[PATCH v3 1/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding

[PATCH v3 1/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding

[Adam Serbinski]

This patch adds bindings required for PCM ports on AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 include/dt-bindings/sound/qcom,q6afe.h | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h
index 1df06f8ad5c3..f3a435a112cb 100644
--- a/include/dt-bindings/sound/qcom,q6afe.h
+++ b/include/dt-bindings/sound/qcom,q6afe.h
@@ -107,6 +107,14 @@
 #define QUINARY_TDM_RX_7	102
 #define QUINARY_TDM_TX_7	103
 #define DISPLAY_PORT_RX		104
+#define PRIMARY_PCM_RX		105
+#define PRIMARY_PCM_TX		106
+#define SECONDARY_PCM_RX	107
+#define SECONDARY_PCM_TX	108
+#define TERTIARY_PCM_RX		109
+#define TERTIARY_PCM_TX		110
+#define QUATERNARY_PCM_RX	111
+#define QUATERNARY_PCM_TX	112
 
 #endif /* __DT_BINDINGS_Q6_AFE_H__ */
 
-- 
2.21.1

[alsa-devel] [PATCH v3 1/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding

[Adam Serbinski]

This patch adds bindings required for PCM ports on AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 include/dt-bindings/sound/qcom,q6afe.h | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h
index 1df06f8ad5c3..f3a435a112cb 100644
--- a/include/dt-bindings/sound/qcom,q6afe.h
+++ b/include/dt-bindings/sound/qcom,q6afe.h
@@ -107,6 +107,14 @@
 #define QUINARY_TDM_RX_7	102
 #define QUINARY_TDM_TX_7	103
 #define DISPLAY_PORT_RX		104
+#define PRIMARY_PCM_RX		105
+#define PRIMARY_PCM_TX		106
+#define SECONDARY_PCM_RX	107
+#define SECONDARY_PCM_TX	108
+#define TERTIARY_PCM_RX		109
+#define TERTIARY_PCM_TX		110
+#define QUATERNARY_PCM_RX	111
+#define QUATERNARY_PCM_TX	112
 
 #endif /* __DT_BINDINGS_Q6_AFE_H__ */
 
-- 
2.21.1

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Alsa-devel@alsa-project.org
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[PATCH v3 2/6] ASoC: qdsp6: q6afe: add support to pcm ports

This patch adds support to pcm ports in AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6afe.c | 118 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6afe.h |  13 +++-
 2 files changed, 130 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7a58c8..11de01679cad 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -40,6 +40,7 @@
 
 #define AFE_PARAM_ID_SLIMBUS_CONFIG    0x00010212
 #define AFE_PARAM_ID_I2S_CONFIG	0x0001020D
+#define AFE_PARAM_ID_PCM_CONFIG        0x0001020E
 #define AFE_PARAM_ID_TDM_CONFIG	0x0001029D
 #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG	0x00010297
 
@@ -117,6 +118,15 @@
 #define AFE_PORT_ID_QUATERNARY_MI2S_RX      0x1006
 #define AFE_PORT_ID_QUATERNARY_MI2S_TX      0x1007
 
+#define AFE_PORT_ID_PRIMARY_PCM_RX          0x100A
+#define AFE_PORT_ID_PRIMARY_PCM_TX          0x100B
+#define AFE_PORT_ID_SECONDARY_PCM_RX        0x100C
+#define AFE_PORT_ID_SECONDARY_PCM_TX        0x100D
+#define AFE_PORT_ID_TERTIARY_PCM_RX         0x1012
+#define AFE_PORT_ID_TERTIARY_PCM_TX         0x1013
+#define AFE_PORT_ID_QUATERNARY_PCM_RX       0x1014
+#define AFE_PORT_ID_QUATERNARY_PCM_TX       0x1015
+
 /* Start of the range of port IDs for TDM devices. */
 #define AFE_PORT_ID_TDM_PORT_RANGE_START	0x9000
 
@@ -421,6 +431,29 @@ struct afe_digital_clk_cfg {
 	u16                  reserved;
 } __packed;
 
+#define AFE_API_VERSION_PCM_CONFIG		0x1
+#define AFE_PORT_PCM_QUANT_MAX			0x5
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL		0x0
+#define AFE_PORT_PCM_SYNC_SRC_INTERNAL		0x1
+#define AFE_PORT_PCM_AUX_MODE_PCM		0x0
+#define AFE_PORT_PCM_BITS_PER_FRAME_64		0x3
+#define AFE_PORT_PCM_BITS_PER_FRAME_128		0x4
+#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE	0x0
+
+struct afe_param_id_pcm_cfg {
+	u32	pcm_cfg_minor_version;
+	u16	aux_mode;
+	u16	sync_src;
+	u16	frame_setting;
+	u16	quantype;
+	u16	ctrl_data_out_enable;
+	u16	reserved;
+	u32	sample_rate;
+	u16	bit_width;
+	u16	num_channels;
+	u16	slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
+} __packed;
+
 struct afe_param_id_i2s_cfg {
 	u32	i2s_cfg_minor_version;
 	u16	bit_width;
@@ -452,6 +485,7 @@ union afe_port_config {
 	struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
 	struct afe_param_id_slimbus_cfg           slim_cfg;
 	struct afe_param_id_i2s_cfg	i2s_cfg;
+	struct afe_param_id_pcm_cfg	pcm_cfg;
 	struct afe_param_id_tdm_cfg	tdm_cfg;
 } __packed;
 
@@ -707,6 +741,22 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				QUINARY_TDM_TX_7, 0, 1},
 	[DISPLAY_PORT_RX] = { AFE_PORT_ID_HDMI_OVER_DP_RX,
 				DISPLAY_PORT_RX, 1, 1},
+	[PRIMARY_PCM_RX] = { AFE_PORT_ID_PRIMARY_PCM_RX,
+				PRIMARY_PCM_RX, 1, 1},
+	[PRIMARY_PCM_TX] = { AFE_PORT_ID_PRIMARY_PCM_TX,
+				PRIMARY_PCM_RX, 0, 1},
+	[SECONDARY_PCM_RX] = { AFE_PORT_ID_SECONDARY_PCM_RX,
+				SECONDARY_PCM_RX, 1, 1},
+	[SECONDARY_PCM_TX] = { AFE_PORT_ID_SECONDARY_PCM_TX,
+				SECONDARY_PCM_TX, 0, 1},
+	[TERTIARY_PCM_RX] = { AFE_PORT_ID_TERTIARY_PCM_RX,
+				TERTIARY_PCM_RX, 1, 1},
+	[TERTIARY_PCM_TX] = { AFE_PORT_ID_TERTIARY_PCM_TX,
+				TERTIARY_PCM_TX, 0, 1},
+	[QUATERNARY_PCM_RX] = { AFE_PORT_ID_QUATERNARY_PCM_RX,
+				QUATERNARY_PCM_RX, 1, 1},
+	[QUATERNARY_PCM_TX] = { AFE_PORT_ID_QUATERNARY_PCM_TX,
+				QUATERNARY_PCM_TX, 0, 1},
 };
 
 static void q6afe_port_free(struct kref *ref)
@@ -993,6 +1043,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
 		break;
 	case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR:
 	case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+	/* TDM cases overlap with PCM */
 	case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT:
 		cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET;
 		cset.clk_id = clk_id;
@@ -1145,6 +1196,63 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 }
 EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
 
+/**
+ * q6afe_pcm_port_prepare() - Prepare pcm afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: PCM configuration for the afe port
+ *
+ */
+int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg)
+{
+	union afe_port_config *pcfg = &port->port_cfg;
+	struct device *dev = port->afe->dev;
+
+	pcfg->pcm_cfg.pcm_cfg_minor_version = AFE_API_VERSION_PCM_CONFIG;
+
+	if (cfg->quantype > AFE_PORT_PCM_QUANT_MAX) {
+		dev_err(dev, "Invalid pcm quantization\n");
+		return -EINVAL;
+	}
+	pcfg->pcm_cfg.quantype = cfg->quantype;
+
+	memcpy(pcfg->pcm_cfg.slot_number_mapping, cfg->slot_number_mapping,
+	       sizeof(u16) * AFE_PORT_MAX_PCM_AUDIO_CHAN_CNT);
+
+	pcfg->pcm_cfg.aux_mode = AFE_PORT_PCM_AUX_MODE_PCM;
+	switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL;
+		break;
+	default:
+		dev_err(dev, "Unsupported format\n");
+		return -EINVAL;
+	}
+
+	switch (cfg->sample_rate) {
+	case 8000:
+		pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128;
+		break;
+	case 16000:
+		pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64;
+		break;
+	default:
+		dev_err(dev, "Unsupported sample rate\n");
+		return -EINVAL;
+	}
+	pcfg->pcm_cfg.ctrl_data_out_enable = AFE_PORT_PCM_CTRL_DATA_OE_DISABLE;
+	pcfg->pcm_cfg.reserved = 0;
+	pcfg->pcm_cfg.sample_rate = cfg->sample_rate;
+	pcfg->pcm_cfg.bit_width = 16;
+	pcfg->pcm_cfg.num_channels = 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(q6afe_pcm_port_prepare);
+
 /**
  * q6afe_i2s_port_prepare() - Prepare i2s afe port.
  *
@@ -1417,6 +1525,16 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_QUATERNARY_MI2S_TX:
 		cfg_type = AFE_PARAM_ID_I2S_CONFIG;
 		break;
+	case AFE_PORT_ID_PRIMARY_PCM_RX:
+	case AFE_PORT_ID_PRIMARY_PCM_TX:
+	case AFE_PORT_ID_SECONDARY_PCM_RX:
+	case AFE_PORT_ID_SECONDARY_PCM_TX:
+	case AFE_PORT_ID_TERTIARY_PCM_RX:
+	case AFE_PORT_ID_TERTIARY_PCM_TX:
+	case AFE_PORT_ID_QUATERNARY_PCM_RX:
+	case AFE_PORT_ID_QUATERNARY_PCM_TX:
+		cfg_type = AFE_PARAM_ID_PCM_CONFIG;
+		break;
 	case AFE_PORT_ID_PRIMARY_TDM_RX ... AFE_PORT_ID_QUINARY_TDM_TX_7:
 		cfg_type = AFE_PARAM_ID_TDM_CONFIG;
 		break;
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed5422baff..d3b79e391e44 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -5,7 +5,7 @@
 
 #include <dt-bindings/sound/qcom,q6afe.h>
 
-#define AFE_PORT_MAX		105
+#define AFE_PORT_MAX		113
 
 #define MSM_AFE_PORT_TYPE_RX 0
 #define MSM_AFE_PORT_TYPE_TX 1
@@ -14,7 +14,9 @@
 #define Q6AFE_MAX_MI2S_LINES	4
 
 #define AFE_MAX_CHAN_COUNT	8
+#define AFE_MAX_PCM_SLOT_COUNT	4
 #define AFE_PORT_MAX_AUDIO_CHAN_CNT	0x8
+#define AFE_PORT_MAX_PCM_AUDIO_CHAN_CNT	0x4
 
 #define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
 #define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
@@ -170,6 +172,13 @@ struct q6afe_i2s_cfg {
 	int fmt;
 };
 
+struct q6afe_pcm_cfg {
+	u16	quantype;
+	u32	sample_rate;
+	u16	slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
+	int fmt;
+};
+
 struct q6afe_tdm_cfg {
 	u16	num_channels;
 	u32	sample_rate;
@@ -188,6 +197,7 @@ struct q6afe_port_config {
 	struct q6afe_hdmi_cfg hdmi;
 	struct q6afe_slim_cfg slim;
 	struct q6afe_i2s_cfg i2s_cfg;
+	struct q6afe_pcm_cfg pcm_cfg;
 	struct q6afe_tdm_cfg tdm;
 };
 
@@ -203,6 +213,7 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 void q6afe_slim_port_prepare(struct q6afe_port *port,
 			  struct q6afe_slim_cfg *cfg);
 int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
+int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg);
 void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
 
 int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
-- 
2.21.1

[alsa-devel] [PATCH v3 2/6] ASoC: qdsp6: q6afe: add support to pcm ports

[Adam Serbinski]

This patch adds support to pcm ports in AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6afe.c | 118 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6afe.h |  13 +++-
 2 files changed, 130 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7a58c8..11de01679cad 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -40,6 +40,7 @@
 
 #define AFE_PARAM_ID_SLIMBUS_CONFIG    0x00010212
 #define AFE_PARAM_ID_I2S_CONFIG	0x0001020D
+#define AFE_PARAM_ID_PCM_CONFIG        0x0001020E
 #define AFE_PARAM_ID_TDM_CONFIG	0x0001029D
 #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG	0x00010297
 
@@ -117,6 +118,15 @@
 #define AFE_PORT_ID_QUATERNARY_MI2S_RX      0x1006
 #define AFE_PORT_ID_QUATERNARY_MI2S_TX      0x1007
 
+#define AFE_PORT_ID_PRIMARY_PCM_RX          0x100A
+#define AFE_PORT_ID_PRIMARY_PCM_TX          0x100B
+#define AFE_PORT_ID_SECONDARY_PCM_RX        0x100C
+#define AFE_PORT_ID_SECONDARY_PCM_TX        0x100D
+#define AFE_PORT_ID_TERTIARY_PCM_RX         0x1012
+#define AFE_PORT_ID_TERTIARY_PCM_TX         0x1013
+#define AFE_PORT_ID_QUATERNARY_PCM_RX       0x1014
+#define AFE_PORT_ID_QUATERNARY_PCM_TX       0x1015
+
 /* Start of the range of port IDs for TDM devices. */
 #define AFE_PORT_ID_TDM_PORT_RANGE_START	0x9000
 
@@ -421,6 +431,29 @@ struct afe_digital_clk_cfg {
 	u16                  reserved;
 } __packed;
 
+#define AFE_API_VERSION_PCM_CONFIG		0x1
+#define AFE_PORT_PCM_QUANT_MAX			0x5
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL		0x0
+#define AFE_PORT_PCM_SYNC_SRC_INTERNAL		0x1
+#define AFE_PORT_PCM_AUX_MODE_PCM		0x0
+#define AFE_PORT_PCM_BITS_PER_FRAME_64		0x3
+#define AFE_PORT_PCM_BITS_PER_FRAME_128		0x4
+#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE	0x0
+
+struct afe_param_id_pcm_cfg {
+	u32	pcm_cfg_minor_version;
+	u16	aux_mode;
+	u16	sync_src;
+	u16	frame_setting;
+	u16	quantype;
+	u16	ctrl_data_out_enable;
+	u16	reserved;
+	u32	sample_rate;
+	u16	bit_width;
+	u16	num_channels;
+	u16	slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
+} __packed;
+
 struct afe_param_id_i2s_cfg {
 	u32	i2s_cfg_minor_version;
 	u16	bit_width;
@@ -452,6 +485,7 @@ union afe_port_config {
 	struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
 	struct afe_param_id_slimbus_cfg           slim_cfg;
 	struct afe_param_id_i2s_cfg	i2s_cfg;
+	struct afe_param_id_pcm_cfg	pcm_cfg;
 	struct afe_param_id_tdm_cfg	tdm_cfg;
 } __packed;
 
@@ -707,6 +741,22 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				QUINARY_TDM_TX_7, 0, 1},
 	[DISPLAY_PORT_RX] = { AFE_PORT_ID_HDMI_OVER_DP_RX,
 				DISPLAY_PORT_RX, 1, 1},
+	[PRIMARY_PCM_RX] = { AFE_PORT_ID_PRIMARY_PCM_RX,
+				PRIMARY_PCM_RX, 1, 1},
+	[PRIMARY_PCM_TX] = { AFE_PORT_ID_PRIMARY_PCM_TX,
+				PRIMARY_PCM_RX, 0, 1},
+	[SECONDARY_PCM_RX] = { AFE_PORT_ID_SECONDARY_PCM_RX,
+				SECONDARY_PCM_RX, 1, 1},
+	[SECONDARY_PCM_TX] = { AFE_PORT_ID_SECONDARY_PCM_TX,
+				SECONDARY_PCM_TX, 0, 1},
+	[TERTIARY_PCM_RX] = { AFE_PORT_ID_TERTIARY_PCM_RX,
+				TERTIARY_PCM_RX, 1, 1},
+	[TERTIARY_PCM_TX] = { AFE_PORT_ID_TERTIARY_PCM_TX,
+				TERTIARY_PCM_TX, 0, 1},
+	[QUATERNARY_PCM_RX] = { AFE_PORT_ID_QUATERNARY_PCM_RX,
+				QUATERNARY_PCM_RX, 1, 1},
+	[QUATERNARY_PCM_TX] = { AFE_PORT_ID_QUATERNARY_PCM_TX,
+				QUATERNARY_PCM_TX, 0, 1},
 };
 
 static void q6afe_port_free(struct kref *ref)
@@ -993,6 +1043,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
 		break;
 	case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR:
 	case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+	/* TDM cases overlap with PCM */
 	case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT:
 		cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET;
 		cset.clk_id = clk_id;
@@ -1145,6 +1196,63 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 }
 EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
 
+/**
+ * q6afe_pcm_port_prepare() - Prepare pcm afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: PCM configuration for the afe port
+ *
+ */
+int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg)
+{
+	union afe_port_config *pcfg = &port->port_cfg;
+	struct device *dev = port->afe->dev;
+
+	pcfg->pcm_cfg.pcm_cfg_minor_version = AFE_API_VERSION_PCM_CONFIG;
+
+	if (cfg->quantype > AFE_PORT_PCM_QUANT_MAX) {
+		dev_err(dev, "Invalid pcm quantization\n");
+		return -EINVAL;
+	}
+	pcfg->pcm_cfg.quantype = cfg->quantype;
+
+	memcpy(pcfg->pcm_cfg.slot_number_mapping, cfg->slot_number_mapping,
+	       sizeof(u16) * AFE_PORT_MAX_PCM_AUDIO_CHAN_CNT);
+
+	pcfg->pcm_cfg.aux_mode = AFE_PORT_PCM_AUX_MODE_PCM;
+	switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL;
+		break;
+	default:
+		dev_err(dev, "Unsupported format\n");
+		return -EINVAL;
+	}
+
+	switch (cfg->sample_rate) {
+	case 8000:
+		pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128;
+		break;
+	case 16000:
+		pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64;
+		break;
+	default:
+		dev_err(dev, "Unsupported sample rate\n");
+		return -EINVAL;
+	}
+	pcfg->pcm_cfg.ctrl_data_out_enable = AFE_PORT_PCM_CTRL_DATA_OE_DISABLE;
+	pcfg->pcm_cfg.reserved = 0;
+	pcfg->pcm_cfg.sample_rate = cfg->sample_rate;
+	pcfg->pcm_cfg.bit_width = 16;
+	pcfg->pcm_cfg.num_channels = 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(q6afe_pcm_port_prepare);
+
 /**
  * q6afe_i2s_port_prepare() - Prepare i2s afe port.
  *
@@ -1417,6 +1525,16 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_QUATERNARY_MI2S_TX:
 		cfg_type = AFE_PARAM_ID_I2S_CONFIG;
 		break;
+	case AFE_PORT_ID_PRIMARY_PCM_RX:
+	case AFE_PORT_ID_PRIMARY_PCM_TX:
+	case AFE_PORT_ID_SECONDARY_PCM_RX:
+	case AFE_PORT_ID_SECONDARY_PCM_TX:
+	case AFE_PORT_ID_TERTIARY_PCM_RX:
+	case AFE_PORT_ID_TERTIARY_PCM_TX:
+	case AFE_PORT_ID_QUATERNARY_PCM_RX:
+	case AFE_PORT_ID_QUATERNARY_PCM_TX:
+		cfg_type = AFE_PARAM_ID_PCM_CONFIG;
+		break;
 	case AFE_PORT_ID_PRIMARY_TDM_RX ... AFE_PORT_ID_QUINARY_TDM_TX_7:
 		cfg_type = AFE_PARAM_ID_TDM_CONFIG;
 		break;
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed5422baff..d3b79e391e44 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -5,7 +5,7 @@
 
 #include <dt-bindings/sound/qcom,q6afe.h>
 
-#define AFE_PORT_MAX		105
+#define AFE_PORT_MAX		113
 
 #define MSM_AFE_PORT_TYPE_RX 0
 #define MSM_AFE_PORT_TYPE_TX 1
@@ -14,7 +14,9 @@
 #define Q6AFE_MAX_MI2S_LINES	4
 
 #define AFE_MAX_CHAN_COUNT	8
+#define AFE_MAX_PCM_SLOT_COUNT	4
 #define AFE_PORT_MAX_AUDIO_CHAN_CNT	0x8
+#define AFE_PORT_MAX_PCM_AUDIO_CHAN_CNT	0x4
 
 #define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
 #define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
@@ -170,6 +172,13 @@ struct q6afe_i2s_cfg {
 	int fmt;
 };
 
+struct q6afe_pcm_cfg {
+	u16	quantype;
+	u32	sample_rate;
+	u16	slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
+	int fmt;
+};
+
 struct q6afe_tdm_cfg {
 	u16	num_channels;
 	u32	sample_rate;
@@ -188,6 +197,7 @@ struct q6afe_port_config {
 	struct q6afe_hdmi_cfg hdmi;
 	struct q6afe_slim_cfg slim;
 	struct q6afe_i2s_cfg i2s_cfg;
+	struct q6afe_pcm_cfg pcm_cfg;
 	struct q6afe_tdm_cfg tdm;
 };
 
@@ -203,6 +213,7 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 void q6afe_slim_port_prepare(struct q6afe_port *port,
 			  struct q6afe_slim_cfg *cfg);
 int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
+int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg);
 void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
 
 int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
-- 
2.21.1

_______________________________________________
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Alsa-devel@alsa-project.org
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[PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Adam Serbinski]

This patch adds support of AFE DAI for PCM port.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 210 ++++++++++++++++++++++++++++++-
 1 file changed, 205 insertions(+), 5 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624eaf17..2deb8d6da462 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -63,6 +63,8 @@ struct q6afe_dai_priv_data {
 	uint32_t invert_sync;
 	uint32_t data_delay;
 	uint32_t data_align;
+	uint32_t quantype;
+	uint32_t slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
 };
 
 struct q6afe_dai_data {
@@ -151,6 +153,52 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int q6pcm_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
+{
+	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg;
+	int i = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+		pcm->sample_rate = params_rate(params);
+		break;
+	default:
+		dev_err(dai->dev, "invalid sample rate = %u\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	pcm->quantype = dai_data->priv[dai->id].quantype;
+
+	for (i = 0; i < AFE_MAX_PCM_SLOT_COUNT; i++)
+		pcm->slot_number_mapping[i] =
+			dai_data->priv[dai->id].slot_number_mapping[i];
+
+	return 0;
+}
+
+static int q6pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFM:
+		pcm->fmt = fmt;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported format = %u\n", fmt);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int q6i2s_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
@@ -358,6 +406,15 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
 			return rc;
 		}
 		break;
+	case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX:
+		rc = q6afe_pcm_port_prepare(dai_data->port[dai->id],
+				&dai_data->port_config[dai->id].pcm_cfg);
+		if (rc < 0) {
+			dev_err(dai->dev, "fail to prepare AFE port %x\n",
+				dai->id);
+			return rc;
+		}
+		break;
 	case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
 		q6afe_tdm_port_prepare(dai_data->port[dai->id],
 					&dai_data->port_config[dai->id].tdm);
@@ -429,16 +486,25 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai,
 					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
 					     freq, dir);
 	case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR:
+	case Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR:
+		switch (dai->id) {
+		case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
+			return q6afe_port_set_sysclk(port, clk_id,
+				Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO,
+				Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+				freq, dir);
+			break;
+		default:
+			return q6afe_port_set_sysclk(port, clk_id,
+				Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO,
+				Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+				freq, dir);
+		}
 	case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
 		return q6afe_port_set_sysclk(port, clk_id,
 					     Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO,
 					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
 					     freq, dir);
-	case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT:
-		return q6afe_port_set_sysclk(port, clk_id,
-					     Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO,
-					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
-					     freq, dir);
 	}
 
 	return 0;
@@ -468,6 +534,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
 	{"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
 
+	{"Primary PCM Playback", NULL, "PRI_PCM_RX"},
+	{"Secondary PCM Playback", NULL, "SEC_PCM_RX"},
+	{"Tertiary PCM Playback", NULL, "TERT_PCM_RX"},
+	{"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"},
+
 	{"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"},
 	{"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"},
 	{"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"},
@@ -562,6 +633,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"PRI_MI2S_TX", NULL, "Primary MI2S Capture"},
 	{"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"},
 	{"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"},
+
+	{"PRI_PCM_TX", NULL, "Primary PCM Capture"},
+	{"SEC_PCM_TX", NULL, "Secondary PCM Capture"},
+	{"TERT_PCM_TX", NULL, "Tertiary PCM Capture"},
+	{"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"},
 };
 
 static const struct snd_soc_dai_ops q6hdmi_ops = {
@@ -578,6 +654,14 @@ static const struct snd_soc_dai_ops q6i2s_ops = {
 	.set_sysclk	= q6afe_mi2s_set_sysclk,
 };
 
+static const struct snd_soc_dai_ops q6pcm_ops = {
+	.prepare	= q6afe_dai_prepare,
+	.hw_params	= q6pcm_hw_params,
+	.set_fmt	= q6pcm_set_fmt,
+	.shutdown	= q6afe_dai_shutdown,
+	.set_sysclk	= q6afe_mi2s_set_sysclk,
+};
+
 static const struct snd_soc_dai_ops q6slim_ops = {
 	.prepare	= q6afe_dai_prepare,
 	.hw_params	= q6slim_hw_params,
@@ -1012,6 +1096,94 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.ops = &q6i2s_ops,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Primary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = PRIMARY_PCM_RX,
+		.name = "PRI_PCM_RX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Primary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = PRIMARY_PCM_TX,
+		.name = "PRI_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Secondary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "SEC_PCM_RX",
+		.id = SECONDARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Secondary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = SECONDARY_PCM_TX,
+		.name = "SEC_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Tertiary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "TERT_PCM_RX",
+		.id = TERTIARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Tertiary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = TERTIARY_PCM_TX,
+		.name = "TERT_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Quaternary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "QUAT_PCM_RX",
+		.id = QUATERNARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Quaternary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = QUATERNARY_PCM_TX,
+		.name = "QUAT_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
 	},
 	Q6AFE_TDM_PB_DAI("Primary", 0, PRIMARY_TDM_RX_0),
 	Q6AFE_TDM_PB_DAI("Primary", 1, PRIMARY_TDM_RX_1),
@@ -1169,6 +1341,15 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
 						0, 0, 0, 0),
 
+	SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL, 0, 0, 0, 0),
+
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
@@ -1381,6 +1562,25 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 			for (i = 0; i < num_lines; i++)
 				priv->sd_line_mask |= BIT(lines[i]);
 
+			break;
+		case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX:
+			priv = &data->priv[id];
+			ret = of_property_read_u32(node, "qcom,pcm-quantype",
+						   &priv->quantype);
+			if (ret) {
+				dev_err(dev, "No quantype from DT\n");
+				break;
+			}
+
+			ret = of_property_read_variable_u32_array(node,
+						"qcom,pcm-slot-mapping",
+						priv->slot_number_mapping,
+						1, AFE_MAX_PCM_SLOT_COUNT);
+			if (ret < 0) {
+				dev_err(dev, "No slot mapping from DT\n");
+				break;
+			}
+
 			break;
 		case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
 			priv = &data->priv[id];
-- 
2.21.1

[alsa-devel] [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Adam Serbinski]

This patch adds support of AFE DAI for PCM port.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 210 ++++++++++++++++++++++++++++++-
 1 file changed, 205 insertions(+), 5 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624eaf17..2deb8d6da462 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -63,6 +63,8 @@ struct q6afe_dai_priv_data {
 	uint32_t invert_sync;
 	uint32_t data_delay;
 	uint32_t data_align;
+	uint32_t quantype;
+	uint32_t slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT];
 };
 
 struct q6afe_dai_data {
@@ -151,6 +153,52 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int q6pcm_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
+{
+	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg;
+	int i = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+		pcm->sample_rate = params_rate(params);
+		break;
+	default:
+		dev_err(dai->dev, "invalid sample rate = %u\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	pcm->quantype = dai_data->priv[dai->id].quantype;
+
+	for (i = 0; i < AFE_MAX_PCM_SLOT_COUNT; i++)
+		pcm->slot_number_mapping[i] =
+			dai_data->priv[dai->id].slot_number_mapping[i];
+
+	return 0;
+}
+
+static int q6pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFM:
+		pcm->fmt = fmt;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported format = %u\n", fmt);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int q6i2s_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
@@ -358,6 +406,15 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
 			return rc;
 		}
 		break;
+	case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX:
+		rc = q6afe_pcm_port_prepare(dai_data->port[dai->id],
+				&dai_data->port_config[dai->id].pcm_cfg);
+		if (rc < 0) {
+			dev_err(dai->dev, "fail to prepare AFE port %x\n",
+				dai->id);
+			return rc;
+		}
+		break;
 	case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
 		q6afe_tdm_port_prepare(dai_data->port[dai->id],
 					&dai_data->port_config[dai->id].tdm);
@@ -429,16 +486,25 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai,
 					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
 					     freq, dir);
 	case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR:
+	case Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR:
+		switch (dai->id) {
+		case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
+			return q6afe_port_set_sysclk(port, clk_id,
+				Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO,
+				Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+				freq, dir);
+			break;
+		default:
+			return q6afe_port_set_sysclk(port, clk_id,
+				Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO,
+				Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+				freq, dir);
+		}
 	case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
 		return q6afe_port_set_sysclk(port, clk_id,
 					     Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO,
 					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
 					     freq, dir);
-	case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT:
-		return q6afe_port_set_sysclk(port, clk_id,
-					     Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO,
-					     Q6AFE_LPASS_CLK_ROOT_DEFAULT,
-					     freq, dir);
 	}
 
 	return 0;
@@ -468,6 +534,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
 	{"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
 
+	{"Primary PCM Playback", NULL, "PRI_PCM_RX"},
+	{"Secondary PCM Playback", NULL, "SEC_PCM_RX"},
+	{"Tertiary PCM Playback", NULL, "TERT_PCM_RX"},
+	{"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"},
+
 	{"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"},
 	{"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"},
 	{"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"},
@@ -562,6 +633,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"PRI_MI2S_TX", NULL, "Primary MI2S Capture"},
 	{"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"},
 	{"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"},
+
+	{"PRI_PCM_TX", NULL, "Primary PCM Capture"},
+	{"SEC_PCM_TX", NULL, "Secondary PCM Capture"},
+	{"TERT_PCM_TX", NULL, "Tertiary PCM Capture"},
+	{"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"},
 };
 
 static const struct snd_soc_dai_ops q6hdmi_ops = {
@@ -578,6 +654,14 @@ static const struct snd_soc_dai_ops q6i2s_ops = {
 	.set_sysclk	= q6afe_mi2s_set_sysclk,
 };
 
+static const struct snd_soc_dai_ops q6pcm_ops = {
+	.prepare	= q6afe_dai_prepare,
+	.hw_params	= q6pcm_hw_params,
+	.set_fmt	= q6pcm_set_fmt,
+	.shutdown	= q6afe_dai_shutdown,
+	.set_sysclk	= q6afe_mi2s_set_sysclk,
+};
+
 static const struct snd_soc_dai_ops q6slim_ops = {
 	.prepare	= q6afe_dai_prepare,
 	.hw_params	= q6slim_hw_params,
@@ -1012,6 +1096,94 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.ops = &q6i2s_ops,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Primary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = PRIMARY_PCM_RX,
+		.name = "PRI_PCM_RX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Primary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = PRIMARY_PCM_TX,
+		.name = "PRI_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Secondary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "SEC_PCM_RX",
+		.id = SECONDARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Secondary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = SECONDARY_PCM_TX,
+		.name = "SEC_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Tertiary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "TERT_PCM_RX",
+		.id = TERTIARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Tertiary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = TERTIARY_PCM_TX,
+		.name = "TERT_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.playback = {
+			.stream_name = "Quaternary PCM Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.name = "QUAT_PCM_RX",
+		.id = QUATERNARY_PCM_RX,
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.capture = {
+			.stream_name = "Quaternary PCM Capture",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.id = QUATERNARY_PCM_TX,
+		.name = "QUAT_PCM_TX",
+		.ops = &q6pcm_ops,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
 	},
 	Q6AFE_TDM_PB_DAI("Primary", 0, PRIMARY_TDM_RX_0),
 	Q6AFE_TDM_PB_DAI("Primary", 1, PRIMARY_TDM_RX_1),
@@ -1169,6 +1341,15 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
 						0, 0, 0, 0),
 
+	SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL, 0, 0, 0, 0),
+
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
@@ -1381,6 +1562,25 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 			for (i = 0; i < num_lines; i++)
 				priv->sd_line_mask |= BIT(lines[i]);
 
+			break;
+		case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX:
+			priv = &data->priv[id];
+			ret = of_property_read_u32(node, "qcom,pcm-quantype",
+						   &priv->quantype);
+			if (ret) {
+				dev_err(dev, "No quantype from DT\n");
+				break;
+			}
+
+			ret = of_property_read_variable_u32_array(node,
+						"qcom,pcm-slot-mapping",
+						priv->slot_number_mapping,
+						1, AFE_MAX_PCM_SLOT_COUNT);
+			if (ret < 0) {
+				dev_err(dev, "No slot mapping from DT\n");
+				break;
+			}
+
 			break;
 		case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7:
 			priv = &data->priv[id];
-- 
2.21.1

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Re: [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Srinivas Kandagatla]

On 12/02/2020 01:52, Adam Serbinski wrote:
> +			ret = of_property_read_u32(node, "qcom,pcm-quantype",
> +						   &priv->quantype);
> +			if (ret) {
> +				dev_err(dev, "No quantype from DT\n");
> +				break;
> +			}
> +
> +			ret = of_property_read_variable_u32_array(node,
> +						"qcom,pcm-slot-mapping",
> +						priv->slot_number_mapping,
> +						1, AFE_MAX_PCM_SLOT_COUNT);

These new bindings needs to be documented in 
Documentation/devicetree/bindings/sound/qcom,q6afe.txt

--srini

Re: [alsa-devel] [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Srinivas Kandagatla]

On 12/02/2020 01:52, Adam Serbinski wrote:
> +			ret = of_property_read_u32(node, "qcom,pcm-quantype",
> +						   &priv->quantype);
> +			if (ret) {
> +				dev_err(dev, "No quantype from DT\n");
> +				break;
> +			}
> +
> +			ret = of_property_read_variable_u32_array(node,
> +						"qcom,pcm-slot-mapping",
> +						priv->slot_number_mapping,
> +						1, AFE_MAX_PCM_SLOT_COUNT);

These new bindings needs to be documented in 
Documentation/devicetree/bindings/sound/qcom,q6afe.txt

--srini
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Re: [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Srinivas Kandagatla]

On 12/02/2020 09:52, Srinivas Kandagatla wrote:
> 
> 
> On 12/02/2020 01:52, Adam Serbinski wrote:
>> +            ret = of_property_read_u32(node, "qcom,pcm-quantype",
>> +                           &priv->quantype);
>> +            if (ret) {
>> +                dev_err(dev, "No quantype from DT\n");
>> +                break;
>> +            }
>> +
>> +            ret = of_property_read_variable_u32_array(node,
>> +                        "qcom,pcm-slot-mapping",
>> +                        priv->slot_number_mapping,
>> +                        1, AFE_MAX_PCM_SLOT_COUNT);
> 
> These new bindings needs to be documented in 
> Documentation/devicetree/bindings/sound/qcom,q6afe.txt

Replied too quickly!

I would normally have expected bindings patch before this patch though..

--srini

> 
> --srini

Re: [alsa-devel] [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais

[Srinivas Kandagatla]

On 12/02/2020 09:52, Srinivas Kandagatla wrote:
> 
> 
> On 12/02/2020 01:52, Adam Serbinski wrote:
>> +            ret = of_property_read_u32(node, "qcom,pcm-quantype",
>> +                           &priv->quantype);
>> +            if (ret) {
>> +                dev_err(dev, "No quantype from DT\n");
>> +                break;
>> +            }
>> +
>> +            ret = of_property_read_variable_u32_array(node,
>> +                        "qcom,pcm-slot-mapping",
>> +                        priv->slot_number_mapping,
>> +                        1, AFE_MAX_PCM_SLOT_COUNT);
> 
> These new bindings needs to be documented in 
> Documentation/devicetree/bindings/sound/qcom,q6afe.txt

Replied too quickly!

I would normally have expected bindings patch before this patch though..

--srini

> 
> --srini
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[PATCH v3 4/6] ASoC: qdsp6: q6routing: add pcm port routing

[Adam Serbinski]

This patch adds support to PCM_PORT mixers required to
select path between ASM stream and AFE ports.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++
 1 file changed, 44 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 20724102e85a..3a81d2161707 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -67,6 +67,10 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "PRI_PCM_TX", "PRI_PCM_TX" },		\
+	{ mix_name, "SEC_PCM_TX", "SEC_PCM_TX" },		\
+	{ mix_name, "TERT_PCM_TX", "TERT_PCM_TX" },		\
+	{ mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" },		\
 	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
 	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
 	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
@@ -128,6 +132,18 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX,		\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
@@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
 static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
 	Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) };
 
+static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) };
+
+static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) };
+
+static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) };
+
+static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) };
+
 static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
 	Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) };
 
@@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
 			   tertiary_mi2s_rx_mixer_controls,
 			   ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   primary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(primary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   secondary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(secondary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   tertiary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   quaternary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)),
 	SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0,
 				pri_tdm_rx_0_mixer_controls,
 				ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)),
@@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer",
 				"PRIMARY_TDM_RX_0"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer",
-- 
2.21.1

[alsa-devel] [PATCH v3 4/6] ASoC: qdsp6: q6routing: add pcm port routing

[Adam Serbinski]

This patch adds support to PCM_PORT mixers required to
select path between ASM stream and AFE ports.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++
 1 file changed, 44 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 20724102e85a..3a81d2161707 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -67,6 +67,10 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "PRI_PCM_TX", "PRI_PCM_TX" },		\
+	{ mix_name, "SEC_PCM_TX", "SEC_PCM_TX" },		\
+	{ mix_name, "TERT_PCM_TX", "TERT_PCM_TX" },		\
+	{ mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" },		\
 	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
 	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
 	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
@@ -128,6 +132,18 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX,		\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
@@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
 static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
 	Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) };
 
+static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) };
+
+static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) };
+
+static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) };
+
+static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) };
+
 static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
 	Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) };
 
@@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
 			   tertiary_mi2s_rx_mixer_controls,
 			   ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   primary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(primary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   secondary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(secondary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   tertiary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)),
+	SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   quaternary_pcm_rx_mixer_controls,
+			   ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)),
 	SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0,
 				pri_tdm_rx_0_mixer_controls,
 				ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)),
@@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"),
+	Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer",
 				"PRIMARY_TDM_RX_0"),
 	Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer",
-- 
2.21.1

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Alsa-devel@alsa-project.org
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[PATCH v3 5/6] ASoC: qcom: apq8096: add support for primary and quaternary I2S/PCM

[Adam Serbinski]

This adds support to primary and quarternary I2S and PCM ports.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/apq8096.c | 86 +++++++++++++++++++++++++++++++++-------
 1 file changed, 71 insertions(+), 15 deletions(-)

diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd6846a..0dd58cdf34dd 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -8,24 +8,13 @@
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
 #include "common.h"
+#include "qdsp6/q6afe.h"
 
 #define SLIM_MAX_TX_PORTS 16
 #define SLIM_MAX_RX_PORTS 16
 #define WCD9335_DEFAULT_MCLK_RATE	9600000
-
-static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
-				      struct snd_pcm_hw_params *params)
-{
-	struct snd_interval *rate = hw_param_interval(params,
-					SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
-					SNDRV_PCM_HW_PARAM_CHANNELS);
-
-	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
-
-	return 0;
-}
+#define MI2S_BCLK_RATE			1536000
+#define PCM_BCLK_RATE			1024000
 
 static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
@@ -33,10 +22,32 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_interval *rate = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
 	u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
 	u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
 	int ret = 0;
 
+	switch (cpu_dai->id) {
+	case PRIMARY_PCM_RX:
+	case PRIMARY_PCM_TX:
+	case QUATERNARY_PCM_RX:
+	case QUATERNARY_PCM_TX:
+		rate->min = 16000;
+		rate->max = 16000;
+		channels->min = 1;
+		channels->max = 1;
+		break;
+	default:
+		rate->min = 48000;
+		rate->max = 48000;
+		channels->min = 2;
+		channels->max = 2;
+		break;
+	}
+
 	ret = snd_soc_dai_get_channel_map(codec_dai,
 				&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
 	if (ret != 0 && ret != -ENOTSUPP) {
@@ -60,8 +71,54 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int msm_snd_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+			MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case QUATERNARY_MI2S_RX:
+	case QUATERNARY_MI2S_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT,
+			MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case PRIMARY_PCM_RX:
+	case PRIMARY_PCM_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT,
+			PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case QUATERNARY_PCM_RX:
+	case QUATERNARY_PCM_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT,
+			PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	default:
+		return -1;
+	}
+	return 0;
+}
+
 static struct snd_soc_ops apq8096_ops = {
 	.hw_params = msm_snd_hw_params,
+	.startup = msm_snd_startup,
 };
 
 static int apq8096_init(struct snd_soc_pcm_runtime *rtd)
@@ -96,7 +153,6 @@ static void apq8096_add_be_ops(struct snd_soc_card *card)
 
 	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1) {
-			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
 			link->init = apq8096_init;
 			link->ops = &apq8096_ops;
 		}
-- 
2.21.1

[alsa-devel] [PATCH v3 5/6] ASoC: qcom: apq8096: add support for primary and quaternary I2S/PCM

[Adam Serbinski]

This adds support to primary and quarternary I2S and PCM ports.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 sound/soc/qcom/apq8096.c | 86 +++++++++++++++++++++++++++++++++-------
 1 file changed, 71 insertions(+), 15 deletions(-)

diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd6846a..0dd58cdf34dd 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -8,24 +8,13 @@
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
 #include "common.h"
+#include "qdsp6/q6afe.h"
 
 #define SLIM_MAX_TX_PORTS 16
 #define SLIM_MAX_RX_PORTS 16
 #define WCD9335_DEFAULT_MCLK_RATE	9600000
-
-static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
-				      struct snd_pcm_hw_params *params)
-{
-	struct snd_interval *rate = hw_param_interval(params,
-					SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
-					SNDRV_PCM_HW_PARAM_CHANNELS);
-
-	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
-
-	return 0;
-}
+#define MI2S_BCLK_RATE			1536000
+#define PCM_BCLK_RATE			1024000
 
 static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
@@ -33,10 +22,32 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_interval *rate = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
 	u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
 	u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
 	int ret = 0;
 
+	switch (cpu_dai->id) {
+	case PRIMARY_PCM_RX:
+	case PRIMARY_PCM_TX:
+	case QUATERNARY_PCM_RX:
+	case QUATERNARY_PCM_TX:
+		rate->min = 16000;
+		rate->max = 16000;
+		channels->min = 1;
+		channels->max = 1;
+		break;
+	default:
+		rate->min = 48000;
+		rate->max = 48000;
+		channels->min = 2;
+		channels->max = 2;
+		break;
+	}
+
 	ret = snd_soc_dai_get_channel_map(codec_dai,
 				&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
 	if (ret != 0 && ret != -ENOTSUPP) {
@@ -60,8 +71,54 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int msm_snd_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+			MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case QUATERNARY_MI2S_RX:
+	case QUATERNARY_MI2S_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT,
+			MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case PRIMARY_PCM_RX:
+	case PRIMARY_PCM_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT,
+			PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	case QUATERNARY_PCM_RX:
+	case QUATERNARY_PCM_TX:
+		snd_soc_dai_set_sysclk(cpu_dai,
+			Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT,
+			PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+		snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS);
+		break;
+	default:
+		return -1;
+	}
+	return 0;
+}
+
 static struct snd_soc_ops apq8096_ops = {
 	.hw_params = msm_snd_hw_params,
+	.startup = msm_snd_startup,
 };
 
 static int apq8096_init(struct snd_soc_pcm_runtime *rtd)
@@ -96,7 +153,6 @@ static void apq8096_add_be_ops(struct snd_soc_card *card)
 
 	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1) {
-			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
 			link->init = apq8096_init;
 			link->ops = &apq8096_ops;
 		}
-- 
2.21.1

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[PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

This patch adds documentation of bindings required for PCM ports on AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 +++++++++++++++++++
 1 file changed, 42 insertions(+)

diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
index d74888b9f1bb..6b1b17d31a2a 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties.
 	Definition: Must be list of serial data lines used by this dai.
 	should be one or more of the 0-3 sd lines.
 
+ - qcom,pcm-quantype
+	Usage: required for pcm interface
+	Value type: <u32>
+	Definition: PCM quantization type
+		0 - ALAW, no padding
+		1 - MULAW, no padding
+		2 - Linear, no padding
+		3 - ALAW, padding
+		4 - MULAW, padding
+		5 - Linear, padding
+
+ - qcom,pcm-slot-mapping
+	Usage: required for pcm interface
+	Value type: <prop-encoded-array>
+	Definition: Slot mapping for audio channels. Array size is the number
+		of slots, minimum 1, maximum 4. The value is 0 for no mapping
+		to the slot, or the channel number from 1 to 32.
+
  - qcom,tdm-sync-mode:
 	Usage: required for tdm interface
 	Value type: <prop-encoded-array>
@@ -174,5 +192,29 @@ q6afe@4 {
 			reg = <23>;
 			qcom,sd-lines = <1>;
 		};
+
+		pri-pcm-rx@105 {
+			reg = <105>;
+			qcom,pcm-quantype = <2>;
+			qcom,pcm-slot-mapping = <1>;
+		};
+
+		pri-pcm-tx@106 {
+			reg = <106>;
+			qcom,pcm-quantype = <2>;
+			qcom,pcm-slot-mapping = <1>;
+		};
+
+		quat-pcm-rx@111 {
+			reg = <111>;
+			qcom,pcm-quantype = <5>;
+			qcom,pcm-slot-mapping = <0 0 1>;
+		};
+
+		quat-pcm-tx@112 {
+			reg = <112>;
+			qcom,pcm-quantype = <5>;
+			qcom,pcm-slot-mapping = <0 0 1>;
+		};
 	};
 };
-- 
2.21.1

[alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

This patch adds documentation of bindings required for PCM ports on AFE.

Signed-off-by: Adam Serbinski <adam@serbinski.com>
CC: Andy Gross <agross@kernel.org>
CC: Mark Rutland <mark.rutland@arm.com>
CC: Liam Girdwood <lgirdwood@gmail.com>
CC: Patrick Lai <plai@codeaurora.org>
CC: Banajit Goswami <bgoswami@codeaurora.org>
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: alsa-devel@alsa-project.org
CC: linux-arm-msm@vger.kernel.org
CC: devicetree@vger.kernel.org
CC: linux-kernel@vger.kernel.org
---
 .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 +++++++++++++++++++
 1 file changed, 42 insertions(+)

diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
index d74888b9f1bb..6b1b17d31a2a 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties.
 	Definition: Must be list of serial data lines used by this dai.
 	should be one or more of the 0-3 sd lines.
 
+ - qcom,pcm-quantype
+	Usage: required for pcm interface
+	Value type: <u32>
+	Definition: PCM quantization type
+		0 - ALAW, no padding
+		1 - MULAW, no padding
+		2 - Linear, no padding
+		3 - ALAW, padding
+		4 - MULAW, padding
+		5 - Linear, padding
+
+ - qcom,pcm-slot-mapping
+	Usage: required for pcm interface
+	Value type: <prop-encoded-array>
+	Definition: Slot mapping for audio channels. Array size is the number
+		of slots, minimum 1, maximum 4. The value is 0 for no mapping
+		to the slot, or the channel number from 1 to 32.
+
  - qcom,tdm-sync-mode:
 	Usage: required for tdm interface
 	Value type: <prop-encoded-array>
@@ -174,5 +192,29 @@ q6afe@4 {
 			reg = <23>;
 			qcom,sd-lines = <1>;
 		};
+
+		pri-pcm-rx@105 {
+			reg = <105>;
+			qcom,pcm-quantype = <2>;
+			qcom,pcm-slot-mapping = <1>;
+		};
+
+		pri-pcm-tx@106 {
+			reg = <106>;
+			qcom,pcm-quantype = <2>;
+			qcom,pcm-slot-mapping = <1>;
+		};
+
+		quat-pcm-rx@111 {
+			reg = <111>;
+			qcom,pcm-quantype = <5>;
+			qcom,pcm-slot-mapping = <0 0 1>;
+		};
+
+		quat-pcm-tx@112 {
+			reg = <112>;
+			qcom,pcm-quantype = <5>;
+			qcom,pcm-slot-mapping = <0 0 1>;
+		};
 	};
 };
-- 
2.21.1

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Re: [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Srinivas Kandagatla]

On 12/02/2020 01:52, Adam Serbinski wrote:
> This patch adds documentation of bindings required for PCM ports on AFE.
> 
> Signed-off-by: Adam Serbinski <adam@serbinski.com>
> CC: Andy Gross <agross@kernel.org>
> CC: Mark Rutland <mark.rutland@arm.com>
> CC: Liam Girdwood <lgirdwood@gmail.com>
> CC: Patrick Lai <plai@codeaurora.org>
> CC: Banajit Goswami <bgoswami@codeaurora.org>
> CC: Jaroslav Kysela <perex@perex.cz>
> CC: Takashi Iwai <tiwai@suse.com>
> CC: alsa-devel@alsa-project.org
> CC: linux-arm-msm@vger.kernel.org
> CC: devicetree@vger.kernel.org
> CC: linux-kernel@vger.kernel.org
> ---
>   .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 +++++++++++++++++++
>   1 file changed, 42 insertions(+)
> 
> diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> index d74888b9f1bb..6b1b17d31a2a 100644
> --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> @@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties.
>   	Definition: Must be list of serial data lines used by this dai.
>   	should be one or more of the 0-3 sd lines.
>   
> + - qcom,pcm-quantype
> +	Usage: required for pcm interface
> +	Value type: <u32>
> +	Definition: PCM quantization type
> +		0 - ALAW, no padding
> +		1 - MULAW, no padding
> +		2 - Linear, no padding
> +		3 - ALAW, padding
> +		4 - MULAW, padding
> +		5 - Linear, padding
> +
> + - qcom,pcm-slot-mapping
> +	Usage: required for pcm interface

Are these not specific to 8k and 16k mode ?
We should probably list values for both modes here.

--srini

> +	Value type: <prop-encoded-array>
> +	Definition: Slot mapping for audio channels. Array size is the number
> +		of slots, minimum 1, maximum 4. The value is 0 for no mapping
> +		to the slot, or the channel number from 1 to 32.
> +
>    - qcom,tdm-sync-mode:
>   	Usage: required for tdm interface
>   	Value type: <prop-encoded-array>
> @@ -174,5 +192,29 @@ q6afe@4 {
>   			reg = <23>;
>   			qcom,sd-lines = <1>;
>   		};
> +
> +		pri-pcm-rx@105 {
> +			reg = <105>;
> +			qcom,pcm-quantype = <2>;
> +			qcom,pcm-slot-mapping = <1>;
> +		};
> +
> +		pri-pcm-tx@106 {
> +			reg = <106>;
> +			qcom,pcm-quantype = <2>;
> +			qcom,pcm-slot-mapping = <1>;
> +		};
> +
> +		quat-pcm-rx@111 {
> +			reg = <111>;
> +			qcom,pcm-quantype = <5>;
> +			qcom,pcm-slot-mapping = <0 0 1>;
> +		};
> +
> +		quat-pcm-tx@112 {
> +			reg = <112>;
> +			qcom,pcm-quantype = <5>;
> +			qcom,pcm-slot-mapping = <0 0 1>;
> +		};
>   	};
>   };
> 

Re: [alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Srinivas Kandagatla]

On 12/02/2020 01:52, Adam Serbinski wrote:
> This patch adds documentation of bindings required for PCM ports on AFE.
> 
> Signed-off-by: Adam Serbinski <adam@serbinski.com>
> CC: Andy Gross <agross@kernel.org>
> CC: Mark Rutland <mark.rutland@arm.com>
> CC: Liam Girdwood <lgirdwood@gmail.com>
> CC: Patrick Lai <plai@codeaurora.org>
> CC: Banajit Goswami <bgoswami@codeaurora.org>
> CC: Jaroslav Kysela <perex@perex.cz>
> CC: Takashi Iwai <tiwai@suse.com>
> CC: alsa-devel@alsa-project.org
> CC: linux-arm-msm@vger.kernel.org
> CC: devicetree@vger.kernel.org
> CC: linux-kernel@vger.kernel.org
> ---
>   .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 +++++++++++++++++++
>   1 file changed, 42 insertions(+)
> 
> diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> index d74888b9f1bb..6b1b17d31a2a 100644
> --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
> @@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties.
>   	Definition: Must be list of serial data lines used by this dai.
>   	should be one or more of the 0-3 sd lines.
>   
> + - qcom,pcm-quantype
> +	Usage: required for pcm interface
> +	Value type: <u32>
> +	Definition: PCM quantization type
> +		0 - ALAW, no padding
> +		1 - MULAW, no padding
> +		2 - Linear, no padding
> +		3 - ALAW, padding
> +		4 - MULAW, padding
> +		5 - Linear, padding
> +
> + - qcom,pcm-slot-mapping
> +	Usage: required for pcm interface

Are these not specific to 8k and 16k mode ?
We should probably list values for both modes here.

--srini

> +	Value type: <prop-encoded-array>
> +	Definition: Slot mapping for audio channels. Array size is the number
> +		of slots, minimum 1, maximum 4. The value is 0 for no mapping
> +		to the slot, or the channel number from 1 to 32.
> +
>    - qcom,tdm-sync-mode:
>   	Usage: required for tdm interface
>   	Value type: <prop-encoded-array>
> @@ -174,5 +192,29 @@ q6afe@4 {
>   			reg = <23>;
>   			qcom,sd-lines = <1>;
>   		};
> +
> +		pri-pcm-rx@105 {
> +			reg = <105>;
> +			qcom,pcm-quantype = <2>;
> +			qcom,pcm-slot-mapping = <1>;
> +		};
> +
> +		pri-pcm-tx@106 {
> +			reg = <106>;
> +			qcom,pcm-quantype = <2>;
> +			qcom,pcm-slot-mapping = <1>;
> +		};
> +
> +		quat-pcm-rx@111 {
> +			reg = <111>;
> +			qcom,pcm-quantype = <5>;
> +			qcom,pcm-slot-mapping = <0 0 1>;
> +		};
> +
> +		quat-pcm-tx@112 {
> +			reg = <112>;
> +			qcom,pcm-quantype = <5>;
> +			qcom,pcm-slot-mapping = <0 0 1>;
> +		};
>   	};
>   };
> 
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Re: [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

On 2020-02-12 04:59, Srinivas Kandagatla wrote:
> On 12/02/2020 01:52, Adam Serbinski wrote:
>> This patch adds documentation of bindings required for PCM ports on 
>> AFE.
>> 
>> Signed-off-by: Adam Serbinski <adam@serbinski.com>
>> CC: Andy Gross <agross@kernel.org>
>> CC: Mark Rutland <mark.rutland@arm.com>
>> CC: Liam Girdwood <lgirdwood@gmail.com>
>> CC: Patrick Lai <plai@codeaurora.org>
>> CC: Banajit Goswami <bgoswami@codeaurora.org>
>> CC: Jaroslav Kysela <perex@perex.cz>
>> CC: Takashi Iwai <tiwai@suse.com>
>> CC: alsa-devel@alsa-project.org
>> CC: linux-arm-msm@vger.kernel.org
>> CC: devicetree@vger.kernel.org
>> CC: linux-kernel@vger.kernel.org
>> ---
>>   .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 
>> +++++++++++++++++++
>>   1 file changed, 42 insertions(+)
>> 
>> diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt 
>> b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> index d74888b9f1bb..6b1b17d31a2a 100644
>> --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> @@ -51,6 +51,24 @@ configuration of each dai. Must contain the 
>> following properties.
>>   	Definition: Must be list of serial data lines used by this dai.
>>   	should be one or more of the 0-3 sd lines.
>>   + - qcom,pcm-quantype
>> +	Usage: required for pcm interface
>> +	Value type: <u32>
>> +	Definition: PCM quantization type
>> +		0 - ALAW, no padding
>> +		1 - MULAW, no padding
>> +		2 - Linear, no padding
>> +		3 - ALAW, padding
>> +		4 - MULAW, padding
>> +		5 - Linear, padding
>> +
>> + - qcom,pcm-slot-mapping
>> +	Usage: required for pcm interface
> 
> Are these not specific to 8k and 16k mode ?
> We should probably list values for both modes here.

No, this is just the offset that the audio sample is placed in with 
respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
pulse.

When switching between 8 and 16k sample rate, it is just the sync pulse 
rate that is changed. The audio sample will be delivered in the same 
slot, just at a different frequency.


>> +	Value type: <prop-encoded-array>
>> +	Definition: Slot mapping for audio channels. Array size is the 
>> number
>> +		of slots, minimum 1, maximum 4. The value is 0 for no mapping
>> +		to the slot, or the channel number from 1 to 32.
>> +
>>    - qcom,tdm-sync-mode:
>>   	Usage: required for tdm interface
>>   	Value type: <prop-encoded-array>
>> @@ -174,5 +192,29 @@ q6afe@4 {
>>   			reg = <23>;
>>   			qcom,sd-lines = <1>;
>>   		};
>> +
>> +		pri-pcm-rx@105 {
>> +			reg = <105>;
>> +			qcom,pcm-quantype = <2>;
>> +			qcom,pcm-slot-mapping = <1>;
>> +		};
>> +
>> +		pri-pcm-tx@106 {
>> +			reg = <106>;
>> +			qcom,pcm-quantype = <2>;
>> +			qcom,pcm-slot-mapping = <1>;
>> +		};
>> +
>> +		quat-pcm-rx@111 {
>> +			reg = <111>;
>> +			qcom,pcm-quantype = <5>;
>> +			qcom,pcm-slot-mapping = <0 0 1>;
>> +		};
>> +
>> +		quat-pcm-tx@112 {
>> +			reg = <112>;
>> +			qcom,pcm-quantype = <5>;
>> +			qcom,pcm-slot-mapping = <0 0 1>;
>> +		};
>>   	};
>>   };
>> 

Re: [alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

On 2020-02-12 04:59, Srinivas Kandagatla wrote:
> On 12/02/2020 01:52, Adam Serbinski wrote:
>> This patch adds documentation of bindings required for PCM ports on 
>> AFE.
>> 
>> Signed-off-by: Adam Serbinski <adam@serbinski.com>
>> CC: Andy Gross <agross@kernel.org>
>> CC: Mark Rutland <mark.rutland@arm.com>
>> CC: Liam Girdwood <lgirdwood@gmail.com>
>> CC: Patrick Lai <plai@codeaurora.org>
>> CC: Banajit Goswami <bgoswami@codeaurora.org>
>> CC: Jaroslav Kysela <perex@perex.cz>
>> CC: Takashi Iwai <tiwai@suse.com>
>> CC: alsa-devel@alsa-project.org
>> CC: linux-arm-msm@vger.kernel.org
>> CC: devicetree@vger.kernel.org
>> CC: linux-kernel@vger.kernel.org
>> ---
>>   .../devicetree/bindings/sound/qcom,q6afe.txt  | 42 
>> +++++++++++++++++++
>>   1 file changed, 42 insertions(+)
>> 
>> diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt 
>> b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> index d74888b9f1bb..6b1b17d31a2a 100644
>> --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
>> @@ -51,6 +51,24 @@ configuration of each dai. Must contain the 
>> following properties.
>>   	Definition: Must be list of serial data lines used by this dai.
>>   	should be one or more of the 0-3 sd lines.
>>   + - qcom,pcm-quantype
>> +	Usage: required for pcm interface
>> +	Value type: <u32>
>> +	Definition: PCM quantization type
>> +		0 - ALAW, no padding
>> +		1 - MULAW, no padding
>> +		2 - Linear, no padding
>> +		3 - ALAW, padding
>> +		4 - MULAW, padding
>> +		5 - Linear, padding
>> +
>> + - qcom,pcm-slot-mapping
>> +	Usage: required for pcm interface
> 
> Are these not specific to 8k and 16k mode ?
> We should probably list values for both modes here.

No, this is just the offset that the audio sample is placed in with 
respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
pulse.

When switching between 8 and 16k sample rate, it is just the sync pulse 
rate that is changed. The audio sample will be delivered in the same 
slot, just at a different frequency.


>> +	Value type: <prop-encoded-array>
>> +	Definition: Slot mapping for audio channels. Array size is the 
>> number
>> +		of slots, minimum 1, maximum 4. The value is 0 for no mapping
>> +		to the slot, or the channel number from 1 to 32.
>> +
>>    - qcom,tdm-sync-mode:
>>   	Usage: required for tdm interface
>>   	Value type: <prop-encoded-array>
>> @@ -174,5 +192,29 @@ q6afe@4 {
>>   			reg = <23>;
>>   			qcom,sd-lines = <1>;
>>   		};
>> +
>> +		pri-pcm-rx@105 {
>> +			reg = <105>;
>> +			qcom,pcm-quantype = <2>;
>> +			qcom,pcm-slot-mapping = <1>;
>> +		};
>> +
>> +		pri-pcm-tx@106 {
>> +			reg = <106>;
>> +			qcom,pcm-quantype = <2>;
>> +			qcom,pcm-slot-mapping = <1>;
>> +		};
>> +
>> +		quat-pcm-rx@111 {
>> +			reg = <111>;
>> +			qcom,pcm-quantype = <5>;
>> +			qcom,pcm-slot-mapping = <0 0 1>;
>> +		};
>> +
>> +		quat-pcm-tx@112 {
>> +			reg = <112>;
>> +			qcom,pcm-quantype = <5>;
>> +			qcom,pcm-slot-mapping = <0 0 1>;
>> +		};
>>   	};
>>   };
>> 
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Re: [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Srinivas Kandagatla]

On 12/02/2020 11:01, Adam Serbinski wrote:
>>>
>>> +
>>> + - qcom,pcm-slot-mapping
>>> +    Usage: required for pcm interface
>>
>> Are these not specific to 8k and 16k mode ?
>> We should probably list values for both modes here.
> 
> No, this is just the offset that the audio sample is placed in with 
> respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
> pulse.


That's not true atleast by the QDSP documentation,
according to it we will use more slots to transfer at higher sample 
rate. ex:
16 kHz data can be transferred using 8 kHz samples in two
slots.

Also there are 32 slots for each of 4 supported channels for PCM AFE port.


> 
> When switching between 8 and 16k sample rate, it is just the sync pulse 
> rate that is changed. The audio sample will be delivered in the same 
> slot, just at a different frequency.

Re: [alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Srinivas Kandagatla]

On 12/02/2020 11:01, Adam Serbinski wrote:
>>>
>>> +
>>> + - qcom,pcm-slot-mapping
>>> +    Usage: required for pcm interface
>>
>> Are these not specific to 8k and 16k mode ?
>> We should probably list values for both modes here.
> 
> No, this is just the offset that the audio sample is placed in with 
> respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
> pulse.


That's not true atleast by the QDSP documentation,
according to it we will use more slots to transfer at higher sample 
rate. ex:
16 kHz data can be transferred using 8 kHz samples in two
slots.

Also there are 32 slots for each of 4 supported channels for PCM AFE port.


> 
> When switching between 8 and 16k sample rate, it is just the sync pulse 
> rate that is changed. The audio sample will be delivered in the same 
> slot, just at a different frequency.
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Re: [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

On 2020-02-12 06:43, Srinivas Kandagatla wrote:
> On 12/02/2020 11:01, Adam Serbinski wrote:
>>>> 
>>>> +
>>>> + - qcom,pcm-slot-mapping
>>>> +    Usage: required for pcm interface
>>> 
>>> Are these not specific to 8k and 16k mode ?
>>> We should probably list values for both modes here.
>> 
>> No, this is just the offset that the audio sample is placed in with 
>> respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
>> pulse.
> 
> 
> That's not true atleast by the QDSP documentation,
> according to it we will use more slots to transfer at higher sample 
> rate. ex:
> 16 kHz data can be transferred using 8 kHz samples in two
> slots.
> 
> Also there are 32 slots for each of 4 supported channels for PCM AFE 
> port.

Ok, if that's the case, then it sounds like someone else is going to 
have to implement it. I have no way to test that kind of a 
configuration, so attempting to implement it would be futile.

> 
> 
>> 
>> When switching between 8 and 16k sample rate, it is just the sync 
>> pulse rate that is changed. The audio sample will be delivered in the 
>> same slot, just at a different frequency.

Re: [alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation

[Adam Serbinski]

On 2020-02-12 06:43, Srinivas Kandagatla wrote:
> On 12/02/2020 11:01, Adam Serbinski wrote:
>>>> 
>>>> +
>>>> + - qcom,pcm-slot-mapping
>>>> +    Usage: required for pcm interface
>>> 
>>> Are these not specific to 8k and 16k mode ?
>>> We should probably list values for both modes here.
>> 
>> No, this is just the offset that the audio sample is placed in with 
>> respect to a maximum of 4 slots, 16 bits wide, beginning with the sync 
>> pulse.
> 
> 
> That's not true atleast by the QDSP documentation,
> according to it we will use more slots to transfer at higher sample 
> rate. ex:
> 16 kHz data can be transferred using 8 kHz samples in two
> slots.
> 
> Also there are 32 slots for each of 4 supported channels for PCM AFE 
> port.

Ok, if that's the case, then it sounds like someone else is going to 
have to implement it. I have no way to test that kind of a 
configuration, so attempting to implement it would be futile.

> 
> 
>> 
>> When switching between 8 and 16k sample rate, it is just the sync 
>> pulse rate that is changed. The audio sample will be delivered in the 
>> same slot, just at a different frequency.
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