linux/sound/soc/qcom/qdsp6/q6asm.c
Krzysztof Kozlowski 528a4a0bb0
ASoC: qcom: reduce number of binding headers includes
Move the includes of binding headers from Qualcomm SoC sound drivers
headers to unit files actually using these bindings.  This reduces the
amount of work for C preprocessor and makes usage of bindings easier to
follow.  No impact expected on the final binaries.

Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20231005075250.88159-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
2023-10-09 13:14:19 +01:00

1754 lines
42 KiB
C

// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
// Copyright (c) 2018, Linaro Limited
#include <dt-bindings/sound/qcom,q6asm.h>
#include <linux/mutex.h>
#include <linux/wait.h>
#include <linux/module.h>
#include <linux/soc/qcom/apr.h>
#include <linux/device.h>
#include <linux/of_platform.h>
#include <linux/spinlock.h>
#include <linux/kref.h>
#include <linux/of.h>
#include <uapi/sound/asound.h>
#include <uapi/sound/compress_params.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
#include "q6asm.h"
#include "q6core.h"
#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_DATA_EVENT_RENDERED_EOS 0x00010C1C
#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_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
#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_MEDIA_FMT_ALAC 0x00012f31
#define ASM_MEDIA_FMT_APE 0x00012f32
#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67
#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
#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 */
#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
#define ASM_SHIFT_GAPLESS_MODE_FLAG 31
#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
struct avs_cmd_shared_mem_map_regions {
u16 mem_pool_id;
u16 num_regions;
u32 property_flag;
} __packed;
struct avs_shared_map_region_payload {
u32 shm_addr_lsw;
u32 shm_addr_msw;
u32 mem_size_bytes;
} __packed;
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_flac_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 is_stream_info_present;
u16 num_channels;
u16 min_blk_size;
u16 max_blk_size;
u16 md5_sum[8];
u32 sample_rate;
u32 min_frame_size;
u32 max_frame_size;
u16 sample_size;
u16 reserved;
} __packed;
struct asm_wmastdv9_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 fmtag;
u16 num_channels;
u32 sample_rate;
u32 bytes_per_sec;
u16 blk_align;
u16 bits_per_sample;
u32 channel_mask;
u16 enc_options;
u16 reserved;
} __packed;
struct asm_wmaprov10_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 fmtag;
u16 num_channels;
u32 sample_rate;
u32 bytes_per_sec;
u16 blk_align;
u16 bits_per_sample;
u32 channel_mask;
u16 enc_options;
u16 advanced_enc_options1;
u32 advanced_enc_options2;
} __packed;
struct asm_alac_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u32 frame_length;
u8 compatible_version;
u8 bit_depth;
u8 pb;
u8 mb;
u8 kb;
u8 num_channels;
u16 max_run;
u32 max_frame_bytes;
u32 avg_bit_rate;
u32 sample_rate;
u32 channel_layout_tag;
} __packed;
struct asm_ape_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 compatible_version;
u16 compression_level;
u32 format_flags;
u32 blocks_per_frame;
u32 final_frame_blocks;
u32 total_frames;
u16 bits_per_sample;
u16 num_channels;
u32 sample_rate;
u32 seek_table_present;
} __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 */
};
struct audio_port_data {
struct audio_buffer *buf;
uint32_t num_periods;
uint32_t dsp_buf;
uint32_t mem_map_handle;
};
struct q6asm {
struct apr_device *adev;
struct device *dev;
struct q6core_svc_api_info ainfo;
wait_queue_head_t mem_wait;
spinlock_t slock;
struct audio_client *session[MAX_SESSIONS + 1];
};
struct audio_client {
int session;
q6asm_cb cb;
void *priv;
uint32_t io_mode;
struct apr_device *adev;
struct mutex cmd_lock;
spinlock_t lock;
struct kref refcount;
/* idx:1 out port, 0: in port */
struct audio_port_data port[2];
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
struct q6asm *q6asm;
struct device *dev;
};
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)
{
hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
hdr->pkt_size = pkt_size;
if (cmd_flg)
hdr->token = ac->session;
}
static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
struct apr_pkt *pkt, uint32_t rsp_opcode)
{
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(a->adev, pkt);
if (rc < 0)
goto err;
if (rsp_opcode)
rc = wait_event_timeout(a->mem_wait,
(ac->result.opcode == hdr->opcode) ||
(ac->result.opcode == rsp_opcode),
5 * HZ);
else
rc = wait_event_timeout(a->mem_wait,
(ac->result.opcode == hdr->opcode),
5 * HZ);
if (!rc) {
dev_err(a->dev, "CMD %x timeout\n", hdr->opcode);
rc = -ETIMEDOUT;
} else if (ac->result.status > 0) {
dev_err(a->dev, "DSP returned error[%x]\n",
ac->result.status);
rc = -EINVAL;
}
err:
mutex_unlock(&ac->cmd_lock);
return rc;
}
static int __q6asm_memory_unmap(struct audio_client *ac,
phys_addr_t buf_add, int dir)
{
struct avs_cmd_shared_mem_unmap_regions *mem_unmap;
struct q6asm *a = dev_get_drvdata(ac->dev->parent);
struct apr_pkt *pkt;
int rc, pkt_size;
void *p;
if (ac->port[dir].mem_map_handle == 0) {
dev_err(ac->dev, "invalid mem handle\n");
return -EINVAL;
}
pkt_size = APR_HDR_SIZE + sizeof(*mem_unmap);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
mem_unmap = p + APR_HDR_SIZE;
pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
pkt->hdr.src_port = 0;
pkt->hdr.dest_port = 0;
pkt->hdr.pkt_size = pkt_size;
pkt->hdr.token = ((ac->session << 8) | dir);
pkt->hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
mem_unmap->mem_map_handle = ac->port[dir].mem_map_handle;
rc = q6asm_apr_send_session_pkt(a, ac, pkt, 0);
if (rc < 0) {
kfree(pkt);
return rc;
}
ac->port[dir].mem_map_handle = 0;
kfree(pkt);
return 0;
}
static void q6asm_audio_client_free_buf(struct audio_client *ac,
struct audio_port_data *port)
{
unsigned long flags;
spin_lock_irqsave(&ac->lock, flags);
port->num_periods = 0;
kfree(port->buf);
port->buf = NULL;
spin_unlock_irqrestore(&ac->lock, flags);
}
/**
* q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
*
* @dir: direction of audio stream
* @ac: audio client instanace
*
* Return: Will be an negative value on failure or zero on success
*/
int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
{
struct audio_port_data *port;
int cnt = 0;
int rc = 0;
port = &ac->port[dir];
if (!port->buf) {
rc = -EINVAL;
goto err;
}
cnt = port->num_periods - 1;
if (cnt >= 0) {
rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
if (rc < 0) {
dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
__func__, rc);
goto err;
}
}
q6asm_audio_client_free_buf(ac, port);
err:
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
size_t period_sz, unsigned int periods,
bool is_contiguous)
{
struct avs_cmd_shared_mem_map_regions *cmd = NULL;
struct avs_shared_map_region_payload *mregions = NULL;
struct q6asm *a = dev_get_drvdata(ac->dev->parent);
struct audio_port_data *port = NULL;
struct audio_buffer *ab = NULL;
struct apr_pkt *pkt;
void *p;
unsigned long flags;
uint32_t num_regions, buf_sz;
int rc, i, pkt_size;
if (is_contiguous) {
num_regions = 1;
buf_sz = period_sz * periods;
} else {
buf_sz = period_sz;
num_regions = periods;
}
/* DSP expects size should be aligned to 4K */
buf_sz = ALIGN(buf_sz, 4096);
pkt_size = APR_HDR_SIZE + sizeof(*cmd) +
(sizeof(*mregions) * num_regions);
p = kzalloc(pkt_size, GFP_KERNEL);
if (!p)
return -ENOMEM;
pkt = p;
cmd = p + APR_HDR_SIZE;
mregions = p + APR_HDR_SIZE + sizeof(*cmd);
pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
pkt->hdr.src_port = 0;
pkt->hdr.dest_port = 0;
pkt->hdr.pkt_size = pkt_size;
pkt->hdr.token = ((ac->session << 8) | dir);
pkt->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
cmd->num_regions = num_regions;
cmd->property_flag = 0x00;
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[dir];
for (i = 0; i < num_regions; i++) {
ab = &port->buf[i];
mregions->shm_addr_lsw = lower_32_bits(ab->phys);
mregions->shm_addr_msw = upper_32_bits(ab->phys);
mregions->mem_size_bytes = buf_sz;
++mregions;
}
spin_unlock_irqrestore(&ac->lock, flags);
rc = q6asm_apr_send_session_pkt(a, ac, pkt,
ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
kfree(pkt);
return rc;
}
/**
* q6asm_map_memory_regions() - map memory regions in the dsp.
*
* @dir: direction of audio stream
* @ac: audio client instanace
* @phys: physical address that needs mapping.
* @period_sz: audio period size
* @periods: number of periods
*
* Return: Will be an negative value on failure or zero on success
*/
int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
phys_addr_t phys,
size_t period_sz, unsigned int periods)
{
struct audio_buffer *buf;
unsigned long flags;
int cnt;
int rc;
spin_lock_irqsave(&ac->lock, flags);
if (ac->port[dir].buf) {
dev_err(ac->dev, "Buffer already allocated\n");
spin_unlock_irqrestore(&ac->lock, flags);
return 0;
}
buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC);
if (!buf) {
spin_unlock_irqrestore(&ac->lock, flags);
return -ENOMEM;
}
ac->port[dir].buf = buf;
buf[0].phys = phys;
buf[0].size = period_sz;
for (cnt = 1; cnt < periods; cnt++) {
if (period_sz > 0) {
buf[cnt].phys = buf[0].phys + (cnt * period_sz);
buf[cnt].size = period_sz;
}
}
ac->port[dir].num_periods = periods;
spin_unlock_irqrestore(&ac->lock, flags);
rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
if (rc < 0) {
dev_err(ac->dev, "Memory_map_regions failed\n");
q6asm_audio_client_free_buf(ac, &ac->port[dir]);
}
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
static void q6asm_audio_client_release(struct kref *ref)
{
struct audio_client *ac;
struct q6asm *a;
unsigned long flags;
ac = container_of(ref, struct audio_client, refcount);
a = ac->q6asm;
spin_lock_irqsave(&a->slock, flags);
a->session[ac->session] = NULL;
spin_unlock_irqrestore(&a->slock, flags);
kfree(ac);
}
/**
* q6asm_audio_client_free() - Freee allocated audio client
*
* @ac: audio client to free
*/
void q6asm_audio_client_free(struct audio_client *ac)
{
kref_put(&ac->refcount, q6asm_audio_client_release);
}
EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
int session_id)
{
struct audio_client *ac = NULL;
unsigned long flags;
spin_lock_irqsave(&a->slock, flags);
if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
dev_err(a->dev, "invalid session: %d\n", session_id);
goto err;
}
/* check for valid session */
if (!a->session[session_id])
goto err;
else if (a->session[session_id]->session != session_id)
goto err;
ac = a->session[session_id];
kref_get(&ac->refcount);
err:
spin_unlock_irqrestore(&a->slock, flags);
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_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:
case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
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;
int token = hdr->token & ASM_WRITE_TOKEN_MASK;
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[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[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;
case ASM_DATA_EVENT_RENDERED_EOS:
client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
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)
{
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
struct aprv2_ibasic_rsp_result_t *result;
struct audio_port_data *port;
struct audio_client *ac = NULL;
struct apr_hdr *hdr = &data->hdr;
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);
if (!ac) {
dev_err(&adev->dev, "Audio Client not active\n");
return 0;
}
a = dev_get_drvdata(ac->dev->parent);
dir = (hdr->token & 0x0F);
port = &ac->port[dir];
result = data->payload;
switch (hdr->opcode) {
case APR_BASIC_RSP_RESULT:
switch (result->opcode) {
case ASM_CMD_SHARED_MEM_MAP_REGIONS:
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
ac->result = *result;
wake_up(&a->mem_wait);
break;
default:
dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
result->opcode);
break;
}
goto done;
case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
ac->result.status = 0;
ac->result.opcode = hdr->opcode;
port->mem_map_handle = result->opcode;
wake_up(&a->mem_wait);
break;
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
ac->result.opcode = hdr->opcode;
ac->result.status = 0;
port->mem_map_handle = 0;
wake_up(&a->mem_wait);
break;
default:
dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
result->opcode, result->status);
break;
}
if (ac->cb)
ac->cb(hdr->opcode, hdr->token, data->payload, ac->priv);
done:
kref_put(&ac->refcount, q6asm_audio_client_release);
return 0;
}
/**
* q6asm_get_session_id() - get session id for audio client
*
* @c: audio client pointer
*
* Return: Will be an session id of the audio client.
*/
int q6asm_get_session_id(struct audio_client *c)
{
return c->session;
}
EXPORT_SYMBOL_GPL(q6asm_get_session_id);
/**
* q6asm_audio_client_alloc() - Allocate a new audio client
*
* @dev: Pointer to asm child device.
* @cb: event callback.
* @priv: private data associated with this client.
* @session_id: session id
* @perf_mode: performace mode for this client
*
* Return: Will be an error pointer on error or a valid audio client
* on success.
*/
struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
void *priv, int session_id,
int perf_mode)
{
struct q6asm *a = dev_get_drvdata(dev->parent);
struct audio_client *ac;
unsigned long flags;
ac = q6asm_get_audio_client(a, session_id + 1);
if (ac) {
dev_err(dev, "Audio Client already active\n");
return ac;
}
ac = kzalloc(sizeof(*ac), GFP_KERNEL);
if (!ac)
return ERR_PTR(-ENOMEM);
spin_lock_irqsave(&a->slock, flags);
a->session[session_id + 1] = ac;
spin_unlock_irqrestore(&a->slock, flags);
ac->session = session_id + 1;
ac->cb = cb;
ac->dev = dev;
ac->q6asm = a;
ac->priv = priv;
ac->io_mode = ASM_SYNC_IO_MODE;
ac->perf_mode = perf_mode;
ac->adev = a->adev;
kref_init(&ac->refcount);
init_waitqueue_head(&ac->cmd_wait);
mutex_init(&ac->cmd_lock);
spin_lock_init(&ac->lock);
return ac;
}
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 %x timeout\n", hdr->opcode);
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
* @stream_id: stream id of q6asm session
* @format: audio sample format
* @codec_profile: compressed format profile
* @bits_per_sample: bits per sample
* @is_gapless: flag to indicate if this is a gapless stream
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
uint32_t format, u32 codec_profile,
uint16_t bits_per_sample, bool is_gapless)
{
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, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open->mode_flags = 0x00;
open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
if (is_gapless)
open->mode_flags |= BIT(ASM_SHIFT_GAPLESS_MODE_FLAG);
/* 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 SND_AUDIOCODEC_MP3:
open->dec_fmt_id = ASM_MEDIA_FMT_MP3;
break;
case FORMAT_LINEAR_PCM:
open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
case SND_AUDIOCODEC_WMA:
switch (codec_profile) {
case SND_AUDIOPROFILE_WMA9:
open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
break;
case SND_AUDIOPROFILE_WMA10:
case SND_AUDIOPROFILE_WMA9_PRO:
case SND_AUDIOPROFILE_WMA9_LOSSLESS:
case SND_AUDIOPROFILE_WMA10_LOSSLESS:
open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
break;
default:
dev_err(ac->dev, "Invalid codec profile 0x%x\n",
codec_profile);
rc = -EINVAL;
goto err;
}
break;
case SND_AUDIOCODEC_ALAC:
open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
break;
case SND_AUDIOCODEC_APE:
open->dec_fmt_id = ASM_MEDIA_FMT_APE;
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 stream_id,
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, 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
* @stream_id: stream id of q6asm session
* @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 stream_id, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true);
}
EXPORT_SYMBOL_GPL(q6asm_run);
/**
* q6asm_run_nowait() - start the audio client withou blocking
*
* @ac: audio client pointer
* @stream_id: stream id
* @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 stream_id,
uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts)
{
return __q6asm_run(ac, stream_id, 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
* @stream_id: stream id
* @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 stream_id,
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, 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);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
uint32_t stream_id,
struct q6asm_flac_cfg *cfg)
{
struct asm_flac_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
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, 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->is_stream_info_present = cfg->stream_info_present;
fmt->num_channels = cfg->ch_cfg;
fmt->min_blk_size = cfg->min_blk_size;
fmt->max_blk_size = cfg->max_blk_size;
fmt->sample_rate = cfg->sample_rate;
fmt->min_frame_size = cfg->min_frame_size;
fmt->max_frame_size = cfg->max_frame_size;
fmt->sample_size = cfg->sample_size;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
uint32_t stream_id,
struct q6asm_wma_cfg *cfg)
{
struct asm_wmastdv9_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
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, 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->fmtag = cfg->fmtag;
fmt->num_channels = cfg->num_channels;
fmt->sample_rate = cfg->sample_rate;
fmt->bytes_per_sec = cfg->bytes_per_sec;
fmt->blk_align = cfg->block_align;
fmt->bits_per_sample = cfg->bits_per_sample;
fmt->channel_mask = cfg->channel_mask;
fmt->enc_options = cfg->enc_options;
fmt->reserved = 0;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
uint32_t stream_id,
struct q6asm_wma_cfg *cfg)
{
struct asm_wmaprov10_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
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, 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->fmtag = cfg->fmtag;
fmt->num_channels = cfg->num_channels;
fmt->sample_rate = cfg->sample_rate;
fmt->bytes_per_sec = cfg->bytes_per_sec;
fmt->blk_align = cfg->block_align;
fmt->bits_per_sample = cfg->bits_per_sample;
fmt->channel_mask = cfg->channel_mask;
fmt->enc_options = cfg->enc_options;
fmt->advanced_enc_options1 = cfg->adv_enc_options;
fmt->advanced_enc_options2 = cfg->adv_enc_options2;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
int q6asm_stream_media_format_block_alac(struct audio_client *ac,
uint32_t stream_id,
struct q6asm_alac_cfg *cfg)
{
struct asm_alac_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
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, 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->frame_length = cfg->frame_length;
fmt->compatible_version = cfg->compatible_version;
fmt->bit_depth = cfg->bit_depth;
fmt->num_channels = cfg->num_channels;
fmt->max_run = cfg->max_run;
fmt->max_frame_bytes = cfg->max_frame_bytes;
fmt->avg_bit_rate = cfg->avg_bit_rate;
fmt->sample_rate = cfg->sample_rate;
fmt->channel_layout_tag = cfg->channel_layout_tag;
fmt->pb = cfg->pb;
fmt->mb = cfg->mb;
fmt->kb = cfg->kb;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
int q6asm_stream_media_format_block_ape(struct audio_client *ac,
uint32_t stream_id,
struct q6asm_ape_cfg *cfg)
{
struct asm_ape_fmt_blk_v2 *fmt;
struct apr_pkt *pkt;
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, 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->compatible_version = cfg->compatible_version;
fmt->compression_level = cfg->compression_level;
fmt->format_flags = cfg->format_flags;
fmt->blocks_per_frame = cfg->blocks_per_frame;
fmt->final_frame_blocks = cfg->final_frame_blocks;
fmt->total_frames = cfg->total_frames;
fmt->bits_per_sample = cfg->bits_per_sample;
fmt->num_channels = cfg->num_channels;
fmt->sample_rate = cfg->sample_rate;
fmt->seek_table_present = cfg->seek_table_present;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
static int q6asm_stream_remove_silence(struct audio_client *ac, uint32_t stream_id,
uint32_t cmd,
uint32_t num_samples)
{
uint32_t *samples;
struct apr_pkt *pkt;
void *p;
int rc, pkt_size;
pkt_size = APR_HDR_SIZE + sizeof(uint32_t);
p = kzalloc(pkt_size, GFP_ATOMIC);
if (!p)
return -ENOMEM;
pkt = p;
samples = p + APR_HDR_SIZE;
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = cmd;
*samples = num_samples;
rc = apr_send_pkt(ac->adev, pkt);
if (rc == pkt_size)
rc = 0;
kfree(pkt);
return rc;
}
int q6asm_stream_remove_initial_silence(struct audio_client *ac,
uint32_t stream_id,
uint32_t initial_samples)
{
return q6asm_stream_remove_silence(ac, stream_id,
ASM_DATA_CMD_REMOVE_INITIAL_SILENCE,
initial_samples);
}
EXPORT_SYMBOL_GPL(q6asm_stream_remove_initial_silence);
int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id,
uint32_t trailing_samples)
{
return q6asm_stream_remove_silence(ac, stream_id,
ASM_DATA_CMD_REMOVE_TRAILING_SILENCE,
trailing_samples);
}
EXPORT_SYMBOL_GPL(q6asm_stream_remove_trailing_silence);
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
* @ac: audio client pointer
* @stream_id: stream id
* @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 stream_id, 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, 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
* @stream_id: stream id
*
* Return: Will be an negative value on error or zero on success
*/
int q6asm_read(struct audio_client *ac, uint32_t stream_id)
{
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, 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 stream_id,
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, 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
* @stream_id: stream id
* @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 stream_id,
uint32_t format, uint16_t bits_per_sample)
{
return __q6asm_open_read(ac, stream_id, format, bits_per_sample);
}
EXPORT_SYMBOL_GPL(q6asm_open_read);
/**
* q6asm_write_async() - non blocking write
*
* @ac: audio client pointer
* @stream_id: stream id
* @len: length 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 stream_id, 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, stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.token = port->dsp_buf | (len << ASM_WRITE_TOKEN_LEN_SHIFT);
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;
write->flags = 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;
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, uint32_t stream_id, int cmd,
bool wait)
{
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
* @stream_id: stream id
* @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, uint32_t stream_id, int cmd)
{
return __q6asm_cmd(ac, stream_id, cmd, true);
}
EXPORT_SYMBOL_GPL(q6asm_cmd);
/**
* q6asm_cmd_nowait() - non blocking, run cmd on audio client
*
* @ac: audio client pointer
* @stream_id: stream id
* @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, uint32_t stream_id, int cmd)
{
return __q6asm_cmd(ac, stream_id, cmd, false);
}
EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
static int q6asm_probe(struct apr_device *adev)
{
struct device *dev = &adev->dev;
struct q6asm *q6asm;
q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
if (!q6asm)
return -ENOMEM;
q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
q6asm->dev = dev;
q6asm->adev = adev;
init_waitqueue_head(&q6asm->mem_wait);
spin_lock_init(&q6asm->slock);
dev_set_drvdata(dev, q6asm);
return devm_of_platform_populate(dev);
}
#ifdef CONFIG_OF
static const struct of_device_id q6asm_device_id[] = {
{ .compatible = "qcom,q6asm" },
{},
};
MODULE_DEVICE_TABLE(of, q6asm_device_id);
#endif
static struct apr_driver qcom_q6asm_driver = {
.probe = q6asm_probe,
.callback = q6asm_srvc_callback,
.driver = {
.name = "qcom-q6asm",
.of_match_table = of_match_ptr(q6asm_device_id),
},
};
module_apr_driver(qcom_q6asm_driver);
MODULE_DESCRIPTION("Q6 Audio Stream Manager driver");
MODULE_LICENSE("GPL v2");