[PATCH v7 0/7] Add mrfld DSP topology and widgets

[PATCH v7 0/7] Add mrfld DSP topology and widgets

[Subhransu S. Prusty]

This patch series adds DPCM and DAPM widgets to handle DSP topology in the
SST platform driver for merrifield.

Changes in v7:
	Handle return values for some function calls.
	Added some comment for the vb_timer command operation.
	Addressed other review comments from Mark.

Subhransu S. Prusty (1):
  ASoC: Export dapm_kcontrol_get_value

Vinod Koul (6):
  ASoC: Intel: mrfld: add the gain controls
  ASoC: Intel: mfld-pcm: add control for powering up/down dsp
  ASoC: Intel: mrfld: add DSP core controls
  ASoC: Intel: mrfld: add the DSP DAPM widgets
  ASoC: Intel: mfld-pcm: add FE and BE ops
  ASoC: Intel: mrfld: add the DSP mixers

 include/sound/soc-dapm.h                |    1 +
 sound/soc/intel/sst-atom-controls.c     | 1140 ++++++++++++++++++++++++++++++-
 sound/soc/intel/sst-atom-controls.h     |  428 ++++++++++++
 sound/soc/intel/sst-mfld-platform-pcm.c |  169 ++++-
 sound/soc/intel/sst-mfld-platform.h     |   21 +-
 sound/soc/soc-dapm.c                    |    3 +-
 6 files changed, 1723 insertions(+), 39 deletions(-)

-- 
1.9.0

[PATCH v7 1/7] ASoC: Intel: mrfld: add the gain controls

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

The DSP has various gain modules in the path,
add these as ALSA gain controls

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-atom-controls.c | 178 +++++++++++++++++++++++++++++++++++-
 sound/soc/intel/sst-atom-controls.h | 119 ++++++++++++++++++++++++
 2 files changed, 295 insertions(+), 2 deletions(-)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index 7104a34..49b54da 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -162,6 +162,167 @@ static int sst_algo_control_set(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
+static int sst_gain_ctl_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = mc->stereo ? 2 : 1;
+	uinfo->value.integer.min = mc->min;
+	uinfo->value.integer.max = mc->max;
+	return 0;
+}
+
+/**
+ * sst_send_gain_cmd - send the gain algorithm IPC to the FW
+ * @gv:		the stored value of gain (also contains rampduration)
+ * @mute:	flag that indicates whether this was called from the
+ *		digital_mute callback or directly. If called from the
+ *		digital_mute callback, module will be muted/unmuted based on this
+ *		flag. The flag is always 0 if called directly.
+ *
+ * The user-set gain value is sent only if the user-controllable 'mute' control
+ * is OFF (indicated by gv->mute). Otherwise, the mute value (MIN value) is
+ * sent.
+ */
+static int sst_send_gain_cmd(struct sst_data *drv, struct sst_gain_value *gv,
+			      u16 task_id, u16 loc_id, u16 module_id, int mute)
+{
+	struct sst_cmd_set_gain_dual cmd;
+
+	dev_dbg(&drv->pdev->dev, "Enter\n");
+
+	cmd.header.command_id = MMX_SET_GAIN;
+	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+	cmd.gain_cell_num = 1;
+
+	if (mute || gv->mute) {
+		cmd.cell_gains[0].cell_gain_left = SST_GAIN_MIN_VALUE;
+		cmd.cell_gains[0].cell_gain_right = SST_GAIN_MIN_VALUE;
+	} else {
+		cmd.cell_gains[0].cell_gain_left = gv->l_gain;
+		cmd.cell_gains[0].cell_gain_right = gv->r_gain;
+	}
+	SST_FILL_DESTINATION(2, cmd.cell_gains[0].dest,
+			     loc_id, module_id);
+	cmd.cell_gains[0].gain_time_constant = gv->ramp_duration;
+
+	cmd.header.length = sizeof(struct sst_cmd_set_gain_dual)
+				- sizeof(struct sst_dsp_header);
+
+	return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS,
+				SST_FLAG_BLOCKED, task_id, 0, &cmd,
+			      sizeof(cmd.header) + cmd.header.length);
+}
+
+static int sst_gain_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+	struct sst_gain_value *gv = mc->gain_val;
+
+	switch (mc->type) {
+	case SST_GAIN_TLV:
+		ucontrol->value.integer.value[0] = gv->l_gain;
+		ucontrol->value.integer.value[1] = gv->r_gain;
+		break;
+	case SST_GAIN_MUTE:
+		ucontrol->value.integer.value[0] = gv->mute ? 1 : 0;
+		break;
+	case SST_GAIN_RAMP_DURATION:
+		ucontrol->value.integer.value[0] = gv->ramp_duration;
+		break;
+	default:
+		dev_err(component->dev, "Invalid Input- gain type:%d\n",
+				mc->type);
+		return -EINVAL;
+	};
+	return 0;
+}
+
+static int sst_gain_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	int ret = 0;
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+	struct sst_gain_value *gv = mc->gain_val;
+
+	mutex_lock(&drv->lock);
+	switch (mc->type) {
+	case SST_GAIN_TLV:
+		gv->l_gain = ucontrol->value.integer.value[0];
+		gv->r_gain = ucontrol->value.integer.value[1];
+		dev_dbg(cmpnt->dev, "%s: Volume %d, %d\n",
+				mc->pname, gv->l_gain, gv->r_gain);
+		break;
+	case SST_GAIN_MUTE:
+		gv->mute = !!ucontrol->value.integer.value[0];
+		dev_dbg(cmpnt->dev, "%s: Mute %d\n", mc->pname, gv->mute);
+		break;
+	case SST_GAIN_RAMP_DURATION:
+		gv->ramp_duration = ucontrol->value.integer.value[0];
+		dev_dbg(cmpnt->dev, "%s: Ramp Delay%d\n",
+					mc->pname, gv->ramp_duration);
+		break;
+	default:
+		mutex_unlock(&drv->lock);
+		dev_err(cmpnt->dev, "Invalid Input- gain type:%d\n",
+				mc->type);
+		return -EINVAL;
+	};
+
+	if (mc->w && mc->w->power)
+		ret = sst_send_gain_cmd(drv, gv, mc->task_id,
+			mc->pipe_id | mc->instance_id, mc->module_id, 0);
+	mutex_unlock(&drv->lock);
+	return ret;
+}
+
+static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0);
+
+/* Gain helper with min/max set */
+#define SST_GAIN(name, path_id, task_id, instance, gain_var)				\
+	SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE,	\
+		SST_GAIN_TC_MIN, SST_GAIN_TC_MAX,					\
+		sst_gain_get, sst_gain_put,						\
+		SST_MODULE_ID_GAIN_CELL, path_id, instance, task_id,			\
+		sst_gain_tlv_common, gain_var)
+
+#define SST_VOLUME(name, path_id, task_id, instance, gain_var)				\
+	SST_GAIN_KCONTROLS(name, "Volume", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE,	\
+		SST_GAIN_TC_MIN, SST_GAIN_TC_MAX,					\
+		sst_gain_get, sst_gain_put,						\
+		SST_MODULE_ID_VOLUME, path_id, instance, task_id,			\
+		sst_gain_tlv_common, gain_var)
+
+#define SST_NUM_GAINS 36
+static struct sst_gain_value sst_gains[SST_NUM_GAINS];
+
+static const struct snd_kcontrol_new sst_gain_controls[] = {
+	SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]),
+	SST_GAIN("media1_in", SST_PATH_INDEX_MEDIA1_IN, SST_TASK_MMX, 0, &sst_gains[1]),
+	SST_GAIN("media2_in", SST_PATH_INDEX_MEDIA2_IN, SST_TASK_MMX, 0, &sst_gains[2]),
+	SST_GAIN("media3_in", SST_PATH_INDEX_MEDIA3_IN, SST_TASK_MMX, 0, &sst_gains[3]),
+
+	SST_GAIN("pcm0_in", SST_PATH_INDEX_PCM0_IN, SST_TASK_SBA, 0, &sst_gains[4]),
+	SST_GAIN("pcm1_in", SST_PATH_INDEX_PCM1_IN, SST_TASK_SBA, 0, &sst_gains[5]),
+	SST_GAIN("pcm1_out", SST_PATH_INDEX_PCM1_OUT, SST_TASK_SBA, 0, &sst_gains[7]),
+	SST_GAIN("pcm2_out", SST_PATH_INDEX_PCM2_OUT, SST_TASK_SBA, 0, &sst_gains[8]),
+
+	SST_GAIN("codec_in0", SST_PATH_INDEX_CODEC_IN0, SST_TASK_SBA, 0, &sst_gains[20]),
+	SST_GAIN("codec_in1", SST_PATH_INDEX_CODEC_IN1, SST_TASK_SBA, 0, &sst_gains[21]),
+	SST_GAIN("codec_out0", SST_PATH_INDEX_CODEC_OUT0, SST_TASK_SBA, 0, &sst_gains[22]),
+	SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[23]),
+	SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[30]),
+	SST_GAIN("media_loop2_out", SST_PATH_INDEX_MEDIA_LOOP2_OUT, SST_TASK_SBA, 0, &sst_gains[31]),
+	SST_GAIN("sprot_loop_out", SST_PATH_INDEX_SPROT_LOOP_OUT, SST_TASK_SBA, 0, &sst_gains[32]),
+	SST_VOLUME("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[33]),
+};
+
 static const struct snd_kcontrol_new sst_algo_controls[] = {
 	SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24,
 		 SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR),
@@ -200,7 +361,7 @@ static int sst_algo_control_init(struct device *dev)
 
 int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 {
-	int ret = 0;
+	int i, ret = 0;
 	struct sst_data *drv = snd_soc_platform_get_drvdata(platform);
 
 	drv->byte_stream = devm_kzalloc(platform->dev,
@@ -208,11 +369,24 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 	if (!drv->byte_stream)
 		return -ENOMEM;
 
-	/*Initialize algo control params*/
+	for (i = 0; i < SST_NUM_GAINS; i++) {
+		sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT;
+		sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT;
+		sst_gains[i].r_gain = SST_GAIN_VOLUME_DEFAULT;
+		sst_gains[i].ramp_duration = SST_GAIN_RAMP_DURATION_DEFAULT;
+	}
+
+	ret = snd_soc_add_platform_controls(platform, sst_gain_controls,
+			ARRAY_SIZE(sst_gain_controls));
+	if (ret)
+		return ret;
+
+	/* Initialize algo control params */
 	ret = sst_algo_control_init(platform->dev);
 	if (ret)
 		return ret;
 	ret = snd_soc_add_platform_controls(platform, sst_algo_controls,
 			ARRAY_SIZE(sst_algo_controls));
+
 	return ret;
 }
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index a73e894..261bc7f 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -360,16 +360,135 @@ struct sst_dsp_header {
 struct sst_cmd_generic {
 	struct sst_dsp_header header;
 } __packed;
+
+struct gain_cell {
+	struct sst_destination_id dest;
+	s16 cell_gain_left;
+	s16 cell_gain_right;
+	u16 gain_time_constant;
+} __packed;
+
+#define NUM_GAIN_CELLS 1
+struct sst_cmd_set_gain_dual {
+	struct sst_dsp_header header;
+	u16    gain_cell_num;
+	struct gain_cell cell_gains[NUM_GAIN_CELLS];
+} __packed;
 struct sst_cmd_set_params {
 	struct sst_destination_id dst;
 	u16 command_id;
 	char params[0];
 } __packed;
+/**** widget defines *****/
+
+#include <sound/soc.h>
+#include <sound/tlv.h>
+struct sst_ids {
+	u16 location_id;
+	u16 module_id;
+	u8  task_id;
+	u8  format;
+	u8  reg;
+	const char *parent_wname;
+	struct snd_soc_dapm_widget *parent_w;
+	struct list_head algo_list;
+	struct list_head gain_list;
+	const struct sst_pcm_format *pcm_fmt;
+};
+enum sst_gain_kcontrol_type {
+	SST_GAIN_TLV,
+	SST_GAIN_MUTE,
+	SST_GAIN_RAMP_DURATION,
+};
+
+struct sst_gain_mixer_control {
+	bool stereo;
+	enum sst_gain_kcontrol_type type;
+	struct sst_gain_value *gain_val;
+	int max;
+	int min;
+	u16 instance_id;
+	u16 module_id;
+	u16 pipe_id;
+	u16 task_id;
+	char pname[44];
+	struct snd_soc_dapm_widget *w;
+};
+
+struct sst_gain_value {
+	u16 ramp_duration;
+	s16 l_gain;
+	s16 r_gain;
+	bool mute;
+};
+#define SST_GAIN_VOLUME_DEFAULT		(-1440)
+#define SST_GAIN_RAMP_DURATION_DEFAULT	5 /* timeconstant */
+#define SST_GAIN_MUTE_DEFAULT		true
+
+#define SST_GAIN_KCONTROL_TLV(xname, xhandler_get, xhandler_put, \
+			      xmod, xpipe, xinstance, xtask, tlv_array, xgain_val, \
+			      xmin, xmax, xpname) \
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		  SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p = (tlv_array), \
+	.info = sst_gain_ctl_info,\
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+	{ .stereo = true, .max = xmax, .min = xmin, .type = SST_GAIN_TLV, \
+	  .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+	  .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname}
+
+#define SST_GAIN_KCONTROL_INT(xname, xhandler_get, xhandler_put, \
+			      xmod, xpipe, xinstance, xtask, xtype, xgain_val, \
+			      xmin, xmax, xpname) \
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = sst_gain_ctl_info, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+	{ .stereo = false, .max = xmax, .min = xmin, .type = xtype, \
+	  .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+	  .instance_id = xinstance, .gain_val = xgain_val, .pname =  xpname}
+
+#define SST_GAIN_KCONTROL_BOOL(xname, xhandler_get, xhandler_put,\
+			       xmod, xpipe, xinstance, xtask, xgain_val, xpname) \
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_bool_ext, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+	{ .stereo = false, .type = SST_GAIN_MUTE, \
+	  .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+	  .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname}
 #define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \
 	xpname " " xmname " " #xinstance " " xtype
 
 #define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \
 	xpname " " xmname " " #xinstance " " xtype " " xsubmodule
+
+/*
+ * 3 Controls for each Gain module
+ * e.g.	- pcm0_in Gain 0 Volume
+ *	- pcm0_in Gain 0 Ramp Delay
+ *	- pcm0_in Gain 0 Switch
+ */
+#define SST_GAIN_KCONTROLS(xpname, xmname, xmin_gain, xmax_gain, xmin_tc, xmax_tc, \
+			   xhandler_get, xhandler_put, \
+			   xmod, xpipe, xinstance, xtask, tlv_array, xgain_val) \
+	{ SST_GAIN_KCONTROL_INT(SST_CONTROL_NAME(xpname, xmname, xinstance, "Ramp Delay"), \
+		xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, SST_GAIN_RAMP_DURATION, \
+		xgain_val, xmin_tc, xmax_tc, xpname) }, \
+	{ SST_GAIN_KCONTROL_BOOL(SST_CONTROL_NAME(xpname, xmname, xinstance, "Switch"), \
+		xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, \
+		xgain_val, xpname) } ,\
+	{ SST_GAIN_KCONTROL_TLV(SST_CONTROL_NAME(xpname, xmname, xinstance, "Volume"), \
+		xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, tlv_array, \
+		xgain_val, xmin_gain, xmax_gain, xpname) }
+
+#define SST_GAIN_TC_MIN		5
+#define SST_GAIN_TC_MAX		5000
+#define SST_GAIN_MIN_VALUE	-1440 /* in 0.1 DB units */
+#define SST_GAIN_MAX_VALUE	360
+
 enum sst_algo_kcontrol_type {
 	SST_ALGO_PARAMS,
 	SST_ALGO_BYPASS,
-- 
1.9.0

Re: [PATCH v7 1/7] ASoC: Intel: mrfld: add the gain controls

[Mark Brown]

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On Fri, Sep 19, 2014 at 04:46:02PM +0530, Subhransu S. Prusty wrote:

> +/* Gain helper with min/max set */
> +#define SST_GAIN(name, path_id, task_id, instance, gain_var)				\
> +	SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE,	\
> +		SST_GAIN_TC_MIN, SST_GAIN_TC_MAX,					\

As far as I can tell from following the macros through this looks like
it will create controls called "name Gain Volume" which doesn't seem
excellent.

> +#define SST_NUM_GAINS 36
> +static struct sst_gain_value sst_gains[SST_NUM_GAINS];

> +static const struct snd_kcontrol_new sst_gain_controls[] = {
> +	SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]),

This table is full of magic number references into the array...

> +	SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[23]),
> +	SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[30]),

...with holes in the array.  This seems really error prone.

>  	char params[0];
>  } __packed;
> +/**** widget defines *****/
> +
> +#include <sound/soc.h>
> +#include <sound/tlv.h>
> +struct sst_ids {

This is adding includes in the middle of the file (which isn't the usual
kernel coding style) and is missing a blank lines.  This sort of stuff
is getting quite repetitive and hence annoying, especially the blank
lines thing which comes up again and again and just makes everything
even harder to read.

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Re: [PATCH v7 1/7] ASoC: Intel: mrfld: add the gain controls

[Vinod Koul]

[-- Attachment #1.1: Type: text/plain, Size: 1995 bytes --]

On Thu, Sep 25, 2014 at 03:20:26PM +0100, Mark Brown wrote:
> On Fri, Sep 19, 2014 at 04:46:02PM +0530, Subhransu S. Prusty wrote:
> 
> > +/* Gain helper with min/max set */
> > +#define SST_GAIN(name, path_id, task_id, instance, gain_var)				\
> > +	SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE,	\
> > +		SST_GAIN_TC_MIN, SST_GAIN_TC_MAX,					\
> 
> As far as I can tell from following the macros through this looks like
> it will create controls called "name Gain Volume" which doesn't seem
> excellent.
The convenstion we used was <pipeline identifer> and then module. So if gain
block is in media 0 input we refer to it as "media 0 in Gain Volume"
This helps to identify the module location and help in understanding control
by looking at the name
> 
> > +#define SST_NUM_GAINS 36
> > +static struct sst_gain_value sst_gains[SST_NUM_GAINS];
> 
> > +static const struct snd_kcontrol_new sst_gain_controls[] = {
> > +	SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]),
> 
> This table is full of magic number references into the array...
> 
> > +	SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[23]),
> > +	SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[30]),
> 
> ...with holes in the array.  This seems really error prone.
I will fix the holes here..

> >  	char params[0];
> >  } __packed;
> > +/**** widget defines *****/
> > +
> > +#include <sound/soc.h>
> > +#include <sound/tlv.h>
> > +struct sst_ids {
> 
> This is adding includes in the middle of the file (which isn't the usual
> kernel coding style) and is missing a blank lines.  This sort of stuff
> is getting quite repetitive and hence annoying, especially the blank
> lines thing which comes up again and again and just makes everything
> even harder to read.
Sorry about this one, i will make sure this is fixed properly and not
repeated

-- 
~Vinod

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[PATCH v7 2/7] ASoC: Intel: mfld-pcm: add control for powering up/down dsp

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

When we have PCM (FE/BE) opened or DAPM widgets triggered we need power
up/down DSP accordingly. The DSP will do ref count of these requests
i.e. link these runtime_get/put calls of DSP

Also fix some preexisting spacing error.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-mfld-platform-pcm.c | 16 ++++++++++++++++
 sound/soc/intel/sst-mfld-platform.h     | 17 +++++++++--------
 2 files changed, 25 insertions(+), 8 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 9906b7c..6f5edd6 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -322,6 +322,16 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 
 }
 
+static int power_up_sst(struct sst_runtime_stream *stream)
+{
+	return stream->ops->power(sst->dev, true);
+}
+
+static void power_down_sst(struct sst_runtime_stream *stream)
+{
+	stream->ops->power(sst->dev, false);
+}
+
 static int sst_media_open(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -351,6 +361,10 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	/* allocate memory for SST API set */
 	runtime->private_data = stream;
 
+	ret_val = power_up_sst(stream);
+	if (ret_val < 0)
+		return ret_val;
+
 	/* Make sure, that the period size is always even */
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
 			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
@@ -370,6 +384,8 @@ static void sst_media_close(struct snd_pcm_substream *substream,
 	int ret_val = 0, str_id;
 
 	stream = substream->runtime->private_data;
+	power_down_sst(stream);
+
 	str_id = stream->stream_info.str_id;
 	if (str_id)
 		ret_val = stream->ops->close(sst->dev, str_id);
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 7092ee3..19f83ec 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -120,15 +120,16 @@ struct compress_sst_ops {
 };
 
 struct sst_ops {
-	int (*open) (struct device *dev, struct snd_sst_params *str_param);
-	int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info);
-	int (*stream_start) (struct device *dev, int str_id);
-	int (*stream_drop) (struct device *dev, int str_id);
-	int (*stream_pause) (struct device *dev, int str_id);
-	int (*stream_pause_release) (struct device *dev, int str_id);
-	int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info);
+	int (*open)(struct device *dev, struct snd_sst_params *str_param);
+	int (*stream_init)(struct device *dev, struct pcm_stream_info *str_info);
+	int (*stream_start)(struct device *dev, int str_id);
+	int (*stream_drop)(struct device *dev, int str_id);
+	int (*stream_pause)(struct device *dev, int str_id);
+	int (*stream_pause_release)(struct device *dev, int str_id);
+	int (*stream_read_tstamp)(struct device *dev, struct pcm_stream_info *str_info);
 	int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes);
-	int (*close) (struct device *dev, unsigned int str_id);
+	int (*close)(struct device *dev, unsigned int str_id);
+	int (*power)(struct device *dev, bool state);
 };
 
 struct sst_runtime_stream {
-- 
1.9.0

Re: [PATCH v7 2/7] ASoC: Intel: mfld-pcm: add control for powering up/down dsp

[Mark Brown]

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On Fri, Sep 19, 2014 at 04:46:03PM +0530, Subhransu S. Prusty wrote:
> From: Vinod Koul <vinod.koul@intel.com>
> 
> When we have PCM (FE/BE) opened or DAPM widgets triggered we need power
> up/down DSP accordingly. The DSP will do ref count of these requests
> i.e. link these runtime_get/put calls of DSP

Applied, thanks.

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[PATCH v7 3/7] ASoC: Intel: mrfld: add DSP core controls

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

This patch adds core controls like interleavers, SSP BEs, and also
logic of sending pipeline and module commands to the DSP.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-atom-controls.c | 711 ++++++++++++++++++++++++++++++++++++
 sound/soc/intel/sst-atom-controls.h | 309 ++++++++++++++++
 2 files changed, 1020 insertions(+)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index 49b54da..dd830f7 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -15,6 +15,9 @@
  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  *  General Public License for more details.
  *
+ *  In the dpcm driver modelling when a particular FE/BE/Mixer/Pipe is active
+ *  we forward the settings and parameters, rest we keep the values  in
+ *  driver and forward when DAPM enables them
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
@@ -81,6 +84,156 @@ static int sst_fill_and_send_cmd(struct sst_data *drv,
 	return ret;
 }
 
+/*
+ * slot map value is a bitfield where each bit represents a FW channel
+ *
+ *			3 2 1 0		# 0 = codec0, 1 = codec1
+ *			RLRLRLRL	# 3, 4 = reserved
+ *
+ * e.g. slot 0 rx map =	00001100b -> data from slot 0 goes into codec_in1 L,R
+ */
+static u8 sst_ssp_slot_map[SST_MAX_TDM_SLOTS] = {
+	0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default rx map */
+};
+
+/*
+ * channel map value is a bitfield where each bit represents a slot
+ *
+ *			  76543210	# 0 = slot 0, 1 = slot 1
+ *
+ * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot 0, 2
+ */
+static u8 sst_ssp_channel_map[SST_MAX_TDM_SLOTS] = {
+	0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default tx map */
+};
+
+static int sst_send_slot_map(struct sst_data *drv)
+{
+	struct sst_param_sba_ssp_slot_map cmd;
+
+	dev_dbg(&drv->pdev->dev, "Enter\n");
+
+	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+	cmd.header.command_id = SBA_SET_SSP_SLOT_MAP;
+	cmd.header.length = sizeof(struct sst_param_sba_ssp_slot_map)
+				- sizeof(struct sst_dsp_header);
+
+	cmd.param_id = SBA_SET_SSP_SLOT_MAP;
+	cmd.param_len = sizeof(cmd.rx_slot_map) + sizeof(cmd.tx_slot_map)
+					+ sizeof(cmd.ssp_index);
+	cmd.ssp_index = SSP_CODEC;
+
+	memcpy(cmd.rx_slot_map, &sst_ssp_slot_map[0], sizeof(cmd.rx_slot_map));
+	memcpy(cmd.tx_slot_map, &sst_ssp_channel_map[0], sizeof(cmd.tx_slot_map));
+
+	return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS,
+			SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd,
+			      sizeof(cmd.header) + cmd.header.length);
+}
+
+int sst_slot_enum_info(struct snd_kcontrol *kcontrol,
+		       struct snd_ctl_elem_info *uinfo)
+{
+	struct sst_enum *e = (struct sst_enum *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = e->max;
+
+	if (uinfo->value.enumerated.item > e->max - 1)
+		uinfo->value.enumerated.item = e->max - 1;
+	strcpy(uinfo->value.enumerated.name,
+		e->texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+/**
+ * sst_slot_get - get the status of the interleaver/deinterleaver control
+ *
+ * Searches the map where the control status is stored, and gets the
+ * channel/slot which is currently set for this enumerated control. Since it is
+ * an enumerated control, there is only one possible value.
+ */
+static int sst_slot_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct sst_enum *e = (void *)kcontrol->private_value;
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	unsigned int ctl_no = e->reg;
+	unsigned int is_tx = e->tx;
+	unsigned int val, mux;
+	u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;
+
+	val = 1 << ctl_no;
+	/* search which slot/channel has this bit set - there should be only one */
+	for (mux = e->max; mux > 0;  mux--)
+		if (map[mux - 1] & val)
+			break;
+
+	ucontrol->value.enumerated.item[0] = mux;
+	dev_dbg(component->dev, "%s - %s map = %#x\n",
+			is_tx ? "tx channel" : "rx slot",
+			 e->texts[mux], mux ? map[mux - 1] : -1);
+	return 0;
+}
+
+/**
+ * sst_slot_put - set the status of interleaver/deinterleaver control
+ *
+ * (de)interleaver controls are defined in opposite sense to be user-friendly
+ *
+ * Instead of the enum value being the value written to the register, it is the
+ * register address; and the kcontrol number (register num) is the value written
+ * to the register. This is so that there can be only one value for each
+ * slot/channel since there is only one control for each slot/channel.
+ *
+ * This means that whenever an enum is set, we need to clear the bit
+ * for that kcontrol_no for all the interleaver OR deinterleaver registers
+ */
+static int sst_slot_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_enum *e = (void *)kcontrol->private_value;
+	int i, ret = 0;
+	unsigned int ctl_no = e->reg;
+	unsigned int is_tx = e->tx;
+	unsigned int slot_channel_no;
+	unsigned int val, mux;
+	u8 *map;
+
+	map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;
+
+	val = 1 << ctl_no;
+	mux = ucontrol->value.enumerated.item[0];
+	if (mux > e->max - 1)
+		return -EINVAL;
+
+	mutex_lock(&drv->lock);
+	/* first clear all registers of this bit */
+	for (i = 0; i < e->max; i++)
+		map[i] &= ~val;
+
+	if (mux == 0) {/* kctl set to 'none' */
+		mutex_unlock(&drv->lock);
+		return 0;
+	}
+
+	/* offset by one to take "None" into account */
+	slot_channel_no = mux - 1;
+	map[slot_channel_no] |= val;
+
+	dev_dbg(cmpnt->dev, "%s %s map = %#x\n",
+			is_tx ? "tx channel" : "rx slot",
+			e->texts[mux], map[slot_channel_no]);
+
+	if (e->w && e->w->power)
+		ret = sst_send_slot_map(drv);
+	mutex_unlock(&drv->lock);
+	return ret;
+}
+
 static int sst_send_algo_cmd(struct sst_data *drv,
 			      struct sst_algo_control *bc)
 {
@@ -104,6 +257,32 @@ static int sst_send_algo_cmd(struct sst_data *drv,
 	return ret;
 }
 
+/**
+ * sst_find_and_send_pipe_algo - send all the algo parameters for a pipe
+ *
+ * The algos which are in each pipeline are sent to the firmware one by one
+ */
+static int sst_find_and_send_pipe_algo(struct sst_data *drv,
+					const char *pipe, struct sst_ids *ids)
+{
+	int ret = 0;
+	struct sst_algo_control *bc;
+	struct sst_module *algo = NULL;
+
+	dev_dbg(&drv->pdev->dev, "Enter: widget=%s\n", pipe);
+
+	list_for_each_entry(algo, &ids->algo_list, node) {
+		bc = (void *)algo->kctl->private_value;
+
+		dev_dbg(&drv->pdev->dev, "Found algo control name=%s pipe=%s\n",
+				algo->kctl->id.name, pipe);
+		ret = sst_send_algo_cmd(drv, bc);
+		if (ret)
+			return ret;
+	}
+	return ret;
+}
+
 static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol,
 			    struct snd_ctl_elem_info *uinfo)
 {
@@ -282,8 +461,313 @@ static int sst_gain_put(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
+static int sst_set_pipe_gain(struct sst_ids *ids,
+				struct sst_data *drv, int mute);
+
+static int sst_send_pipe_module_params(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol)
+{
+	int ret = 0;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_ids *ids = w->priv;
+
+	ret = sst_find_and_send_pipe_algo(drv, w->name, ids);
+	if (ret)
+		return ret;
+	return sst_set_pipe_gain(ids, drv, 0);
+}
+
+static int sst_generic_modules_event(struct snd_soc_dapm_widget *w,
+				     struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		return sst_send_pipe_module_params(w, k);
+	return 0;
+}
+
 static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0);
 
+/* Look up table to convert MIXER SW bit regs to SWM inputs */
+static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = {
+	[SST_IP_CODEC0]		= SST_SWM_IN_CODEC0,
+	[SST_IP_CODEC1]		= SST_SWM_IN_CODEC1,
+	[SST_IP_LOOP0]		= SST_SWM_IN_SPROT_LOOP,
+	[SST_IP_LOOP1]		= SST_SWM_IN_MEDIA_LOOP1,
+	[SST_IP_LOOP2]		= SST_SWM_IN_MEDIA_LOOP2,
+	[SST_IP_PCM0]		= SST_SWM_IN_PCM0,
+	[SST_IP_PCM1]		= SST_SWM_IN_PCM1,
+	[SST_IP_MEDIA0]		= SST_SWM_IN_MEDIA0,
+	[SST_IP_MEDIA1]		= SST_SWM_IN_MEDIA1,
+	[SST_IP_MEDIA2]		= SST_SWM_IN_MEDIA2,
+	[SST_IP_MEDIA3]		= SST_SWM_IN_MEDIA3,
+};
+
+static int sst_set_pipe_gain(struct sst_ids *ids,
+			struct sst_data *drv, int mute)
+{
+	int ret = 0;
+	struct sst_gain_mixer_control *mc;
+	struct sst_gain_value *gv;
+	struct sst_module *gain = NULL;
+
+	list_for_each_entry(gain, &ids->gain_list, node) {
+		struct snd_kcontrol *kctl = gain->kctl;
+
+		dev_dbg(&drv->pdev->dev, "control name=%s\n", kctl->id.name);
+		mc = (void *)kctl->private_value;
+		gv = mc->gain_val;
+
+		ret = sst_send_gain_cmd(drv, gv, mc->task_id,
+			mc->pipe_id | mc->instance_id, mc->module_id, mute);
+		if (ret)
+			return ret;
+	}
+	return ret;
+}
+
+/*
+ * sst_handle_vb_timer - Start/Stop the DSP scheduler
+ *
+ * The DSP expects first cmd to be SBA_VB_START, so at first startup send
+ * that.
+ * DSP expects last cmd to be SBA_VB_IDLE, so at last shutdown send that.
+ *
+ * Do refcount internally so that we send command only at first start
+ * and last end. Since SST driver does its own ref count, invoke sst's
+ * power ops always!
+ */
+int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable)
+{
+	int ret = 0;
+	struct sst_cmd_generic cmd;
+	struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+	static int timer_usage;
+
+	if (enable)
+		cmd.header.command_id = SBA_VB_START;
+	else
+		cmd.header.command_id = SBA_IDLE;
+	dev_dbg(dai->dev, "enable=%u, usage=%d\n", enable, timer_usage);
+
+	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+	cmd.header.length = 0;
+
+	if (enable) {
+		ret = sst->ops->power(sst->dev, true);
+		if (ret < 0)
+			return ret;
+	}
+
+	mutex_lock(&drv->lock);
+	if (enable)
+		timer_usage++;
+	else
+		timer_usage--;
+
+	/*
+	 * Send the command only if this call is the first enable or last
+	 * disable
+	 */
+	if ((enable && (timer_usage == 1)) ||
+	    (!enable && (timer_usage == 0))) {
+		ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_CMD,
+				SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd,
+				sizeof(cmd.header) + cmd.header.length);
+		if (ret && enable) {
+			timer_usage--;
+			enable  = false;
+		}
+	}
+	mutex_unlock(&drv->lock);
+
+	if (!enable)
+		sst->ops->power(sst->dev, false);
+	return ret;
+}
+
+/**
+ * sst_ssp_config - contains SSP configuration for media UC
+ */
+static const struct sst_ssp_config sst_ssp_configs = {
+	.ssp_id = SSP_CODEC,
+	.bits_per_slot = 24,
+	.slots = 4,
+	.ssp_mode = SSP_MODE_MASTER,
+	.pcm_mode = SSP_PCM_MODE_NETWORK,
+	.duplex = SSP_DUPLEX,
+	.ssp_protocol = SSP_MODE_PCM,
+	.fs_width = 1,
+	.fs_frequency = SSP_FS_48_KHZ,
+	.active_slot_map = 0xF,
+	.start_delay = 0,
+};
+
+int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable)
+{
+	struct sst_cmd_sba_hw_set_ssp cmd;
+	struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+	const struct sst_ssp_config *config;
+
+	dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id);
+
+	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+	cmd.header.command_id = SBA_HW_SET_SSP;
+	cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp)
+				- sizeof(struct sst_dsp_header);
+
+	config = &sst_ssp_configs;
+	dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id);
+
+	if (enable)
+		cmd.switch_state = SST_SWITCH_ON;
+	else
+		cmd.switch_state = SST_SWITCH_OFF;
+
+	cmd.selection = config->ssp_id;
+	cmd.nb_bits_per_slots = config->bits_per_slot;
+	cmd.nb_slots = config->slots;
+	cmd.mode = config->ssp_mode | (config->pcm_mode << 1);
+	cmd.duplex = config->duplex;
+	cmd.active_tx_slot_map = config->active_slot_map;
+	cmd.active_rx_slot_map = config->active_slot_map;
+	cmd.frame_sync_frequency = config->fs_frequency;
+	cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH;
+	cmd.data_polarity = 1;
+	cmd.frame_sync_width = config->fs_width;
+	cmd.ssp_protocol = config->ssp_protocol;
+	cmd.start_delay = config->start_delay;
+	cmd.reserved1 = cmd.reserved2 = 0xFF;
+
+	return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+				SST_TASK_SBA, 0, &cmd,
+				sizeof(cmd.header) + cmd.header.length);
+}
+
+static int sst_set_be_modules(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *k, int event)
+{
+	int ret = 0;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+
+	dev_dbg(cmpnt->dev, "Enter: widget=%s\n", w->name);
+
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = sst_send_slot_map(drv);
+		if (ret)
+			return ret;
+		ret = sst_send_pipe_module_params(w, k);
+	}
+	return ret;
+}
+
+static int sst_set_media_path(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *k, int event)
+{
+	int ret = 0;
+	struct sst_cmd_set_media_path cmd;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_ids *ids = w->priv;
+
+	dev_dbg(cmpnt->dev, "widget=%s\n", w->name);
+	dev_dbg(cmpnt->dev, "task=%u, location=%#x\n",
+				ids->task_id, ids->location_id);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		cmd.switch_state = SST_PATH_ON;
+	else
+		cmd.switch_state = SST_PATH_OFF;
+
+	SST_FILL_DESTINATION(2, cmd.header.dst,
+			     ids->location_id, SST_DEFAULT_MODULE_ID);
+
+	/* MMX_SET_MEDIA_PATH == SBA_SET_MEDIA_PATH */
+	cmd.header.command_id = MMX_SET_MEDIA_PATH;
+	cmd.header.length = sizeof(struct sst_cmd_set_media_path)
+				- sizeof(struct sst_dsp_header);
+
+	ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+			      ids->task_id, 0, &cmd,
+			      sizeof(cmd.header) + cmd.header.length);
+	if (ret)
+		return ret;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ret = sst_send_pipe_module_params(w, k);
+	return ret;
+}
+
+static int sst_set_media_loop(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	int ret = 0;
+	struct sst_cmd_sba_set_media_loop_map cmd;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_ids *ids = w->priv;
+
+	dev_dbg(cmpnt->dev, "Enter:widget=%s\n", w->name);
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		cmd.switch_state = SST_SWITCH_ON;
+	else
+		cmd.switch_state = SST_SWITCH_OFF;
+
+	SST_FILL_DESTINATION(2, cmd.header.dst,
+			     ids->location_id, SST_DEFAULT_MODULE_ID);
+
+	cmd.header.command_id = SBA_SET_MEDIA_LOOP_MAP;
+	cmd.header.length = sizeof(struct sst_cmd_sba_set_media_loop_map)
+				 - sizeof(struct sst_dsp_header);
+	cmd.param.part.cfg.rate = 2; /* 48khz */
+
+	cmd.param.part.cfg.format = ids->format; /* stereo/Mono */
+	cmd.param.part.cfg.s_length = 1; /* 24bit left justified */
+	cmd.map = 0; /* Algo sequence: Gain - DRP - FIR - IIR */
+
+	ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+			      SST_TASK_SBA, 0, &cmd,
+			      sizeof(cmd.header) + cmd.header.length);
+	if (ret)
+		return ret;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ret = sst_send_pipe_module_params(w, k);
+	return ret;
+}
+
+static const char * const slot_names[] = {
+	"none",
+	"slot 0", "slot 1", "slot 2", "slot 3",
+	"slot 4", "slot 5", "slot 6", "slot 7", /* not supported by FW */
+};
+
+static const char * const channel_names[] = {
+	"none",
+	"codec_out0_0", "codec_out0_1", "codec_out1_0", "codec_out1_1",
+	"codec_out2_0", "codec_out2_1", "codec_out3_0", "codec_out3_1", /* not supported by FW */
+};
+
+#define SST_INTERLEAVER(xpname, slot_name, slotno) \
+	SST_SSP_SLOT_CTL(xpname, "interleaver", slot_name, slotno, true, \
+			 channel_names, sst_slot_get, sst_slot_put)
+
+#define SST_DEINTERLEAVER(xpname, channel_name, channel_no) \
+	SST_SSP_SLOT_CTL(xpname, "deinterleaver", channel_name, channel_no, false, \
+			 slot_names, sst_slot_get, sst_slot_put)
+
+static const struct snd_kcontrol_new sst_slot_controls[] = {
+	SST_INTERLEAVER("codec_out", "slot 0", 0),
+	SST_INTERLEAVER("codec_out", "slot 1", 1),
+	SST_INTERLEAVER("codec_out", "slot 2", 2),
+	SST_INTERLEAVER("codec_out", "slot 3", 3),
+	SST_DEINTERLEAVER("codec_in", "codec_in0_0", 0),
+	SST_DEINTERLEAVER("codec_in", "codec_in0_1", 1),
+	SST_DEINTERLEAVER("codec_in", "codec_in1_0", 2),
+	SST_DEINTERLEAVER("codec_in", "codec_in1_1", 3),
+};
+
 /* Gain helper with min/max set */
 #define SST_GAIN(name, path_id, task_id, instance, gain_var)				\
 	SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE,	\
@@ -359,6 +843,224 @@ static int sst_algo_control_init(struct device *dev)
 	return 0;
 }
 
+static bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w)
+{
+	switch (w->id) {
+	case snd_soc_dapm_pga:
+	case snd_soc_dapm_aif_in:
+	case snd_soc_dapm_aif_out:
+	case snd_soc_dapm_input:
+	case snd_soc_dapm_output:
+	case snd_soc_dapm_mixer:
+		return true;
+	default:
+		return false;
+	}
+}
+
+/**
+ * sst_send_pipe_gains - send gains for the front-end DAIs
+ *
+ * The gains in the pipes connected to the front-ends are muted/unmuted
+ * automatically via the digital_mute() DAPM callback. This function sends the
+ * gains for the front-end pipes.
+ */
+int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute)
+{
+	struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_path *p = NULL;
+
+	dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream);
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dev_dbg(dai->dev, "Stream name=%s\n",
+				dai->playback_widget->name);
+		w = dai->playback_widget;
+		list_for_each_entry(p, &w->sinks, list_source) {
+			if (p->connected && !p->connected(w, p->sink))
+				continue;
+
+			if (p->connect && p->sink->power &&
+					is_sst_dapm_widget(p->sink)) {
+				struct sst_ids *ids = p->sink->priv;
+
+				dev_dbg(dai->dev, "send gains for widget=%s\n",
+						p->sink->name);
+				sst_set_pipe_gain(ids, drv, mute);
+			}
+		}
+	} else {
+		dev_dbg(dai->dev, "Stream name=%s\n",
+				dai->capture_widget->name);
+		w = dai->capture_widget;
+		list_for_each_entry(p, &w->sources, list_sink) {
+			if (p->connected && !p->connected(w, p->sink))
+				continue;
+
+			if (p->connect &&  p->source->power &&
+					is_sst_dapm_widget(p->source)) {
+				struct sst_ids *ids = p->source->priv;
+
+				dev_dbg(dai->dev, "send gain for widget=%s\n",
+						p->source->name);
+				sst_set_pipe_gain(ids, drv, mute);
+			}
+		}
+	}
+	return 0;
+}
+
+/**
+ * sst_fill_module_list - populate the list of modules/gains for a pipe
+ *
+ *
+ * Fills the widget pointer in the kcontrol private data, and also fills the
+ * kcontrol pointer in the widget private data.
+ *
+ * Widget pointer is used to send the algo/gain in the .put() handler if the
+ * widget is powerd on.
+ *
+ * Kcontrol pointer is used to send the algo/gain in the widget power ON/OFF
+ * event handler. Each widget (pipe) has multiple algos stored in the algo_list.
+ */
+static int sst_fill_module_list(struct snd_kcontrol *kctl,
+	 struct snd_soc_dapm_widget *w, int type)
+{
+	struct sst_module *module = NULL;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_ids *ids = w->priv;
+
+	module = devm_kzalloc(cmpnt->dev, sizeof(*module), GFP_KERNEL);
+	if (!module)
+		return -ENOMEM;
+
+	if (type == SST_MODULE_GAIN) {
+		struct sst_gain_mixer_control *mc = (void *)kctl->private_value;
+
+		mc->w = w;
+		module->kctl = kctl;
+		list_add_tail(&module->node, &ids->gain_list);
+	} else if (type == SST_MODULE_ALGO) {
+		struct sst_algo_control *bc = (void *)kctl->private_value;
+
+		bc->w = w;
+		module->kctl = kctl;
+		list_add_tail(&module->node, &ids->algo_list);
+	}
+
+	return 0;
+}
+
+/**
+ * sst_fill_widget_module_info - fill list of gains/algos for the pipe
+ * @widget:	pipe modelled as a DAPM widget
+ *
+ * Fill the list of gains/algos for the widget by looking at all the card
+ * controls and comparing the name of the widget with the first part of control
+ * name. First part of control name contains the pipe name (widget name).
+ */
+static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w,
+	struct snd_soc_platform *platform)
+{
+	struct snd_kcontrol *kctl;
+	int index, ret = 0;
+	struct snd_card *card = platform->component.card->snd_card;
+	char *idx;
+
+	down_read(&card->controls_rwsem);
+
+	list_for_each_entry(kctl, &card->controls, list) {
+		idx = strstr(kctl->id.name, " ");
+		if (idx == NULL)
+			continue;
+		index  = strlen(kctl->id.name) - strlen(idx);
+
+		if (strstr(kctl->id.name, "Volume") &&
+		    !strncmp(kctl->id.name, w->name, index))
+			ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN);
+
+		else if (strstr(kctl->id.name, "params") &&
+			 !strncmp(kctl->id.name, w->name, index))
+			ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO);
+
+		else if (strstr(kctl->id.name, "Switch") &&
+			 !strncmp(kctl->id.name, w->name, index)) {
+			struct sst_gain_mixer_control *mc =
+						(void *)kctl->private_value;
+
+			mc->w = w;
+
+		} else if (strstr(kctl->id.name, "interleaver") &&
+			 !strncmp(kctl->id.name, w->name, index)) {
+			struct sst_enum *e = (void *)kctl->private_value;
+
+			e->w = w;
+
+		} else if (strstr(kctl->id.name, "deinterleaver") &&
+			 !strncmp(kctl->id.name, w->name, index)) {
+
+			struct sst_enum *e = (void *)kctl->private_value;
+
+			e->w = w;
+		}
+
+		if (ret < 0) {
+			up_read(&card->controls_rwsem);
+			return ret;
+		}
+	}
+
+	up_read(&card->controls_rwsem);
+	return 0;
+}
+
+/**
+ * sst_fill_linked_widgets - fill the parent pointer for the linked widget
+ */
+static void sst_fill_linked_widgets(struct snd_soc_platform *platform,
+						struct sst_ids *ids)
+{
+	struct snd_soc_dapm_widget *w;
+	unsigned int len = strlen(ids->parent_wname);
+
+	list_for_each_entry(w, &platform->component.card->widgets, list) {
+		if (!strncmp(ids->parent_wname, w->name, len)) {
+			ids->parent_w = w;
+			break;
+		}
+	}
+}
+
+/**
+ * sst_map_modules_to_pipe - fill algo/gains list for all pipes
+ */
+static int sst_map_modules_to_pipe(struct snd_soc_platform *platform)
+{
+	struct snd_soc_dapm_widget *w;
+	int ret = 0;
+
+	list_for_each_entry(w, &platform->component.card->widgets, list) {
+		if (platform && is_sst_dapm_widget(w) && (w->priv)) {
+			struct sst_ids *ids = w->priv;
+
+			dev_dbg(platform->dev, "widget type=%d name=%s\n",
+					w->id, w->name);
+			INIT_LIST_HEAD(&ids->algo_list);
+			INIT_LIST_HEAD(&ids->gain_list);
+			ret = sst_fill_widget_module_info(w, platform);
+
+			if (ret < 0)
+				return ret;
+
+			/* fill linked widgets */
+			if (ids->parent_wname !=  NULL)
+				sst_fill_linked_widgets(platform, ids);
+		}
+	}
+	return 0;
+}
+
 int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 {
 	int i, ret = 0;
@@ -387,6 +1089,15 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 		return ret;
 	ret = snd_soc_add_platform_controls(platform, sst_algo_controls,
 			ARRAY_SIZE(sst_algo_controls));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_add_platform_controls(platform, sst_slot_controls,
+			ARRAY_SIZE(sst_slot_controls));
+	if (ret)
+		return ret;
+
+	ret = sst_map_modules_to_pipe(platform);
 
 	return ret;
 }
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index 261bc7f..40f249c 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -361,6 +361,38 @@ struct sst_cmd_generic {
 	struct sst_dsp_header header;
 } __packed;
 
+struct swm_input_ids {
+	struct sst_destination_id input_id;
+} __packed;
+
+struct sst_cmd_set_swm {
+	struct sst_dsp_header header;
+	struct sst_destination_id output_id;
+	u16    switch_state;
+	u16    nb_inputs;
+	struct swm_input_ids input[SST_CMD_SWM_MAX_INPUTS];
+} __packed;
+
+struct sst_cmd_set_media_path {
+	struct sst_dsp_header header;
+	u16    switch_state;
+} __packed;
+
+struct pcm_cfg {
+		u8 s_length:2;
+		u8 rate:3;
+		u8 format:3;
+} __packed;
+
+struct sst_cmd_set_speech_path {
+	struct sst_dsp_header header;
+	u16    switch_state;
+	struct {
+		u16 rsvd:8;
+		struct pcm_cfg cfg;
+	} config;
+} __packed;
+
 struct gain_cell {
 	struct sst_destination_id dest;
 	s16 cell_gain_left;
@@ -379,10 +411,166 @@ struct sst_cmd_set_params {
 	u16 command_id;
 	char params[0];
 } __packed;
+
+
+struct sst_cmd_sba_vb_start {
+	struct sst_dsp_header header;
+} __packed;
+
+union sba_media_loop_params {
+	struct {
+		u16 rsvd:8;
+		struct pcm_cfg cfg;
+	} part;
+	u16 full;
+} __packed;
+
+struct sst_cmd_sba_set_media_loop_map {
+	struct	sst_dsp_header header;
+	u16	switch_state;
+	union	sba_media_loop_params param;
+	u16	map;
+} __packed;
+
+struct sst_cmd_tone_stop {
+	struct	sst_dsp_header header;
+	u16	switch_state;
+} __packed;
+
+enum sst_ssp_mode {
+	SSP_MODE_MASTER = 0,
+	SSP_MODE_SLAVE = 1,
+};
+
+enum sst_ssp_pcm_mode {
+	SSP_PCM_MODE_NORMAL = 0,
+	SSP_PCM_MODE_NETWORK = 1,
+};
+
+enum sst_ssp_duplex {
+	SSP_DUPLEX = 0,
+	SSP_RX = 1,
+	SSP_TX = 2,
+};
+
+enum sst_ssp_fs_frequency {
+	SSP_FS_8_KHZ = 0,
+	SSP_FS_16_KHZ = 1,
+	SSP_FS_44_1_KHZ = 2,
+	SSP_FS_48_KHZ = 3,
+};
+
+enum sst_ssp_fs_polarity {
+	SSP_FS_ACTIVE_LOW = 0,
+	SSP_FS_ACTIVE_HIGH = 1,
+};
+
+enum sst_ssp_protocol {
+	SSP_MODE_PCM = 0,
+	SSP_MODE_I2S = 1,
+};
+
+enum sst_ssp_port_id {
+	SSP_MODEM = 0,
+	SSP_BT = 1,
+	SSP_FM = 2,
+	SSP_CODEC = 3,
+};
+
+struct sst_cmd_sba_hw_set_ssp {
+	struct sst_dsp_header header;
+	u16 selection;			/* 0:SSP0(def), 1:SSP1, 2:SSP2 */
+
+	u16 switch_state;
+
+	u16 nb_bits_per_slots:6;        /* 0-32 bits, 24 (def) */
+	u16 nb_slots:4;			/* 0-8: slots per frame  */
+	u16 mode:3;			/* 0:Master, 1: Slave  */
+	u16 duplex:3;
+
+	u16 active_tx_slot_map:8;       /* Bit map, 0:off, 1:on */
+	u16 reserved1:8;
+
+	u16 active_rx_slot_map:8;       /* Bit map 0: Off, 1:On */
+	u16 reserved2:8;
+
+	u16 frame_sync_frequency;
+
+	u16 frame_sync_polarity:8;
+	u16 data_polarity:8;
+
+	u16 frame_sync_width;           /* 1 to N clocks */
+	u16 ssp_protocol:8;
+	u16 start_delay:8;		/* Start delay in terms of clock ticks */
+} __packed;
+
+#define SST_MAX_TDM_SLOTS 8
+
+struct sst_param_sba_ssp_slot_map {
+	struct sst_dsp_header header;
+
+	u16 param_id;
+	u16 param_len;
+	u16 ssp_index;
+
+	u8 rx_slot_map[SST_MAX_TDM_SLOTS];
+	u8 tx_slot_map[SST_MAX_TDM_SLOTS];
+} __packed;
+
+enum {
+	SST_PROBE_EXTRACTOR = 0,
+	SST_PROBE_INJECTOR = 1,
+};
+
 /**** widget defines *****/
 
 #include <sound/soc.h>
 #include <sound/tlv.h>
+
+#define SST_MODULE_GAIN 1
+#define SST_MODULE_ALGO 2
+
+#define SST_FMT_MONO 0
+#define SST_FMT_STEREO 3
+
+/* physical SSP numbers */
+enum {
+	SST_SSP0 = 0,
+	SST_SSP1,
+	SST_SSP2,
+	SST_SSP_LAST = SST_SSP2,
+};
+
+#define SST_NUM_SSPS		(SST_SSP_LAST + 1)	/* physical SSPs */
+#define SST_MAX_SSP_MUX		2			/* single SSP muxed between pipes */
+#define SST_MAX_SSP_DOMAINS	2			/* domains present in each pipe */
+
+struct sst_module {
+	struct snd_kcontrol *kctl;
+	struct list_head node;
+};
+
+struct sst_ssp_config {
+	u8 ssp_id;
+	u8 bits_per_slot;
+	u8 slots;
+	u8 ssp_mode;
+	u8 pcm_mode;
+	u8 duplex;
+	u8 ssp_protocol;
+	u8 fs_frequency;
+	u8 active_slot_map;
+	u8 start_delay;
+	u16 fs_width;
+};
+
+struct sst_ssp_cfg {
+	const u8 ssp_number;
+	const int *mux_shift;
+	const int (*domain_shift)[SST_MAX_SSP_MUX];
+	const struct sst_ssp_config (*ssp_config)[SST_MAX_SSP_MUX][SST_MAX_SSP_DOMAINS];
+};
+
 struct sst_ids {
 	u16 location_id;
 	u16 module_id;
@@ -395,6 +583,102 @@ struct sst_ids {
 	struct list_head gain_list;
 	const struct sst_pcm_format *pcm_fmt;
 };
+
+
+#define SST_AIF_IN(wname, wevent)							\
+{	.id = snd_soc_dapm_aif_in, .name = wname, .sname = NULL,			\
+	.reg = SND_SOC_NOPM, .shift = 0,					\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD,	\
+	.priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 }		\
+}
+
+#define SST_AIF_OUT(wname, wevent)							\
+{	.id = snd_soc_dapm_aif_out, .name = wname, .sname = NULL,			\
+	.reg = SND_SOC_NOPM, .shift = 0,						\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD,	\
+	.priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 }		\
+}
+
+#define SST_INPUT(wname, wevent)							\
+{	.id = snd_soc_dapm_input, .name = wname, .sname = NULL,				\
+	.reg = SND_SOC_NOPM, .shift = 0,						\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD,	\
+	.priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 }		\
+}
+
+#define SST_OUTPUT(wname, wevent)							\
+{	.id = snd_soc_dapm_output, .name = wname, .sname = NULL,			\
+	.reg = SND_SOC_NOPM, .shift = 0,						\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD,	\
+	.priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 }		\
+}
+
+#define SST_DAPM_OUTPUT(wname, wloc_id, wtask_id, wformat, wevent)                      \
+{	.id = snd_soc_dapm_output, .name = wname, .sname = NULL,                        \
+	.reg = SND_SOC_NOPM, .shift = 0,						\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD,   \
+	.priv = (void *)&(struct sst_ids) { .location_id = wloc_id, .task_id = wtask_id,\
+						.pcm_fmt = wformat, }			\
+}
+
+#define SST_PATH(wname, wtask, wloc_id, wevent, wflags)					\
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0,		\
+	.kcontrol_news = NULL, .num_kcontrols = 0,				\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = wflags,						\
+	.priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, }	\
+}
+
+#define SST_LINKED_PATH(wname, wtask, wloc_id, linked_wname, wevent, wflags)		\
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0,		\
+	.kcontrol_news = NULL, .num_kcontrols = 0,				\
+	.on_val = 1, .off_val = 0,							\
+	.event = wevent, .event_flags = wflags,						\
+	.priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id,	\
+					.parent_wname = linked_wname}			\
+}
+
+#define SST_PATH_MEDIA_LOOP(wname, wtask, wloc_id, wformat, wevent, wflags)             \
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0,         \
+	.kcontrol_news = NULL, .num_kcontrols = 0,                         \
+	.event = wevent, .event_flags = wflags,                                         \
+	.priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id,	\
+					    .format = wformat,}				\
+}
+
+/* output is triggered before input */
+#define SST_PATH_INPUT(name, task_id, loc_id, event)					\
+	SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
+
+#define SST_PATH_LINKED_INPUT(name, task_id, loc_id, linked_wname, event)		\
+	SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event,			\
+					SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
+
+#define SST_PATH_OUTPUT(name, task_id, loc_id, event)					\
+	SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+#define SST_PATH_LINKED_OUTPUT(name, task_id, loc_id, linked_wname, event)		\
+	SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event,			\
+					SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+#define SST_PATH_MEDIA_LOOP_OUTPUT(name, task_id, loc_id, format, event)		\
+	SST_PATH_MEDIA_LOOP(name, task_id, loc_id, format, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+
+#define SST_SWM_MIXER(wname, wreg, wtask, wloc_id, wcontrols, wevent)			\
+{	.id = snd_soc_dapm_mixer, .name = wname, .reg = SND_SOC_NOPM, .shift = 0,	\
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols),\
+	.event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD |	\
+					SND_SOC_DAPM_POST_REG,				\
+	.priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id,	\
+					    .reg = wreg }				\
+}
+
 enum sst_gain_kcontrol_type {
 	SST_GAIN_TLV,
 	SST_GAIN_MUTE,
@@ -558,4 +842,29 @@ struct sst_enum {
 	struct snd_soc_dapm_widget *w;
 };
 
+/* only 4 slots/channels supported atm */
+#define SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts) \
+	(struct sst_enum){ .reg = s_ch_no, .tx = is_tx, .max = 4+1, .texts = xtexts, }
+
+#define SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name) \
+	xpname " " xmname " " s_ch_name
+
+#define SST_SSP_SLOT_CTL(xpname, xmname, s_ch_name, s_ch_no, is_tx, xtexts, xget, xput) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+	.name = SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name), \
+	.info = sst_slot_enum_info, \
+	.get = xget, .put = xput, \
+	.private_value = (unsigned long)&SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts), \
+}
+
+#define SST_MUX_CTL_NAME(xpname, xinstance) \
+	xpname " " #xinstance
+
+#define SST_SSP_MUX_ENUM(xreg, xshift, xtexts) \
+	(struct soc_enum) SOC_ENUM_DOUBLE(xreg, xshift, xshift, ARRAY_SIZE(xtexts), xtexts)
+
+#define SST_SSP_MUX_CTL(xpname, xinstance, xreg, xshift, xtexts) \
+	SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \
+			  SST_SSP_MUX_ENUM(xreg, xshift, xtexts))
+
 #endif
-- 
1.9.0

Re: [PATCH v7 3/7] ASoC: Intel: mrfld: add DSP core controls

[Mark Brown]

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On Fri, Sep 19, 2014 at 04:46:04PM +0530, Subhransu S. Prusty wrote:

> +static int sst_slot_get(struct snd_kcontrol *kcontrol,
> +                       struct snd_ctl_elem_value *ucontrol)
> +{
> +       struct sst_enum *e = (void *)kcontrol->private_value;
> +       struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
> +       unsigned int ctl_no = e->reg;
> +       unsigned int is_tx = e->tx;
> +       unsigned int val, mux;
> +       u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;

So the "channel" map is for transmit and the "slot" map is for receive?
That naming doesn't seem at all obvious, I'd expect some confusion and
resulting bugs there.  

This is a really big block of code and there's lots of things like this
in the code which may well *work* but don't seem robust - not flagging
up easy to detect errors for example - and I've been spotting things
like locking trouble as well.  This is all setting off alarm bells which
is worrying.

> +	mutex_lock(&drv->lock);
> +	/* first clear all registers of this bit */
> +	for (i = 0; i < e->max; i++)
> +		map[i] &= ~val;
> +
> +	if (mux == 0) {/* kctl set to 'none' */
> +		mutex_unlock(&drv->lock);
> +		return 0;
> +	}

It's still not at all clear to me why we don't need to interact with the
hardware if we're settinng this to zero.  AFAICT from the previous
discussion this comment at the top of the file:

> + *  In the dpcm driver modelling when a particular FE/BE/Mixer/Pipe is active
> + *  we forward the settings and parameters, rest we keep the values  in
> + *  driver and forward when DAPM enables them

is supposed to explain what's going on but since it's nowhere near the
code it's unlikely that people will have seen it and it's not terribly
easy to relate to the code here.  As far as I can tell the theory is
that something is going to trigger a power down of the DSP and it won't
matter but if this is the case it isn't at all clear from the immediate
code.

> +	if (type == SST_MODULE_GAIN) {
> +		struct sst_gain_mixer_control *mc = (void *)kctl->private_value;
> +
> +		mc->w = w;
> +		module->kctl = kctl;
> +		list_add_tail(&module->node, &ids->gain_list);
> +	} else if (type == SST_MODULE_ALGO) {
> +		struct sst_algo_control *bc = (void *)kctl->private_value;
> +
> +		bc->w = w;
> +		module->kctl = kctl;
> +		list_add_tail(&module->node, &ids->algo_list);
> +	}

This looks like it should be a switch statement with a default case to
trap any errors.

> +		if (idx == NULL)
> +			continue;
> +		index  = strlen(kctl->id.name) - strlen(idx);

I keep on seeing lots of code with random double instead of single
spaces.

> +		if (strstr(kctl->id.name, "Volume") &&
> +		    !strncmp(kctl->id.name, w->name, index))
> +			ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN);
> +
> +		else if (strstr(kctl->id.name, "params") &&
> +			 !strncmp(kctl->id.name, w->name, index))
> +			ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO);
> +
> +		else if (strstr(kctl->id.name, "Switch") &&
> +			 !strncmp(kctl->id.name, w->name, index)) {
> +			struct sst_gain_mixer_control *mc =
> +						(void *)kctl->private_value;
> +
> +			mc->w = w;
> +
> +		} else if (strstr(kctl->id.name, "interleaver") &&

Both or no branches of an if statement should use { }.

> +			 !strncmp(kctl->id.name, w->name, index)) {
> +			struct sst_enum *e = (void *)kctl->private_value;
> +
> +			e->w = w;
> +
> +		} else if (strstr(kctl->id.name, "deinterleaver") &&
> +			 !strncmp(kctl->id.name, w->name, index)) {
> +
> +			struct sst_enum *e = (void *)kctl->private_value;
> +
> +			e->w = w;
> +		}

Again no fallthrough case.

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Re: [PATCH v7 3/7] ASoC: Intel: mrfld: add DSP core controls

[Vinod Koul]

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On Thu, Sep 25, 2014 at 04:08:13PM +0100, Mark Brown wrote:
> On Fri, Sep 19, 2014 at 04:46:04PM +0530, Subhransu S. Prusty wrote:
> 
> > +static int sst_slot_get(struct snd_kcontrol *kcontrol,
> > +                       struct snd_ctl_elem_value *ucontrol)
> > +{
> > +       struct sst_enum *e = (void *)kcontrol->private_value;
> > +       struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
> > +       unsigned int ctl_no = e->reg;
> > +       unsigned int is_tx = e->tx;
> > +       unsigned int val, mux;
> > +       u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;
> 
> So the "channel" map is for transmit and the "slot" map is for receive?
> That naming doesn't seem at all obvious, I'd expect some confusion and
> resulting bugs there.  
yes as multiple channels get interleaved to slot while transmitting and on
recive side various lots get de-interleaved to channels :)
These controls allow you to route stuff.

> This is a really big block of code and there's lots of things like this
> in the code which may well *work* but don't seem robust - not flagging
> up easy to detect errors for example - and I've been spotting things
> like locking trouble as well.  This is all setting off alarm bells which
> is worrying.
Ok will recheck this part and fix wherever required.

> > +	mutex_lock(&drv->lock);
> > +	/* first clear all registers of this bit */
> > +	for (i = 0; i < e->max; i++)
> > +		map[i] &= ~val;
> > +
> > +	if (mux == 0) {/* kctl set to 'none' */
> > +		mutex_unlock(&drv->lock);
> > +		return 0;
> > +	}
> 
> It's still not at all clear to me why we don't need to interact with the
> hardware if we're settinng this to zero.  AFAICT from the previous
> discussion this comment at the top of the file:
> 
> > + *  In the dpcm driver modelling when a particular FE/BE/Mixer/Pipe is active
> > + *  we forward the settings and parameters, rest we keep the values  in
> > + *  driver and forward when DAPM enables them
> 
> is supposed to explain what's going on but since it's nowhere near the
> code it's unlikely that people will have seen it and it's not terribly
> easy to relate to the code here.  As far as I can tell the theory is
> that something is going to trigger a power down of the DSP and it won't
> matter but if this is the case it isn't at all clear from the immediate
> code.
None is just a SW state thats why :)

The DSP has the bitmap for interlever as explained in sst_send_slot_map()
/*
 * channel map value is a bitfield where each bit represents a slot
 *
 *                        76543210      # 0 = slot 0, 1 = slot 1
 *
 * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot
 * 0, 2
 */
so based on the channel selected here we will set the map and send to DSP.
For none case we don't need to do anything. Yes this bit need this
explanation so will add up

> 
> > +	if (type == SST_MODULE_GAIN) {
> > +		struct sst_gain_mixer_control *mc = (void *)kctl->private_value;
> > +
> > +		mc->w = w;
> > +		module->kctl = kctl;
> > +		list_add_tail(&module->node, &ids->gain_list);
> > +	} else if (type == SST_MODULE_ALGO) {
> > +		struct sst_algo_control *bc = (void *)kctl->private_value;
> > +
> > +		bc->w = w;
> > +		module->kctl = kctl;
> > +		list_add_tail(&module->node, &ids->algo_list);
> > +	}
> 
> This looks like it should be a switch statement with a default case to
> trap any errors.
> 
> > +		if (idx == NULL)
> > +			continue;
> > +		index  = strlen(kctl->id.name) - strlen(idx);
> 
> I keep on seeing lots of code with random double instead of single
> spaces.
this shouldn't be, will fix

> > +		if (strstr(kctl->id.name, "Volume") &&
> > +		    !strncmp(kctl->id.name, w->name, index))
> > +			ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN);
> > +
> > +		else if (strstr(kctl->id.name, "params") &&
> > +			 !strncmp(kctl->id.name, w->name, index))
> > +			ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO);
> > +
> > +		else if (strstr(kctl->id.name, "Switch") &&
> > +			 !strncmp(kctl->id.name, w->name, index)) {
> > +			struct sst_gain_mixer_control *mc =
> > +						(void *)kctl->private_value;
> > +
> > +			mc->w = w;
> > +
> > +		} else if (strstr(kctl->id.name, "interleaver") &&
> 
> Both or no branches of an if statement should use { }.
will fix

> 
> > +			 !strncmp(kctl->id.name, w->name, index)) {
> > +			struct sst_enum *e = (void *)kctl->private_value;
> > +
> > +			e->w = w;
> > +
> > +		} else if (strstr(kctl->id.name, "deinterleaver") &&
> > +			 !strncmp(kctl->id.name, w->name, index)) {
> > +
> > +			struct sst_enum *e = (void *)kctl->private_value;
> > +
> > +			e->w = w;
> > +		}
> 
> Again no fallthrough case.
would that be valid. We have only volume, params, mute, interlevaer and
deinterleaver controls only :)
Can add fall thru for therotical if you feel that is right thing to do.

Thanks
-- 
~Vinod

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Re: [PATCH v7 3/7] ASoC: Intel: mrfld: add DSP core controls

[Mark Brown]

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On Thu, Sep 25, 2014 at 09:38:53PM +0530, Vinod Koul wrote:
> On Thu, Sep 25, 2014 at 04:08:13PM +0100, Mark Brown wrote:
> > On Fri, Sep 19, 2014 at 04:46:04PM +0530, Subhransu S. Prusty wrote:

> > > +       u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;

> > So the "channel" map is for transmit and the "slot" map is for receive?
> > That naming doesn't seem at all obvious, I'd expect some confusion and
> > resulting bugs there.  

> yes as multiple channels get interleaved to slot while transmitting and on
> recive side various lots get de-interleaved to channels :)
> These controls allow you to route stuff.

I can tell what they're supposed to do given the above code, what I'm
saying is that when I see the above code I'm expecting to find bugs
elsewhere since it's not going to be clear if you see either variable in
isolation.  Better naming please, for example use the direction.

> The DSP has the bitmap for interlever as explained in sst_send_slot_map()

> /*
>  * channel map value is a bitfield where each bit represents a slot
>  *
>  *                        76543210      # 0 = slot 0, 1 = slot 1
>  *
>  * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot
>  * 0, 2
>  */

> so based on the channel selected here we will set the map and send to DSP.
> For none case we don't need to do anything. Yes this bit need this
> explanation so will add up

But why not - why is the effect of clearing all the bits to do nothing?

> > > +			 !strncmp(kctl->id.name, w->name, index)) {
> > > +			struct sst_enum *e = (void *)kctl->private_value;
> > > +
> > > +			e->w = w;
> > > +
> > > +		} else if (strstr(kctl->id.name, "deinterleaver") &&
> > > +			 !strncmp(kctl->id.name, w->name, index)) {
> > > +
> > > +			struct sst_enum *e = (void *)kctl->private_value;
> > > +
> > > +			e->w = w;
> > > +		}

> > Again no fallthrough case.

> would that be valid. We have only volume, params, mute, interlevaer and
> deinterleaver controls only :)
> Can add fall thru for therotical if you feel that is right thing to do.

Yes, it's better.  This is part of the whole thing about the code
setting off alarm bells - things look fragile and the missing error
checking does nothing to dispel that.

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Re: [PATCH v7 3/7] ASoC: Intel: mrfld: add DSP core controls

[Vinod Koul]

[-- Attachment #1.1: Type: text/plain, Size: 2940 bytes --]

On Sat, Sep 27, 2014 at 12:13:33PM +0100, Mark Brown wrote:
> On Thu, Sep 25, 2014 at 09:38:53PM +0530, Vinod Koul wrote:
> > On Thu, Sep 25, 2014 at 04:08:13PM +0100, Mark Brown wrote:
> > > On Fri, Sep 19, 2014 at 04:46:04PM +0530, Subhransu S. Prusty wrote:
> 
> > > > +       u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map;
> 
> > > So the "channel" map is for transmit and the "slot" map is for receive?
> > > That naming doesn't seem at all obvious, I'd expect some confusion and
> > > resulting bugs there.  
> 
> > yes as multiple channels get interleaved to slot while transmitting and on
> > recive side various lots get de-interleaved to channels :)
> > These controls allow you to route stuff.
> 
> I can tell what they're supposed to do given the above code, what I'm
> saying is that when I see the above code I'm expecting to find bugs
> elsewhere since it's not going to be clear if you see either variable in
> isolation.  Better naming please, for example use the direction.
Sure will add tx and rx to the control names to make it explict. Thanks for
the suggestion

> > The DSP has the bitmap for interlever as explained in sst_send_slot_map()
> 
> > /*
> >  * channel map value is a bitfield where each bit represents a slot
> >  *
> >  *                        76543210      # 0 = slot 0, 1 = slot 1
> >  *
> >  * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot
> >  * 0, 2
> >  */
> 
> > so based on the channel selected here we will set the map and send to DSP.
> > For none case we don't need to do anything. Yes this bit need this
> > explanation so will add up
> 
> But why not - why is the effect of clearing all the bits to do nothing?
Okay thanks for pointing out, We did check on this bit. So initially DSP
wasn't supporting the dynamic updates of the  slots so thsi was fine but
later we added support. On re-examining again as you pointed yes 'none'
value needs to reset the bitmap and send to DSP so we just need to forward
the IPC here as well. I will fix it up.

> > > > +			 !strncmp(kctl->id.name, w->name, index)) {
> > > > +			struct sst_enum *e = (void *)kctl->private_value;
> > > > +
> > > > +			e->w = w;
> > > > +
> > > > +		} else if (strstr(kctl->id.name, "deinterleaver") &&
> > > > +			 !strncmp(kctl->id.name, w->name, index)) {
> > > > +
> > > > +			struct sst_enum *e = (void *)kctl->private_value;
> > > > +
> > > > +			e->w = w;
> > > > +		}
> 
> > > Again no fallthrough case.
> 
> > would that be valid. We have only volume, params, mute, interlevaer and
> > deinterleaver controls only :)
> > Can add fall thru for therotical if you feel that is right thing to do.
> 
> Yes, it's better.  This is part of the whole thing about the code
> setting off alarm bells - things look fragile and the missing error
> checking does nothing to dispel that.
Yup, will add these

-- 
~Vinod

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[PATCH v7 4/7] ASoC: Export dapm_kcontrol_get_value

[Subhransu S. Prusty]

The DSP driver needs to know widget control value in its event handler for
widgets like mixers. This is required in the subsequent patches

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
---
 include/sound/soc-dapm.h | 1 +
 sound/soc/soc-dapm.c     | 3 ++-
 2 files changed, 3 insertions(+), 1 deletion(-)

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index f955d65..e6dd703 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -432,6 +432,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
 				const char *pin);
 void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card);
+unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol);
 
 /* Mostly internal - should not normally be used */
 void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7efe4fa..a2905cd 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list(
 	list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \
 		list_kcontrol)
 
-static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol)
+unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol)
 {
 	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
 
 	return data->value;
 }
+EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value);
 
 static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
 	unsigned int value)
-- 
1.9.0

Re: [PATCH v7 4/7] ASoC: Export dapm_kcontrol_get_value

[Mark Brown]

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On Fri, Sep 19, 2014 at 04:46:05PM +0530, Subhransu S. Prusty wrote:
> The DSP driver needs to know widget control value in its event handler for
> widgets like mixers. This is required in the subsequent patches

Applied, thanks.

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[PATCH v7 5/7] ASoC: Intel: mrfld: add the DSP DAPM widgets

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

This patch adds all DAPM widgets and the event handlers for
DSP except the mixers.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-atom-controls.c | 225 ++++++++++++++++++++++++++++++++++++
 sound/soc/intel/sst-mfld-platform.h |   4 +
 2 files changed, 229 insertions(+)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index dd830f7..28ec5b1 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -503,6 +503,40 @@ static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = {
 	[SST_IP_MEDIA3]		= SST_SWM_IN_MEDIA3,
 };
 
+/**
+ * fill_swm_input - fill in the SWM input ids given the register
+ *
+ * The register value is a bit-field inicated which mixer inputs are ON. Use the
+ * lookup table to get the input-id and fill it in the structure.
+ */
+static int fill_swm_input(struct snd_soc_component *cmpnt,
+		struct swm_input_ids *swm_input, unsigned int reg)
+{
+	uint i, is_set, nb_inputs = 0;
+	u16 input_loc_id;
+
+	dev_dbg(cmpnt->dev, "reg: %#x\n", reg);
+	for (i = 0; i < SST_SWM_INPUT_COUNT; i++) {
+		is_set = reg & BIT(i);
+		if (!is_set)
+			continue;
+
+		input_loc_id = swm_mixer_input_ids[i];
+		SST_FILL_DESTINATION(2, swm_input->input_id,
+				     input_loc_id, SST_DEFAULT_MODULE_ID);
+		nb_inputs++;
+		swm_input++;
+		dev_dbg(cmpnt->dev, "input id: %#x, nb_inputs: %d\n",
+				input_loc_id, nb_inputs);
+
+		if (nb_inputs == SST_CMD_SWM_MAX_INPUTS) {
+			dev_warn(cmpnt->dev, "SET_SWM cmd max inputs reached");
+			break;
+		}
+	}
+	return nb_inputs;
+}
+
 static int sst_set_pipe_gain(struct sst_ids *ids,
 			struct sst_data *drv, int mute)
 {
@@ -526,6 +560,112 @@ static int sst_set_pipe_gain(struct sst_ids *ids,
 	return ret;
 }
 
+static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	struct sst_cmd_set_swm cmd;
+	struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_ids *ids = w->priv;
+	bool set_mixer = false;
+	struct soc_mixer_control *mc;
+	int val = 0;
+	int i = 0;
+
+	dev_dbg(cmpnt->dev, "widget = %s\n", w->name);
+	/*
+	 * Identify which mixer input is on and send the bitmap of the
+	 * inputs as an IPC to the DSP.
+	 */
+	for (i = 0; i < w->num_kcontrols; i++) {
+		if (dapm_kcontrol_get_value(w->kcontrols[i])) {
+			mc = (struct soc_mixer_control *)(w->kcontrols[i])->private_value;
+			val |= 1 << mc->shift;
+		}
+	}
+	dev_dbg(cmpnt->dev, "val = %#x\n", val);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+	case SND_SOC_DAPM_POST_PMD:
+		set_mixer = true;
+		break;
+	case SND_SOC_DAPM_POST_REG:
+		if (w->power)
+			set_mixer = true;
+		break;
+	default:
+		set_mixer = false;
+	}
+
+	if (set_mixer == false)
+		return 0;
+
+	if (SND_SOC_DAPM_EVENT_ON(event) ||
+	    event == SND_SOC_DAPM_POST_REG)
+		cmd.switch_state = SST_SWM_ON;
+	else
+		cmd.switch_state = SST_SWM_OFF;
+
+	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+	/* MMX_SET_SWM == SBA_SET_SWM */
+	cmd.header.command_id = SBA_SET_SWM;
+
+	SST_FILL_DESTINATION(2, cmd.output_id,
+			     ids->location_id, SST_DEFAULT_MODULE_ID);
+	cmd.nb_inputs =	fill_swm_input(cmpnt, &cmd.input[0], val);
+	cmd.header.length = offsetof(struct sst_cmd_set_swm, input)
+				- sizeof(struct sst_dsp_header)
+				+ (cmd.nb_inputs * sizeof(cmd.input[0]));
+
+	return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+			      ids->task_id, 0, &cmd,
+			      sizeof(cmd.header) + cmd.header.length);
+}
+
+/* SBA mixers - 16 inputs */
+#define SST_SBA_DECLARE_MIX_CONTROLS(kctl_name)						\
+	static const struct snd_kcontrol_new kctl_name[] = {				\
+		SOC_DAPM_SINGLE("codec_in0", SND_SOC_NOPM, SST_IP_CODEC0, 1, 0),	\
+		SOC_DAPM_SINGLE("codec_in1", SND_SOC_NOPM, SST_IP_CODEC1, 1, 0),	\
+		SOC_DAPM_SINGLE("sprot_loop_in", SND_SOC_NOPM, SST_IP_LOOP0, 1, 0),	\
+		SOC_DAPM_SINGLE("media_loop1_in", SND_SOC_NOPM, SST_IP_LOOP1, 1, 0),	\
+		SOC_DAPM_SINGLE("media_loop2_in", SND_SOC_NOPM, SST_IP_LOOP2, 1, 0),	\
+		SOC_DAPM_SINGLE("pcm0_in", SND_SOC_NOPM, SST_IP_PCM0, 1, 0),		\
+		SOC_DAPM_SINGLE("pcm1_in", SND_SOC_NOPM, SST_IP_PCM1, 1, 0),		\
+	}
+
+#define SST_SBA_MIXER_GRAPH_MAP(mix_name)			\
+	{ mix_name, "codec_in0",	"codec_in0" },		\
+	{ mix_name, "codec_in1",	"codec_in1" },		\
+	{ mix_name, "sprot_loop_in",	"sprot_loop_in" },	\
+	{ mix_name, "media_loop1_in",	"media_loop1_in" },	\
+	{ mix_name, "media_loop2_in",	"media_loop2_in" },	\
+	{ mix_name, "pcm0_in",		"pcm0_in" },		\
+	{ mix_name, "pcm1_in",		"pcm1_in" }
+
+#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name)			\
+	static const struct snd_kcontrol_new kctl_name[] = {			\
+		SOC_DAPM_SINGLE("media0_in", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0),	\
+		SOC_DAPM_SINGLE("media1_in", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0),	\
+		SOC_DAPM_SINGLE("media2_in", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0),	\
+		SOC_DAPM_SINGLE("media3_in", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0),	\
+	}
+
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media0_controls);
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media1_controls);
+
+/* 18 SBA mixers */
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls);
+
 /*
  * sst_handle_vb_timer - Start/Stop the DSP scheduler
  *
@@ -737,6 +877,83 @@ static int sst_set_media_loop(struct snd_soc_dapm_widget *w,
 	return ret;
 }
 
+static const struct snd_soc_dapm_widget sst_dapm_widgets[] = {
+	SST_AIF_IN("codec_in0", sst_set_be_modules),
+	SST_AIF_IN("codec_in1", sst_set_be_modules),
+	SST_AIF_OUT("codec_out0", sst_set_be_modules),
+	SST_AIF_OUT("codec_out1", sst_set_be_modules),
+
+	/* Media Paths */
+	/* MediaX IN paths are set via ALLOC, so no SET_MEDIA_PATH command */
+	SST_PATH_INPUT("media0_in", SST_TASK_MMX, SST_SWM_IN_MEDIA0, sst_generic_modules_event),
+	SST_PATH_INPUT("media1_in", SST_TASK_MMX, SST_SWM_IN_MEDIA1, NULL),
+	SST_PATH_INPUT("media2_in", SST_TASK_MMX, SST_SWM_IN_MEDIA2, sst_set_media_path),
+	SST_PATH_INPUT("media3_in", SST_TASK_MMX, SST_SWM_IN_MEDIA3, NULL),
+	SST_PATH_OUTPUT("media0_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA0, sst_set_media_path),
+	SST_PATH_OUTPUT("media1_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA1, sst_set_media_path),
+
+	/* SBA PCM Paths */
+	SST_PATH_INPUT("pcm0_in", SST_TASK_SBA, SST_SWM_IN_PCM0, sst_set_media_path),
+	SST_PATH_INPUT("pcm1_in", SST_TASK_SBA, SST_SWM_IN_PCM1, sst_set_media_path),
+	SST_PATH_OUTPUT("pcm0_out", SST_TASK_SBA, SST_SWM_OUT_PCM0, sst_set_media_path),
+	SST_PATH_OUTPUT("pcm1_out", SST_TASK_SBA, SST_SWM_OUT_PCM1, sst_set_media_path),
+	SST_PATH_OUTPUT("pcm2_out", SST_TASK_SBA, SST_SWM_OUT_PCM2, sst_set_media_path),
+
+	/* SBA Loops */
+	SST_PATH_INPUT("sprot_loop_in", SST_TASK_SBA, SST_SWM_IN_SPROT_LOOP, NULL),
+	SST_PATH_INPUT("media_loop1_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP1, NULL),
+	SST_PATH_INPUT("media_loop2_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP2, NULL),
+	SST_PATH_MEDIA_LOOP_OUTPUT("sprot_loop_out", SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, SST_FMT_MONO, sst_set_media_loop),
+	SST_PATH_MEDIA_LOOP_OUTPUT("media_loop1_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, SST_FMT_MONO, sst_set_media_loop),
+	SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop),
+
+	/* Media Mixers */
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"media0_in", NULL, "Compress Playback"},
+	{"media1_in", NULL, "Headset Playback"},
+	{"media2_in", NULL, "pcm0_out"},
+
+	{"media0_out mix 0", "media0_in", "media0_in"},
+	{"media0_out mix 0", "media1_in", "media1_in"},
+	{"media0_out mix 0", "media2_in", "media2_in"},
+	{"media0_out mix 0", "media3_in", "media3_in"},
+	{"media1_out mix 0", "media0_in", "media0_in"},
+	{"media1_out mix 0", "media1_in", "media1_in"},
+	{"media1_out mix 0", "media2_in", "media2_in"},
+	{"media1_out mix 0", "media3_in", "media3_in"},
+
+	{"media0_out", NULL, "media0_out mix 0"},
+	{"media1_out", NULL, "media1_out mix 0"},
+	{"pcm0_in", NULL, "media0_out"},
+	{"pcm1_in", NULL, "media1_out"},
+
+	{"Headset Capture", NULL, "pcm1_out"},
+	{"Headset Capture", NULL, "pcm2_out"},
+	{"pcm0_out", NULL, "pcm0_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("pcm0_out mix 0"),
+	{"pcm1_out", NULL, "pcm1_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("pcm1_out mix 0"),
+	{"pcm2_out", NULL, "pcm2_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("pcm2_out mix 0"),
+
+	{"media_loop1_in", NULL, "media_loop1_out"},
+	{"media_loop1_out", NULL, "media_loop1_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("media_loop1_out mix 0"),
+	{"media_loop2_in", NULL, "media_loop2_out"},
+	{"media_loop2_out", NULL, "media_loop2_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("media_loop2_out mix 0"),
+	{"sprot_loop_in", NULL, "sprot_loop_out"},
+	{"sprot_loop_out", NULL, "sprot_loop_out mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("sprot_loop_out mix 0"),
+
+	{"codec_out0", NULL, "codec_out0 mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("codec_out0 mix 0"),
+	{"codec_out1", NULL, "codec_out1 mix 0"},
+	SST_SBA_MIXER_GRAPH_MAP("codec_out1 mix 0"),
+
+};
 static const char * const slot_names[] = {
 	"none",
 	"slot 0", "slot 1", "slot 2", "slot 3",
@@ -1064,6 +1281,8 @@ static int sst_map_modules_to_pipe(struct snd_soc_platform *platform)
 int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 {
 	int i, ret = 0;
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(&platform->component);
 	struct sst_data *drv = snd_soc_platform_get_drvdata(platform);
 
 	drv->byte_stream = devm_kzalloc(platform->dev,
@@ -1071,6 +1290,12 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
 	if (!drv->byte_stream)
 		return -ENOMEM;
 
+	snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
+			ARRAY_SIZE(sst_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon,
+			ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_widgets(dapm->card);
+
 	for (i = 0; i < SST_NUM_GAINS; i++) {
 		sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT;
 		sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT;
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 19f83ec..d41d1c3 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -153,6 +153,10 @@ struct sst_device {
 struct sst_data;
 
 int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform);
+int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute);
+int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable);
+int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable);
+
 void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
 int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
 			   struct snd_sst_params *str_params, bool is_compress);
-- 
1.9.0

Re: [PATCH v7 5/7] ASoC: Intel: mrfld: add the DSP DAPM widgets

[Mark Brown]

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On Fri, Sep 19, 2014 at 04:46:06PM +0530, Subhransu S. Prusty wrote:

> +#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name)			\
> +	static const struct snd_kcontrol_new kctl_name[] = {			\
> +		SOC_DAPM_SINGLE("media0_in", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0),	\
> +		SOC_DAPM_SINGLE("media1_in", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0),	\
> +		SOC_DAPM_SINGLE("media2_in", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0),	\
> +		SOC_DAPM_SINGLE("media3_in", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0),	\
> +	}

These should be Switch controls?

> +	snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
> +			ARRAY_SIZE(sst_dapm_widgets));
> +	snd_soc_dapm_add_routes(dapm, intercon,
> +			ARRAY_SIZE(intercon));
> +	snd_soc_dapm_new_widgets(dapm->card);

We should be able to do this from the component but we can't yet...

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Re: [PATCH v7 5/7] ASoC: Intel: mrfld: add the DSP DAPM widgets

[Vinod Koul]

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On Thu, Oct 02, 2014 at 07:12:28PM +0100, Mark Brown wrote:
> On Fri, Sep 19, 2014 at 04:46:06PM +0530, Subhransu S. Prusty wrote:
> 
> > +#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name)			\
> > +	static const struct snd_kcontrol_new kctl_name[] = {			\
> > +		SOC_DAPM_SINGLE("media0_in", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0),	\
> > +		SOC_DAPM_SINGLE("media1_in", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0),	\
> > +		SOC_DAPM_SINGLE("media2_in", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0),	\
> > +		SOC_DAPM_SINGLE("media3_in", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0),	\
> > +	}
> 
> These should be Switch controls?
Yup I will fix this one.

> 
> > +	snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
> > +			ARRAY_SIZE(sst_dapm_widgets));
> > +	snd_soc_dapm_add_routes(dapm, intercon,
> > +			ARRAY_SIZE(intercon));
> > +	snd_soc_dapm_new_widgets(dapm->card);
> 
> We should be able to do this from the component but we can't yet...
We can convert once ready :)

I still have lots of stuff to do once this series is in. We also need to
enable dynamic firmware here once liam upstreams basic support :)

Thanks
-- 
~Vinod

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Re: [PATCH v7 5/7] ASoC: Intel: mrfld: add the DSP DAPM widgets

[Vinod Koul]

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On Mon, Oct 06, 2014 at 08:10:10AM +0530, Vinod Koul wrote:
> On Thu, Oct 02, 2014 at 07:12:28PM +0100, Mark Brown wrote:
> > On Fri, Sep 19, 2014 at 04:46:06PM +0530, Subhransu S. Prusty wrote:
> > 
> > > +#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name)			\
> > > +	static const struct snd_kcontrol_new kctl_name[] = {			\
> > > +		SOC_DAPM_SINGLE("media0_in", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0),	\
> > > +		SOC_DAPM_SINGLE("media1_in", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0),	\
> > > +		SOC_DAPM_SINGLE("media2_in", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0),	\
> > > +		SOC_DAPM_SINGLE("media3_in", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0),	\
> > > +	}
> > 
> > These should be Switch controls?
> Yup I will fix this one.
> 
> > 
> > > +	snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
> > > +			ARRAY_SIZE(sst_dapm_widgets));
> > > +	snd_soc_dapm_add_routes(dapm, intercon,
> > > +			ARRAY_SIZE(intercon));
> > > +	snd_soc_dapm_new_widgets(dapm->card);
> > 
> > We should be able to do this from the component but we can't yet...
> We can convert once ready :)
> 
> I still have lots of stuff to do once this series is in. We also need to
> enable dynamic firmware here once liam upstreams basic support :)

Wanted to check if you have any comments on last two patches, before I
send updated series :)

-- 
~Vinod

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Re: [PATCH v7 5/7] ASoC: Intel: mrfld: add the DSP DAPM widgets

[Mark Brown]

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On Mon, Oct 06, 2014 at 08:14:35AM +0530, Vinod Koul wrote:

> Wanted to check if you have any comments on last two patches, before I
> send updated series :)

Not at the minute.

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[PATCH v7 6/7] ASoC: Intel: mfld-pcm: add FE and BE ops

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

Now that we have added code for managing DSP pipelines we need to
add the code for DSPs FrontEnd and Backend dai.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-mfld-platform-pcm.c | 153 ++++++++++++++++++++++++++------
 1 file changed, 125 insertions(+), 28 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 6f5edd6..6962105 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -101,35 +101,11 @@ static struct sst_dev_stream_map dpcm_strm_map[] = {
 	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
 };
 
-/* MFLD - MSIC */
-static struct snd_soc_dai_driver sst_platform_dai[] = {
+static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
-	.name = "Headset-cpu-dai",
-	.id = 0,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 5,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Compress-cpu-dai",
-	.compress_dai = 1,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-},
-};
+
+	return sst_send_pipe_gains(dai, stream, mute);
+}
 
 /* helper functions */
 void sst_set_stream_status(struct sst_runtime_stream *stream,
@@ -451,12 +427,133 @@ static int sst_media_hw_free(struct snd_pcm_substream *substream,
 	return snd_pcm_lib_free_pages(substream);
 }
 
+static int sst_enable_ssp(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	int ret = 0;
+
+	if (!dai->active) {
+		ret = sst_handle_vb_timer(dai, true);
+		if (ret)
+			return ret;
+		ret = send_ssp_cmd(dai, dai->name, 1);
+	}
+	return ret;
+}
+
+static void sst_disable_ssp(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	if (!dai->active) {
+		send_ssp_cmd(dai, dai->name, 0);
+		sst_handle_vb_timer(dai, false);
+	}
+}
+
 static struct snd_soc_dai_ops sst_media_dai_ops = {
 	.startup = sst_media_open,
 	.shutdown = sst_media_close,
 	.prepare = sst_media_prepare,
 	.hw_params = sst_media_hw_params,
 	.hw_free = sst_media_hw_free,
+	.mute_stream = sst_media_digital_mute,
+};
+
+static struct snd_soc_dai_ops sst_compr_dai_ops = {
+	.mute_stream = sst_media_digital_mute,
+};
+
+static struct snd_soc_dai_ops sst_be_dai_ops = {
+	.startup = sst_enable_ssp,
+	.shutdown = sst_disable_ssp,
+};
+
+static struct snd_soc_dai_driver sst_platform_dai[] = {
+{
+	.name = "media-cpu-dai",
+	.ops = &sst_media_dai_ops,
+	.playback = {
+		.stream_name = "Headset Playback",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "Headset Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "compress-cpu-dai",
+	.compress_dai = 1,
+	.ops = &sst_compr_dai_ops,
+	.playback = {
+		.stream_name = "Compress Playback",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+/* BE CPU  Dais */
+{
+	.name = "ssp0-port",
+	.ops = &sst_be_dai_ops,
+	.playback = {
+		.stream_name = "ssp0 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp0 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "ssp1-port",
+	.ops = &sst_be_dai_ops,
+	.playback = {
+		.stream_name = "ssp1 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp1 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "ssp2-port",
+	.ops = &sst_be_dai_ops,
+	.playback = {
+		.stream_name = "ssp2 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp2 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
 };
 
 static int sst_platform_open(struct snd_pcm_substream *substream)
-- 
1.9.0

[PATCH v7 7/7] ASoC: Intel: mrfld: add the DSP mixers

[Subhransu S. Prusty]

From: Vinod Koul <vinod.koul@intel.com>

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
---
 sound/soc/intel/sst-atom-controls.c | 26 ++++++++++++++++++++++++++
 1 file changed, 26 insertions(+)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index 28ec5b1..cb5d295 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -908,6 +908,32 @@ static const struct snd_soc_dapm_widget sst_dapm_widgets[] = {
 	SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop),
 
 	/* Media Mixers */
+	SST_SWM_MIXER("media0_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA0,
+		      sst_mix_media0_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("media1_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA1,
+		      sst_mix_media1_controls, sst_swm_mixer_event),
+
+	/* SBA PCM mixers */
+	SST_SWM_MIXER("pcm0_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM0,
+		      sst_mix_pcm0_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("pcm1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM1,
+		      sst_mix_pcm1_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("pcm2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM2,
+		      sst_mix_pcm2_controls, sst_swm_mixer_event),
+
+	/* SBA Loop mixers */
+	SST_SWM_MIXER("sprot_loop_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP,
+		      sst_mix_sprot_l0_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("media_loop1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1,
+		      sst_mix_media_l1_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("media_loop2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2,
+		      sst_mix_media_l2_controls, sst_swm_mixer_event),
+
+	/* SBA Backend mixers */
+	SST_SWM_MIXER("codec_out0 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC0,
+		      sst_mix_codec0_controls, sst_swm_mixer_event),
+	SST_SWM_MIXER("codec_out1 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC1,
+		      sst_mix_codec1_controls, sst_swm_mixer_event),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-- 
1.9.0