| // SPDX-License-Identifier: GPL-2.0-only |
| /* |
| * 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/q6common.h> |
| #include <dsp/q6core.h> |
| #include "adsp_err.h" |
| |
| #define TIMEOUT_MS 1000 |
| #define TRUE 0x01 |
| #define FALSE 0x00 |
| #define SESSION_MAX 8 |
| |
| #define ENC_FRAMES_PER_BUFFER 0x01 |
| |
| 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); |
| |
| 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 |
| }; |
| |
| /* |
| * get_monotonic_timeval - |
| * This method returns a structure in timeval |
| * format (sec,microsec) by using ktime kernel |
| * API to get time in nano secs and then converts |
| * it to timeval format |
| * |
| * ktime_get [nsec]-> ktime_to_timespec [sec,nsec]-> timeval[sec,usec] |
| * |
| * Returns struct timeval |
| */ |
| static struct timeval get_monotonic_timeval(void) |
| { |
| static struct timeval out_tval; |
| |
| /* Get time from monotonic clock in nanoseconds */ |
| ktime_t kTimeNsec = ktime_get(); |
| |
| /* Convert it to timespec format and later to timeval as expected by audio HAL */ |
| struct timespec temp_tspec = ktime_to_timespec(kTimeNsec); |
| |
| /* Time returned above is in sec,nanosec format, needs to convert to sec,microsec */ |
| out_tval.tv_usec = temp_tspec.tv_nsec/1000; |
| out_tval.tv_sec = temp_tspec.tv_sec; |
| return out_tval; |
| } |
| |
| 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) { |
| out_cold_tv = get_monotonic_timeval(); |
| 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) { |
| in_cont_tv = get_monotonic_timeval(); |
| 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) { |
| out_cold_tv = get_monotonic_timeval(); |
| 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))) { |
| out_warm_tv = get_monotonic_timeval(); |
| 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))) { |
| out_cont_tv = get_monotonic_timeval(); |
| 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.dma_buf == 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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].dma_buf); |
| |
| port->buf[cnt].dma_buf = 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], dma_buf[%pK]\n", |
| __func__, |
| port->buf[0].data, |
| &port->buf[0].phys, |
| port->buf[0].dma_buf); |
| if (!rc || atomic_read(&ac->reset)) |
| msm_audio_ion_free(port->buf[0].dma_buf); |
| port->buf[0].dma_buf = 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; |
| uint32_t sz = 0; |
| uint64_t actual_sz = 0; |
| int session_id = 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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( |
| &buf[cnt].dma_buf, |
| bufsz, |
| &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; |
| } |
| 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(&buf[0].dma_buf, |
| bytes_to_alloc, |
| &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; |
| } |
| |
| 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: |
| case ASM_STREAM_CMD_SET_PP_PARAMS_V3: |
| 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_READ_COMPRESSED: |
| case ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED: |
| if (data->payload_size >= 2 * sizeof(uint32_t)) { |
| 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); |
| if (payload[1] != 0) { |
| 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) || |
| (payload[0] == |
| ASM_STREAM_CMD_SET_PP_PARAMS_V3)) |
| 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; |
| } |
| } else { |
| pr_err("%s: payload size of %x is less than expected.\n", |
| __func__, data->payload_size); |
| } |
| if ((is_adsp_reg_event(payload[0]) >= 0) || |
| (payload[0] == ASM_STREAM_CMD_SET_PP_PARAMS_V2) || |
| (payload[0] == ASM_STREAM_CMD_SET_PP_PARAMS_V3)) { |
| 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)) { |
| pr_debug("%s:Payload = [0x%x]stat[0x%x]\n", |
| __func__, payload[0], payload[1]); |
| if (payload[1] != 0) { |
| 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; |
| } |
| } else { |
| pr_err("%s: payload size of %x is less than expected.\n", |
| __func__, data->payload_size); |
| } |
| 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: |
| case ASM_STREAM_CMD_GET_PP_PARAMS_V3: |
| pr_debug("%s: ASM_STREAM_CMD_GET_PP_PARAMS 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: |
| case ASM_STREAM_CMDRSP_GET_PP_PARAMS_V3: |
| pr_debug("%s: ASM_STREAM_CMDRSP_GET_PP_PARAMS 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 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 Read 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_IEC_61937_MEDIA_FMT_EVENT: |
| 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 if (data->payload_size >= sizeof(uint32_t)) |
| pr_debug("%s: ASM_STREAM_EVENT payload[0][0x%x]", |
| __func__, payload[0]); |
| 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); |
| if ((data->opcode == ASM_IEC_61937_MEDIA_FMT_EVENT) && |
| (payload_size == 4)) { |
| switch (payload[0]) { |
| case ASM_MEDIA_FMT_AC3: |
| ((uint32_t *)pp_event_package->payload)[0] = |
| SND_AUDIOCODEC_AC3; |
| break; |
| case ASM_MEDIA_FMT_EAC3: |
| ((uint32_t *)pp_event_package->payload)[0] = |
| SND_AUDIOCODEC_EAC3; |
| break; |
| case ASM_MEDIA_FMT_DTS: |
| ((uint32_t *)pp_event_package->payload)[0] = |
| SND_AUDIOCODEC_DTS; |
| break; |
| case ASM_MEDIA_FMT_TRUEHD: |
| ((uint32_t *)pp_event_package->payload)[0] = |
| SND_AUDIOCODEC_TRUEHD; |
| break; |
| case ASM_MEDIA_FMT_AAC_V2: |
| ((uint32_t *)pp_event_package->payload)[0] = |
| SND_AUDIOCODEC_AAC; |
| break; |
| default: |
| pr_debug("%s: Event with unknown media_fmt 0x%x\n", |
| __func__, payload[0]); |
| } |
| } |
| 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 (payload[0] == 0 && data->payload_size >= 2 * sizeof(uint32_t)) { |
| 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; |
| } |
| |
| /** |
| * q6asm_set_pp_params |
| * command to set ASM parameter data |
| * send memory mapping header for out of band case |
| * send pre-packed parameter data for in band case |
| * |
| * @ac: audio client handle |
| * @mem_hdr: memory mapping header |
| * @param_data: pre-packed parameter payload |
| * @param_size: size of pre-packed parameter data |
| * |
| * Returns 0 on success or error on failure |
| */ |
| int q6asm_set_pp_params(struct audio_client *ac, |
| struct mem_mapping_hdr *mem_hdr, u8 *param_data, |
| u32 param_size) |
| { |
| struct asm_stream_cmd_set_pp_params *asm_set_param = NULL; |
| int pkt_size = 0; |
| int ret = 0; |
| int session_id = 0; |
| |
| if (ac == NULL) { |
| pr_err("%s: Audio Client is NULL\n", __func__); |
| return -EINVAL; |
| } else if (ac->apr == NULL) { |
| pr_err("%s: APR pointer is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| session_id = q6asm_get_session_id_from_audio_client(ac); |
| if (!session_id) |
| return -EINVAL; |
| |
| pkt_size = sizeof(struct asm_stream_cmd_set_pp_params); |
| /* Add param size to packet size when sending in-band only */ |
| if (param_data != NULL) |
| pkt_size += param_size; |
| asm_set_param = kzalloc(pkt_size, GFP_KERNEL); |
| if (!asm_set_param) |
| return -ENOMEM; |
| |
| mutex_lock(&session[session_id].mutex_lock_per_session); |
| if (!q6asm_is_valid_audio_client(ac)) { |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if (ac->apr == NULL) { |
| pr_err("%s: AC APR handle NULL\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| q6asm_add_hdr_async(ac, &asm_set_param->apr_hdr, pkt_size, TRUE); |
| |
| /* With pre-packed data, only the opcode differs from V2 and V3. */ |
| if (q6common_is_instance_id_supported()) |
| asm_set_param->apr_hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V3; |
| else |
| asm_set_param->apr_hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2; |
| |
| asm_set_param->payload_size = param_size; |
| |
| if (mem_hdr != NULL) { |
| /* Out of band case */ |
| asm_set_param->mem_hdr = *mem_hdr; |
| } else if (param_data != NULL) { |
| /* |
| * In band case. Parameter data must be pre-packed with its |
| * header before calling this function. Use |
| * q6common_pack_pp_params to pack parameter data and header |
| * correctly. |
| */ |
| memcpy(&asm_set_param->param_data, param_data, param_size); |
| } else { |
| pr_err("%s: Received NULL pointers for both mem header and param data\n", |
| __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| atomic_set(&ac->cmd_state_pp, -1); |
| ret = apr_send_pkt(ac->apr, (uint32_t *)asm_set_param); |
| if (ret < 0) { |
| pr_err("%s: apr send failed rc %d\n", __func__, ret); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ret = wait_event_timeout(ac->cmd_wait, |
| atomic_read(&ac->cmd_state_pp) >= 0, |
| msecs_to_jiffies(TIMEOUT_MS)); |
| if (!ret) { |
| pr_err("%s: timeout sending apr pkt\n", __func__); |
| ret = -ETIMEDOUT; |
| goto done; |
| } |
| |
| if (atomic_read(&ac->cmd_state_pp) > 0) { |
| pr_err("%s: DSP returned error[%s]\n", __func__, |
| adsp_err_get_err_str(atomic_read(&ac->cmd_state_pp))); |
| ret = adsp_err_get_lnx_err_code(atomic_read(&ac->cmd_state_pp)); |
| goto done; |
| } |
| ret = 0; |
| done: |
| mutex_unlock(&session[session_id].mutex_lock_per_session); |
| kfree(asm_set_param); |
| return ret; |
| } |
| EXPORT_SYMBOL(q6asm_set_pp_params); |
| |
| /** |
| * q6asm_pack_and_set_pp_param_in_band |
| * command to pack and set parameter data for in band case |
| * |
| * @ac: audio client handle |
| * @param_hdr: parameter header |
| * @param_data: parameter data |
| * |
| * Returns 0 on success or error on failure |
| */ |
| int q6asm_pack_and_set_pp_param_in_band(struct audio_client *ac, |
| struct param_hdr_v3 param_hdr, |
| u8 *param_data) |
| { |
| u8 *packed_data = NULL; |
| u32 packed_size = sizeof(union param_hdrs) + param_hdr.param_size; |
| int ret = 0; |
| |
| if (ac == NULL) { |
| pr_err("%s: Audio Client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| packed_data = kzalloc(packed_size, GFP_KERNEL); |
| if (packed_data == NULL) |
| return -ENOMEM; |
| |
| ret = q6common_pack_pp_params(packed_data, ¶m_hdr, param_data, |
| &packed_size); |
| if (ret) { |
| pr_err("%s: Failed to pack params, error %d\n", __func__, ret); |
| goto done; |
| } |
| |
| ret = q6asm_set_pp_params(ac, NULL, packed_data, packed_size); |
| done: |
| kfree(packed_data); |
| return ret; |
| } |
| EXPORT_SYMBOL(q6asm_pack_and_set_pp_param_in_band); |
| |
| /** |
| * q6asm_set_soft_volume_module_instance_ids |
| * command to set module and instance ids for soft volume |
| * |
| * @instance: soft volume instance |
| * @param_hdr: parameter header |
| * |
| * Returns 0 on success or error on failure |
| */ |
| int q6asm_set_soft_volume_module_instance_ids(int instance, |
| struct param_hdr_v3 *param_hdr) |
| { |
| if (param_hdr == NULL) { |
| pr_err("%s: Param header is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| switch (instance) { |
| case SOFT_VOLUME_INSTANCE_2: |
| param_hdr->module_id = ASM_MODULE_ID_VOL_CTRL2; |
| param_hdr->instance_id = INSTANCE_ID_0; |
| return 0; |
| case SOFT_VOLUME_INSTANCE_1: |
| param_hdr->module_id = ASM_MODULE_ID_VOL_CTRL; |
| param_hdr->instance_id = INSTANCE_ID_0; |
| return 0; |
| default: |
| pr_err("%s: Invalid instance %d\n", __func__, instance); |
| return -EINVAL; |
| } |
| } |
| EXPORT_SYMBOL(q6asm_set_soft_volume_module_instance_ids); |
| |
| /** |
| * q6asm_open_read_compressed - |
| * command to open ASM in compressed read mode |
| * |
| * @ac: Audio client handle |
| * @format: capture format for ASM |
| * @passthrough_flag: flag to indicate passthrough option |
| * |
| * Returns 0 on success or error on failure |
| */ |
| int q6asm_open_read_compressed(struct audio_client *ac, uint32_t format, |
| uint32_t passthrough_flag) |
| { |
| int rc = 0; |
| struct asm_stream_cmd_open_read_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]\n", __func__, ac->session, |
| format); |
| |
| q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE); |
| open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_COMPRESSED; |
| atomic_set(&ac->cmd_state, -1); |
| |
| /* |
| * Below flag indicates whether DSP shall keep IEC61937 packing or |
| * unpack to raw compressed format |
| */ |
| if (format == FORMAT_IEC61937) { |
| open.mode_flags = 0x1; |
| pr_debug("%s: Flag 1 IEC61937 output\n", __func__); |
| } else { |
| open.mode_flags = 0; |
| open.frames_per_buf = 1; |
| pr_debug("%s: Flag 0 RAW_COMPR output\n", __func__); |
| } |
| |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| if (!rc) { |
| pr_err("%s: timeout. waited for OPEN_READ_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_read_compressed); |
| |
| 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, uint32_t enc_cfg_id) |
| { |
| 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; |
| case FORMAT_BESPOKE: |
| open.mode_flags |= BUFFER_META_ENABLE; |
| open.enc_cfg_id = enc_cfg_id; |
| if (ts_mode) |
| open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE; |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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*/, ENC_CFG_ID_NONE); |
| } |
| 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*/, ENC_CFG_ID_NONE); |
| } |
| |
| /* |
| * 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*/, ENC_CFG_ID_NONE); |
| } |
| 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, |
| uint32_t enc_cfg_id) |
| { |
| return __q6asm_open_read(ac, format, bits_per_sample, |
| PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/, |
| ts_mode, enc_cfg_id); |
| } |
| 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, enc_cfg_id); |
| } |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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(&buf_circ->dma_buf, |
| bytes_to_alloc, |
| &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(&buf_pos->dma_buf, |
| bytes_to_alloc, |
| &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) |
| * use_default_chmap: true if default channel map to be used |
| * channel_map: input channel map |
| * returns 0 if successful, error code otherwise |
| */ |
| int q6asm_open_shared_io(struct audio_client *ac, |
| struct shared_io_config *config, |
| int dir, bool use_default_chmap, u8 *channel_map) |
| { |
| 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 (!use_default_chmap && (channel_map == NULL)) { |
| pr_err("%s: No valid chan map and can't use default\n", |
| __func__); |
| 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); |
| |
| if (use_default_chmap) { |
| rc = q6asm_map_channels(channel_mapping, config->channels, |
| false); |
| if (rc) { |
| pr_err("%s: Map channels failed, ret: %d\n", |
| __func__, rc); |
| goto done; |
| } |
| } else { |
| memcpy(channel_mapping, channel_map, |
| PCM_FORMAT_MAX_NUM_CHANNEL); |
| } |
| |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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->dma_buf); |
| port->buf->dma_buf = 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.dma_buf); |
| ac->shared_pos_buf.dma_buf = 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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_custom - |
| * command to set encode cfg block for custom |
| * |
| * @ac: Audio client handle |
| * @sample_rate: Sample rate |
| * @channels: number of ASM channels |
| * @format: custom format flag |
| * @cfg: generic encoder config |
| * |
| * Returns 0 on success or error on failure |
| */ |
| int q6asm_enc_cfg_blk_custom(struct audio_client *ac, |
| uint32_t sample_rate, uint32_t channels, |
| uint32_t format, void *cfg) |
| { |
| struct asm_custom_enc_cfg_t_v2 enc_cfg; |
| int rc = 0; |
| uint32_t custom_size; |
| struct snd_enc_generic *enc_generic = (struct snd_enc_generic *) cfg; |
| |
| custom_size = enc_generic->reserved[1]; |
| |
| if (channels > PCM_FORMAT_MAX_NUM_CHANNEL) { |
| pr_err("%s: Invalid channel count %d\n", __func__, channels); |
| return -EINVAL; |
| } |
| |
| pr_debug("%s: session[%d] size[%d] res[2]=[%d] res[3]=[%d]\n", |
| __func__, ac->session, custom_size, enc_generic->reserved[2], |
| enc_generic->reserved[3]); |
| |
| pr_debug("%s: res[4]=[%d] sr[%d] ch[%d] format[%d]\n", |
| __func__, enc_generic->reserved[4], sample_rate, |
| channels, format); |
| |
| memset(&enc_cfg, 0, sizeof(struct asm_custom_enc_cfg_t_v2)); |
| 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_custom_enc_cfg_t_v2) - |
| sizeof(struct asm_stream_cmd_set_encdec_param); |
| enc_cfg.encblk.frames_per_buf = ENC_FRAMES_PER_BUFFER; |
| 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.sample_rate = sample_rate; |
| |
| if (q6asm_map_channels(enc_cfg.channel_mapping, channels, false)) { |
| pr_err("%s: map channels failed %d\n", |
| __func__, channels); |
| rc = -EINVAL; |
| goto fail_cmd; |
| } |
| |
| if (format == FORMAT_BESPOKE && custom_size && |
| custom_size <= sizeof(enc_cfg.custom_data)) { |
| memcpy(enc_cfg.custom_data, &enc_generic->reserved[2], |
| custom_size); |
| enc_cfg.custom_size = custom_size; |
| } |
| |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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_custom); |
| |
| /** |
| * 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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); |
| |
| /* |
| * q6asm_map_channels: |
| * Provide default asm channel mapping for given channel count. |
| * |
| * @channel_mapping: buffer pointer to write back channel maps. |
| * @channels: channel count for which channel map is required. |
| * @use_back_flavor: use back channels instead of surround channels. |
| * Returns 0 for success, -EINVAL for unsupported channel count. |
| */ |
| 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 { |
| pr_err("%s: ERROR.unsupported num_ch = %u\n", |
| __func__, channels); |
| return -EINVAL; |
| } |
| return 0; |
| } |
| EXPORT_SYMBOL(q6asm_map_channels); |
| |
| /** |
| * 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| |
| memset(¶m_info, 0, sizeof(param_info)); |
| memset(&multi_ch_gain, 0, sizeof(multi_ch_gain)); |
| |
| param_info.module_id = ASM_MODULE_ID_VOL_CTRL; |
| param_info.instance_id = INSTANCE_ID_0; |
| param_info.param_id = ASM_PARAM_ID_MULTICHANNEL_GAIN; |
| param_info.param_size = sizeof(multi_ch_gain); |
| |
| 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 = q6asm_pack_and_set_pp_param_in_band(ac, param_info, |
| (u8 *) &multi_ch_gain); |
| if (rc < 0) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| int i; |
| u8 default_chmap[VOLUME_CONTROL_MAX_CHANNELS]; |
| |
| if (ac == NULL) { |
| pr_err("%s: Audio client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| if (gains == NULL) { |
| dev_err(ac->dev, "%s: gain_list is NULL\n", __func__); |
| rc = -EINVAL; |
| goto done; |
| } |
| if (channels > VOLUME_CONTROL_MAX_CHANNELS) { |
| dev_err(ac->dev, "%s: Invalid channel count %d\n", |
| __func__, channels); |
| rc = -EINVAL; |
| goto done; |
| } |
| if (!use_default && ch_map == NULL) { |
| dev_err(ac->dev, "%s: NULL channel map\n", __func__); |
| rc = -EINVAL; |
| goto done; |
| } |
| |
| memset(¶m_info, 0, sizeof(param_info)); |
| memset(&multich_gain, 0, sizeof(multich_gain)); |
| param_info.module_id = ASM_MODULE_ID_VOL_CTRL; |
| param_info.instance_id = INSTANCE_ID_0; |
| param_info.param_id = ASM_PARAM_ID_MULTICHANNEL_GAIN; |
| param_info.param_size = sizeof(multich_gain); |
| |
| if (use_default) { |
| rc = q6asm_map_channels(default_chmap, channels, false); |
| if (rc < 0) |
| goto done; |
| 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 = q6asm_pack_and_set_pp_param_in_band(ac, param_info, |
| (u8 *) &multich_gain); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| |
| memset(&mute, 0, sizeof(mute)); |
| memset(¶m_info, 0, sizeof(param_info)); |
| param_info.module_id = ASM_MODULE_ID_VOL_CTRL; |
| param_info.instance_id = INSTANCE_ID_0; |
| param_info.param_id = ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG; |
| param_info.param_size = sizeof(mute); |
| mute.mute_flag = muteflag; |
| |
| rc = q6asm_pack_and_set_pp_param_in_band(ac, param_info, (u8 *) &mute); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| |
| memset(&vol, 0, sizeof(vol)); |
| memset(¶m_info, 0, sizeof(param_info)); |
| |
| rc = q6asm_set_soft_volume_module_instance_ids(instance, ¶m_info); |
| if (rc) { |
| pr_err("%s: Failed to pack soft volume module and instance IDs, error %d\n", |
| __func__, rc); |
| return rc; |
| } |
| |
| param_info.param_id = ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN; |
| param_info.param_size = sizeof(vol); |
| vol.master_gain = volume; |
| |
| rc = q6asm_pack_and_set_pp_param_in_band(ac, param_info, (u8 *) &vol); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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 dma_buf *dma_buf; |
| dma_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(&dma_buf, |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| |
| memset(&softpause, 0, sizeof(softpause)); |
| memset(¶m_info, 0, sizeof(param_info)); |
| param_info.module_id = ASM_MODULE_ID_VOL_CTRL; |
| param_info.instance_id = INSTANCE_ID_0; |
| param_info.param_id = ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS; |
| param_info.param_size = sizeof(softpause); |
| |
| softpause.enable_flag = pause_param->enable; |
| softpause.period = pause_param->period; |
| softpause.step = pause_param->step; |
| softpause.ramping_curve = pause_param->rampingcurve; |
| |
| rc = q6asm_pack_and_set_pp_param_in_band(ac, param_info, |
| (u8 *) &softpause); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| |
| 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; |
| struct param_hdr_v3 param_info; |
| int rc = 0; |
| |
| memset(&softvol, 0, sizeof(softvol)); |
| memset(¶m_info, 0, sizeof(param_info)); |
| |
| rc = q6asm_set_soft_volume_module_instance_ids(instance, ¶m_info); |
| if (rc) { |
| pr_err("%s: Failed to pack soft volume module and instance IDs, error %d\n", |
| __func__, rc); |
| return rc; |
| } |
| |
| param_info.param_id = ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS; |
| param_info.param_size = sizeof(softvol); |
| |
| softvol.period = softvol_param->period; |
| softvol.step = softvol_param->step; |
| softvol.ramping_curve = softvol_param->rampingcurve; |
| |
| rc = q6asm_pack_and_set_pp_param_in_band(ac, param_info, |
| (u8 *) &softvol); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| |
| 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; |
| struct param_hdr_v3 param_info; |
| int i = 0; |
| int rc = 0; |
| |
| if (ac == NULL) { |
| pr_err("%s: Audio client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| if (eq_p == NULL) { |
| pr_err("%s: [%d]: Invalid Eq param\n", __func__, ac->session); |
| rc = -EINVAL; |
| goto fail_cmd; |
| } |
| |
| memset(&eq, 0, sizeof(eq)); |
| memset(¶m_info, 0, sizeof(param_info)); |
| eq_params = (struct msm_audio_eq_stream_config *) eq_p; |
| param_info.module_id = ASM_MODULE_ID_EQUALIZER; |
| param_info.instance_id = INSTANCE_ID_0; |
| param_info.param_id = ASM_PARAM_ID_EQUALIZER_PARAMETERS; |
| param_info.param_size = sizeof(eq); |
| 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 = q6asm_pack_and_set_pp_param_in_band(ac, param_info, (u8 *) &eq); |
| if (rc) |
| pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", |
| __func__, param_info.param_id, rc); |
| |
| fail_cmd: |
| 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_ratelimited("%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, ¶m->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_ratelimited("%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 param_hdr_v1) + |
| 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) { |
| dev_err_ratelimited(ac->dev, "%s: Get Session Time failed %d\n", |
| __func__, rc); |
| return rc; |
| } |
| |
| rc = wait_event_timeout(ac->time_wait, |
| (atomic_read(&ac->time_flag) == 0), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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_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 param_hdr_v1) + |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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 param_hdr_v1) + |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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 param_hdr_v1) + |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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 param_hdr_v1) + |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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_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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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), |
| msecs_to_jiffies(TIMEOUT_MS)); |
| 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 mem_mapping_hdr mem_hdr; |
| u32 payload_size = 0; |
| int rc = -EINVAL; |
| pr_debug("%s:\n", __func__); |
| |
| if (!ac) { |
| pr_err("%s: Audio client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| if (ac->io_mode & NT_MODE) { |
| pr_debug("%s: called for NT MODE, exiting\n", __func__); |
| goto done; |
| } |
| |
| if (cal_data[ASM_AUDSTRM_CAL] == NULL) |
| goto done; |
| |
| if (ac->perf_mode == ULTRA_LOW_LATENCY_PCM_MODE) { |
| rc = 0; /* no cal is required, not error case */ |
| goto done; |
| } |
| |
| memset(&mem_hdr, 0, sizeof(mem_hdr)); |
| 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) { |
| pr_err("%s: cal_block is NULL\n", |
| __func__); |
| goto unlock; |
| } |
| |
| if (cal_utils_is_cal_stale(cal_block)) { |
| rc = 0; /* not error case */ |
| pr_debug("%s: cal_block is 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; |
| } |
| |
| mem_hdr.data_payload_addr_lsw = |
| lower_32_bits(cal_block->cal_data.paddr); |
| mem_hdr.data_payload_addr_msw = |
| msm_audio_populate_upper_32_bits(cal_block->cal_data.paddr); |
| mem_hdr.mem_map_handle = cal_block->map_data.q6map_handle; |
| payload_size = cal_block->cal_data.size; |
| |
| pr_debug("%s: phyaddr lsw = %x msw = %x, maphdl = %x calsize = %d\n", |
| __func__, mem_hdr.data_payload_addr_lsw, |
| mem_hdr.data_payload_addr_msw, mem_hdr.mem_map_handle, |
| payload_size); |
| |
| rc = q6asm_set_pp_params(ac, &mem_hdr, NULL, payload_size); |
| if (rc) { |
| pr_err("%s: audio audstrm cal send failed\n", __func__); |
| goto unlock; |
| } |
| |
| if (cal_block) |
| cal_utils_mark_cal_used(cal_block); |
| rc = 0; |
| |
| 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(); |
| } |