ASoC: msm: qdsp6v2: Extend LSM support to 48khz, 24bit and 4 channel

Enhance LSM code to support maximum configuration of 48khz, 24bit
and 4 channel.

Change-Id: I03895c983527d87389ca69e85235b1def5b4a2fa
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Revathi Uddaraju <revathiu@codeaurora.org>

5 files changed, 202 insertions, 54 deletions

diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index d8629fed8ab2..1f8bba7e9ab7 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -9063,6 +9063,7 @@ struct asm_aptx_dec_fmt_blk_v2 {
 #define LSM_PARAM_ID_SWMAD_MODEL			(0x00012C19)
 #define LSM_PARAM_ID_SWMAD_ENABLE			(0x00012C1A)
 #define LSM_PARAM_ID_POLLING_ENABLE			(0x00012C1B)
+#define LSM_PARAM_ID_MEDIA_FMT				(0x00012C1E)
 #define LSM_PARAM_ID_FWK_MODE_CONFIG			(0x00012C27)
 
 /* HW MAD specific */
diff --git a/include/sound/q6lsm.h b/include/sound/q6lsm.h
index 3bcc6c34b39f..4805246766d6 100644
--- a/include/sound/q6lsm.h
+++ b/include/sound/q6lsm.h
@@ -23,6 +23,8 @@
 
 #define ADM_LSM_PORT_ID 0xADCB
 
+#define LSM_MAX_NUM_CHANNELS 8
+
 typedef void (*lsm_app_cb)(uint32_t opcode, uint32_t token,
 		       uint32_t *payload, void *priv);
 
@@ -51,11 +53,12 @@ struct lsm_lab_buffer {
 	uint32_t mem_map_handle;
 };
 
-struct lsm_lab_hw_params {
+struct lsm_hw_params {
 	u16 sample_rate;
 	u16 sample_size;
 	u32 buf_sz;
 	u32 period_count;
+	u16 num_chs;
 };
 
 struct lsm_client {
@@ -81,7 +84,7 @@ struct lsm_client {
 	bool		lab_enable;
 	bool		lab_started;
 	struct lsm_lab_buffer *lab_buffer;
-	struct lsm_lab_hw_params hw_params;
+	struct lsm_hw_params hw_params;
 	bool		use_topology;
 	int		session_state;
 	bool		poll_enable;
@@ -153,6 +156,15 @@ struct lsm_param_fwk_mode_cfg {
 	uint32_t	mode;
 } __packed;
 
+struct lsm_param_media_fmt {
+	struct lsm_param_payload_common common;
+	uint32_t	minor_version;
+	uint32_t	sample_rate;
+	uint16_t	num_channels;
+	uint16_t	bit_width;
+	uint8_t		channel_mapping[LSM_MAX_NUM_CHANNELS];
+} __packed;
+
 /*
  * This param cannot be sent in this format.
  * The actual number of confidence level values
@@ -285,6 +297,13 @@ struct lsm_cmd_set_fwk_mode_cfg {
 	struct lsm_param_fwk_mode_cfg fwk_mode_cfg;
 } __packed;
 
+struct lsm_cmd_set_media_fmt {
+	struct apr_hdr  msg_hdr;
+	struct lsm_set_params_hdr params_hdr;
+	struct lsm_param_media_fmt media_fmt;
+} __packed;
+
+
 struct lsm_client *q6lsm_client_alloc(lsm_app_cb cb, void *priv);
 void q6lsm_client_free(struct lsm_client *client);
 int q6lsm_open(struct lsm_client *client, uint16_t app_id);
@@ -315,4 +334,5 @@ void q6lsm_sm_set_param_data(struct lsm_client *client,
 		size_t *offset);
 int q6lsm_set_port_connected(struct lsm_client *client);
 int q6lsm_set_fwk_mode_cfg(struct lsm_client *client, uint32_t event_mode);
+int q6lsm_set_media_fmt_params(struct lsm_client *client);
 #endif /* __Q6LSM_H__ */
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
index 6c1897340e74..44a6a245c7a2 100644
--- a/sound/soc/msm/msm-dai-fe.c
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -1,4 +1,4 @@
-/* Copyright (c) 2012-2016, The Linux Foundation. All rights reserved.
+/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved.
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 and
@@ -2203,12 +2203,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 1 Audio Service Capture",
 			.aif_name = "LSM1_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM1",
@@ -2218,12 +2220,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 2 Audio Service Capture",
 			.aif_name = "LSM2_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM2",
@@ -2233,12 +2237,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 3 Audio Service Capture",
 			.aif_name = "LSM3_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM3",
@@ -2248,12 +2254,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 4 Audio Service Capture",
 			.aif_name = "LSM4_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM4",
@@ -2263,12 +2271,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 5 Audio Service Capture",
 			.aif_name = "LSM5_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM5",
@@ -2278,12 +2288,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 6 Audio Service Capture",
 			.aif_name = "LSM6_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM6",
@@ -2293,12 +2305,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 7 Audio Service Capture",
 			.aif_name = "LSM7_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM7",
@@ -2308,12 +2322,14 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.capture = {
 			.stream_name = "Listen 8 Audio Service Capture",
 			.aif_name = "LSM8_UL_HL",
-			.rates = SNDRV_PCM_RATE_16000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.rates = (SNDRV_PCM_RATE_16000 |
+				  SNDRV_PCM_RATE_48000),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE),
 			.channels_min = 1,
-			.channels_max = 1,
+			.channels_max = 4,
 			.rate_min = 16000,
-			.rate_max = 16000,
+			.rate_max = 48000,
 		},
 		.ops = &msm_fe_dai_ops,
 		.name = "LSM8",
diff --git a/sound/soc/msm/qdsp6v2/msm-lsm-client.c b/sound/soc/msm/qdsp6v2/msm-lsm-client.c
index 3c266651847a..98e3cbc48068 100644
--- a/sound/soc/msm/qdsp6v2/msm-lsm-client.c
+++ b/sound/soc/msm/qdsp6v2/msm-lsm-client.c
@@ -35,7 +35,7 @@
 
 #define CAPTURE_MIN_NUM_PERIODS     2
 #define CAPTURE_MAX_NUM_PERIODS     8
-#define CAPTURE_MAX_PERIOD_SIZE     4096
+#define CAPTURE_MAX_PERIOD_SIZE     61440
 #define CAPTURE_MIN_PERIOD_SIZE     320
 #define LISTEN_MAX_STATUS_PAYLOAD_SIZE 256
 
@@ -47,12 +47,14 @@ static struct snd_pcm_hardware msm_pcm_hardware_capture = {
 				SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				SNDRV_PCM_INFO_INTERLEAVED |
 				SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
-	.formats =              SNDRV_PCM_FMTBIT_S16_LE,
-	.rates =                SNDRV_PCM_RATE_16000,
+	.formats =              (SNDRV_PCM_FMTBIT_S16_LE |
+				SNDRV_PCM_FMTBIT_S24_LE),
+	.rates =		(SNDRV_PCM_RATE_16000 |
+				SNDRV_PCM_RATE_48000),
 	.rate_min =             16000,
-	.rate_max =             16000,
+	.rate_max =             48000,
 	.channels_min =         1,
-	.channels_max =         1,
+	.channels_max =         4,
 	.buffer_bytes_max =     CAPTURE_MAX_NUM_PERIODS *
 				CAPTURE_MAX_PERIOD_SIZE,
 	.period_bytes_min =	CAPTURE_MIN_PERIOD_SIZE,
@@ -64,7 +66,7 @@ static struct snd_pcm_hardware msm_pcm_hardware_capture = {
 
 /* Conventional and unconventional sample rate supported */
 static unsigned int supported_sample_rates[] = {
-	16000,
+	16000, 48000,
 };
 
 static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
@@ -1962,6 +1964,10 @@ static int msm_lsm_prepare(struct snd_pcm_substream *substream)
 		return -EINVAL;
 	}
 
+	if (q6lsm_set_media_fmt_params(prtd->lsm_client))
+		dev_dbg(rtd->dev,
+			"%s: failed to set lsm media fmt params\n", __func__);
+
 	if (prtd->lsm_client->session_state == IDLE) {
 		ret = msm_pcm_routing_reg_phy_compr_stream(
 				rtd->dai_link->be_id,
@@ -2053,7 +2059,7 @@ static int msm_lsm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct lsm_priv *prtd = runtime->private_data;
-	struct lsm_lab_hw_params *hw_params = NULL;
+	struct lsm_hw_params *hw_params = NULL;
 	struct snd_soc_pcm_runtime *rtd;
 
 	if (!substream->private_data) {
@@ -2069,25 +2075,36 @@ static int msm_lsm_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 	hw_params = &prtd->lsm_client->hw_params;
-	hw_params->sample_rate = params_rate(params);
-	hw_params->sample_size =
-	(params_format(params) == SNDRV_PCM_FORMAT_S16_LE) ? 16 : 0;
+	hw_params->num_chs = params_channels(params);
 	hw_params->period_count = params_periods(params);
-	if (hw_params->sample_rate != 16000 || hw_params->sample_size != 16 ||
-		hw_params->period_count == 0) {
+	hw_params->sample_rate = params_rate(params);
+	if (((hw_params->sample_rate != 16000) &&
+		(hw_params->sample_rate != 48000)) ||
+		(hw_params->period_count == 0)) {
 		dev_err(rtd->dev,
-			"%s: Invalid params sample rate %d sample size %d period count %d",
+			"%s: Invalid Params sample rate %d period count %d\n",
 			__func__, hw_params->sample_rate,
-			hw_params->sample_size,
-		hw_params->period_count);
+			hw_params->period_count);
 		return -EINVAL;
 	}
+
+	if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) {
+		hw_params->sample_size = 16;
+	} else if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE) {
+		hw_params->sample_size = 24;
+	} else {
+		dev_err(rtd->dev, "%s: Invalid Format 0x%x\n",
+			__func__, params_format(params));
+		return -EINVAL;
+	}
+
 	hw_params->buf_sz = params_buffer_bytes(params) /
-	hw_params->period_count;
+			hw_params->period_count;
 	dev_dbg(rtd->dev,
-		"%s: sample rate %d sample size %d buffer size %d period count %d\n",
-		__func__, hw_params->sample_rate, hw_params->sample_size,
-		hw_params->buf_sz, hw_params->period_count);
+		"%s: channels %d sample rate %d sample size %d buffer size %d period count %d\n",
+		__func__, hw_params->num_chs, hw_params->sample_rate,
+		hw_params->sample_size, hw_params->buf_sz,
+		hw_params->period_count);
 	return 0;
 }
 
diff --git a/sound/soc/msm/qdsp6v2/q6lsm.c b/sound/soc/msm/qdsp6v2/q6lsm.c
index 455516ff6e41..525ec1c30f48 100644
--- a/sound/soc/msm/qdsp6v2/q6lsm.c
+++ b/sound/soc/msm/qdsp6v2/q6lsm.c
@@ -38,6 +38,8 @@
 #define LSM_ALIGN_BOUNDARY 512
 #define LSM_SAMPLE_RATE 16000
 #define QLSM_PARAM_ID_MINOR_VERSION 1
+#define QLSM_PARAM_ID_MINOR_VERSION_2 2
+
 static int lsm_afe_port;
 
 enum {
@@ -897,6 +899,98 @@ int q6lsm_set_fwk_mode_cfg(struct lsm_client *client,
 	return rc;
 }
 
+static int q6lsm_arrange_mch_map(struct lsm_param_media_fmt *media_fmt,
+			 int channel_count)
+{
+	int rc = 0;
+
+	memset(media_fmt->channel_mapping, 0, LSM_MAX_NUM_CHANNELS);
+
+	switch (channel_count) {
+	case 1:
+		media_fmt->channel_mapping[0] = PCM_CHANNEL_FC;
+		break;
+	case 2:
+		media_fmt->channel_mapping[0] = PCM_CHANNEL_FL;
+		media_fmt->channel_mapping[1] = PCM_CHANNEL_FR;
+		break;
+	case 3:
+		media_fmt->channel_mapping[0] = PCM_CHANNEL_FL;
+		media_fmt->channel_mapping[1] = PCM_CHANNEL_FR;
+		media_fmt->channel_mapping[2] = PCM_CHANNEL_FC;
+		break;
+	case 4:
+		media_fmt->channel_mapping[0] = PCM_CHANNEL_FL;
+		media_fmt->channel_mapping[1] = PCM_CHANNEL_FR;
+		media_fmt->channel_mapping[2] = PCM_CHANNEL_LS;
+		media_fmt->channel_mapping[3] = PCM_CHANNEL_RS;
+		break;
+	default:
+		pr_err("%s: invalid num_chan %d\n", __func__, channel_count);
+		rc = -EINVAL;
+		break;
+	}
+	return rc;
+}
+
+int q6lsm_set_media_fmt_params(struct lsm_client *client)
+{
+	int rc = 0;
+	struct lsm_cmd_set_media_fmt cmd;
+	struct lsm_module_param_ids media_fmt_ids;
+	struct apr_hdr  *msg_hdr;
+	struct lsm_param_media_fmt *media_fmt;
+	u32 data_payload_size, param_size, set_param_opcode;
+	struct lsm_hw_params param = client->hw_params;
+
+	if (client->use_topology) {
+		set_param_opcode = LSM_SESSION_CMD_SET_PARAMS_V2;
+		media_fmt_ids.module_id = LSM_MODULE_ID_FRAMEWORK;
+		media_fmt_ids.param_id = LSM_PARAM_ID_MEDIA_FMT;
+	} else {
+		pr_debug("%s: Ignore sending media format\n", __func__);
+		goto err_ret;
+	}
+
+	msg_hdr = &cmd.msg_hdr;
+	q6lsm_add_hdr(client, msg_hdr,
+		      sizeof(struct lsm_cmd_set_media_fmt), true);
+	msg_hdr->opcode = set_param_opcode;
+	data_payload_size = sizeof(struct lsm_cmd_set_media_fmt) -
+			    sizeof(struct apr_hdr) -
+			    sizeof(struct lsm_set_params_hdr);
+	q6lsm_set_param_hdr_info(&cmd.params_hdr,
+				 data_payload_size, 0, 0, 0);
+	media_fmt = &cmd.media_fmt;
+
+	param_size = (sizeof(struct lsm_param_media_fmt) -
+		      sizeof(media_fmt->common));
+	q6lsm_set_param_common(&media_fmt->common,
+			       &media_fmt_ids, param_size,
+			       set_param_opcode);
+
+	media_fmt->minor_version = QLSM_PARAM_ID_MINOR_VERSION_2;
+	media_fmt->sample_rate = param.sample_rate;
+	media_fmt->num_channels = param.num_chs;
+	media_fmt->bit_width = param.sample_size;
+
+	rc = q6lsm_arrange_mch_map(media_fmt, media_fmt->num_channels);
+	if (rc)
+		goto err_ret;
+
+	pr_debug("%s: sample rate= %d, channels %d bit width %d\n",
+		 __func__, media_fmt->sample_rate, media_fmt->num_channels,
+		 media_fmt->bit_width);
+
+	rc = q6lsm_apr_send_pkt(client, client->apr,
+				&cmd, true, NULL);
+	if (rc)
+		pr_err("%s: Failed set_params opcode 0x%x, rc %d\n",
+		       __func__, msg_hdr->opcode, rc);
+err_ret:
+	return rc;
+}
+
 int q6lsm_set_data(struct lsm_client *client,
 			   enum lsm_detection_mode mode,
 			   bool detectfailure)