struct snd_pcm_hardware aml_pdm_hardware = {
 					SNDRV_PCM_FMTBIT_S24 |
 					SNDRV_PCM_FMTBIT_S32,
 
-	.rate_min		=	8000,
-	.rate_max		=	48000,
+	.rate_min			=	8000,
+	.rate_max			=	48000,
 
-	.channels_min		=	1,
-	.channels_max		=	8,
+	.channels_min		=	PDM_CHANNELS_MIN,
+	.channels_max		=	PDM_CHANNELS_MAX,
 
 	.buffer_bytes_max	=	32 * 1024,
 	.period_bytes_max	=	16 * 1024,
@@ -473,10 +473,10 @@ static int aml_pdm_dai_prepare(
 
 	switch (bitwidth) {
 	case 16:
-		toddr_type = 2;
+	case 32:
+		toddr_type = 0;
 		break;
 	case 24:
-	case 32:
 		toddr_type = 4;
 		break;
 	default:

sound/soc/amlogic/auge/pdm.h

@@ -27,9 +27,13 @@
 #define DEFAULT_FS_RATIO		256
 
 #define PDM_CHANNELS_MIN		1
-#define PDM_CHANNELS_MAX		8
+#define PDM_CHANNELS_MAX		(8 + 8) /* 8ch pdm in, 8 ch tdmin_lb */
+
+#define PDM_RATES			(SNDRV_PCM_RATE_48000 |\
+						SNDRV_PCM_RATE_32000 |\
+						SNDRV_PCM_RATE_16000 |\
+						SNDRV_PCM_RATE_8000)
 
-#define PDM_RATES			SNDRV_PCM_RATE_8000_48000
 #define PDM_FORMATS			(SNDRV_PCM_FMTBIT_S16_LE |\
 						SNDRV_PCM_FMTBIT_S24_LE |\
 						SNDRV_PCM_FMTBIT_S32_LE)

sound/soc/amlogic/auge/spdif.c

@@ -34,7 +34,7 @@
 
 #include "ddr_mngr.h"
 #include "spdif_hw.h"
-
+#include "audio_utils.h"
 
 #define DRV_NAME "aml_spdif"
 
@@ -412,36 +412,55 @@ static int aml_dai_spdif_prepare(
 		aml_frddr_set_fifos(fr, 0x40, 0x20);
 	} else {
 		struct toddr *to = p_spdif->tddr;
-		unsigned int lsb, toddr_type;
+		unsigned int msb, lsb, toddr_type;
 
-		switch (bit_depth) {
-		case 8:
-			toddr_type = 0;
-			break;
-		case 16:
-			toddr_type = 1;
-			break;
-		case 24:
-			toddr_type = 4;
-			break;
-		case 32:
-			toddr_type = 3;
-			break;
-		default:
-			dev_err(p_spdif->dev,
-				"runtime format invalid bit_depth: %d\n",
-				bit_depth);
-			return -EINVAL;
+		if (loopback_is_enable()) {
+			switch (bit_depth) {
+			case 8:
+			case 16:
+			case 32:
+				toddr_type = 0;
+				break;
+			case 24:
+				toddr_type = 4;
+				break;
+			default:
+				pr_err(
+					"runtime format invalid bit_depth: %d\n",
+					bit_depth);
+				return -EINVAL;
+			}
+			msb = 32 - 1;
+			lsb = 32 - bit_depth;
+		} else {
+			switch (bit_depth) {
+			case 8:
+			case 16:
+				toddr_type = 0;
+				break;
+			case 24:
+				toddr_type = 4;
+				break;
+			case 32:
+				toddr_type = 3;
+				break;
+			default:
+				dev_err(p_spdif->dev,
+					"runtime format invalid bit_depth: %d\n",
+					bit_depth);
+				return -EINVAL;
+			}
+
+			msb = 28 - 1;
+			if (bit_depth <= 24)
+				lsb = 28 - bit_depth;
+			else
+				lsb = 4;
 		}
 
-		if (bit_depth <= 24)
-			lsb = 28 - bit_depth;
-		else
-			lsb = 4;
-
 		// to ddr spdifin
 		aml_toddr_select_src(to, SPDIFIN);
-		aml_toddr_set_format(to, toddr_type, 27, lsb);
+		aml_toddr_set_format(to, toddr_type, msb, lsb);
 		aml_toddr_set_fifos(to, 0x40);
 	}
 
@@ -576,7 +595,8 @@ static struct snd_soc_dai_driver aml_spdif_dai = {
 	},
 	.capture = {
 	     .channels_min = 1,
-	     .channels_max = 2,
+		 /* spdif 2ch + tdmin_lb 8ch(fake for loopback) */
+	     .channels_max = 10,
 	     .rates = AML_DAI_SPDIF_RATES,
 	     .formats = AML_DAI_SPDIF_FORMATS,
 	},

sound/soc/amlogic/auge/tdm.c

@@ -403,13 +403,11 @@ static int aml_dai_tdm_prepare(struct snd_pcm_substream *substream,
 
 		switch (bit_depth) {
 		case 8:
+		case 16:
+		case 32:
 			toddr_type = 0;
 			break;
-		case 16:
-			toddr_type = 2;
-			break;
 		case 24:
-		case 32:
 			toddr_type = 4;
 			break;
 		default:
@@ -816,7 +814,7 @@ static struct snd_soc_dai_driver aml_tdm_dai[] = {
 	},
 	.capture = {
 	     .channels_min = 1,
-	     .channels_max = 8,
+	     .channels_max = 32,
 	     .rates = AML_DAI_TDM_RATES,
 	     .formats = AML_DAI_TDM_FORMATS,
 	},
@@ -836,7 +834,7 @@ static struct snd_soc_dai_driver aml_tdm_dai[] = {
 	},
 	.capture = {
 	     .channels_min = 1,
-	     .channels_max = 8,
+	     .channels_max = 32,
 	     .rates = AML_DAI_TDM_RATES,
 	     .formats = AML_DAI_TDM_FORMATS,
 	},

sound/soc/codecs/amlogic/pdm_dummy.c

@@ -20,7 +20,11 @@
 #include <linux/of.h>
 #include <sound/soc.h>
 
-#define DUMMY_RATES			SNDRV_PCM_RATE_8000_48000
+#define DUMMY_RATES			(SNDRV_PCM_RATE_48000 |\
+						SNDRV_PCM_RATE_32000 |\
+						SNDRV_PCM_RATE_16000 |\
+						SNDRV_PCM_RATE_8000)
+
 #define DUMMY_FORMATS		(SNDRV_PCM_FMTBIT_S16_LE |\
 						SNDRV_PCM_FMTBIT_S24_LE |\
 						SNDRV_PCM_FMTBIT_S32_LE)
@@ -59,7 +63,7 @@ struct snd_soc_dai_driver pdm_dummy_dai = {
 	.capture = {
 		.stream_name = "PDM Capture",
 		.channels_min = 1,
-		.channels_max = 8,
+		.channels_max = 16,
 		.rates = DUMMY_RATES,
 		.formats = DUMMY_FORMATS,
 	},

sound/soc/codecs/amlogic/tlv320adc3101.c

@@ -565,7 +565,7 @@ static int adc3101_mute(struct snd_soc_dai *dai, int mute)
 static int adc3101_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
-	pr_info("%s ..\n", __func__);
+	pr_debug("%s ..\n", __func__);
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* Switch on PLL */