Message-Id: <E1fKn20-00078m-SE@debutante>
Date:   Mon, 21 May 2018 16:47:28 +0100
From:   Mark Brown <broonie@...nel.org>
To:     Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Cc:     Mark Brown <broonie@...nel.org>, broonie@...nel.org,
        linux-arm-msm@...r.kernel.org, alsa-devel@...a-project.org,
        bgoswami@...eaurora.org, mark.rutland@....com,
        devicetree@...r.kernel.org, rohkumar@....qualcomm.com,
        gregkh@...uxfoundation.org, plai@...eaurora.org,
        lgirdwood@...il.com, tiwai@...e.com, david.brown@...aro.org,
        robh+dt@...nel.org, linux-arm-kernel@...ts.infradead.org,
        spatakok@....qualcomm.com, linux-kernel@...r.kernel.org,
        alsa-devel@...a-project.org
Subject: Applied "ASoC: qdsp6: q6asm: Add support to audio stream apis" to the asoc tree

The patch

   ASoC: qdsp6: q6asm: Add support to audio stream apis

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 68fd8480bb7baaf361e983f75d8571f25e017c67 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Date: Fri, 18 May 2018 13:56:03 +0100
Subject: [PATCH] ASoC: qdsp6: q6asm: Add support to audio stream apis

This patch adds support to open, write and media format commands
in the q6asm module.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@...eaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@...eaurora.org>
Signed-off-by: Mark Brown <broonie@...nel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 802 ++++++++++++++++++++++++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.h |  49 +++
 2 files changed, 850 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index a20d243ed10a..530852385cad 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,6 +11,8 @@
 #include <linux/spinlock.h>
 #include <linux/kref.h>
 #include <linux/of.h>
+#include <linux/of_platform.h>
+#include <uapi/sound/asound.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <linux/mm.h>
@@ -19,10 +21,36 @@
 #include "q6dsp-errno.h"
 #include "q6dsp-common.h"
 
+#define ASM_STREAM_CMD_CLOSE			0x00010BCD
+#define ASM_STREAM_CMD_FLUSH			0x00010BCE
+#define ASM_SESSION_CMD_PAUSE			0x00010BD3
+#define ASM_DATA_CMD_EOS			0x00010BDB
+#define ASM_NULL_POPP_TOPOLOGY			0x00010C68
+#define ASM_STREAM_CMD_FLUSH_READBUFS		0x00010C09
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM		0x00010C10
+#define ASM_STREAM_POSTPROC_TOPO_ID_NONE	0x00010C68
 #define ASM_CMD_SHARED_MEM_MAP_REGIONS		0x00010D92
 #define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS	0x00010D93
 #define ASM_CMD_SHARED_MEM_UNMAP_REGIONS	0x00010D94
-
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2	0x00010D98
+#define ASM_DATA_EVENT_WRITE_DONE_V2		0x00010D99
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2	0x00010DA3
+#define ASM_SESSION_CMD_RUN_V2			0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
+#define ASM_DATA_CMD_WRITE_V2			0x00010DAB
+#define ASM_DATA_CMD_READ_V2			0x00010DAC
+#define ASM_SESSION_CMD_SUSPEND			0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3		0x00010DB3
+#define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+
+
+#define ASM_LEGACY_STREAM_SESSION	0
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ              29
+#define ASM_END_POINT_DEVICE_MATRIX	0
+#define ASM_DEFAULT_APP_TYPE		0
 #define ASM_SYNC_IO_MODE		0x0001
 #define ASM_ASYNC_IO_MODE		0x0002
 #define ASM_TUN_READ_IO_MODE		0x0004	/* tunnel read write mode */
@@ -46,6 +74,89 @@ struct avs_cmd_shared_mem_unmap_regions {
 	u32 mem_map_handle;
 } __packed;
 
+struct asm_data_cmd_media_fmt_update_v2 {
+	u32 fmt_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 num_channels;
+	u16 bits_per_sample;
+	u32 sample_rate;
+	u16 is_signed;
+	u16 reserved;
+	u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+	u32                  param_id;
+	u32                  param_size;
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+	u32                  frames_per_buf;
+	u32                  enc_cfg_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	uint16_t  num_channels;
+	uint16_t  bits_per_sample;
+	uint32_t  sample_rate;
+	uint16_t  is_signed;
+	uint16_t  reserved;
+	uint8_t   channel_mapping[8];
+} __packed;
+
+struct asm_data_cmd_read_v2 {
+	u32                  buf_addr_lsw;
+	u32                  buf_addr_msw;
+	u32                  mem_map_handle;
+	u32                  buf_size;
+	u32                  seq_id;
+} __packed;
+
+struct asm_data_cmd_read_v2_done {
+	u32	status;
+	u32	buf_addr_lsw;
+	u32	buf_addr_msw;
+};
+
+struct asm_stream_cmd_open_read_v3 {
+	u32                    mode_flags;
+	u32                    src_endpointype;
+	u32                    preprocopo_id;
+	u32                    enc_cfg_id;
+	u16                    bits_per_sample;
+	u16                    reserved;
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+	u32 buf_addr_lsw;
+	u32 buf_addr_msw;
+	u32 mem_map_handle;
+	u32 buf_size;
+	u32 seq_id;
+	u32 timestamp_lsw;
+	u32 timestamp_msw;
+	u32 flags;
+} __packed;
+
+struct asm_stream_cmd_open_write_v3 {
+	uint32_t mode_flags;
+	uint16_t sink_endpointype;
+	uint16_t bits_per_sample;
+	uint32_t postprocopo_id;
+	uint32_t dec_fmt_id;
+} __packed;
+
+struct asm_session_cmd_run_v2 {
+	u32 flags;
+	u32 time_lsw;
+	u32 time_msw;
+} __packed;
+
 struct audio_buffer {
 	phys_addr_t phys;
 	uint32_t size;		/* size of buffer */
@@ -409,6 +520,149 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
 	return ac;
 }
 
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+				     struct apr_resp_pkt *data,
+				     int session_id)
+{
+	struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+	struct aprv2_ibasic_rsp_result_t *result;
+	struct apr_hdr *hdr = &data->hdr;
+	struct audio_port_data *port;
+	struct audio_client *ac;
+	uint32_t client_event = 0;
+	int ret = 0;
+
+	ac = q6asm_get_audio_client(q6asm, session_id);
+	if (!ac)/* Audio client might already be freed by now */
+		return 0;
+
+	result = data->payload;
+
+	switch (hdr->opcode) {
+	case APR_BASIC_RSP_RESULT:
+		switch (result->opcode) {
+		case ASM_SESSION_CMD_PAUSE:
+			client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
+			break;
+		case ASM_SESSION_CMD_SUSPEND:
+			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
+			break;
+		case ASM_DATA_CMD_EOS:
+			client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+			break;
+		case ASM_STREAM_CMD_FLUSH:
+			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
+			break;
+		case ASM_SESSION_CMD_RUN_V2:
+			client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
+			break;
+		case ASM_STREAM_CMD_CLOSE:
+			client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+			break;
+		case ASM_STREAM_CMD_FLUSH_READBUFS:
+			client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
+			break;
+		case ASM_STREAM_CMD_OPEN_WRITE_V3:
+		case ASM_STREAM_CMD_OPEN_READ_V3:
+		case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+		case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+		case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+			if (result->status != 0) {
+				dev_err(ac->dev,
+					"cmd = 0x%x returned error = 0x%x\n",
+					result->opcode, result->status);
+				ac->result = *result;
+				wake_up(&ac->cmd_wait);
+				ret = 0;
+				goto done;
+			}
+			break;
+		default:
+			dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+				result->opcode);
+			break;
+		}
+
+		ac->result = *result;
+		wake_up(&ac->cmd_wait);
+
+		if (ac->cb)
+			ac->cb(client_event, hdr->token,
+			       data->payload, ac->priv);
+
+		ret = 0;
+		goto done;
+
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+		client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+		if (ac->io_mode & ASM_SYNC_IO_MODE) {
+			phys_addr_t phys;
+			unsigned long flags;
+
+			spin_lock_irqsave(&ac->lock, flags);
+
+			port =  &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+			if (!port->buf) {
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = 0;
+				goto done;
+			}
+
+			phys = port->buf[hdr->token].phys;
+
+			if (lower_32_bits(phys) != result->opcode ||
+			    upper_32_bits(phys) != result->status) {
+				dev_err(ac->dev, "Expected addr %pa\n",
+					&port->buf[hdr->token].phys);
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = -EINVAL;
+				goto done;
+			}
+			spin_unlock_irqrestore(&ac->lock, flags);
+		}
+		break;
+	case ASM_DATA_EVENT_READ_DONE_V2:
+		client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
+		if (ac->io_mode & ASM_SYNC_IO_MODE) {
+			struct asm_data_cmd_read_v2_done *done = data->payload;
+			unsigned long flags;
+			phys_addr_t phys;
+
+			spin_lock_irqsave(&ac->lock, flags);
+			port =  &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+			if (!port->buf) {
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = 0;
+				goto done;
+			}
+
+			phys = port->buf[hdr->token].phys;
+
+			if (upper_32_bits(phys) != done->buf_addr_msw ||
+			    lower_32_bits(phys) != done->buf_addr_lsw) {
+				dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
+					&port->buf[hdr->token].phys,
+					done->buf_addr_lsw,
+					done->buf_addr_msw);
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = -EINVAL;
+				goto done;
+			}
+			spin_unlock_irqrestore(&ac->lock, flags);
+		}
+
+		break;
+	}
+
+	if (ac->cb)
+		ac->cb(client_event, hdr->token, data->payload, ac->priv);
+
+done:
+	kref_put(&ac->refcount, q6asm_audio_client_release);
+	return ret;
+}
+
 static int q6asm_srvc_callback(struct apr_device *adev,
 			       struct apr_resp_pkt *data)
 {
@@ -420,6 +674,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
 	struct q6asm *a;
 	uint32_t sid = 0;
 	uint32_t dir = 0;
+	int session_id;
+
+	session_id = (hdr->dest_port >> 8) & 0xFF;
+	if (session_id)
+		return q6asm_stream_callback(adev, data, session_id);
 
 	sid = (hdr->token >> 8) & 0x0F;
 	ac = q6asm_get_audio_client(q6asm, sid);
@@ -540,6 +799,547 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
 }
 EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
 
+static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
+{
+	struct apr_hdr *hdr = &pkt->hdr;
+	int rc;
+
+	mutex_lock(&ac->cmd_lock);
+	ac->result.opcode = 0;
+	ac->result.status = 0;
+
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc < 0)
+		goto err;
+
+	rc = wait_event_timeout(ac->cmd_wait,
+				(ac->result.opcode == hdr->opcode), 5 * HZ);
+	if (!rc) {
+		dev_err(ac->dev, "CMD timeout\n");
+		rc =  -ETIMEDOUT;
+		goto err;
+	}
+
+	if (ac->result.status > 0) {
+		dev_err(ac->dev, "DSP returned error[%x]\n",
+			ac->result.status);
+		rc = -EINVAL;
+	} else {
+		rc = 0;
+	}
+
+
+err:
+	mutex_unlock(&ac->cmd_lock);
+	return rc;
+}
+
+/**
+ * q6asm_open_write() - Open audio client for writing
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+		     uint16_t bits_per_sample)
+{
+	struct asm_stream_cmd_open_write_v3 *open;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*open);
+
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	open = p + APR_HDR_SIZE;
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
+	open->mode_flags = 0x00;
+	open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
+
+	/* source endpoint : matrix */
+	open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+	open->bits_per_sample = bits_per_sample;
+	open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
+
+	switch (format) {
+	case FORMAT_LINEAR_PCM:
+		open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		break;
+	default:
+		dev_err(ac->dev, "Invalid format 0x%x\n", format);
+		rc = -EINVAL;
+		goto err;
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	if (rc < 0)
+		goto err;
+
+	ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_open_write);
+
+static int __q6asm_run(struct audio_client *ac, uint32_t flags,
+	      uint32_t msw_ts, uint32_t lsw_ts, bool wait)
+{
+	struct asm_session_cmd_run_v2 *run;
+	struct apr_pkt *pkt;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*run);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	run = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+	run->flags = flags;
+	run->time_lsw = lsw_ts;
+	run->time_msw = msw_ts;
+	if (wait) {
+		rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	} else {
+		rc = apr_send_pkt(ac->adev, pkt);
+		if (rc == pkt_size)
+			rc = 0;
+	}
+
+	kfree(pkt);
+	return rc;
+}
+
+/**
+ * q6asm_run() - start the audio client
+ *
+ * @ac: audio client pointer
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+	      uint32_t msw_ts, uint32_t lsw_ts)
+{
+	return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_run);
+
+/**
+ * q6asm_run_nowait() - start the audio client withou blocking
+ *
+ * @ac: audio client pointer
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+	      uint32_t msw_ts, uint32_t lsw_ts)
+{
+	return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_run_nowait);
+
+/**
+ * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @channel_map: channel map pointer
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+					  uint32_t rate, uint32_t channels,
+					  u8 channel_map[PCM_MAX_NUM_CHANNEL],
+					  uint16_t bits_per_sample)
+{
+	struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	u8 *channel_mapping;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->num_channels = channels;
+	fmt->bits_per_sample = bits_per_sample;
+	fmt->sample_rate = rate;
+	fmt->is_signed = 1;
+
+	channel_mapping = fmt->channel_mapping;
+
+	if (channel_map) {
+		memcpy(channel_mapping, channel_map, PCM_MAX_NUM_CHANNEL);
+	} else {
+		if (q6dsp_map_channels(channel_mapping, channels)) {
+			dev_err(ac->dev, " map channels failed %d\n", channels);
+			rc = -EINVAL;
+			goto err;
+		}
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+		uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+{
+	struct asm_multi_channel_pcm_enc_cfg_v2  *enc_cfg;
+	struct apr_pkt *pkt;
+	u8 *channel_mapping;
+	u32 frames_per_buf = 0;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	enc_cfg = p + APR_HDR_SIZE;
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+	enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+	enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
+	enc_cfg->encblk.frames_per_buf = frames_per_buf;
+	enc_cfg->encblk.enc_cfg_blk_size  = enc_cfg->encdec.param_size -
+					sizeof(struct asm_enc_cfg_blk_param_v2);
+
+	enc_cfg->num_channels = channels;
+	enc_cfg->bits_per_sample = bits_per_sample;
+	enc_cfg->sample_rate = rate;
+	enc_cfg->is_signed = 1;
+	channel_mapping = enc_cfg->channel_mapping;
+
+	if (q6dsp_map_channels(channel_mapping, channels)) {
+		rc = -EINVAL;
+		goto err;
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
+
+/**
+ * q6asm_read() - read data of period size from audio client
+ *
+ * @ac: audio client pointer
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_read(struct audio_client *ac)
+{
+	struct asm_data_cmd_read_v2 *read;
+	struct audio_port_data *port;
+	struct audio_buffer *ab;
+	struct apr_pkt *pkt;
+	unsigned long flags;
+	int pkt_size;
+	int rc = 0;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*read);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	read = p + APR_HDR_SIZE;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+	ab = &port->buf[port->dsp_buf];
+	pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
+	read->buf_addr_lsw = lower_32_bits(ab->phys);
+	read->buf_addr_msw = upper_32_bits(ab->phys);
+	read->mem_map_handle = port->mem_map_handle;
+
+	read->buf_size = ab->size;
+	read->seq_id = port->dsp_buf;
+	pkt->hdr.token = port->dsp_buf;
+
+	port->dsp_buf++;
+
+	if (port->dsp_buf >= port->num_periods)
+		port->dsp_buf = 0;
+
+	spin_unlock_irqrestore(&ac->lock, flags);
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc == pkt_size)
+		rc = 0;
+	else
+		pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
+
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_read);
+
+static int __q6asm_open_read(struct audio_client *ac,
+		uint32_t format, uint16_t bits_per_sample)
+{
+	struct asm_stream_cmd_open_read_v3 *open;
+	struct apr_pkt *pkt;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*open);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	open = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr,  pkt_size, true, ac->stream_id);
+	pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
+	/* Stream prio : High, provide meta info with encoded frames */
+	open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+	open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
+	open->bits_per_sample = bits_per_sample;
+	open->mode_flags = 0x0;
+
+	open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+				ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+
+	switch (format) {
+	case FORMAT_LINEAR_PCM:
+		open->mode_flags |= 0x00;
+		open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		break;
+	default:
+		pr_err("Invalid format[%d]\n", format);
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+	kfree(pkt);
+	return rc;
+}
+
+/**
+ * q6asm_open_read() - Open audio client for reading
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+			uint16_t bits_per_sample)
+{
+	return __q6asm_open_read(ac, format, bits_per_sample);
+}
+EXPORT_SYMBOL_GPL(q6asm_open_read);
+
+/**
+ * q6asm_write_async() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @len: lenght in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @wflags: flags associated with write
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+		       uint32_t lsw_ts, uint32_t wflags)
+{
+	struct asm_data_cmd_write_v2 *write;
+	struct audio_port_data *port;
+	struct audio_buffer *ab;
+	unsigned long flags;
+	struct apr_pkt *pkt;
+	int pkt_size;
+	int rc = 0;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*write);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	write = p + APR_HDR_SIZE;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+
+	ab = &port->buf[port->dsp_buf];
+	pkt->hdr.token = port->dsp_buf;
+	pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+	write->buf_addr_lsw = lower_32_bits(ab->phys);
+	write->buf_addr_msw = upper_32_bits(ab->phys);
+	write->buf_size = len;
+	write->seq_id = port->dsp_buf;
+	write->timestamp_lsw = lsw_ts;
+	write->timestamp_msw = msw_ts;
+	write->mem_map_handle =
+	    ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
+
+	if (wflags == NO_TIMESTAMP)
+		write->flags = (wflags & 0x800000FF);
+	else
+		write->flags = (0x80000000 | wflags);
+
+	port->dsp_buf++;
+
+	if (port->dsp_buf >= port->num_periods)
+		port->dsp_buf = 0;
+
+	spin_unlock_irqrestore(&ac->lock, flags);
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc == pkt_size)
+		rc = 0;
+
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+	struct audio_port_data *port = NULL;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+	port->dsp_buf = 0;
+	port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+	port->dsp_buf = 0;
+	spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+	int stream_id = ac->stream_id;
+	struct apr_pkt pkt;
+	int rc;
+
+	q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
+
+	switch (cmd) {
+	case CMD_PAUSE:
+		pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
+		break;
+	case CMD_SUSPEND:
+		pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+		break;
+	case CMD_FLUSH:
+		pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
+		break;
+	case CMD_OUT_FLUSH:
+		pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+		break;
+	case CMD_EOS:
+		pkt.hdr.opcode = ASM_DATA_CMD_EOS;
+		break;
+	case CMD_CLOSE:
+		pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (wait)
+		rc = q6asm_ac_send_cmd_sync(ac, &pkt);
+	else
+		return apr_send_pkt(ac->adev, &pkt);
+
+	if (rc < 0)
+		return rc;
+
+	if (cmd == CMD_FLUSH)
+		q6asm_reset_buf_state(ac);
+
+	return 0;
+}
+
+/**
+ * q6asm_cmd() - run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd(struct audio_client *ac, int cmd)
+{
+	return __q6asm_cmd(ac, cmd, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd);
+
+/**
+ * q6asm_cmd_nowait() - non blocking, run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+{
+	return __q6asm_cmd(ac, cmd, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
 
 static int q6asm_probe(struct apr_device *adev)
 {
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 8c317b7b63c3..9f5fb573e4a0 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -1,8 +1,36 @@
 /* SPDX-License-Identifier: GPL-2.0 */
 #ifndef __Q6_ASM_H__
 #define __Q6_ASM_H__
+#include "q6dsp-common.h"
+#include <dt-bindings/sound/qcom,q6asm.h>
+
+/* ASM client callback events */
+#define CMD_PAUSE			0x0001
+#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE		0x1001
+#define CMD_FLUSH				0x0002
+#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE		0x1002
+#define CMD_EOS				0x0003
+#define ASM_CLIENT_EVENT_CMD_EOS_DONE		0x1003
+#define CMD_CLOSE				0x0004
+#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE		0x1004
+#define CMD_OUT_FLUSH				0x0005
+#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE	0x1005
+#define CMD_SUSPEND				0x0006
+#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE	0x1006
+#define ASM_CLIENT_EVENT_CMD_RUN_DONE		0x1008
+#define ASM_CLIENT_EVENT_DATA_WRITE_DONE	0x1009
+#define ASM_CLIENT_EVENT_DATA_READ_DONE		0x100a
+
+enum {
+	LEGACY_PCM_MODE = 0,
+	LOW_LATENCY_PCM_MODE,
+	ULTRA_LOW_LATENCY_PCM_MODE,
+	ULL_POST_PROCESSING_PCM_MODE,
+};
 
 #define MAX_SESSIONS	8
+#define NO_TIMESTAMP    0xFF00
+#define FORMAT_LINEAR_PCM   0x0000
 
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
@@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
 					      q6asm_cb cb, void *priv,
 					      int session_id, int perf_mode);
 void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+		       uint32_t lsw_ts, uint32_t flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+		     uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+		     uint16_t bits_per_sample);
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+		uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
+int q6asm_read(struct audio_client *ac);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+					  uint32_t rate, uint32_t channels,
+					  u8 channel_map[PCM_MAX_NUM_CHANNEL],
+					  uint16_t bits_per_sample);
+int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+	      uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+		     uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
 int q6asm_get_session_id(struct audio_client *ac);
 int q6asm_map_memory_regions(unsigned int dir,
 			     struct audio_client *ac,
-- 
2.17.0

Powered by blists - more mailing lists