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gerrit-public.fairphone.software / platform / vendor / opensource / audio-kernel / refs/heads/int/10/fp3 / . / dsp / q6asm.c
blob: 80aa2a6bbdb75dbb47f22a0bd9612c0bd05a67a4 [file] [log] [blame]
/*
* Copyright (c) 2012-2019, The Linux Foundation. All rights reserved.
* Author: Brian Swetland <swetland@google.com>
*
* This software is licensed under the terms of the GNU General Public
* License version 2, as published by the Free Software Foundation, and
* may be copied, distributed, and modified under those terms.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
*/
#include <linux/fs.h>
#include <linux/mutex.h>
#include <linux/wait.h>
#include <linux/miscdevice.h>
#include <linux/uaccess.h>
#include <linux/sched.h>
#include <linux/dma-mapping.h>
#include <linux/miscdevice.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/debugfs.h>
#include <linux/time.h>
#include <linux/atomic.h>
#include <linux/mm.h>
#include <asm/ioctls.h>
#include <linux/memory.h>
#include <sound/compress_params.h>
#include <dsp/msm_audio_ion.h>
#include <dsp/apr_audio-v2.h>
#include <dsp/audio_cal_utils.h>
#include <dsp/q6asm-v2.h>
#include <dsp/q6audio-v2.h>
#include <dsp/q6core.h>
#include "adsp_err.h"
#define TRUE 0x01
#define FALSE 0x00
#define SESSION_MAX 8
enum {
ASM_TOPOLOGY_CAL = 0,
ASM_CUSTOM_TOP_CAL,
ASM_AUDSTRM_CAL,
ASM_RTAC_APR_CAL,
ASM_MAX_CAL_TYPES
};
union asm_token_struct {
struct {
u8 stream_id;
u8 session_id;
u8 buf_index;
u8 flags;
} _token;
u32 token;
} __packed;
enum {
ASM_DIRECTION_OFFSET,
ASM_CMD_NO_WAIT_OFFSET,
/*
* Offset is limited to 7 because flags is stored in u8
* field in asm_token_structure defined above. The offset
* starts from 0.
*/
ASM_MAX_OFFSET = 7,
};
enum {
WAIT_CMD,
NO_WAIT_CMD
};
#define ASM_SET_BIT(n, x) (n |= 1 << x)
#define ASM_TEST_BIT(n, x) ((n >> x) & 1)
/* TODO, combine them together */
static DEFINE_MUTEX(session_lock);
struct asm_mmap {
atomic_t ref_cnt;
void *apr;
};
static struct asm_mmap this_mmap;
struct audio_session {
struct audio_client *ac;
spinlock_t session_lock;
struct mutex mutex_lock_per_session;
};
/* session id: 0 reserved */
static struct audio_session session[ASM_ACTIVE_STREAMS_ALLOWED + 1];
struct asm_buffer_node {
struct list_head list;
phys_addr_t buf_phys_addr;
uint32_t mmap_hdl;
};
static int32_t q6asm_srvc_callback(struct apr_client_data *data, void *priv);
static int32_t q6asm_callback(struct apr_client_data *data, void *priv);
static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg);
static void q6asm_add_hdr_custom_topology(struct audio_client *ac,
struct apr_hdr *hdr,
uint32_t pkt_size);
static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg);
static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
uint32_t bufsz, uint32_t bufcnt,
bool is_contiguous);
static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir);
static void q6asm_reset_buf_state(struct audio_client *ac);
static int q6asm_map_channels(u8 *channel_mapping, uint32_t channels,
bool use_back_flavor);
void *q6asm_mmap_apr_reg(void);
static int q6asm_is_valid_session(struct apr_client_data *data, void *priv);
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info);
/* for ASM custom topology */
static struct cal_type_data *cal_data[ASM_MAX_CAL_TYPES];
static struct audio_buffer common_buf[2];
static struct audio_client common_client;
static int set_custom_topology;
static int topology_map_handle;
struct generic_get_data_ {
int valid;
int is_inband;
int size_in_ints;
int ints[];
};
static struct generic_get_data_ *generic_get_data;
#ifdef CONFIG_DEBUG_FS
#define OUT_BUFFER_SIZE 56
#define IN_BUFFER_SIZE 24
static struct timeval out_cold_tv;
static struct timeval out_warm_tv;
static struct timeval out_cont_tv;
static struct timeval in_cont_tv;
static long out_enable_flag;
static long in_enable_flag;
static struct dentry *out_dentry;
static struct dentry *in_dentry;
static int in_cont_index;
/*This var is used to keep track of first write done for cold output latency */
static int out_cold_index;
static char *out_buffer;
static char *in_buffer;
static uint32_t adsp_reg_event_opcode[] = {
ASM_STREAM_CMD_REGISTER_PP_EVENTS,
ASM_STREAM_CMD_REGISTER_ENCDEC_EVENTS,
ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE };
static uint32_t adsp_raise_event_opcode[] = {
ASM_STREAM_PP_EVENT,
ASM_STREAM_CMD_ENCDEC_EVENTS,
ASM_IEC_61937_MEDIA_FMT_EVENT };
static int is_adsp_reg_event(uint32_t cmd)
{
int i;
for (i = 0; i < ARRAY_SIZE(adsp_reg_event_opcode); i++) {
if (cmd == adsp_reg_event_opcode[i])
return i;
}
return -EINVAL;
}
static int is_adsp_raise_event(uint32_t cmd)
{
int i;
for (i = 0; i < ARRAY_SIZE(adsp_raise_event_opcode); i++) {
if (cmd == adsp_raise_event_opcode[i])
return i;
}
return -EINVAL;
}
static inline void q6asm_set_flag_in_token(union asm_token_struct *asm_token,
int flag, int flag_offset)
{
if (flag)
ASM_SET_BIT(asm_token->_token.flags, flag_offset);
}
static inline int q6asm_get_flag_from_token(union asm_token_struct *asm_token,
int flag_offset)
{
return ASM_TEST_BIT(asm_token->_token.flags, flag_offset);
}
static inline void q6asm_update_token(u32 *token, u8 session_id, u8 stream_id,
u8 buf_index, u8 dir, u8 nowait_flag)
{
union asm_token_struct asm_token;
asm_token.token = 0;
asm_token._token.session_id = session_id;
asm_token._token.stream_id = stream_id;
asm_token._token.buf_index = buf_index;
q6asm_set_flag_in_token(&asm_token, dir, ASM_DIRECTION_OFFSET);
q6asm_set_flag_in_token(&asm_token, nowait_flag,
ASM_CMD_NO_WAIT_OFFSET);
*token = asm_token.token;
}
static inline uint32_t q6asm_get_pcm_format_id(uint32_t media_format_block_ver)
{
uint32_t pcm_format_id;
switch (media_format_block_ver) {
case PCM_MEDIA_FORMAT_V5:
pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5;
break;
case PCM_MEDIA_FORMAT_V4:
pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4;
break;
case PCM_MEDIA_FORMAT_V3:
pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3;
break;
case PCM_MEDIA_FORMAT_V2:
default:
pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
}
return pcm_format_id;
}
/*
* q6asm_get_buf_index_from_token:
* Retrieve buffer index from token.
*
* @token: token value sent to ASM service on q6.
* Returns buffer index in the read/write commands.
*/
uint8_t q6asm_get_buf_index_from_token(uint32_t token)
{
union asm_token_struct asm_token;
asm_token.token = token;
return asm_token._token.buf_index;
}
EXPORT_SYMBOL(q6asm_get_buf_index_from_token);
/*
* q6asm_get_stream_id_from_token:
* Retrieve stream id from token.
*
* @token: token value sent to ASM service on q6.
* Returns stream id.
*/
uint8_t q6asm_get_stream_id_from_token(uint32_t token)
{
union asm_token_struct asm_token;
asm_token.token = token;
return asm_token._token.stream_id;
}
EXPORT_SYMBOL(q6asm_get_stream_id_from_token);
static int audio_output_latency_dbgfs_open(struct inode *inode,
struct file *file)
{
file->private_data = inode->i_private;
return 0;
}
static ssize_t audio_output_latency_dbgfs_read(struct file *file,
char __user *buf, size_t count, loff_t *ppos)
{
if (out_buffer == NULL) {
pr_err("%s: out_buffer is null\n", __func__);
return 0;
}
if (count < OUT_BUFFER_SIZE) {
pr_err("%s: read size %d exceeds buf size %zd\n", __func__,
OUT_BUFFER_SIZE, count);
return 0;
}
snprintf(out_buffer, OUT_BUFFER_SIZE, "%ld,%ld,%ld,%ld,%ld,%ld,",
out_cold_tv.tv_sec, out_cold_tv.tv_usec, out_warm_tv.tv_sec,
out_warm_tv.tv_usec, out_cont_tv.tv_sec, out_cont_tv.tv_usec);
return simple_read_from_buffer(buf, OUT_BUFFER_SIZE, ppos,
out_buffer, OUT_BUFFER_SIZE);
}
static ssize_t audio_output_latency_dbgfs_write(struct file *file,
const char __user *buf, size_t count, loff_t *ppos)
{
char *temp;
if (count > 2*sizeof(char)) {
pr_err("%s: err count is more %zd\n", __func__, count);
return -EINVAL;
}
temp = kmalloc(2*sizeof(char), GFP_KERNEL);
out_cold_index = 0;
if (temp) {
if (copy_from_user(temp, buf, 2*sizeof(char))) {
pr_err("%s: copy from user failed for size %zd\n",
__func__, 2*sizeof(char));
kfree(temp);
return -EFAULT;
}
if (!kstrtol(temp, 10, &out_enable_flag)) {
kfree(temp);
return count;
}
kfree(temp);
}
return -EINVAL;
}
static const struct file_operations audio_output_latency_debug_fops = {
.open = audio_output_latency_dbgfs_open,
.read = audio_output_latency_dbgfs_read,
.write = audio_output_latency_dbgfs_write
};
static int audio_input_latency_dbgfs_open(struct inode *inode,
struct file *file)
{
file->private_data = inode->i_private;
return 0;
}
static ssize_t audio_input_latency_dbgfs_read(struct file *file,
char __user *buf, size_t count, loff_t *ppos)
{
if (in_buffer == NULL) {
pr_err("%s: in_buffer is null\n", __func__);
return 0;
}
if (count < IN_BUFFER_SIZE) {
pr_err("%s: read size %d exceeds buf size %zd\n", __func__,
IN_BUFFER_SIZE, count);
return 0;
}
snprintf(in_buffer, IN_BUFFER_SIZE, "%ld,%ld,",
in_cont_tv.tv_sec, in_cont_tv.tv_usec);
return simple_read_from_buffer(buf, IN_BUFFER_SIZE, ppos,
in_buffer, IN_BUFFER_SIZE);
}
static ssize_t audio_input_latency_dbgfs_write(struct file *file,
const char __user *buf, size_t count, loff_t *ppos)
{
char *temp;
if (count > 2*sizeof(char)) {
pr_err("%s: err count is more %zd\n", __func__, count);
return -EINVAL;
}
temp = kmalloc(2*sizeof(char), GFP_KERNEL);
if (temp) {
if (copy_from_user(temp, buf, 2*sizeof(char))) {
pr_err("%s: copy from user failed for size %zd\n",
__func__, 2*sizeof(char));
kfree(temp);
return -EFAULT;
}
if (!kstrtol(temp, 10, &in_enable_flag)) {
kfree(temp);
return count;
}
kfree(temp);
}
return -EINVAL;
}
static const struct file_operations audio_input_latency_debug_fops = {
.open = audio_input_latency_dbgfs_open,
.read = audio_input_latency_dbgfs_read,
.write = audio_input_latency_dbgfs_write
};
static void config_debug_fs_write_cb(void)
{
if (out_enable_flag) {
/* For first Write done log the time and reset
* out_cold_index
*/
if (out_cold_index != 1) {
do_gettimeofday(&out_cold_tv);
pr_debug("COLD: apr_send_pkt at %ld sec %ld microsec\n",
out_cold_tv.tv_sec,
out_cold_tv.tv_usec);
out_cold_index = 1;
}
pr_debug("%s: out_enable_flag %ld\n",
__func__, out_enable_flag);
}
}
static void config_debug_fs_read_cb(void)
{
if (in_enable_flag) {
/* when in_cont_index == 7, DSP would be
* writing into the 8th 512 byte buffer and this
* timestamp is tapped here.Once done it then writes
* to 9th 512 byte buffer.These two buffers(8th, 9th)
* reach the test application in 5th iteration and that
* timestamp is tapped at user level. The difference
* of these two timestamps gives us the time between
* the time at which dsp started filling the sample
* required and when it reached the test application.
* Hence continuous input latency
*/
if (in_cont_index == 7) {
do_gettimeofday(&in_cont_tv);
pr_info("%s: read buffer at %ld sec %ld microsec\n",
__func__,
in_cont_tv.tv_sec, in_cont_tv.tv_usec);
}
in_cont_index++;
}
}
static void config_debug_fs_reset_index(void)
{
in_cont_index = 0;
}
static void config_debug_fs_run(void)
{
if (out_enable_flag) {
do_gettimeofday(&out_cold_tv);
pr_debug("%s: COLD apr_send_pkt at %ld sec %ld microsec\n",
__func__, out_cold_tv.tv_sec, out_cold_tv.tv_usec);
}
}
static void config_debug_fs_write(struct audio_buffer *ab)
{
if (out_enable_flag) {
char zero_pattern[2] = {0x00, 0x00};
/* If First two byte is non zero and last two byte
* is zero then it is warm output pattern
*/
if ((strcmp(((char *)ab->data), zero_pattern)) &&
(!strcmp(((char *)ab->data + 2), zero_pattern))) {
do_gettimeofday(&out_warm_tv);
pr_debug("%s: WARM:apr_send_pkt at %ld sec %ld microsec\n",
__func__,
out_warm_tv.tv_sec,
out_warm_tv.tv_usec);
pr_debug("%s: Warm Pattern Matched\n", __func__);
}
/* If First two byte is zero and last two byte is
* non zero then it is cont output pattern
*/
else if ((!strcmp(((char *)ab->data), zero_pattern))
&& (strcmp(((char *)ab->data + 2), zero_pattern))) {
do_gettimeofday(&out_cont_tv);
pr_debug("%s: CONT:apr_send_pkt at %ld sec %ld microsec\n",
__func__,
out_cont_tv.tv_sec,
out_cont_tv.tv_usec);
pr_debug("%s: Cont Pattern Matched\n", __func__);
}
}
}
static void config_debug_fs_init(void)
{
out_buffer = kzalloc(OUT_BUFFER_SIZE, GFP_KERNEL);
if (out_buffer == NULL)
goto outbuf_fail;
in_buffer = kzalloc(IN_BUFFER_SIZE, GFP_KERNEL);
if (in_buffer == NULL)
goto inbuf_fail;
out_dentry = debugfs_create_file("audio_out_latency_measurement_node",
0664,
NULL, NULL, &audio_output_latency_debug_fops);
if (IS_ERR(out_dentry)) {
pr_err("%s: debugfs_create_file failed\n", __func__);
goto file_fail;
}
in_dentry = debugfs_create_file("audio_in_latency_measurement_node",
0664,
NULL, NULL, &audio_input_latency_debug_fops);
if (IS_ERR(in_dentry)) {
pr_err("%s: debugfs_create_file failed\n", __func__);
goto file_fail;
}
return;
file_fail:
kfree(in_buffer);
inbuf_fail:
kfree(out_buffer);
outbuf_fail:
in_buffer = NULL;
out_buffer = NULL;
}
#else
static void config_debug_fs_write(struct audio_buffer *ab)
{
}
static void config_debug_fs_run(void)
{
}
static void config_debug_fs_reset_index(void)
{
}
static void config_debug_fs_read_cb(void)
{
}
static void config_debug_fs_write_cb(void)
{
}
static void config_debug_fs_init(void)
{
}
#endif
int q6asm_mmap_apr_dereg(void)
{
int c;
c = atomic_sub_return(1, &this_mmap.ref_cnt);
if (c == 0) {
apr_deregister(this_mmap.apr);
common_client.mmap_apr = NULL;
pr_debug("%s: APR De-Register common port\n", __func__);
} else if (c < 0) {
pr_err("%s: APR Common Port Already Closed %d\n",
__func__, c);
atomic_set(&this_mmap.ref_cnt, 0);
}
return 0;
}
static int q6asm_session_alloc(struct audio_client *ac)
{
int n;
for (n = 1; n <= ASM_ACTIVE_STREAMS_ALLOWED; n++) {
if (!(session[n].ac)) {
session[n].ac = ac;
return n;
}
}
pr_err("%s: session not available\n", __func__);
return -ENOMEM;
}
static int q6asm_get_session_id_from_audio_client(struct audio_client *ac)
{
int n;
for (n = 1; n <= ASM_ACTIVE_STREAMS_ALLOWED; n++) {
if (session[n].ac == ac)
return n;
}
pr_debug("%s: cannot find matching audio client. ac = %pK\n",
__func__, ac);
return 0;
}
static bool q6asm_is_valid_audio_client(struct audio_client *ac)
{
return q6asm_get_session_id_from_audio_client(ac) ? 1 : 0;
}
static void q6asm_session_free(struct audio_client *ac)
{
int session_id;
unsigned long flags = 0;
pr_debug("%s: sessionid[%d]\n", __func__, ac->session);
session_id = ac->session;
mutex_lock(&session[session_id].mutex_lock_per_session);
rtac_remove_popp_from_adm_devices(ac->session);
spin_lock_irqsave(&(session[session_id].session_lock), flags);
session[ac->session].ac = NULL;
ac->session = 0;
ac->perf_mode = LEGACY_PCM_MODE;
ac->fptr_cache_ops = NULL;
ac->cb = NULL;
ac->priv = NULL;
kfree(ac);
ac = NULL;
spin_unlock_irqrestore(&(session[session_id].session_lock), flags);
mutex_unlock(&session[session_id].mutex_lock_per_session);
}
static uint32_t q6asm_get_next_buf(struct audio_client *ac,
uint32_t curr_buf, uint32_t max_buf_cnt)
{
dev_vdbg(ac->dev, "%s: curr_buf = %d, max_buf_cnt = %d\n",
__func__, curr_buf, max_buf_cnt);
curr_buf += 1;
return (curr_buf >= max_buf_cnt) ? 0 : curr_buf;
}
static int q6asm_map_cal_memory(int32_t cal_type,
struct cal_block_data *cal_block)
{
int result = 0;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
if (cal_block == NULL) {
pr_err("%s: cal_block is NULL!\n",
__func__);
goto done;
}
if (cal_block->cal_data.paddr == 0) {
pr_debug("%s: No address to map!\n",
__func__);
goto done;
}
common_client.mmap_apr = q6asm_mmap_apr_reg();
if (common_client.mmap_apr == NULL) {
pr_err("%s: q6asm_mmap_apr_reg failed\n",
__func__);
result = -EPERM;
goto done;
}
common_client.apr = common_client.mmap_apr;
if (cal_block->map_data.map_size == 0) {
pr_debug("%s: map size is 0!\n",
__func__);
goto done;
}
/* Use second asm buf to map memory */
if (common_client.port[IN].buf == NULL) {
pr_err("%s: common buf is NULL\n",
__func__);
result = -EINVAL;
goto done;
}
common_client.port[IN].buf->phys = cal_block->cal_data.paddr;
result = q6asm_memory_map_regions(&common_client,
IN, cal_block->map_data.map_size, 1, 1);
if (result < 0) {
pr_err("%s: mmap did not work! size = %zd result %d\n",
__func__,
cal_block->map_data.map_size, result);
pr_debug("%s: mmap did not work! addr = 0x%pK, size = %zd\n",
__func__,
&cal_block->cal_data.paddr,
cal_block->map_data.map_size);
goto done;
}
list_for_each_safe(ptr, next,
&common_client.port[IN].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == cal_block->cal_data.paddr) {
cal_block->map_data.q6map_handle = buf_node->mmap_hdl;
break;
}
}
done:
return result;
}
static int remap_cal_data(int32_t cal_type, struct cal_block_data *cal_block)
{
int ret = 0;
if (cal_block->map_data.ion_client == NULL) {
pr_err("%s: No ION allocation for cal type %d!\n",
__func__, cal_type);
ret = -EINVAL;
goto done;
}
if ((cal_block->map_data.map_size > 0) &&
(cal_block->map_data.q6map_handle == 0)) {
ret = q6asm_map_cal_memory(cal_type, cal_block);
if (ret < 0) {
pr_err("%s: mmap did not work! size = %zd ret %d\n",
__func__, cal_block->map_data.map_size, ret);
goto done;
}
}
done:
return ret;
}
static int q6asm_unmap_cal_memory(int32_t cal_type,
struct cal_block_data *cal_block)
{
int result = 0;
int result2 = 0;
if (cal_block == NULL) {
pr_err("%s: cal_block is NULL!\n",
__func__);
result = -EINVAL;
goto done;
}
if (cal_block->map_data.q6map_handle == 0) {
pr_debug("%s: No address to unmap!\n",
__func__);
result = -EINVAL;
goto done;
}
if (common_client.mmap_apr == NULL) {
common_client.mmap_apr = q6asm_mmap_apr_reg();
if (common_client.mmap_apr == NULL) {
pr_err("%s: q6asm_mmap_apr_reg failed\n",
__func__);
result = -EPERM;
goto done;
}
}
result2 = q6asm_memory_unmap_regions(&common_client, IN);
if (result2 < 0) {
pr_err("%s: unmap failed, err %d\n",
__func__, result2);
result = result2;
}
cal_block->map_data.q6map_handle = 0;
done:
return result;
}
int q6asm_unmap_cal_data(int cal_type, struct cal_block_data *cal_block)
{
int ret = 0;
if ((cal_block->map_data.map_size > 0) &&
(cal_block->map_data.q6map_handle != 0)) {
ret = q6asm_unmap_cal_memory(cal_type, cal_block);
if (ret < 0) {
pr_err("%s: unmap did not work! size = %zd ret %d\n",
__func__, cal_block->map_data.map_size, ret);
goto done;
}
}
done:
return ret;
}
int send_asm_custom_topology(struct audio_client *ac)
{
struct cal_block_data *cal_block = NULL;
struct cmd_set_topologies asm_top;
int result = 0;
int result1 = 0;
if (cal_data[ASM_CUSTOM_TOP_CAL] == NULL)
goto done;
mutex_lock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
if (!set_custom_topology)
goto unlock;
set_custom_topology = 0;
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_CUSTOM_TOP_CAL]);
if (cal_block == NULL || cal_utils_is_cal_stale(cal_block))
goto unlock;
if (cal_block->cal_data.size == 0) {
pr_debug("%s: No cal to send!\n", __func__);
goto unlock;
}
pr_debug("%s: Sending cal_index %d\n", __func__, ASM_CUSTOM_TOP_CAL);
result = remap_cal_data(ASM_CUST_TOPOLOGY_CAL_TYPE, cal_block);
if (result) {
pr_err("%s: Remap_cal_data failed for cal %d!\n",
__func__, ASM_CUSTOM_TOP_CAL);
goto unlock;
}
q6asm_add_hdr_custom_topology(ac, &asm_top.hdr, sizeof(asm_top));
atomic_set(&ac->mem_state, -1);
asm_top.hdr.opcode = ASM_CMD_ADD_TOPOLOGIES;
asm_top.payload_addr_lsw = lower_32_bits(cal_block->cal_data.paddr);
asm_top.payload_addr_msw = msm_audio_populate_upper_32_bits(
cal_block->cal_data.paddr);
asm_top.mem_map_handle = cal_block->map_data.q6map_handle;
asm_top.payload_size = cal_block->cal_data.size;
pr_debug("%s: Sending ASM_CMD_ADD_TOPOLOGIES payload = %pK, size = %d, map handle = 0x%x\n",
__func__, &cal_block->cal_data.paddr,
asm_top.payload_size, asm_top.mem_map_handle);
result = apr_send_pkt(ac->apr, (uint32_t *) &asm_top);
if (result < 0) {
pr_err("%s: Set topologies failed result %d\n",
__func__, result);
pr_debug("%s: Set topologies failed payload = 0x%pK\n",
__func__, &cal_block->cal_data.paddr);
goto unmap;
}
result = wait_event_timeout(ac->mem_wait,
(atomic_read(&ac->mem_state) >= 0), 5*HZ);
if (!result) {
pr_err("%s: Set topologies failed timeout\n", __func__);
pr_debug("%s: Set topologies failed after timedout payload = 0x%pK\n",
__func__, &cal_block->cal_data.paddr);
result = -ETIMEDOUT;
goto unmap;
}
if (atomic_read(&ac->mem_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->mem_state)));
result = adsp_err_get_lnx_err_code(
atomic_read(&ac->mem_state));
goto unmap;
}
unmap:
result1 = q6asm_unmap_cal_memory(ASM_CUST_TOPOLOGY_CAL_TYPE,
cal_block);
if (result1 < 0) {
result = result1;
pr_debug("%s: unmap cal failed! %d\n", __func__, result);
}
unlock:
mutex_unlock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
done:
return result;
}
int q6asm_map_rtac_block(struct rtac_cal_block_data *cal_block)
{
int result = 0;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
pr_debug("%s:\n", __func__);
if (cal_block == NULL) {
pr_err("%s: cal_block is NULL!\n",
__func__);
result = -EINVAL;
goto done;
}
if (cal_block->cal_data.paddr == 0) {
pr_debug("%s: No address to map!\n",
__func__);
result = -EINVAL;
goto done;
}
if (common_client.mmap_apr == NULL) {
common_client.mmap_apr = q6asm_mmap_apr_reg();
if (common_client.mmap_apr == NULL) {
pr_err("%s: q6asm_mmap_apr_reg failed\n",
__func__);
result = -EPERM;
goto done;
}
}
if (cal_block->map_data.map_size == 0) {
pr_debug("%s: map size is 0!\n",
__func__);
result = -EINVAL;
goto done;
}
/* Use second asm buf to map memory */
if (common_client.port[OUT].buf == NULL) {
pr_err("%s: common buf is NULL\n",
__func__);
result = -EINVAL;
goto done;
}
common_client.port[OUT].buf->phys = cal_block->cal_data.paddr;
result = q6asm_memory_map_regions(&common_client,
OUT, cal_block->map_data.map_size, 1, 1);
if (result < 0) {
pr_err("%s: mmap did not work! size = %d result %d\n",
__func__,
cal_block->map_data.map_size, result);
pr_debug("%s: mmap did not work! addr = 0x%pK, size = %d\n",
__func__,
&cal_block->cal_data.paddr,
cal_block->map_data.map_size);
goto done;
}
list_for_each_safe(ptr, next,
&common_client.port[OUT].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == cal_block->cal_data.paddr) {
cal_block->map_data.map_handle = buf_node->mmap_hdl;
break;
}
}
done:
return result;
}
int q6asm_unmap_rtac_block(uint32_t *mem_map_handle)
{
int result = 0;
int result2 = 0;
pr_debug("%s:\n", __func__);
if (mem_map_handle == NULL) {
pr_debug("%s: Map handle is NULL, nothing to unmap\n",
__func__);
goto done;
}
if (*mem_map_handle == 0) {
pr_debug("%s: Map handle is 0, nothing to unmap\n",
__func__);
goto done;
}
if (common_client.mmap_apr == NULL) {
common_client.mmap_apr = q6asm_mmap_apr_reg();
if (common_client.mmap_apr == NULL) {
pr_err("%s: q6asm_mmap_apr_reg failed\n",
__func__);
result = -EPERM;
goto done;
}
}
result2 = q6asm_memory_unmap_regions(&common_client, OUT);
if (result2 < 0) {
pr_err("%s: unmap failed, err %d\n",
__func__, result2);
result = result2;
} else {
*mem_map_handle = 0;
}
result2 = q6asm_mmap_apr_dereg();
if (result2 < 0) {
pr_err("%s: q6asm_mmap_apr_dereg failed, err %d\n",
__func__, result2);
result = result2;
}
done:
return result;
}
int q6asm_audio_client_buf_free(unsigned int dir,
struct audio_client *ac)
{
struct audio_port_data *port;
int cnt = 0;
int rc = 0;
pr_debug("%s: Session id %d\n", __func__, ac->session);
mutex_lock(&ac->cmd_lock);
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[dir];
if (!port->buf) {
pr_err("%s: buf NULL\n", __func__);
mutex_unlock(&ac->cmd_lock);
return 0;
}
cnt = port->max_buf_cnt - 1;
if (cnt >= 0) {
rc = q6asm_memory_unmap_regions(ac, dir);
if (rc < 0)
pr_err("%s: Memory_unmap_regions failed %d\n",
__func__, rc);
}
while (cnt >= 0) {
if (port->buf[cnt].data) {
if (!rc || atomic_read(&ac->reset))
msm_audio_ion_free(
port->buf[cnt].client,
port->buf[cnt].handle);
port->buf[cnt].client = NULL;
port->buf[cnt].handle = NULL;
port->buf[cnt].data = NULL;
port->buf[cnt].phys = 0;
--(port->max_buf_cnt);
}
--cnt;
}
kfree(port->buf);
port->buf = NULL;
}
mutex_unlock(&ac->cmd_lock);
return 0;
}
/**
* q6asm_audio_client_buf_free_contiguous -
* frees the memory buffers for ASM
*
* @dir: RX or TX direction
* @ac: audio client handle
*
* Returns 0 on success or error on failure
*/
int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
struct audio_client *ac)
{
struct audio_port_data *port;
int cnt = 0;
int rc = 0;
pr_debug("%s: Session id %d\n", __func__, ac->session);
mutex_lock(&ac->cmd_lock);
port = &ac->port[dir];
if (!port->buf) {
mutex_unlock(&ac->cmd_lock);
return 0;
}
cnt = port->max_buf_cnt - 1;
if (cnt >= 0) {
rc = q6asm_memory_unmap(ac, port->buf[0].phys, dir);
if (rc < 0)
pr_err("%s: Memory_unmap_regions failed %d\n",
__func__, rc);
}
if (port->buf[0].data) {
pr_debug("%s: data[%pK]phys[%pK][%pK] , client[%pK] handle[%pK]\n",
__func__,
port->buf[0].data,
&port->buf[0].phys,
&port->buf[0].phys,
port->buf[0].client,
port->buf[0].handle);
if (!rc || atomic_read(&ac->reset))
msm_audio_ion_free(port->buf[0].client,
port->buf[0].handle);
port->buf[0].client = NULL;
port->buf[0].handle = NULL;
}
while (cnt >= 0) {
port->buf[cnt].data = NULL;
port->buf[cnt].phys = 0;
cnt--;
}
port->max_buf_cnt = 0;
kfree(port->buf);
port->buf = NULL;
mutex_unlock(&ac->cmd_lock);
return 0;
}
EXPORT_SYMBOL(q6asm_audio_client_buf_free_contiguous);
/**
* q6asm_audio_client_free -
* frees the audio client for ASM
*
* @ac: audio client handle
*
*/
void q6asm_audio_client_free(struct audio_client *ac)
{
int loopcnt;
struct audio_port_data *port;
if (!ac) {
pr_err("%s: ac %pK\n", __func__, ac);
return;
}
if (!ac->session) {
pr_err("%s: ac session invalid\n", __func__);
return;
}
mutex_lock(&session_lock);
pr_debug("%s: Session id %d\n", __func__, ac->session);
if (ac->io_mode & SYNC_IO_MODE) {
for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
port = &ac->port[loopcnt];
if (!port->buf)
continue;
pr_debug("%s: loopcnt = %d\n",
__func__, loopcnt);
q6asm_audio_client_buf_free(loopcnt, ac);
}
}
rtac_set_asm_handle(ac->session, NULL);
apr_deregister(ac->apr2);
apr_deregister(ac->apr);
q6asm_mmap_apr_dereg();
ac->apr2 = NULL;
ac->apr = NULL;
ac->mmap_apr = NULL;
q6asm_session_free(ac);
pr_debug("%s: APR De-Register\n", __func__);
/*done:*/
mutex_unlock(&session_lock);
}
EXPORT_SYMBOL(q6asm_audio_client_free);
/**
* q6asm_set_io_mode -
* Update IO mode for ASM
*
* @ac: audio client handle
* @mode1: IO mode to update
*
* Returns 0 on success or error on failure
*/
int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode1)
{
uint32_t mode;
int ret = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
ac->io_mode &= 0xFF00;
mode = (mode1 & 0xF);
pr_debug("%s: ac->mode after anding with FF00:0x%x,\n",
__func__, ac->io_mode);
if ((mode == ASYNC_IO_MODE) || (mode == SYNC_IO_MODE)) {
ac->io_mode |= mode1;
pr_debug("%s: Set Mode to 0x%x\n", __func__, ac->io_mode);
} else {
pr_err("%s: Not an valid IO Mode:%d\n", __func__, ac->io_mode);
ret = -EINVAL;
}
return ret;
}
EXPORT_SYMBOL(q6asm_set_io_mode);
void *q6asm_mmap_apr_reg(void)
{
if ((atomic_read(&this_mmap.ref_cnt) == 0) ||
(this_mmap.apr == NULL)) {
this_mmap.apr = apr_register("ADSP", "ASM",
(apr_fn)q6asm_srvc_callback,
0x0FFFFFFFF, &this_mmap);
if (this_mmap.apr == NULL) {
pr_debug("%s: Unable to register APR ASM common port\n",
__func__);
goto fail;
}
}
atomic_inc(&this_mmap.ref_cnt);
return this_mmap.apr;
fail:
return NULL;
}
/**
* q6asm_send_stream_cmd -
* command to send for ASM stream
*
* @ac: audio client handle
* @data: event data
*
* Returns 0 on success or error on failure
*/
int q6asm_send_stream_cmd(struct audio_client *ac,
struct msm_adsp_event_data *data)
{
char *asm_params = NULL;
struct apr_hdr hdr;
int rc, session_id = 0;
uint32_t sz = 0;
uint64_t actual_sz = 0;
if (!data || !ac) {
pr_err("%s: %s is NULL\n", __func__,
(!data) ? "data" : "ac");
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
if (data->event_type >= ARRAY_SIZE(adsp_reg_event_opcode)) {
pr_err("%s: event %u out of boundary of array size of (%lu)\n",
__func__, data->event_type,
(long)ARRAY_SIZE(adsp_reg_event_opcode));
rc = -EINVAL;
goto done;
}
actual_sz = sizeof(struct apr_hdr) + data->payload_len;
if (actual_sz > U32_MAX) {
pr_err("%s: payload size 0x%X exceeds limit\n",
__func__, data->payload_len);
rc = -EINVAL;
goto done;
}
sz = (uint32_t)actual_sz;
asm_params = kzalloc(sz, GFP_KERNEL);
if (!asm_params) {
rc = -ENOMEM;
goto done;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_send_param;
}
q6asm_add_hdr_async(ac, &hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
hdr.opcode = adsp_reg_event_opcode[data->event_type];
memcpy(asm_params, &hdr, sizeof(struct apr_hdr));
memcpy(asm_params + sizeof(struct apr_hdr),
data->payload, data->payload_len);
rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params);
if (rc < 0) {
pr_err("%s: stream event cmd apr pkt failed\n", __func__);
rc = -EINVAL;
goto fail_send_param;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 1 * HZ);
if (!rc) {
pr_err("%s: timeout for stream event cmd resp\n", __func__);
rc = -ETIMEDOUT;
goto fail_send_param;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] for stream event cmd\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_send_param;
}
rc = 0;
fail_send_param:
mutex_unlock(&session[session_id].mutex_lock_per_session);
kfree(asm_params);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_send_stream_cmd);
/**
* q6asm_audio_client_alloc -
* Alloc audio client for ASM
*
* @cb: callback fn
* @priv: private data
*
* Returns ac pointer on success or NULL on failure
*/
struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv)
{
struct audio_client *ac;
int n;
int lcnt = 0;
int rc = 0;
ac = kzalloc(sizeof(struct audio_client), GFP_KERNEL);
if (!ac)
return NULL;
mutex_lock(&session_lock);
n = q6asm_session_alloc(ac);
if (n <= 0) {
pr_err("%s: ASM Session alloc fail n=%d\n", __func__, n);
mutex_unlock(&session_lock);
kfree(ac);
goto fail_session;
}
ac->session = n;
ac->cb = cb;
ac->path_delay = UINT_MAX;
ac->priv = priv;
ac->io_mode = SYNC_IO_MODE;
ac->perf_mode = LEGACY_PCM_MODE;
ac->fptr_cache_ops = NULL;
/* DSP expects stream id from 1 */
ac->stream_id = 1;
ac->apr = apr_register("ADSP", "ASM",
(apr_fn)q6asm_callback,
((ac->session) << 8 | 0x0001),
ac);
if (ac->apr == NULL) {
pr_err("%s: Registration with APR failed\n", __func__);
mutex_unlock(&session_lock);
goto fail_apr1;
}
ac->apr2 = apr_register("ADSP", "ASM",
(apr_fn)q6asm_callback,
((ac->session) << 8 | 0x0002),
ac);
if (ac->apr2 == NULL) {
pr_err("%s: Registration with APR-2 failed\n", __func__);
mutex_unlock(&session_lock);
goto fail_apr2;
}
rtac_set_asm_handle(n, ac->apr);
pr_debug("%s: Registering the common port with APR\n", __func__);
ac->mmap_apr = q6asm_mmap_apr_reg();
if (ac->mmap_apr == NULL) {
mutex_unlock(&session_lock);
goto fail_mmap;
}
init_waitqueue_head(&ac->cmd_wait);
init_waitqueue_head(&ac->time_wait);
init_waitqueue_head(&ac->mem_wait);
atomic_set(&ac->time_flag, 1);
atomic_set(&ac->reset, 0);
INIT_LIST_HEAD(&ac->port[0].mem_map_handle);
INIT_LIST_HEAD(&ac->port[1].mem_map_handle);
pr_debug("%s: mem_map_handle list init'ed\n", __func__);
mutex_init(&ac->cmd_lock);
for (lcnt = 0; lcnt <= OUT; lcnt++) {
mutex_init(&ac->port[lcnt].lock);
spin_lock_init(&ac->port[lcnt].dsp_lock);
}
atomic_set(&ac->cmd_state, 0);
atomic_set(&ac->cmd_state_pp, 0);
atomic_set(&ac->mem_state, 0);
rc = send_asm_custom_topology(ac);
if (rc < 0) {
mutex_unlock(&session_lock);
goto fail_mmap;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
mutex_unlock(&session_lock);
return ac;
fail_mmap:
apr_deregister(ac->apr2);
fail_apr2:
apr_deregister(ac->apr);
fail_apr1:
q6asm_session_free(ac);
fail_session:
return NULL;
}
EXPORT_SYMBOL(q6asm_audio_client_alloc);
/**
* q6asm_get_audio_client -
* Retrieve audio client for ASM
*
* @session_id: ASM session id
*
* Returns valid pointer on success or NULL on failure
*/
struct audio_client *q6asm_get_audio_client(int session_id)
{
if (session_id == ASM_CONTROL_SESSION)
return &common_client;
if ((session_id <= 0) || (session_id > ASM_ACTIVE_STREAMS_ALLOWED)) {
pr_err("%s: invalid session: %d\n", __func__, session_id);
goto err;
}
if (!(session[session_id].ac)) {
pr_err("%s: session not active: %d\n", __func__, session_id);
goto err;
}
return session[session_id].ac;
err:
return NULL;
}
EXPORT_SYMBOL(q6asm_get_audio_client);
/**
* q6asm_audio_client_buf_alloc -
* Allocs memory from ION for ASM
*
* @dir: RX or TX direction
* @ac: Audio client handle
* @bufsz: size of each buffer
* @bufcnt: number of buffers to alloc
*
* Returns 0 on success or error on failure
*/
int q6asm_audio_client_buf_alloc(unsigned int dir,
struct audio_client *ac,
unsigned int bufsz,
uint32_t bufcnt)
{
int cnt = 0;
int rc = 0;
struct audio_buffer *buf;
size_t len;
if (!(ac) || !(bufsz) || ((dir != IN) && (dir != OUT))) {
pr_err("%s: ac %pK bufsz %d dir %d\n", __func__, ac, bufsz,
dir);
return -EINVAL;
}
pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n", __func__, ac->session,
bufsz, bufcnt);
if (ac->session <= 0 || ac->session > 8) {
pr_err("%s: Session ID is invalid, session = %d\n", __func__,
ac->session);
goto fail;
}
if (ac->io_mode & SYNC_IO_MODE) {
if (ac->port[dir].buf) {
pr_debug("%s: buffer already allocated\n", __func__);
return 0;
}
mutex_lock(&ac->cmd_lock);
if (bufcnt > (U32_MAX/sizeof(struct audio_buffer))) {
pr_err("%s: Buffer size overflows", __func__);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
GFP_KERNEL);
if (!buf) {
mutex_unlock(&ac->cmd_lock);
goto fail;
}
ac->port[dir].buf = buf;
while (cnt < bufcnt) {
if (bufsz > 0) {
if (!buf[cnt].data) {
rc = msm_audio_ion_alloc("asm_client",
&buf[cnt].client, &buf[cnt].handle,
bufsz,
(ion_phys_addr_t *)&buf[cnt].phys,
&len,
&buf[cnt].data);
if (rc) {
pr_err("%s: ION Get Physical for AUDIO failed, rc = %d\n",
__func__, rc);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
buf[cnt].used = 1;
buf[cnt].size = bufsz;
buf[cnt].actual_size = bufsz;
pr_debug("%s: data[%pK]phys[%pK][%pK]\n",
__func__,
buf[cnt].data,
&buf[cnt].phys,
&buf[cnt].phys);
cnt++;
}
}
}
ac->port[dir].max_buf_cnt = cnt;
mutex_unlock(&ac->cmd_lock);
rc = q6asm_memory_map_regions(ac, dir, bufsz, cnt, 0);
if (rc < 0) {
pr_err("%s: CMD Memory_map_regions failed %d for size %d\n",
__func__, rc, bufsz);
goto fail;
}
}
return 0;
fail:
q6asm_audio_client_buf_free(dir, ac);
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_audio_client_buf_alloc);
/**
* q6asm_audio_client_buf_alloc_contiguous -
* Alloc contiguous memory from ION for ASM
*
* @dir: RX or TX direction
* @ac: Audio client handle
* @bufsz: size of each buffer
* @bufcnt: number of buffers to alloc
*
* Returns 0 on success or error on failure
*/
int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir,
struct audio_client *ac,
unsigned int bufsz,
unsigned int bufcnt)
{
int cnt = 0;
int rc = 0;
struct audio_buffer *buf;
size_t len;
int bytes_to_alloc;
if (!(ac) || ((dir != IN) && (dir != OUT))) {
pr_err("%s: ac %pK dir %d\n", __func__, ac, dir);
return -EINVAL;
}
pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n",
__func__, ac->session,
bufsz, bufcnt);
if (ac->session <= 0 || ac->session > 8) {
pr_err("%s: Session ID is invalid, session = %d\n", __func__,
ac->session);
goto fail;
}
if (ac->port[dir].buf) {
pr_err("%s: buffer already allocated\n", __func__);
return 0;
}
if (bufcnt == 0) {
pr_err("%s: invalid buffer count\n", __func__);
return -EINVAL;
}
mutex_lock(&ac->cmd_lock);
buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
GFP_KERNEL);
if (!buf) {
pr_err("%s: buffer allocation failed\n", __func__);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
ac->port[dir].buf = buf;
/* check for integer overflow */
if ((bufcnt > 0) && ((INT_MAX / bufcnt) < bufsz)) {
pr_err("%s: integer overflow\n", __func__);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
bytes_to_alloc = bufsz * bufcnt;
/* The size to allocate should be multiple of 4K bytes */
bytes_to_alloc = PAGE_ALIGN(bytes_to_alloc);
rc = msm_audio_ion_alloc("asm_client", &buf[0].client, &buf[0].handle,
bytes_to_alloc,
(ion_phys_addr_t *)&buf[0].phys, &len,
&buf[0].data);
if (rc) {
pr_err("%s: Audio ION alloc is failed, rc = %d\n",
__func__, rc);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
buf[0].used = dir ^ 1;
buf[0].size = bufsz;
buf[0].actual_size = bufsz;
cnt = 1;
while (cnt < bufcnt) {
if (bufsz > 0) {
buf[cnt].data = buf[0].data + (cnt * bufsz);
buf[cnt].phys = buf[0].phys + (cnt * bufsz);
if (!buf[cnt].data) {
pr_err("%s: Buf alloc failed\n",
__func__);
mutex_unlock(&ac->cmd_lock);
goto fail;
}
buf[cnt].used = dir ^ 1;
buf[cnt].size = bufsz;
buf[cnt].actual_size = bufsz;
pr_debug("%s: data[%pK]phys[%pK][%pK]\n",
__func__,
buf[cnt].data,
&buf[cnt].phys,
&buf[cnt].phys);
}
cnt++;
}
ac->port[dir].max_buf_cnt = cnt;
mutex_unlock(&ac->cmd_lock);
rc = q6asm_memory_map_regions(ac, dir, bufsz, cnt, 1);
if (rc < 0) {
pr_err("%s: CMD Memory_map_regions failed %d for size %d\n",
__func__, rc, bufsz);
goto fail;
}
return 0;
fail:
q6asm_audio_client_buf_free_contiguous(dir, ac);
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_audio_client_buf_alloc_contiguous);
static int32_t q6asm_srvc_callback(struct apr_client_data *data, void *priv)
{
uint32_t dir = 0;
uint32_t i = IN;
uint32_t *payload;
unsigned long dsp_flags = 0;
unsigned long flags = 0;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
union asm_token_struct asm_token;
struct audio_client *ac = NULL;
struct audio_port_data *port;
int session_id;
if (!data) {
pr_err("%s: Invalid CB\n", __func__);
return 0;
}
payload = data->payload;
if (data->opcode == RESET_EVENTS) {
pr_debug("%s: Reset event is received: %d %d apr[%pK]\n",
__func__,
data->reset_event,
data->reset_proc,
this_mmap.apr);
atomic_set(&this_mmap.ref_cnt, 0);
apr_reset(this_mmap.apr);
this_mmap.apr = NULL;
for (; i <= OUT; i++) {
list_for_each_safe(ptr, next,
&common_client.port[i].mem_map_handle) {
buf_node = list_entry(ptr,
struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr ==
common_client.port[i].buf->phys) {
list_del(&buf_node->list);
kfree(buf_node);
}
}
pr_debug("%s: Clearing custom topology\n", __func__);
}
cal_utils_clear_cal_block_q6maps(ASM_MAX_CAL_TYPES, cal_data);
common_client.mmap_apr = NULL;
mutex_lock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
set_custom_topology = 1;
mutex_unlock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
topology_map_handle = 0;
rtac_clear_mapping(ASM_RTAC_CAL);
return 0;
}
asm_token.token = data->token;
session_id = asm_token._token.session_id;
if ((session_id > 0 && session_id <= ASM_ACTIVE_STREAMS_ALLOWED))
spin_lock_irqsave(&(session[session_id].session_lock), flags);
ac = q6asm_get_audio_client(session_id);
dir = q6asm_get_flag_from_token(&asm_token, ASM_DIRECTION_OFFSET);
if (!ac) {
pr_debug("%s: session[%d] already freed\n",
__func__, session_id);
if ((session_id > 0 &&
session_id <= ASM_ACTIVE_STREAMS_ALLOWED))
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
if (data->payload_size >= 2 * sizeof(uint32_t)) {
pr_debug("%s:ptr0[0x%x]ptr1[0x%x]opcode[0x%x] token[0x%x]payload_s[%d] src[%d] dest[%d]sid[%d]dir[%d]\n",
__func__, payload[0], payload[1], data->opcode,
data->token, data->payload_size, data->src_port,
data->dest_port, asm_token._token.session_id, dir);
pr_debug("%s:Payload = [0x%x] status[0x%x]\n",
__func__, payload[0], payload[1]);
} else if (data->payload_size == sizeof(uint32_t)) {
pr_debug("%s:ptr0[0x%x]opcode[0x%x] token[0x%x]payload_s[%d] src[%d] dest[%d]sid[%d]dir[%d]\n",
__func__, payload[0], data->opcode,
data->token, data->payload_size, data->src_port,
data->dest_port, asm_token._token.session_id, dir);
pr_debug("%s:Payload = [0x%x]\n",
__func__, payload[0]);
}
if (data->opcode == APR_BASIC_RSP_RESULT) {
switch (payload[0]) {
case ASM_CMD_SHARED_MEM_MAP_REGIONS:
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
case ASM_CMD_ADD_TOPOLOGIES:
if (data->payload_size >=
2 * sizeof(uint32_t)
&& payload[1] != 0) {
pr_err("%s: cmd = 0x%x returned error = 0x%x sid:%d\n",
__func__, payload[0], payload[1],
asm_token._token.session_id);
if (payload[0] ==
ASM_CMD_SHARED_MEM_UNMAP_REGIONS)
atomic_set(&ac->unmap_cb_success, 0);
atomic_set(&ac->mem_state, payload[1]);
wake_up(&ac->mem_wait);
} else {
if (payload[0] ==
ASM_CMD_SHARED_MEM_UNMAP_REGIONS)
atomic_set(&ac->unmap_cb_success, 1);
}
if (atomic_cmpxchg(&ac->mem_state, -1, 0) == -1)
wake_up(&ac->mem_wait);
if (data->payload_size >= 2 * sizeof(uint32_t))
dev_vdbg(ac->dev, "%s: Payload = [0x%x] status[0x%x]\n",
__func__, payload[0], payload[1]);
else
dev_vdbg(ac->dev, "%s: Payload size of %d is less than expected.\n",
__func__, data->payload_size);
break;
default:
pr_debug("%s: command[0x%x] not expecting rsp\n",
__func__, payload[0]);
break;
}
if ((session_id > 0 &&
session_id <= ASM_ACTIVE_STREAMS_ALLOWED))
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
if (dir != IN && dir != OUT) {
pr_err("%s: Invalid audio port index: %d\n", __func__, dir);
if ((session_id > 0 && session_id <= SESSION_MAX))
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
port = &ac->port[dir];
switch (data->opcode) {
case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:{
pr_debug("%s:PL#0[0x%x] dir=0x%x s_id=0x%x\n",
__func__, payload[0], dir, asm_token._token.session_id);
spin_lock_irqsave(&port->dsp_lock, dsp_flags);
if (atomic_cmpxchg(&ac->mem_state, -1, 0) == -1) {
ac->port[dir].tmp_hdl = payload[0];
wake_up(&ac->mem_wait);
}
spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
break;
}
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:{
if (data->payload_size >= 2 * sizeof(uint32_t))
pr_debug("%s: PL#0[0x%x]PL#1 [0x%x]\n",
__func__, payload[0], payload[1]);
else
pr_debug("%s: Payload size of %d is less than expected.\n",
__func__, data->payload_size);
spin_lock_irqsave(&port->dsp_lock, dsp_flags);
if (atomic_cmpxchg(&ac->mem_state, -1, 0) == -1)
wake_up(&ac->mem_wait);
spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
break;
}
default:
if (data->payload_size >= 2 * sizeof(uint32_t))
pr_debug("%s: command[0x%x]success [0x%x]\n",
__func__, payload[0], payload[1]);
else
pr_debug("%s: Payload size of %d is less than expected.\n",
__func__, data->payload_size);
}
if (ac->cb)
ac->cb(data->opcode, data->token,
data->payload, ac->priv);
if ((session_id > 0 && session_id <= ASM_ACTIVE_STREAMS_ALLOWED))
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
static void q6asm_process_mtmx_get_param_rsp(struct audio_client *ac,
struct asm_mtmx_strtr_get_params_cmdrsp *cmdrsp)
{
struct asm_session_mtmx_strtr_param_session_time_v3_t *time;
if (cmdrsp->err_code) {
dev_err_ratelimited(ac->dev,
"%s: err=%x, mod_id=%x, param_id=%x\n",
__func__, cmdrsp->err_code,
cmdrsp->param_info.module_id,
cmdrsp->param_info.param_id);
return;
}
dev_dbg_ratelimited(ac->dev,
"%s: mod_id=%x, param_id=%x\n", __func__,
cmdrsp->param_info.module_id,
cmdrsp->param_info.param_id);
switch (cmdrsp->param_info.module_id) {
case ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC:
switch (cmdrsp->param_info.param_id) {
case ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3:
time = &cmdrsp->param_data.session_time;
dev_vdbg(ac->dev, "%s: GET_TIME_V3, time_lsw=%x, time_msw=%x\n",
__func__, time->session_time_lsw,
time->session_time_msw);
ac->time_stamp = (uint64_t)(((uint64_t)
time->session_time_msw << 32) |
time->session_time_lsw);
if (time->flags &
ASM_SESSION_MTMX_STRTR_PARAM_STIME_TSTMP_FLG_BMASK)
dev_warn_ratelimited(ac->dev,
"%s: recv inval tstmp\n",
__func__);
if (atomic_cmpxchg(&ac->time_flag, 1, 0))
wake_up(&ac->time_wait);
break;
default:
dev_err(ac->dev, "%s: unexpected param_id %x\n",
__func__, cmdrsp->param_info.param_id);
break;
}
break;
default:
dev_err(ac->dev, "%s: unexpected mod_id %x\n", __func__,
cmdrsp->param_info.module_id);
break;
}
}
static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
{
int i = 0;
struct audio_client *ac = (struct audio_client *)priv;
unsigned long dsp_flags = 0;
uint32_t *payload;
uint32_t wakeup_flag = 1;
int32_t ret = 0;
union asm_token_struct asm_token;
uint8_t buf_index;
struct msm_adsp_event_data *pp_event_package = NULL;
uint32_t payload_size = 0;
unsigned long flags = 0;
int session_id;
if (ac == NULL) {
pr_err("%s: ac NULL\n", __func__);
return -EINVAL;
}
if (data == NULL) {
pr_err("%s: data NULL\n", __func__);
return -EINVAL;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (session_id <= 0 || session_id > ASM_ACTIVE_STREAMS_ALLOWED) {
pr_err("%s: Session ID is invalid, session = %d\n", __func__,
session_id);
return -EINVAL;
}
spin_lock_irqsave(&(session[session_id].session_lock), flags);
if (!q6asm_is_valid_audio_client(ac)) {
pr_err("%s: audio client pointer is invalid, ac = %pK\n",
__func__, ac);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return -EINVAL;
}
payload = data->payload;
asm_token.token = data->token;
if (q6asm_get_flag_from_token(&asm_token, ASM_CMD_NO_WAIT_OFFSET)) {
pr_debug("%s: No wait command opcode[0x%x] cmd_opcode:%x\n",
__func__, data->opcode, payload ? payload[0] : 0);
wakeup_flag = 0;
}
if (data->opcode == RESET_EVENTS) {
atomic_set(&ac->reset, 1);
if (ac->apr == NULL) {
ac->apr = ac->apr2;
ac->apr2 = NULL;
}
pr_debug("%s: Reset event is received: %d %d apr[%pK]\n",
__func__,
data->reset_event, data->reset_proc, ac->apr);
if (ac->cb)
ac->cb(data->opcode, data->token,
(uint32_t *)data->payload, ac->priv);
apr_reset(ac->apr);
ac->apr = NULL;
atomic_set(&ac->time_flag, 0);
atomic_set(&ac->cmd_state, 0);
atomic_set(&ac->mem_state, 0);
atomic_set(&ac->cmd_state_pp, 0);
wake_up(&ac->time_wait);
wake_up(&ac->cmd_wait);
wake_up(&ac->mem_wait);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
dev_vdbg(ac->dev, "%s: session[%d]opcode[0x%x] token[0x%x]payload_size[%d] src[%d] dest[%d]\n",
__func__,
ac->session, data->opcode,
data->token, data->payload_size, data->src_port,
data->dest_port);
if ((data->opcode != ASM_DATA_EVENT_RENDERED_EOS) &&
(data->opcode != ASM_DATA_EVENT_EOS) &&
(data->opcode != ASM_SESSION_EVENTX_OVERFLOW) &&
(data->opcode != ASM_SESSION_EVENT_RX_UNDERFLOW)) {
if (payload == NULL) {
pr_err("%s: payload is null\n", __func__);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return -EINVAL;
}
if (data->payload_size >=
2 * sizeof(uint32_t))
dev_vdbg(ac->dev, "%s: Payload = [0x%x] status[0x%x] opcode 0x%x\n",
__func__, payload[0], payload[1], data->opcode);
else
dev_vdbg(ac->dev, "%s: Payload size of %d is less than expected.\n",
__func__, data->payload_size);
}
if (data->opcode == APR_BASIC_RSP_RESULT) {
switch (payload[0]) {
case ASM_STREAM_CMD_SET_PP_PARAMS_V2:
if (rtac_make_asm_callback(ac->session, payload,
data->payload_size))
break;
case ASM_SESSION_CMD_PAUSE:
case ASM_SESSION_CMD_SUSPEND:
case ASM_DATA_CMD_EOS:
case ASM_STREAM_CMD_CLOSE:
case ASM_STREAM_CMD_FLUSH:
case ASM_SESSION_CMD_RUN_V2:
case ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS:
case ASM_STREAM_CMD_FLUSH_READBUFS:
pr_debug("%s: session %d opcode 0x%x token 0x%x Payload = [0x%x] src %d dest %d\n",
__func__, ac->session, data->opcode, data->token,
payload[0], data->src_port, data->dest_port);
ret = q6asm_is_valid_session(data, priv);
if (ret != 0) {
pr_err("%s: session invalid %d\n", __func__, ret);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return ret;
}
case ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2:
case ASM_STREAM_CMD_OPEN_READ_V3:
case ASM_STREAM_CMD_OPEN_WRITE_V3:
case ASM_STREAM_CMD_OPEN_PULL_MODE_WRITE:
case ASM_STREAM_CMD_OPEN_PUSH_MODE_READ:
case ASM_STREAM_CMD_OPEN_READWRITE_V2:
case ASM_STREAM_CMD_OPEN_LOOPBACK_V2:
case ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK:
case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
case ASM_DATA_CMD_IEC_60958_MEDIA_FMT:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM_V2:
case ASM_STREAM_CMD_REGISTER_ENCDEC_EVENTS:
case ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE:
case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
case ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS:
case ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED:
if (data->payload_size >=
2 * sizeof(uint32_t) &&
payload[1] != 0) {
pr_debug("%s: session %d opcode 0x%x token 0x%x Payload = [0x%x] stat 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
payload[0], payload[1],
data->src_port, data->dest_port);
pr_err("%s: cmd = 0x%x returned error = 0x%x\n",
__func__, payload[0], payload[1]);
if (wakeup_flag) {
if ((is_adsp_reg_event(payload[0]) >= 0)
|| (payload[0] ==
ASM_STREAM_CMD_SET_PP_PARAMS_V2))
atomic_set(&ac->cmd_state_pp,
payload[1]);
else
atomic_set(&ac->cmd_state,
payload[1]);
wake_up(&ac->cmd_wait);
}
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return 0;
}
if ((is_adsp_reg_event(payload[0]) >= 0) ||
(payload[0] == ASM_STREAM_CMD_SET_PP_PARAMS_V2)) {
if (atomic_read(&ac->cmd_state_pp) &&
wakeup_flag) {
atomic_set(&ac->cmd_state_pp, 0);
wake_up(&ac->cmd_wait);
}
} else {
if (atomic_read(&ac->cmd_state) &&
wakeup_flag) {
atomic_set(&ac->cmd_state, 0);
wake_up(&ac->cmd_wait);
}
}
if (ac->cb)
ac->cb(data->opcode, data->token,
(uint32_t *)data->payload, ac->priv);
break;
case ASM_CMD_ADD_TOPOLOGIES:
if (data->payload_size >=
2 * sizeof(uint32_t) &&
payload[1] != 0) {
pr_debug("%s:Payload = [0x%x]stat[0x%x]\n",
__func__, payload[0], payload[1]);
pr_err("%s: cmd = 0x%x returned error = 0x%x\n",
__func__, payload[0], payload[1]);
if (wakeup_flag) {
atomic_set(&ac->mem_state, payload[1]);
wake_up(&ac->mem_wait);
}
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return 0;
}
if (atomic_read(&ac->mem_state) && wakeup_flag) {
atomic_set(&ac->mem_state, 0);
wake_up(&ac->mem_wait);
}
if (ac->cb)
ac->cb(data->opcode, data->token,
(uint32_t *)data->payload, ac->priv);
break;
case ASM_DATA_EVENT_WATERMARK: {
if (data->payload_size >= 2 * sizeof(uint32_t))
pr_debug("%s: Watermark opcode[0x%x] status[0x%x]",
__func__, payload[0], payload[1]);
else
pr_err("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
break;
}
case ASM_STREAM_CMD_GET_PP_PARAMS_V2:
pr_debug("%s: ASM_STREAM_CMD_GET_PP_PARAMS_V2 session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
/* Should only come here if there is an APR */
/* error or malformed APR packet. Otherwise */
/* response will be returned as */
/* ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 */
if (data->payload_size >= 2 * sizeof(uint32_t)) {
if (payload[1] != 0) {
pr_err("%s: ASM get param error = %d, resuming\n",
__func__, payload[1]);
rtac_make_asm_callback(ac->session,
payload,
data->payload_size);
}
} else {
pr_err("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
}
break;
case ASM_STREAM_CMD_REGISTER_PP_EVENTS:
pr_debug("%s: ASM_STREAM_CMD_REGISTER_PP_EVENTS session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
if (data->payload_size >= 2 * sizeof(uint32_t)) {
if (payload[1] != 0)
pr_err("%s: ASM get param error = %d, resuming\n",
__func__, payload[1]);
atomic_set(&ac->cmd_state_pp, payload[1]);
wake_up(&ac->cmd_wait);
} else {
pr_err("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
}
break;
default:
pr_debug("%s: command[0x%x] not expecting rsp\n",
__func__, payload[0]);
break;
}
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
switch (data->opcode) {
case ASM_DATA_EVENT_WRITE_DONE_V2:{
struct audio_port_data *port = &ac->port[IN];
if (data->payload_size >= 2 * sizeof(uint32_t))
dev_vdbg(ac->dev, "%s: Rxed opcode[0x%x] status[0x%x] token[%d]",
__func__, payload[0], payload[1],
data->token);
else
dev_err(ac->dev, "%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
if (ac->io_mode & SYNC_IO_MODE) {
if (port->buf == NULL) {
pr_err("%s: Unexpected Write Done\n",
__func__);
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return -EINVAL;
}
spin_lock_irqsave(&port->dsp_lock, dsp_flags);
buf_index = asm_token._token.buf_index;
if (buf_index < 0 || buf_index >= port->max_buf_cnt) {
pr_debug("%s: Invalid buffer index %u\n",
__func__, buf_index);
spin_unlock_irqrestore(&port->dsp_lock,
dsp_flags);
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return -EINVAL;
}
if (data->payload_size >= 2 * sizeof(uint32_t) &&
(lower_32_bits(port->buf[buf_index].phys) !=
payload[0] ||
msm_audio_populate_upper_32_bits(
port->buf[buf_index].phys) !=
payload[1])) {
pr_debug("%s: Expected addr %pK\n",
__func__, &port->buf[buf_index].phys);
pr_err("%s: rxedl[0x%x] rxedu [0x%x]\n",
__func__, payload[0], payload[1]);
spin_unlock_irqrestore(&port->dsp_lock,
dsp_flags);
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return -EINVAL;
}
port->buf[buf_index].used = 1;
spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
config_debug_fs_write_cb();
for (i = 0; i < port->max_buf_cnt; i++)
dev_vdbg(ac->dev, "%s %d\n",
__func__, port->buf[i].used);
}
break;
}
case ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2:
pr_debug("%s: ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session, data->opcode,
data->token,
data->src_port, data->dest_port);
if (payload[0] != 0) {
pr_err("%s: ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 returned error = 0x%x\n",
__func__, payload[0]);
} else if (generic_get_data) {
generic_get_data->valid = 1;
if (generic_get_data->is_inband) {
if (data->payload_size >= 4 * sizeof(uint32_t))
pr_debug("%s: payload[1] = 0x%x, payload[2]=0x%x, payload[3]=0x%x\n",
__func__, payload[1],
payload[2], payload[3]);
else
pr_err("%s: payload size of %x is less than expected.\n",
__func__,
data->payload_size);
if (data->payload_size >=
(4 + (payload[3]>>2))
* sizeof(uint32_t)) {
generic_get_data->size_in_ints =
payload[3]>>2;
for (i = 0; i < payload[3]>>2; i++) {
generic_get_data->ints[i] =
payload[4+i];
pr_debug("%s: ASM callback val %i = %i\n",
__func__, i,
payload[4+i]);
}
} else {
pr_err("%s: payload size of %x is less than expected.\n",
__func__,
data->payload_size);
}
pr_debug("%s: callback size in ints = %i\n",
__func__,
generic_get_data->size_in_ints);
}
if (atomic_read(&ac->cmd_state) && wakeup_flag) {
atomic_set(&ac->cmd_state, 0);
wake_up(&ac->cmd_wait);
}
break;
}
rtac_make_asm_callback(ac->session, payload,
data->payload_size);
break;
case ASM_DATA_EVENT_READ_DONE_V2:{
struct audio_port_data *port = &ac->port[OUT];
config_debug_fs_read_cb();
dev_vdbg(ac->dev, "%s: ReadDone: status=%d buff_add=0x%x act_size=%d offset=%d\n",
__func__, payload[READDONE_IDX_STATUS],
payload[READDONE_IDX_BUFADD_LSW],
payload[READDONE_IDX_SIZE],
payload[READDONE_IDX_OFFSET]);
dev_vdbg(ac->dev, "%s: ReadDone:msw_ts=%d lsw_ts=%d memmap_hdl=0x%x flags=%d id=%d num=%d\n",
__func__, payload[READDONE_IDX_MSW_TS],
payload[READDONE_IDX_LSW_TS],
payload[READDONE_IDX_MEMMAP_HDL],
payload[READDONE_IDX_FLAGS],
payload[READDONE_IDX_SEQ_ID],
payload[READDONE_IDX_NUMFRAMES]);
if (ac->io_mode & SYNC_IO_MODE) {
if (port->buf == NULL) {
pr_err("%s: Unexpected Write Done\n", __func__);
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return -EINVAL;
}
spin_lock_irqsave(&port->dsp_lock, dsp_flags);
buf_index = asm_token._token.buf_index;
if (buf_index < 0 || buf_index >= port->max_buf_cnt) {
pr_debug("%s: Invalid buffer index %u\n",
__func__, buf_index);
spin_unlock_irqrestore(&port->dsp_lock,
dsp_flags);
spin_unlock_irqrestore(
&(session[session_id].session_lock),
flags);
return -EINVAL;
}
port->buf[buf_index].used = 0;
if (lower_32_bits(port->buf[buf_index].phys) !=
payload[READDONE_IDX_BUFADD_LSW] ||
msm_audio_populate_upper_32_bits(
port->buf[buf_index].phys) !=
payload[READDONE_IDX_BUFADD_MSW]) {
dev_vdbg(ac->dev, "%s: Expected addr %pK\n",
__func__, &port->buf[buf_index].phys);
pr_err("%s: rxedl[0x%x] rxedu[0x%x]\n",
__func__,
payload[READDONE_IDX_BUFADD_LSW],
payload[READDONE_IDX_BUFADD_MSW]);
spin_unlock_irqrestore(&port->dsp_lock,
dsp_flags);
break;
}
port->buf[buf_index].actual_size =
payload[READDONE_IDX_SIZE];
spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
}
break;
}
case ASM_DATA_EVENT_EOS:
case ASM_DATA_EVENT_RENDERED_EOS:
pr_debug("%s: EOS ACK received: rxed session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
break;
case ASM_SESSION_EVENTX_OVERFLOW:
pr_debug("%s: ASM_SESSION_EVENTX_OVERFLOW session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
break;
case ASM_SESSION_EVENT_RX_UNDERFLOW:
pr_debug("%s: ASM_SESSION_EVENT_RX_UNDERFLOW session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
break;
case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3:
if (data->payload_size >= 3 * sizeof(uint32_t)) {
dev_vdbg(ac->dev, "%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3, payload[0] = %d, payload[1] = %d, payload[2] = %d\n",
__func__,
payload[0], payload[1], payload[2]);
ac->time_stamp =
(uint64_t)(((uint64_t)payload[2] << 32) |
payload[1]);
} else {
dev_err(ac->dev, "%s: payload size of %x is less than expected.n",
__func__, data->payload_size);
}
if (atomic_cmpxchg(&ac->time_flag, 1, 0))
wake_up(&ac->time_wait);
break;
case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY:
pr_debug("%s: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY session %d opcode 0x%x token 0x%x src %d dest %d\n",
__func__, ac->session,
data->opcode, data->token,
data->src_port, data->dest_port);
if (data->payload_size >= 4 * sizeof(uint32_t))
pr_debug("%s: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY, payload[0] = %d, payload[1] = %d, payload[2] = %d, payload[3] = %d\n",
__func__,
payload[0], payload[1], payload[2],
payload[3]);
else
pr_debug("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
break;
case ASM_SESSION_CMDRSP_GET_MTMX_STRTR_PARAMS_V2:
q6asm_process_mtmx_get_param_rsp(ac, (void *) payload);
break;
case ASM_STREAM_PP_EVENT:
case ASM_STREAM_CMD_ENCDEC_EVENTS:
case ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE:
if (data->payload_size >= 2 * sizeof(uint32_t))
pr_debug("%s: ASM_STREAM_EVENT payload[0][0x%x] payload[1][0x%x]",
__func__, payload[0], payload[1]);
else
pr_debug("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
i = is_adsp_raise_event(data->opcode);
if (i < 0) {
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
/* repack payload for asm_stream_pp_event
* package is composed of event type + size + actual payload
*/
payload_size = data->payload_size;
if (payload_size > UINT_MAX
- sizeof(struct msm_adsp_event_data)) {
pr_err("%s: payload size = %d exceeds limit.\n",
__func__, payload_size);
spin_unlock(&(session[session_id].session_lock));
return -EINVAL;
}
pp_event_package = kzalloc(payload_size
+ sizeof(struct msm_adsp_event_data),
GFP_ATOMIC);
if (!pp_event_package) {
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return -ENOMEM;
}
pp_event_package->event_type = i;
pp_event_package->payload_len = payload_size;
memcpy((void *)pp_event_package->payload,
data->payload, payload_size);
ac->cb(data->opcode, data->token,
(void *)pp_event_package, ac->priv);
kfree(pp_event_package);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
case ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2:
if (data->payload_size >= 3 * sizeof(uint32_t))
pr_debug("%s: ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 sesion %d status 0x%x msw %u lsw %u\n",
__func__, ac->session, payload[0], payload[2],
payload[1]);
else
pr_err("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
wake_up(&ac->cmd_wait);
break;
case ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2:
if (data->payload_size >= 3 * sizeof(uint32_t))
pr_debug("%s: ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 session %d status 0x%x msw %u lsw %u\n",
__func__, ac->session,
payload[0], payload[2],
payload[1]);
else
pr_err("%s: payload size of %x is less than expected.\n",
__func__, data->payload_size);
if (data->payload_size >= 2 * sizeof(uint32_t) &&
payload[0] == 0) {
atomic_set(&ac->cmd_state, 0);
/* ignore msw, as a delay that large shouldn't happen */
ac->path_delay = payload[1];
} else {
atomic_set(&ac->cmd_state, payload[0]);
ac->path_delay = UINT_MAX;
}
wake_up(&ac->cmd_wait);
break;
}
if (ac->cb)
ac->cb(data->opcode, data->token,
data->payload, ac->priv);
spin_unlock_irqrestore(
&(session[session_id].session_lock), flags);
return 0;
}
/**
* q6asm_is_cpu_buf_avail -
* retrieve next CPU buf avail
*
* @dir: RX or TX direction
* @ac: Audio client handle
* @size: size pointer to be updated with size of buffer
* @index: index pointer to be updated with
* CPU buffer index available
*
* Returns buffer pointer on success or NULL on failure
*/
void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac, uint32_t *size,
uint32_t *index)
{
void *data;
unsigned char idx;
struct audio_port_data *port;
if (!ac || ((dir != IN) && (dir != OUT))) {
pr_err("%s: ac %pK dir %d\n", __func__, ac, dir);
return NULL;
}
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[dir];
mutex_lock(&port->lock);
idx = port->cpu_buf;
if (port->buf == NULL) {
pr_err("%s: Buffer pointer null\n", __func__);
mutex_unlock(&port->lock);
return NULL;
}
/* dir 0: used = 0 means buf in use
* dir 1: used = 1 means buf in use
*/
if (port->buf[idx].used == dir) {
/* To make it more robust, we could loop and get the
* next avail buf, its risky though
*/
pr_err("%s: Next buf idx[0x%x] not available, dir[%d]\n",
__func__, idx, dir);
mutex_unlock(&port->lock);
return NULL;
}
*size = port->buf[idx].actual_size;
*index = port->cpu_buf;
data = port->buf[idx].data;
dev_vdbg(ac->dev, "%s: session[%d]index[%d] data[%pK]size[%d]\n",
__func__,
ac->session,
port->cpu_buf,
data, *size);
/* By default increase the cpu_buf cnt
* user accesses this function,increase cpu
* buf(to avoid another api)
*/
port->buf[idx].used = dir;
port->cpu_buf = q6asm_get_next_buf(ac, port->cpu_buf,
port->max_buf_cnt);
mutex_unlock(&port->lock);
return data;
}
return NULL;
}
EXPORT_SYMBOL(q6asm_is_cpu_buf_avail);
/**
* q6asm_cpu_buf_release -
* releases cpu buffer for ASM
*
* @dir: RX or TX direction
* @ac: Audio client handle
*
* Returns 0 on success or error on failure
*/
int q6asm_cpu_buf_release(int dir, struct audio_client *ac)
{
struct audio_port_data *port;
int ret = 0;
int idx;
if (!ac || ((dir != IN) && (dir != OUT))) {
pr_err("%s: ac %pK dir %d\n", __func__, ac, dir);
ret = -EINVAL;
goto exit;
}
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[dir];
mutex_lock(&port->lock);
idx = port->cpu_buf;
if (port->cpu_buf == 0) {
port->cpu_buf = port->max_buf_cnt - 1;
} else if (port->cpu_buf < port->max_buf_cnt) {
port->cpu_buf = port->cpu_buf - 1;
} else {
pr_err("%s: buffer index(%d) out of range\n",
__func__, port->cpu_buf);
ret = -EINVAL;
mutex_unlock(&port->lock);
goto exit;
}
port->buf[port->cpu_buf].used = dir ^ 1;
mutex_unlock(&port->lock);
}
exit:
return ret;
}
EXPORT_SYMBOL(q6asm_cpu_buf_release);
/**
* q6asm_is_cpu_buf_avail_nolock -
* retrieve next CPU buf avail without lock acquire
*
* @dir: RX or TX direction
* @ac: Audio client handle
* @size: size pointer to be updated with size of buffer
* @index: index pointer to be updated with
* CPU buffer index available
*
* Returns buffer pointer on success or NULL on failure
*/
void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
uint32_t *size, uint32_t *index)
{
void *data;
unsigned char idx;
struct audio_port_data *port;
if (!ac || ((dir != IN) && (dir != OUT))) {
pr_err("%s: ac %pK dir %d\n", __func__, ac, dir);
return NULL;
}
port = &ac->port[dir];
idx = port->cpu_buf;
if (port->buf == NULL) {
pr_err("%s: Buffer pointer null\n", __func__);
return NULL;
}
/*
* dir 0: used = 0 means buf in use
* dir 1: used = 1 means buf in use
*/
if (port->buf[idx].used == dir) {
/*
* To make it more robust, we could loop and get the
* next avail buf, its risky though
*/
pr_err("%s: Next buf idx[0x%x] not available, dir[%d]\n",
__func__, idx, dir);
return NULL;
}
*size = port->buf[idx].actual_size;
*index = port->cpu_buf;
data = port->buf[idx].data;
dev_vdbg(ac->dev, "%s: session[%d]index[%d] data[%pK]size[%d]\n",
__func__, ac->session, port->cpu_buf,
data, *size);
/*
* By default increase the cpu_buf cnt
* user accesses this function,increase cpu
* buf(to avoid another api)
*/
port->buf[idx].used = dir;
port->cpu_buf = q6asm_get_next_buf(ac, port->cpu_buf,
port->max_buf_cnt);
return data;
}
EXPORT_SYMBOL(q6asm_is_cpu_buf_avail_nolock);
int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac)
{
int ret = -1;
struct audio_port_data *port;
uint32_t idx;
if (!ac || (dir != OUT)) {
pr_err("%s: ac %pK dir %d\n", __func__, ac, dir);
return ret;
}
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[dir];
mutex_lock(&port->lock);
idx = port->dsp_buf;
if (port->buf[idx].used == (dir ^ 1)) {
/* To make it more robust, we could loop and get the
* next avail buf, its risky though
*/
pr_err("%s: Next buf idx[0x%x] not available, dir[%d]\n",
__func__, idx, dir);
mutex_unlock(&port->lock);
return ret;
}
dev_vdbg(ac->dev, "%s: session[%d]dsp_buf=%d cpu_buf=%d\n",
__func__,
ac->session, port->dsp_buf, port->cpu_buf);
ret = ((port->dsp_buf != port->cpu_buf) ? 0 : -1);
mutex_unlock(&port->lock);
}
return ret;
}
static void __q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg, uint32_t stream_id)
{
unsigned long flags = 0;
dev_vdbg(ac->dev, "%s: pkt_size=%d cmd_flg=%d session=%d stream_id=%d\n",
__func__, pkt_size, cmd_flg, ac->session, stream_id);
mutex_lock(&ac->cmd_lock);
spin_lock_irqsave(&(session[ac->session].session_lock), flags);
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL", __func__);
spin_unlock_irqrestore(
&(session[ac->session].session_lock), flags);
mutex_unlock(&ac->cmd_lock);
return;
}
hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
APR_HDR_LEN(sizeof(struct apr_hdr)),
APR_PKT_VER);
hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
hdr->src_domain = APR_DOMAIN_APPS;
hdr->dest_svc = APR_SVC_ASM;
hdr->dest_domain = APR_DOMAIN_ADSP;
hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
if (cmd_flg)
q6asm_update_token(&hdr->token,
ac->session,
0, /* Stream ID is NA */
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
hdr->pkt_size = pkt_size;
spin_unlock_irqrestore(
&(session[ac->session].session_lock), flags);
mutex_unlock(&ac->cmd_lock);
}
static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg)
{
__q6asm_add_hdr(ac, hdr, pkt_size, cmd_flg, ac->stream_id);
}
static void q6asm_stream_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg, int32_t stream_id)
{
__q6asm_add_hdr(ac, hdr, pkt_size, cmd_flg, stream_id);
}
static void __q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg,
uint32_t stream_id, u8 no_wait_flag)
{
dev_vdbg(ac->dev, "%s: pkt_size = %d, cmd_flg = %d, session = %d stream_id=%d\n",
__func__, pkt_size, cmd_flg, ac->session, stream_id);
hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
APR_HDR_LEN(sizeof(struct apr_hdr)),
APR_PKT_VER);
if (ac->apr == NULL) {
pr_err("%s: AC APR is NULL", __func__);
return;
}
hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
hdr->src_domain = APR_DOMAIN_APPS;
hdr->dest_svc = APR_SVC_ASM;
hdr->dest_domain = APR_DOMAIN_ADSP;
hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
if (cmd_flg) {
q6asm_update_token(&hdr->token,
ac->session,
0, /* Stream ID is NA */
0, /* Buffer index is NA */
0, /* Direction flag is NA */
no_wait_flag);
}
hdr->pkt_size = pkt_size;
}
static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
uint32_t pkt_size, uint32_t cmd_flg)
{
__q6asm_add_hdr_async(ac, hdr, pkt_size, cmd_flg,
ac->stream_id, WAIT_CMD);
}
static void q6asm_stream_add_hdr_async(struct audio_client *ac,
struct apr_hdr *hdr, uint32_t pkt_size,
uint32_t cmd_flg, int32_t stream_id)
{
__q6asm_add_hdr_async(ac, hdr, pkt_size, cmd_flg,
stream_id, NO_WAIT_CMD);
}
static void q6asm_add_hdr_custom_topology(struct audio_client *ac,
struct apr_hdr *hdr,
uint32_t pkt_size)
{
pr_debug("%s: pkt_size=%d session=%d\n",
__func__, pkt_size, ac->session);
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return;
}
mutex_lock(&ac->cmd_lock);
hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
APR_HDR_LEN(sizeof(struct apr_hdr)),
APR_PKT_VER);
hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
hdr->src_domain = APR_DOMAIN_APPS;
hdr->dest_svc = APR_SVC_ASM;
hdr->dest_domain = APR_DOMAIN_ADSP;
hdr->src_port = ((ac->session << 8) & 0xFF00) | 0x01;
hdr->dest_port = 0;
q6asm_update_token(&hdr->token,
ac->session,
0, /* Stream ID is NA */
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
hdr->pkt_size = pkt_size;
mutex_unlock(&ac->cmd_lock);
}
static void q6asm_add_mmaphdr(struct audio_client *ac, struct apr_hdr *hdr,
u32 pkt_size, int dir)
{
pr_debug("%s: pkt size=%d\n",
__func__, pkt_size);
hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
hdr->src_port = 0;
hdr->dest_port = 0;
q6asm_update_token(&hdr->token,
ac->session,
0, /* Stream ID is NA */
0, /* Buffer index is NA */
dir,
WAIT_CMD);
hdr->pkt_size = pkt_size;
}
static int __q6asm_open_read(struct audio_client *ac,
uint32_t format, uint16_t bits_per_sample,
uint32_t pcm_format_block_ver,
bool ts_mode)
{
int rc = 0x00;
struct asm_stream_cmd_open_read_v3 open;
struct q6asm_cal_info cal_info;
config_debug_fs_reset_index();
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
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;
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.preprocopo_id = cal_info.topology_id;
open.bits_per_sample = bits_per_sample;
open.mode_flags = 0x0;
ac->topology = open.preprocopo_id;
ac->app_type = cal_info.app_type;
if (ac->perf_mode == LOW_LATENCY_PCM_MODE) {
open.mode_flags |= ASM_LOW_LATENCY_TX_STREAM_SESSION <<
ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
} else {
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 = q6asm_get_pcm_format_id(pcm_format_block_ver);
if (ts_mode)
open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE;
break;
case FORMAT_MPEG4_AAC:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_G711_ALAW_FS:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_G711_ALAW_FS;
break;
case FORMAT_G711_MLAW_FS:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_G711_MLAW_FS;
break;
case FORMAT_V13K:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_V13K_FS;
break;
case FORMAT_EVRC:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_EVRC_FS;
break;
case FORMAT_AMRNB:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_AMRNB_FS;
break;
case FORMAT_AMRWB:
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
break;
default:
pr_err("%s: Invalid format 0x%x\n",
__func__, format);
rc = -EINVAL;
goto fail_cmd;
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open read\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
ac->io_mode |= TUN_READ_IO_MODE;
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_open_read -
* command to open ASM in read mode
*
* @ac: Audio client handle
* @format: capture format for ASM
*
* Returns 0 on success or error on failure
*/
int q6asm_open_read(struct audio_client *ac,
uint32_t format)
{
return __q6asm_open_read(ac, format, 16,
PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
false/*ts_mode*/);
}
EXPORT_SYMBOL(q6asm_open_read);
int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
false/*ts_mode*/);
}
/*
* asm_open_read_v3 - Opens audio capture session
*
* @ac: Client session handle
* @format: encoder format
* @bits_per_sample: bit width of capture session
*/
int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V3/*media fmt block ver*/,
false/*ts_mode*/);
}
EXPORT_SYMBOL(q6asm_open_read_v3);
/*
* asm_open_read_v4 - Opens audio capture session
*
* @ac: Client session handle
* @format: encoder format
* @bits_per_sample: bit width of capture session
* @ts_mode: timestamp mode
*/
int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, bool ts_mode)
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/,
ts_mode);
}
EXPORT_SYMBOL(q6asm_open_read_v4);
/*
* asm_open_read_v5 - Opens audio capture session
*
* @ac: Client session handle
* @format: encoder format
* @bits_per_sample: bit width of capture session
* @ts_mode: timestamp mode
*/
int q6asm_open_read_v5(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, bool ts_mode,
uint32_t enc_cfg_id)
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V5 /*media fmt block ver*/,
ts_mode);
}
EXPORT_SYMBOL(q6asm_open_read_v5);
/**
* q6asm_open_write_compressed -
* command to open ASM in compressed write mode
*
* @ac: Audio client handle
* @format: playback format for ASM
* @passthrough_flag: flag to indicate passthrough option
*
* Returns 0 on success or error on failure
*/
int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format,
uint32_t passthrough_flag)
{
int rc = 0;
struct asm_stream_cmd_open_write_compressed open;
if (ac == NULL) {
pr_err("%s: ac[%pK] NULL\n", __func__, ac);
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: APR handle[%pK] NULL\n", __func__, ac->apr);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s: session[%d] wr_format[0x%x]", __func__, ac->session,
format);
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED;
atomic_set(&ac->cmd_state, -1);
switch (format) {
case FORMAT_AC3:
open.fmt_id = ASM_MEDIA_FMT_AC3;
break;
case FORMAT_EAC3:
open.fmt_id = ASM_MEDIA_FMT_EAC3;
break;
case FORMAT_DTS:
open.fmt_id = ASM_MEDIA_FMT_DTS;
break;
case FORMAT_DSD:
open.fmt_id = ASM_MEDIA_FMT_DSD;
break;
case FORMAT_GEN_COMPR:
open.fmt_id = ASM_MEDIA_FMT_GENERIC_COMPRESSED;
break;
case FORMAT_TRUEHD:
open.fmt_id = ASM_MEDIA_FMT_TRUEHD;
break;
case FORMAT_IEC61937:
open.fmt_id = ASM_MEDIA_FMT_IEC;
break;
default:
pr_err("%s: Invalid format[%d]\n", __func__, format);
rc = -EINVAL;
goto fail_cmd;
}
/* Below flag indicates the DSP that Compressed audio input
* stream is not IEC 61937 or IEC 60958 packetizied
*/
if (passthrough_flag == COMPRESSED_PASSTHROUGH ||
passthrough_flag == COMPRESSED_PASSTHROUGH_DSD ||
passthrough_flag == COMPRESSED_PASSTHROUGH_GEN) {
open.flags = 0x0;
pr_debug("%s: Flag 0 COMPRESSED_PASSTHROUGH\n", __func__);
} else if (passthrough_flag == COMPRESSED_PASSTHROUGH_CONVERT) {
open.flags = 0x8;
pr_debug("%s: Flag 8 - COMPRESSED_PASSTHROUGH_CONVERT\n",
__func__);
} else if (passthrough_flag == COMPRESSED_PASSTHROUGH_IEC61937) {
open.flags = 0x1;
pr_debug("%s: Flag 1 - COMPRESSED_PASSTHROUGH_IEC61937\n",
__func__);
} else {
pr_err("%s: Invalid passthrough type[%d]\n",
__func__, passthrough_flag);
rc = -EINVAL;
goto fail_cmd;
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 1*HZ);
if (!rc) {
pr_err("%s: timeout. waited for OPEN_WRITE_COMPR rc[%d]\n",
__func__, rc);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_open_write_compressed);
static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, uint32_t stream_id,
bool is_gapless_mode,
uint32_t pcm_format_block_ver)
{
int rc = 0x00;
struct asm_stream_cmd_open_write_v3 open;
struct q6asm_cal_info cal_info;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
dev_vdbg(ac->dev, "%s: session[%d] wr_format[0x%x]\n",
__func__, ac->session, format);
q6asm_stream_add_hdr(ac, &open.hdr, sizeof(open), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&open.hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
dev_vdbg(ac->dev, "%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, open.hdr.token, stream_id, ac->session);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open.mode_flags = 0x00;
if (ac->perf_mode == ULL_POST_PROCESSING_PCM_MODE)
open.mode_flags |= ASM_ULL_POST_PROCESSING_STREAM_SESSION;
else if (ac->perf_mode == ULTRA_LOW_LATENCY_PCM_MODE)
open.mode_flags |= ASM_ULTRA_LOW_LATENCY_STREAM_SESSION;
else if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION;
else {
open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
if (is_gapless_mode)
open.mode_flags |= 1 << ASM_SHIFT_GAPLESS_MODE_FLAG;
}
/* source endpoint : matrix */
open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
open.bits_per_sample = bits_per_sample;
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.postprocopo_id = cal_info.topology_id;
if (ac->perf_mode != LEGACY_PCM_MODE)
open.postprocopo_id = ASM_STREAM_POSTPROCOPO_ID_NONE;
pr_debug("%s: perf_mode %d asm_topology 0x%x bps %d\n", __func__,
ac->perf_mode, open.postprocopo_id, open.bits_per_sample);
/*
* For Gapless playback it will use the same session for next stream,
* So use the same topology
*/
if (!ac->topology) {
ac->topology = open.postprocopo_id;
ac->app_type = cal_info.app_type;
}
switch (format) {
case FORMAT_LINEAR_PCM:
open.dec_fmt_id = q6asm_get_pcm_format_id(pcm_format_block_ver);
break;
case FORMAT_MPEG4_AAC:
open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_MPEG4_MULTI_AAC:
open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_WMA_V9:
open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V9_V2;
break;
case FORMAT_WMA_V10PRO:
open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V10PRO_V2;
break;
case FORMAT_MP3:
open.dec_fmt_id = ASM_MEDIA_FMT_MP3;
break;
case FORMAT_AC3:
open.dec_fmt_id = ASM_MEDIA_FMT_AC3;
break;
case FORMAT_EAC3:
open.dec_fmt_id = ASM_MEDIA_FMT_EAC3;
break;
case FORMAT_MP2:
open.dec_fmt_id = ASM_MEDIA_FMT_MP2;
break;
case FORMAT_FLAC:
open.dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
case FORMAT_ALAC:
open.dec_fmt_id = ASM_MEDIA_FMT_ALAC;
break;
case FORMAT_VORBIS:
open.dec_fmt_id = ASM_MEDIA_FMT_VORBIS;
break;
case FORMAT_APE:
open.dec_fmt_id = ASM_MEDIA_FMT_APE;
break;
case FORMAT_DSD:
open.dec_fmt_id = ASM_MEDIA_FMT_DSD;
break;
case FORMAT_APTX:
open.dec_fmt_id = ASM_MEDIA_FMT_APTX;
break;
case FORMAT_GEN_COMPR:
open.dec_fmt_id = ASM_MEDIA_FMT_GENERIC_COMPRESSED;
break;
default:
pr_err("%s: Invalid format 0x%x\n", __func__, format);
rc = -EINVAL;
goto fail_cmd;
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open write\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
ac->io_mode |= TUN_WRITE_IO_MODE;
return 0;
fail_cmd:
return rc;
}
int q6asm_open_write(struct audio_client *ac, uint32_t format)
{
return __q6asm_open_write(ac, format, 16, ac->stream_id,
false /*gapless*/,
PCM_MEDIA_FORMAT_V2 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_open_write);
int q6asm_open_write_v2(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_write(ac, format, bits_per_sample,
ac->stream_id, false /*gapless*/,
PCM_MEDIA_FORMAT_V2 /*pcm_format_block_ver*/);
}
/*
* q6asm_open_write_v3 - Opens audio playback session
*
* @ac: Client session handle
* @format: decoder format
* @bits_per_sample: bit width of playback session
*/
int q6asm_open_write_v3(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_write(ac, format, bits_per_sample,
ac->stream_id, false /*gapless*/,
PCM_MEDIA_FORMAT_V3 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_open_write_v3);
/*
* q6asm_open_write_v4 - Opens audio playback session
*
* @ac: Client session handle
* @format: decoder format
* @bits_per_sample: bit width of playback session
*/
int q6asm_open_write_v4(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_write(ac, format, bits_per_sample,
ac->stream_id, false /*gapless*/,
PCM_MEDIA_FORMAT_V4 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_open_write_v4);
int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode)
{
return __q6asm_open_write(ac, format, bits_per_sample,
stream_id, is_gapless_mode,
PCM_MEDIA_FORMAT_V2 /*pcm_format_block_ver*/);
}
/*
* q6asm_stream_open_write_v3 - Creates audio stream for playback
*
* @ac: Client session handle
* @format: asm playback format
* @bits_per_sample: bit width of requested stream
* @stream_id: stream id of stream to be associated with this session
* @is_gapless_mode: true if gapless mode needs to be enabled
*/
int q6asm_stream_open_write_v3(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode)
{
return __q6asm_open_write(ac, format, bits_per_sample,
stream_id, is_gapless_mode,
PCM_MEDIA_FORMAT_V3 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_stream_open_write_v3);
/*
* q6asm_open_write_v5 - Opens audio playback session
*
* @ac: Client session handle
* @format: decoder format
* @bits_per_sample: bit width of playback session
*/
int q6asm_open_write_v5(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample)
{
return __q6asm_open_write(ac, format, bits_per_sample,
ac->stream_id, false /*gapless*/,
PCM_MEDIA_FORMAT_V5 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_open_write_v5);
/*
* q6asm_stream_open_write_v4 - Creates audio stream for playback
*
* @ac: Client session handle
* @format: asm playback format
* @bits_per_sample: bit width of requested stream
* @stream_id: stream id of stream to be associated with this session
* @is_gapless_mode: true if gapless mode needs to be enabled
*/
int q6asm_stream_open_write_v4(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode)
{
return __q6asm_open_write(ac, format, bits_per_sample,
stream_id, is_gapless_mode,
PCM_MEDIA_FORMAT_V4 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_stream_open_write_v4);
/*
* q6asm_stream_open_write_v5 - Creates audio stream for playback
*
* @ac: Client session handle
* @format: asm playback format
* @bits_per_sample: bit width of requested stream
* @stream_id: stream id of stream to be associated with this session
* @is_gapless_mode: true if gapless mode needs to be enabled
*/
int q6asm_stream_open_write_v5(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode)
{
return __q6asm_open_write(ac, format, bits_per_sample,
stream_id, is_gapless_mode,
PCM_MEDIA_FORMAT_V5 /*pcm_format_block_ver*/);
}
EXPORT_SYMBOL(q6asm_stream_open_write_v5);
static int __q6asm_open_read_write(struct audio_client *ac, uint32_t rd_format,
uint32_t wr_format, bool is_meta_data_mode,
uint32_t bits_per_sample,
bool overwrite_topology, int topology)
{
int rc = 0x00;
struct asm_stream_cmd_open_readwrite_v2 open;
struct q6asm_cal_info cal_info;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
pr_debug("%s: wr_format[0x%x]rd_format[0x%x]\n",
__func__, wr_format, rd_format);
ac->io_mode |= NT_MODE;
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_READWRITE_V2;
open.mode_flags = is_meta_data_mode ? BUFFER_META_ENABLE : 0;
open.bits_per_sample = bits_per_sample;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.postprocopo_id = cal_info.topology_id;
open.postprocopo_id = overwrite_topology ?
topology : open.postprocopo_id;
ac->topology = open.postprocopo_id;
ac->app_type = cal_info.app_type;
switch (wr_format) {
case FORMAT_LINEAR_PCM:
case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
case FORMAT_MPEG4_AAC:
open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_MPEG4_MULTI_AAC:
open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_WMA_V9:
open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V9_V2;
break;
case FORMAT_WMA_V10PRO:
open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V10PRO_V2;
break;
case FORMAT_AMRNB:
open.dec_fmt_id = ASM_MEDIA_FMT_AMRNB_FS;
break;
case FORMAT_AMRWB:
open.dec_fmt_id = ASM_MEDIA_FMT_AMRWB_FS;
break;
case FORMAT_AMR_WB_PLUS:
open.dec_fmt_id = ASM_MEDIA_FMT_AMR_WB_PLUS_V2;
break;
case FORMAT_V13K:
open.dec_fmt_id = ASM_MEDIA_FMT_V13K_FS;
break;
case FORMAT_EVRC:
open.dec_fmt_id = ASM_MEDIA_FMT_EVRC_FS;
break;
case FORMAT_EVRCB:
open.dec_fmt_id = ASM_MEDIA_FMT_EVRCB_FS;
break;
case FORMAT_EVRCWB:
open.dec_fmt_id = ASM_MEDIA_FMT_EVRCWB_FS;
break;
case FORMAT_MP3:
open.dec_fmt_id = ASM_MEDIA_FMT_MP3;
break;
case FORMAT_ALAC:
open.dec_fmt_id = ASM_MEDIA_FMT_ALAC;
break;
case FORMAT_APE:
open.dec_fmt_id = ASM_MEDIA_FMT_APE;
break;
case FORMAT_DSD:
open.dec_fmt_id = ASM_MEDIA_FMT_DSD;
break;
case FORMAT_G711_ALAW_FS:
open.dec_fmt_id = ASM_MEDIA_FMT_G711_ALAW_FS;
break;
case FORMAT_G711_MLAW_FS:
open.dec_fmt_id = ASM_MEDIA_FMT_G711_MLAW_FS;
break;
default:
pr_err("%s: Invalid format 0x%x\n",
__func__, wr_format);
rc = -EINVAL;
goto fail_cmd;
}
switch (rd_format) {
case FORMAT_LINEAR_PCM:
case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
case FORMAT_MPEG4_AAC:
open.enc_cfg_id = ASM_MEDIA_FMT_AAC_V2;
break;
case FORMAT_G711_ALAW_FS:
open.enc_cfg_id = ASM_MEDIA_FMT_G711_ALAW_FS;
break;
case FORMAT_G711_MLAW_FS:
open.enc_cfg_id = ASM_MEDIA_FMT_G711_MLAW_FS;
break;
case FORMAT_V13K:
open.enc_cfg_id = ASM_MEDIA_FMT_V13K_FS;
break;
case FORMAT_EVRC:
open.enc_cfg_id = ASM_MEDIA_FMT_EVRC_FS;
break;
case FORMAT_AMRNB:
open.enc_cfg_id = ASM_MEDIA_FMT_AMRNB_FS;
break;
case FORMAT_AMRWB:
open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
break;
case FORMAT_ALAC:
open.enc_cfg_id = ASM_MEDIA_FMT_ALAC;
break;
case FORMAT_APE:
open.enc_cfg_id = ASM_MEDIA_FMT_APE;
break;
default:
pr_err("%s: Invalid format 0x%x\n",
__func__, rd_format);
rc = -EINVAL;
goto fail_cmd;
}
dev_vdbg(ac->dev, "%s: rdformat[0x%x]wrformat[0x%x]\n", __func__,
open.enc_cfg_id, open.dec_fmt_id);
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open read-write\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_open_read_write -
* command to open ASM in read/write mode
*
* @ac: Audio client handle
* @rd_format: capture format for ASM
* @wr_format: playback format for ASM
*
* Returns 0 on success or error on failure
*/
int q6asm_open_read_write(struct audio_client *ac, uint32_t rd_format,
uint32_t wr_format)
{
return __q6asm_open_read_write(ac, rd_format, wr_format,
true/*meta data mode*/,
16 /*bits_per_sample*/,
false /*overwrite_topology*/, 0);
}
EXPORT_SYMBOL(q6asm_open_read_write);
/**
* q6asm_open_read_write_v2 -
* command to open ASM in bi-directional read/write mode
*
* @ac: Audio client handle
* @rd_format: capture format for ASM
* @wr_format: playback format for ASM
* @is_meta_data_mode: mode to indicate if meta data present
* @bits_per_sample: number of bits per sample
* @overwrite_topology: topology to be overwritten flag
* @topology: Topology for ASM
*
* Returns 0 on success or error on failure
*/
int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format,
uint32_t wr_format, bool is_meta_data_mode,
uint32_t bits_per_sample, bool overwrite_topology,
int topology)
{
return __q6asm_open_read_write(ac, rd_format, wr_format,
is_meta_data_mode, bits_per_sample,
overwrite_topology, topology);
}
EXPORT_SYMBOL(q6asm_open_read_write_v2);
/**
* q6asm_open_loopback_v2 -
* command to open ASM in loopback mode
*
* @ac: Audio client handle
* @bits_per_sample: number of bits per sample
*
* Returns 0 on success or error on failure
*/
int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
{
int rc = 0x00;
struct q6asm_cal_info cal_info;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
if (ac->perf_mode == LOW_LATENCY_PCM_MODE) {
struct asm_stream_cmd_open_transcode_loopback_t open;
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK;
open.mode_flags = 0;
open.src_endpoint_type = 0;
open.sink_endpoint_type = 0;
open.src_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
open.sink_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.audproc_topo_id = cal_info.topology_id;
ac->app_type = cal_info.app_type;
if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION;
else
open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
ac->topology = open.audproc_topo_id;
open.bits_per_sample = bits_per_sample;
open.reserved = 0;
pr_debug("%s: opening a transcode_loopback with mode_flags =[%d] session[%d]\n",
__func__, open.mode_flags, ac->session);
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
} else {/*if(ac->perf_mode == LEGACY_PCM_MODE)*/
struct asm_stream_cmd_open_loopback_v2 open;
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK_V2;
open.mode_flags = 0;
open.src_endpointype = 0;
open.sink_endpointype = 0;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.postprocopo_id = cal_info.topology_id;
ac->app_type = cal_info.app_type;
ac->topology = open.postprocopo_id;
open.bits_per_sample = bits_per_sample;
open.reserved = 0;
pr_debug("%s: opening a loopback_v2 with mode_flags =[%d] session[%d]\n",
__func__, open.mode_flags, ac->session);
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open_loopback\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_open_loopback_v2);
/**
* q6asm_open_transcode_loopback -
* command to open ASM in transcode loopback mode
*
* @ac: Audio client handle
* @bits_per_sample: number of bits per sample
* @source_format: Format of clip
* @sink_format: end device supported format
*
* Returns 0 on success or error on failure
*/
int q6asm_open_transcode_loopback(struct audio_client *ac,
uint16_t bits_per_sample,
uint32_t source_format, uint32_t sink_format)
{
int rc = 0x00;
struct asm_stream_cmd_open_transcode_loopback_t open;
struct q6asm_cal_info cal_info;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK;
open.mode_flags = 0;
open.src_endpoint_type = 0;
open.sink_endpoint_type = 0;
switch (source_format) {
case FORMAT_LINEAR_PCM:
case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
open.src_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3;
break;
case FORMAT_AC3:
open.src_format_id = ASM_MEDIA_FMT_AC3;
break;
case FORMAT_EAC3:
open.src_format_id = ASM_MEDIA_FMT_EAC3;
break;
default:
pr_err("%s: Unsupported src fmt [%d]\n",
__func__, source_format);
return -EINVAL;
}
switch (sink_format) {
case FORMAT_LINEAR_PCM:
case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
open.sink_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3;
break;
default:
pr_err("%s: Unsupported sink fmt [%d]\n",
__func__, sink_format);
return -EINVAL;
}
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
open.audproc_topo_id = cal_info.topology_id;
ac->app_type = cal_info.app_type;
if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION;
else
open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
ac->topology = open.audproc_topo_id;
open.bits_per_sample = bits_per_sample;
open.reserved = 0;
pr_debug("%s: opening a transcode_loopback with mode_flags =[%d] session[%d]\n",
__func__, open.mode_flags, ac->session);
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open_transcode_loopback\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_open_transcode_loopback);
static
int q6asm_set_shared_circ_buff(struct audio_client *ac,
struct asm_stream_cmd_open_shared_io *open,
int bufsz, int bufcnt,
int dir)
{
struct audio_buffer *buf_circ;
int bytes_to_alloc, rc;
size_t len;
mutex_lock(&ac->cmd_lock);
if (ac->port[dir].buf) {
pr_err("%s: Buffer already allocated\n", __func__);
rc = -EINVAL;
goto done;
}
buf_circ = kzalloc(sizeof(struct audio_buffer), GFP_KERNEL);
if (!buf_circ) {
rc = -ENOMEM;
goto done;
}
bytes_to_alloc = bufsz * bufcnt;
bytes_to_alloc = PAGE_ALIGN(bytes_to_alloc);
rc = msm_audio_ion_alloc("audio_client", &buf_circ->client,
&buf_circ->handle, bytes_to_alloc,
(ion_phys_addr_t *)&buf_circ->phys,
&len, &buf_circ->data);
if (rc) {
pr_err("%s: Audio ION alloc is failed, rc = %d\n", __func__,
rc);
kfree(buf_circ);
goto done;
}
ac->port[dir].buf = buf_circ;
buf_circ->used = dir ^ 1;
buf_circ->size = bytes_to_alloc;
buf_circ->actual_size = bytes_to_alloc;
memset(buf_circ->data, 0, buf_circ->actual_size);
ac->port[dir].max_buf_cnt = 1;
open->shared_circ_buf_mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
open->shared_circ_buf_num_regions = 1;
open->shared_circ_buf_property_flag = 0x00;
open->shared_circ_buf_start_phy_addr_lsw =
lower_32_bits(buf_circ->phys);
open->shared_circ_buf_start_phy_addr_msw =
msm_audio_populate_upper_32_bits(buf_circ->phys);
open->shared_circ_buf_size = bufsz * bufcnt;
open->map_region_circ_buf.shm_addr_lsw = lower_32_bits(buf_circ->phys);
open->map_region_circ_buf.shm_addr_msw =
msm_audio_populate_upper_32_bits(buf_circ->phys);
open->map_region_circ_buf.mem_size_bytes = bytes_to_alloc;
done:
mutex_unlock(&ac->cmd_lock);
return rc;
}
static
int q6asm_set_shared_pos_buff(struct audio_client *ac,
struct asm_stream_cmd_open_shared_io *open,
int dir)
{
struct audio_buffer *buf_pos = &ac->shared_pos_buf;
int rc;
size_t len;
int bytes_to_alloc = sizeof(struct asm_shared_position_buffer);
mutex_lock(&ac->cmd_lock);
bytes_to_alloc = PAGE_ALIGN(bytes_to_alloc);
rc = msm_audio_ion_alloc("audio_client", &buf_pos->client,
&buf_pos->handle, bytes_to_alloc,
(ion_phys_addr_t *)&buf_pos->phys, &len,
&buf_pos->data);
if (rc) {
pr_err("%s: Audio pos buf ION alloc is failed, rc = %d\n",
__func__, rc);
goto done;
}
buf_pos->used = dir ^ 1;
buf_pos->size = bytes_to_alloc;
buf_pos->actual_size = bytes_to_alloc;
open->shared_pos_buf_mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
open->shared_pos_buf_num_regions = 1;
open->shared_pos_buf_property_flag = 0x00;
open->shared_pos_buf_phy_addr_lsw = lower_32_bits(buf_pos->phys);
open->shared_pos_buf_phy_addr_msw =
msm_audio_populate_upper_32_bits(buf_pos->phys);
open->map_region_pos_buf.shm_addr_lsw = lower_32_bits(buf_pos->phys);
open->map_region_pos_buf.shm_addr_msw =
msm_audio_populate_upper_32_bits(buf_pos->phys);
open->map_region_pos_buf.mem_size_bytes = bytes_to_alloc;
done:
mutex_unlock(&ac->cmd_lock);
return rc;
}
/*
* q6asm_open_shared_io: Open an ASM session for pull mode (playback)
* or push mode (capture).
* parameters
* config - session parameters (channels, bits_per_sample, sr)
* dir - stream direction (IN for playback, OUT for capture)
* returns 0 if successful, error code otherwise
*/
int q6asm_open_shared_io(struct audio_client *ac,
struct shared_io_config *config,
int dir)
{
struct asm_stream_cmd_open_shared_io *open;
u8 *channel_mapping;
int i, size_of_open, num_watermarks, bufsz, bufcnt, rc, flags = 0;
struct q6asm_cal_info cal_info;
if (!ac || !config)
return -EINVAL;
if (config->channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__,
config->channels);
return -EINVAL;
}
bufsz = config->bufsz;
bufcnt = config->bufcnt;
num_watermarks = 0;
ac->config = *config;
if (ac->session <= 0 || ac->session > SESSION_MAX) {
pr_err("%s: Session %d is out of bounds\n",
__func__, ac->session);
return -EINVAL;
}
size_of_open = sizeof(struct asm_stream_cmd_open_shared_io) +
(sizeof(struct asm_shared_watermark_level) * num_watermarks);
open = kzalloc(PAGE_ALIGN(size_of_open), GFP_KERNEL);
if (!open)
return -ENOMEM;
q6asm_stream_add_hdr(ac, &open->hdr, size_of_open, TRUE,
ac->stream_id);
atomic_set(&ac->cmd_state, 1);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x, perf %d\n",
__func__, open->hdr.token, ac->stream_id, ac->session,
ac->perf_mode);
open->hdr.opcode =
dir == IN ? ASM_STREAM_CMD_OPEN_PULL_MODE_WRITE :
ASM_STREAM_CMD_OPEN_PUSH_MODE_READ;
pr_debug("%s perf_mode %d\n", __func__, ac->perf_mode);
if (dir == IN)
if (ac->perf_mode == ULL_POST_PROCESSING_PCM_MODE)
flags = 4 << ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE;
else if (ac->perf_mode == ULTRA_LOW_LATENCY_PCM_MODE)
flags = 2 << ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE;
else if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
flags = 1 << ASM_SHIFT_STREAM_PERF_FLAG_PULL_MODE_WRITE;
else
pr_err("Invalid perf mode for pull write\n");
else
if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
flags = ASM_LOW_LATENCY_TX_STREAM_SESSION <<
ASM_SHIFT_STREAM_PERF_FLAG_PUSH_MODE_READ;
else
pr_err("Invalid perf mode for push read\n");
if (flags == 0) {
pr_err("%s: Invalid mode[%d]\n", __func__,
ac->perf_mode);
kfree(open);
return -EINVAL;
}
pr_debug("open.mode_flags = 0x%x\n", flags);
open->mode_flags = flags;
open->endpoint_type = ASM_END_POINT_DEVICE_MATRIX;
open->topo_bits_per_sample = config->bits_per_sample;
rc = q6asm_get_asm_topology_apptype(&cal_info);
open->topo_id = cal_info.topology_id;
if (config->format == FORMAT_LINEAR_PCM)
open->fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3;
else {
pr_err("%s: Invalid format[%d]\n", __func__, config->format);
rc = -EINVAL;
goto done;
}
rc = q6asm_set_shared_circ_buff(ac, open, bufsz, bufcnt, dir);
if (rc)
goto done;
ac->port[dir].tmp_hdl = 0;
rc = q6asm_set_shared_pos_buff(ac, open, dir);
if (rc)
goto done;
/* asm_multi_channel_pcm_fmt_blk_v3 */
open->fmt.num_channels = config->channels;
open->fmt.bits_per_sample = config->bits_per_sample;
open->fmt.sample_rate = config->rate;
open->fmt.is_signed = 1;
open->fmt.sample_word_size = config->sample_word_size;
channel_mapping = open->fmt.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
rc = q6asm_map_channels(channel_mapping, config->channels, false);
if (rc) {
pr_err("%s: Map channels failed, ret: %d\n", __func__, rc);
goto done;
}
open->num_watermark_levels = num_watermarks;
for (i = 0; i < num_watermarks; i++) {
open->watermark[i].watermark_level_bytes = i *
((bufsz * bufcnt) / num_watermarks);
pr_debug("%s: Watermark level set for %i\n",
__func__,
open->watermark[i].watermark_level_bytes);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) open);
if (rc < 0) {
pr_err("%s: Open failed op[0x%x]rc[%d]\n",
__func__, open->hdr.opcode, rc);
goto done;
}
pr_debug("%s: sent open apr pkt\n", __func__);
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) <= 0), 5*HZ);
if (!rc) {
pr_err("%s: Timeout. Waited for open write apr pkt rc[%d]\n",
__func__, rc);
rc = -ETIMEDOUT;
goto done;
}
if (atomic_read(&ac->cmd_state) < 0) {
pr_err("%s: DSP returned error [%d]\n", __func__,
atomic_read(&ac->cmd_state));
rc = -EINVAL;
goto done;
}
ac->io_mode |= TUN_WRITE_IO_MODE;
rc = 0;
done:
kfree(open);
return rc;
}
EXPORT_SYMBOL(q6asm_open_shared_io);
/*
* q6asm_shared_io_buf: Returns handle to the shared circular buffer being
* used for pull/push mode.
* parameters
* dir - used to identify input/output port
* returns buffer handle
*/
struct audio_buffer *q6asm_shared_io_buf(struct audio_client *ac,
int dir)
{
struct audio_port_data *port;
if (!ac) {
pr_err("%s: ac is null\n", __func__);
return NULL;
}
port = &ac->port[dir];
return port->buf;
}
EXPORT_SYMBOL(q6asm_shared_io_buf);
/*
* q6asm_shared_io_free: Frees memory allocated for a pull/push session
* parameters
* dir - port direction
* returns 0 if successful, error otherwise
*/
int q6asm_shared_io_free(struct audio_client *ac, int dir)
{
struct audio_port_data *port;
if (!ac) {
pr_err("%s: audio client is null\n", __func__);
return -EINVAL;
}
port = &ac->port[dir];
mutex_lock(&ac->cmd_lock);
if (port->buf && port->buf->data) {
msm_audio_ion_free(port->buf->client, port->buf->handle);
port->buf->client = NULL;
port->buf->handle = NULL;
port->max_buf_cnt = 0;
kfree(port->buf);
port->buf = NULL;
}
if (ac->shared_pos_buf.data) {
msm_audio_ion_free(ac->shared_pos_buf.client,
ac->shared_pos_buf.handle);
ac->shared_pos_buf.client = NULL;
ac->shared_pos_buf.handle = NULL;
}
mutex_unlock(&ac->cmd_lock);
return 0;
}
EXPORT_SYMBOL(q6asm_shared_io_free);
/*
* q6asm_get_shared_pos: Returns current read index/write index as observed
* by the DSP. Note that this is an offset and iterates from [0,BUF_SIZE - 1]
* parameters - (all output)
* read_index - offset
* wall_clk_msw1 - ADSP wallclock msw
* wall_clk_lsw1 - ADSP wallclock lsw
* returns 0 if successful, -EAGAIN if DSP failed to update after some
* retries
*/
int q6asm_get_shared_pos(struct audio_client *ac, uint32_t *read_index,
uint32_t *wall_clk_msw1, uint32_t *wall_clk_lsw1)
{
struct asm_shared_position_buffer *pos_buf;
uint32_t frame_cnt1, frame_cnt2;
int i, j;
if (!ac) {
pr_err("%s: audio client is null\n", __func__);
return -EINVAL;
}
pos_buf = ac->shared_pos_buf.data;
/* always try to get the latest update in the shared pos buffer */
for (i = 0; i < 2; i++) {
/* retry until there is an update from DSP */
for (j = 0; j < 5; j++) {
frame_cnt1 = pos_buf->frame_counter;
if (frame_cnt1 != 0)
break;
}
*wall_clk_msw1 = pos_buf->wall_clock_us_msw;
*wall_clk_lsw1 = pos_buf->wall_clock_us_lsw;
*read_index = pos_buf->index;
frame_cnt2 = pos_buf->frame_counter;
if (frame_cnt1 != frame_cnt2)
continue;
return 0;
}
pr_err("%s out of tries trying to get a good read, try again\n",
__func__);
return -EAGAIN;
}
EXPORT_SYMBOL(q6asm_get_shared_pos);
/**
* q6asm_run -
* command to set ASM to run state
*
* @ac: Audio client handle
* @flags: Flags for session
* @msw_ts: upper 32bits timestamp
* @lsw_ts: lower 32bits timestamp
*
* Returns 0 on success or error on failure
*/
int q6asm_run(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
struct asm_session_cmd_run_v2 run;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_add_hdr(ac, &run.hdr, sizeof(run), TRUE);
atomic_set(&ac->cmd_state, -1);
run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
run.flags = flags;
run.time_lsw = lsw_ts;
run.time_msw = msw_ts;
config_debug_fs_run();
rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
if (rc < 0) {
pr_err("%s: Commmand run failed[%d]",
__func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for run success",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_run);
static int __q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts, uint32_t stream_id)
{
struct asm_session_cmd_run_v2 run;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_stream_add_hdr_async(ac, &run.hdr, sizeof(run), TRUE, stream_id);
atomic_set(&ac->cmd_state, 1);
run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
run.flags = flags;
run.time_lsw = lsw_ts;
run.time_msw = msw_ts;
rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
if (rc < 0) {
pr_err("%s: Commmand run failed[%d]", __func__, rc);
return -EINVAL;
}
return 0;
}
/**
* q6asm_run_nowait -
* command to set ASM to run state with no wait for ack
*
* @ac: Audio client handle
* @flags: Flags for session
* @msw_ts: upper 32bits timestamp
* @lsw_ts: lower 32bits timestamp
*
* Returns 0 on success or error on failure
*/
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
return __q6asm_run_nowait(ac, flags, msw_ts, lsw_ts, ac->stream_id);
}
EXPORT_SYMBOL(q6asm_run_nowait);
int q6asm_stream_run_nowait(struct audio_client *ac, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts, uint32_t stream_id)
{
return __q6asm_run_nowait(ac, flags, msw_ts, lsw_ts, stream_id);
}
/**
* q6asm_enc_cfg_blk_aac -
* command to set encode cfg block for aac
*
* @ac: Audio client handle
* @frames_per_buf: number of frames per buffer
* @sample_rate: Sample rate
* @channels: number of ASM channels
* @bit_rate: Bit rate info
* @mode: mode of AAC stream encode
* @format: aac format flag
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
uint32_t frames_per_buf,
uint32_t sample_rate, uint32_t channels,
uint32_t bit_rate, uint32_t mode, uint32_t format)
{
struct asm_aac_enc_cfg_v2 enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]SR[%d]ch[%d]bitrate[%d]mode[%d] format[%d]\n",
__func__, ac->session, frames_per_buf,
sample_rate, channels, bit_rate, mode, format);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_aac_enc_cfg_v2) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.bit_rate = bit_rate;
enc_cfg.enc_mode = mode;
enc_cfg.aac_fmt_flag = format;
enc_cfg.channel_cfg = channels;
enc_cfg.sample_rate = sample_rate;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_aac);
/**
* q6asm_enc_cfg_blk_g711 -
* command to set encode cfg block for g711
*
* @ac: Audio client handle
* @frames_per_buf: number of frames per buffer
* @sample_rate: Sample rate
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_g711(struct audio_client *ac,
uint32_t frames_per_buf,
uint32_t sample_rate)
{
struct asm_g711_enc_cfg_v2 enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]SR[%d]\n",
__func__, ac->session, frames_per_buf,
sample_rate);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_g711_enc_cfg_v2) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.sample_rate = sample_rate;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_g711);
/**
* q6asm_set_encdec_chan_map -
* command to set encdec channel map
*
* @ac: Audio client handle
* @channels: number of channels
*
* Returns 0 on success or error on failure
*/
int q6asm_set_encdec_chan_map(struct audio_client *ac,
uint32_t num_channels)
{
struct asm_dec_out_chan_map_param chan_map;
u8 *channel_mapping;
int rc = 0;
if (num_channels > MAX_CHAN_MAP_CHANNELS) {
pr_err("%s: Invalid channel count %d\n", __func__,
num_channels);
return -EINVAL;
}
pr_debug("%s: Session %d, num_channels = %d\n",
__func__, ac->session, num_channels);
q6asm_add_hdr(ac, &chan_map.hdr, sizeof(chan_map), TRUE);
atomic_set(&ac->cmd_state, -1);
chan_map.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
chan_map.encdec.param_id = ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP;
chan_map.encdec.param_size = sizeof(struct asm_dec_out_chan_map_param) -
(sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_encdec_param));
chan_map.num_channels = num_channels;
channel_mapping = chan_map.channel_mapping;
memset(channel_mapping, PCM_CHANNEL_NULL, MAX_CHAN_MAP_CHANNELS);
if (q6asm_map_channels(channel_mapping, num_channels, false)) {
pr_err("%s: map channels failed %d\n", __func__, num_channels);
return -EINVAL;
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &chan_map);
if (rc < 0) {
pr_err("%s: Command opcode[0x%x]paramid[0x%x] failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n", __func__,
chan_map.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_set_encdec_chan_map);
/*
* q6asm_enc_cfg_blk_pcm_v5 - sends encoder configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @use_back_flavor: to configure back left and right channel
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
static int q6asm_enc_cfg_blk_pcm_v5(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample, bool use_default_chmap,
bool use_back_flavor, u8 *channel_map,
uint16_t sample_word_size, uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_enc_cfg_v5 enc_cfg;
struct asm_enc_cfg_blk_param_v2 enc_fg_blk;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int rc;
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
rc = -EINVAL;
goto fail_cmd;
}
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL_V8) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&enc_cfg, 0, sizeof(enc_cfg));
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(enc_fg_blk);
enc_cfg.num_channels = channels;
enc_cfg.bits_per_sample = bits_per_sample;
enc_cfg.sample_rate = rate;
enc_cfg.is_signed = 1;
enc_cfg.sample_word_size = sample_word_size;
enc_cfg.endianness = endianness;
enc_cfg.mode = mode;
channel_mapping = enc_cfg.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL_V8);
if (use_default_chmap) {
pr_debug("%s: setting default channel map for %d channels",
__func__, channels);
if (q6asm_map_channels(channel_mapping, channels,
use_back_flavor)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
pr_debug("%s: Using pre-defined channel map", __func__);
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL_V8);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Command open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, enc_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_v5);
/*
* q6asm_enc_cfg_blk_pcm_v4 - sends encoder configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @use_back_flavor: to configure back left and right channel
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_enc_cfg_blk_pcm_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample, bool use_default_chmap,
bool use_back_flavor, u8 *channel_map,
uint16_t sample_word_size, uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_enc_cfg_v4 enc_cfg;
struct asm_enc_cfg_blk_param_v2 enc_fg_blk;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int rc;
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
rc = -EINVAL;
goto fail_cmd;
}
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&enc_cfg, 0, sizeof(enc_cfg));
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(enc_fg_blk);
enc_cfg.num_channels = channels;
enc_cfg.bits_per_sample = bits_per_sample;
enc_cfg.sample_rate = rate;
enc_cfg.is_signed = 1;
enc_cfg.sample_word_size = sample_word_size;
enc_cfg.endianness = endianness;
enc_cfg.mode = mode;
channel_mapping = enc_cfg.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
pr_debug("%s: setting default channel map for %d channels",
__func__, channels);
if (q6asm_map_channels(channel_mapping, channels,
use_back_flavor)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
pr_debug("%s: Using pre-defined channel map", __func__);
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Command open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, enc_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_v4);
/*
* q6asm_enc_cfg_blk_pcm_v3 - sends encoder configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @use_back_flavor: to configure back left and right channel
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
*/
int q6asm_enc_cfg_blk_pcm_v3(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample, bool use_default_chmap,
bool use_back_flavor, u8 *channel_map,
uint16_t sample_word_size)
{
struct asm_multi_channel_pcm_enc_cfg_v3 enc_cfg;
struct asm_enc_cfg_blk_param_v2 enc_fg_blk;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int rc;
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
rc = -EINVAL;
goto fail_cmd;
}
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&enc_cfg, 0, sizeof(enc_cfg));
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(enc_fg_blk);
enc_cfg.num_channels = channels;
enc_cfg.bits_per_sample = bits_per_sample;
enc_cfg.sample_rate = rate;
enc_cfg.is_signed = 1;
enc_cfg.sample_word_size = sample_word_size;
channel_mapping = enc_cfg.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
pr_debug("%s: setting default channel map for %d channels",
__func__, channels);
if (q6asm_map_channels(channel_mapping, channels,
use_back_flavor)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
pr_debug("%s: Using pre-defined channel map", __func__);
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, enc_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_v3);
/**
* q6asm_enc_cfg_blk_pcm_v2 -
* command to set encode config block for pcm_v2
*
* @ac: Audio client handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: number of bits per sample
* @use_default_chmap: Flag indicating to use default ch_map or not
* @use_back_flavor: back flavor flag
* @channel_map: Custom channel map settings
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_pcm_v2(struct audio_client *ac,
uint32_t rate, uint32_t channels, uint16_t bits_per_sample,
bool use_default_chmap, bool use_back_flavor, u8 *channel_map)
{
struct asm_multi_channel_pcm_enc_cfg_v2 enc_cfg;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int rc = 0;
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
return -EINVAL;
}
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: Session %d, rate = %d, channels = %d\n", __func__,
ac->session, rate, channels);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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 (use_default_chmap) {
pr_debug("%s: setting default channel map for %d channels",
__func__, channels);
if (q6asm_map_channels(channel_mapping, channels,
use_back_flavor)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
return -EINVAL;
}
} else {
pr_debug("%s: Using pre-defined channel map", __func__);
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, enc_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_v2);
static int __q6asm_enc_cfg_blk_pcm_v5(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return q6asm_enc_cfg_blk_pcm_v5(ac, rate, channels,
bits_per_sample, true, false, NULL,
sample_word_size, endianness, mode);
}
static int __q6asm_enc_cfg_blk_pcm_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return q6asm_enc_cfg_blk_pcm_v4(ac, rate, channels,
bits_per_sample, true, false, NULL,
sample_word_size, endianness, mode);
}
static int __q6asm_enc_cfg_blk_pcm_v3(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size)
{
return q6asm_enc_cfg_blk_pcm_v3(ac, rate, channels,
bits_per_sample, true, false, NULL,
sample_word_size);
}
static int __q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
{
return q6asm_enc_cfg_blk_pcm_v2(ac, rate, channels,
bits_per_sample, true, false, NULL);
}
/**
* q6asm_enc_cfg_blk_pcm -
* command to set encode config block for pcm
*
* @ac: Audio client handle
* @rate: sample rate
* @channels: number of channels
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels)
{
return __q6asm_enc_cfg_blk_pcm(ac, rate, channels, 16);
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm);
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
{
return __q6asm_enc_cfg_blk_pcm(ac, rate, channels, bits_per_sample);
}
/*
* q6asm_enc_cfg_blk_pcm_format_support_v3 - sends encoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @sample_word_size: Size in bits of the word that holds a sample of a channel
*/
int q6asm_enc_cfg_blk_pcm_format_support_v3(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size)
{
return __q6asm_enc_cfg_blk_pcm_v3(ac, rate, channels,
bits_per_sample, sample_word_size);
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_format_support_v3);
/*
* q6asm_enc_cfg_blk_pcm_format_support_v4 - sends encoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return __q6asm_enc_cfg_blk_pcm_v4(ac, rate, channels,
bits_per_sample, sample_word_size,
endianness, mode);
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_format_support_v4);
/*
* q6asm_enc_cfg_blk_pcm_format_support_v5 - sends encoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_enc_cfg_blk_pcm_format_support_v5(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return __q6asm_enc_cfg_blk_pcm_v5(ac, rate, channels,
bits_per_sample, sample_word_size,
endianness, mode);
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_format_support_v5);
/**
* q6asm_enc_cfg_blk_pcm_native -
* command to set encode config block for pcm_native
*
* @ac: Audio client handle
* @rate: sample rate
* @channels: number of channels
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_pcm_native(struct audio_client *ac,
uint32_t rate, uint32_t channels)
{
struct asm_multi_channel_pcm_enc_cfg_v2 enc_cfg;
u8 *channel_mapping;
u32 frames_per_buf = 0;
int rc = 0;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: Session %d, rate = %d, channels = %d\n", __func__,
ac->session, rate, channels);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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 = 0;/*channels;*/
enc_cfg.bits_per_sample = 16;
enc_cfg.sample_rate = 0;/*rate;*/
enc_cfg.is_signed = 1;
channel_mapping = enc_cfg.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n", __func__, channels);
return -EINVAL;
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, enc_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_native);
static int q6asm_map_channels(u8 *channel_mapping, uint32_t channels,
bool use_back_flavor)
{
u8 *lchannel_mapping;
lchannel_mapping = channel_mapping;
pr_debug("%s: channels passed: %d\n", __func__, channels);
if (channels == 1) {
lchannel_mapping[0] = PCM_CHANNEL_FC;
} else if (channels == 2) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
} else if (channels == 3) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
} else if (channels == 4) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = use_back_flavor ?
PCM_CHANNEL_LB : PCM_CHANNEL_LS;
lchannel_mapping[3] = use_back_flavor ?
PCM_CHANNEL_RB : PCM_CHANNEL_RS;
} else if (channels == 5) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = use_back_flavor ?
PCM_CHANNEL_LB : PCM_CHANNEL_LS;
lchannel_mapping[4] = use_back_flavor ?
PCM_CHANNEL_RB : PCM_CHANNEL_RS;
} else if (channels == 6) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = PCM_CHANNEL_LFE;
lchannel_mapping[4] = use_back_flavor ?
PCM_CHANNEL_LB : PCM_CHANNEL_LS;
lchannel_mapping[5] = use_back_flavor ?
PCM_CHANNEL_RB : PCM_CHANNEL_RS;
} else if (channels == 7) {
/*
* Configured for 5.1 channel mapping + 1 channel for debug
* Can be customized based on DSP.
*/
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = PCM_CHANNEL_LFE;
lchannel_mapping[4] = use_back_flavor ?
PCM_CHANNEL_LB : PCM_CHANNEL_LS;
lchannel_mapping[5] = use_back_flavor ?
PCM_CHANNEL_RB : PCM_CHANNEL_RS;
lchannel_mapping[6] = PCM_CHANNEL_CS;
} else if (channels == 8) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = PCM_CHANNEL_LFE;
lchannel_mapping[4] = PCM_CHANNEL_LB;
lchannel_mapping[5] = PCM_CHANNEL_RB;
lchannel_mapping[6] = PCM_CHANNEL_LS;
lchannel_mapping[7] = PCM_CHANNEL_RS;
} else if (channels == 12) {
/*
* Configured for 7.1.4 channel mapping
* Todo: Needs to be checked
*/
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = PCM_CHANNEL_LFE;
lchannel_mapping[4] = PCM_CHANNEL_LB;
lchannel_mapping[5] = PCM_CHANNEL_RB;
lchannel_mapping[6] = PCM_CHANNEL_LS;
lchannel_mapping[7] = PCM_CHANNEL_RS;
lchannel_mapping[8] = PCM_CHANNEL_TFL;
lchannel_mapping[9] = PCM_CHANNEL_TFR;
lchannel_mapping[10] = PCM_CHANNEL_TSL;
lchannel_mapping[11] = PCM_CHANNEL_TSR;
} else if (channels == 16) {
/*
* Configured for 7.1.8 channel mapping
* Todo: Needs to be checked
*/
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_FC;
lchannel_mapping[3] = PCM_CHANNEL_LFE;
lchannel_mapping[4] = PCM_CHANNEL_LB;
lchannel_mapping[5] = PCM_CHANNEL_RB;
lchannel_mapping[6] = PCM_CHANNEL_LS;
lchannel_mapping[7] = PCM_CHANNEL_RS;
lchannel_mapping[8] = PCM_CHANNEL_TFL;
lchannel_mapping[9] = PCM_CHANNEL_TFR;
lchannel_mapping[10] = PCM_CHANNEL_TSL;
lchannel_mapping[11] = PCM_CHANNEL_TSR;
lchannel_mapping[12] = PCM_CHANNEL_FLC;
lchannel_mapping[13] = PCM_CHANNEL_FRC;
lchannel_mapping[14] = PCM_CHANNEL_RLC;
lchannel_mapping[15] = PCM_CHANNEL_RRC;
} else if (channels == 32) {
lchannel_mapping[0] = PCM_CHANNEL_FL;
lchannel_mapping[1] = PCM_CHANNEL_FR;
lchannel_mapping[2] = PCM_CHANNEL_LFE;
lchannel_mapping[3] = PCM_CHANNEL_FC;
lchannel_mapping[4] = PCM_CHANNEL_LS;
lchannel_mapping[5] = PCM_CHANNEL_RS;
lchannel_mapping[6] = PCM_CHANNEL_LB;
lchannel_mapping[7] = PCM_CHANNEL_RB;
lchannel_mapping[8] = PCM_CHANNEL_CS;
lchannel_mapping[9] = PCM_CHANNELS;
lchannel_mapping[10] = PCM_CHANNEL_CVH;
lchannel_mapping[11] = PCM_CHANNEL_MS;
lchannel_mapping[12] = PCM_CHANNEL_FLC;
lchannel_mapping[13] = PCM_CHANNEL_FRC;
lchannel_mapping[14] = PCM_CHANNEL_RLC;
lchannel_mapping[15] = PCM_CHANNEL_RRC;
lchannel_mapping[16] = PCM_CHANNEL_LFE2;
lchannel_mapping[17] = PCM_CHANNEL_SL;
lchannel_mapping[18] = PCM_CHANNEL_SR;
lchannel_mapping[19] = PCM_CHANNEL_TFL;
lchannel_mapping[20] = PCM_CHANNEL_TFR;
lchannel_mapping[21] = PCM_CHANNEL_TC;
lchannel_mapping[22] = PCM_CHANNEL_TBL;
lchannel_mapping[23] = PCM_CHANNEL_TBR;
lchannel_mapping[24] = PCM_CHANNEL_TSL;
lchannel_mapping[25] = PCM_CHANNEL_TSR;
lchannel_mapping[26] = PCM_CHANNEL_TBC;
lchannel_mapping[27] = PCM_CHANNEL_BFC;
lchannel_mapping[28] = PCM_CHANNEL_BFL;
lchannel_mapping[29] = PCM_CHANNEL_BFR;
lchannel_mapping[30] = PCM_CHANNEL_LW;
lchannel_mapping[31] = PCM_CHANNEL_RW;
} else {
pr_err("%s: ERROR.unsupported num_ch = %u\n",
__func__, channels);
return -EINVAL;
}
return 0;
}
/**
* q6asm_enable_sbrps -
* command to enable sbrps for ASM
*
* @ac: Audio client handle
* @sbr_ps_enable: flag for sbr_ps enable or disable
*
* Returns 0 on success or error on failure
*/
int q6asm_enable_sbrps(struct audio_client *ac,
uint32_t sbr_ps_enable)
{
struct asm_aac_sbr_ps_flag_param sbrps;
u32 frames_per_buf = 0;
int rc = 0;
pr_debug("%s: Session %d\n", __func__, ac->session);
q6asm_add_hdr(ac, &sbrps.hdr, sizeof(sbrps), TRUE);
atomic_set(&ac->cmd_state, -1);
sbrps.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
sbrps.encdec.param_id = ASM_PARAM_ID_AAC_SBR_PS_FLAG;
sbrps.encdec.param_size = sizeof(struct asm_aac_sbr_ps_flag_param) -
sizeof(struct asm_stream_cmd_set_encdec_param);
sbrps.encblk.frames_per_buf = frames_per_buf;
sbrps.encblk.enc_cfg_blk_size = sbrps.encdec.param_size -
sizeof(struct asm_enc_cfg_blk_param_v2);
sbrps.sbr_ps_flag = sbr_ps_enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &sbrps);
if (rc < 0) {
pr_err("%s: Command opcode[0x%x]paramid[0x%x] failed %d\n",
__func__,
ASM_STREAM_CMD_SET_ENCDEC_PARAM,
ASM_PARAM_ID_AAC_SBR_PS_FLAG, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x] ", __func__, sbrps.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enable_sbrps);
/**
* q6asm_cfg_dual_mono_aac -
* command to set config for dual mono aac
*
* @ac: Audio client handle
* @sce_left: left sce val
* @sce_right: right sce val
*
* Returns 0 on success or error on failure
*/
int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
uint16_t sce_left, uint16_t sce_right)
{
struct asm_aac_dual_mono_mapping_param dual_mono;
int rc = 0;
pr_debug("%s: Session %d, sce_left = %d, sce_right = %d\n",
__func__, ac->session, sce_left, sce_right);
q6asm_add_hdr(ac, &dual_mono.hdr, sizeof(dual_mono), TRUE);
atomic_set(&ac->cmd_state, -1);
dual_mono.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
dual_mono.encdec.param_id = ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING;
dual_mono.encdec.param_size = sizeof(dual_mono.left_channel_sce) +
sizeof(dual_mono.right_channel_sce);
dual_mono.left_channel_sce = sce_left;
dual_mono.right_channel_sce = sce_right;
rc = apr_send_pkt(ac->apr, (uint32_t *) &dual_mono);
if (rc < 0) {
pr_err("%s: Command opcode[0x%x]paramid[0x%x] failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n", __func__,
dual_mono.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_cfg_dual_mono_aac);
/* Support for selecting stereo mixing coefficients for B family not done */
int q6asm_cfg_aac_sel_mix_coef(struct audio_client *ac, uint32_t mix_coeff)
{
struct asm_aac_stereo_mix_coeff_selection_param_v2 aac_mix_coeff;
int rc = 0;
q6asm_add_hdr(ac, &aac_mix_coeff.hdr, sizeof(aac_mix_coeff), TRUE);
atomic_set(&ac->cmd_state, -1);
aac_mix_coeff.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
aac_mix_coeff.param_id =
ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2;
aac_mix_coeff.param_size =
sizeof(struct asm_aac_stereo_mix_coeff_selection_param_v2);
aac_mix_coeff.aac_stereo_mix_coeff_flag = mix_coeff;
pr_debug("%s: mix_coeff = %u\n", __func__, mix_coeff);
rc = apr_send_pkt(ac->apr, (uint32_t *) &aac_mix_coeff);
if (rc < 0) {
pr_err("%s: Command opcode[0x%x]paramid[0x%x] failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2,
rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n",
__func__, aac_mix_coeff.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_cfg_aac_sel_mix_coef);
/**
* q6asm_enc_cfg_blk_qcelp -
* command to set encode config block for QCELP
*
* @ac: Audio client handle
* @frames_per_buf: Number of frames per buffer
* @min_rate: Minimum Enc rate
* @max_rate: Maximum Enc rate
* reduced_rate_level: Reduced rate level
* @rate_modulation_cmd: rate modulation command
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
uint16_t min_rate, uint16_t max_rate,
uint16_t reduced_rate_level, uint16_t rate_modulation_cmd)
{
struct asm_v13k_enc_cfg enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x] reduced_rate_level[0x%4x]rate_modulation_cmd[0x%4x]\n",
__func__,
ac->session, frames_per_buf, min_rate, max_rate,
reduced_rate_level, rate_modulation_cmd);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_v13k_enc_cfg) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.min_rate = min_rate;
enc_cfg.max_rate = max_rate;
enc_cfg.reduced_rate_cmd = reduced_rate_level;
enc_cfg.rate_mod_cmd = rate_modulation_cmd;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for setencdec v13k resp\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_qcelp);
/**
* q6asm_enc_cfg_blk_evrc -
* command to set encode config block for EVRC
*
* @ac: Audio client handle
* @frames_per_buf: Number of frames per buffer
* @min_rate: Minimum Enc rate
* @max_rate: Maximum Enc rate
* @rate_modulation_cmd: rate modulation command
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
uint16_t min_rate, uint16_t max_rate,
uint16_t rate_modulation_cmd)
{
struct asm_evrc_enc_cfg enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x] rate_modulation_cmd[0x%4x]\n",
__func__, ac->session,
frames_per_buf, min_rate, max_rate, rate_modulation_cmd);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_evrc_enc_cfg) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.min_rate = min_rate;
enc_cfg.max_rate = max_rate;
enc_cfg.rate_mod_cmd = rate_modulation_cmd;
enc_cfg.reserved = 0;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for encdec evrc\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_evrc);
/**
* q6asm_enc_cfg_blk_amrnb -
* command to set encode config block for AMRNB
*
* @ac: Audio client handle
* @frames_per_buf: Number of frames per buffer
* @band_mode: Band mode used
* @dtx_enable: DTX en flag
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
uint16_t band_mode, uint16_t dtx_enable)
{
struct asm_amrnb_enc_cfg enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]\n",
__func__, ac->session, frames_per_buf, band_mode, dtx_enable);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_amrnb_enc_cfg) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.enc_mode = band_mode;
enc_cfg.dtx_mode = dtx_enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for set encdec amrnb\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_amrnb);
/**
* q6asm_enc_cfg_blk_amrwb -
* command to set encode config block for AMRWB
*
* @ac: Audio client handle
* @frames_per_buf: Number of frames per buffer
* @band_mode: Band mode used
* @dtx_enable: DTX en flag
*
* Returns 0 on success or error on failure
*/
int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
uint16_t band_mode, uint16_t dtx_enable)
{
struct asm_amrwb_enc_cfg enc_cfg;
int rc = 0;
pr_debug("%s: session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]\n",
__func__, ac->session, frames_per_buf, band_mode, dtx_enable);
q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
atomic_set(&ac->cmd_state, -1);
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(struct asm_amrwb_enc_cfg) -
sizeof(struct asm_stream_cmd_set_encdec_param);
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.enc_mode = band_mode;
enc_cfg.dtx_mode = dtx_enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
if (rc < 0) {
pr_err("%s: Comamnd %d failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_amrwb);
static int __q6asm_media_format_block_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample, int stream_id,
bool use_default_chmap, char *channel_map)
{
struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
u8 *channel_mapping;
int rc = 0;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
channels);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&fmt.hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, fmt.hdr.token, stream_id, ac->session);
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;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
return -EINVAL;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_pcm_v3(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size)
{
struct asm_multi_channel_pcm_fmt_blk_param_v3 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
fmt.hdr.token = ((ac->session << 8) & 0xFFFF00) |
(stream_id & 0xFF);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, fmt.hdr.token, stream_id, ac->session);
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.param.num_channels = channels;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_pcm_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_fmt_blk_param_v4 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
fmt.hdr.token = ((ac->session << 8) & 0xFFFF00) |
(stream_id & 0xFF);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, fmt.hdr.token, stream_id, ac->session);
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.param.num_channels = channels;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
fmt.param.endianness = endianness;
fmt.param.mode = mode;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_pcm_v5(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_fmt_blk_param_v5 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL_V8) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
fmt.hdr.token = ((ac->session << 8) & 0xFFFF00) |
(stream_id & 0xFF);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, fmt.hdr.token, stream_id, ac->session);
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.param.num_channels = (uint16_t) channels & 0xFFFF;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
fmt.param.endianness = endianness;
fmt.param.mode = mode;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL_V8);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, fmt.param.num_channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL_V8);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_media_format_block_pcm -
* command to set mediafmt block for PCM on ASM stream
*
* @ac: Audio client handle
* @rate: sample rate
* @channels: number of ASM channels
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels)
{
return __q6asm_media_format_block_pcm(ac, rate,
channels, 16, ac->stream_id,
true, NULL);
}
EXPORT_SYMBOL(q6asm_media_format_block_pcm);
/**
* q6asm_media_format_block_pcm_format_support -
* command to set mediafmt block for PCM format support
*
* @ac: Audio client handle
* @rate: sample rate
* @channels: number of ASM channels
* @bits_per_sample: number of bits per sample
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_pcm_format_support(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample)
{
return __q6asm_media_format_block_pcm(ac, rate,
channels, bits_per_sample, ac->stream_id,
true, NULL);
}
EXPORT_SYMBOL(q6asm_media_format_block_pcm_format_support);
int q6asm_media_format_block_pcm_format_support_v2(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample, int stream_id,
bool use_default_chmap, char *channel_map)
{
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
return -EINVAL;
}
return __q6asm_media_format_block_pcm(ac, rate,
channels, bits_per_sample, stream_id,
use_default_chmap, channel_map);
}
/*
* q6asm_media_format_block_pcm_format_support_v3- sends pcm decoder
* configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @stream_id: stream id of stream to be associated with this session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
*/
int q6asm_media_format_block_pcm_format_support_v3(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size)
{
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
return -EINVAL;
}
return __q6asm_media_format_block_pcm_v3(ac, rate,
channels, bits_per_sample, stream_id,
use_default_chmap, channel_map,
sample_word_size);
}
EXPORT_SYMBOL(q6asm_media_format_block_pcm_format_support_v3);
/*
* q6asm_media_format_block_pcm_format_support_v4- sends pcm decoder
* configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @stream_id: stream id of stream to be associated with this session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_media_format_block_pcm_format_support_v4(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
return -EINVAL;
}
return __q6asm_media_format_block_pcm_v4(ac, rate,
channels, bits_per_sample, stream_id,
use_default_chmap, channel_map,
sample_word_size, endianness,
mode);
}
EXPORT_SYMBOL(q6asm_media_format_block_pcm_format_support_v4);
/*
* q6asm_media_format_block_pcm_format_support_v5- sends pcm decoder
* configuration parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @stream_id: stream id of stream to be associated with this session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_media_format_block_pcm_format_support_v5(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
uint16_t bits_per_sample,
int stream_id,
bool use_default_chmap,
char *channel_map,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
if (!use_default_chmap && (channel_map == NULL)) {
pr_err("%s: No valid chan map and can't use default\n",
__func__);
return -EINVAL;
}
return __q6asm_media_format_block_pcm_v5(ac, rate,
channels, bits_per_sample, stream_id,
use_default_chmap, channel_map,
sample_word_size, endianness,
mode);
}
EXPORT_SYMBOL(q6asm_media_format_block_pcm_format_support_v5);
static int __q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap, char *channel_map,
uint16_t bits_per_sample)
{
struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
u8 *channel_mapping;
int rc = 0;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
channels);
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
atomic_set(&ac->cmd_state, -1);
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;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
return -EINVAL;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_multi_ch_pcm_v3(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size)
{
struct asm_multi_channel_pcm_fmt_blk_param_v3 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
atomic_set(&ac->cmd_state, -1);
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.param.num_channels = channels;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_fmt_blk_param_v4 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
atomic_set(&ac->cmd_state, -1);
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.param.num_channels = channels;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
fmt.param.endianness = endianness;
fmt.param.mode = mode;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static int __q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac,
uint32_t rate,
uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
struct asm_multi_channel_pcm_fmt_blk_param_v5 fmt;
u8 *channel_mapping;
int rc;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL_V8) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
ac->session, rate, channels,
bits_per_sample, sample_word_size);
memset(&fmt, 0, sizeof(fmt));
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
atomic_set(&ac->cmd_state, -1);
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.param.num_channels = channels;
fmt.param.bits_per_sample = bits_per_sample;
fmt.param.sample_rate = rate;
fmt.param.is_signed = 1;
fmt.param.sample_word_size = sample_word_size;
fmt.param.endianness = endianness;
fmt.param.mode = mode;
channel_mapping = fmt.param.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL_V8);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL_V8);
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap, char *channel_map)
{
return __q6asm_media_format_block_multi_ch_pcm(ac, rate,
channels, use_default_chmap, channel_map, 16);
}
int q6asm_media_format_block_multi_ch_pcm_v2(
struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap, char *channel_map,
uint16_t bits_per_sample)
{
return __q6asm_media_format_block_multi_ch_pcm(ac, rate,
channels, use_default_chmap, channel_map,
bits_per_sample);
}
/*
* q6asm_media_format_block_multi_ch_pcm_v3 - sends pcm decoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
*/
int q6asm_media_format_block_multi_ch_pcm_v3(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size)
{
return __q6asm_media_format_block_multi_ch_pcm_v3(ac, rate, channels,
use_default_chmap,
channel_map,
bits_per_sample,
sample_word_size);
}
EXPORT_SYMBOL(q6asm_media_format_block_multi_ch_pcm_v3);
/*
* q6asm_media_format_block_multi_ch_pcm_v4 - sends pcm decoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return __q6asm_media_format_block_multi_ch_pcm_v4(ac, rate, channels,
use_default_chmap,
channel_map,
bits_per_sample,
sample_word_size,
endianness,
mode);
}
EXPORT_SYMBOL(q6asm_media_format_block_multi_ch_pcm_v4);
/*
* q6asm_media_format_block_multi_ch_pcm_v5 - sends pcm decoder configuration
* parameters
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @bits_per_sample: bit width of encoder session
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @sample_word_size: Size in bits of the word that holds a sample of a channel
* @endianness: endianness of the pcm data
* @mode: Mode to provide additional info about the pcm input data
*/
int q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap,
char *channel_map,
uint16_t bits_per_sample,
uint16_t sample_word_size,
uint16_t endianness,
uint16_t mode)
{
return __q6asm_media_format_block_multi_ch_pcm_v5(ac, rate, channels,
use_default_chmap,
channel_map,
bits_per_sample,
sample_word_size,
endianness,
mode);
}
EXPORT_SYMBOL(q6asm_media_format_block_multi_ch_pcm_v5);
/*
* q6asm_media_format_block_gen_compr - set up generic compress format params
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
* @use_default_chmap: true if default channel map to be used
* @channel_map: input channel map
* @bits_per_sample: bit width of gen compress stream
*/
int q6asm_media_format_block_gen_compr(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap, char *channel_map,
uint16_t bits_per_sample)
{
struct asm_generic_compressed_fmt_blk_t fmt;
u8 *channel_mapping;
int rc = 0;
if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) {
pr_err("%s: Invalid channel count %d\n", __func__, channels);
return -EINVAL;
}
pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]\n",
__func__, ac->session, rate,
channels, bits_per_sample);
memset(&fmt, 0, sizeof(fmt));
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
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.sampling_rate = rate;
channel_mapping = fmt.channel_mapping;
memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
if (use_default_chmap) {
if (q6asm_map_channels(channel_mapping, channels, false)) {
pr_err("%s: map channels failed %d\n",
__func__, channels);
return -EINVAL;
}
} else {
memcpy(channel_mapping, channel_map,
PCM_FORMAT_MAX_NUM_CHANNEL);
}
atomic_set(&ac->cmd_state, -1);
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_gen_compr);
/*
* q6asm_media_format_block_iec - set up IEC61937 (compressed) or IEC60958
* (pcm) format params. Both audio standards
* use the same format and are used for
* HDMI or SPDIF.
*
* @ac: Client session handle
* @rate: sample rate
* @channels: number of channels
*/
int q6asm_media_format_block_iec(struct audio_client *ac,
uint32_t rate, uint32_t channels)
{
struct asm_iec_compressed_fmt_blk_t fmt;
int rc = 0;
pr_debug("%s: session[%d]rate[%d]ch[%d]\n",
__func__, ac->session, rate,
channels);
memset(&fmt, 0, sizeof(fmt));
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
fmt.hdr.opcode = ASM_DATA_CMD_IEC_60958_MEDIA_FMT;
fmt.num_channels = channels;
fmt.sampling_rate = rate;
atomic_set(&ac->cmd_state, -1);
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for format update\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_iec);
static int __q6asm_media_format_block_multi_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg, int stream_id)
{
struct asm_aac_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s: session[%d]rate[%d]ch[%d]\n", __func__, ac->session,
cfg->sample_rate, cfg->ch_cfg);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&fmt.hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, fmt.hdr.token, stream_id, ac->session);
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.aac_fmt_flag = cfg->format;
fmt.audio_objype = cfg->aot;
/* If zero, PCE is assumed to be available in bitstream*/
fmt.total_size_of_PCE_bits = 0;
fmt.channel_config = cfg->ch_cfg;
fmt.sample_rate = cfg->sample_rate;
pr_debug("%s: format=0x%x cfg_size=%d aac-cfg=0x%x aot=%d ch=%d sr=%d\n",
__func__, fmt.aac_fmt_flag, fmt.fmt_blk.fmt_blk_size,
fmt.aac_fmt_flag,
fmt.audio_objype,
fmt.channel_config,
fmt.sample_rate);
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_media_format_block_multi_aac -
* command to set mediafmt block for multi_aac on ASM stream
*
* @ac: Audio client handle
* @cfg: multi_aac config
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_multi_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg)
{
return __q6asm_media_format_block_multi_aac(ac, cfg, ac->stream_id);
}
EXPORT_SYMBOL(q6asm_media_format_block_multi_aac);
/**
* q6asm_media_format_block_aac -
* command to set mediafmt block for aac on ASM
*
* @ac: Audio client handle
* @cfg: aac config
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg)
{
return __q6asm_media_format_block_multi_aac(ac, cfg, ac->stream_id);
}
EXPORT_SYMBOL(q6asm_media_format_block_aac);
/**
* q6asm_stream_media_format_block_aac -
* command to set mediafmt block for aac on ASM stream
*
* @ac: Audio client handle
* @cfg: aac config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_media_format_block_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg, int stream_id)
{
return __q6asm_media_format_block_multi_aac(ac, cfg, stream_id);
}
EXPORT_SYMBOL(q6asm_stream_media_format_block_aac);
/**
* q6asm_media_format_block_wma -
* command to set mediafmt block for wma on ASM stream
*
* @ac: Audio client handle
* @cfg: wma config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_wma(struct audio_client *ac,
void *cfg, int stream_id)
{
struct asm_wmastdv9_fmt_blk_v2 fmt;
struct asm_wma_cfg *wma_cfg = (struct asm_wma_cfg *)cfg;
int rc = 0;
pr_debug("session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d], balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x]\n",
ac->session, wma_cfg->format_tag, wma_cfg->sample_rate,
wma_cfg->ch_cfg, wma_cfg->avg_bytes_per_sec,
wma_cfg->block_align, wma_cfg->valid_bits_per_sample,
wma_cfg->ch_mask, wma_cfg->encode_opt);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.fmtag = wma_cfg->format_tag;
fmt.num_channels = wma_cfg->ch_cfg;
fmt.sample_rate = wma_cfg->sample_rate;
fmt.avg_bytes_per_sec = wma_cfg->avg_bytes_per_sec;
fmt.blk_align = wma_cfg->block_align;
fmt.bits_per_sample =
wma_cfg->valid_bits_per_sample;
fmt.channel_mask = wma_cfg->ch_mask;
fmt.enc_options = wma_cfg->encode_opt;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_wma);
/**
* q6asm_media_format_block_wmapro -
* command to set mediafmt block for wmapro on ASM stream
*
* @ac: Audio client handle
* @cfg: wmapro config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_wmapro(struct audio_client *ac,
void *cfg, int stream_id)
{
struct asm_wmaprov10_fmt_blk_v2 fmt;
struct asm_wmapro_cfg *wmapro_cfg = (struct asm_wmapro_cfg *)cfg;
int rc = 0;
pr_debug("%s: session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d], balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x], adv_enc_opt[0x%4x], adv_enc_opt2[0x%8x]\n",
__func__,
ac->session, wmapro_cfg->format_tag, wmapro_cfg->sample_rate,
wmapro_cfg->ch_cfg, wmapro_cfg->avg_bytes_per_sec,
wmapro_cfg->block_align, wmapro_cfg->valid_bits_per_sample,
wmapro_cfg->ch_mask, wmapro_cfg->encode_opt,
wmapro_cfg->adv_encode_opt, wmapro_cfg->adv_encode_opt2);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.fmtag = wmapro_cfg->format_tag;
fmt.num_channels = wmapro_cfg->ch_cfg;
fmt.sample_rate = wmapro_cfg->sample_rate;
fmt.avg_bytes_per_sec =
wmapro_cfg->avg_bytes_per_sec;
fmt.blk_align = wmapro_cfg->block_align;
fmt.bits_per_sample = wmapro_cfg->valid_bits_per_sample;
fmt.channel_mask = wmapro_cfg->ch_mask;
fmt.enc_options = wmapro_cfg->encode_opt;
fmt.usAdvancedEncodeOpt = wmapro_cfg->adv_encode_opt;
fmt.advanced_enc_options2 = wmapro_cfg->adv_encode_opt2;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd open failed %d\n", __func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_wmapro);
/**
* q6asm_media_format_block_amrwbplus -
* command to set mediafmt block for amrwbplus on ASM stream
*
* @ac: Audio client handle
* @cfg: amrwbplus config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_amrwbplus(struct audio_client *ac,
struct asm_amrwbplus_cfg *cfg)
{
struct asm_amrwbplus_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s: session[%d]band-mode[%d]frame-fmt[%d]ch[%d]\n",
__func__,
ac->session,
cfg->amr_band_mode,
cfg->amr_frame_fmt,
cfg->num_channels);
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.amr_frame_fmt = cfg->amr_frame_fmt;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Comamnd media format update failed.. %d\n",
__func__, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_amrwbplus);
/**
* q6asm_stream_media_format_block_flac -
* command to set mediafmt block for flac on ASM stream
*
* @ac: Audio client handle
* @cfg: FLAC config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct asm_flac_cfg *cfg, int stream_id)
{
struct asm_flac_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s :session[%d] rate[%d] ch[%d] size[%d] stream_id[%d]\n",
__func__, ac->session, cfg->sample_rate, cfg->ch_cfg,
cfg->sample_size, stream_id);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
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 = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s :Comamnd media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_stream_media_format_block_flac);
/**
* q6asm_media_format_block_alac -
* command to set mediafmt block for alac on ASM stream
*
* @ac: Audio client handle
* @cfg: ALAC config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_alac(struct audio_client *ac,
struct asm_alac_cfg *cfg, int stream_id)
{
struct asm_alac_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s :session[%d]rate[%d]ch[%d]\n", __func__,
ac->session, cfg->sample_rate, cfg->num_channels);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.frame_length = cfg->frame_length;
fmt.compatible_version = cfg->compatible_version;
fmt.bit_depth = cfg->bit_depth;
fmt.pb = cfg->pb;
fmt.mb = cfg->mb;
fmt.kb = cfg->kb;
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;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s :Comamnd media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_alac);
/*
* q6asm_media_format_block_g711 - sends g711 decoder configuration
* parameters
* @ac: Client session handle
* @cfg: Audio stream manager configuration parameters
* @stream_id: Stream id
*/
int q6asm_media_format_block_g711(struct audio_client *ac,
struct asm_g711_dec_cfg *cfg, int stream_id)
{
struct asm_g711_dec_fmt_blk_v2 fmt;
int rc = 0;
if (!ac) {
pr_err("%s: audio client is null\n", __func__);
return -EINVAL;
}
if (!cfg) {
pr_err("%s: Invalid ASM config\n", __func__);
return -EINVAL;
}
if (stream_id <= 0) {
pr_err("%s: Invalid stream id\n", __func__);
return -EINVAL;
}
pr_debug("%s :session[%d]rate[%d]\n", __func__,
ac->session, cfg->sample_rate);
memset(&fmt, 0, sizeof(struct asm_g711_dec_fmt_blk_v2));
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.sample_rate = cfg->sample_rate;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s :Command media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_g711);
/**
* q6asm_stream_media_format_block_vorbis -
* command to set mediafmt block for vorbis on ASM stream
*
* @ac: Audio client handle
* @cfg: vorbis config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_media_format_block_vorbis(struct audio_client *ac,
struct asm_vorbis_cfg *cfg, int stream_id)
{
struct asm_vorbis_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s :session[%d] bit_stream_fmt[%d] stream_id[%d]\n",
__func__, ac->session, cfg->bit_stream_fmt, stream_id);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.bit_stream_fmt = cfg->bit_stream_fmt;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s :Comamnd media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_stream_media_format_block_vorbis);
/**
* q6asm_media_format_block_ape -
* command to set mediafmt block for APE on ASM stream
*
* @ac: Audio client handle
* @cfg: APE config
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_media_format_block_ape(struct audio_client *ac,
struct asm_ape_cfg *cfg, int stream_id)
{
struct asm_ape_fmt_blk_v2 fmt;
int rc = 0;
pr_debug("%s :session[%d]rate[%d]ch[%d]\n", __func__,
ac->session, cfg->sample_rate, cfg->num_channels);
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
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 = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s :Comamnd media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_ape);
/*
* q6asm_media_format_block_dsd- Sends DSD Decoder
* configuration parameters
*
* @ac: Client session handle
* @cfg: DSD Media Format Configuration.
* @stream_id: stream id of stream to be associated with this session
*
* Return 0 on success or negative error code on failure
*/
int q6asm_media_format_block_dsd(struct audio_client *ac,
struct asm_dsd_cfg *cfg, int stream_id)
{
struct asm_dsd_fmt_blk_v2 fmt;
int rc;
pr_debug("%s: session[%d] data_rate[%d] ch[%d]\n", __func__,
ac->session, cfg->dsd_data_rate, cfg->num_channels);
memset(&fmt, 0, sizeof(fmt));
q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
sizeof(fmt.fmtblk);
fmt.num_version = cfg->num_version;
fmt.is_bitwise_big_endian = cfg->is_bitwise_big_endian;
fmt.dsd_channel_block_size = cfg->dsd_channel_block_size;
fmt.num_channels = cfg->num_channels;
fmt.dsd_data_rate = cfg->dsd_data_rate;
atomic_set(&ac->cmd_state, -1);
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
if (rc < 0) {
pr_err("%s: Command DSD media format update failed, err: %d\n",
__func__, rc);
goto done;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for DSD FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto done;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto done;
}
return 0;
done:
return rc;
}
EXPORT_SYMBOL(q6asm_media_format_block_dsd);
/**
* q6asm_stream_media_format_block_aptx_dec -
* command to set mediafmt block for APTX dec on ASM stream
*
* @ac: Audio client handle
* @srate: sample rate
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_media_format_block_aptx_dec(struct audio_client *ac,
uint32_t srate, int stream_id)
{
struct asm_aptx_dec_fmt_blk_v2 aptx_fmt;
int rc = 0;
if (!ac->session) {
pr_err("%s: ac session invalid\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s :session[%d] rate[%d] stream_id[%d]\n",
__func__, ac->session, srate, stream_id);
q6asm_stream_add_hdr(ac, &aptx_fmt.hdr, sizeof(aptx_fmt), TRUE,
stream_id);
atomic_set(&ac->cmd_state, -1);
aptx_fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
aptx_fmt.fmtblk.fmt_blk_size = sizeof(aptx_fmt) - sizeof(aptx_fmt.hdr) -
sizeof(aptx_fmt.fmtblk);
aptx_fmt.sample_rate = srate;
rc = apr_send_pkt(ac->apr, (uint32_t *) &aptx_fmt);
if (rc < 0) {
pr_err("%s :Comamnd media format update failed %d\n",
__func__, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
rc = 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_stream_media_format_block_aptx_dec);
static int __q6asm_ds1_set_endp_params(struct audio_client *ac, int param_id,
int param_value, int stream_id)
{
struct asm_dec_ddp_endp_param_v2 ddp_cfg;
int rc = 0;
pr_debug("%s: session[%d] stream[%d],param_id[%d]param_value[%d]",
__func__, ac->session, stream_id, param_id, param_value);
q6asm_stream_add_hdr(ac, &ddp_cfg.hdr, sizeof(ddp_cfg), TRUE,
stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&ddp_cfg.hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
ddp_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
ddp_cfg.encdec.param_id = param_id;
ddp_cfg.encdec.param_size = sizeof(struct asm_dec_ddp_endp_param_v2) -
(sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_encdec_param));
ddp_cfg.endp_param_value = param_value;
rc = apr_send_pkt(ac->apr, (uint32_t *) &ddp_cfg);
if (rc < 0) {
pr_err("%s: Command opcode[0x%x] failed %d\n",
__func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout opcode[0x%x]\n", __func__,
ddp_cfg.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_ds1_set_endp_params -
* command to set DS1 params for ASM
*
* @ac: Audio client handle
* @param_id: param id
* @param_value: value of param
*
* Returns 0 on success or error on failure
*/
int q6asm_ds1_set_endp_params(struct audio_client *ac,
int param_id, int param_value)
{
return __q6asm_ds1_set_endp_params(ac, param_id, param_value,
ac->stream_id);
}
/**
* q6asm_ds1_set_stream_endp_params -
* command to set DS1 params for ASM stream
*
* @ac: Audio client handle
* @param_id: param id
* @param_value: value of param
* @stream_id: stream ID info
*
* Returns 0 on success or error on failure
*/
int q6asm_ds1_set_stream_endp_params(struct audio_client *ac,
int param_id, int param_value,
int stream_id)
{
return __q6asm_ds1_set_endp_params(ac, param_id, param_value,
stream_id);
}
EXPORT_SYMBOL(q6asm_ds1_set_stream_endp_params);
/**
* q6asm_memory_map -
* command to send memory map for ASM
*
* @ac: Audio client handle
* @buf_add: buffer address to map
* @dir: RX or TX session
* @bufsz: size of each buffer
* @bufcnt: buffer count
*
* Returns 0 on success or error on failure
*/
int q6asm_memory_map(struct audio_client *ac, phys_addr_t buf_add, int dir,
uint32_t bufsz, uint32_t bufcnt)
{
struct avs_cmd_shared_mem_map_regions *mmap_regions = NULL;
struct avs_shared_map_region_payload *mregions = NULL;
struct audio_port_data *port = NULL;
void *mmap_region_cmd = NULL;
void *payload = NULL;
struct asm_buffer_node *buffer_node = NULL;
int rc = 0;
int cmd_size = 0;
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->mmap_apr == NULL) {
pr_err("%s: mmap APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: Session[%d]\n", __func__, ac->session);
buffer_node = kmalloc(sizeof(struct asm_buffer_node), GFP_KERNEL);
if (!buffer_node)
return -ENOMEM;
cmd_size = sizeof(struct avs_cmd_shared_mem_map_regions)
+ sizeof(struct avs_shared_map_region_payload) * bufcnt;
mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
if (mmap_region_cmd == NULL) {
rc = -EINVAL;
kfree(buffer_node);
return rc;
}
mmap_regions = (struct avs_cmd_shared_mem_map_regions *)
mmap_region_cmd;
q6asm_add_mmaphdr(ac, &mmap_regions->hdr, cmd_size, dir);
atomic_set(&ac->mem_state, -1);
mmap_regions->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
mmap_regions->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
mmap_regions->num_regions = bufcnt & 0x00ff;
mmap_regions->property_flag = 0x00;
payload = ((u8 *) mmap_region_cmd +
sizeof(struct avs_cmd_shared_mem_map_regions));
mregions = (struct avs_shared_map_region_payload *)payload;
ac->port[dir].tmp_hdl = 0;
port = &ac->port[dir];
pr_debug("%s: buf_add 0x%pK, bufsz: %d\n", __func__,
&buf_add, bufsz);
mregions->shm_addr_lsw = lower_32_bits(buf_add);
mregions->shm_addr_msw = msm_audio_populate_upper_32_bits(buf_add);
mregions->mem_size_bytes = bufsz;
++mregions;
rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) mmap_region_cmd);
if (rc < 0) {
pr_err("%s: mmap op[0x%x]rc[%d]\n", __func__,
mmap_regions->hdr.opcode, rc);
rc = -EINVAL;
kfree(buffer_node);
goto fail_cmd;
}
rc = wait_event_timeout(ac->mem_wait,
(atomic_read(&ac->mem_state) >= 0 &&
ac->port[dir].tmp_hdl), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for memory_map\n", __func__);
rc = -ETIMEDOUT;
kfree(buffer_node);
goto fail_cmd;
}
if (atomic_read(&ac->mem_state) > 0) {
pr_err("%s: DSP returned error[%s] for memory_map\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->mem_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->mem_state));
kfree(buffer_node);
goto fail_cmd;
}
buffer_node->buf_phys_addr = buf_add;
buffer_node->mmap_hdl = ac->port[dir].tmp_hdl;
list_add_tail(&buffer_node->list, &ac->port[dir].mem_map_handle);
ac->port[dir].tmp_hdl = 0;
rc = 0;
fail_cmd:
kfree(mmap_region_cmd);
return rc;
}
EXPORT_SYMBOL(q6asm_memory_map);
/**
* q6asm_memory_unmap -
* command to send memory unmap for ASM
*
* @ac: Audio client handle
* @buf_add: buffer address to unmap
* @dir: RX or TX session
*
* Returns 0 on success or error on failure
*/
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 asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
int rc = 0;
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (this_mmap.apr == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: Session[%d]\n", __func__, ac->session);
q6asm_add_mmaphdr(ac, &mem_unmap.hdr,
sizeof(struct avs_cmd_shared_mem_unmap_regions),
dir);
atomic_set(&ac->mem_state, -1);
mem_unmap.hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
mem_unmap.mem_map_handle = 0;
list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == buf_add) {
pr_debug("%s: Found the element\n", __func__);
mem_unmap.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
pr_debug("%s: mem_unmap-mem_map_handle: 0x%x\n",
__func__, mem_unmap.mem_map_handle);
if (mem_unmap.mem_map_handle == 0) {
pr_err("%s: Do not send null mem handle to DSP\n", __func__);
rc = 0;
goto fail_cmd;
}
rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) &mem_unmap);
if (rc < 0) {
pr_err("%s: mem_unmap op[0x%x]rc[%d]\n", __func__,
mem_unmap.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->mem_wait,
(atomic_read(&ac->mem_state) >= 0), 5 * HZ);
if (!rc) {
pr_err("%s: timeout. waited for memory_unmap of handle 0x%x\n",
__func__, mem_unmap.mem_map_handle);
rc = -ETIMEDOUT;
goto fail_cmd;
} else if (atomic_read(&ac->mem_state) > 0) {
pr_err("%s DSP returned error [%s] map handle 0x%x\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->mem_state)),
mem_unmap.mem_map_handle);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->mem_state));
goto fail_cmd;
} else if (atomic_read(&ac->unmap_cb_success) == 0) {
pr_err("%s: Error in mem unmap callback of handle 0x%x\n",
__func__, mem_unmap.mem_map_handle);
rc = -EINVAL;
goto fail_cmd;
}
rc = 0;
fail_cmd:
list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == buf_add) {
list_del(&buf_node->list);
kfree(buf_node);
break;
}
}
return rc;
}
EXPORT_SYMBOL(q6asm_memory_unmap);
/**
* q6asm_memory_map_regions -
* command to send memory map regions for ASM
*
* @ac: Audio client handle
* @dir: RX or TX session
* @bufsz: size of each buffer
* @bufcnt: buffer count
* @is_contiguous: alloc contiguous mem or not
*
* Returns 0 on success or error on failure
*/
static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
uint32_t bufsz, uint32_t bufcnt,
bool is_contiguous)
{
struct avs_cmd_shared_mem_map_regions *mmap_regions = NULL;
struct avs_shared_map_region_payload *mregions = NULL;
struct audio_port_data *port = NULL;
struct audio_buffer *ab = NULL;
void *mmap_region_cmd = NULL;
void *payload = NULL;
struct asm_buffer_node *buffer_node = NULL;
int rc = 0;
int i = 0;
uint32_t cmd_size = 0;
uint32_t bufcnt_t;
uint32_t bufsz_t;
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->mmap_apr == NULL) {
pr_err("%s: mmap APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: Session[%d]\n", __func__, ac->session);
bufcnt_t = (is_contiguous) ? 1 : bufcnt;
bufsz_t = (is_contiguous) ? (bufsz * bufcnt) : bufsz;
if (is_contiguous) {
/* The size to memory map should be multiple of 4K bytes */
bufsz_t = PAGE_ALIGN(bufsz_t);
}
if (bufcnt_t > (UINT_MAX
- sizeof(struct avs_cmd_shared_mem_map_regions))
/ sizeof(struct avs_shared_map_region_payload)) {
pr_err("%s: Unsigned Integer Overflow. bufcnt_t = %u\n",
__func__, bufcnt_t);
return -EINVAL;
}
cmd_size = sizeof(struct avs_cmd_shared_mem_map_regions)
+ (sizeof(struct avs_shared_map_region_payload)
* bufcnt_t);
if (bufcnt > (UINT_MAX / sizeof(struct asm_buffer_node))) {
pr_err("%s: Unsigned Integer Overflow. bufcnt = %u\n",
__func__, bufcnt);
return -EINVAL;
}
buffer_node = kzalloc(sizeof(struct asm_buffer_node) * bufcnt,
GFP_KERNEL);
if (!buffer_node)
return -ENOMEM;
mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
if (mmap_region_cmd == NULL) {
rc = -EINVAL;
kfree(buffer_node);
return rc;
}
mmap_regions = (struct avs_cmd_shared_mem_map_regions *)
mmap_region_cmd;
q6asm_add_mmaphdr(ac, &mmap_regions->hdr, cmd_size, dir);
atomic_set(&ac->mem_state, -1);
pr_debug("%s: mmap_region=0x%pK token=0x%x\n", __func__,
mmap_regions, ((ac->session << 8) | dir));
mmap_regions->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
mmap_regions->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
mmap_regions->num_regions = bufcnt_t; /*bufcnt & 0x00ff; */
mmap_regions->property_flag = 0x00;
pr_debug("%s: map_regions->nregions = %d\n", __func__,
mmap_regions->num_regions);
payload = ((u8 *) mmap_region_cmd +
sizeof(struct avs_cmd_shared_mem_map_regions));
mregions = (struct avs_shared_map_region_payload *)payload;
ac->port[dir].tmp_hdl = 0;
port = &ac->port[dir];
for (i = 0; i < bufcnt_t; i++) {
ab = &port->buf[i];
mregions->shm_addr_lsw = lower_32_bits(ab->phys);
mregions->shm_addr_msw =
msm_audio_populate_upper_32_bits(ab->phys);
mregions->mem_size_bytes = bufsz_t;
++mregions;
}
rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) mmap_region_cmd);
if (rc < 0) {
pr_err("%s: mmap_regions op[0x%x]rc[%d]\n", __func__,
mmap_regions->hdr.opcode, rc);
rc = -EINVAL;
kfree(buffer_node);
goto fail_cmd;
}
rc = wait_event_timeout(ac->mem_wait,
(atomic_read(&ac->mem_state) >= 0 &&
ac->port[dir].tmp_hdl), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for memory_map\n", __func__);
rc = -ETIMEDOUT;
kfree(buffer_node);
goto fail_cmd;
}
if (atomic_read(&ac->mem_state) > 0) {
pr_err("%s DSP returned error for memory_map [%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->mem_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->mem_state));
kfree(buffer_node);
goto fail_cmd;
}
mutex_lock(&ac->cmd_lock);
for (i = 0; i < bufcnt; i++) {
ab = &port->buf[i];
buffer_node[i].buf_phys_addr = ab->phys;
buffer_node[i].mmap_hdl = ac->port[dir].tmp_hdl;
list_add_tail(&buffer_node[i].list,
&ac->port[dir].mem_map_handle);
pr_debug("%s: i=%d, bufadd[i] = 0x%pK, maphdl[i] = 0x%x\n",
__func__, i, &buffer_node[i].buf_phys_addr,
buffer_node[i].mmap_hdl);
}
ac->port[dir].tmp_hdl = 0;
mutex_unlock(&ac->cmd_lock);
rc = 0;
fail_cmd:
kfree(mmap_region_cmd);
return rc;
}
EXPORT_SYMBOL(q6asm_memory_map_regions);
/**
* q6asm_memory_unmap_regions -
* command to send memory unmap regions for ASM
*
* @ac: Audio client handle
* @dir: RX or TX session
*
* Returns 0 on success or error on failure
*/
static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir)
{
struct avs_cmd_shared_mem_unmap_regions mem_unmap;
struct audio_port_data *port = NULL;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
phys_addr_t buf_add;
int rc = 0;
int cmd_size = 0;
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->mmap_apr == NULL) {
pr_err("%s: mmap APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: Session[%d]\n", __func__, ac->session);
cmd_size = sizeof(struct avs_cmd_shared_mem_unmap_regions);
q6asm_add_mmaphdr(ac, &mem_unmap.hdr, cmd_size, dir);
atomic_set(&ac->mem_state, -1);
port = &ac->port[dir];
buf_add = port->buf->phys;
mem_unmap.hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
mem_unmap.mem_map_handle = 0;
list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == buf_add) {
pr_debug("%s: Found the element\n", __func__);
mem_unmap.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
pr_debug("%s: mem_unmap-mem_map_handle: 0x%x\n",
__func__, mem_unmap.mem_map_handle);
if (mem_unmap.mem_map_handle == 0) {
pr_err("%s: Do not send null mem handle to DSP\n", __func__);
rc = 0;
goto fail_cmd;
}
rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) &mem_unmap);
if (rc < 0) {
pr_err("mmap_regions op[0x%x]rc[%d]\n",
mem_unmap.hdr.opcode, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->mem_wait,
(atomic_read(&ac->mem_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for memory_unmap of handle 0x%x\n",
__func__, mem_unmap.mem_map_handle);
rc = -ETIMEDOUT;
goto fail_cmd;
} else if (atomic_read(&ac->mem_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->mem_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->mem_state));
goto fail_cmd;
} else if (atomic_read(&ac->unmap_cb_success) == 0) {
pr_err("%s: Error in mem unmap callback of handle 0x%x\n",
__func__, mem_unmap.mem_map_handle);
rc = -EINVAL;
goto fail_cmd;
}
rc = 0;
fail_cmd:
list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == buf_add) {
list_del(&buf_node->list);
kfree(buf_node);
break;
}
}
return rc;
}
EXPORT_SYMBOL(q6asm_memory_unmap_regions);
int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain)
{
struct asm_volume_ctrl_multichannel_gain multi_ch_gain;
int sz = 0;
int rc = 0;
int session_id = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
memset(&multi_ch_gain, 0, sizeof(multi_ch_gain));
sz = sizeof(struct asm_volume_ctrl_multichannel_gain);
q6asm_add_hdr_async(ac, &multi_ch_gain.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
multi_ch_gain.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
multi_ch_gain.param.data_payload_addr_lsw = 0;
multi_ch_gain.param.data_payload_addr_msw = 0;
multi_ch_gain.param.mem_map_handle = 0;
multi_ch_gain.param.data_payload_size = sizeof(multi_ch_gain) -
sizeof(multi_ch_gain.hdr) - sizeof(multi_ch_gain.param);
multi_ch_gain.data.module_id = ASM_MODULE_ID_VOL_CTRL;
multi_ch_gain.data.param_id = ASM_PARAM_ID_MULTICHANNEL_GAIN;
multi_ch_gain.data.param_size = multi_ch_gain.param.data_payload_size -
sizeof(multi_ch_gain.data);
multi_ch_gain.data.reserved = 0;
multi_ch_gain.gain_data[0].channeltype = PCM_CHANNEL_FL;
multi_ch_gain.gain_data[0].gain = left_gain << 15;
multi_ch_gain.gain_data[1].channeltype = PCM_CHANNEL_FR;
multi_ch_gain.gain_data[1].gain = right_gain << 15;
multi_ch_gain.num_channels = 2;
rc = apr_send_pkt(ac->apr, (uint32_t *) &multi_ch_gain);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, multi_ch_gain.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
multi_ch_gain.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] , set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
multi_ch_gain.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
/*
* q6asm_set_multich_gain: set multiple channel gains on an ASM session
* @ac: audio client handle
* @channels: number of channels caller intends to set gains
* @gains: list of gains of audio channels
* @ch_map: list of channel mapping. Only valid if use_default is false
* @use_default: flag to indicate whether to use default mapping
*/
int q6asm_set_multich_gain(struct audio_client *ac, uint32_t channels,
uint32_t *gains, uint8_t *ch_map, bool use_default)
{
struct asm_volume_ctrl_multichannel_gain multich_gain;
int sz = 0;
int rc = 0;
int i, session_id = 0;
u8 default_chmap[VOLUME_CONTROL_MAX_CHANNELS];
if (ac == NULL) {
pr_err("%s: ac is NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
memset(&multich_gain, 0, sizeof(multich_gain));
sz = sizeof(struct asm_volume_ctrl_multichannel_gain);
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
dev_err(ac->dev, "%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (gains == NULL) {
dev_err(ac->dev, "%s: gain_list is NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (channels > VOLUME_CONTROL_MAX_CHANNELS) {
dev_err(ac->dev, "%s: Invalid channel count %d\n",
__func__, channels);
rc = -EINVAL;
goto fail_cmd;
}
if (!use_default && ch_map == NULL) {
dev_err(ac->dev, "%s: NULL channel map\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
q6asm_add_hdr_async(ac, &multich_gain.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
multich_gain.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
multich_gain.param.data_payload_addr_lsw = 0;
multich_gain.param.data_payload_addr_msw = 0;
multich_gain.param.mem_map_handle = 0;
multich_gain.param.data_payload_size = sizeof(multich_gain) -
sizeof(multich_gain.hdr) - sizeof(multich_gain.param);
multich_gain.data.module_id = ASM_MODULE_ID_VOL_CTRL;
multich_gain.data.param_id = ASM_PARAM_ID_MULTICHANNEL_GAIN;
multich_gain.data.param_size = multich_gain.param.data_payload_size -
sizeof(multich_gain.data);
multich_gain.data.reserved = 0;
if (use_default) {
rc = q6asm_map_channels(default_chmap, channels, false);
if (rc < 0)
goto fail_cmd;
for (i = 0; i < channels; i++) {
multich_gain.gain_data[i].channeltype =
default_chmap[i];
multich_gain.gain_data[i].gain = gains[i] << 15;
}
} else {
for (i = 0; i < channels; i++) {
multich_gain.gain_data[i].channeltype = ch_map[i];
multich_gain.gain_data[i].gain = gains[i] << 15;
}
}
multich_gain.num_channels = channels;
rc = apr_send_pkt(ac->apr, (uint32_t *) &multich_gain);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, multich_gain.data.param_id, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
multich_gain.data.param_id);
rc = -EINVAL;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%d] , set-params paramid[0x%x]\n",
__func__, atomic_read(&ac->cmd_state_pp),
multich_gain.data.param_id);
rc = -EINVAL;
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_set_multich_gain);
/**
* q6asm_set_mute -
* command to set mute for ASM
*
* @ac: Audio client handle
* @muteflag: mute value
*
* Returns 0 on success or error on failure
*/
int q6asm_set_mute(struct audio_client *ac, int muteflag)
{
struct asm_volume_ctrl_mute_config mute;
int sz = 0;
int rc = 0;
int session_id = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct asm_volume_ctrl_mute_config);
q6asm_add_hdr_async(ac, &mute.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
mute.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
mute.param.data_payload_addr_lsw = 0;
mute.param.data_payload_addr_msw = 0;
mute.param.mem_map_handle = 0;
mute.param.data_payload_size = sizeof(mute) -
sizeof(mute.hdr) - sizeof(mute.param);
mute.data.module_id = ASM_MODULE_ID_VOL_CTRL;
mute.data.param_id = ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG;
mute.data.param_size = mute.param.data_payload_size - sizeof(mute.data);
mute.data.reserved = 0;
mute.mute_flag = muteflag;
rc = apr_send_pkt(ac->apr, (uint32_t *) &mute);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, mute.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
mute.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
mute.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_set_mute);
static int __q6asm_set_volume(struct audio_client *ac, int volume, int instance)
{
struct asm_volume_ctrl_master_gain vol;
int sz = 0;
int rc = 0;
int module_id, session_id = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
switch (instance) {
case SOFT_VOLUME_INSTANCE_2:
module_id = ASM_MODULE_ID_VOL_CTRL2;
break;
case SOFT_VOLUME_INSTANCE_1:
default:
module_id = ASM_MODULE_ID_VOL_CTRL;
break;
}
sz = sizeof(struct asm_volume_ctrl_master_gain);
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
q6asm_add_hdr_async(ac, &vol.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
vol.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
vol.param.data_payload_addr_lsw = 0;
vol.param.data_payload_addr_msw = 0;
vol.param.mem_map_handle = 0;
vol.param.data_payload_size = sizeof(vol) -
sizeof(vol.hdr) - sizeof(vol.param);
vol.data.module_id = module_id;
vol.data.param_id = ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN;
vol.data.param_size = vol.param.data_payload_size - sizeof(vol.data);
vol.data.reserved = 0;
vol.master_gain = volume;
rc = apr_send_pkt(ac->apr, (uint32_t *) &vol);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, vol.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
vol.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
vol.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
/**
* q6asm_set_volume -
* command to set volume for ASM
*
* @ac: Audio client handle
* @volume: volume level
*
* Returns 0 on success or error on failure
*/
int q6asm_set_volume(struct audio_client *ac, int volume)
{
return __q6asm_set_volume(ac, volume, SOFT_VOLUME_INSTANCE_1);
}
EXPORT_SYMBOL(q6asm_set_volume);
int q6asm_set_volume_v2(struct audio_client *ac, int volume, int instance)
{
return __q6asm_set_volume(ac, volume, instance);
}
/**
* q6asm_set_aptx_dec_bt_addr -
* command to aptx decoder BT addr for ASM
*
* @ac: Audio client handle
* @cfg: APTX decoder bt addr config
*
* Returns 0 on success or error on failure
*/
int q6asm_set_aptx_dec_bt_addr(struct audio_client *ac,
struct aptx_dec_bt_addr_cfg *cfg)
{
struct aptx_dec_bt_dev_addr paylod;
int sz = 0;
int rc = 0;
pr_debug("%s: BT addr nap %d, uap %d, lap %d\n", __func__, cfg->nap,
cfg->uap, cfg->lap);
if (ac == NULL) {
pr_err("%s: AC handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct aptx_dec_bt_dev_addr);
q6asm_add_hdr_async(ac, &paylod.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
paylod.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
paylod.encdec.param_id = APTX_DECODER_BT_ADDRESS;
paylod.encdec.param_size = sz - sizeof(paylod.hdr)
- sizeof(paylod.encdec);
paylod.bt_addr_cfg.lap = cfg->lap;
paylod.bt_addr_cfg.uap = cfg->uap;
paylod.bt_addr_cfg.nap = cfg->nap;
rc = apr_send_pkt(ac->apr, (uint32_t *) &paylod);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, paylod.encdec.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
paylod.encdec.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)),
paylod.encdec.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
pr_debug("%s: set BT addr is success\n", __func__);
rc = 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_set_aptx_dec_bt_addr);
/**
* q6asm_send_ion_fd -
* command to send ION memory map for ASM
*
* @ac: Audio client handle
* @fd: ION file desc
*
* Returns 0 on success or error on failure
*/
int q6asm_send_ion_fd(struct audio_client *ac, int fd)
{
struct ion_client *client;
struct ion_handle *handle;
ion_phys_addr_t paddr;
size_t pa_len = 0;
void *vaddr;
int ret;
int sz = 0;
struct avs_rtic_shared_mem_addr shm;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
ret = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
ret = -EINVAL;
goto fail_cmd;
}
ret = msm_audio_ion_import("audio_mem_client",
&client,
&handle,
fd,
NULL,
0,
&paddr,
&pa_len,
&vaddr);
if (ret) {
pr_err("%s: audio ION import failed, rc = %d\n",
__func__, ret);
ret = -ENOMEM;
goto fail_cmd;
}
/* get payload length */
sz = sizeof(struct avs_rtic_shared_mem_addr);
q6asm_add_hdr_async(ac, &shm.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
shm.shm_buf_addr_lsw = lower_32_bits(paddr);
shm.shm_buf_addr_msw = msm_audio_populate_upper_32_bits(paddr);
shm.buf_size = pa_len;
shm.shm_buf_num_regions = 1;
shm.shm_buf_mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
shm.shm_buf_flag = 0x00;
shm.encdec.param_id = AVS_PARAM_ID_RTIC_SHARED_MEMORY_ADDR;
shm.encdec.param_size = sizeof(struct avs_rtic_shared_mem_addr) -
sizeof(struct apr_hdr) -
sizeof(struct asm_stream_cmd_set_encdec_param_v2);
shm.encdec.service_id = OUT;
shm.encdec.reserved = 0;
shm.map_region.shm_addr_lsw = shm.shm_buf_addr_lsw;
shm.map_region.shm_addr_msw = shm.shm_buf_addr_msw;
shm.map_region.mem_size_bytes = pa_len;
shm.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM_V2;
ret = apr_send_pkt(ac->apr, (uint32_t *) &shm);
if (ret < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, shm.encdec.param_id, ret);
ret = -EINVAL;
goto fail_cmd;
}
ret = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 1*HZ);
if (!ret) {
pr_err("%s: timeout, shm.encdec paramid[0x%x]\n", __func__,
shm.encdec.param_id);
ret = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s] shm.encdec paramid[0x%x]\n",
__func__,
adsp_err_get_err_str(atomic_read(&ac->cmd_state)),
shm.encdec.param_id);
ret = adsp_err_get_lnx_err_code(atomic_read(&ac->cmd_state));
goto fail_cmd;
}
ret = 0;
fail_cmd:
return ret;
}
EXPORT_SYMBOL(q6asm_send_ion_fd);
/**
* q6asm_send_rtic_event_ack -
* command to send RTIC event ack
*
* @ac: Audio client handle
* @param: params for event ack
* @params_length: length of params
*
* Returns 0 on success or error on failure
*/
int q6asm_send_rtic_event_ack(struct audio_client *ac,
void *param, uint32_t params_length)
{
char *asm_params = NULL;
int sz, rc;
struct avs_param_rtic_event_ack ack;
if (!param || !ac) {
pr_err("%s: %s is NULL\n", __func__,
(!param) ? "param" : "ac");
rc = -EINVAL;
goto done;
}
sz = sizeof(struct avs_param_rtic_event_ack) + params_length;
asm_params = kzalloc(sz, GFP_KERNEL);
if (!asm_params) {
rc = -ENOMEM;
goto done;
}
q6asm_add_hdr_async(ac, &ack.hdr,
sizeof(struct avs_param_rtic_event_ack) +
params_length, TRUE);
atomic_set(&ac->cmd_state, -1);
ack.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM_V2;
ack.encdec.param_id = AVS_PARAM_ID_RTIC_EVENT_ACK;
ack.encdec.param_size = params_length;
ack.encdec.reserved = 0;
ack.encdec.service_id = OUT;
memcpy(asm_params, &ack, sizeof(struct avs_param_rtic_event_ack));
memcpy(asm_params + sizeof(struct avs_param_rtic_event_ack),
param, params_length);
rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params);
if (rc < 0) {
pr_err("%s: apr pkt failed for rtic event ack\n", __func__);
rc = -EINVAL;
goto fail_send_param;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 1 * HZ);
if (!rc) {
pr_err("%s: timeout for rtic event ack cmd\n", __func__);
rc = -ETIMEDOUT;
goto fail_send_param;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s] for rtic event ack cmd\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_send_param;
}
rc = 0;
fail_send_param:
kfree(asm_params);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_send_rtic_event_ack);
/**
* q6asm_set_softpause -
* command to set pause for ASM
*
* @ac: Audio client handle
* @pause_param: params for pause
*
* Returns 0 on success or error on failure
*/
int q6asm_set_softpause(struct audio_client *ac,
struct asm_softpause_params *pause_param)
{
struct asm_soft_pause_params softpause;
int sz = 0;
int rc = 0;
int session_id = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct asm_soft_pause_params);
q6asm_add_hdr_async(ac, &softpause.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
softpause.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
softpause.param.data_payload_addr_lsw = 0;
softpause.param.data_payload_addr_msw = 0;
softpause.param.mem_map_handle = 0;
softpause.param.data_payload_size = sizeof(softpause) -
sizeof(softpause.hdr) - sizeof(softpause.param);
softpause.data.module_id = ASM_MODULE_ID_VOL_CTRL;
softpause.data.param_id = ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS;
softpause.data.param_size = softpause.param.data_payload_size -
sizeof(softpause.data);
softpause.data.reserved = 0;
softpause.enable_flag = pause_param->enable;
softpause.period = pause_param->period;
softpause.step = pause_param->step;
softpause.ramping_curve = pause_param->rampingcurve;
rc = apr_send_pkt(ac->apr, (uint32_t *) &softpause);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, softpause.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
softpause.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
softpause.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_set_softpause);
static int __q6asm_set_softvolume(struct audio_client *ac,
struct asm_softvolume_params *softvol_param,
int instance)
{
struct asm_soft_step_volume_params softvol;
int sz = 0;
int rc = 0;
int module_id, session_id;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
switch (instance) {
case SOFT_VOLUME_INSTANCE_2:
module_id = ASM_MODULE_ID_VOL_CTRL2;
break;
case SOFT_VOLUME_INSTANCE_1:
default:
module_id = ASM_MODULE_ID_VOL_CTRL;
break;
}
sz = sizeof(struct asm_soft_step_volume_params);
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
q6asm_add_hdr_async(ac, &softvol.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
softvol.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
softvol.param.data_payload_addr_lsw = 0;
softvol.param.data_payload_addr_msw = 0;
softvol.param.mem_map_handle = 0;
softvol.param.data_payload_size = sizeof(softvol) -
sizeof(softvol.hdr) - sizeof(softvol.param);
softvol.data.module_id = module_id;
softvol.data.param_id = ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS;
softvol.data.param_size = softvol.param.data_payload_size -
sizeof(softvol.data);
softvol.data.reserved = 0;
softvol.period = softvol_param->period;
softvol.step = softvol_param->step;
softvol.ramping_curve = softvol_param->rampingcurve;
rc = apr_send_pkt(ac->apr, (uint32_t *) &softvol);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, softvol.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
softvol.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
softvol.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
/**
* q6asm_set_softvolume -
* command to set softvolume for ASM
*
* @ac: Audio client handle
* @softvol_param: params for softvol
*
* Returns 0 on success or error on failure
*/
int q6asm_set_softvolume(struct audio_client *ac,
struct asm_softvolume_params *softvol_param)
{
return __q6asm_set_softvolume(ac, softvol_param,
SOFT_VOLUME_INSTANCE_1);
}
EXPORT_SYMBOL(q6asm_set_softvolume);
/**
* q6asm_set_softvolume_v2 -
* command to set softvolume V2 for ASM
*
* @ac: Audio client handle
* @softvol_param: params for softvol
* @instance: instance to apply softvol
*
* Returns 0 on success or error on failure
*/
int q6asm_set_softvolume_v2(struct audio_client *ac,
struct asm_softvolume_params *softvol_param,
int instance)
{
return __q6asm_set_softvolume(ac, softvol_param, instance);
}
EXPORT_SYMBOL(q6asm_set_softvolume_v2);
/**
* q6asm_equalizer -
* command to set equalizer for ASM
*
* @ac: Audio client handle
* @eq_p: Equalizer params
*
* Returns 0 on success or error on failure
*/
int q6asm_equalizer(struct audio_client *ac, void *eq_p)
{
struct asm_eq_params eq;
struct msm_audio_eq_stream_config *eq_params = NULL;
int i = 0;
int sz = 0;
int rc = 0;
int session_id = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
rc = -EINVAL;
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id) {
rc = -EINVAL;
goto done;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (eq_p == NULL) {
pr_err("%s: [%d]: Invalid Eq param\n", __func__, ac->session);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct asm_eq_params);
eq_params = (struct msm_audio_eq_stream_config *) eq_p;
q6asm_add_hdr(ac, &eq.hdr, sz, TRUE);
atomic_set(&ac->cmd_state_pp, -1);
eq.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
eq.param.data_payload_addr_lsw = 0;
eq.param.data_payload_addr_msw = 0;
eq.param.mem_map_handle = 0;
eq.param.data_payload_size = sizeof(eq) -
sizeof(eq.hdr) - sizeof(eq.param);
eq.data.module_id = ASM_MODULE_ID_EQUALIZER;
eq.data.param_id = ASM_PARAM_ID_EQUALIZER_PARAMETERS;
eq.data.param_size = eq.param.data_payload_size - sizeof(eq.data);
eq.enable_flag = eq_params->enable;
eq.num_bands = eq_params->num_bands;
pr_debug("%s: enable:%d numbands:%d\n", __func__, eq_params->enable,
eq_params->num_bands);
for (i = 0; i < eq_params->num_bands; i++) {
eq.eq_bands[i].band_idx =
eq_params->eq_bands[i].band_idx;
eq.eq_bands[i].filterype =
eq_params->eq_bands[i].filter_type;
eq.eq_bands[i].center_freq_hz =
eq_params->eq_bands[i].center_freq_hz;
eq.eq_bands[i].filter_gain =
eq_params->eq_bands[i].filter_gain;
eq.eq_bands[i].q_factor =
eq_params->eq_bands[i].q_factor;
pr_debug("%s: filter_type:%u bandnum:%d\n", __func__,
eq_params->eq_bands[i].filter_type, i);
pr_debug("%s: center_freq_hz:%u bandnum:%d\n", __func__,
eq_params->eq_bands[i].center_freq_hz, i);
pr_debug("%s: filter_gain:%d bandnum:%d\n", __func__,
eq_params->eq_bands[i].filter_gain, i);
pr_debug("%s: q_factor:%d bandnum:%d\n", __func__,
eq_params->eq_bands[i].q_factor, i);
}
rc = apr_send_pkt(ac->apr, (uint32_t *)&eq);
if (rc < 0) {
pr_err("%s: set-params send failed paramid[0x%x] rc %d\n",
__func__, eq.data.param_id, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
eq.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)),
eq.data.param_id);
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_cmd;
}
rc = 0;
fail_cmd:
mutex_unlock(&session[session_id].mutex_lock_per_session);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_equalizer);
static int __q6asm_read(struct audio_client *ac, bool is_custom_len_reqd,
int len)
{
struct asm_data_cmd_read_v2 read;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
struct audio_buffer *ab;
int dsp_buf;
struct audio_port_data *port;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[OUT];
q6asm_add_hdr(ac, &read.hdr, sizeof(read), FALSE);
mutex_lock(&port->lock);
dsp_buf = port->dsp_buf;
if (port->buf == NULL) {
pr_err("%s: buf is NULL\n", __func__);
mutex_unlock(&port->lock);
return -EINVAL;
}
ab = &port->buf[dsp_buf];
dev_vdbg(ac->dev, "%s: session[%d]dsp-buf[%d][%pK]cpu_buf[%d][%pK]\n",
__func__,
ac->session,
dsp_buf,
port->buf[dsp_buf].data,
port->cpu_buf,
&port->buf[port->cpu_buf].phys);
read.hdr.opcode = ASM_DATA_CMD_READ_V2;
read.buf_addr_lsw = lower_32_bits(ab->phys);
read.buf_addr_msw = msm_audio_populate_upper_32_bits(ab->phys);
list_for_each_safe(ptr, next, &ac->port[OUT].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == ab->phys) {
read.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
dev_vdbg(ac->dev, "memory_map handle in q6asm_read: [%0x]:",
read.mem_map_handle);
read.buf_size = is_custom_len_reqd ? len : ab->size;
read.seq_id = port->dsp_buf;
q6asm_update_token(&read.hdr.token,
0, /* Session ID is NA */
0, /* Stream ID is NA */
port->dsp_buf,
0, /* Direction flag is NA */
WAIT_CMD);
port->dsp_buf = q6asm_get_next_buf(ac, port->dsp_buf,
port->max_buf_cnt);
mutex_unlock(&port->lock);
dev_vdbg(ac->dev, "%s: buf add[%pK] token[0x%x] uid[%d]\n",
__func__, &ab->phys, read.hdr.token,
read.seq_id);
rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
if (rc < 0) {
pr_err("%s: read op[0x%x]rc[%d]\n",
__func__, read.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
}
fail_cmd:
return -EINVAL;
}
/**
* q6asm_read -
* command to read buffer data from DSP
*
* @ac: Audio client handle
*
* Returns 0 on success or error on failure
*/
int q6asm_read(struct audio_client *ac)
{
return __q6asm_read(ac, false/*is_custom_len_reqd*/, 0);
}
EXPORT_SYMBOL(q6asm_read);
/**
* q6asm_read_v2 -
* command to read buffer data from DSP
*
* @ac: Audio client handle
* @len: buffer size to read
*
* Returns 0 on success or error on failure
*/
int q6asm_read_v2(struct audio_client *ac, uint32_t len)
{
return __q6asm_read(ac, true /*is_custom_len_reqd*/, len);
}
EXPORT_SYMBOL(q6asm_read_v2);
/**
* q6asm_read_nolock -
* command to read buffer data from DSP
* with no wait for ack.
*
* @ac: Audio client handle
*
* Returns 0 on success or error on failure
*/
int q6asm_read_nolock(struct audio_client *ac)
{
struct asm_data_cmd_read_v2 read;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
struct audio_buffer *ab;
int dsp_buf;
struct audio_port_data *port;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[OUT];
q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
dsp_buf = port->dsp_buf;
ab = &port->buf[dsp_buf];
dev_vdbg(ac->dev, "%s: session[%d]dsp-buf[%d][%pK]cpu_buf[%d][%pK]\n",
__func__,
ac->session,
dsp_buf,
port->buf[dsp_buf].data,
port->cpu_buf,
&port->buf[port->cpu_buf].phys);
read.hdr.opcode = ASM_DATA_CMD_READ_V2;
read.buf_addr_lsw = lower_32_bits(ab->phys);
read.buf_addr_msw = msm_audio_populate_upper_32_bits(ab->phys);
read.buf_size = ab->size;
read.seq_id = port->dsp_buf;
q6asm_update_token(&read.hdr.token,
0, /* Session ID is NA */
0, /* Stream ID is NA */
port->dsp_buf,
0, /* Direction flag is NA */
WAIT_CMD);
list_for_each_safe(ptr, next, &ac->port[OUT].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == ab->phys) {
read.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
port->dsp_buf = q6asm_get_next_buf(ac, port->dsp_buf,
port->max_buf_cnt);
dev_vdbg(ac->dev, "%s: buf add[%pK] token[0x%x] uid[%d]\n",
__func__, &ab->phys, read.hdr.token,
read.seq_id);
rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
if (rc < 0) {
pr_err("%s: read op[0x%x]rc[%d]\n",
__func__, read.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
}
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_read_nolock);
/**
* q6asm_async_write -
* command to write DSP buffer
*
* @ac: Audio client handle
* @param: params for async write
*
* Returns 0 on success or error on failure
*/
int q6asm_async_write(struct audio_client *ac,
struct audio_aio_write_param *param)
{
int rc = 0;
struct asm_data_cmd_write_v2 write;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
struct audio_buffer *ab;
struct audio_port_data *port;
phys_addr_t lbuf_phys_addr;
u32 liomode;
u32 io_compressed;
u32 io_compressed_stream;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
q6asm_stream_add_hdr_async(
ac, &write.hdr, sizeof(write), TRUE, ac->stream_id);
port = &ac->port[IN];
ab = &port->buf[port->dsp_buf];
/* Pass session id as token for AIO scheme */
write.hdr.token = param->uid;
write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
write.buf_addr_lsw = lower_32_bits(param->paddr);
write.buf_addr_msw = msm_audio_populate_upper_32_bits(param->paddr);
write.buf_size = param->len;
write.timestamp_msw = param->msw_ts;
write.timestamp_lsw = param->lsw_ts;
liomode = (ASYNC_IO_MODE | NT_MODE);
io_compressed = (ASYNC_IO_MODE | COMPRESSED_IO);
io_compressed_stream = (ASYNC_IO_MODE | COMPRESSED_STREAM_IO);
if (ac->io_mode == liomode)
lbuf_phys_addr = (param->paddr - 32);
else if (ac->io_mode == io_compressed ||
ac->io_mode == io_compressed_stream)
lbuf_phys_addr = (param->paddr - param->metadata_len);
else {
if (param->flags & SET_TIMESTAMP)
lbuf_phys_addr = param->paddr -
sizeof(struct snd_codec_metadata);
else
lbuf_phys_addr = param->paddr;
}
dev_vdbg(ac->dev, "%s: token[0x%x], buf_addr[%pK], buf_size[0x%x], ts_msw[0x%x], ts_lsw[0x%x], lbuf_phys_addr: 0x[%pK]\n",
__func__,
write.hdr.token, &param->paddr,
write.buf_size, write.timestamp_msw,
write.timestamp_lsw, &lbuf_phys_addr);
/* Use 0xFF00 for disabling timestamps */
if (param->flags == 0xFF00)
write.flags = (0x00000000 | (param->flags & 0x800000FF));
else
write.flags = (0x80000000 | param->flags);
write.flags |= param->last_buffer << ASM_SHIFT_LAST_BUFFER_FLAG;
write.seq_id = param->uid;
list_for_each_safe(ptr, next, &ac->port[IN].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == lbuf_phys_addr) {
write.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
if (rc < 0) {
pr_err("%s: write op[0x%x]rc[%d]\n", __func__,
write.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_async_write);
/**
* q6asm_async_read -
* command to read DSP buffer
*
* @ac: Audio client handle
* @param: params for async read
*
* Returns 0 on success or error on failure
*/
int q6asm_async_read(struct audio_client *ac,
struct audio_aio_read_param *param)
{
int rc = 0;
struct asm_data_cmd_read_v2 read;
struct asm_buffer_node *buf_node = NULL;
struct list_head *ptr, *next;
phys_addr_t lbuf_phys_addr;
u32 liomode;
u32 io_compressed;
int dir = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
/* Pass session id as token for AIO scheme */
read.hdr.token = param->uid;
read.hdr.opcode = ASM_DATA_CMD_READ_V2;
read.buf_addr_lsw = lower_32_bits(param->paddr);
read.buf_addr_msw = msm_audio_populate_upper_32_bits(param->paddr);
read.buf_size = param->len;
read.seq_id = param->uid;
liomode = (NT_MODE | ASYNC_IO_MODE);
io_compressed = (ASYNC_IO_MODE | COMPRESSED_IO);
if (ac->io_mode == liomode) {
lbuf_phys_addr = (param->paddr - 32);
/*legacy wma driver case*/
dir = IN;
} else if (ac->io_mode == io_compressed) {
lbuf_phys_addr = (param->paddr - 64);
dir = OUT;
} else {
if (param->flags & COMPRESSED_TIMESTAMP_FLAG)
lbuf_phys_addr = param->paddr -
sizeof(struct snd_codec_metadata);
else
lbuf_phys_addr = param->paddr;
dir = OUT;
}
list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
buf_node = list_entry(ptr, struct asm_buffer_node,
list);
if (buf_node->buf_phys_addr == lbuf_phys_addr) {
read.mem_map_handle = buf_node->mmap_hdl;
break;
}
}
rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
if (rc < 0) {
pr_err("%s: read op[0x%x]rc[%d]\n", __func__,
read.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_async_read);
/**
* q6asm_write -
* command to write buffer data to DSP
*
* @ac: Audio client handle
* @len: buffer size
* @msw_ts: upper 32bits of timestamp
* @lsw_ts: lower 32bits of timestamp
* @flags: Flags for timestamp mode
*
* Returns 0 on success or error on failure
*/
int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags)
{
int rc = 0;
struct asm_data_cmd_write_v2 write;
struct asm_buffer_node *buf_node = NULL;
struct audio_port_data *port;
struct audio_buffer *ab;
int dsp_buf = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
dev_vdbg(ac->dev, "%s: session[%d] len=%d\n",
__func__, ac->session, len);
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[IN];
q6asm_add_hdr(ac, &write.hdr, sizeof(write),
FALSE);
mutex_lock(&port->lock);
dsp_buf = port->dsp_buf;
ab = &port->buf[dsp_buf];
q6asm_update_token(&write.hdr.token,
0, /* Session ID is NA */
0, /* Stream ID is NA */
port->dsp_buf,
0, /* Direction flag is NA */
NO_WAIT_CMD);
write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
write.buf_addr_lsw = lower_32_bits(ab->phys);
write.buf_addr_msw = msm_audio_populate_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;
/* Use 0xFF00 for disabling timestamps */
if (flags == 0xFF00)
write.flags = (0x00000000 | (flags & 0x800000FF));
else
write.flags = (0x80000000 | flags);
port->dsp_buf = q6asm_get_next_buf(ac, port->dsp_buf,
port->max_buf_cnt);
buf_node = list_first_entry(&ac->port[IN].mem_map_handle,
struct asm_buffer_node,
list);
write.mem_map_handle = buf_node->mmap_hdl;
dev_vdbg(ac->dev, "%s: ab->phys[%pK]bufadd[0x%x] token[0x%x]buf_id[0x%x]buf_size[0x%x]mmaphdl[0x%x]"
, __func__,
&ab->phys,
write.buf_addr_lsw,
write.hdr.token,
write.seq_id,
write.buf_size,
write.mem_map_handle);
mutex_unlock(&port->lock);
config_debug_fs_write(ab);
rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
if (rc < 0) {
pr_err("%s: write op[0x%x]rc[%d]\n",
__func__, write.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
}
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_write);
/**
* q6asm_write_nolock -
* command to write buffer data to DSP
* with no wait for ack.
*
* @ac: Audio client handle
* @len: buffer size
* @msw_ts: upper 32bits of timestamp
* @lsw_ts: lower 32bits of timestamp
* @flags: Flags for timestamp mode
*
* Returns 0 on success or error on failure
*/
int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags)
{
int rc = 0;
struct asm_data_cmd_write_v2 write;
struct asm_buffer_node *buf_node = NULL;
struct audio_port_data *port;
struct audio_buffer *ab;
int dsp_buf = 0;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
dev_vdbg(ac->dev, "%s: session[%d] len=%d\n",
__func__, ac->session, len);
if (ac->io_mode & SYNC_IO_MODE) {
port = &ac->port[IN];
q6asm_add_hdr_async(ac, &write.hdr, sizeof(write),
FALSE);
dsp_buf = port->dsp_buf;
ab = &port->buf[dsp_buf];
q6asm_update_token(&write.hdr.token,
0, /* Session ID is NA */
0, /* Stream ID is NA */
port->dsp_buf,
0, /* Direction flag is NA */
NO_WAIT_CMD);
write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
write.buf_addr_lsw = lower_32_bits(ab->phys);
write.buf_addr_msw = msm_audio_populate_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;
buf_node = list_first_entry(&ac->port[IN].mem_map_handle,
struct asm_buffer_node,
list);
write.mem_map_handle = buf_node->mmap_hdl;
/* Use 0xFF00 for disabling timestamps */
if (flags == 0xFF00)
write.flags = (0x00000000 | (flags & 0x800000FF));
else
write.flags = (0x80000000 | flags);
port->dsp_buf = q6asm_get_next_buf(ac, port->dsp_buf,
port->max_buf_cnt);
dev_vdbg(ac->dev, "%s: ab->phys[%pK]bufadd[0x%x]token[0x%x] buf_id[0x%x]buf_size[0x%x]mmaphdl[0x%x]"
, __func__,
&ab->phys,
write.buf_addr_lsw,
write.hdr.token,
write.seq_id,
write.buf_size,
write.mem_map_handle);
rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
if (rc < 0) {
pr_err("%s: write op[0x%x]rc[%d]\n",
__func__, write.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
}
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_write_nolock);
/**
* q6asm_get_session_time -
* command to retrieve timestamp info
*
* @ac: Audio client handle
* @tstamp: pointer to fill with timestamp info
*
* Returns 0 on success or error on failure
*/
int q6asm_get_session_time(struct audio_client *ac, uint64_t *tstamp)
{
struct asm_mtmx_strtr_get_params mtmx_params;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
if (tstamp == NULL) {
pr_err("%s: tstamp NULL\n", __func__);
return -EINVAL;
}
q6asm_add_hdr(ac, &mtmx_params.hdr, sizeof(mtmx_params), TRUE);
mtmx_params.hdr.opcode = ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2;
mtmx_params.param_info.data_payload_addr_lsw = 0;
mtmx_params.param_info.data_payload_addr_msw = 0;
mtmx_params.param_info.mem_map_handle = 0;
mtmx_params.param_info.direction = (ac->io_mode & TUN_READ_IO_MODE
? 1 : 0);
mtmx_params.param_info.module_id =
ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC;
mtmx_params.param_info.param_id =
ASM_SESSION_MTMX_STRTR_PARAM_SESSION_TIME_V3;
mtmx_params.param_info.param_max_size =
sizeof(struct asm_stream_param_data_v2) +
sizeof(struct asm_session_mtmx_strtr_param_session_time_v3_t);
atomic_set(&ac->time_flag, 1);
dev_vdbg(ac->dev, "%s: session[%d]opcode[0x%x]\n", __func__,
ac->session, mtmx_params.hdr.opcode);
rc = apr_send_pkt(ac->apr, (uint32_t *) &mtmx_params);
if (rc < 0) {
pr_err("%s: Commmand 0x%x failed %d\n", __func__,
mtmx_params.hdr.opcode, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->time_wait,
(atomic_read(&ac->time_flag) == 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout in getting session time from DSP\n",
__func__);
goto fail_cmd;
}
*tstamp = ac->time_stamp;
return 0;
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_get_session_time);
/**
* q6asm_get_session_time_legacy -
* command to retrieve timestamp info
*
* @ac: Audio client handle
* @tstamp: pointer to fill with timestamp info
*
* Returns 0 on success or error on failure
*/
int q6asm_get_session_time_legacy(struct audio_client *ac, uint64_t *tstamp)
{
struct apr_hdr hdr;
int rc;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
if (tstamp == NULL) {
pr_err("%s: tstamp NULL\n", __func__);
return -EINVAL;
}
q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
hdr.opcode = ASM_SESSION_CMD_GET_SESSIONTIME_V3;
atomic_set(&ac->time_flag, 1);
dev_vdbg(ac->dev, "%s: session[%d]opcode[0x%x]\n", __func__,
ac->session,
hdr.opcode);
rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
if (rc < 0) {
pr_err("%s: Commmand 0x%x failed %d\n",
__func__, hdr.opcode, rc);
goto fail_cmd;
}
rc = wait_event_timeout(ac->time_wait,
(atomic_read(&ac->time_flag) == 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout in getting session time from DSP\n",
__func__);
goto fail_cmd;
}
*tstamp = ac->time_stamp;
return 0;
fail_cmd:
return -EINVAL;
}
EXPORT_SYMBOL(q6asm_get_session_time_legacy);
/**
* q6asm_send_audio_effects_params -
* command to send audio effects params
*
* @ac: Audio client handle
* @params: audio effects params
* @params_length: size of params
*
* Returns 0 on success or error on failure
*/
int q6asm_send_audio_effects_params(struct audio_client *ac, char *params,
uint32_t params_length)
{
char *asm_params = NULL;
struct apr_hdr hdr;
struct asm_stream_cmd_set_pp_params_v2 payload_params;
int sz, rc, session_id = 0;
pr_debug("%s:\n", __func__);
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id)
return -EINVAL;
if (params == NULL) {
pr_err("%s: params NULL\n", __func__);
return -EINVAL;
}
sz = sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_pp_params_v2) +
params_length;
asm_params = kzalloc(sz, GFP_KERNEL);
if (!asm_params) {
pr_err("%s, asm params memory alloc failed", __func__);
return -ENOMEM;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto fail_send_param;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
rc = -EINVAL;
goto fail_send_param;
}
q6asm_add_hdr_async(ac, &hdr, (sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_pp_params_v2) +
params_length), TRUE);
atomic_set(&ac->cmd_state_pp, -1);
hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
payload_params.data_payload_addr_lsw = 0;
payload_params.data_payload_addr_msw = 0;
payload_params.mem_map_handle = 0;
payload_params.data_payload_size = params_length;
memcpy(((u8 *)asm_params), &hdr, sizeof(struct apr_hdr));
memcpy(((u8 *)asm_params + sizeof(struct apr_hdr)), &payload_params,
sizeof(struct asm_stream_cmd_set_pp_params_v2));
memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_pp_params_v2)),
params, params_length);
rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params);
if (rc < 0) {
pr_err("%s: audio effects set-params send failed\n", __func__);
rc = -EINVAL;
goto fail_send_param;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 1*HZ);
if (!rc) {
pr_err("%s: timeout, audio effects set-params\n", __func__);
rc = -ETIMEDOUT;
goto fail_send_param;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%s] set-params\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state_pp)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state_pp));
goto fail_send_param;
}
rc = 0;
fail_send_param:
mutex_unlock(&session[session_id].mutex_lock_per_session);
kfree(asm_params);
return rc;
}
EXPORT_SYMBOL(q6asm_send_audio_effects_params);
/**
* q6asm_send_mtmx_strtr_window -
* command to send matrix for window params
*
* @ac: Audio client handle
* @window_param: window params
* @param_id: param id for window
*
* Returns 0 on success or error on failure
*/
int q6asm_send_mtmx_strtr_window(struct audio_client *ac,
struct asm_session_mtmx_strtr_param_window_v2_t *window_param,
uint32_t param_id)
{
struct asm_mtmx_strtr_params matrix;
int sz = 0;
int rc = 0;
pr_debug("%s: Window lsw is %d, window msw is %d\n", __func__,
window_param->window_lsw, window_param->window_msw);
if (!ac) {
pr_err("%s: audio client handle is NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: ac->apr is NULL", __func__);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct asm_mtmx_strtr_params);
q6asm_add_hdr(ac, &matrix.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
matrix.hdr.opcode = ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2;
matrix.param.data_payload_addr_lsw = 0;
matrix.param.data_payload_addr_msw = 0;
matrix.param.mem_map_handle = 0;
matrix.param.data_payload_size =
sizeof(struct asm_stream_param_data_v2) +
sizeof(struct asm_session_mtmx_strtr_param_window_v2_t);
matrix.param.direction = 0; /* RX */
matrix.data.module_id = ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC;
matrix.data.param_id = param_id;
matrix.data.param_size =
sizeof(struct asm_session_mtmx_strtr_param_window_v2_t);
matrix.data.reserved = 0;
memcpy(&(matrix.config.window_param),
window_param,
sizeof(struct asm_session_mtmx_strtr_param_window_v2_t));
rc = apr_send_pkt(ac->apr, (uint32_t *) &matrix);
if (rc < 0) {
pr_err("%s: Render window start send failed paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, Render window start paramid[0x%x]\n",
__func__, matrix.data.param_id);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
rc = 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_send_mtmx_strtr_window);
/**
* q6asm_send_mtmx_strtr_render_mode -
* command to send matrix for render mode
*
* @ac: Audio client handle
* @render_mode: rendering mode
*
* Returns 0 on success or error on failure
*/
int q6asm_send_mtmx_strtr_render_mode(struct audio_client *ac,
uint32_t render_mode)
{
struct asm_mtmx_strtr_params matrix;
struct asm_session_mtmx_strtr_param_render_mode_t render_param;
int sz = 0;
int rc = 0;
pr_debug("%s: render mode is %d\n", __func__, render_mode);
if (!ac) {
pr_err("%s: audio client handle is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
if (ac->apr == NULL) {
pr_err("%s: ac->apr is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
if ((render_mode != ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT) &&
(render_mode != ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC)) {
pr_err("%s: Invalid render mode %d\n", __func__, render_mode);
rc = -EINVAL;
goto exit;
}
memset(&render_param, 0,
sizeof(struct asm_session_mtmx_strtr_param_render_mode_t));
render_param.flags = render_mode;
memset(&matrix, 0, sizeof(struct asm_mtmx_strtr_params));
sz = sizeof(struct asm_mtmx_strtr_params);
q6asm_add_hdr(ac, &matrix.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
matrix.hdr.opcode = ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2;
matrix.param.data_payload_addr_lsw = 0;
matrix.param.data_payload_addr_msw = 0;
matrix.param.mem_map_handle = 0;
matrix.param.data_payload_size =
sizeof(struct asm_stream_param_data_v2) +
sizeof(struct asm_session_mtmx_strtr_param_render_mode_t);
matrix.param.direction = 0; /* RX */
matrix.data.module_id = ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC;
matrix.data.param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD;
matrix.data.param_size =
sizeof(struct asm_session_mtmx_strtr_param_render_mode_t);
matrix.data.reserved = 0;
memcpy(&(matrix.config.render_param),
&render_param,
sizeof(struct asm_session_mtmx_strtr_param_render_mode_t));
rc = apr_send_pkt(ac->apr, (uint32_t *) &matrix);
if (rc < 0) {
pr_err("%s: Render mode send failed paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -EINVAL;
goto exit;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, Render mode send paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -ETIMEDOUT;
goto exit;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto exit;
}
rc = 0;
exit:
return rc;
}
EXPORT_SYMBOL(q6asm_send_mtmx_strtr_render_mode);
/**
* q6asm_send_mtmx_strtr_clk_rec_mode -
* command to send matrix for clock rec
*
* @ac: Audio client handle
* @clk_rec_mode: mode for clock rec
*
* Returns 0 on success or error on failure
*/
int q6asm_send_mtmx_strtr_clk_rec_mode(struct audio_client *ac,
uint32_t clk_rec_mode)
{
struct asm_mtmx_strtr_params matrix;
struct asm_session_mtmx_strtr_param_clk_rec_t clk_rec_param;
int sz = 0;
int rc = 0;
pr_debug("%s: clk rec mode is %d\n", __func__, clk_rec_mode);
if (!ac) {
pr_err("%s: audio client handle is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
if (ac->apr == NULL) {
pr_err("%s: ac->apr is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
if ((clk_rec_mode != ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE) &&
(clk_rec_mode != ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO)) {
pr_err("%s: Invalid clk rec mode %d\n", __func__, clk_rec_mode);
rc = -EINVAL;
goto exit;
}
memset(&clk_rec_param, 0,
sizeof(struct asm_session_mtmx_strtr_param_clk_rec_t));
clk_rec_param.flags = clk_rec_mode;
memset(&matrix, 0, sizeof(struct asm_mtmx_strtr_params));
sz = sizeof(struct asm_mtmx_strtr_params);
q6asm_add_hdr(ac, &matrix.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
matrix.hdr.opcode = ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2;
matrix.param.data_payload_addr_lsw = 0;
matrix.param.data_payload_addr_msw = 0;
matrix.param.mem_map_handle = 0;
matrix.param.data_payload_size =
sizeof(struct asm_stream_param_data_v2) +
sizeof(struct asm_session_mtmx_strtr_param_clk_rec_t);
matrix.param.direction = 0; /* RX */
matrix.data.module_id = ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC;
matrix.data.param_id = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD;
matrix.data.param_size =
sizeof(struct asm_session_mtmx_strtr_param_clk_rec_t);
matrix.data.reserved = 0;
memcpy(&(matrix.config.clk_rec_param),
&clk_rec_param,
sizeof(struct asm_session_mtmx_strtr_param_clk_rec_t));
rc = apr_send_pkt(ac->apr, (uint32_t *) &matrix);
if (rc < 0) {
pr_err("%s: clk rec mode send failed paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -EINVAL;
goto exit;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, clk rec mode send paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -ETIMEDOUT;
goto exit;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto exit;
}
rc = 0;
exit:
return rc;
}
EXPORT_SYMBOL(q6asm_send_mtmx_strtr_clk_rec_mode);
/**
* q6asm_send_mtmx_strtr_enable_adjust_session_clock -
* command to send matrix for adjust time
*
* @ac: Audio client handle
* @enable: flag to adjust time or not
*
* Returns 0 on success or error on failure
*/
int q6asm_send_mtmx_strtr_enable_adjust_session_clock(struct audio_client *ac,
bool enable)
{
struct asm_mtmx_strtr_params matrix;
struct asm_session_mtmx_param_adjust_session_time_ctl_t adjust_time;
int sz = 0;
int rc = 0;
pr_debug("%s: adjust session enable %d\n", __func__, enable);
if (!ac) {
pr_err("%s: audio client handle is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
if (ac->apr == NULL) {
pr_err("%s: ac->apr is NULL\n", __func__);
rc = -EINVAL;
goto exit;
}
adjust_time.enable = enable;
memset(&matrix, 0, sizeof(struct asm_mtmx_strtr_params));
sz = sizeof(struct asm_mtmx_strtr_params);
q6asm_add_hdr(ac, &matrix.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
matrix.hdr.opcode = ASM_SESSION_CMD_SET_MTMX_STRTR_PARAMS_V2;
matrix.param.data_payload_addr_lsw = 0;
matrix.param.data_payload_addr_msw = 0;
matrix.param.mem_map_handle = 0;
matrix.param.data_payload_size =
sizeof(struct asm_stream_param_data_v2) +
sizeof(struct asm_session_mtmx_param_adjust_session_time_ctl_t);
matrix.param.direction = 0; /* RX */
matrix.data.module_id = ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC;
matrix.data.param_id = ASM_SESSION_MTMX_PARAM_ADJUST_SESSION_TIME_CTL;
matrix.data.param_size =
sizeof(struct asm_session_mtmx_param_adjust_session_time_ctl_t);
matrix.data.reserved = 0;
matrix.config.adj_time_param.enable = adjust_time.enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &matrix);
if (rc < 0) {
pr_err("%s: enable adjust session failed failed paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -EINVAL;
goto exit;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: enable adjust session failed failed paramid [0x%x]\n",
__func__, matrix.data.param_id);
rc = -ETIMEDOUT;
goto exit;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto exit;
}
rc = 0;
exit:
return rc;
}
EXPORT_SYMBOL(q6asm_send_mtmx_strtr_enable_adjust_session_clock);
static int __q6asm_cmd(struct audio_client *ac, int cmd, uint32_t stream_id)
{
struct apr_hdr hdr;
int rc;
atomic_t *state;
int cnt = 0;
if (!ac) {
pr_err_ratelimited("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err_ratelimited("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
q6asm_stream_add_hdr(ac, &hdr, sizeof(hdr), TRUE, stream_id);
atomic_set(&ac->cmd_state, -1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
WAIT_CMD);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, hdr.token, stream_id, ac->session);
switch (cmd) {
case CMD_PAUSE:
pr_debug("%s: CMD_PAUSE\n", __func__);
hdr.opcode = ASM_SESSION_CMD_PAUSE;
state = &ac->cmd_state;
break;
case CMD_SUSPEND:
pr_debug("%s: CMD_SUSPEND\n", __func__);
hdr.opcode = ASM_SESSION_CMD_SUSPEND;
state = &ac->cmd_state;
break;
case CMD_FLUSH:
pr_debug("%s: CMD_FLUSH\n", __func__);
hdr.opcode = ASM_STREAM_CMD_FLUSH;
state = &ac->cmd_state;
break;
case CMD_OUT_FLUSH:
pr_debug("%s: CMD_OUT_FLUSH\n", __func__);
hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
state = &ac->cmd_state;
break;
case CMD_EOS:
pr_debug("%s: CMD_EOS\n", __func__);
hdr.opcode = ASM_DATA_CMD_EOS;
atomic_set(&ac->cmd_state, 0);
state = &ac->cmd_state;
break;
case CMD_CLOSE:
pr_debug("%s: CMD_CLOSE\n", __func__);
hdr.opcode = ASM_STREAM_CMD_CLOSE;
state = &ac->cmd_state;
break;
default:
pr_err("%s: Invalid format[%d]\n", __func__, cmd);
rc = -EINVAL;
goto fail_cmd;
}
pr_debug("%s: session[%d]opcode[0x%x]\n", __func__,
ac->session,
hdr.opcode);
rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
if (rc < 0) {
pr_err("%s: Commmand 0x%x failed %d\n",
__func__, hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait, (atomic_read(state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for response opcode[0x%x]\n",
__func__, hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(state) > 0) {
pr_err("%s: DSP returned error[%s] opcode %d\n",
__func__, adsp_err_get_err_str(
atomic_read(state)),
hdr.opcode);
rc = adsp_err_get_lnx_err_code(atomic_read(state));
goto fail_cmd;
}
if (cmd == CMD_FLUSH)
q6asm_reset_buf_state(ac);
if (cmd == CMD_CLOSE) {
/* check if DSP return all buffers */
if (ac->port[IN].buf) {
for (cnt = 0; cnt < ac->port[IN].max_buf_cnt;
cnt++) {
if (ac->port[IN].buf[cnt].used == IN) {
dev_vdbg(ac->dev, "Write Buf[%d] not returned\n",
cnt);
}
}
}
if (ac->port[OUT].buf) {
for (cnt = 0; cnt < ac->port[OUT].max_buf_cnt; cnt++) {
if (ac->port[OUT].buf[cnt].used == OUT) {
dev_vdbg(ac->dev, "Read Buf[%d] not returned\n",
cnt);
}
}
}
}
return 0;
fail_cmd:
return rc;
}
/**
* q6asm_cmd -
* Function used to send commands for
* ASM with wait for ack.
*
* @ac: Audio client handle
* @cmd: command to send
*
* Returns 0 on success or error on failure
*/
int q6asm_cmd(struct audio_client *ac, int cmd)
{
return __q6asm_cmd(ac, cmd, ac->stream_id);
}
EXPORT_SYMBOL(q6asm_cmd);
/**
* q6asm_stream_cmd -
* Function used to send commands for
* ASM stream with wait for ack.
*
* @ac: Audio client handle
* @cmd: command to send
* @stream_id: Stream ID
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_cmd(struct audio_client *ac, int cmd, uint32_t stream_id)
{
return __q6asm_cmd(ac, cmd, stream_id);
}
EXPORT_SYMBOL(q6asm_stream_cmd);
/**
* q6asm_cmd_nowait -
* Function used to send commands for
* ASM stream without wait for ack.
*
* @ac: Audio client handle
* @cmd: command to send
* @stream_id: Stream ID
*
* Returns 0 on success or error on failure
*/
static int __q6asm_cmd_nowait(struct audio_client *ac, int cmd,
uint32_t stream_id)
{
struct apr_hdr hdr;
int rc;
if (!ac) {
pr_err_ratelimited("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err_ratelimited("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
q6asm_stream_add_hdr_async(ac, &hdr, sizeof(hdr), TRUE, stream_id);
atomic_set(&ac->cmd_state, 1);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
NO_WAIT_CMD);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, hdr.token, stream_id, ac->session);
switch (cmd) {
case CMD_PAUSE:
pr_debug("%s: CMD_PAUSE\n", __func__);
hdr.opcode = ASM_SESSION_CMD_PAUSE;
break;
case CMD_FLUSH:
pr_debug("%s: CMD_FLUSH\n", __func__);
hdr.opcode = ASM_STREAM_CMD_FLUSH;
break;
case CMD_EOS:
pr_debug("%s: CMD_EOS\n", __func__);
hdr.opcode = ASM_DATA_CMD_EOS;
break;
case CMD_CLOSE:
pr_debug("%s: CMD_CLOSE\n", __func__);
hdr.opcode = ASM_STREAM_CMD_CLOSE;
break;
default:
pr_err("%s: Invalid format[%d]\n", __func__, cmd);
goto fail_cmd;
}
pr_debug("%s: session[%d]opcode[0x%x]\n", __func__,
ac->session,
hdr.opcode);
rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
if (rc < 0) {
pr_err("%s: Commmand 0x%x failed %d\n",
__func__, hdr.opcode, rc);
goto fail_cmd;
}
return 0;
fail_cmd:
return -EINVAL;
}
int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
{
pr_debug("%s: stream_id: %d\n", __func__, ac->stream_id);
return __q6asm_cmd_nowait(ac, cmd, ac->stream_id);
}
EXPORT_SYMBOL(q6asm_cmd_nowait);
/**
* q6asm_stream_cmd_nowait -
* Function used to send commands for
* ASM stream without wait for ack.
*
* @ac: Audio client handle
* @cmd: command to send
* @stream_id: Stream ID
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_cmd_nowait(struct audio_client *ac, int cmd,
uint32_t stream_id)
{
pr_debug("%s: stream_id: %d\n", __func__, stream_id);
return __q6asm_cmd_nowait(ac, cmd, stream_id);
}
EXPORT_SYMBOL(q6asm_stream_cmd_nowait);
int __q6asm_send_meta_data(struct audio_client *ac, uint32_t stream_id,
uint32_t initial_samples, uint32_t trailing_samples)
{
struct asm_data_cmd_remove_silence silence;
int rc = 0;
if (!ac) {
pr_err_ratelimited("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err_ratelimited("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_stream_add_hdr_async(ac, &silence.hdr, sizeof(silence), TRUE,
stream_id);
/*
* Updated the token field with stream/session for compressed playback
* Platform driver must know the the stream with which the command is
* associated
*/
if (ac->io_mode & COMPRESSED_STREAM_IO)
q6asm_update_token(&silence.hdr.token,
ac->session,
stream_id,
0, /* Buffer index is NA */
0, /* Direction flag is NA */
NO_WAIT_CMD);
pr_debug("%s: token = 0x%x, stream_id %d, session 0x%x\n",
__func__, silence.hdr.token, stream_id, ac->session);
silence.hdr.opcode = ASM_DATA_CMD_REMOVE_INITIAL_SILENCE;
silence.num_samples_to_remove = initial_samples;
rc = apr_send_pkt(ac->apr, (uint32_t *) &silence);
if (rc < 0) {
pr_err("%s: Commmand silence failed[%d]", __func__, rc);
goto fail_cmd;
}
silence.hdr.opcode = ASM_DATA_CMD_REMOVE_TRAILING_SILENCE;
silence.num_samples_to_remove = trailing_samples;
rc = apr_send_pkt(ac->apr, (uint32_t *) &silence);
if (rc < 0) {
pr_err("%s: Commmand silence failed[%d]", __func__, rc);
goto fail_cmd;
}
return 0;
fail_cmd:
return -EINVAL;
}
/**
* q6asm_stream_send_meta_data -
* command to send meta data for stream
*
* @ac: Audio client handle
* @stream_id: Stream ID
* @initial_samples: Initial samples of stream
* @trailing_samples: Trailing samples of stream
*
* Returns 0 on success or error on failure
*/
int q6asm_stream_send_meta_data(struct audio_client *ac, uint32_t stream_id,
uint32_t initial_samples, uint32_t trailing_samples)
{
return __q6asm_send_meta_data(ac, stream_id, initial_samples,
trailing_samples);
}
EXPORT_SYMBOL(q6asm_stream_send_meta_data);
int q6asm_send_meta_data(struct audio_client *ac, uint32_t initial_samples,
uint32_t trailing_samples)
{
return __q6asm_send_meta_data(ac, ac->stream_id, initial_samples,
trailing_samples);
}
static void q6asm_reset_buf_state(struct audio_client *ac)
{
int cnt = 0;
int loopcnt = 0;
int used;
struct audio_port_data *port = NULL;
if (ac->io_mode & SYNC_IO_MODE) {
used = (ac->io_mode & TUN_WRITE_IO_MODE ? 1 : 0);
mutex_lock(&ac->cmd_lock);
for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
port = &ac->port[loopcnt];
cnt = port->max_buf_cnt - 1;
port->dsp_buf = 0;
port->cpu_buf = 0;
while (cnt >= 0) {
if (!port->buf)
continue;
port->buf[cnt].used = used;
cnt--;
}
}
mutex_unlock(&ac->cmd_lock);
}
}
/**
* q6asm_reg_tx_overflow -
* command to register for TX overflow events
*
* @ac: Audio client handle
* @enable: flag to enable or disable events
*
* Returns 0 on success or error on failure
*/
int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable)
{
struct asm_session_cmd_regx_overflow tx_overflow;
int rc;
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]enable[%d]\n", __func__,
ac->session, enable);
q6asm_add_hdr(ac, &tx_overflow.hdr, sizeof(tx_overflow), TRUE);
atomic_set(&ac->cmd_state, -1);
tx_overflow.hdr.opcode =
ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS;
/* tx overflow event: enable */
tx_overflow.enable_flag = enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &tx_overflow);
if (rc < 0) {
pr_err("%s: tx overflow op[0x%x]rc[%d]\n",
__func__, tx_overflow.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for tx overflow\n", __func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_reg_tx_overflow);
int q6asm_reg_rx_underflow(struct audio_client *ac, uint16_t enable)
{
struct asm_session_cmd_rgstr_rx_underflow rx_underflow;
int rc;
if (!ac) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]enable[%d]\n", __func__,
ac->session, enable);
q6asm_add_hdr_async(ac, &rx_underflow.hdr, sizeof(rx_underflow), FALSE);
rx_underflow.hdr.opcode =
ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS;
/* tx overflow event: enable */
rx_underflow.enable_flag = enable;
rc = apr_send_pkt(ac->apr, (uint32_t *) &rx_underflow);
if (rc < 0) {
pr_err("%s: tx overflow op[0x%x]rc[%d]\n",
__func__, rx_underflow.hdr.opcode, rc);
goto fail_cmd;
}
return 0;
fail_cmd:
return -EINVAL;
}
/**
* q6asm_adjust_session_clock -
* command to adjust session clock
*
* @ac: Audio client handle
* @adjust_time_lsw: lower 32bits
* @adjust_time_msw: upper 32bits
*
* Returns 0 on success or error on failure
*/
int q6asm_adjust_session_clock(struct audio_client *ac,
uint32_t adjust_time_lsw,
uint32_t adjust_time_msw)
{
int rc = 0;
int sz = 0;
struct asm_session_cmd_adjust_session_clock_v2 adjust_clock;
pr_debug("%s: adjust_time_lsw is %x, adjust_time_msw is %x\n", __func__,
adjust_time_lsw, adjust_time_msw);
if (!ac) {
pr_err("%s: audio client handle is NULL\n", __func__);
rc = -EINVAL;
goto fail_cmd;
}
if (ac->apr == NULL) {
pr_err("%s: ac->apr is NULL", __func__);
rc = -EINVAL;
goto fail_cmd;
}
sz = sizeof(struct asm_session_cmd_adjust_session_clock_v2);
q6asm_add_hdr(ac, &adjust_clock.hdr, sz, TRUE);
atomic_set(&ac->cmd_state, -1);
adjust_clock.hdr.opcode = ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2;
adjust_clock.adjustime_lsw = adjust_time_lsw;
adjust_clock.adjustime_msw = adjust_time_msw;
rc = apr_send_pkt(ac->apr, (uint32_t *) &adjust_clock);
if (rc < 0) {
pr_err("%s: adjust_clock send failed paramid [0x%x]\n",
__func__, adjust_clock.hdr.opcode);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout, adjust_clock paramid[0x%x]\n",
__func__, adjust_clock.hdr.opcode);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
rc = 0;
fail_cmd:
return rc;
}
EXPORT_SYMBOL(q6asm_adjust_session_clock);
/*
* q6asm_get_path_delay() - get the path delay for an audio session
* @ac: audio client handle
*
* Retrieves the current audio DSP path delay for the given audio session.
*
* Return: 0 on success, error code otherwise
*/
int q6asm_get_path_delay(struct audio_client *ac)
{
int rc = 0;
struct apr_hdr hdr;
if (!ac || ac->apr == NULL) {
pr_err("%s: invalid audio client\n", __func__);
return -EINVAL;
}
hdr.opcode = ASM_SESSION_CMD_GET_PATH_DELAY_V2;
q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
atomic_set(&ac->cmd_state, -1);
rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
if (rc < 0) {
pr_err("%s: Commmand 0x%x failed %d\n", __func__,
hdr.opcode, rc);
return rc;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5 * HZ);
if (!rc) {
pr_err("%s: timeout. waited for response opcode[0x%x]\n",
__func__, hdr.opcode);
return -ETIMEDOUT;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
return rc;
}
return 0;
}
EXPORT_SYMBOL(q6asm_get_path_delay);
int q6asm_get_apr_service_id(int session_id)
{
pr_debug("%s:\n", __func__);
if (session_id <= 0 || session_id > ASM_ACTIVE_STREAMS_ALLOWED) {
pr_err("%s: invalid session_id = %d\n", __func__, session_id);
return -EINVAL;
}
return ((struct apr_svc *)(session[session_id].ac)->apr)->id;
}
int q6asm_get_asm_topology(int session_id)
{
int topology = -EINVAL;
if (session_id <= 0 || session_id > ASM_ACTIVE_STREAMS_ALLOWED) {
pr_err("%s: invalid session_id = %d\n", __func__, session_id);
goto done;
}
if (session[session_id].ac == NULL) {
pr_err("%s: session not created for session id = %d\n",
__func__, session_id);
goto done;
}
topology = (session[session_id].ac)->topology;
done:
return topology;
}
int q6asm_get_asm_app_type(int session_id)
{
int app_type = -EINVAL;
if (session_id <= 0 || session_id > ASM_ACTIVE_STREAMS_ALLOWED) {
pr_err("%s: invalid session_id = %d\n", __func__, session_id);
goto done;
}
if (session[session_id].ac == NULL) {
pr_err("%s: session not created for session id = %d\n",
__func__, session_id);
goto done;
}
app_type = (session[session_id].ac)->app_type;
done:
return app_type;
}
/*
* Retrieving cal_block will mark cal_block as stale.
* Hence it cannot be reused or resent unless the flag
* is reset.
*/
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info)
{
struct cal_block_data *cal_block = NULL;
cal_info->topology_id = DEFAULT_POPP_TOPOLOGY;
cal_info->app_type = DEFAULT_APP_TYPE;
if (cal_data[ASM_TOPOLOGY_CAL] == NULL)
goto done;
mutex_lock(&cal_data[ASM_TOPOLOGY_CAL]->lock);
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_TOPOLOGY_CAL]);
if (cal_block == NULL || cal_utils_is_cal_stale(cal_block))
goto unlock;
cal_info->topology_id = ((struct audio_cal_info_asm_top *)
cal_block->cal_info)->topology;
cal_info->app_type = ((struct audio_cal_info_asm_top *)
cal_block->cal_info)->app_type;
cal_utils_mark_cal_used(cal_block);
unlock:
mutex_unlock(&cal_data[ASM_TOPOLOGY_CAL]->lock);
done:
pr_debug("%s: Using topology %d app_type %d\n", __func__,
cal_info->topology_id, cal_info->app_type);
return 0;
}
/**
* q6asm_send_cal -
* command to send ASM calibration
*
* @ac: Audio client handle
*
* Returns 0 on success or error on failure
*/
int q6asm_send_cal(struct audio_client *ac)
{
struct cal_block_data *cal_block = NULL;
struct apr_hdr hdr;
char *asm_params = NULL;
struct asm_stream_cmd_set_pp_params_v2 payload_params;
int sz, rc = -EINVAL, session_id = 0;
pr_debug("%s:\n", __func__);
if (!ac) {
pr_err("%s: APR handle NULL\n", __func__);
goto done;
}
session_id = q6asm_get_session_id_from_audio_client(ac);
if (!session_id)
goto done;
if (cal_data[ASM_AUDSTRM_CAL] == NULL)
goto done;
mutex_lock(&cal_data[ASM_AUDSTRM_CAL]->lock);
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_AUDSTRM_CAL]);
if (cal_block == NULL || cal_utils_is_cal_stale(cal_block)) {
rc = 0; /* not error case */
pr_err("%s: cal_block is NULL or stale\n",
__func__);
goto unlock;
}
if (cal_block->cal_data.size == 0) {
rc = 0; /* not error case */
pr_debug("%s: cal_data.size is 0, don't send cal data\n",
__func__);
goto unlock;
}
rc = remap_cal_data(ASM_AUDSTRM_CAL_TYPE, cal_block);
if (rc) {
pr_err("%s: Remap_cal_data failed for cal %d!\n",
__func__, ASM_AUDSTRM_CAL);
goto unlock;
}
sz = sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_pp_params_v2);
asm_params = kzalloc(sz, GFP_KERNEL);
if (!asm_params) {
pr_err("%s, asm params memory alloc failed", __func__);
rc = -ENOMEM;
goto unlock;
}
mutex_lock(&session[session_id].mutex_lock_per_session);
if (!q6asm_is_valid_audio_client(ac)) {
rc = -EINVAL;
goto free;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
goto free;
}
if (ac->io_mode & NT_MODE) {
pr_debug("%s: called for NT MODE, exiting\n", __func__);
goto free;
}
if (ac->perf_mode == ULTRA_LOW_LATENCY_PCM_MODE) {
rc = 0; /* no cal is required, not error case */
goto free;
}
/* asm_stream_cmd_set_pp_params_v2 has no APR header in it */
q6asm_add_hdr_async(ac, &hdr, (sizeof(struct apr_hdr) +
sizeof(struct asm_stream_cmd_set_pp_params_v2)), TRUE);
atomic_set(&ac->cmd_state_pp, -1);
hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
payload_params.data_payload_addr_lsw =
lower_32_bits(cal_block->cal_data.paddr);
payload_params.data_payload_addr_msw =
msm_audio_populate_upper_32_bits(
cal_block->cal_data.paddr);
payload_params.mem_map_handle = cal_block->map_data.q6map_handle;
payload_params.data_payload_size = cal_block->cal_data.size;
memcpy(((u8 *)asm_params), &hdr, sizeof(struct apr_hdr));
memcpy(((u8 *)asm_params + sizeof(struct apr_hdr)), &payload_params,
sizeof(struct asm_stream_cmd_set_pp_params_v2));
pr_debug("%s: phyaddr lsw = %x msw = %x, maphdl = %x calsize = %d\n",
__func__, payload_params.data_payload_addr_lsw,
payload_params.data_payload_addr_msw,
payload_params.mem_map_handle,
payload_params.data_payload_size);
rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params);
if (rc < 0) {
pr_err("%s: audio audstrm cal send failed\n", __func__);
rc = -EINVAL;
goto free;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state_pp) >= 0), 5 * HZ);
if (!rc) {
pr_err("%s: timeout, audio audstrm cal send\n", __func__);
rc = -ETIMEDOUT;
goto free;
}
if (atomic_read(&ac->cmd_state_pp) > 0) {
pr_err("%s: DSP returned error[%d] audio audstrm cal send\n",
__func__, atomic_read(&ac->cmd_state_pp));
rc = -EINVAL;
goto free;
}
if (cal_block)
cal_utils_mark_cal_used(cal_block);
rc = 0;
free:
mutex_unlock(&session[session_id].mutex_lock_per_session);
kfree(asm_params);
unlock:
mutex_unlock(&cal_data[ASM_AUDSTRM_CAL]->lock);
done:
return rc;
}
EXPORT_SYMBOL(q6asm_send_cal);
static int get_cal_type_index(int32_t cal_type)
{
int ret = -EINVAL;
switch (cal_type) {
case ASM_TOPOLOGY_CAL_TYPE:
ret = ASM_TOPOLOGY_CAL;
break;
case ASM_CUST_TOPOLOGY_CAL_TYPE:
ret = ASM_CUSTOM_TOP_CAL;
break;
case ASM_AUDSTRM_CAL_TYPE:
ret = ASM_AUDSTRM_CAL;
break;
case ASM_RTAC_APR_CAL_TYPE:
ret = ASM_RTAC_APR_CAL;
break;
default:
pr_err("%s: invalid cal type %d!\n", __func__, cal_type);
}
return ret;
}
static int q6asm_alloc_cal(int32_t cal_type,
size_t data_size, void *data)
{
int ret = 0;
int cal_index;
pr_debug("%s:\n", __func__);
cal_index = get_cal_type_index(cal_type);
if (cal_index < 0) {
pr_err("%s: could not get cal index %d!\n",
__func__, cal_index);
ret = -EINVAL;
goto done;
}
ret = cal_utils_alloc_cal(data_size, data,
cal_data[cal_index], 0, NULL);
if (ret < 0) {
pr_err("%s: cal_utils_alloc_block failed, ret = %d, cal type = %d!\n",
__func__, ret, cal_type);
ret = -EINVAL;
goto done;
}
done:
return ret;
}
static int q6asm_dealloc_cal(int32_t cal_type,
size_t data_size, void *data)
{
int ret = 0;
int cal_index;
pr_debug("%s:\n", __func__);
cal_index = get_cal_type_index(cal_type);
if (cal_index < 0) {
pr_err("%s: could not get cal index %d!\n",
__func__, cal_index);
ret = -EINVAL;
goto done;
}
ret = cal_utils_dealloc_cal(data_size, data,
cal_data[cal_index]);
if (ret < 0) {
pr_err("%s: cal_utils_dealloc_block failed, ret = %d, cal type = %d!\n",
__func__, ret, cal_type);
ret = -EINVAL;
goto done;
}
done:
return ret;
}
static int q6asm_set_cal(int32_t cal_type,
size_t data_size, void *data)
{
int ret = 0;
int cal_index;
pr_debug("%s:\n", __func__);
cal_index = get_cal_type_index(cal_type);
if (cal_index < 0) {
pr_err("%s: could not get cal index %d!\n",
__func__, cal_index);
ret = -EINVAL;
goto done;
}
ret = cal_utils_set_cal(data_size, data,
cal_data[cal_index], 0, NULL);
if (ret < 0) {
pr_err("%s: cal_utils_set_cal failed, ret = %d, cal type = %d!\n",
__func__, ret, cal_type);
ret = -EINVAL;
goto done;
}
if (cal_index == ASM_CUSTOM_TOP_CAL) {
mutex_lock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
set_custom_topology = 1;
mutex_unlock(&cal_data[ASM_CUSTOM_TOP_CAL]->lock);
}
done:
return ret;
}
static void q6asm_delete_cal_data(void)
{
pr_debug("%s:\n", __func__);
cal_utils_destroy_cal_types(ASM_MAX_CAL_TYPES, cal_data);
}
static int q6asm_init_cal_data(void)
{
int ret = 0;
struct cal_type_info cal_type_info[] = {
{{ASM_TOPOLOGY_CAL_TYPE,
{NULL, NULL, NULL,
q6asm_set_cal, NULL, NULL} },
{NULL, NULL, cal_utils_match_buf_num} },
{{ASM_CUST_TOPOLOGY_CAL_TYPE,
{q6asm_alloc_cal, q6asm_dealloc_cal, NULL,
q6asm_set_cal, NULL, NULL} },
{NULL, q6asm_unmap_cal_memory, cal_utils_match_buf_num} },
{{ASM_AUDSTRM_CAL_TYPE,
{q6asm_alloc_cal, q6asm_dealloc_cal, NULL,
q6asm_set_cal, NULL, NULL} },
{NULL, q6asm_unmap_cal_memory, cal_utils_match_buf_num} },
{{ASM_RTAC_APR_CAL_TYPE,
{NULL, NULL, NULL, NULL, NULL, NULL} },
{NULL, NULL, cal_utils_match_buf_num} }
};
pr_debug("%s\n", __func__);
ret = cal_utils_create_cal_types(ASM_MAX_CAL_TYPES, cal_data,
cal_type_info);
if (ret < 0) {
pr_err("%s: could not create cal type! %d\n",
__func__, ret);
ret = -EINVAL;
goto err;
}
return ret;
err:
q6asm_delete_cal_data();
return ret;
}
static int q6asm_is_valid_session(struct apr_client_data *data, void *priv)
{
struct audio_client *ac = (struct audio_client *)priv;
union asm_token_struct asm_token;
asm_token.token = data->token;
if (asm_token._token.session_id != ac->session) {
pr_err("%s: Invalid session[%d] rxed expected[%d]",
__func__, asm_token._token.session_id, ac->session);
return -EINVAL;
}
return 0;
}
int __init q6asm_init(void)
{
int lcnt, ret;
pr_debug("%s:\n", __func__);
memset(session, 0, sizeof(struct audio_session) *
(ASM_ACTIVE_STREAMS_ALLOWED + 1));
for (lcnt = 0; lcnt <= ASM_ACTIVE_STREAMS_ALLOWED; lcnt++) {
spin_lock_init(&(session[lcnt].session_lock));
mutex_init(&(session[lcnt].mutex_lock_per_session));
}
set_custom_topology = 1;
/*setup common client used for cal mem map */
common_client.session = ASM_CONTROL_SESSION;
common_client.port[0].buf = &common_buf[0];
common_client.port[1].buf = &common_buf[1];
init_waitqueue_head(&common_client.cmd_wait);
init_waitqueue_head(&common_client.time_wait);
init_waitqueue_head(&common_client.mem_wait);
atomic_set(&common_client.time_flag, 1);
INIT_LIST_HEAD(&common_client.port[0].mem_map_handle);
INIT_LIST_HEAD(&common_client.port[1].mem_map_handle);
mutex_init(&common_client.cmd_lock);
for (lcnt = 0; lcnt <= OUT; lcnt++) {
mutex_init(&common_client.port[lcnt].lock);
spin_lock_init(&common_client.port[lcnt].dsp_lock);
}
atomic_set(&common_client.cmd_state, 0);
atomic_set(&common_client.mem_state, 0);
ret = q6asm_init_cal_data();
if (ret)
pr_err("%s: could not init cal data! ret %d\n",
__func__, ret);
config_debug_fs_init();
return 0;
}
void q6asm_exit(void)
{
q6asm_delete_cal_data();
}