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Message-Id: <20180426094606.4775-17-srinivas.kandagatla@linaro.org>
Date: Thu, 26 Apr 2018 10:45:58 +0100
From: srinivas.kandagatla@...aro.org
To: andy.gross@...aro.org, broonie@...nel.org,
linux-arm-msm@...r.kernel.org, alsa-devel@...a-project.org,
robh+dt@...nel.org, bgoswami@...eaurora.org
Cc: gregkh@...uxfoundation.org, david.brown@...aro.org,
mark.rutland@....com, lgirdwood@...il.com, plai@...eaurora.org,
tiwai@...e.com, perex@...ex.cz, devicetree@...r.kernel.org,
linux-kernel@...r.kernel.org, linux-arm-kernel@...ts.infradead.org,
rohkumar@....qualcomm.com, spatakok@....qualcomm.com,
Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Subject: [PATCH v6 16/24] ASoC: qdsp6: q6asm: Add support to audio stream apis
From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
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>
---
sound/soc/qcom/qdsp6/q6asm.c | 760 ++++++++++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 49 +++
2 files changed, 808 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index fc1b505dcca5..593f53191c03 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -10,6 +10,8 @@
#include <linux/of_platform.h>
#include <linux/spinlock.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>
@@ -18,10 +20,36 @@
#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 */
@@ -47,6 +75,96 @@ 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 apr_hdr hdr;
+ 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_FORMAT_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 apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ 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 {
+ struct apr_hdr hdr;
+ u32 flags;
+ u32 time_lsw;
+ u32 time_msw;
+} __packed;
+
struct audio_buffer {
phys_addr_t phys;
uint32_t used;
@@ -87,6 +205,22 @@ struct q6asm {
struct platform_device *pdev_dais;
};
+static bool q6asm_is_valid_audio_client(struct audio_client *ac)
+{
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ int n;
+
+ if (!ac)
+ return false;
+
+ for (n = 1; n <= MAX_SESSIONS; n++) {
+ if (a->session[n] == ac)
+ return true;
+ }
+
+ return false;
+}
+
static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, bool cmd_flg,
uint32_t stream_id)
@@ -379,6 +513,153 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
return a->session[session_id];
}
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+ struct apr_client_message *data,
+ int session_id)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct audio_port_data *port;
+ struct audio_client *ac;
+ uint32_t token;
+ uint32_t client_event = 0;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!ac)/* Audio client might already be freed by now */
+ return 0;
+
+ if (!q6asm_is_valid_audio_client(ac))
+ return -EINVAL;
+
+ result = data->payload;
+
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ token = data->token;
+ 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;
+ 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_FLUSH_READBUFS:
+ if (token != ac->session) {
+ dev_err(ac->dev, "session invalid\n");
+ return -EINVAL;
+ }
+ case ASM_STREAM_CMD_CLOSE:
+ client_event = ASM_CLIENT_EVENT_CMD_CLOSE_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);
+ return 0;
+ }
+ 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, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ 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->buf_lock, flags);
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return 0;
+ }
+
+ phys = port->buf[data->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[data->token].phys);
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return -EINVAL;
+ }
+ token = data->token;
+ port->buf[token].used = 1;
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ }
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2:
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ 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->buf_lock, flags);
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return 0;
+ }
+
+ phys = port->buf[data->token].phys;
+ token = data->token;
+ port->buf[token].used = 0;
+
+ 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[data->token].phys,
+ done->buf_addr_lsw,
+ done->buf_addr_msw);
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return -EINVAL;
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ }
+
+ break;
+ }
+
+ if (ac->cb)
+ ac->cb(client_event, data->token, data->payload, ac->priv);
+
+ return 0;
+}
+
static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_client_message *data)
{
@@ -389,6 +670,11 @@ 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 = (data->dest_port >> 8) & 0xFF;
+ if (session_id)
+ return q6asm_stream_callback(adev, data, session_id);
result = data->payload;
sid = (data->token >> 8) & 0x0F;
@@ -496,6 +782,478 @@ 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, void *cmd)
+{
+ struct apr_hdr *hdr = cmd;
+ int rc;
+
+ mutex_lock(&ac->lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+
+ rc = apr_send_pkt(ac->adev, cmd);
+ 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;
+ }
+
+
+err:
+ mutex_unlock(&ac->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;
+ int rc;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
+
+ open.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);
+ return -EINVAL;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, &open);
+ if (rc < 0)
+ return rc;
+
+ ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+ return 0;
+}
+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;
+
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+ if (wait)
+ return q6asm_ac_send_cmd_sync(ac, &run);
+ else
+ return apr_send_pkt(ac->adev, &run);
+
+}
+
+/**
+ * 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.
+ * @use_default_chmap: flag to use default ch map.
+ * @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_FORMAT_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
+ u8 *channel_mapping;
+ int rc;
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ 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_FORMAT_MAX_NUM_CHANNEL);
+ } else {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ return -EINVAL;
+ }
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, &fmt);
+ if (rc < 0)
+ goto fail_cmd;
+
+ return 0;
+fail_cmd:
+ 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.
+ * @use_default_chmap: flag to use default ch map.
+ * @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_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;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), true, ac->stream_id);
+ enc_cfg.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.hdr) -
+ 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;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (q6dsp_map_channels(channel_mapping, channels))
+ return -EINVAL;
+
+
+ return q6asm_ac_send_cmd_sync(ac, &enc_cfg);
+}
+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;
+ int rc;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return 0;
+
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ q6asm_add_hdr(ac, &read.hdr, sizeof(read), false, ac->stream_id);
+ ab = &port->buf[port->dsp_buf];
+ read.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 = ac->port[SNDRV_PCM_STREAM_CAPTURE].mem_map_handle;
+
+ read.buf_size = ab->size;
+ read.seq_id = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ return rc;
+ }
+
+ return 0;
+}
+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;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
+ open.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);
+ }
+
+ return q6asm_ac_send_cmd_sync(ac, &open);
+}
+
+/**
+ * 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
+ * @flags: 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 flags)
+{
+ struct asm_data_cmd_write_v2 write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int rc = 0;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return 0;
+
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write), false,
+ ac->stream_id);
+
+ ab = &port->buf[port->dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.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 (flags == NO_TIMESTAMP)
+ write.flags = (flags & 0x800000FF);
+ else
+ write.flags = (0x80000000 | flags);
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, &write);
+ if (rc < 0)
+ return rc;
+
+ return 0;
+}
+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;
+ int loopcnt = 0;
+ int cnt = 0;
+ int used;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return;
+
+ used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0);
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE; loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->num_periods - 1;
+ port->dsp_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = used;
+ cnt--;
+ }
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+ int stream_id = ac->stream_id;
+ struct apr_hdr hdr;
+ int rc;
+
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), true, stream_id);
+
+ switch (cmd) {
+ case CMD_PAUSE:
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_SUSPEND:
+ hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+ break;
+ case CMD_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ break;
+ case CMD_OUT_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ break;
+ case CMD_EOS:
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ case CMD_CLOSE:
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (wait)
+ rc = q6asm_ac_send_cmd_sync(ac, &hdr);
+ else
+ return apr_send_pkt(ac->adev, &hdr);
+
+ 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 93e86d922087..0ddba5206165 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 16
+#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_FORMAT_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,
--
2.16.2
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