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体验新版 GitCode,发现更多精彩内容 >>
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fcfa986b
编写于
8月 08, 2019
作者:
B
Bernard Xiong
提交者:
GitHub
8月 08, 2019
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差异文件
Merge pull request #2928 from Zero-Free/audio_developer
添加qemu和stm32l475-atk-pandora对音频设备驱动框架的支持
上级
15e01877
baac77c8
变更
21
展开全部
隐藏空白更改
内联
并排
Showing
21 changed file
with
1923 addition
and
808 deletion
+1923
-808
bsp/qemu-vexpress-a9/drivers/audio/audio_device.c
bsp/qemu-vexpress-a9/drivers/audio/audio_device.c
+0
-191
bsp/qemu-vexpress-a9/drivers/audio/audio_device.h
bsp/qemu-vexpress-a9/drivers/audio/audio_device.h
+0
-51
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.c
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.c
+0
-328
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.h
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.h
+0
-29
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.c
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.c
+1
-1
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.h
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.h
+1
-1
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.c
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.c
+342
-0
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.h
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.h
+15
-0
bsp/qemu-vexpress-a9/drivers/audio/wav_play.c
bsp/qemu-vexpress-a9/drivers/audio/wav_play.c
+0
-144
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/.mxproject
...tm32/stm32l475-atk-pandora/board/CubeMX_Config/.mxproject
+5
-5
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Inc/stm32l4xx_hal_conf.h
...-atk-pandora/board/CubeMX_Config/Inc/stm32l4xx_hal_conf.h
+1
-1
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/STM32L475VE.ioc
...stm32l475-atk-pandora/board/CubeMX_Config/STM32L475VE.ioc
+89
-53
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Src/main.c
...tm32/stm32l475-atk-pandora/board/CubeMX_Config/Src/main.c
+126
-4
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Src/stm32l4xx_hal_msp.c
...5-atk-pandora/board/CubeMX_Config/Src/stm32l4xx_hal_msp.c
+91
-0
bsp/stm32/stm32l475-atk-pandora/board/Kconfig
bsp/stm32/stm32l475-atk-pandora/board/Kconfig
+17
-0
bsp/stm32/stm32l475-atk-pandora/board/SConscript
bsp/stm32/stm32l475-atk-pandora/board/SConscript
+10
-0
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.c
...tm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.c
+287
-0
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.h
...tm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.h
+95
-0
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_mic.c
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_mic.c
+369
-0
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.c
...stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.c
+458
-0
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.h
...stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.h
+16
-0
未找到文件。
bsp/qemu-vexpress-a9/drivers/audio/audio_device.c
已删除
100644 → 0
浏览文件 @
15e01877
/*
* File : audio_device.c
* This file is part of RT-Thread RTOS
* COPYRIGHT (C) 2017, RT-Thread Development Team
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Change Logs:
* Date Author Notes
* 2018-05-26 RT-Thread the first version
*/
#include <rtthread.h>
#include <rthw.h>
#include <rtdevice.h>
#include <string.h>
#include "drv_pl041.h"
#include "drv_ac97.h"
#include "audio_device.h"
struct
audio_device
{
struct
rt_device
*
snd
;
struct
rt_mempool
mp
;
int
state
;
void
(
*
evt_handler
)(
void
*
parameter
,
int
state
);
void
*
parameter
;
};
static
struct
audio_device
*
_audio_device
=
NULL
;
void
*
audio_device_get_buffer
(
int
*
bufsz
)
{
if
(
bufsz
)
{
*
bufsz
=
AUDIO_DEVICE_DECODE_MP_BLOCK_SZ
*
2
;
}
return
rt_mp_alloc
(
&
(
_audio_device
->
mp
),
RT_WAITING_FOREVER
);
}
void
audio_device_put_buffer
(
void
*
ptr
)
{
if
(
ptr
)
rt_mp_free
(
ptr
);
return
;
}
static
rt_err_t
audio_device_write_done
(
struct
rt_device
*
device
,
void
*
ptr
)
{
if
(
!
ptr
)
{
rt_kprintf
(
"device buf_release NULL
\n
"
);
return
-
RT_ERROR
;
}
rt_mp_free
(
ptr
);
return
RT_EOK
;
}
void
audio_device_write
(
void
*
buffer
,
int
size
)
{
if
(
_audio_device
->
snd
&&
size
!=
0
)
{
if
(
_audio_device
->
state
==
AUDIO_DEVICE_IDLE
)
{
if
(
_audio_device
->
evt_handler
)
_audio_device
->
evt_handler
(
_audio_device
->
parameter
,
AUDIO_DEVICE_PLAYBACK
);
/* change audio device state */
_audio_device
->
state
=
AUDIO_DEVICE_PLAYBACK
;
}
rt_device_write
(
_audio_device
->
snd
,
0
,
buffer
,
size
);
}
else
{
/* release buffer directly */
rt_mp_free
(
buffer
);
}
return
;
}
int
audio_device_init
(
void
)
{
uint8_t
*
mempool_ptr
;
if
(
!
_audio_device
)
{
_audio_device
=
(
struct
audio_device
*
)
rt_malloc
(
sizeof
(
struct
audio_device
)
+
AUDIO_DEVICE_DECODE_MP_SZ
);
if
(
_audio_device
==
NULL
)
{
rt_kprintf
(
"malloc memeory for _audio_device failed!
\n
"
);
return
-
RT_ERROR
;
}
_audio_device
->
evt_handler
=
NULL
;
_audio_device
->
parameter
=
NULL
;
mempool_ptr
=
(
uint8_t
*
)(
_audio_device
+
1
);
rt_mp_init
(
&
(
_audio_device
->
mp
),
"adbuf"
,
mempool_ptr
,
AUDIO_DEVICE_DECODE_MP_SZ
,
AUDIO_DEVICE_DECODE_MP_BLOCK_SZ
*
2
);
/* find snd device */
_audio_device
->
snd
=
rt_device_find
(
"sound"
);
if
(
_audio_device
->
snd
==
NULL
)
{
rt_kprintf
(
"sound device not found
\n
"
);
return
-
1
;
}
/* set tx complete call back function */
rt_device_set_tx_complete
(
_audio_device
->
snd
,
audio_device_write_done
);
}
return
RT_EOK
;
}
int
audio_device_set_evt_handler
(
void
(
*
handler
)(
void
*
parameter
,
int
state
),
void
*
parameter
)
{
if
(
_audio_device
)
{
_audio_device
->
evt_handler
=
handler
;
_audio_device
->
parameter
=
parameter
;
}
return
0
;
}
void
audio_device_set_rate
(
int
sample_rate
)
{
if
(
_audio_device
->
snd
)
{
int
rate
=
sample_rate
;
rt_device_control
(
_audio_device
->
snd
,
CODEC_CMD_SAMPLERATE
,
&
rate
);
}
}
void
audio_device_set_volume
(
int
value
)
{
if
(
_audio_device
->
snd
)
{
rt_device_control
(
_audio_device
->
snd
,
CODEC_CMD_SET_VOLUME
,
&
value
);
}
}
int
audio_device_get_volume
(
void
)
{
int
value
=
0
;
if
(
_audio_device
->
snd
)
{
rt_device_control
(
_audio_device
->
snd
,
CODEC_CMD_GET_VOLUME
,
&
value
);
}
return
value
;
}
void
audio_device_open
(
void
)
{
_audio_device
->
state
=
AUDIO_DEVICE_IDLE
;
rt_device_open
(
_audio_device
->
snd
,
RT_DEVICE_OFLAG_WRONLY
);
}
void
audio_device_close
(
void
)
{
rt_device_close
(
_audio_device
->
snd
);
if
(
_audio_device
->
state
==
AUDIO_DEVICE_PLAYBACK
)
{
if
(
_audio_device
->
evt_handler
)
_audio_device
->
evt_handler
(
_audio_device
->
parameter
,
AUDIO_DEVICE_CLOSE
);
}
/* set to idle */
_audio_device
->
state
=
AUDIO_DEVICE_CLOSE
;
}
bsp/qemu-vexpress-a9/drivers/audio/audio_device.h
已删除
100644 → 0
浏览文件 @
15e01877
/*
* File : audio_device.h
* This file is part of RT-Thread RTOS
* COPYRIGHT (C) 2017, RT-Thread Development Team
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Change Logs:
* Date Author Notes
* 2018-05-26 RT-Thread the first version
*/
#ifndef AUDIO_DEVICE_H__
#define AUDIO_DEVICE_H__
enum
AUDIO_DEVICE_STATE
{
AUDIO_DEVICE_IDLE
,
AUDIO_DEVICE_PLAYBACK
,
AUDIO_DEVICE_CLOSE
,
};
void
*
audio_device_get_buffer
(
int
*
bufsz
);
void
audio_device_put_buffer
(
void
*
ptr
);
void
audio_device_write
(
void
*
buffer
,
int
size
);
int
audio_device_init
(
void
);
void
audio_device_close
(
void
);
void
audio_device_open
(
void
);
int
audio_device_set_evt_handler
(
void
(
*
handler
)(
void
*
parameter
,
int
state
),
void
*
parameter
);
void
audio_device_set_rate
(
int
sample_rate
);
void
audio_device_set_volume
(
int
volume
);
void
audio_device_wait_free
(
void
);
#endif
\ No newline at end of file
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.c
已删除
100644 → 0
浏览文件 @
15e01877
/*
* File : drv_audio.c
* This file is part of RT-Thread RTOS
* COPYRIGHT (C) 2017, RT-Thread Development Team
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Change Logs:
* Date Author Notes
* 2018-05-26 RT-Thread the first version
*/
#include <rtthread.h>
#include <rthw.h>
#include <rtdevice.h>
#include <string.h>
#include "drv_pl041.h"
#include "drv_ac97.h"
#include "drv_audio.h"
#define DATA_NODE_MAX (10)
#define CODEC_TX_FIFO_SIZE (256)
#define AUDIO_DEVICE_DECODE_MP_SIZE (4096)
#define AUDIO_DEVICE_DECODE_MP_CONUT (4)
struct
codec_data_node
{
char
*
data_ptr
;
rt_size_t
data_size
;
};
struct
audio_buff_des
{
struct
codec_data_node
*
data_list
;
void
(
*
free_fun
)(
void
*
);
rt_uint32_t
read_offset
;
rt_uint16_t
node_num
;
rt_uint16_t
read_index
,
put_index
;
};
struct
audio_device
{
/* inherit from rt_device */
struct
rt_device
parent
;
};
static
struct
audio_device
audio_device_drive
;
static
struct
audio_buff_des
*
audio_buff
;
static
int
irq_flag
=
0
;
static
void
_audio_buff_cb
(
void
*
buff
)
{
if
(
audio_device_drive
.
parent
.
tx_complete
!=
RT_NULL
)
{
audio_device_drive
.
parent
.
tx_complete
(
&
audio_device_drive
.
parent
,
buff
);
}
}
static
rt_size_t
_audio_buff_push
(
struct
audio_buff_des
*
hdle
,
void
*
buff
,
int
size
)
{
struct
codec_data_node
*
node
;
rt_uint16_t
next_index
;
rt_uint32_t
level
;
if
((
buff
==
RT_NULL
)
||
(
size
==
0
))
{
return
0
;
}
next_index
=
hdle
->
put_index
+
1
;
if
(
next_index
>=
hdle
->
node_num
)
next_index
=
0
;
/* check data_list full */
if
(
next_index
==
hdle
->
read_index
)
{
rt_kprintf
(
"data_list full
\n
"
);
rt_set_errno
(
-
RT_EFULL
);
return
0
;
}
level
=
rt_hw_interrupt_disable
();
node
=
&
hdle
->
data_list
[
hdle
->
put_index
];
hdle
->
put_index
=
next_index
;
/* set node attribute */
node
->
data_ptr
=
(
char
*
)
buff
;
node
->
data_size
=
size
;
rt_hw_interrupt_enable
(
level
);
return
size
;
}
static
rt_size_t
_audio_buff_pop
(
struct
audio_buff_des
*
hdle
,
void
*
buff
,
int
size
)
{
struct
codec_data_node
*
node
;
rt_uint32_t
next_index
,
count
=
0
,
cp_size
=
0
,
offset
=
0
;
node
=
&
hdle
->
data_list
[
hdle
->
read_index
];
if
((
hdle
->
read_index
==
hdle
->
put_index
)
&&
(
node
->
data_ptr
==
RT_NULL
))
{
memset
(
buff
,
0xff
,
size
);
return
0
;
}
while
(
count
<
size
)
{
node
=
&
hdle
->
data_list
[
hdle
->
read_index
];
offset
=
hdle
->
read_offset
;
cp_size
=
(
node
->
data_size
-
offset
)
>
(
size
-
count
)
?
(
size
-
count
)
:
(
node
->
data_size
-
offset
);
if
(
node
->
data_ptr
==
RT_NULL
)
{
memset
(
buff
,
0
,
size
-
count
);
return
count
;
}
memcpy
((
rt_uint8_t
*
)
buff
+
count
,
(
rt_uint8_t
*
)(
node
->
data_ptr
)
+
offset
,
cp_size
);
hdle
->
read_offset
+=
cp_size
;
count
+=
cp_size
;
if
(
hdle
->
read_offset
>=
node
->
data_size
)
{
/* notify transmitted complete. */
if
(
hdle
->
free_fun
!=
RT_NULL
)
{
hdle
->
free_fun
(
node
->
data_ptr
);
}
/* clear current node */
memset
(
node
,
0
,
sizeof
(
struct
codec_data_node
));
next_index
=
hdle
->
read_index
+
1
;
if
(
next_index
>=
hdle
->
node_num
)
{
next_index
=
0
;
}
hdle
->
read_offset
=
0
;
hdle
->
read_index
=
next_index
;
}
}
return
count
;
}
static
void
transit_wav_data
(
rt_uint32_t
status
)
{
rt_uint16_t
sample
[
CODEC_TX_FIFO_SIZE
];
int
i
=
0
,
size
;
size
=
_audio_buff_pop
(
audio_buff
,
sample
,
CODEC_TX_FIFO_SIZE
*
sizeof
(
rt_uint16_t
));
if
((
size
==
0
)
&&
(
irq_flag
==
1
))
{
aaci_pl041_irq_disable
(
0
,
AACI_IE_UR
|
AACI_IE_TX
|
AACI_IE_TXC
);
irq_flag
=
0
;
}
for
(
i
=
0
;
i
<
(
size
>>
1
);
i
++
)
{
aaci_pl041_channle_write
(
0
,
&
sample
[
i
],
1
);
}
}
static
void
rt_hw_aaci_isr
(
rt_uint32_t
status
,
void
*
user_data
)
{
if
(
status
&
AACI_SR_TXHE
)
{
transit_wav_data
(
status
);
}
}
static
rt_err_t
codec_init
(
rt_device_t
dev
)
{
struct
pl041_cfg
_cfg
;
_cfg
.
itype
=
PL041_CHANNLE_LEFT_ADC
|
PL041_CHANNLE_RIGHT_ADC
;
_cfg
.
otype
=
PL041_CHANNLE_LEFT_DAC
|
PL041_CHANNLE_RIGHT_DAC
;
_cfg
.
vol
=
50
;
_cfg
.
rate
=
8000
;
ac97_reset
();
aaci_pl041_channle_cfg
(
0
,
&
_cfg
);
aaci_pl041_irq_register
(
0
,
rt_hw_aaci_isr
,
RT_NULL
);
return
RT_EOK
;
}
static
rt_err_t
codec_open
(
rt_device_t
dev
,
rt_uint16_t
oflag
)
{
return
RT_EOK
;
}
static
rt_err_t
codec_close
(
rt_device_t
dev
)
{
rt_uint16_t
temp
=
0
,
i
=
1024
*
10
;
while
(
PL041
->
sr1
&
AACI_SR_TXB
);
while
(
i
)
{
if
(
aaci_pl041_channle_write
(
0
,
&
temp
,
1
)
!=
0
)
{
i
--
;
}
}
return
RT_EOK
;
}
static
rt_size_t
codec_read
(
rt_device_t
dev
,
rt_off_t
pos
,
void
*
buffer
,
rt_size_t
size
)
{
return
0
;
}
static
rt_size_t
codec_write
(
rt_device_t
dev
,
rt_off_t
pos
,
const
void
*
buffer
,
rt_size_t
size
)
{
_audio_buff_push
(
audio_buff
,
(
void
*
)
buffer
,
size
);
if
(
irq_flag
==
0
)
{
//open irq
irq_flag
=
1
;
aaci_pl041_channle_enable
(
0
);
aaci_pl041_irq_enable
(
0
,
AACI_IE_UR
|
AACI_IE_TX
|
AACI_IE_TXC
);
}
return
0
;
}
static
rt_err_t
codec_control
(
rt_device_t
dev
,
int
cmd
,
void
*
args
)
{
rt_err_t
result
=
RT_EOK
;
switch
(
cmd
)
{
case
CODEC_CMD_RESET
:
{
break
;
}
case
CODEC_CMD_SET_VOLUME
:
{
uint32_t
v
;
v
=
*
(
rt_uint32_t
*
)
args
;
result
=
ac97_set_vol
(
v
);
break
;
}
case
CODEC_CMD_GET_VOLUME
:
{
int
*
v
=
args
;
*
v
=
ac97_get_vol
();
break
;
}
case
CODEC_CMD_SAMPLERATE
:
{
int
v
;
v
=
*
(
rt_uint32_t
*
)
args
;
ac97_set_rate
(
v
);
break
;
}
default:
result
=
RT_ERROR
;
}
return
result
;
}
#ifdef RT_USING_DEVICE_OPS
const
static
struct
rt_device_ops
codec_ops
=
{
codec_init
,
codec_open
,
codec_close
,
codec_read
,
codec_write
,
codec_control
};
#endif
int
audio_hw_init
(
void
)
{
struct
audio_device
*
codec
=
&
audio_device_drive
;
codec
->
parent
.
type
=
RT_Device_Class_Sound
;
codec
->
parent
.
rx_indicate
=
RT_NULL
;
codec
->
parent
.
tx_complete
=
RT_NULL
;
#ifdef RT_USING_DEVICE_OPS
codec
->
parent
.
ops
=
&
codec_ops
;
#else
codec
->
parent
.
init
=
codec_init
;
codec
->
parent
.
open
=
codec_open
;
codec
->
parent
.
close
=
codec_close
;
codec
->
parent
.
read
=
codec_read
;
codec
->
parent
.
write
=
codec_write
;
codec
->
parent
.
control
=
codec_control
;
#endif
codec
->
parent
.
user_data
=
RT_NULL
;
audio_buff
=
rt_malloc
(
sizeof
(
struct
audio_buff_des
)
+
sizeof
(
struct
codec_data_node
)
*
DATA_NODE_MAX
);
if
(
audio_buff
==
RT_NULL
)
{
rt_kprintf
(
"audio buff malloc fail
\n
"
);
return
-
1
;
}
rt_memset
(
audio_buff
,
0
,
sizeof
(
struct
audio_buff_des
)
+
sizeof
(
struct
codec_data_node
)
*
DATA_NODE_MAX
);
audio_buff
->
data_list
=
(
struct
codec_data_node
*
)((
rt_uint8_t
*
)
audio_buff
+
sizeof
(
struct
audio_buff_des
));
audio_buff
->
free_fun
=
_audio_buff_cb
;
audio_buff
->
node_num
=
DATA_NODE_MAX
;
/* register the device */
rt_device_register
(
&
codec
->
parent
,
"sound"
,
RT_DEVICE_FLAG_WRONLY
|
RT_DEVICE_FLAG_DMA_TX
);
aaci_pl041_init
();
rt_device_init
(
&
codec
->
parent
);
return
0
;
}
INIT_DEVICE_EXPORT
(
audio_hw_init
);
bsp/qemu-vexpress-a9/drivers/audio/drv_audio.h
已删除
100644 → 0
浏览文件 @
15e01877
/*
* File : drv_audio.h
* This file is part of RT-Thread RTOS
* COPYRIGHT (C) 2017, RT-Thread Development Team
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Change Logs:
* Date Author Notes
* 2018-05-26 RT-Thread the first version
*/
#ifndef __DRV_AUDIO_H__
#define __DRV_AUDIO_H__
#endif
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.c
浏览文件 @
fcfa986b
...
...
@@ -302,7 +302,7 @@ static void aaci_pl041_irq_handle(int irqno, void *param)
void
*
p_status
;
mask
=
PL041_READ
(
&
PL041
->
allints
);
PL041_WRITE
(
PL041
->
intclr
,
mask
);
PL041_WRITE
(
&
PL041
->
intclr
,
mask
);
for
(
channle
=
0
;
(
channle
<
PL041_CHANNLE_NUM
)
&&
(
mask
);
channle
++
)
{
...
...
bsp/qemu-vexpress-a9/drivers/audio/drv_pl041.h
浏览文件 @
fcfa986b
...
...
@@ -225,7 +225,7 @@ typedef void (*pl041_irq_fun_t)(rt_uint32_t status, void * user_data);
rt_err_t
aaci_pl041_init
(
void
);
void
aaci_ac97_write
(
rt_uint16_t
reg
,
rt_uint16_t
val
);
rt_uint16_t
aaci_ac97_read
(
rt_uint16_t
reg
);
int
aaci_pl041_channle_cfg
(
int
channle
,
pl041_cfg_t
c
gf
);
int
aaci_pl041_channle_cfg
(
int
channle
,
pl041_cfg_t
c
fg
);
int
aaci_pl041_channle_write
(
int
channle
,
rt_uint16_t
*
buff
,
int
count
);
int
aaci_pl041_channle_read
(
int
channle
,
rt_uint16_t
*
buff
,
int
count
);
int
aaci_pl041_channle_enable
(
int
channle
);
...
...
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.c
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-23 Zero-Free first implementation
*/
#include <rtthread.h>
#include <rtdevice.h>
#include <drv_sound.h>
#include <drv_pl041.h>
#include <drv_ac97.h>
#define DBG_TAG "drv.sound"
#define DBG_LVL DBG_INFO
#include <rtdbg.h>
#define TX_FIFO_SIZE (3840)
struct
sound_device
{
struct
rt_audio_device
audio
;
struct
rt_audio_configure
replay_config
;
rt_uint8_t
*
tx_fifo
;
rt_uint8_t
volume
;
};
static
struct
sound_device
snd_dev
=
{
0
};
static
void
rt_hw_aaci_isr
(
rt_uint32_t
status
,
void
*
user_data
)
{
if
(
status
&
AACI_SR_TXHE
)
{
rt_audio_tx_complete
(
&
snd_dev
.
audio
);
}
}
static
rt_err_t
sound_getcaps
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_QUERY
:
/* qurey the types of hw_codec device */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_TYPE_QUERY
:
caps
->
udata
.
mask
=
AUDIO_TYPE_OUTPUT
|
AUDIO_TYPE_MIXER
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_OUTPUT
:
/* Provide capabilities of OUTPUT unit */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
caps
->
udata
.
config
.
samplerate
=
snd_dev
->
replay_config
.
samplerate
;
caps
->
udata
.
config
.
channels
=
snd_dev
->
replay_config
.
channels
;
caps
->
udata
.
config
.
samplebits
=
snd_dev
->
replay_config
.
samplebits
;
break
;
case
AUDIO_DSP_SAMPLERATE
:
caps
->
udata
.
config
.
samplerate
=
snd_dev
->
replay_config
.
samplerate
;
break
;
case
AUDIO_DSP_CHANNELS
:
caps
->
udata
.
config
.
channels
=
snd_dev
->
replay_config
.
channels
;
break
;
case
AUDIO_DSP_SAMPLEBITS
:
caps
->
udata
.
config
.
samplebits
=
snd_dev
->
replay_config
.
samplebits
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_MIXER
:
/* report the Mixer Units */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_QUERY
:
caps
->
udata
.
mask
=
AUDIO_MIXER_VOLUME
;
break
;
case
AUDIO_MIXER_VOLUME
:
caps
->
udata
.
value
=
snd_dev
->
volume
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
return
result
;
}
static
rt_err_t
sound_configure
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
struct
rt_audio_replay
*
replay
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_MIXER
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_VOLUME
:
{
rt_uint8_t
volume
=
caps
->
udata
.
value
;
snd_dev
->
volume
=
volume
;
ac97_set_vol
(
volume
);
LOG_I
(
"set volume %d"
,
volume
);
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_OUTPUT
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
{
/* set samplerate */
ac97_set_rate
(
caps
->
udata
.
config
.
samplerate
);
/* update buffer fifo informaition according samplerate */
replay
=
snd_dev
->
audio
.
replay
;
replay
->
buf_info
.
total_size
=
caps
->
udata
.
config
.
samplerate
/
50
*
4
;
replay
->
buf_info
.
block_size
=
replay
->
buf_info
.
total_size
/
2
;
/* save configs */
snd_dev
->
replay_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
snd_dev
->
replay_config
.
channels
=
caps
->
udata
.
config
.
channels
;
snd_dev
->
replay_config
.
samplebits
=
caps
->
udata
.
config
.
samplebits
;
LOG_D
(
"set samplerate %d"
,
snd_dev
->
replay_config
.
samplerate
);
break
;
}
case
AUDIO_DSP_SAMPLERATE
:
{
ac97_set_rate
(
caps
->
udata
.
config
.
samplerate
);
snd_dev
->
replay_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
LOG_D
(
"set samplerate %d"
,
snd_dev
->
replay_config
.
samplerate
);
break
;
}
case
AUDIO_DSP_CHANNELS
:
{
/* not support */
snd_dev
->
replay_config
.
channels
=
caps
->
udata
.
config
.
channels
;
LOG_D
(
"set channels %d"
,
snd_dev
->
replay_config
.
channels
);
break
;
}
case
AUDIO_DSP_SAMPLEBITS
:
{
/* not support */
snd_dev
->
replay_config
.
samplebits
=
caps
->
udata
.
config
.
samplebits
;
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
default:
break
;
}
return
result
;
}
static
rt_err_t
sound_init
(
struct
rt_audio_device
*
audio
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
struct
pl041_cfg
_cfg
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
aaci_pl041_init
();
_cfg
.
itype
=
PL041_CHANNLE_LEFT_ADC
|
PL041_CHANNLE_RIGHT_ADC
;
_cfg
.
otype
=
PL041_CHANNLE_LEFT_DAC
|
PL041_CHANNLE_RIGHT_DAC
;
_cfg
.
vol
=
snd_dev
->
volume
;
_cfg
.
rate
=
snd_dev
->
replay_config
.
samplerate
;
ac97_reset
();
aaci_pl041_channle_cfg
(
0
,
&
_cfg
);
aaci_pl041_irq_register
(
0
,
rt_hw_aaci_isr
,
RT_NULL
);
return
result
;
}
static
rt_err_t
sound_start
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
RT_ASSERT
(
audio
!=
RT_NULL
);
if
(
stream
==
AUDIO_STREAM_REPLAY
)
{
LOG_D
(
"open sound device"
);
aaci_pl041_channle_enable
(
0
);
aaci_pl041_irq_enable
(
0
,
AACI_IE_UR
|
AACI_IE_TX
|
AACI_IE_TXC
);
}
return
RT_EOK
;
}
static
rt_err_t
sound_stop
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
RT_ASSERT
(
audio
!=
RT_NULL
);
if
(
stream
==
AUDIO_STREAM_REPLAY
)
{
/* wait codec free */
rt_thread_mdelay
(
100
);
/* disable irq and channels 0 */
aaci_pl041_irq_disable
(
0
,
AACI_IE_UR
|
AACI_IE_TX
|
AACI_IE_TXC
);
aaci_pl041_channle_disable
(
0
);
LOG_D
(
"close sound device"
);
}
return
RT_EOK
;
}
static
void
sound_buffer_info
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_buf_info
*
info
)
{
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
/**
* TX_FIFO
* +----------------+----------------+
* | block1 | block2 |
* +----------------+----------------+
* \ block_size /
*/
info
->
buffer
=
snd_dev
->
tx_fifo
;
info
->
total_size
=
TX_FIFO_SIZE
;
info
->
block_size
=
TX_FIFO_SIZE
/
2
;
info
->
block_count
=
2
;
}
static
rt_size_t
sound_transmit
(
struct
rt_audio_device
*
audio
,
const
void
*
writeBuf
,
void
*
readBuf
,
rt_size_t
size
)
{
RT_ASSERT
(
audio
!=
RT_NULL
);
/* write data to channel_0 fifo */
aaci_pl041_channle_write
(
0
,
(
rt_uint16_t
*
)
writeBuf
,
size
>>
1
);
return
size
;
}
static
struct
rt_audio_ops
snd_ops
=
{
.
getcaps
=
sound_getcaps
,
.
configure
=
sound_configure
,
.
init
=
sound_init
,
.
start
=
sound_start
,
.
stop
=
sound_stop
,
.
transmit
=
sound_transmit
,
.
buffer_info
=
sound_buffer_info
,
};
int
rt_hw_audio_init
(
void
)
{
rt_uint8_t
*
tx_fifo
;
if
(
snd_dev
.
tx_fifo
)
return
RT_EOK
;
tx_fifo
=
rt_malloc
(
TX_FIFO_SIZE
);
if
(
tx_fifo
==
RT_NULL
)
return
-
RT_ENOMEM
;
rt_memset
(
tx_fifo
,
0
,
TX_FIFO_SIZE
);
snd_dev
.
tx_fifo
=
tx_fifo
;
/* init default configuration */
{
snd_dev
.
replay_config
.
samplerate
=
44100
;
snd_dev
.
replay_config
.
channels
=
2
;
snd_dev
.
replay_config
.
samplebits
=
16
;
snd_dev
.
volume
=
55
;
}
/* register sound device */
snd_dev
.
audio
.
ops
=
&
snd_ops
;
rt_audio_register
(
&
snd_dev
.
audio
,
"sound0"
,
RT_DEVICE_FLAG_WRONLY
,
&
snd_dev
);
return
RT_EOK
;
}
INIT_DEVICE_EXPORT
(
rt_hw_audio_init
);
bsp/qemu-vexpress-a9/drivers/audio/drv_sound.h
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-23 Zero-Free first implementation
*/
#ifndef __DRV_SOUND_H__
#define __DRV_SOUND_H__
int
rt_hw_audio_init
(
void
);
#endif
bsp/qemu-vexpress-a9/drivers/audio/wav_play.c
已删除
100644 → 0
浏览文件 @
15e01877
#include <rtthread.h>
#include <rtdevice.h>
#include <finsh.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include "audio_device.h"
#define BUFSZ 2048
struct
RIFF_HEADER_DEF
{
char
riff_id
[
4
];
// 'R','I','F','F'
uint32_t
riff_size
;
char
riff_format
[
4
];
// 'W','A','V','E'
};
struct
WAVE_FORMAT_DEF
{
uint16_t
FormatTag
;
uint16_t
Channels
;
uint32_t
SamplesPerSec
;
uint32_t
AvgBytesPerSec
;
uint16_t
BlockAlign
;
uint16_t
BitsPerSample
;
};
struct
FMT_BLOCK_DEF
{
char
fmt_id
[
4
];
// 'f','m','t',' '
uint32_t
fmt_size
;
struct
WAVE_FORMAT_DEF
wav_format
;
};
struct
DATA_BLOCK_DEF
{
char
data_id
[
4
];
// 'R','I','F','F'
uint32_t
data_size
;
};
struct
wav_info
{
struct
RIFF_HEADER_DEF
header
;
struct
FMT_BLOCK_DEF
fmt_block
;
struct
DATA_BLOCK_DEF
data_block
;
};
static
char
file_name
[
32
];
void
wavplay_thread_entry
(
void
*
parameter
)
{
FILE
*
fp
=
NULL
;
uint16_t
*
buffer
=
NULL
;
struct
wav_info
*
info
=
NULL
;
fp
=
fopen
(
file_name
,
"rb"
);
if
(
!
fp
)
{
printf
(
"open file failed!
\n
"
);
goto
__exit
;
}
info
=
(
struct
wav_info
*
)
malloc
(
sizeof
(
*
info
));
if
(
!
info
)
goto
__exit
;
if
(
fread
(
&
(
info
->
header
),
sizeof
(
struct
RIFF_HEADER_DEF
),
1
,
fp
)
!=
1
)
goto
__exit
;
if
(
fread
(
&
(
info
->
fmt_block
),
sizeof
(
struct
FMT_BLOCK_DEF
),
1
,
fp
)
!=
1
)
goto
__exit
;
if
(
fread
(
&
(
info
->
data_block
),
sizeof
(
struct
DATA_BLOCK_DEF
),
1
,
fp
)
!=
1
)
goto
__exit
;
printf
(
"wav information:
\n
"
);
printf
(
"samplerate %u
\n
"
,
info
->
fmt_block
.
wav_format
.
SamplesPerSec
);
printf
(
"channel %u
\n
"
,
info
->
fmt_block
.
wav_format
.
Channels
);
audio_device_init
();
audio_device_open
();
audio_device_set_rate
(
info
->
fmt_block
.
wav_format
.
SamplesPerSec
);
while
(
!
feof
(
fp
))
{
int
length
;
buffer
=
(
uint16_t
*
)
audio_device_get_buffer
(
RT_NULL
);
length
=
fread
(
buffer
,
1
,
BUFSZ
,
fp
);
if
(
length
)
{
if
(
info
->
fmt_block
.
wav_format
.
Channels
==
1
)
{
/* extend to stereo channels */
int
index
;
uint16_t
*
ptr
;
ptr
=
(
uint16_t
*
)((
uint8_t
*
)
buffer
+
BUFSZ
*
2
);
for
(
index
=
1
;
index
<
BUFSZ
/
2
;
index
++
)
{
*
ptr
=
*
(
ptr
-
1
)
=
buffer
[
BUFSZ
/
2
-
index
];
ptr
-=
2
;
}
length
=
length
*
2
;
}
audio_device_write
((
uint8_t
*
)
buffer
,
length
);
}
else
{
audio_device_put_buffer
((
uint8_t
*
)
buffer
);
break
;
}
}
audio_device_close
();
__exit:
if
(
fp
)
fclose
(
fp
);
if
(
info
)
free
(
info
);
}
int
wavplay
(
int
argc
,
char
**
argv
)
{
rt_thread_t
tid
=
RT_NULL
;
if
(
argc
!=
2
)
{
printf
(
"Usage:
\n
"
);
printf
(
"wavplay song.wav
\n
"
);
return
0
;
}
memset
(
file_name
,
0
,
sizeof
(
file_name
));
memcpy
(
file_name
,
argv
[
1
],
strlen
(
argv
[
1
]));
tid
=
rt_thread_create
(
"wayplay"
,
wavplay_thread_entry
,
RT_NULL
,
1024
*
8
,
22
,
10
);
if
(
tid
!=
RT_NULL
)
rt_thread_startup
(
tid
);
}
MSH_CMD_EXPORT
(
wavplay
,
wavplay
song
.
wav
);
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/.mxproject
浏览文件 @
fcfa986b
此差异已折叠。
点击以展开。
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Inc/stm32l4xx_hal_conf.h
浏览文件 @
fcfa986b
...
...
@@ -81,7 +81,7 @@
#define HAL_QSPI_MODULE_ENABLED
/*#define HAL_RNG_MODULE_ENABLED */
#define HAL_RTC_MODULE_ENABLED
/*#define HAL_SAI_MODULE_ENABLED */
#define HAL_SAI_MODULE_ENABLED
/*#define HAL_SD_MODULE_ENABLED */
/*#define HAL_SMBUS_MODULE_ENABLED */
/*#define HAL_SMARTCARD_MODULE_ENABLED */
...
...
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/STM32L475VE.ioc
浏览文件 @
fcfa986b
...
...
@@ -11,65 +11,73 @@ KeepUserPlacement=false
Mcu.Family=STM32L4
Mcu.IP0=ADC1
Mcu.IP1=IWDG
Mcu.IP10=TIM2
Mcu.IP11=TIM4
Mcu.IP12=TIM15
Mcu.IP13=TIM16
Mcu.IP14=TIM17
Mcu.IP15=USART1
Mcu.IP16=USART2
Mcu.IP17=USB_OTG_FS
Mcu.IP10=TIM1
Mcu.IP11=TIM2
Mcu.IP12=TIM4
Mcu.IP13=TIM15
Mcu.IP14=TIM16
Mcu.IP15=TIM17
Mcu.IP16=USART1
Mcu.IP17=USART2
Mcu.IP18=USB_OTG_FS
Mcu.IP2=NVIC
Mcu.IP3=QUADSPI
Mcu.IP4=RCC
Mcu.IP5=RTC
Mcu.IP6=S
P
I1
Mcu.IP7=SPI
2
Mcu.IP8=S
YS
Mcu.IP9=
TIM1
Mcu.IPNb=1
8
Mcu.IP6=S
A
I1
Mcu.IP7=SPI
1
Mcu.IP8=S
PI2
Mcu.IP9=
SYS
Mcu.IPNb=1
9
Mcu.Name=STM32L475V(C-E-G)Tx
Mcu.Package=LQFP100
Mcu.Pin0=PC14-OSC32_IN (PC14)
Mcu.Pin1=PC15-OSC32_OUT (PC15)
Mcu.Pin10=PE9
Mcu.Pin11=PE10
Mcu.Pin12=PE11
Mcu.Pin13=PE12
Mcu.Pin14=PE13
Mcu.Pin15=PE14
Mcu.Pin16=PE15
Mcu.Pin17=PB10
Mcu.Pin18=PB11
Mcu.Pin19=PB13
Mcu.Pin2=PH0-OSC_IN (PH0)
Mcu.Pin20=PB14
Mcu.Pin21=PB15
Mcu.Pin22=PA9
Mcu.Pin23=PA10
Mcu.Pin24=PA11
Mcu.Pin25=PA12
Mcu.Pin26=PA13 (JTMS-SWDIO)
Mcu.Pin27=PA14 (JTCK-SWCLK)
Mcu.Pin28=PB7
Mcu.Pin29=PB8
Mcu.Pin3=PH1-OSC_OUT (PH1)
Mcu.Pin30=VP_IWDG_VS_IWDG
Mcu.Pin31=VP_RTC_VS_RTC_Activate
Mcu.Pin32=VP_SYS_VS_Systick
Mcu.Pin33=VP_TIM1_VS_ClockSourceINT
Mcu.Pin34=VP_TIM2_VS_ClockSourceINT
Mcu.Pin35=VP_TIM4_VS_ClockSourceINT
Mcu.Pin36=VP_TIM15_VS_ClockSourceINT
Mcu.Pin37=VP_TIM16_VS_ClockSourceINT
Mcu.Pin38=VP_TIM17_VS_ClockSourceINT
Mcu.Pin4=PA2
Mcu.Pin5=PA3
Mcu.Pin6=PA5
Mcu.Pin7=PA6
Mcu.Pin8=PA7
Mcu.Pin9=PC5
Mcu.PinsNb=39
Mcu.Pin0=PE2
Mcu.Pin1=PE3
Mcu.Pin10=PA3
Mcu.Pin11=PA5
Mcu.Pin12=PA6
Mcu.Pin13=PA7
Mcu.Pin14=PC5
Mcu.Pin15=PE9
Mcu.Pin16=PE10
Mcu.Pin17=PE11
Mcu.Pin18=PE12
Mcu.Pin19=PE13
Mcu.Pin2=PE4
Mcu.Pin20=PE14
Mcu.Pin21=PE15
Mcu.Pin22=PB10
Mcu.Pin23=PB11
Mcu.Pin24=PB13
Mcu.Pin25=PB14
Mcu.Pin26=PB15
Mcu.Pin27=PA9
Mcu.Pin28=PA10
Mcu.Pin29=PA11
Mcu.Pin3=PE5
Mcu.Pin30=PA12
Mcu.Pin31=PA13 (JTMS-SWDIO)
Mcu.Pin32=PA14 (JTCK-SWCLK)
Mcu.Pin33=PB7
Mcu.Pin34=PB8
Mcu.Pin35=VP_IWDG_VS_IWDG
Mcu.Pin36=VP_RTC_VS_RTC_Activate
Mcu.Pin37=VP_SAI1_VP_$IpInstance_SAIA_SAI_BASIC
Mcu.Pin38=VP_SAI1_VP_$IpInstance_SAIB_SAI_BASIC
Mcu.Pin39=VP_SYS_VS_Systick
Mcu.Pin4=PE6
Mcu.Pin40=VP_TIM1_VS_ClockSourceINT
Mcu.Pin41=VP_TIM2_VS_ClockSourceINT
Mcu.Pin42=VP_TIM4_VS_ClockSourceINT
Mcu.Pin43=VP_TIM15_VS_ClockSourceINT
Mcu.Pin44=VP_TIM16_VS_ClockSourceINT
Mcu.Pin45=VP_TIM17_VS_ClockSourceINT
Mcu.Pin5=PC14-OSC32_IN (PC14)
Mcu.Pin6=PC15-OSC32_OUT (PC15)
Mcu.Pin7=PH0-OSC_IN (PH0)
Mcu.Pin8=PH1-OSC_OUT (PH1)
Mcu.Pin9=PA2
Mcu.PinsNb=46
Mcu.ThirdPartyNb=0
Mcu.UserConstants=
Mcu.UserName=STM32L475VETx
...
...
@@ -152,6 +160,16 @@ PE14.Signal=QUADSPI_BK1_IO2
PE15.Locked=true
PE15.Mode=Single Bank
PE15.Signal=QUADSPI_BK1_IO3
PE2.Mode=SAI_A_MasterWithClock
PE2.Signal=SAI1_MCLK_A
PE3.Mode=SAI_B_SyncSlave
PE3.Signal=SAI1_SD_B
PE4.Mode=SAI_A_MasterWithClock
PE4.Signal=SAI1_FS_A
PE5.Mode=SAI_A_MasterWithClock
PE5.Signal=SAI1_SCK_A
PE6.Mode=SAI_A_MasterWithClock
PE6.Signal=SAI1_SD_A
PE9.Signal=S_TIM1_CH1
PH0-OSC_IN\ (PH0).Mode=HSE-External-Oscillator
PH0-OSC_IN\ (PH0).Signal=RCC_OSC_IN
...
...
@@ -240,6 +258,20 @@ RCC.VCOInputFreq_Value=8000000
RCC.VCOOutputFreq_Value=160000000
RCC.VCOSAI1OutputFreq_Value=96000000
RCC.VCOSAI2OutputFreq_Value=64000000
SAI1.AudioFrequency-SAI_A_MasterWithClock=SAI_AUDIO_FREQUENCY_44K
SAI1.ErrorAudioFreq-SAI_A_MasterWithClock=-72.09 %
SAI1.IPParameters=Instance-SAI_A_MasterWithClock,VirtualMode-SAI_A_MasterWithClock,MClockEnable-SAI_A_MasterWithClock,RealAudioFreq-SAI_A_MasterWithClock,ErrorAudioFreq-SAI_A_MasterWithClock,InitProtocol-SAI_A_MasterWithClock,VirtualProtocol-SAI_A_BASIC,AudioFrequency-SAI_A_MasterWithClock,OutputDrive-SAI_A_MasterWithClock,Instance-SAI_B_SyncSlave,VirtualMode-SAI_B_SyncSlave,InitProtocol-SAI_B_SyncSlave,VirtualProtocol-SAI_B_BASIC
SAI1.InitProtocol-SAI_A_MasterWithClock=Enable
SAI1.InitProtocol-SAI_B_SyncSlave=Enable
SAI1.Instance-SAI_A_MasterWithClock=SAI$Index_Block_A
SAI1.Instance-SAI_B_SyncSlave=SAI$Index_Block_B
SAI1.MClockEnable-SAI_A_MasterWithClock=SAI_MASTERCLOCK_ENABLE
SAI1.OutputDrive-SAI_A_MasterWithClock=SAI_OUTPUTDRIVE_ENABLE
SAI1.RealAudioFreq-SAI_A_MasterWithClock=53.571 KHz
SAI1.VirtualMode-SAI_A_MasterWithClock=VM_MASTER
SAI1.VirtualMode-SAI_B_SyncSlave=VM_SLAVE
SAI1.VirtualProtocol-SAI_A_BASIC=VM_BASIC_PROTOCOL
SAI1.VirtualProtocol-SAI_B_BASIC=VM_BASIC_PROTOCOL
SH.ADCx_IN14.0=ADC1_IN14,IN14-Single-Ended
SH.ADCx_IN14.ConfNb=1
SH.S_TIM1_CH1.0=TIM1_CH1,PWM Generation1 CH1
...
...
@@ -280,6 +312,10 @@ VP_IWDG_VS_IWDG.Mode=IWDG_Activate
VP_IWDG_VS_IWDG.Signal=IWDG_VS_IWDG
VP_RTC_VS_RTC_Activate.Mode=RTC_Enabled
VP_RTC_VS_RTC_Activate.Signal=RTC_VS_RTC_Activate
VP_SAI1_VP_$IpInstance_SAIA_SAI_BASIC.Mode=SAI_A_BASIC
VP_SAI1_VP_$IpInstance_SAIA_SAI_BASIC.Signal=SAI1_VP_$IpInstance_SAIA_SAI_BASIC
VP_SAI1_VP_$IpInstance_SAIB_SAI_BASIC.Mode=SAI_B_BASIC
VP_SAI1_VP_$IpInstance_SAIB_SAI_BASIC.Signal=SAI1_VP_$IpInstance_SAIB_SAI_BASIC
VP_SYS_VS_Systick.Mode=SysTick
VP_SYS_VS_Systick.Signal=SYS_VS_Systick
VP_TIM15_VS_ClockSourceINT.Mode=Internal
...
...
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Src/main.c
浏览文件 @
fcfa986b
...
...
@@ -70,10 +70,14 @@ QSPI_HandleTypeDef hqspi;
RTC_HandleTypeDef
hrtc
;
SAI_HandleTypeDef
hsai_BlockA1
;
SAI_HandleTypeDef
hsai_BlockB1
;
SPI_HandleTypeDef
hspi1
;
SPI_HandleTypeDef
hspi2
;
TIM_HandleTypeDef
htim1
;
TIM_HandleTypeDef
htim2
;
TIM_HandleTypeDef
htim4
;
TIM_HandleTypeDef
htim15
;
TIM_HandleTypeDef
htim16
;
...
...
@@ -105,6 +109,8 @@ static void MX_TIM16_Init(void);
static
void
MX_TIM15_Init
(
void
);
static
void
MX_TIM4_Init
(
void
);
static
void
MX_TIM1_Init
(
void
);
static
void
MX_SAI1_Init
(
void
);
static
void
MX_TIM2_Init
(
void
);
static
void
MX_USB_OTG_FS_PCD_Init
(
void
);
/* USER CODE BEGIN PFP */
/* Private function prototypes -----------------------------------------------*/
...
...
@@ -158,6 +164,8 @@ int main(void)
MX_TIM15_Init
();
MX_TIM4_Init
();
MX_TIM1_Init
();
MX_SAI1_Init
();
MX_TIM2_Init
();
MX_USB_OTG_FS_PCD_Init
();
/* USER CODE BEGIN 2 */
...
...
@@ -222,10 +230,11 @@ void SystemClock_Config(void)
Error_Handler
();
}
PeriphClkInit
.
PeriphClockSelection
=
RCC_PERIPHCLK_RTC
|
RCC_PERIPHCLK_USART1
|
RCC_PERIPHCLK_USART2
|
RCC_PERIPHCLK_
USB
|
RCC_PERIPHCLK_ADC
;
|
RCC_PERIPHCLK_USART2
|
RCC_PERIPHCLK_
SAI1
|
RCC_PERIPHCLK_
USB
|
RCC_PERIPHCLK_
ADC
;
PeriphClkInit
.
Usart1ClockSelection
=
RCC_USART1CLKSOURCE_PCLK2
;
PeriphClkInit
.
Usart2ClockSelection
=
RCC_USART2CLKSOURCE_PCLK1
;
PeriphClkInit
.
Sai1ClockSelection
=
RCC_SAI1CLKSOURCE_PLLSAI1
;
PeriphClkInit
.
AdcClockSelection
=
RCC_ADCCLKSOURCE_PLLSAI1
;
PeriphClkInit
.
RTCClockSelection
=
RCC_RTCCLKSOURCE_LSE
;
PeriphClkInit
.
UsbClockSelection
=
RCC_USBCLKSOURCE_PLLSAI1
;
...
...
@@ -235,7 +244,8 @@ void SystemClock_Config(void)
PeriphClkInit
.
PLLSAI1
.
PLLSAI1P
=
RCC_PLLP_DIV7
;
PeriphClkInit
.
PLLSAI1
.
PLLSAI1Q
=
RCC_PLLQ_DIV2
;
PeriphClkInit
.
PLLSAI1
.
PLLSAI1R
=
RCC_PLLR_DIV2
;
PeriphClkInit
.
PLLSAI1
.
PLLSAI1ClockOut
=
RCC_PLLSAI1_48M2CLK
|
RCC_PLLSAI1_ADC1CLK
;
PeriphClkInit
.
PLLSAI1
.
PLLSAI1ClockOut
=
RCC_PLLSAI1_SAI1CLK
|
RCC_PLLSAI1_48M2CLK
|
RCC_PLLSAI1_ADC1CLK
;
if
(
HAL_RCCEx_PeriphCLKConfig
(
&
PeriphClkInit
)
!=
HAL_OK
)
{
Error_Handler
();
...
...
@@ -409,6 +419,55 @@ static void MX_RTC_Init(void)
}
/**
* @brief SAI1 Initialization Function
* @param None
* @retval None
*/
static
void
MX_SAI1_Init
(
void
)
{
/* USER CODE BEGIN SAI1_Init 0 */
/* USER CODE END SAI1_Init 0 */
/* USER CODE BEGIN SAI1_Init 1 */
/* USER CODE END SAI1_Init 1 */
hsai_BlockA1
.
Instance
=
SAI1_Block_A
;
hsai_BlockA1
.
Init
.
AudioMode
=
SAI_MODEMASTER_TX
;
hsai_BlockA1
.
Init
.
Synchro
=
SAI_ASYNCHRONOUS
;
hsai_BlockA1
.
Init
.
OutputDrive
=
SAI_OUTPUTDRIVE_ENABLE
;
hsai_BlockA1
.
Init
.
NoDivider
=
SAI_MASTERDIVIDER_ENABLE
;
hsai_BlockA1
.
Init
.
FIFOThreshold
=
SAI_FIFOTHRESHOLD_EMPTY
;
hsai_BlockA1
.
Init
.
AudioFrequency
=
SAI_AUDIO_FREQUENCY_44K
;
hsai_BlockA1
.
Init
.
SynchroExt
=
SAI_SYNCEXT_DISABLE
;
hsai_BlockA1
.
Init
.
MonoStereoMode
=
SAI_STEREOMODE
;
hsai_BlockA1
.
Init
.
CompandingMode
=
SAI_NOCOMPANDING
;
hsai_BlockA1
.
Init
.
TriState
=
SAI_OUTPUT_NOTRELEASED
;
if
(
HAL_SAI_InitProtocol
(
&
hsai_BlockA1
,
SAI_I2S_STANDARD
,
SAI_PROTOCOL_DATASIZE_16BIT
,
2
)
!=
HAL_OK
)
{
Error_Handler
();
}
hsai_BlockB1
.
Instance
=
SAI1_Block_B
;
hsai_BlockB1
.
Init
.
AudioMode
=
SAI_MODESLAVE_RX
;
hsai_BlockB1
.
Init
.
Synchro
=
SAI_SYNCHRONOUS
;
hsai_BlockB1
.
Init
.
OutputDrive
=
SAI_OUTPUTDRIVE_DISABLE
;
hsai_BlockB1
.
Init
.
FIFOThreshold
=
SAI_FIFOTHRESHOLD_EMPTY
;
hsai_BlockB1
.
Init
.
SynchroExt
=
SAI_SYNCEXT_DISABLE
;
hsai_BlockB1
.
Init
.
MonoStereoMode
=
SAI_STEREOMODE
;
hsai_BlockB1
.
Init
.
CompandingMode
=
SAI_NOCOMPANDING
;
hsai_BlockB1
.
Init
.
TriState
=
SAI_OUTPUT_NOTRELEASED
;
if
(
HAL_SAI_InitProtocol
(
&
hsai_BlockB1
,
SAI_I2S_STANDARD
,
SAI_PROTOCOL_DATASIZE_16BIT
,
2
)
!=
HAL_OK
)
{
Error_Handler
();
}
/* USER CODE BEGIN SAI1_Init 2 */
/* USER CODE END SAI1_Init 2 */
}
/**
* @brief SPI1 Initialization Function
* @param None
...
...
@@ -569,6 +628,69 @@ static void MX_TIM1_Init(void)
}
/**
* @brief TIM2 Initialization Function
* @param None
* @retval None
*/
static
void
MX_TIM2_Init
(
void
)
{
/* USER CODE BEGIN TIM2_Init 0 */
/* USER CODE END TIM2_Init 0 */
TIM_ClockConfigTypeDef
sClockSourceConfig
=
{
0
};
TIM_MasterConfigTypeDef
sMasterConfig
=
{
0
};
TIM_OC_InitTypeDef
sConfigOC
=
{
0
};
/* USER CODE BEGIN TIM2_Init 1 */
/* USER CODE END TIM2_Init 1 */
htim2
.
Instance
=
TIM2
;
htim2
.
Init
.
Prescaler
=
0
;
htim2
.
Init
.
CounterMode
=
TIM_COUNTERMODE_UP
;
htim2
.
Init
.
Period
=
0
;
htim2
.
Init
.
ClockDivision
=
TIM_CLOCKDIVISION_DIV1
;
htim2
.
Init
.
AutoReloadPreload
=
TIM_AUTORELOAD_PRELOAD_DISABLE
;
if
(
HAL_TIM_Base_Init
(
&
htim2
)
!=
HAL_OK
)
{
Error_Handler
();
}
sClockSourceConfig
.
ClockSource
=
TIM_CLOCKSOURCE_INTERNAL
;
if
(
HAL_TIM_ConfigClockSource
(
&
htim2
,
&
sClockSourceConfig
)
!=
HAL_OK
)
{
Error_Handler
();
}
if
(
HAL_TIM_PWM_Init
(
&
htim2
)
!=
HAL_OK
)
{
Error_Handler
();
}
sMasterConfig
.
MasterOutputTrigger
=
TIM_TRGO_RESET
;
sMasterConfig
.
MasterSlaveMode
=
TIM_MASTERSLAVEMODE_DISABLE
;
if
(
HAL_TIMEx_MasterConfigSynchronization
(
&
htim2
,
&
sMasterConfig
)
!=
HAL_OK
)
{
Error_Handler
();
}
sConfigOC
.
OCMode
=
TIM_OCMODE_PWM1
;
sConfigOC
.
Pulse
=
0
;
sConfigOC
.
OCPolarity
=
TIM_OCPOLARITY_HIGH
;
sConfigOC
.
OCFastMode
=
TIM_OCFAST_DISABLE
;
if
(
HAL_TIM_PWM_ConfigChannel
(
&
htim2
,
&
sConfigOC
,
TIM_CHANNEL_3
)
!=
HAL_OK
)
{
Error_Handler
();
}
if
(
HAL_TIM_PWM_ConfigChannel
(
&
htim2
,
&
sConfigOC
,
TIM_CHANNEL_4
)
!=
HAL_OK
)
{
Error_Handler
();
}
/* USER CODE BEGIN TIM2_Init 2 */
/* USER CODE END TIM2_Init 2 */
HAL_TIM_MspPostInit
(
&
htim2
);
}
/**
* @brief TIM4 Initialization Function
* @param None
...
...
@@ -856,10 +978,10 @@ static void MX_GPIO_Init(void)
{
/* GPIO Ports Clock Enable */
__HAL_RCC_GPIOE_CLK_ENABLE
();
__HAL_RCC_GPIOC_CLK_ENABLE
();
__HAL_RCC_GPIOH_CLK_ENABLE
();
__HAL_RCC_GPIOA_CLK_ENABLE
();
__HAL_RCC_GPIOE_CLK_ENABLE
();
__HAL_RCC_GPIOB_CLK_ENABLE
();
}
...
...
bsp/stm32/stm32l475-atk-pandora/board/CubeMX_Config/Src/stm32l4xx_hal_msp.c
浏览文件 @
fcfa986b
...
...
@@ -814,6 +814,97 @@ void HAL_PCD_MspDeInit(PCD_HandleTypeDef* hpcd)
}
static
uint32_t
SAI1_client
=
0
;
void
HAL_SAI_MspInit
(
SAI_HandleTypeDef
*
hsai
)
{
GPIO_InitTypeDef
GPIO_InitStruct
;
/* SAI1 */
if
(
hsai
->
Instance
==
SAI1_Block_A
)
{
/* Peripheral clock enable */
if
(
SAI1_client
==
0
)
{
__HAL_RCC_SAI1_CLK_ENABLE
();
}
SAI1_client
++
;
/**SAI1_A_Block_A GPIO Configuration
PE2 ------> SAI1_MCLK_A
PE4 ------> SAI1_FS_A
PE5 ------> SAI1_SCK_A
PE6 ------> SAI1_SD_A
*/
GPIO_InitStruct
.
Pin
=
GPIO_PIN_2
|
GPIO_PIN_4
|
GPIO_PIN_5
|
GPIO_PIN_6
;
GPIO_InitStruct
.
Mode
=
GPIO_MODE_AF_PP
;
GPIO_InitStruct
.
Pull
=
GPIO_NOPULL
;
GPIO_InitStruct
.
Speed
=
GPIO_SPEED_FREQ_LOW
;
GPIO_InitStruct
.
Alternate
=
GPIO_AF13_SAI1
;
HAL_GPIO_Init
(
GPIOE
,
&
GPIO_InitStruct
);
}
if
(
hsai
->
Instance
==
SAI1_Block_B
)
{
/* Peripheral clock enable */
if
(
SAI1_client
==
0
)
{
__HAL_RCC_SAI1_CLK_ENABLE
();
}
SAI1_client
++
;
/**SAI1_B_Block_B GPIO Configuration
PE3 ------> SAI1_SD_B
*/
GPIO_InitStruct
.
Pin
=
GPIO_PIN_3
;
GPIO_InitStruct
.
Mode
=
GPIO_MODE_AF_PP
;
GPIO_InitStruct
.
Pull
=
GPIO_NOPULL
;
GPIO_InitStruct
.
Speed
=
GPIO_SPEED_FREQ_LOW
;
GPIO_InitStruct
.
Alternate
=
GPIO_AF13_SAI1
;
HAL_GPIO_Init
(
GPIOE
,
&
GPIO_InitStruct
);
}
}
void
HAL_SAI_MspDeInit
(
SAI_HandleTypeDef
*
hsai
)
{
/* SAI1 */
if
(
hsai
->
Instance
==
SAI1_Block_A
)
{
SAI1_client
--
;
if
(
SAI1_client
==
0
)
{
/* Peripheral clock disable */
__HAL_RCC_SAI1_CLK_DISABLE
();
}
/**SAI1_A_Block_A GPIO Configuration
PE2 ------> SAI1_MCLK_A
PE4 ------> SAI1_FS_A
PE5 ------> SAI1_SCK_A
PE6 ------> SAI1_SD_A
*/
HAL_GPIO_DeInit
(
GPIOE
,
GPIO_PIN_2
|
GPIO_PIN_4
|
GPIO_PIN_5
|
GPIO_PIN_6
);
}
if
(
hsai
->
Instance
==
SAI1_Block_B
)
{
SAI1_client
--
;
if
(
SAI1_client
==
0
)
{
/* Peripheral clock disable */
__HAL_RCC_SAI1_CLK_DISABLE
();
}
/**SAI1_B_Block_B GPIO Configuration
PE3 ------> SAI1_SD_B
*/
HAL_GPIO_DeInit
(
GPIOE
,
GPIO_PIN_3
);
}
}
/* USER CODE BEGIN 1 */
/* USER CODE END 1 */
...
...
bsp/stm32/stm32l475-atk-pandora/board/Kconfig
浏览文件 @
fcfa986b
...
...
@@ -47,6 +47,23 @@ menu "Onboard Peripheral Drivers"
select PKG_USING_AHT10_LATEST_VERSION
default n
menuconfig BSP_USING_AUDIO
bool "Enable Audio Device"
select RT_USING_AUDIO
select BSP_USING_I2C
select BSP_USING_I2C3
default n
if BSP_USING_AUDIO
config BSP_USING_AUDIO_PLAY
bool "Enable Audio Play"
default y
config BSP_USING_AUDIO_RECORD
bool "Enable Audio Record"
default n
endif
endmenu
menu "On-chip Peripheral Drivers"
...
...
bsp/stm32/stm32l475-atk-pandora/board/SConscript
浏览文件 @
fcfa986b
...
...
@@ -21,10 +21,20 @@ if GetDepend(['BSP_USING_SDCARD']):
if
GetDepend
([
'BSP_USING_ICM20608'
])
or
GetDepend
([
'BSP_USING_AHT10'
]):
src
+=
Glob
(
'ports/sensor_port.c'
)
if
GetDepend
([
'BSP_USING_AUDIO'
]):
src
+=
Glob
(
'ports/audio/drv_es8388.c'
)
src
+=
Glob
(
'ports/audio/drv_sound.c'
)
if
GetDepend
([
'BSP_USING_AUDIO_RECORD'
]):
src
+=
Glob
(
'ports/audio/drv_mic.c'
)
path
=
[
cwd
]
path
+=
[
cwd
+
'/CubeMX_Config/Inc'
]
path
+=
[
cwd
+
'/ports'
]
if
GetDepend
([
'BSP_USING_AUDIO'
]):
path
+=
[
cwd
+
'/ports/audio'
]
startup_path_prefix
=
SDK_LIB
if
rtconfig
.
CROSS_TOOL
==
'gcc'
:
...
...
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.c
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-31 Zero-Free first implementation
*/
#include <rtthread.h>
#include <rtdevice.h>
#include "drv_es8388.h"
/* ES8388 address */
#define ES8388_ADDR 0x10
/*0x11:CE=1;0x10:CE=0*/
struct
es8388_device
{
struct
rt_i2c_bus_device
*
i2c
;
rt_uint16_t
pin
;
};
static
struct
es8388_device
es_dev
=
{
0
};
static
rt_uint16_t
reg_read
(
rt_uint8_t
addr
)
{
struct
rt_i2c_msg
msg
[
2
]
=
{
0
};
uint8_t
val
=
0xff
;
RT_ASSERT
(
es_dev
.
i2c
!=
RT_NULL
);
msg
[
0
].
addr
=
ES8388_ADDR
;
msg
[
0
].
flags
=
RT_I2C_WR
;
msg
[
0
].
len
=
1
;
msg
[
0
].
buf
=
&
addr
;
msg
[
1
].
addr
=
ES8388_ADDR
;
msg
[
1
].
flags
=
RT_I2C_RD
;
msg
[
1
].
len
=
1
;
msg
[
1
].
buf
=
&
val
;
if
(
rt_i2c_transfer
(
es_dev
.
i2c
,
msg
,
2
)
!=
2
)
{
rt_kprintf
(
"I2C read data failed, reg = 0x%02x.
\n
"
,
addr
);
return
0xff
;
}
return
val
;
}
static
void
reg_write
(
rt_uint8_t
addr
,
rt_uint8_t
val
)
{
struct
rt_i2c_msg
msgs
[
1
]
=
{
0
};
rt_uint8_t
buff
[
2
]
=
{
0
};
RT_ASSERT
(
es_dev
.
i2c
!=
RT_NULL
);
buff
[
0
]
=
addr
;
buff
[
1
]
=
val
;
msgs
[
0
].
addr
=
ES8388_ADDR
;
msgs
[
0
].
flags
=
RT_I2C_WR
;
msgs
[
0
].
buf
=
buff
;
msgs
[
0
].
len
=
2
;
if
(
rt_i2c_transfer
(
es_dev
.
i2c
,
msgs
,
1
)
!=
1
)
{
rt_kprintf
(
"I2C write data failed, reg = 0x%2x.
\n
"
,
addr
);
return
;
}
}
static
int
es8388_set_adc_dac_volume
(
int
mode
,
int
volume
,
int
dot
)
{
int
res
=
0
;
if
(
volume
<
-
96
||
volume
>
0
)
{
if
(
volume
<
-
96
)
volume
=
-
96
;
else
volume
=
0
;
}
dot
=
(
dot
>=
5
?
1
:
0
);
volume
=
(
-
volume
<<
1
)
+
dot
;
if
(
mode
==
ES_MODE_ADC
||
mode
==
ES_MODE_DAC_ADC
)
{
reg_write
(
ES8388_ADCCONTROL8
,
volume
);
reg_write
(
ES8388_ADCCONTROL9
,
volume
);
//ADC Right Volume=0db
}
if
(
mode
==
ES_MODE_DAC
||
mode
==
ES_MODE_DAC_ADC
)
{
reg_write
(
ES8388_DACCONTROL5
,
volume
);
reg_write
(
ES8388_DACCONTROL4
,
volume
);
}
return
res
;
}
void
es8388_set_voice_mute
(
rt_bool_t
enable
)
{
uint8_t
reg
=
0
;
reg
=
reg_read
(
ES8388_DACCONTROL3
);
reg
=
reg
&
0xFB
;
reg_write
(
ES8388_DACCONTROL3
,
reg
|
(((
int
)
enable
)
<<
2
));
}
rt_err_t
es8388_init
(
const
char
*
i2c_name
,
rt_uint16_t
pin
)
{
es_dev
.
i2c
=
rt_i2c_bus_device_find
(
i2c_name
);
if
(
es_dev
.
i2c
==
RT_NULL
)
{
rt_kprintf
(
"%s bus not found
\n
"
,
i2c_name
);
return
-
RT_ERROR
;
}
es_dev
.
pin
=
pin
;
reg_write
(
ES8388_DACCONTROL3
,
0x04
);
// 0x04 mute/0x00 unmute&ramp;DAC unmute and disabled digital volume control soft ramp
/* Chip Control and Power Management */
reg_write
(
ES8388_CONTROL2
,
0x50
);
reg_write
(
ES8388_CHIPPOWER
,
0x00
);
//normal all and power up all
reg_write
(
ES8388_MASTERMODE
,
0x00
);
//TODO:CODEC IN I2S SLAVE MODE
/* dac */
reg_write
(
ES8388_DACPOWER
,
0xC0
);
//disable DAC and disable Lout/Rout/1/2
reg_write
(
ES8388_CONTROL1
,
0x12
);
//Enfr=0,Play&Record Mode,(0x17-both of mic&paly)
// reg_write(ES8388_CONTROL2, 0); //LPVrefBuf=0,Pdn_ana=0
reg_write
(
ES8388_DACCONTROL1
,
0x18
);
//1a 0x18:16bit iis , 0x00:24
reg_write
(
ES8388_DACCONTROL2
,
0x02
);
//DACFsMode,SINGLE SPEED; DACFsRatio,256
reg_write
(
ES8388_DACCONTROL16
,
0x00
);
// 0x00 audio on LIN1&RIN1, 0x09 LIN2&RIN2
reg_write
(
ES8388_DACCONTROL17
,
0x90
);
// only left DAC to left mixer enable 0db
reg_write
(
ES8388_DACCONTROL20
,
0x90
);
// only right DAC to right mixer enable 0db
reg_write
(
ES8388_DACCONTROL21
,
0x80
);
//set internal ADC and DAC use the same LRCK clock, ADC LRCK as internal LRCK
reg_write
(
ES8388_DACCONTROL23
,
0x00
);
//vroi=0
es8388_set_adc_dac_volume
(
ES_MODE_DAC
,
0
,
0
);
// 0db
reg_write
(
ES8388_DACPOWER
,
0x3c
);
//0x3c Enable DAC and Enable Lout/Rout/1/2
/* adc */
reg_write
(
ES8388_ADCPOWER
,
0xFF
);
reg_write
(
ES8388_ADCCONTROL1
,
0xbb
);
// MIC Left and Right channel PGA gain
reg_write
(
ES8388_ADCCONTROL2
,
0x00
);
//0x00 LINSEL & RINSEL, LIN1/RIN1 as ADC Input; DSSEL,use one DS Reg11; DSR, LINPUT1-RINPUT1
reg_write
(
ES8388_ADCCONTROL3
,
0x02
);
reg_write
(
ES8388_ADCCONTROL4
,
0x0d
);
// Left/Right data, Left/Right justified mode, Bits length, I2S format
reg_write
(
ES8388_ADCCONTROL5
,
0x02
);
//ADCFsMode,singel SPEED,RATIO=256
//ALC for Microphone
es8388_set_adc_dac_volume
(
ES_MODE_ADC
,
0
,
0
);
// 0db
reg_write
(
ES8388_ADCPOWER
,
0x09
);
//Power on ADC, Enable LIN&RIN, Power off MICBIAS, set int1lp to low power mode
/* enable es8388 PA */
es8388_pa_power
(
RT_TRUE
);
return
RT_EOK
;
}
rt_err_t
es8388_start
(
enum
es8388_mode
mode
)
{
int
res
=
0
;
uint8_t
prev_data
=
0
,
data
=
0
;
prev_data
=
reg_read
(
ES8388_DACCONTROL21
);
if
(
mode
==
ES_MODE_LINE
)
{
reg_write
(
ES8388_DACCONTROL16
,
0x09
);
// 0x00 audio on LIN1&RIN1, 0x09 LIN2&RIN2 by pass enable
reg_write
(
ES8388_DACCONTROL17
,
0x50
);
// left DAC to left mixer enable and LIN signal to left mixer enable 0db : bupass enable
reg_write
(
ES8388_DACCONTROL20
,
0x50
);
// right DAC to right mixer enable and LIN signal to right mixer enable 0db : bupass enable
reg_write
(
ES8388_DACCONTROL21
,
0xC0
);
//enable adc
}
else
{
reg_write
(
ES8388_DACCONTROL21
,
0x80
);
//enable dac
}
data
=
reg_read
(
ES8388_DACCONTROL21
);
if
(
prev_data
!=
data
)
{
reg_write
(
ES8388_CHIPPOWER
,
0xF0
);
//start state machine
// reg_write(ES8388_ADDR, ES8388_CONTROL1, 0x16);
// reg_write(ES8388_ADDR, ES8388_CONTROL2, 0x50);
reg_write
(
ES8388_CHIPPOWER
,
0x00
);
//start state machine
}
if
(
mode
==
ES_MODE_ADC
||
mode
==
ES_MODE_DAC_ADC
||
mode
==
ES_MODE_LINE
)
{
reg_write
(
ES8388_ADCPOWER
,
0x00
);
//power up adc and line in
}
if
(
mode
==
ES_MODE_DAC
||
mode
==
ES_MODE_DAC_ADC
||
mode
==
ES_MODE_LINE
)
{
reg_write
(
ES8388_DACPOWER
,
0x3c
);
//power up dac and line out
es8388_set_voice_mute
(
RT_FALSE
);
}
return
res
;
}
rt_err_t
es8388_stop
(
enum
es8388_mode
mode
)
{
int
res
=
0
;
if
(
mode
==
ES_MODE_LINE
)
{
reg_write
(
ES8388_DACCONTROL21
,
0x80
);
//enable dac
reg_write
(
ES8388_DACCONTROL16
,
0x00
);
// 0x00 audio on LIN1&RIN1, 0x09 LIN2&RIN2
reg_write
(
ES8388_DACCONTROL17
,
0x90
);
// only left DAC to left mixer enable 0db
reg_write
(
ES8388_DACCONTROL20
,
0x90
);
// only right DAC to right mixer enable 0db
return
res
;
}
if
(
mode
==
ES_MODE_DAC
||
mode
==
ES_MODE_DAC_ADC
)
{
reg_write
(
ES8388_DACPOWER
,
0x00
);
es8388_set_voice_mute
(
RT_TRUE
);
//res |= Es8388SetAdcDacVolume(ES_MODULE_DAC, -96, 5); // 0db
// reg_write(ES8388_ADDR, ES8388_DACPOWER, 0xC0); //power down dac and line out
}
if
(
mode
==
ES_MODE_ADC
||
mode
==
ES_MODE_DAC_ADC
)
{
// Es8388SetAdcDacVolume(ES_MODULE_ADC, -96, 5); // 0db
reg_write
(
ES8388_ADCPOWER
,
0xFF
);
//power down adc and line in
}
if
(
mode
==
ES_MODE_DAC_ADC
)
{
reg_write
(
ES8388_DACCONTROL21
,
0x9C
);
//disable mclk
// reg_write(ES8388_CONTROL1, 0x00);
// reg_write(ES8388_CONTROL2, 0x58);
// reg_write(ES8388_CHIPPOWER, 0xF3); //stop state machine
}
return
RT_EOK
;
}
rt_err_t
es8388_fmt_set
(
enum
es8388_mode
mode
,
enum
es8388_format
fmt
)
{
uint8_t
reg
=
0
;
if
(
mode
==
ES_MODE_ADC
||
mode
==
ES_MODE_DAC_ADC
)
{
reg
=
reg_read
(
ES8388_ADCCONTROL4
);
reg
=
reg
&
0xfc
;
reg_write
(
ES8388_ADCCONTROL4
,
reg
|
fmt
);
}
if
(
mode
==
ES_MODE_DAC
||
mode
==
ES_MODE_DAC_ADC
)
{
reg
=
reg_read
(
ES8388_DACCONTROL1
);
reg
=
reg
&
0xf9
;
reg_write
(
ES8388_DACCONTROL1
,
reg
|
(
fmt
<<
1
));
}
return
RT_EOK
;
}
void
es8388_volume_set
(
rt_uint8_t
volume
)
{
if
(
volume
>
100
)
volume
=
100
;
volume
/=
3
;
reg_write
(
ES8388_DACCONTROL24
,
volume
);
reg_write
(
ES8388_DACCONTROL25
,
volume
);
}
rt_uint8_t
es8388_volume_get
(
void
)
{
rt_uint8_t
volume
;
volume
=
reg_read
(
ES8388_DACCONTROL24
);
if
(
volume
==
0xff
)
{
volume
=
0
;
}
else
{
volume
*=
3
;
if
(
volume
==
99
)
volume
=
100
;
}
return
volume
;
}
void
es8388_pa_power
(
rt_bool_t
enable
)
{
rt_pin_mode
(
es_dev
.
pin
,
PIN_MODE_OUTPUT
);
if
(
enable
)
{
rt_pin_write
(
es_dev
.
pin
,
PIN_HIGH
);
}
else
{
rt_pin_write
(
es_dev
.
pin
,
PIN_LOW
);
}
}
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_es8388.h
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-31 Zero-Free first implementation
*/
#ifndef __DRV_ES8388_H__
#define __DRV_ES8388_H__
/* ES8388 register space */
#define ES8388_CONTROL1 0x00
#define ES8388_CONTROL2 0x01
#define ES8388_CHIPPOWER 0x02
#define ES8388_ADCPOWER 0x03
#define ES8388_DACPOWER 0x04
#define ES8388_CHIPLOPOW1 0x05
#define ES8388_CHIPLOPOW2 0x06
#define ES8388_ANAVOLMANAG 0x07
#define ES8388_MASTERMODE 0x08
#define ES8388_ADCCONTROL1 0x09
#define ES8388_ADCCONTROL2 0x0a
#define ES8388_ADCCONTROL3 0x0b
#define ES8388_ADCCONTROL4 0x0c
#define ES8388_ADCCONTROL5 0x0d
#define ES8388_ADCCONTROL6 0x0e
#define ES8388_ADCCONTROL7 0x0f
#define ES8388_ADCCONTROL8 0x10
#define ES8388_ADCCONTROL9 0x11
#define ES8388_ADCCONTROL10 0x12
#define ES8388_ADCCONTROL11 0x13
#define ES8388_ADCCONTROL12 0x14
#define ES8388_ADCCONTROL13 0x15
#define ES8388_ADCCONTROL14 0x16
#define ES8388_DACCONTROL1 0x17
#define ES8388_DACCONTROL2 0x18
#define ES8388_DACCONTROL3 0x19
#define ES8388_DACCONTROL4 0x1a
#define ES8388_DACCONTROL5 0x1b
#define ES8388_DACCONTROL6 0x1c
#define ES8388_DACCONTROL7 0x1d
#define ES8388_DACCONTROL8 0x1e
#define ES8388_DACCONTROL9 0x1f
#define ES8388_DACCONTROL10 0x20
#define ES8388_DACCONTROL11 0x21
#define ES8388_DACCONTROL12 0x22
#define ES8388_DACCONTROL13 0x23
#define ES8388_DACCONTROL14 0x24
#define ES8388_DACCONTROL15 0x25
#define ES8388_DACCONTROL16 0x26
#define ES8388_DACCONTROL17 0x27
#define ES8388_DACCONTROL18 0x28
#define ES8388_DACCONTROL19 0x29
#define ES8388_DACCONTROL20 0x2a
#define ES8388_DACCONTROL21 0x2b
#define ES8388_DACCONTROL22 0x2c
#define ES8388_DACCONTROL23 0x2d
#define ES8388_DACCONTROL24 0x2e
#define ES8388_DACCONTROL25 0x2f
#define ES8388_DACCONTROL26 0x30
#define ES8388_DACCONTROL27 0x31
#define ES8388_DACCONTROL28 0x32
#define ES8388_DACCONTROL29 0x33
#define ES8388_DACCONTROL30 0x34
enum
es8388_mode
{
ES_MODE_NONE
=
0x00
,
ES_MODE_DAC
=
0x01
,
ES_MODE_ADC
=
0x02
,
ES_MODE_DAC_ADC
=
0x03
,
ES_MODE_LINE
=
0x04
,
ES_MODE_MAX
=
0x06
,
};
enum
es8388_format
{
ES_FMT_NORMAL
=
0
,
ES_FMT_LEFT
=
1
,
ES_FMT_RIGHT
=
2
,
ES_FMT_DSP
=
3
,
};
rt_err_t
es8388_init
(
const
char
*
i2c_name
,
rt_uint16_t
pin
);
rt_err_t
es8388_start
(
enum
es8388_mode
mode
);
rt_err_t
es8388_stop
(
enum
es8388_mode
mode
);
rt_err_t
es8388_fmt_set
(
enum
es8388_mode
mode
,
enum
es8388_format
fmt
);
void
es8388_volume_set
(
rt_uint8_t
volume
);
rt_uint8_t
es8388_volume_get
(
void
);
void
es8388_pa_power
(
rt_bool_t
enable
);
#endif
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_mic.c
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-31 Zero-Free first implementation
*/
#include <board.h>
#include "drv_es8388.h"
#define DBG_TAG "drv.mic"
#define DBG_LVL DBG_INFO
#include <rtdbg.h>
#define RX_FIFO_SIZE (1024)
struct
mic_device
{
struct
rt_audio_device
audio
;
struct
rt_audio_configure
record_config
;
rt_uint8_t
*
rx_fifo
;
rt_uint8_t
volume
;
};
static
struct
mic_device
mic_dev
=
{
0
};
static
rt_uint16_t
zero_frame
[
2
]
=
{
0
};
static
SAI_HandleTypeDef
SAI1B_Handler
=
{
0
};
static
DMA_HandleTypeDef
SAI1_RXDMA_Handler
=
{
0
};
extern
SAI_HandleTypeDef
SAI1A_Handler
;
extern
DMA_HandleTypeDef
SAI1_RXDMA_Handler
;
extern
void
SAIA_Frequency_Set
(
uint32_t
frequency
);
void
SAIB_Init
(
void
)
{
HAL_SAI_DeInit
(
&
SAI1B_Handler
);
SAI1B_Handler
.
Init
.
AudioFrequency
=
SAI_AUDIO_FREQUENCY_44K
;
SAI1B_Handler
.
Instance
=
SAI1_Block_B
;
SAI1B_Handler
.
Init
.
AudioMode
=
SAI_MODESLAVE_RX
;
SAI1B_Handler
.
Init
.
Synchro
=
SAI_SYNCHRONOUS
;
SAI1B_Handler
.
Init
.
OutputDrive
=
SAI_OUTPUTDRIVE_ENABLE
;
SAI1B_Handler
.
Init
.
NoDivider
=
SAI_MASTERDIVIDER_ENABLE
;
SAI1B_Handler
.
Init
.
FIFOThreshold
=
SAI_FIFOTHRESHOLD_EMPTY
;
SAI1B_Handler
.
Init
.
MonoStereoMode
=
SAI_MONOMODE
;
SAI1B_Handler
.
Init
.
Protocol
=
SAI_FREE_PROTOCOL
;
SAI1B_Handler
.
Init
.
DataSize
=
SAI_DATASIZE_16
;
SAI1B_Handler
.
Init
.
FirstBit
=
SAI_FIRSTBIT_MSB
;
SAI1B_Handler
.
Init
.
ClockStrobing
=
SAI_CLOCKSTROBING_RISINGEDGE
;
SAI1B_Handler
.
FrameInit
.
FrameLength
=
64
;
SAI1B_Handler
.
FrameInit
.
ActiveFrameLength
=
32
;
SAI1B_Handler
.
FrameInit
.
FSDefinition
=
SAI_FS_CHANNEL_IDENTIFICATION
;
SAI1B_Handler
.
FrameInit
.
FSPolarity
=
SAI_FS_ACTIVE_LOW
;
SAI1B_Handler
.
FrameInit
.
FSOffset
=
SAI_FS_BEFOREFIRSTBIT
;
SAI1B_Handler
.
SlotInit
.
FirstBitOffset
=
0
;
SAI1B_Handler
.
SlotInit
.
SlotSize
=
SAI_SLOTSIZE_32B
;
SAI1B_Handler
.
SlotInit
.
SlotNumber
=
2
;
SAI1B_Handler
.
SlotInit
.
SlotActive
=
SAI_SLOTACTIVE_0
|
SAI_SLOTACTIVE_1
;
HAL_SAI_Init
(
&
SAI1B_Handler
);
__HAL_SAI_ENABLE
(
&
SAI1B_Handler
);
/* Configure DMA used for SAI1 */
__HAL_RCC_DMA2_CLK_ENABLE
();
SAI1_RXDMA_Handler
.
Init
.
Request
=
DMA_REQUEST_1
;
SAI1_RXDMA_Handler
.
Init
.
Direction
=
DMA_PERIPH_TO_MEMORY
;
SAI1_RXDMA_Handler
.
Init
.
PeriphInc
=
DMA_PINC_DISABLE
;
SAI1_RXDMA_Handler
.
Init
.
MemInc
=
DMA_MINC_ENABLE
;
SAI1_RXDMA_Handler
.
Init
.
PeriphDataAlignment
=
DMA_PDATAALIGN_HALFWORD
;
SAI1_RXDMA_Handler
.
Init
.
MemDataAlignment
=
DMA_MDATAALIGN_HALFWORD
;
SAI1_RXDMA_Handler
.
Init
.
Mode
=
DMA_CIRCULAR
;
SAI1_RXDMA_Handler
.
Init
.
Priority
=
DMA_PRIORITY_HIGH
;
SAI1_RXDMA_Handler
.
Instance
=
DMA2_Channel2
;
__HAL_LINKDMA
(
&
SAI1B_Handler
,
hdmarx
,
SAI1_RXDMA_Handler
);
HAL_DMA_DeInit
(
&
SAI1_RXDMA_Handler
);
HAL_DMA_Init
(
&
SAI1_RXDMA_Handler
);
__HAL_DMA_ENABLE
(
&
SAI1_RXDMA_Handler
);
HAL_NVIC_SetPriority
(
DMA2_Channel2_IRQn
,
0x01
,
0
);
HAL_NVIC_EnableIRQ
(
DMA2_Channel2_IRQn
);
}
void
SAIB_Channels_Set
(
uint8_t
channels
)
{
if
(
channels
==
1
)
{
SAI1B_Handler
.
Init
.
MonoStereoMode
=
SAI_MONOMODE
;
}
else
{
SAI1B_Handler
.
Init
.
MonoStereoMode
=
SAI_STEREOMODE
;
}
__HAL_SAI_DISABLE
(
&
SAI1B_Handler
);
HAL_SAI_Init
(
&
SAI1B_Handler
);
__HAL_SAI_ENABLE
(
&
SAI1B_Handler
);
}
void
DMA2_Channel2_IRQHandler
(
void
)
{
HAL_DMA_IRQHandler
(
&
SAI1_RXDMA_Handler
);
}
void
HAL_SAI_RxHalfCpltCallback
(
SAI_HandleTypeDef
*
hsai
)
{
rt_audio_rx_done
(
&
mic_dev
.
audio
,
&
mic_dev
.
rx_fifo
[
0
],
RX_FIFO_SIZE
/
2
);
}
void
HAL_SAI_RxCpltCallback
(
SAI_HandleTypeDef
*
hsai
)
{
rt_audio_rx_done
(
&
mic_dev
.
audio
,
&
mic_dev
.
rx_fifo
[
RX_FIFO_SIZE
/
2
],
RX_FIFO_SIZE
/
2
);
}
static
rt_err_t
mic_getcaps
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
mic_device
*
mic_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
mic_dev
=
(
struct
mic_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_QUERY
:
/* qurey the types of hw_codec device */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_TYPE_QUERY
:
caps
->
udata
.
mask
=
AUDIO_TYPE_INPUT
|
AUDIO_TYPE_MIXER
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_INPUT
:
/* Provide capabilities of INPUT unit */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
caps
->
udata
.
config
.
samplerate
=
mic_dev
->
record_config
.
samplerate
;
caps
->
udata
.
config
.
channels
=
mic_dev
->
record_config
.
channels
;
caps
->
udata
.
config
.
samplebits
=
mic_dev
->
record_config
.
samplebits
;
break
;
case
AUDIO_DSP_SAMPLERATE
:
caps
->
udata
.
config
.
samplerate
=
mic_dev
->
record_config
.
samplerate
;
break
;
case
AUDIO_DSP_CHANNELS
:
caps
->
udata
.
config
.
channels
=
mic_dev
->
record_config
.
channels
;
break
;
case
AUDIO_DSP_SAMPLEBITS
:
caps
->
udata
.
config
.
samplebits
=
mic_dev
->
record_config
.
samplebits
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_MIXER
:
/* report the Mixer Units */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_QUERY
:
caps
->
udata
.
mask
=
AUDIO_MIXER_VOLUME
|
AUDIO_MIXER_LINE
;
break
;
case
AUDIO_MIXER_VOLUME
:
caps
->
udata
.
value
=
mic_dev
->
volume
;
break
;
case
AUDIO_MIXER_LINE
:
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
return
result
;
}
static
rt_err_t
mic_configure
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
mic_device
*
mic_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
mic_dev
=
(
struct
mic_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_MIXER
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_VOLUME
:
{
rt_uint32_t
volume
=
caps
->
udata
.
value
;
mic_dev
->
volume
=
volume
;
LOG_D
(
"set volume %d"
,
volume
);
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_INPUT
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
{
SAIA_Frequency_Set
(
caps
->
udata
.
config
.
samplerate
);
HAL_SAI_DMAStop
(
&
SAI1B_Handler
);
SAIB_Channels_Set
(
caps
->
udata
.
config
.
channels
);
HAL_SAI_Transmit
(
&
SAI1A_Handler
,
(
uint8_t
*
)
&
zero_frame
[
0
],
2
,
0
);
HAL_SAI_Receive_DMA
(
&
SAI1B_Handler
,
mic_dev
->
rx_fifo
,
RX_FIFO_SIZE
/
2
);
/* save configs */
mic_dev
->
record_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
mic_dev
->
record_config
.
channels
=
caps
->
udata
.
config
.
channels
;
mic_dev
->
record_config
.
samplebits
=
caps
->
udata
.
config
.
samplebits
;
LOG_D
(
"set samplerate %d"
,
mic_dev
->
record_config
.
samplerate
);
LOG_D
(
"set channels %d"
,
mic_dev
->
record_config
.
channels
);
break
;
}
case
AUDIO_DSP_SAMPLERATE
:
{
mic_dev
->
record_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
LOG_D
(
"set channels %d"
,
mic_dev
->
record_config
.
channels
);
break
;
}
case
AUDIO_DSP_CHANNELS
:
{
mic_dev
->
record_config
.
channels
=
caps
->
udata
.
config
.
channels
;
LOG_D
(
"set channels %d"
,
mic_dev
->
record_config
.
channels
);
break
;
}
default:
break
;
}
break
;
}
default:
break
;
}
return
result
;
}
static
rt_err_t
mic_init
(
struct
rt_audio_device
*
audio
)
{
struct
mic_device
*
mic_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
mic_dev
=
(
struct
mic_device
*
)
audio
->
parent
.
user_data
;
SAIB_Init
();
/* set default params */
SAIB_Channels_Set
(
mic_dev
->
record_config
.
channels
);
return
RT_EOK
;
}
static
rt_err_t
mic_start
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
struct
mic_device
*
mic_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
mic_dev
=
(
struct
mic_device
*
)
audio
->
parent
.
user_data
;
if
(
stream
==
AUDIO_STREAM_RECORD
)
{
es8388_start
(
ES_MODE_ADC
);
HAL_SAI_Transmit
(
&
SAI1A_Handler
,
(
uint8_t
*
)
&
zero_frame
[
0
],
2
,
0
);
HAL_SAI_Receive_DMA
(
&
SAI1B_Handler
,
mic_dev
->
rx_fifo
,
RX_FIFO_SIZE
/
2
);
}
return
RT_EOK
;
}
static
rt_err_t
mic_stop
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
if
(
stream
==
AUDIO_STREAM_RECORD
)
{
HAL_SAI_DMAStop
(
&
SAI1B_Handler
);
HAL_SAI_Abort
(
&
SAI1A_Handler
);
es8388_stop
(
ES_MODE_ADC
);
}
return
RT_EOK
;
}
static
struct
rt_audio_ops
mic_ops
=
{
.
getcaps
=
mic_getcaps
,
.
configure
=
mic_configure
,
.
init
=
mic_init
,
.
start
=
mic_start
,
.
stop
=
mic_stop
,
.
transmit
=
RT_NULL
,
.
buffer_info
=
RT_NULL
,
};
int
rt_hw_mic_init
(
void
)
{
rt_uint8_t
*
rx_fifo
;
if
(
mic_dev
.
rx_fifo
)
return
RT_EOK
;
rx_fifo
=
rt_malloc
(
RX_FIFO_SIZE
);
if
(
rx_fifo
==
RT_NULL
)
return
-
RT_ENOMEM
;
rt_memset
(
rx_fifo
,
0
,
RX_FIFO_SIZE
);
mic_dev
.
rx_fifo
=
rx_fifo
;
/* init default configuration */
{
mic_dev
.
record_config
.
samplerate
=
44100
;
mic_dev
.
record_config
.
channels
=
2
;
mic_dev
.
record_config
.
samplebits
=
16
;
mic_dev
.
volume
=
55
;
}
/* register sound device */
mic_dev
.
audio
.
ops
=
&
mic_ops
;
rt_audio_register
(
&
mic_dev
.
audio
,
"mic0"
,
RT_DEVICE_FLAG_RDONLY
,
&
mic_dev
);
return
RT_EOK
;
}
INIT_DEVICE_EXPORT
(
rt_hw_mic_init
);
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.c
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-31 Zero-Free first implementation
*/
#include <board.h>
#include "drv_sound.h"
#include "drv_es8388.h"
#define DBG_TAG "drv.sound"
#define DBG_LVL DBG_INFO
#include <rtdbg.h>
#define TX_FIFO_SIZE (2048)
struct
sound_device
{
struct
rt_audio_device
audio
;
struct
rt_audio_configure
replay_config
;
rt_uint8_t
*
tx_fifo
;
rt_uint8_t
volume
;
};
static
struct
sound_device
snd_dev
=
{
0
};
SAI_HandleTypeDef
SAI1A_Handler
=
{
0
};
DMA_HandleTypeDef
SAI1_TXDMA_Handler
=
{
0
};
static
void
SAIA_Init
(
void
)
{
RCC_PeriphCLKInitTypeDef
PeriphClkInit
;
/* Configure and enable PLLSAI1 clock to generate 45.714286MHz */
PeriphClkInit
.
PeriphClockSelection
=
RCC_PERIPHCLK_SAI1
;
PeriphClkInit
.
Sai1ClockSelection
=
RCC_SAI1CLKSOURCE_PLLSAI2
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2Source
=
RCC_PLLSOURCE_HSE
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2M
=
1
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2N
=
40
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2P
=
RCC_PLLP_DIV7
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2R
=
RCC_PLLR_DIV2
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2ClockOut
=
RCC_PLLSAI2_SAI2CLK
;
HAL_RCCEx_PeriphCLKConfig
(
&
PeriphClkInit
);
HAL_SAI_DeInit
(
&
SAI1A_Handler
);
SAI1A_Handler
.
Init
.
AudioFrequency
=
SAI_AUDIO_FREQUENCY_44K
;
SAI1A_Handler
.
Instance
=
SAI1_Block_A
;
SAI1A_Handler
.
Init
.
AudioMode
=
SAI_MODEMASTER_TX
;
SAI1A_Handler
.
Init
.
Synchro
=
SAI_ASYNCHRONOUS
;
SAI1A_Handler
.
Init
.
OutputDrive
=
SAI_OUTPUTDRIVE_ENABLE
;
SAI1A_Handler
.
Init
.
NoDivider
=
SAI_MASTERDIVIDER_ENABLE
;
SAI1A_Handler
.
Init
.
FIFOThreshold
=
SAI_FIFOTHRESHOLD_EMPTY
;
SAI1A_Handler
.
Init
.
MonoStereoMode
=
SAI_STEREOMODE
;
SAI1A_Handler
.
Init
.
Protocol
=
SAI_FREE_PROTOCOL
;
SAI1A_Handler
.
Init
.
DataSize
=
SAI_DATASIZE_16
;
SAI1A_Handler
.
Init
.
FirstBit
=
SAI_FIRSTBIT_MSB
;
SAI1A_Handler
.
Init
.
ClockStrobing
=
SAI_CLOCKSTROBING_RISINGEDGE
;
SAI1A_Handler
.
FrameInit
.
FrameLength
=
64
;
SAI1A_Handler
.
FrameInit
.
ActiveFrameLength
=
32
;
SAI1A_Handler
.
FrameInit
.
FSDefinition
=
SAI_FS_CHANNEL_IDENTIFICATION
;
SAI1A_Handler
.
FrameInit
.
FSPolarity
=
SAI_FS_ACTIVE_LOW
;
SAI1A_Handler
.
FrameInit
.
FSOffset
=
SAI_FS_BEFOREFIRSTBIT
;
SAI1A_Handler
.
SlotInit
.
FirstBitOffset
=
0
;
SAI1A_Handler
.
SlotInit
.
SlotSize
=
SAI_SLOTSIZE_32B
;
SAI1A_Handler
.
SlotInit
.
SlotNumber
=
2
;
SAI1A_Handler
.
SlotInit
.
SlotActive
=
SAI_SLOTACTIVE_0
|
SAI_SLOTACTIVE_1
;
HAL_SAI_Init
(
&
SAI1A_Handler
);
__HAL_SAI_ENABLE
(
&
SAI1A_Handler
);
/* Configure DMA used for SAI1 */
__HAL_RCC_DMA2_CLK_ENABLE
();
SAI1_TXDMA_Handler
.
Init
.
Request
=
DMA_REQUEST_1
;
SAI1_TXDMA_Handler
.
Init
.
Direction
=
DMA_MEMORY_TO_PERIPH
;
SAI1_TXDMA_Handler
.
Init
.
PeriphInc
=
DMA_PINC_DISABLE
;
SAI1_TXDMA_Handler
.
Init
.
MemInc
=
DMA_MINC_ENABLE
;
SAI1_TXDMA_Handler
.
Init
.
PeriphDataAlignment
=
DMA_PDATAALIGN_HALFWORD
;
SAI1_TXDMA_Handler
.
Init
.
MemDataAlignment
=
DMA_MDATAALIGN_HALFWORD
;
SAI1_TXDMA_Handler
.
Init
.
Mode
=
DMA_CIRCULAR
;
SAI1_TXDMA_Handler
.
Init
.
Priority
=
DMA_PRIORITY_HIGH
;
SAI1_TXDMA_Handler
.
Instance
=
DMA2_Channel1
;
__HAL_LINKDMA
(
&
SAI1A_Handler
,
hdmatx
,
SAI1_TXDMA_Handler
);
HAL_DMA_DeInit
(
&
SAI1_TXDMA_Handler
);
HAL_DMA_Init
(
&
SAI1_TXDMA_Handler
);
__HAL_DMA_ENABLE
(
&
SAI1_TXDMA_Handler
);
HAL_NVIC_SetPriority
(
DMA2_Channel1_IRQn
,
0x01
,
0
);
HAL_NVIC_EnableIRQ
(
DMA2_Channel1_IRQn
);
}
void
DMA2_Channel1_IRQHandler
(
void
)
{
HAL_DMA_IRQHandler
(
&
SAI1_TXDMA_Handler
);
}
void
HAL_SAI_TxHalfCpltCallback
(
SAI_HandleTypeDef
*
hsai
)
{
if
(
hsai
==
&
SAI1A_Handler
)
{
rt_audio_tx_complete
(
&
snd_dev
.
audio
);
}
}
void
HAL_SAI_TxCpltCallback
(
SAI_HandleTypeDef
*
hsai
)
{
if
(
hsai
==
&
SAI1A_Handler
)
{
rt_audio_tx_complete
(
&
snd_dev
.
audio
);
}
}
void
SAIA_Frequency_Set
(
uint32_t
frequency
)
{
RCC_PeriphCLKInitTypeDef
PeriphClkInit
;
HAL_RCCEx_GetPeriphCLKConfig
(
&
PeriphClkInit
);
if
((
frequency
==
SAI_AUDIO_FREQUENCY_11K
)
||
(
frequency
==
SAI_AUDIO_FREQUENCY_22K
)
||
(
frequency
==
SAI_AUDIO_FREQUENCY_44K
))
{
/* Configure and enable PLLSAI1 clock to generate 45.714286MHz */
PeriphClkInit
.
PeriphClockSelection
=
RCC_PERIPHCLK_SAI1
;
PeriphClkInit
.
Sai1ClockSelection
=
RCC_SAI1CLKSOURCE_PLLSAI2
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2Source
=
RCC_PLLSOURCE_HSE
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2M
=
1
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2N
=
40
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2ClockOut
=
RCC_PLLSAI2_SAI2CLK
;
HAL_RCCEx_PeriphCLKConfig
(
&
PeriphClkInit
);
}
else
{
/* Configure and enable PLLSAI1 clock to generate 49.142857MHz */
PeriphClkInit
.
PeriphClockSelection
=
RCC_PERIPHCLK_SAI1
;
PeriphClkInit
.
Sai1ClockSelection
=
RCC_SAI1CLKSOURCE_PLLSAI2
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2Source
=
RCC_PLLSOURCE_HSE
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2M
=
1
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2N
=
43
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2P
=
RCC_PLLP_DIV7
;
PeriphClkInit
.
PLLSAI2
.
PLLSAI2ClockOut
=
RCC_PLLSAI2_SAI2CLK
;
HAL_RCCEx_PeriphCLKConfig
(
&
PeriphClkInit
);
}
/* Disable SAI peripheral to allow access to SAI internal registers */
__HAL_SAI_DISABLE
(
&
SAI1A_Handler
);
/* Update the SAI audio frequency configuration */
SAI1A_Handler
.
Init
.
AudioFrequency
=
frequency
;
HAL_SAI_Init
(
&
SAI1A_Handler
);
/* Enable SAI peripheral to generate MCLK */
__HAL_SAI_ENABLE
(
&
SAI1A_Handler
);
}
void
SAIA_Channels_Set
(
uint8_t
channels
)
{
if
(
channels
==
1
)
{
SAI1A_Handler
.
Init
.
MonoStereoMode
=
SAI_MONOMODE
;
}
else
{
SAI1A_Handler
.
Init
.
MonoStereoMode
=
SAI_STEREOMODE
;
}
__HAL_SAI_DISABLE
(
&
SAI1A_Handler
);
HAL_SAI_Init
(
&
SAI1A_Handler
);
__HAL_SAI_ENABLE
(
&
SAI1A_Handler
);
}
/**
* RT-Thread Audio Device Driver Interface
*/
static
rt_err_t
sound_getcaps
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_QUERY
:
/* qurey the types of hw_codec device */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_TYPE_QUERY
:
caps
->
udata
.
mask
=
AUDIO_TYPE_OUTPUT
|
AUDIO_TYPE_MIXER
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_OUTPUT
:
/* Provide capabilities of OUTPUT unit */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
caps
->
udata
.
config
.
samplerate
=
snd_dev
->
replay_config
.
samplerate
;
caps
->
udata
.
config
.
channels
=
snd_dev
->
replay_config
.
channels
;
caps
->
udata
.
config
.
samplebits
=
snd_dev
->
replay_config
.
samplebits
;
break
;
case
AUDIO_DSP_SAMPLERATE
:
caps
->
udata
.
config
.
samplerate
=
snd_dev
->
replay_config
.
samplerate
;
break
;
case
AUDIO_DSP_CHANNELS
:
caps
->
udata
.
config
.
channels
=
snd_dev
->
replay_config
.
channels
;
break
;
case
AUDIO_DSP_SAMPLEBITS
:
caps
->
udata
.
config
.
samplebits
=
snd_dev
->
replay_config
.
samplebits
;
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_MIXER
:
/* report the Mixer Units */
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_QUERY
:
caps
->
udata
.
mask
=
AUDIO_MIXER_VOLUME
;
break
;
case
AUDIO_MIXER_VOLUME
:
caps
->
udata
.
value
=
es8388_volume_get
();
break
;
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
return
result
;
}
static
rt_err_t
sound_configure
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_caps
*
caps
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
switch
(
caps
->
main_type
)
{
case
AUDIO_TYPE_MIXER
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_MIXER_VOLUME
:
{
rt_uint8_t
volume
=
caps
->
udata
.
value
;
es8388_volume_set
(
volume
);
snd_dev
->
volume
=
volume
;
LOG_D
(
"set volume %d"
,
volume
);
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
case
AUDIO_TYPE_OUTPUT
:
{
switch
(
caps
->
sub_type
)
{
case
AUDIO_DSP_PARAM
:
{
/* set samplerate */
SAIA_Frequency_Set
(
caps
->
udata
.
config
.
samplerate
);
/* set channels */
SAIA_Channels_Set
(
caps
->
udata
.
config
.
channels
);
/* save configs */
snd_dev
->
replay_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
snd_dev
->
replay_config
.
channels
=
caps
->
udata
.
config
.
channels
;
snd_dev
->
replay_config
.
samplebits
=
caps
->
udata
.
config
.
samplebits
;
LOG_D
(
"set samplerate %d"
,
snd_dev
->
replay_config
.
samplerate
);
break
;
}
case
AUDIO_DSP_SAMPLERATE
:
{
SAIA_Frequency_Set
(
caps
->
udata
.
config
.
samplerate
);
snd_dev
->
replay_config
.
samplerate
=
caps
->
udata
.
config
.
samplerate
;
LOG_D
(
"set samplerate %d"
,
snd_dev
->
replay_config
.
samplerate
);
break
;
}
case
AUDIO_DSP_CHANNELS
:
{
SAIA_Channels_Set
(
caps
->
udata
.
config
.
channels
);
snd_dev
->
replay_config
.
channels
=
caps
->
udata
.
config
.
channels
;
LOG_D
(
"set channels %d"
,
snd_dev
->
replay_config
.
channels
);
break
;
}
case
AUDIO_DSP_SAMPLEBITS
:
{
/* not support */
snd_dev
->
replay_config
.
samplebits
=
caps
->
udata
.
config
.
samplebits
;
break
;
}
default:
result
=
-
RT_ERROR
;
break
;
}
break
;
}
default:
break
;
}
return
result
;
}
static
rt_err_t
sound_init
(
struct
rt_audio_device
*
audio
)
{
rt_err_t
result
=
RT_EOK
;
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
es8388_init
(
"i2c3"
,
GET_PIN
(
A
,
5
));
SAIA_Init
();
/* set default params */
SAIA_Frequency_Set
(
snd_dev
->
replay_config
.
samplerate
);
SAIA_Channels_Set
(
snd_dev
->
replay_config
.
channels
);
return
result
;
}
static
rt_err_t
sound_start
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
if
(
stream
==
AUDIO_STREAM_REPLAY
)
{
LOG_D
(
"open sound device"
);
es8388_start
(
ES_MODE_DAC
);
HAL_SAI_Transmit_DMA
(
&
SAI1A_Handler
,
snd_dev
->
tx_fifo
,
TX_FIFO_SIZE
/
2
);
}
return
RT_EOK
;
}
static
rt_err_t
sound_stop
(
struct
rt_audio_device
*
audio
,
int
stream
)
{
RT_ASSERT
(
audio
!=
RT_NULL
);
if
(
stream
==
AUDIO_STREAM_REPLAY
)
{
HAL_SAI_DMAStop
(
&
SAI1A_Handler
);
es8388_stop
(
ES_MODE_DAC
);
LOG_D
(
"close sound device"
);
}
return
RT_EOK
;
}
static
void
sound_buffer_info
(
struct
rt_audio_device
*
audio
,
struct
rt_audio_buf_info
*
info
)
{
struct
sound_device
*
snd_dev
;
RT_ASSERT
(
audio
!=
RT_NULL
);
snd_dev
=
(
struct
sound_device
*
)
audio
->
parent
.
user_data
;
/**
* TX_FIFO
* +----------------+----------------+
* | block1 | block2 |
* +----------------+----------------+
* \ block_size /
*/
info
->
buffer
=
snd_dev
->
tx_fifo
;
info
->
total_size
=
TX_FIFO_SIZE
;
info
->
block_size
=
TX_FIFO_SIZE
/
2
;
info
->
block_count
=
2
;
}
static
struct
rt_audio_ops
snd_ops
=
{
.
getcaps
=
sound_getcaps
,
.
configure
=
sound_configure
,
.
init
=
sound_init
,
.
start
=
sound_start
,
.
stop
=
sound_stop
,
.
transmit
=
RT_NULL
,
.
buffer_info
=
sound_buffer_info
,
};
int
rt_hw_sound_init
(
void
)
{
rt_uint8_t
*
tx_fifo
;
if
(
snd_dev
.
tx_fifo
)
return
RT_EOK
;
tx_fifo
=
rt_malloc
(
TX_FIFO_SIZE
);
if
(
tx_fifo
==
RT_NULL
)
return
-
RT_ENOMEM
;
rt_memset
(
tx_fifo
,
0
,
TX_FIFO_SIZE
);
snd_dev
.
tx_fifo
=
tx_fifo
;
/* init default configuration */
{
snd_dev
.
replay_config
.
samplerate
=
44100
;
snd_dev
.
replay_config
.
channels
=
2
;
snd_dev
.
replay_config
.
samplebits
=
16
;
snd_dev
.
volume
=
55
;
}
/* register sound device */
snd_dev
.
audio
.
ops
=
&
snd_ops
;
rt_audio_register
(
&
snd_dev
.
audio
,
"sound0"
,
RT_DEVICE_FLAG_WRONLY
,
&
snd_dev
);
return
RT_EOK
;
}
INIT_DEVICE_EXPORT
(
rt_hw_sound_init
);
bsp/stm32/stm32l475-atk-pandora/board/ports/audio/drv_sound.h
0 → 100644
浏览文件 @
fcfa986b
/*
* Copyright (c) 2006-2018, RT-Thread Development Team
*
* SPDX-License-Identifier: Apache-2.0
*
* Date Author Notes
* 2019-07-31 Zero-Free first implementation
*/
#ifndef __DRV_SOUND_H__
#define __DRV_SOUND_H__
int
rt_hw_sound_init
(
void
);
int
rt_hw_mic_init
(
void
);
#endif
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