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audio: nxstyle fixes for core and drivers

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nxstyle fixes for the audio core and drivers

Signed-off-by: Alin Jerpelea <alin.jerpelea@sony.com>
This commit is contained in:
Alin Jerpelea 2020-04-21 09:45:24 +02:00 committed by patacongo
parent 76c47f6b21
commit d9d720b296
24 changed files with 660 additions and 434 deletions
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111
audio/audio.c
View file

@ -68,6 +68,7 @@
****************************************************************************/
/* Debug ********************************************************************/
/* Non-standard debug that may be enabled just for testing Audio */
#ifndef AUDIO_MAX_DEVICE_PATH
@ -86,11 +87,11 @@
struct audio_upperhalf_s
{
uint8_t crefs; /* The number of times the device has been opened */
volatile bool started; /* True: playback is active */
sem_t exclsem; /* Supports mutual exclusion */
uint8_t crefs; /* The number of times the device has been opened */
volatile bool started; /* True: playback is active */
sem_t exclsem; /* Supports mutual exclusion */
FAR struct audio_lowerhalf_s *dev; /* lower-half state */
mqd_t usermq; /* User mode app's message queue */
mqd_t usermq; /* User mode app's message queue */
};
/****************************************************************************
@ -99,17 +100,29 @@ struct audio_upperhalf_s
static int audio_open(FAR struct file *filep);
static int audio_close(FAR struct file *filep);
static ssize_t audio_read(FAR struct file *filep, FAR char *buffer, size_t buflen);
static ssize_t audio_write(FAR struct file *filep, FAR const char *buffer, size_t buflen);
static int audio_ioctl(FAR struct file *filep, int cmd, unsigned long arg);
static ssize_t audio_read(FAR struct file *filep,
FAR char *buffer,
size_t buflen);
static ssize_t audio_write(FAR struct file *filep,
FAR const char *buffer,
size_t buflen);
static int audio_ioctl(FAR struct file *filep,
int cmd,
unsigned long arg);
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int audio_start(FAR struct audio_upperhalf_s *upper, FAR void *session);
static void audio_callback(FAR void *priv, uint16_t reason,
FAR struct ap_buffer_s *apb, uint16_t status, FAR void *session);
static int audio_start(FAR struct audio_upperhalf_s *upper,
FAR void *session);
static void audio_callback(FAR void *priv,
uint16_t reason,
FAR struct ap_buffer_s *apb,
uint16_t status,
FAR void *session);
#else
static int audio_start(FAR struct audio_upperhalf_s *upper);
static void audio_callback(FAR void *priv, uint16_t reason,
FAR struct ap_buffer_s *apb, uint16_t status);
static void audio_callback(FAR void *priv,
uint16_t reason,
FAR struct ap_buffer_s *apb,
uint16_t status);
#endif /* CONFIG_AUDIO_MULTI_SESSION */
/****************************************************************************
@ -131,13 +144,13 @@ static const struct file_operations g_audioops =
* Private Functions
****************************************************************************/
/************************************************************************************
/****************************************************************************
* Name: audio_open
*
* Description:
* This function is called whenever the Audio device is opened.
*
************************************************************************************/
****************************************************************************/
static int audio_open(FAR struct file *filep)
{
@ -184,13 +197,13 @@ errout:
return ret;
}
/************************************************************************************
/****************************************************************************
* Name: audio_close
*
* Description:
* This function is called when the Audio device is closed.
*
************************************************************************************/
****************************************************************************/
static int audio_close(FAR struct file *filep)
{
@ -241,15 +254,17 @@ errout:
return ret;
}
/************************************************************************************
/****************************************************************************
* Name: audio_read
*
* Description:
* A dummy read method. This is provided only to satsify the VFS layer.
*
************************************************************************************/
****************************************************************************/
static ssize_t audio_read(FAR struct file *filep, FAR char *buffer, size_t buflen)
static ssize_t audio_read(FAR struct file *filep,
FAR char *buffer,
size_t buflen)
{
FAR struct inode *inode = filep->f_inode;
FAR struct audio_upperhalf_s *upper = inode->i_private;
@ -267,15 +282,17 @@ static ssize_t audio_read(FAR struct file *filep, FAR char *buffer, size_t bufle
return 0;
}
/************************************************************************************
/****************************************************************************
* Name: audio_write
*
* Description:
* A dummy write method. This is provided only to satsify the VFS layer.
*
************************************************************************************/
****************************************************************************/
static ssize_t audio_write(FAR struct file *filep, FAR const char *buffer, size_t buflen)
static ssize_t audio_write(FAR struct file *filep,
FAR const char *buffer,
size_t buflen)
{
FAR struct inode *inode = filep->f_inode;
FAR struct audio_upperhalf_s *upper = inode->i_private;
@ -293,16 +310,17 @@ static ssize_t audio_write(FAR struct file *filep, FAR const char *buffer, size_
return 0;
}
/************************************************************************************
/****************************************************************************
* Name: audio_start
*
* Description:
* Handle the AUDIOIOC_START ioctl command
*
************************************************************************************/
****************************************************************************/
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int audio_start(FAR struct audio_upperhalf_s *upper, FAR void *session)
static int audio_start(FAR struct audio_upperhalf_s *upper,
FAR void *session)
#else
static int audio_start(FAR struct audio_upperhalf_s *upper)
#endif
@ -339,13 +357,13 @@ static int audio_start(FAR struct audio_upperhalf_s *upper)
return ret;
}
/************************************************************************************
/****************************************************************************
* Name: audio_ioctl
*
* Description:
* The standard ioctl method. This is where ALL of the Audio work is done.
*
************************************************************************************/
****************************************************************************/
static int audio_ioctl(FAR struct file *filep, int cmd, unsigned long arg)
{
@ -379,7 +397,8 @@ static int audio_ioctl(FAR struct file *filep, int cmd, unsigned long arg)
case AUDIOIOC_GETCAPS:
{
FAR struct audio_caps_s *caps = (FAR struct audio_caps_s *)((uintptr_t)arg);
FAR struct audio_caps_s *caps =
(FAR struct audio_caps_s *)((uintptr_t)arg);
DEBUGASSERT(lower->ops->getcaps != NULL);
audinfo("AUDIOIOC_GETCAPS: Device=%d\n", caps->ac_type);
@ -420,7 +439,8 @@ static int audio_ioctl(FAR struct file *filep, int cmd, unsigned long arg)
}
break;
/* AUDIOIOC_START - Start the audio stream. The AUDIOIOC_SETCHARACTERISTICS
/* AUDIOIOC_START - Start the audio stream.
* The AUDIOIOC_SETCHARACTERISTICS
* command must have previously been sent.
*
* ioctl argument: Audio session
@ -647,7 +667,9 @@ static int audio_ioctl(FAR struct file *filep, int cmd, unsigned long arg)
}
break;
/* Any unrecognized IOCTL commands might be platform-specific ioctl commands */
/* Any unrecognized IOCTL commands might be
* platform-specific ioctl commands
*/
default:
{
@ -711,8 +733,8 @@ static inline void audio_dequeuebuffer(FAR struct audio_upperhalf_s *upper,
if (upper->usermq != NULL)
{
msg.msgId = AUDIO_MSG_DEQUEUE;
msg.u.pPtr = apb;
msg.msg_id = AUDIO_MSG_DEQUEUE;
msg.u.ptr = apb;
#ifdef CONFIG_AUDIO_MULTI_SESSION
msg.session = session;
#endif
@ -750,8 +772,8 @@ static inline void audio_complete(FAR struct audio_upperhalf_s *upper,
upper->started = false;
if (upper->usermq != NULL)
{
msg.msgId = AUDIO_MSG_COMPLETE;
msg.u.pPtr = NULL;
msg.msg_id = AUDIO_MSG_COMPLETE;
msg.u.ptr = NULL;
#ifdef CONFIG_AUDIO_MULTI_SESSION
msg.session = session;
#endif
@ -823,7 +845,8 @@ static void audio_callback(FAR void *handle, uint16_t reason,
FAR struct ap_buffer_s *apb, uint16_t status)
#endif
{
FAR struct audio_upperhalf_s *upper = (FAR struct audio_upperhalf_s *)handle;
FAR struct audio_upperhalf_s *upper =
(FAR struct audio_upperhalf_s *)handle;
audinfo("Entry\n");
@ -854,7 +877,9 @@ static void audio_callback(FAR void *handle, uint16_t reason,
case AUDIO_CALLBACK_COMPLETE:
{
/* Send a complete message to the user if a message queue is registered */
/* Send a complete message to the user if a message queue
* is registered
*/
#ifdef CONFIG_AUDIO_MULTI_SESSION
audio_complete(upper, apb, status, session);
@ -902,9 +927,9 @@ static void audio_callback(FAR void *handle, uint16_t reason,
* filesystem. The recommended convention is to name Audio drivers
* based on the function they provide, such as "/dev/pcm0", "/dev/mp31",
* etc.
* dev - A pointer to an instance of lower half audio driver. This instance
* is bound to the Audio driver and must persists as long as the driver
* persists.
* dev - A pointer to an instance of lower half audio driver.
* This instance is bound to the Audio driver and must persists as long
* as the driver persists.
*
* Returned Value:
* Zero on success; a negated errno value on failure.
@ -926,14 +951,17 @@ int audio_register(FAR const char *name, FAR struct audio_lowerhalf_s *dev)
/* Allocate the upper-half data structure */
upper = (FAR struct audio_upperhalf_s *)kmm_zalloc(sizeof(struct audio_upperhalf_s));
upper = (FAR struct audio_upperhalf_s *)kmm_zalloc(
sizeof(struct audio_upperhalf_s));
if (!upper)
{
auderr("ERROR: Allocation failed\n");
return -ENOMEM;
}
/* Initialize the Audio device structure (it was already zeroed by kmm_zalloc()) */
/* Initialize the Audio device structure
* (it was already zeroed by kmm_zalloc())
*/
nxsem_init(&upper->exclsem, 0, 1);
upper->dev = dev;
@ -979,6 +1007,7 @@ int audio_register(FAR const char *name, FAR struct audio_lowerhalf_s *dev)
{
*pathptr++ = *ptr++;
}
*pathptr = '\0';
/* Make this level of directory */

6
audio/audio_comp.c
View file

@ -139,11 +139,13 @@ static int audio_comp_release(FAR struct audio_lowerhalf_s *dev);
#ifdef CONFIG_AUDIO_MULTI_SESSION
static void audio_comp_callback(FAR void *arg, uint16_t reason,
FAR struct ap_buffer_s *apb, uint16_t status,
FAR struct ap_buffer_s *apb,
uint16_t status,
FAR void *session);
#else
static void audio_comp_callback(FAR void *arg, uint16_t reason,
FAR struct ap_buffer_s *apb, uint16_t status);
FAR struct ap_buffer_s *apb,
uint16_t status);
#endif
/****************************************************************************

55
audio/pcm_decode.c
View file

@ -215,13 +215,15 @@ static int pcm_ioctl(FAR struct audio_lowerhalf_s *dev,
int cmd, unsigned long arg);
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int pcm_reserve(FAR struct audio_lowerhalf_s *dev, FAR void **session);
static int pcm_reserve(FAR struct audio_lowerhalf_s *dev,
FAR void **session);
#else
static int pcm_reserve(FAR struct audio_lowerhalf_s *dev);
#endif
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int pcm_release(FAR struct audio_lowerhalf_s *dev, FAR void *session);
static int pcm_release(FAR struct audio_lowerhalf_s *dev,
FAR void *session);
#else
static int pcm_release(FAR struct audio_lowerhalf_s *dev);
#endif
@ -399,14 +401,14 @@ static bool pcm_parsewav(FAR struct pcm_decode_s *priv, uint8_t *data)
if (priv->bpsamp != 8 && priv->bpsamp != 16)
{
auderr("ERROR: Cannot support bits per sample of %d in this mode\n",
auderr("ERROR: %d bits per sample are not suported in this mode\n",
priv->bpsamp);
return -EINVAL;
}
if (priv->nchannels != 1 && priv->nchannels != 2)
{
auderr("ERROR: Cannot support number of channels of %d in this mode\n",
auderr("ERROR: %d channels are not supported in this mode\n",
priv->nchannels);
return -EINVAL;
}
@ -637,7 +639,9 @@ static void pcm_subsample(FAR struct pcm_decode_s *priv,
priv->skip = skipsize;
}
/* Now copy the sample from the end of audio buffer to the beginning. */
/* Now copy the sample from the end of audio buffer
* to the beginning.
*/
for (i = 0; i < copysize; i++)
{
@ -666,10 +670,11 @@ static void pcm_subsample(FAR struct pcm_decode_s *priv,
* Description:
* This method is called to retrieve the lower-half device capabilities.
* It will be called with device type AUDIO_TYPE_QUERY to request the
* overall capabilities, such as to determine the types of devices supported
* audio formats supported, etc. Then it may be called once or more with
* reported supported device types to determine the specific capabilities
* of that device type (such as MP3 encoder, WMA encoder, PCM output, etc.).
* overall capabilities, such as to determine the types of devices
* supported audio formats supported, etc.
* Then it may be called once or more with reported supported device types
* to determine the specific capabilities of that device type
* (such as MP3 encoder, WMA encoder, PCM output, etc.).
*
****************************************************************************/
@ -696,9 +701,9 @@ static int pcm_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
return ret;
}
/* Modify the capabilities reported by the lower driver: PCM is the only
* supported format that we will report, regardless of what the lower driver
* reported.
/* Modify the capabilities reported by the lower driver:
* PCM is the only supported format that we will report,
* regardless of what the lower driver reported.
*/
if (caps->ac_subtype == AUDIO_TYPE_QUERY)
@ -724,7 +729,8 @@ static int pcm_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int pcm_configure(FAR struct audio_lowerhalf_s *dev,
FAR void *session, FAR const struct audio_caps_s *caps)
FAR void *session,
FAR const struct audio_caps_s *caps)
#else
static int pcm_configure(FAR struct audio_lowerhalf_s *dev,
FAR const struct audio_caps_s *caps)
@ -776,8 +782,8 @@ static int pcm_configure(FAR struct audio_lowerhalf_s *dev,
* output generation. It should also disable the audio hardware and put
* it into the lowest possible power usage state.
*
* Any enqueued Audio Pipeline Buffers that have not been processed / dequeued
* should be dequeued by this function.
* Any enqueued Audio Pipeline Buffers that have not been
* processed / dequeued should be dequeued by this function.
*
****************************************************************************/
@ -805,10 +811,10 @@ static int pcm_shutdown(FAR struct audio_lowerhalf_s *dev)
* Name: pcm_start
*
* Description:
* Start audio streaming in the configured mode. For input and synthesis
* devices, this means it should begin sending streaming audio data. For output
* or processing type device, it means it should begin processing of any enqueued
* Audio Pipeline Buffers.
* Start audio streaming in the configured mode.
* For input and synthesis devices, this means it should begin sending
* streaming audio data. For output or processing type device, it means
* it should begin processing of any enqueued Audio Pipeline Buffers.
*
****************************************************************************/
@ -879,8 +885,9 @@ static int pcm_stop(FAR struct audio_lowerhalf_s *dev)
* Name: pcm_pause
*
* Description:
* Pause the audio stream. Should keep current playback context active
* in case a resume is issued. Could be called and then followed by a stop.
* Pause the audio stream.
* Should keep current playback context active in case a resume is issued.
* Could be called and then followed by a stop.
*
****************************************************************************/
@ -1037,6 +1044,7 @@ static int pcm_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
if (priv->streaming)
{
/* Yes, we are streaming */
/* Check for the last audio buffer in the stream */
if ((apb->flags & AUDIO_APB_FINAL) != 0)
@ -1120,8 +1128,9 @@ static int pcm_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
/* Then give the audio buffer to the lower driver */
audinfo("Pass to lower enqueuebuffer: apb=%p curbyte=%d nbytes=%d\n",
apb, apb->curbyte, apb->nbytes);
audinfo(
"Pass to lower enqueuebuffer: apb=%p curbyte=%d nbytes=%d\n",
apb, apb->curbyte, apb->nbytes);
ret = lower->ops->enqueuebuffer(lower, apb);
if (ret == OK)

4
drivers/audio/audio_i2s.c
View file

@ -232,7 +232,9 @@ static int audio_i2s_configure(FAR struct audio_lowerhalf_s *dev,
{
FAR struct audio_i2s_s *audio_i2s = (struct audio_i2s_s *)dev;
FAR struct i2s_dev_s *i2s;
int samprate, nchannels, bpsamp;
int samprate;
int nchannels;
int bpsamp;
int ret = OK;
DEBUGASSERT(audio_i2s != NULL && caps != NULL);

59
drivers/audio/audio_null.c
View file

@ -211,12 +211,14 @@ static int null_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
/* The types of audio units we implement */
caps->ac_controls.b[0] = AUDIO_TYPE_OUTPUT | AUDIO_TYPE_FEATURE |
caps->ac_controls.b[0] = AUDIO_TYPE_OUTPUT |
AUDIO_TYPE_FEATURE |
AUDIO_TYPE_PROCESSING;
break;
case AUDIO_FMT_MIDI:
/* We only support Format 0 */
caps->ac_controls.b[0] = AUDIO_SUBFMT_END;
@ -241,9 +243,12 @@ static int null_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
/* Report the Sample rates we support */
caps->ac_controls.b[0] = AUDIO_SAMP_RATE_8K | AUDIO_SAMP_RATE_11K |
AUDIO_SAMP_RATE_16K | AUDIO_SAMP_RATE_22K |
AUDIO_SAMP_RATE_32K | AUDIO_SAMP_RATE_44K |
caps->ac_controls.b[0] = AUDIO_SAMP_RATE_8K |
AUDIO_SAMP_RATE_11K |
AUDIO_SAMP_RATE_16K |
AUDIO_SAMP_RATE_22K |
AUDIO_SAMP_RATE_32K |
AUDIO_SAMP_RATE_44K |
AUDIO_SAMP_RATE_48K;
break;
@ -262,19 +267,25 @@ static int null_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
case AUDIO_TYPE_FEATURE:
/* If the sub-type is UNDEF, then report the Feature Units we support */
/* If the sub-type is UNDEF,
* then report the Feature Units we support
*/
if (caps->ac_subtype == AUDIO_FU_UNDEF)
{
/* Fill in the ac_controls section with the Feature Units we have */
/* Fill in the ac_controls section with
* the Feature Units we have
*/
caps->ac_controls.b[0] = AUDIO_FU_VOLUME | AUDIO_FU_BASS | AUDIO_FU_TREBLE;
caps->ac_controls.b[0] = AUDIO_FU_VOLUME |
AUDIO_FU_BASS |
AUDIO_FU_TREBLE;
caps->ac_controls.b[1] = AUDIO_FU_BALANCE >> 8;
}
else
{
/* TODO: Do we need to provide specific info for the Feature Units,
* such as volume setting ranges, etc.?
/* TODO: Do we need to provide specific info for the
* Feature Units, such as volume setting ranges, etc.?
*/
}
@ -297,7 +308,8 @@ static int null_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
/* Provide capabilities of our Stereo Extender */
caps->ac_controls.b[0] = AUDIO_STEXT_ENABLE | AUDIO_STEXT_WIDTH;
caps->ac_controls.b[0] = AUDIO_STEXT_ENABLE |
AUDIO_STEXT_WIDTH;
break;
default:
@ -450,7 +462,7 @@ static void *null_workerthread(pthread_addr_t pvarg)
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
case AUDIO_MSG_DATA_REQUEST:
break;
@ -468,7 +480,7 @@ static void *null_workerthread(pthread_addr_t pvarg)
break;
default:
auderr("ERROR: Ignoring message ID %d\n", msg.msgId);
auderr("ERROR: Ignoring message ID %d\n", msg.msg_id);
break;
}
}
@ -584,11 +596,12 @@ static int null_stop(FAR struct audio_lowerhalf_s *dev)
FAR void *value;
/* Send a message to stop all audio streaming */
/* REVISIT: There should be a check to see if the worker thread is still
* running.
/* REVISIT:
* There should be a check to see if the worker thread is still running.
*/
term_msg.msgId = AUDIO_MSG_STOP;
term_msg.msg_id = AUDIO_MSG_STOP;
term_msg.u.data = 0;
nxmq_send(priv->mq, (FAR const char *)&term_msg, sizeof(term_msg),
CONFIG_AUDIO_NULL_MSG_PRIO);
@ -681,9 +694,16 @@ static int null_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
if (done)
{
#ifdef CONFIG_AUDIO_MULTI_SESSION
priv->dev.upper(priv->dev.priv, AUDIO_CALLBACK_COMPLETE, NULL, OK, NULL);
priv->dev.upper(priv->dev.priv,
AUDIO_CALLBACK_COMPLETE,
NULL,
OK,
NULL);
#else
priv->dev.upper(priv->dev.priv, AUDIO_CALLBACK_COMPLETE, NULL, OK);
priv->dev.upper(priv->dev.priv,
AUDIO_CALLBACK_COMPLETE,
NULL,
OK);
#endif
}
@ -834,8 +854,9 @@ FAR struct audio_lowerhalf_s *audio_null_initialize(void)
priv = (FAR struct null_dev_s *)kmm_zalloc(sizeof(struct null_dev_s));
if (priv)
{
/* Initialize the null audio device structure. Since we used kmm_zalloc,
* only the non-zero elements of the structure need to be initialized.
/* Initialize the null audio device structure.
* Since we used kmm_zalloc, only the non-zero elements
* of the structure need to be initialized.
*/
priv->dev.ops = &g_audioops;

10
drivers/audio/cs43l22.c
View file

@ -1014,7 +1014,7 @@ cs43l22_senddone(FAR struct i2s_dev_s *i2s,
* buffers in the done queue that need to be cleaned up.
*/
msg.msgId = AUDIO_MSG_COMPLETE;
msg.msg_id = AUDIO_MSG_COMPLETE;
ret = nxmq_send(priv->mq, (FAR const char *)&msg, sizeof(msg),
CONFIG_CS43L22_MSG_PRIO);
if (ret < 0)
@ -1282,7 +1282,7 @@ static int cs43l22_stop(FAR struct audio_lowerhalf_s *dev)
/* Send a message to stop all audio streaming */
term_msg.msgId = AUDIO_MSG_STOP;
term_msg.msg_id = AUDIO_MSG_STOP;
term_msg.u.data = 0;
nxmq_send(priv->mq, (FAR const char *)&term_msg, sizeof(term_msg),
CONFIG_CS43L22_MSG_PRIO);
@ -1409,7 +1409,7 @@ static int cs43l22_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
ret = OK;
if (priv->mq != NULL)
{
term_msg.msgId = AUDIO_MSG_ENQUEUE;
term_msg.msg_id = AUDIO_MSG_ENQUEUE;
term_msg.u.data = 0;
ret = nxmq_send(priv->mq, (FAR const char *)&term_msg,
@ -1692,7 +1692,7 @@ static void *cs43l22_workerthread(pthread_addr_t pvarg)
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
/* The ISR has requested more data. We will catch this case at
* the top of the loop.
@ -1730,7 +1730,7 @@ static void *cs43l22_workerthread(pthread_addr_t pvarg)
break;
default:
auderr("ERROR: Ignoring message ID %d\n", msg.msgId);
auderr("ERROR: Ignoring message ID %d\n", msg.msg_id);
break;
}
}

57
drivers/audio/cs43l22.h
View file

@ -63,6 +63,7 @@
/****************************************************************************
* Pre-Processor Definitions
****************************************************************************/
/* So far, I have not been able to get FLL lock interrupts. Worse, I have
* been able to get the FLL to claim that it is locked at all even when
* polling. What am I doing wrong?
@ -120,8 +121,9 @@
#define CS43L22_SPKAMUTE (1 << 4)
/* Register Default Values **************************************************/
/* Registers have some undocumented bits set on power up. These probably
* should be retained on writes (?).
/* Registers have some undocumented bits set on power up.
* These probably should be retained on writes (?).
*/
#define CS43L22_ID_REV_DEFAULT 0xe3 /* Chip I.D. and Revision */
@ -179,33 +181,41 @@
/* 0x04 Power Control 2 */
#define CS43L22_PDN_HPB_SHIFT (6) /* Bits 6-7: Headphone channel B Control */
#define CS43L22_PDN_HPB_ON_HW_PIN_LO (0 << CS43L22_PDN_HPB_SHIFT) /* PDN_HPx[1:0] 00 Headphone channel is ON when the SPK/HP_SW pin, 6, is LO
Headphone channel is OFF when the SPK/HP_SW pin, 6, is HI */
* Headphone channel is OFF when the SPK/HP_SW pin, 6, is HI
*/
#define CS43L22_PDN_HPB_ON_HW_PIN_HI (1 << CS43L22_PDN_HPB_SHIFT) /* PDN_HPx[1:0] 01 Headphone channel is ON when the SPK/HP_SW pin, 6, is HI
Headphone channel is OFF when the SPK/HP_SW pin, 6, is LO */
* Headphone channel is OFF when the SPK/HP_SW pin, 6, is LO
*/
#define CS43L22_PDN_HPB_ON (2 << CS43L22_PDN_HPB_SHIFT) /* PDN_HPx[1:0] 10 Headphone channel is always ON */
#define CS43L22_PDN_HPB_OFF (3 << CS43L22_PDN_HPB_SHIFT) /* PDN_HPx[1:0] 11 Headphone channel is always OFF */
#define CS43L22_PDN_HPA_SHIFT (4) /* Bits 4-5: Headphone channel A Control */
#define CS43L22_PDN_HPA_ON_HW_PIN_LO (0 << CS43L22_PDN_HPA_SHIFT) /* PDN_HPx[1:0] 00 Headphone channel is ON when the SPK/HP_SW pin, 6, is LO
Headphone channel is OFF when the SPK/HP_SW pin, 6, is HI */
* Headphone channel is OFF when the SPK/HP_SW pin, 6, is HI */
#define CS43L22_PDN_HPA_ON_HW_PIN_HI (1 << CS43L22_PDN_HPA_SHIFT) /* PDN_HPx[1:0] 01 Headphone channel is ON when the SPK/HP_SW pin, 6, is HI
Headphone channel is OFF when the SPK/HP_SW pin, 6, is LO */
* Headphone channel is OFF when the SPK/HP_SW pin, 6, is LO
*/
#define CS43L22_PDN_HPA_ON (2 << CS43L22_PDN_HPA_SHIFT) /* PDN_HPx[1:0] 10 Headphone channel is always ON */
#define CS43L22_PDN_HPA_OFF (3 << CS43L22_PDN_HPA_SHIFT) /* PDN_HPx[1:0] 11 Headphone channel is always OFF */
#define CS43L22_PDN_SPKB_SHIFT (2) /* Bits 2-3: Speaker channel B Control */
#define CS43L22_PDN_SPKB_ON_HW_PIN_LO (0 << CS43L22_PDN_SPKB_SHIFT) /* PDN_HPx[1:0] 00 Speaker channel is ON when the SPK/HP_SW pin, 6, is LO
Speaker channel is OFF when the SPK/HP_SW pin, 6, is HI */
* Speaker channel is OFF when the SPK/HP_SW pin, 6, is HI
*/
#define CS43L22_PDN_SPKB_ON_HW_PIN_HI (1 << CS43L22_PDN_SPKB_SHIFT) /* PDN_HPx[1:0] 01 Speaker channel is ON when the SPK/HP_SW pin, 6, is HI
Speaker channel is OFF when the SPK/HP_SW pin, 6, is LO */
* Speaker channel is OFF when the SPK/HP_SW pin, 6, is LO
*/
#define CS43L22_PDN_SPKB_ON (2 << CS43L22_PDN_SPKB_SHIFT) /* PDN_HPx[1:0] 10 Speaker channel is always ON */
#define CS43L22_PDN_SPKB_OFF (3 << CS43L22_PDN_SPKB_SHIFT) /* PDN_HPx[1:0] 11 Speaker channel is always OFF */
#define CS43L22_PDN_SPKA_SHIFT (0) /* Bits 0-1: Speaker channel A Control */
#define CS43L22_PDN_SPKA_ON_HW_PIN_LO (0 << CS43L22_PDN_SPKA_SHIFT) /* PDN_HPx[1:0] 00 Speaker channel is ON when the SPK/HP_SW pin, 6, is LO
Speaker channel is OFF when the SPK/HP_SW pin, 6, is HI */
* Speaker channel is OFF when the SPK/HP_SW pin, 6, is HI
*/
#define CS43L22_PDN_SPKA_ON_HW_PIN_HI (1 << CS43L22_PDN_SPKA_SHIFT) /* PDN_HPx[1:0] 01 Speaker channel is ON when the SPK/HP_SW pin, 6, is HI
Speaker channel is OFF when the SPK/HP_SW pin, 6, is LO */
* Speaker channel is OFF when the SPK/HP_SW pin, 6, is LO
*/
#define CS43L22_PDN_SPKA_ON (2 << CS43L22_PDN_SPKA_SHIFT) /* PDN_HPx[1:0] 10 Speaker channel is always ON */
#define CS43L22_PDN_SPKA_OFF (3 << CS43L22_PDN_SPKA_SHIFT) /* PDN_HPx[1:0] 11 Speaker channel is always OFF */
@ -223,6 +233,7 @@
#define CS43L22_VIDEOCLK_ENABLE (1 << 3) /* Bit 3: Specifies whether or not the external MCLK frequency is 27 MHz */
#define CS43L22_MCLK_LRCK_RATIO_SHIFT (1) /* Bits 1-2: Internal MCLK/LRCK Ratio */
#define CS43L22_RATIO_128_64 (0 << CS43L22_MCLK_LRCK_RATIO_SHIFT) /* RATIO[1:0] Internal MCLK Cycles per LRCK=128, SCLK/LRCK=64 Ratio in Master Mode */
#define CS43L22_RATIO_125_62 (1 << CS43L22_MCLK_LRCK_RATIO_SHIFT) /* RATIO[1:0] Internal MCLK Cycles per LRCK=125, SCLK/LRCK=62 Ratio in Master Mode */
#define CS43L22_RATIO_132_66 (2 << CS43L22_MCLK_LRCK_RATIO_SHIFT) /* RATIO[1:0] Internal MCLK Cycles per LRCK=132, SCLK/LRCK=66 Ratio in Master Mode */
@ -237,17 +248,17 @@
#define CS43L22_DSP_MODE_ENABLE (1 << 4) /* Configures a data-packed interface format for the DAC */
#define CS43L22_DAC_IF_FORMAT_SHIFT (2) /* Bits 2-3: Configures the digital interface format for data on SDIN */
#define CS43L22_DAC_IF_FORMAT_SHIFT (2) /* Bits 2-3: Configures the digital interface format for data on SDIN */
#define CS43L22_DAC_IF_LEFT_JUSTIFIED (0 << CS43L22_DAC_IF_FORMAT_SHIFT) /* DACDIF[1:0] Left Justified, up to 24-bit data */
#define CS43L22_DAC_IF_I2S (1 << CS43L22_DAC_IF_FORMAT_SHIFT) /* DACDIF[1:0] I2S, up to 24-bit data */
#define CS43L22_DAC_IF_RIGHT_JUSTIFIED (2 << CS43L22_DAC_IF_FORMAT_SHIFT) /* DACDIF[1:0] Right Justified */
#define CS43L22_DAC_IF_RESERVED (3 << CS43L22_DAC_IF_FORMAT_SHIFT) /* DACDIF[1:0] Reserved */
#define CS43L22_AUDIO_WORD_LENGHT_SHIFT (0) /* Bits 0-1: Configures the audio sample word length used for the data into SDIN */
#define CS43L22_AWL_DSP_32_RJ_24 (0 << CS43L22_AUDIO_WORD_LENGHT_SHIFT)/* AWL[1:0] DSP Mode: 32-bit data, Right Justified: 24-bit data */
#define CS43L22_AWL_DSP_24_RJ_20 (1 << CS43L22_AUDIO_WORD_LENGHT_SHIFT)/* AWL[1:0] DSP Mode: 24-bit data, Right Justified: 20-bit data */
#define CS43L22_AWL_DSP_20_RJ_18 (2 << CS43L22_AUDIO_WORD_LENGHT_SHIFT)/* AWL[1:0] DSP Mode: 20-bit data, Right Justified: 18-bit data */
#define CS43L22_AWL_DSP_16_RJ_16 (3 << CS43L22_AUDIO_WORD_LENGHT_SHIFT)/* AWL[1:0] DSP Mode: 16 bit data, Right Justified: 16-bit data */
#define CS43L22_AUDIO_WORD_LENGHT_SHIFT (0) /* Bits 0-1: Configures the audio sample word length used for the data into SDIN */
#define CS43L22_AWL_DSP_32_RJ_24 (0 << CS43L22_AUDIO_WORD_LENGHT_SHIFT) /* AWL[1:0] DSP Mode: 32-bit data, Right Justified: 24-bit data */
#define CS43L22_AWL_DSP_24_RJ_20 (1 << CS43L22_AUDIO_WORD_LENGHT_SHIFT) /* AWL[1:0] DSP Mode: 24-bit data, Right Justified: 20-bit data */
#define CS43L22_AWL_DSP_20_RJ_18 (2 << CS43L22_AUDIO_WORD_LENGHT_SHIFT) /* AWL[1:0] DSP Mode: 20-bit data, Right Justified: 18-bit data */
#define CS43L22_AWL_DSP_16_RJ_16 (3 << CS43L22_AUDIO_WORD_LENGHT_SHIFT) /* AWL[1:0] DSP Mode: 16 bit data, Right Justified: 16-bit data */
/* 0x0E Miscellaneous Controls */
#define CS43L22_FREEZE (1 << 3) /* Configures a hold on all register settings */
@ -256,7 +267,7 @@
/* 0x1F Tone Control */
#define CS43L22_TREB_GAIN_SHIFT (4) /* Sets the gain of the treble shelving filter */
#define CS43L22_TREB_GAIN(a) ((a) << CS43L22_TREB_GAIN_SHIFT)
/* TREB[3:0] Gain Setting:*/
/* TREB[3:0] Gain Setting: */
/* 0000 +12.0 dB */
/* ··· ··· */
/* 0111 +1.5 dB */
@ -267,7 +278,7 @@
#define CS43L22_BASS_GAIN_SHIFT (0) /* Sets the gain of the bass shelving filter */
#define CS43L22_BASS_GAIN(a) ((a) << CS43L22_BASS_GAIN_SHIFT)
/* BASS[3:0] Gain Setting:*/
/* BASS[3:0] Gain Setting: */
/* 0000 +12.0 dB */
/* ··· ··· */
/* 0111 +1.5 dB */
@ -276,7 +287,8 @@
/* 1111 -10.5 dB */
/* Step Size: 1.5 dB */
/* FLL Configuration *********************************************************/
/* FLL Configuration ********************************************************/
/* Default FLL configuration */
#define CS43L22_DEFAULT_SAMPRATE 11025 /* Initial sample rate */
@ -312,8 +324,9 @@ struct cs43l22_dev_s
/* We are an audio lower half driver (We are also the upper "half" of
* the CS43L22 driver with respect to the board lower half driver).
*
* Terminology: Our "lower" half audio instances will be called dev for the
* publicly visible version and "priv" for the version that only this driver
* Terminology:
* Our "lower" half audio instances will be called dev for the publicly
* visible version and "priv" for the version that only this driver
* knows. From the point of view of this driver, it is the board lower
* "half" that is referred to as "lower".
*/
@ -363,7 +376,7 @@ struct cs43l22_dev_s
****************************************************************************/
#ifdef CONFIG_CS43L22_CLKDEBUG
extern const uint8_t g_sysclk_scaleb1[CS43L22_BCLK_MAXDIV+1];
extern const uint8_t g_sysclk_scaleb1[CS43L22_BCLK_MAXDIV + 1];
extern const uint8_t g_fllratio[CS43L22_NFLLRATIO];
#endif

78
drivers/audio/cs43l22_debug.c
View file

@ -82,47 +82,47 @@ struct cs43l22_regdump_s
#ifdef CONFIG_CS43L22_REGDUMP
static const struct cs43l22_regdump_s g_cs43l22_debug[] =
{
{"CHIP_ID_REV", CS43L22_ID_REV },
{"POWER_CTRL1", CS43L22_POWER_CTRL1 },
{"POWER_CTRL2", CS43L22_POWER_CTRL2 },
{"CLOCK_CTRL", CS43L22_CLOCK_CTRL },
{"INTERFACE_CTRL1", CS43L22_INTERFACE_CTRL1 },
{"INTERFACE_CTRL2", CS43L22_INTERFACE_CTRL2 },
{"PASS_SEL_A", CS43L22_PASS_SEL_A },
{"PASS_SEL_B", CS43L22_PASS_SEL_B },
{"ANLG_ZC_SR_SEL", CS43L22_ANLG_ZC_SR_SEL },
{"PASS_GANG_CTRL", CS43L22_PASS_GANG_CTRL },
{"PLAYBACK_CTRL1", CS43L22_PLAYBACK_CTRL1 },
{"MISCLLNS_CTRL", CS43L22_MISCLLNS_CTRL },
{"PLAYBACK_CTRL2", CS43L22_PLAYBACK_CTRL2 },
{"PASS_VOL_A", CS43L22_PASS_VOL_A },
{"PASS_VOL_B", CS43L22_PASS_VOL_B },
{"PCM_VOL_A", CS43L22_PCM_VOL_A },
{"PCM_VOL_B", CS43L22_PCM_VOL_B },
{"BP_FREQ_ON_T", CS43L22_BP_FREQ_ON_TIME },
{"BP_VOL_OFF_T", CS43L22_BP_VOL_OFF_TIME },
{"BP_TONE_CFG", CS43L22_BP_TONE_CFG },
{"TONE_CTRL", CS43L22_TONE_CTRL },
{"MS_VOL_CTRL_A", CS43L22_MS_VOL_CTRL_A },
{"MS_VOL_CTRL_B", CS43L22_MS_VOL_CTRL_B },
{"HP_VOL_CTRL_A", CS43L22_HP_VOL_CTRL_A },
{"HP_VOL_CTRL_B", CS43L22_HP_VOL_CTRL_B },
{"SPK_VOL_CTRL_A", CS43L22_SPK_VOL_CTRL_A },
{"SPK_VOL_CTRL_B", CS43L22_SPK_VOL_CTRL_B },
{"PCM_CH_SWAP", CS43L22_PCM_CH_SWAP },
{"LIM_CTRL1", CS43L22_LIM_CTRL1 },
{"LIM_CTRL2", CS43L22_LIM_CTRL2 },
{"LIM_ATTACK_RATE", CS43L22_LIM_ATTACK_RATE },
{"STATUS", CS43L22_STATUS },
{"BAT_COMP", CS43L22_BAT_COMP },
{"VP_BAT_LEVEL", CS43L22_VP_BAT_LEVEL },
{"SPK_STATUS", CS43L22_SPK_STATUS },
{"TEMP_MON_CTRL", CS43L22_TEMP_MON_CTRL },
{"THERMAL_FOLDBACK",CS43L22_THERMAL_FOLDBACK},
{"CHRG_PUMP_FREQ", CS43L22_CHRG_PUMP_FREQ }
{"CHIP_ID_REV", CS43L22_ID_REV },
{"POWER_CTRL1", CS43L22_POWER_CTRL1 },
{"POWER_CTRL2", CS43L22_POWER_CTRL2 },
{"CLOCK_CTRL", CS43L22_CLOCK_CTRL },
{"INTERFACE_CTRL1", CS43L22_INTERFACE_CTRL1 },
{"INTERFACE_CTRL2", CS43L22_INTERFACE_CTRL2 },
{"PASS_SEL_A", CS43L22_PASS_SEL_A },
{"PASS_SEL_B", CS43L22_PASS_SEL_B },
{"ANLG_ZC_SR_SEL", CS43L22_ANLG_ZC_SR_SEL },
{"PASS_GANG_CTRL", CS43L22_PASS_GANG_CTRL },
{"PLAYBACK_CTRL1", CS43L22_PLAYBACK_CTRL1 },
{"MISCLLNS_CTRL", CS43L22_MISCLLNS_CTRL },
{"PLAYBACK_CTRL2", CS43L22_PLAYBACK_CTRL2 },
{"PASS_VOL_A", CS43L22_PASS_VOL_A },
{"PASS_VOL_B", CS43L22_PASS_VOL_B },
{"PCM_VOL_A", CS43L22_PCM_VOL_A },
{"PCM_VOL_B", CS43L22_PCM_VOL_B },
{"BP_FREQ_ON_T", CS43L22_BP_FREQ_ON_TIME },
{"BP_VOL_OFF_T", CS43L22_BP_VOL_OFF_TIME },
{"BP_TONE_CFG", CS43L22_BP_TONE_CFG },
{"TONE_CTRL", CS43L22_TONE_CTRL },
{"MS_VOL_CTRL_A", CS43L22_MS_VOL_CTRL_A },
{"MS_VOL_CTRL_B", CS43L22_MS_VOL_CTRL_B },
{"HP_VOL_CTRL_A", CS43L22_HP_VOL_CTRL_A },
{"HP_VOL_CTRL_B", CS43L22_HP_VOL_CTRL_B },
{"SPK_VOL_CTRL_A", CS43L22_SPK_VOL_CTRL_A },
{"SPK_VOL_CTRL_B", CS43L22_SPK_VOL_CTRL_B },
{"PCM_CH_SWAP", CS43L22_PCM_CH_SWAP },
{"LIM_CTRL1", CS43L22_LIM_CTRL1 },
{"LIM_CTRL2", CS43L22_LIM_CTRL2 },
{"LIM_ATTACK_RATE", CS43L22_LIM_ATTACK_RATE },
{"STATUS", CS43L22_STATUS },
{"BAT_COMP", CS43L22_BAT_COMP },
{"VP_BAT_LEVEL", CS43L22_VP_BAT_LEVEL },
{"SPK_STATUS", CS43L22_SPK_STATUS },
{"TEMP_MON_CTRL", CS43L22_TEMP_MON_CTRL },
{"THERMAL_FOLDBACK", CS43L22_THERMAL_FOLDBACK},
{"CHRG_PUMP_FREQ", CS43L22_CHRG_PUMP_FREQ },
};
# define CS43L22_NREGISTERS (sizeof(g_cs43l22_debug)/sizeof(struct cs43l22_regdump_s))
# define CS43L22_NREGISTERS (sizeof(g_cs43l22_debug) / sizeof(struct cs43l22_regdump_s))
#endif /* CONFIG_CS43L22_REGDUMP */
/****************************************************************************

14
drivers/audio/cxd56.c
View file

@ -929,7 +929,7 @@ static void cxd56_dma_int_handler(void)
if (dev->mq != NULL)
{
msg.msgId = AUDIO_MSG_DATA_REQUEST;
msg.msg_id = AUDIO_MSG_DATA_REQUEST;
msg.u.data = 0;
(void)nxmq_send(dev->mq, (FAR const char *) &msg,
sizeof(msg), CONFIG_CXD56_MSG_PRIO);
@ -939,7 +939,7 @@ static void cxd56_dma_int_handler(void)
{
/* End of data */
msg.msgId = AUDIO_MSG_STOP;
msg.msg_id = AUDIO_MSG_STOP;
msg.u.data = 0;
(void)nxmq_send(dev->mq, (FAR const char *)&msg,
sizeof(msg), CONFIG_CXD56_MSG_PRIO);
@ -1743,7 +1743,7 @@ static int cxd56_stop(FAR struct audio_lowerhalf_s *lower)
priv->state = CXD56_DEV_STATE_STOPPING;
msg.msgId = AUDIO_MSG_STOP;
msg.msg_id = AUDIO_MSG_STOP;
msg.u.data = 0;
(void)nxmq_send(priv->mq, (FAR const char *)&msg,
sizeof(msg), CONFIG_CXD56_MSG_PRIO);
@ -1804,7 +1804,7 @@ static int cxd56_resume(FAR struct audio_lowerhalf_s *lower)
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int cxd56_release(FAR struct audio_lowerhalf_s *lower,
FAR void *pContext)
FAR void *session)
#else
static int cxd56_release(FAR struct audio_lowerhalf_s *lower)
#endif
@ -1822,7 +1822,7 @@ static int cxd56_release(FAR struct audio_lowerhalf_s *lower)
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int cxd56_reserve(FAR struct audio_lowerhalf_s *lower,
FAR void **ppContext)
FAR void **session)
#else
static int cxd56_reserve(FAR struct audio_lowerhalf_s *lower)
#endif
@ -2030,7 +2030,7 @@ static int cxd56_enqueuebuffer(FAR struct audio_lowerhalf_s *lower,
if (priv->mq != NULL)
{
msg.msgId = AUDIO_MSG_ENQUEUE;
msg.msg_id = AUDIO_MSG_ENQUEUE;
msg.u.data = 0;
(void)nxmq_send(priv->mq, (FAR const char *) &msg,
sizeof(msg), CONFIG_CXD56_MSG_PRIO);
@ -2143,7 +2143,7 @@ static void *cxd56_workerthread(pthread_addr_t pvarg)
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
case AUDIO_MSG_STOP:
cxd56_stop_dma(priv);

32
drivers/audio/tone.c
View file

@ -8,7 +8,8 @@
* modified to become a NuttX driver and to use the Oneshot Timer API.
*
* The PX4 driver is here:
* https://github.com/PX4/Firmware/blob/master/src/drivers/stm32/tone_alarm/tone_alarm.cpp
* https://github.com/PX4/Firmware/blob/master/ \
* src/drivers/stm32/tone_alarm/tone_alarm.cpp
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -80,7 +81,7 @@
#define MODE_LEGATO 2
#define MODE_STACCATO 3
/* Max tune string length*/
/* Max tune string length */
#define MAX_TUNE_LEN (1 * 256)
@ -114,7 +115,10 @@ static char tune_buf[MAX_TUNE_LEN];
/* Semitone offsets from C for the characters 'A'-'G' */
static const uint8_t g_note_tab[] = { 9, 11, 0, 2, 4, 5, 7 };
static const uint8_t g_note_tab[] =
{
9, 11, 0, 2, 4, 5, 7
};
/* Notes in Frequency */
@ -169,7 +173,6 @@ static ssize_t tone_read(FAR struct file *filep, FAR char *buffer,
static ssize_t tone_write(FAR struct file *filep, FAR const char *buffer,
size_t buflen);
/****************************************************************************
* Private Data
****************************************************************************/
@ -406,7 +409,8 @@ static void next_note(FAR struct tone_upperhalf_s *upper)
}
/* Make sure we still have a tune - may be removed by the write / ioctl
* handler */
* handler
*/
if ((g_next == NULL) || (g_tune == NULL))
{
@ -719,7 +723,7 @@ static uint8_t next_number(void)
uint8_t number = 0;
int c;
for (;;)
for (; ; )
{
c = next_char();
@ -782,9 +786,10 @@ static int tone_open(FAR struct file *filep)
goto errout;
}
/* Increment the count of references to the device. If this the first time
* that the driver has been opened for this device, then initialize the
* device. */
/* Increment the count of references to the device.
* If this the first time that the driver has been opened for this device,
* then initialize the device.
*/
tmp = upper->crefs + 1;
if (tmp == 0)
@ -831,8 +836,10 @@ static int tone_close(FAR struct file *filep)
goto errout;
}
/* Decrement the references to the driver. If the reference count will
* decrement to 0, then uninitialize the driver. */
/* Decrement the references to the driver.
* If the reference count will decrement to 0,
* then uninitialize the driver.
*/
if (upper->crefs > 1)
{
@ -946,7 +953,8 @@ int tone_register(FAR const char *path, FAR struct pwm_lowerhalf_s *tone,
/* Allocate the upper-half data structure */
upper =
(FAR struct tone_upperhalf_s *)kmm_zalloc(sizeof(struct tone_upperhalf_s));
(FAR struct tone_upperhalf_s *)kmm_zalloc(
sizeof(struct tone_upperhalf_s));
if (!upper)
{

307
drivers/audio/vs1053.c
View file

@ -149,11 +149,11 @@ struct vs1053_struct_s
****************************************************************************/
static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
FAR struct audio_caps_s *pCaps);
FAR struct audio_caps_s *caps);
static int vs1053_shutdown(FAR struct audio_lowerhalf_s *lower);
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
FAR void *session, FAR const struct audio_caps_s *pCaps);
FAR void *session, FAR const struct audio_caps_s *caps);
static int vs1053_start(FAR struct audio_lowerhalf_s *lower,
FAR void *session);
#ifndef CONFIG_AUDIO_EXCLUDE_STOP
@ -167,12 +167,12 @@ static int vs1053_resume(FAR struct audio_lowerhalf_s *lower,
FAR void *session);
#endif /* CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME */
static int vs1053_reserve(FAR struct audio_lowerhalf_s *lower,
FAR void** ppContext);
FAR void** session);
static int vs1053_release(FAR struct audio_lowerhalf_s *lower,
FAR void *pContext);
FAR void *session);
#else
static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
FAR const struct audio_caps_s *pCaps);
FAR const struct audio_caps_s *caps);
static int vs1053_start(FAR struct audio_lowerhalf_s *lower);
#ifndef CONFIG_AUDIO_EXCLUDE_STOP
static int vs1053_stop(FAR struct audio_lowerhalf_s *lower);
@ -236,8 +236,8 @@ static const uint8_t g_logtable [] =
22, 21, 19, 18, 17, /* 60 - 68 */
15, 14, 13, 12, 11, /* 70 - 78 */
10, 9, 8, 7, 6, /* 80 - 88 */
5, 4, 3, 2, 1, /* 90 - 98 */
0 /* 100 */
5, 4, 3, 2, 1, /* 90 - 98 */
0 /* 100 */
};
#endif /* CONFIG_AUDIO_EXCLUDE_VOLUME */
@ -245,26 +245,29 @@ static const uint8_t g_logtable [] =
* Private Functions
****************************************************************************/
/************************************************************************************
/****************************************************************************
* Name: vs1053_spi_lock
************************************************************************************/
****************************************************************************/
static void vs1053_spi_lock(FAR struct spi_dev_s *dev, unsigned long freq_mhz)
static void vs1053_spi_lock(FAR struct spi_dev_s *dev,
unsigned long freq_mhz)
{
/* On SPI buses where there are multiple devices, it will be necessary to
* lock SPI to have exclusive access to the buses for a sequence of
* transfers. The bus should be locked before the chip is selected.
*
* This is a blocking call and will not return until we have exclusive access to
* the SPI bus. We will retain that exclusive access until the bus is unlocked.
* This is a blocking call and will not return until we have exclusive
* access to the SPI bus.
* We will retain that exclusive access until the bus is unlocked.
*/
SPI_LOCK(dev, true);
/* After locking the SPI bus, the we also need call the setfrequency, setbits, and
* setmode methods to make sure that the SPI is properly configured for the device.
* If the SPI bus is being shared, then it may have been left in an incompatible
* state.
/* After locking the SPI bus, the we also need call the setfrequency,
* setbits, and setmode methods to make sure that the SPI is properly
* configured for the device.
* If the SPI bus is being shared, then it may have been left in an
* incompatible state.
*/
SPI_SETMODE(dev, CONFIG_VS1053_SPIMODE);
@ -273,19 +276,19 @@ static void vs1053_spi_lock(FAR struct spi_dev_s *dev, unsigned long freq_mhz)
SPI_SETFREQUENCY(dev, freq_mhz);
}
/************************************************************************************
/****************************************************************************
* Name: vs1053_spi_unlock
************************************************************************************/
****************************************************************************/
static inline void vs1053_spi_unlock(FAR struct spi_dev_s *dev)
{
SPI_LOCK(dev, false);
}
/************************************************************************************
/****************************************************************************
* Name: vs1053_readreg - Read the specified 16-bit register from the
* VS1053 device. Caller must hold the SPI lock.
************************************************************************************/
****************************************************************************/
static uint16_t vs1053_readreg(FAR struct vs1053_struct_s *dev, uint8_t reg)
{
@ -313,12 +316,14 @@ static uint16_t vs1053_readreg(FAR struct vs1053_struct_s *dev, uint8_t reg)
return ret;
}
/************************************************************************************
/****************************************************************************
* Name: vs1053_writereg - Write the specified 16-bit register to the
* VS1053 device. Caller must hold the SPI lock.
************************************************************************************/
****************************************************************************/
static void vs1053_writereg(FAR struct vs1053_struct_s *dev, uint8_t reg, uint16_t val)
static void vs1053_writereg(FAR struct vs1053_struct_s *dev,
uint8_t reg,
uint16_t val)
{
FAR struct spi_dev_s *spi = dev->spi;
@ -336,7 +341,7 @@ static void vs1053_writereg(FAR struct vs1053_struct_s *dev, uint8_t reg, uint16
/* Now read the 16-bit value */
SPI_SEND(spi, val >> 8);
SPI_SEND(spi, val & 0xFF);
SPI_SEND(spi, val & 0xff);
/* Deselect the CODEC */
@ -354,7 +359,8 @@ static void vs1053_writereg(FAR struct vs1053_struct_s *dev, uint8_t reg, uint16
*
****************************************************************************/
static int vs1053_setfrequency(FAR struct vs1053_struct_s *dev, uint32_t freq)
static int vs1053_setfrequency(FAR struct vs1053_struct_s *dev,
uint32_t freq)
{
double factor;
uint16_t reg;
@ -395,7 +401,7 @@ static int vs1053_setfrequency(FAR struct vs1053_struct_s *dev, uint32_t freq)
* increase the frequency the maximum amount as needed
*/
reg |= (VS1053_SC_ADD_XTALIx20 << VS1053_SC_ADD_SHIFT);
reg |= (VS1053_SC_ADD_XTALI_X20 << VS1053_SC_ADD_SHIFT);
/* If we aren't running with a 12.228Mhz input crystal, then we
* must tell the chip what the frequency is
@ -431,11 +437,12 @@ static int vs1053_setfrequency(FAR struct vs1053_struct_s *dev, uint32_t freq)
return OK;
}
/************************************************************************************
* Name: vs1053_logapprox - Approximate the register value in .5 dB increments
* level based on the percentage using a log table since
* math libraries aren't available.
************************************************************************************/
/****************************************************************************
* Name: vs1053_logapprox -
* Approximate the register value in .5 dB increments
* level based on the percentage using a log table since
* math libraries aren't available.
****************************************************************************/
#ifndef CONFIG_AUDIO_EXCLUDE_VOLUME
uint8_t vs1053_logapprox(int percent)
@ -447,14 +454,15 @@ uint8_t vs1053_logapprox(int percent)
return 0;
}
return (g_logtable[percent >> 1] + g_logtable[(percent+1) >> 1]) >> 1;
return (g_logtable[percent >> 1] + g_logtable[(percent + 1) >> 1]) >> 1;
}
#endif /* CONFIG_AUDIO_EXCLUDE_VOLUME */
/************************************************************************************
* Name: vs1053_setvolume - Set the right and left volume values in the VS1053
* device based on the current volume and balance settings.
************************************************************************************/
/****************************************************************************
* Name: vs1053_setvolume -
* Set the right and left volume values in the VS1053
* device based on the current volume and balance settings.
****************************************************************************/
#ifndef CONFIG_AUDIO_EXCLUDE_VOLUME
static void vs1053_setvolume(FAR struct vs1053_struct_s *dev)
@ -462,7 +470,8 @@ static void vs1053_setvolume(FAR struct vs1053_struct_s *dev)
FAR struct spi_dev_s *spi = dev->spi;
uint32_t leftlevel;
uint32_t rightlevel;
uint8_t leftreg, rightreg;
uint8_t leftreg;
uint8_t rightreg;
/* Constrain balance */
#ifndef CONFIG_AUDIO_EXCLUDE_BALANCE
@ -505,6 +514,7 @@ static void vs1053_setvolume(FAR struct vs1053_struct_s *dev)
#endif
/* Calculate the left and right register values */
/* The register sets the volume in dB which is a logrithmic scale,
* so we must use log() to calculate the register value.
*/
@ -520,20 +530,22 @@ static void vs1053_setvolume(FAR struct vs1053_struct_s *dev)
}
#endif /* CONFIG_AUDIO_EXCLUDE_VOLUME */
/************************************************************************************
/****************************************************************************
* Name: vs1053_setbass - Set the bass and treble level as specified in the
* context's bass and treble variables..
*
* The level and range are in whole percentage levels (0-100).
*
************************************************************************************/
****************************************************************************/
#ifndef CONFIG_AUDIO_EXCLUDE_TONE
static void vs1053_setbass(FAR struct vs1053_struct_s *dev)
{
FAR struct spi_dev_s *spi = dev->spi;
int bass_range, bass_boost;
int treble_range, treble_boost;
int bass_range;
int bass_boost;
int treble_range;
int treble_boost;
/* Calculate range and boost based on level */
@ -545,8 +557,9 @@ static void vs1053_setbass(FAR struct vs1053_struct_s *dev)
/* Lock the SPI bus to get exclsive access to the chip. */
vs1053_spi_lock(spi, dev->spi_freq);
vs1053_writereg(dev, VS1053_SCI_BASS, (treble_boost << 12) | (treble_range << 8) |
(bass_boost << 4) | bass_range);
vs1053_writereg(dev, VS1053_SCI_BASS,
(treble_boost << 12) | (treble_range << 8) |
(bass_boost << 4) | bass_range);
vs1053_spi_unlock(spi);
}
#endif /* CONFIG_AUDIO_EXCLUDE_TONE */
@ -559,20 +572,20 @@ static void vs1053_setbass(FAR struct vs1053_struct_s *dev)
****************************************************************************/
static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
FAR struct audio_caps_s *pCaps)
FAR struct audio_caps_s *caps)
{
audinfo("Entry\n");
/* Validate the structure */
DEBUGASSERT(pCaps->ac_len >= sizeof(struct audio_caps_s));
DEBUGASSERT(caps->ac_len >= sizeof(struct audio_caps_s));
/* Fill in the caller's structure based on requested info */
pCaps->ac_format.hw = 0;
pCaps->ac_controls.w = 0;
caps->ac_format.hw = 0;
caps->ac_controls.w = 0;
switch (pCaps->ac_type)
switch (caps->ac_type)
{
/* Caller is querying for the types of units we support */
@ -582,14 +595,15 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
* must then call us back for specific info for each capability.
*/
pCaps->ac_channels = 2; /* Stereo output */
caps->ac_channels = 2; /* Stereo output */
switch (pCaps->ac_subtype)
switch (caps->ac_subtype)
{
case AUDIO_TYPE_QUERY:
/* The input formats we can decode / accept */
pCaps->ac_format.hw = 0
caps->ac_format.hw = 0
#ifdef CONFIG_AUDIO_FORMAT_AC3
| (1 << (AUDIO_FMT_AC3 - 1))
#endif
@ -612,8 +626,9 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
/* The types of audio units we implement */
pCaps->ac_controls.b[0] = AUDIO_TYPE_OUTPUT | AUDIO_TYPE_FEATURE |
AUDIO_TYPE_PROCESSING;
caps->ac_controls.b[0] = AUDIO_TYPE_OUTPUT |
AUDIO_TYPE_FEATURE |
AUDIO_TYPE_PROCESSING;
break;
@ -621,15 +636,16 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
#ifdef CONFIG_AUDIO_FORMAT_MIDI
case AUDIO_FMT_MIDI:
/* We only support Format 0 */
pCaps->ac_controls.b[0] = AUDIO_SUBFMT_MIDI_0;
pCaps->ac_controls.b[1] = AUDIO_SUBFMT_END;
caps->ac_controls.b[0] = AUDIO_SUBFMT_MIDI_0;
caps->ac_controls.b[1] = AUDIO_SUBFMT_END;
break;
#endif
default:
pCaps->ac_controls.b[0] = AUDIO_SUBFMT_END;
caps->ac_controls.b[0] = AUDIO_SUBFMT_END;
break;
}
@ -639,28 +655,34 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
case AUDIO_TYPE_OUTPUT:
pCaps->ac_channels = 2;
caps->ac_channels = 2;
switch (pCaps->ac_subtype)
switch (caps->ac_subtype)
{
case AUDIO_TYPE_QUERY:
/* Report the Sample rates we support */
pCaps->ac_controls.b[0] = AUDIO_SAMP_RATE_8K | AUDIO_SAMP_RATE_11K |
AUDIO_SAMP_RATE_16K | AUDIO_SAMP_RATE_22K |
AUDIO_SAMP_RATE_32K | AUDIO_SAMP_RATE_44K |
AUDIO_SAMP_RATE_48K;
caps->ac_controls.b[0] = AUDIO_SAMP_RATE_8K |
AUDIO_SAMP_RATE_11K |
AUDIO_SAMP_RATE_16K |
AUDIO_SAMP_RATE_22K |
AUDIO_SAMP_RATE_32K |
AUDIO_SAMP_RATE_44K |
AUDIO_SAMP_RATE_48K;
break;
case AUDIO_FMT_MP3:
case AUDIO_FMT_WMA:
case AUDIO_FMT_PCM:
/* Report the Bit rates we support. The bit rate support is actually a
* complex function of the format and selected sample rate, and the datasheet
* has multiple tables to indicate the supported bit rate vs sample rate vs
* format. The selected sample rate should be provided in the ac_format
* field of the query, and only a single sample rate should be given.
/* Report the Bit rates we support.
* The bit rate support is actually a complex function of the
* format and selected sample rate, and the datasheet has
* multiple tables to indicate the supported bit rate vs sample
* rate vsformat.
* The selected sample rate should be provided in the ac_format
* field of the query, and only a single sample rate should be
* given.
*/
/* TODO: Create a table or set of tables to report this! */
@ -677,18 +699,25 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
case AUDIO_TYPE_FEATURE:
/* If the sub-type is UNDEF, then report the Feature Units we support */
/* If the sub-type is UNDEF,
* then report the Feature Units we support
*/
if (pCaps->ac_subtype == AUDIO_FU_UNDEF)
if (caps->ac_subtype == AUDIO_FU_UNDEF)
{
/* Fill in the ac_controls section with the Feature Units we have */
/* Fill in the ac_controls section with the
* Feature Units we have
*/
pCaps->ac_controls.b[0] = AUDIO_FU_VOLUME | AUDIO_FU_BASS | AUDIO_FU_TREBLE;
pCaps->ac_controls.b[1] = AUDIO_FU_BALANCE >> 8;
caps->ac_controls.b[0] = AUDIO_FU_VOLUME |
AUDIO_FU_BASS |
AUDIO_FU_TREBLE;
caps->ac_controls.b[1] = AUDIO_FU_BALANCE >> 8;
}
else
{
/* TODO: Do we need to provide specific info for the Feature Units,
/* TODO:
* Do we need to provide specific info for the Feature Units,
* such as volume setting ranges, etc.?
*/
}
@ -699,20 +728,21 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
case AUDIO_TYPE_PROCESSING:
switch (pCaps->ac_subtype)
switch (caps->ac_subtype)
{
case AUDIO_PU_UNDEF:
/* Provide the type of Processing Units we support */
pCaps->ac_controls.b[0] = AUDIO_PU_STEREO_EXTENDER;
caps->ac_controls.b[0] = AUDIO_PU_STEREO_EXTENDER;
break;
case AUDIO_PU_STEREO_EXTENDER:
/* Proivde capabilities of our Stereo Extender */
pCaps->ac_controls.b[0] = AUDIO_STEXT_ENABLE | AUDIO_STEXT_WIDTH;
caps->ac_controls.b[0] = AUDIO_STEXT_ENABLE |
AUDIO_STEXT_WIDTH;
break;
default:
@ -730,8 +760,8 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
/* Zero out the fields to indicate no support */
pCaps->ac_subtype = 0;
pCaps->ac_channels = 0;
caps->ac_subtype = 0;
caps->ac_channels = 0;
break;
}
@ -740,7 +770,7 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
* proper Audio device type.
*/
return pCaps->ac_len;
return caps->ac_len;
}
/****************************************************************************
@ -753,10 +783,10 @@ static int vs1053_getcaps(FAR struct audio_lowerhalf_s *lower, int type,
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
FAR void *session, FAR const struct audio_caps_s *pCaps)
FAR void *session, FAR const struct audio_caps_s *caps)
#else
static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
FAR const struct audio_caps_s *pCaps)
FAR const struct audio_caps_s *caps)
#endif
{
int ret = OK;
@ -768,19 +798,20 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
/* Process the configure operation */
switch (pCaps->ac_type)
switch (caps->ac_type)
{
case AUDIO_TYPE_FEATURE:
/* Process based on Feature Unit */
switch (pCaps->ac_format.hw)
switch (caps->ac_format.hw)
{
#ifndef CONFIG_AUDIO_EXCLUDE_VOLUME
case AUDIO_FU_VOLUME:
/* Set the volume */
dev->volume = pCaps->ac_controls.hw[0];
dev->volume = caps->ac_controls.hw[0];
vs1053_setvolume(dev);
break;
@ -788,9 +819,10 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
#if !defined(CONFIG_AUDIO_EXCLUDE_TONE) && !defined(CONFIG_AUDIO_EXCLUDE_VOLUME)
case AUDIO_FU_BALANCE:
/* Set the volume */
dev->balance = pCaps->ac_controls.hw[0];
dev->balance = caps->ac_controls.hw[0];
vs1053_setvolume(dev);
break;
@ -798,11 +830,12 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
#ifndef CONFIG_AUDIO_EXCLUDE_TONE
case AUDIO_FU_BASS:
/* Set the bass. The percentage level (0-100) is in the
* ac_controls[0] parameter.
*/
dev->bass = pCaps->ac_controls.b[0];
dev->bass = caps->ac_controls.b[0];
if (dev->bass > 100)
dev->bass = 100;
vs1053_setbass(dev);
@ -814,7 +847,7 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
* ac_controls.b[0] parameter.
*/
dev->treble = pCaps->ac_controls.b[0];
dev->treble = caps->ac_controls.b[0];
if (dev->treble > 100)
dev->treble = 100;
vs1053_setbass(dev);
@ -823,6 +856,7 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
#endif /* CONFIG_AUDIO_EXCLUDE_TONE */
default:
/* Others we don't support */
break;
@ -837,7 +871,7 @@ static int vs1053_configure(FAR struct audio_lowerhalf_s *lower,
/* We only support STEREO_EXTENDER */
if (pCaps->ac_format.hw == AUDIO_PU_STEREO_EXTENDER)
if (caps->ac_format.hw == AUDIO_PU_STEREO_EXTENDER)
{
}
@ -922,7 +956,7 @@ static int vs1053_shutdown(FAR struct audio_lowerhalf_s *lower)
audinfo("Entry\n");
vs1053_spi_lock(spi, dev->spi_freq); /* Lock the device */
vs1053_setfrequency(dev, CONFIG_VS1053_XTALI); /* Reduce speed to minimum */
vs1053_writereg(dev, VS1053_SCI_VOL, 0xFEFE); /* Power down the DAC outputs */
vs1053_writereg(dev, VS1053_SCI_VOL, 0xfefe); /* Power down the DAC outputs */
vs1053_spi_unlock(spi); /* Unlock the device */
return OK;
}
@ -940,7 +974,7 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
{
int bytecount;
int ret;
uint8_t *pSamp = NULL;
uint8_t *samp = NULL;
uint16_t reg;
struct ap_buffer_s *apb;
FAR struct spi_dev_s *spi = dev->spi;
@ -957,18 +991,22 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
*/
vs1053_spi_lock(spi, VS1053_DATA_FREQ); /* Lock the SPI bus */
SPI_SELECT(spi, SPIDEV_AUDIO_DATA(0), true); /* Select the VS1053 data bus */
/* Local stack copy of our active buffer */
apb = dev->apb;
//audinfo("Entry apb=%p, Bytes left=%d\n", apb, apb->nbytes - apb->curbyte);
/* audinfo("Entry apb=%p, Bytes left=%d\n",
* apb, apb->nbytes - apb->curbyte);
*/
/* Setup pointer to the next sample in the buffer */
if (apb)
{
pSamp = &apb->samp[apb->curbyte];
samp = &apb->samp[apb->curbyte];
}
else if (!dev->endmode)
{
@ -1004,7 +1042,7 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
* 32 bytes at a time.
*/
if (dev->endfillbytes == 32*65)
if (dev->endfillbytes == 32 * 65)
{
/* After at least 2052 bytes, we send an SM_CANCEL */
@ -1014,7 +1052,7 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
vs1053_writereg(dev, VS1053_SCI_MODE, reg | VS1053_SM_CANCEL);
dev->hw_lower->enable(dev->hw_lower); /* Enable the DREQ interrupt */
}
else if (dev->endfillbytes >= 32*130)
else if (dev->endfillbytes >= 32 * 130)
{
/* Do a hard reset and terminate */
@ -1023,14 +1061,15 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
dev->endmode = false;
break;
}
else if (dev->endfillbytes > 32*65)
else if (dev->endfillbytes > 32 * 65)
{
/* After each 32 byte of endfillchar, check the status
* register to see if SM_CANCEL has been cleared. If
* it has been cleared, then we're done.
*/
if (!(vs1053_readreg(dev, VS1053_SCI_STATUS) & VS1053_SM_CANCEL))
if (!(vs1053_readreg(dev, VS1053_SCI_STATUS) &
VS1053_SM_CANCEL))
{
SPI_SETFREQUENCY(dev->spi, dev->spi_freq);
dev->hw_lower->disable(dev->hw_lower); /* Disable the DREQ interrupt */
@ -1039,13 +1078,18 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
vs1053_readreg(dev, VS1053_SCI_HDAT1),
vs1053_readreg(dev, VS1053_SCI_HDAT0));
vs1053_writereg(dev, VS1053_SCI_WRAMADDR, VS1053_END_FILL_BYTE);
dev->endfillchar = vs1053_readreg(dev, VS1053_SCI_WRAM) >> 8;
vs1053_writereg(dev,
VS1053_SCI_WRAMADDR,
VS1053_END_FILL_BYTE);
dev->endfillchar = vs1053_readreg(dev,
VS1053_SCI_WRAM) >> 8;
audinfo("EndFillChar: 0x%0X\n", dev->endfillchar);
reg = vs1053_readreg(dev, VS1053_SCI_MODE);
vs1053_writereg(dev, VS1053_SCI_MODE, reg | VS1053_SM_RESET);
vs1053_writereg(dev,
VS1053_SCI_MODE,
reg | VS1053_SM_RESET);
dev->running = false;
dev->endmode = false;
@ -1066,18 +1110,20 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
{
bytecount = 32;
}
#if 1
SPI_SNDBLOCK(spi, pSamp, bytecount);
pSamp += bytecount;
SPI_SNDBLOCK(spi, samp, bytecount);
samp += bytecount;
#else
bytecount = bytecount;
while (bytecount--)
{
/* Send next byte from the buffer */
SPI_SEND(spi, *pSamp);
pSamp++;
SPI_SEND(spi, *samp);
samp++;
}
#endif
apb->curbyte += bytecount;
@ -1121,8 +1167,11 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
dev->hw_lower->disable(dev->hw_lower); /* Disable the DREQ interrupt */
SPI_SETFREQUENCY(dev->spi, dev->spi_freq);
vs1053_writereg(dev, VS1053_SCI_WRAMADDR, VS1053_END_FILL_BYTE);
dev->endfillchar = vs1053_readreg(dev, VS1053_SCI_WRAM) >> 8;
vs1053_writereg(dev,
VS1053_SCI_WRAMADDR,
VS1053_END_FILL_BYTE);
dev->endfillchar = vs1053_readreg(dev,
VS1053_SCI_WRAM) >> 8;
SPI_SETFREQUENCY(dev->spi, VS1053_DATA_FREQ);
dev->hw_lower->enable(dev->hw_lower); /* Enable the DREQ interrupt */
@ -1133,12 +1182,12 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
#ifndef CONFIG_AUDIO_EXCLUDE_STOP
if (dev->cancelmode)
{
/* If we are in cancel mode, then we don't dequeue the buffer
* or need to send another SM_CANCEL, so jump into the middle
* of the stop sequence.
/* If we are in cancel mode, then we don't dequeue the
* buffer or need to send another SM_CANCEL, so jump
* into the middle of the stop sequence.
*/
dev->endfillbytes = 32*65+1;
dev->endfillbytes = 32 * 65 + 1;
continue;
}
else
@ -1180,14 +1229,17 @@ static void vs1053_feeddata(FAR struct vs1053_struct_s *dev)
apb = (struct ap_buffer_s *) dq_remfirst(&dev->apbq);
dev->apb = apb;
//audinfo("Next Buffer = %p, bytes = %d\n", apb, apb ? apb->nbytes : 0);
/* audinfo("Next Buffer = %p, bytes = %d\n",
* apb, apb ? apb->nbytes : 0);
*/
if (apb == NULL)
{
nxsem_post(&dev->apbq_sem);
break;
}
pSamp = &apb->samp[apb->curbyte];
samp = &apb->samp[apb->curbyte];
apb_reference(apb); /* Add our buffer reference */
nxsem_post(&dev->apbq_sem);
}
@ -1222,13 +1274,13 @@ static int vs1053_dreq_isr(int irq, FAR void *context, FAR void *arg)
if (dev->running)
{
msg.msgId = AUDIO_MSG_DATA_REQUEST;
msg.msg_id = AUDIO_MSG_DATA_REQUEST;
nxmq_send(dev->mq, (FAR const char *)&msg, sizeof(msg),
CONFIG_VS1053_MSG_PRIO);
}
else
{
msg.msgId = AUDIO_MSG_DATA_REQUEST;
msg.msg_id = AUDIO_MSG_DATA_REQUEST;
}
return 0;
@ -1302,7 +1354,7 @@ static void *vs1053_workerthread(pthread_addr_t pvarg)
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
/* The ISR has requested more data */
@ -1356,7 +1408,8 @@ static void *vs1053_workerthread(pthread_addr_t pvarg)
{
/* Get the next buffer from the queue */
while ((apb = (FAR struct ap_buffer_s *) dq_remfirst(&dev->apbq)) != NULL)
while ((apb = (FAR struct ap_buffer_s *) dq_remfirst(&dev->apbq))
!= NULL)
;
}
@ -1397,7 +1450,8 @@ static void *vs1053_workerthread(pthread_addr_t pvarg)
****************************************************************************/
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_start(FAR struct audio_lowerhalf_s *lower, FAR void *session)
static int vs1053_start(FAR struct audio_lowerhalf_s *lower,
FAR void *session)
#else
static int vs1053_start(FAR struct audio_lowerhalf_s *lower)
#endif
@ -1501,7 +1555,8 @@ static int vs1053_start(FAR struct audio_lowerhalf_s *lower)
#ifndef CONFIG_AUDIO_EXCLUDE_STOP
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_stop(FAR struct audio_lowerhalf_s *lower, FAR void *session)
static int vs1053_stop(FAR struct audio_lowerhalf_s *lower,
FAR void *session)
#else
static int vs1053_stop(FAR struct audio_lowerhalf_s *lower)
#endif
@ -1512,7 +1567,7 @@ static int vs1053_stop(FAR struct audio_lowerhalf_s *lower)
/* Send a message to stop all audio streaming */
term_msg.msgId = AUDIO_MSG_STOP;
term_msg.msg_id = AUDIO_MSG_STOP;
term_msg.u.data = 0;
nxmq_send(dev->mq, (FAR const char *)&term_msg, sizeof(term_msg),
CONFIG_VS1053_MSG_PRIO);
@ -1545,7 +1600,8 @@ static int vs1053_stop(FAR struct audio_lowerhalf_s *lower)
#ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_pause(FAR struct audio_lowerhalf_s *lower, FAR void *session)
static int vs1053_pause(FAR struct audio_lowerhalf_s *lower,
FAR void *session)
#else
static int vs1053_pause(FAR struct audio_lowerhalf_s *lower)
#endif
@ -1574,7 +1630,8 @@ static int vs1053_pause(FAR struct audio_lowerhalf_s *lower)
#ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME
#ifdef CONFIG_AUDIO_MULTI_SESSION
static int vs1053_resume(FAR struct audio_lowerhalf_s *lower, FAR void *session)
static int vs1053_resume(FAR struct audio_lowerhalf_s *lower,
FAR void *session)
#else
static int vs1053_resume(FAR struct audio_lowerhalf_s *lower)
#endif
@ -1626,7 +1683,7 @@ static int vs1053_enqueuebuffer(FAR struct audio_lowerhalf_s *lower,
if (dev->mq != NULL)
{
term_msg.msgId = AUDIO_MSG_ENQUEUE;
term_msg.msg_id = AUDIO_MSG_ENQUEUE;
term_msg.u.data = 0;
nxmq_send(dev->mq, (FAR const char *)&term_msg,
sizeof(term_msg), CONFIG_VS1053_MSG_PRIO);
@ -1660,7 +1717,7 @@ static int vs1053_ioctl(FAR struct audio_lowerhalf_s *lower, int cmd,
unsigned long arg)
{
#ifdef CONFIG_AUDIO_DRIVER_SPECIFIC_BUFFERS
FAR struct ap_buffer_info_s *pBufInfo;
FAR struct ap_buffer_info_s *bufinfo;
#endif
/* Deal with ioctls passed from the upper-half driver */
@ -1680,9 +1737,9 @@ static int vs1053_ioctl(FAR struct audio_lowerhalf_s *lower, int cmd,
#ifdef CONFIG_AUDIO_DRIVER_SPECIFIC_BUFFERS
case AUDIOIOC_GETBUFFERINFO:
pBufInfo = (FAR struct ap_buffer_info_s *) arg;
pBufInfo->buffer_size = CONFIG_VS1053_BUFFER_SIZE;
pBufInfo->nbuffers = CONFIG_VS1053_NUM_BUFFERS;
bufinfo = (FAR struct ap_buffer_info_s *) arg;
bufinfo->buffer_size = CONFIG_VS1053_BUFFER_SIZE;
bufinfo->nbuffers = CONFIG_VS1053_NUM_BUFFERS;
break;
#endif

34
drivers/audio/vs1053.h
View file

@ -93,7 +93,7 @@
#define VS1053_SM_LINE1 0x4000
#define VS1053_SM_CLK_RANGE 0x8000
/* STATUS register bit definitions ****************************************/
/* STATUS register bit definitions ******************************************/
#define VS1053_SS_DO_NOT_JUMP 0x8000
#define VS1053_SS_SWING 0x7000
@ -115,14 +115,14 @@
#define VS1053_VER_VS1063 6
#define VS1053_VER_VS1103 7
/* BASS register bit definitions ******************************************/
/* BASS register bit definitions ********************************************/
#define VS1053_ST_AMPLITUDE 0xF000
#define VS1053_ST_FREQLIMIT 0x0F00
#define VS1053_SB_AMPLITUDE 0x00F0
#define VS1053_SB_FREQLIMIT 0x000F
/* CLOCKF register bit definitions ****************************************/
/* CLOCKF register bit definitions ******************************************/
#define VS1053_SC_MULT 0xE000
#define VS1053_SC_MULT_SHIFT 13
@ -130,21 +130,21 @@
#define VS1053_SC_ADD_SHIFT 11
#define VS1053_SC_FREQ 0x07FF
#define VS1053_SC_MULT_XTALIx10 0
#define VS1053_SC_MULT_XTALIx20 1
#define VS1053_SC_MULT_XTALIx25 2
#define VS1053_SC_MULT_XTALIx30 3
#define VS1053_SC_MULT_XTALIx35 4
#define VS1053_SC_MULT_XTALIx40 5
#define VS1053_SC_MULT_XTALIx45 6
#define VS1053_SC_MULT_XTALIx50 7
#define VS1053_SC_MULT_XTALI_X10 0
#define VS1053_SC_MULT_XTALI_X20 1
#define VS1053_SC_MULT_XTALI_X25 2
#define VS1053_SC_MULT_XTALI_X30 3
#define VS1053_SC_MULT_XTALI_X35 4
#define VS1053_SC_MULT_XTALI_X40 5
#define VS1053_SC_MULT_XTALI_X45 6
#define VS1053_SC_MULT_XTALI_X50 7
#define VS1053_SC_ADD_NONE 0
#define VS1053_SC_ADD_XTALIx10 1
#define VS1053_SC_ADD_XTALIx15 2
#define VS1053_SC_ADD_XTALIx20 3
#define VS1053_SC_ADD_NONE 0
#define VS1053_SC_ADD_XTALI_X10 1
#define VS1053_SC_ADD_XTALI_X15 2
#define VS1053_SC_ADD_XTALI_X20 3
/* WRAM Addresses **********************************************************/
/* WRAM Addresses ***********************************************************/
#define VS1053_XRAM_BASE 0x1800 /* X data RAM */
#define VS1053_XRAM_SIZE 256
@ -158,7 +158,7 @@
#define VS1053_IO_BASE 0xC000
#define VS1053_IO_SIZE 0x4000
/* HDAT1 register values *************************************************/
/* HDAT1 register values ****************************************************/
#define VS1053_HDAT1_WAV 0x7665 /* "ve" (as in Wave) */
#define VS1053_HDAT1_ADTS 0x4154 /* "AT" */

10
drivers/audio/wm8776.c
View file

@ -587,7 +587,7 @@ static void wm8776_senddone(FAR struct i2s_dev_s *i2s,
* buffers in the done queue that need to be cleaned up.
*/
msg.msgId = AUDIO_MSG_COMPLETE;
msg.msg_id = AUDIO_MSG_COMPLETE;
ret = mq_send(priv->mq, (FAR const char *)&msg, sizeof(msg),
CONFIG_WM8776_MSG_PRIO);
if (ret < 0)
@ -832,7 +832,7 @@ static int wm8776_stop(FAR struct audio_lowerhalf_s *dev)
/* Send a message to stop all audio streaming */
term_msg.msgId = AUDIO_MSG_STOP;
term_msg.msg_id = AUDIO_MSG_STOP;
term_msg.u.data = 0;
mq_send(priv->mq, (FAR const char *)&term_msg, sizeof(term_msg),
CONFIG_WM8776_MSG_PRIO);
@ -946,7 +946,7 @@ static int wm8776_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
ret = OK;
if (priv->mq != NULL)
{
term_msg.msgId = AUDIO_MSG_ENQUEUE;
term_msg.msg_id = AUDIO_MSG_ENQUEUE;
term_msg.u.data = 0;
ret = mq_send(priv->mq, (FAR const char *)&term_msg, sizeof(term_msg),
@ -1250,7 +1250,7 @@ repeat:
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
/* The ISR has requested more data. We will catch this case at
* the top of the loop.
@ -1288,7 +1288,7 @@ repeat:
break;
default:
auderr("ERROR: Ignoring message ID %d\n", msg.msgId);
auderr("ERROR: Ignoring message ID %d\n", msg.msg_id);
break;
}

9
drivers/audio/wm8776.h
View file

@ -84,10 +84,11 @@ struct wm8776_dev_s
/* We are an audio lower half driver (We are also the upper "half" of
* the WM8776 driver with respect to the board lower half driver).
*
* Terminology: Our "lower" half audio instances will be called dev for the
* publicly visible version and "priv" for the version that only this driver
* knows. From the point of view of this driver, it is the board lower
* "half" that is referred to as "lower".
* Terminology:
* Our "lower" half audio instances will be called dev for the publicly
* visible version and "priv" for the version that only this driver knows
* From the point of view of this driver, it is the board lower "half"
* that is referred to as "lower".
*/
struct audio_lowerhalf_s dev; /* WM8776 audio lower half (this device) */

18
drivers/audio/wm8904.c
View file

@ -1099,11 +1099,15 @@ static int wm8904_getcaps(FAR struct audio_lowerhalf_s *dev, int type,
case AUDIO_TYPE_FEATURE:
/* If the sub-type is UNDEF, then report the Feature Units we support */
/* If the sub-type is UNDEF,
* then report the Feature Units we support
*/
if (caps->ac_subtype == AUDIO_FU_UNDEF)
{
/* Fill in the ac_controls section with the Feature Units we have */
/* Fill in the ac_controls section with
* the Feature Units we have
*/
caps->ac_controls.b[0] = AUDIO_FU_VOLUME | AUDIO_FU_BASS |
AUDIO_FU_TREBLE;
@ -1400,7 +1404,7 @@ static void wm8904_senddone(FAR struct i2s_dev_s *i2s,
* buffers in the done queue that need to be cleaned up.
*/
msg.msgId = AUDIO_MSG_COMPLETE;
msg.msg_id = AUDIO_MSG_COMPLETE;
ret = nxmq_send(priv->mq, (FAR const char *)&msg, sizeof(msg),
CONFIG_WM8904_MSG_PRIO);
if (ret < 0)
@ -1667,7 +1671,7 @@ static int wm8904_stop(FAR struct audio_lowerhalf_s *dev)
/* Send a message to stop all audio streaming */
term_msg.msgId = AUDIO_MSG_STOP;
term_msg.msg_id = AUDIO_MSG_STOP;
term_msg.u.data = 0;
nxmq_send(priv->mq, (FAR const char *)&term_msg, sizeof(term_msg),
CONFIG_WM8904_MSG_PRIO);
@ -1790,7 +1794,7 @@ static int wm8904_enqueuebuffer(FAR struct audio_lowerhalf_s *dev,
ret = OK;
if (priv->mq != NULL)
{
term_msg.msgId = AUDIO_MSG_ENQUEUE;
term_msg.msg_id = AUDIO_MSG_ENQUEUE;
term_msg.u.data = 0;
ret = nxmq_send(priv->mq, (FAR const char *)&term_msg,
@ -2124,7 +2128,7 @@ static void *wm8904_workerthread(pthread_addr_t pvarg)
/* Process the message */
switch (msg.msgId)
switch (msg.msg_id)
{
/* The ISR has requested more data. We will catch this case at
* the top of the loop.
@ -2162,7 +2166,7 @@ static void *wm8904_workerthread(pthread_addr_t pvarg)
break;
default:
auderr("ERROR: Ignoring message ID %d\n", msg.msgId);
auderr("ERROR: Ignoring message ID %d\n", msg.msg_id);
break;
}
}

90
drivers/audio/wm8904.h
View file

@ -58,6 +58,7 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* So far, I have not been able to get FLL lock interrupts. Worse, I have
* been able to get the FLL to claim that it is locked at all even when
* polling. What am I doing wrong?
@ -175,6 +176,7 @@
#define WM8904_DUMMY 0xff /* Dummy register address */
/* Register Default Values **************************************************/
/* Registers have some undocumented bits set on power up. These probably
* should be retained on writes (?).
*/
@ -287,22 +289,22 @@
/* 0x04 Bias Control */
#define WM8904_ISEL_SHIFT (2) /* Bits 2-3: Master Bias Control */
#define WM8904_ISEL_SHIFT (2) /* Bits 2-3: Master Bias Control */
#define WM8904_ISEL_MASK (3 << WM8904_ISEL_SHIFT)
# define WM8904_ISEL_LOW (0 << WM8904_ISEL_SHIFT) /* Low power bias */
# define WM8904_ISEL_HIGH (2 << WM8904_ISEL_SHIFT) /* High performance bias */
#define WM8904_BIAS_ENA (1 << 0) /* Bit 0: Normal bias current generator */
#define WM8904_BIAS_ENA (1 << 0) /* Bit 0: Normal bias current generator */
/* 0x05 VMID Control */
#define WM8904_VMID_BUF_ENA (1 << 6) /* Bit 6: Enable VMID buffer to unused I/O */
#define WM8904_VMID_RES_SHIFT (1) /* Bits 1-2: VMID divider enable and select */
#define WM8904_VMID_BUF_ENA (1 << 6) /* Bit 6: Enable VMID buffer to unused I/O */
#define WM8904_VMID_RES_SHIFT (1) /* Bits 1-2: VMID divider enable and select */
#define WM8904_VMID_RES_MASK (3 << WM8904_VMID_RES_SHIFT)
# define WM8904_VMID_RES_OFF (0 << WM8904_VMID_RES_SHIFT) /* VMID disabled */
# define WM8904_VMID_RES_NORMAL (1 << WM8904_VMID_RES_SHIFT) /* 2 x 50k divider */
# define WM8904_VMID_RES_STANDBY (2 << WM8904_VMID_RES_SHIFT) /* 2 x 250k divider */
# define WM8904_VMID_RES_FASTSTART (3 << WM8904_VMID_RES_SHIFT) /* 2 x 5k divider */
#define WM8904_VMID_ENA (1 << 0) /* Bit 0: VMID buffer enable */
#define WM8904_VMID_ENA (1 << 0) /* Bit 0: VMID buffer enable */
/* 0x06 Mic Bias Control 0 */
@ -433,29 +435,32 @@
/* 0x1a Audio Interface 2 */
#define WM8904_OPCLK_DIV_SHIFT (8) /* Bits 8-11: GPIO Output Clock Divider */
#define WM8904_OPCLK_DIV_MASK (15 << WM8904_OPCLK_DIV_SHIFT)
# define WM8904_OPCLK_DIV1 (0 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK */
# define WM8904_OPCLK_DIV2 (1 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 2 */
# define WM8904_OPCLK_DIV3 (2 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 3 */
# define WM8904_OPCLK_DIV4 (3 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 4 */
# define WM8904_OPCLK_DIV5p5 (4 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 5.5 */
# define WM8904_OPCLK_DIV6 (5 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 6 */
# define WM8904_OPCLK_DIV8 (6 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 8 */
# define WM8904_OPCLK_DIV12 (7 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 12 */
# define WM8904_OPCLK_DIV16 (8 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 16 */
# define WM8904_OPCLK_DIV1 (0 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK */
# define WM8904_OPCLK_DIV2 (1 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 2 */
# define WM8904_OPCLK_DIV3 (2 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 3 */
# define WM8904_OPCLK_DIV4 (3 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 4 */
# define WM8904_OPCLK_DIV5p5 (4 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 5.5 */
# define WM8904_OPCLK_DIV6 (5 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 6 */
# define WM8904_OPCLK_DIV8 (6 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 8 */
# define WM8904_OPCLK_DIV12 (7 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 12 */
# define WM8904_OPCLK_DIV16 (8 << WM8904_OPCLK_DIV_SHIFT) /* SYSCLK / 16 */
#define WM8904_BCLK_DIV_SHIFT (0) /* Bits 0-4: BCLK Frequency (Master Mode) */
#define WM8904_BCLK_DIV_MASK (31 << WM8904_BCLK_DIV_SHIFT)
# define WM8904_BCLK_DIV(n) ((uint16_t)(n) << WM8904_BCLK_DIV_SHIFT)
# define WM8904_BCLK_DIV1 (0 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK */
# define WM8904_BCLK_DIV1p5 (1 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 1.5 */
# define WM8904_BCLK_DIV2 (2 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 2 */
# define WM8904_BCLK_DIV3 (3 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 3 */
# define WM8904_BCLK_DIV4 (4 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 4 */
# define WM8904_BCLK_DIV5 (5 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 5 */
# define WM8904_BCLK_DIV5p5 (6 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 5.5 */
# define WM8904_BCLK_DIV6 (7 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 6 */
# define WM8904_BCLK_DIV8 (8 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 8 */
# define WM8904_BCLK_DIV10 (9 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 10 */
# define WM8904_BCLK_DIV1 (0 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK */
# define WM8904_BCLK_DIV1p5 (1 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 1.5 */
# define WM8904_BCLK_DIV2 (2 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 2 */
# define WM8904_BCLK_DIV3 (3 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 3 */
# define WM8904_BCLK_DIV4 (4 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 4 */
# define WM8904_BCLK_DIV5 (5 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 5 */
# define WM8904_BCLK_DIV5p5 (6 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 5.5 */
# define WM8904_BCLK_DIV6 (7 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 6 */
# define WM8904_BCLK_DIV8 (8 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 8 */
# define WM8904_BCLK_DIV10 (9 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 10 */
# define WM8904_BCLK_DIV11 (10 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 11 */
# define WM8904_BCLK_DIV12 (11 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 12 */
# define WM8904_BCLK_DIV16 (12 << WM8904_BCLK_DIV_SHIFT) /* SYSCLK / 16 */
@ -476,6 +481,7 @@
# define WM8904_LRCLK_RATE(n) ((uint16_t)(n) << WM8904_LRCLK_RATE_SHIFT)
/* 0x1e DAC Digital Volume Left */
/* 0x1f DAC Digital Volume Right */
#define WM8904_DAC_VU (1 << 8) /* Bit 8: DAC volume update */
@ -493,11 +499,14 @@
# define WM8904_ADCR_DAC_SVOL(n) ((uint16_t)(n) << WM8904_ADCR_DAC_SVOL_SHIFT)
#define WM8904_ADC_TO_DACL_SHIFT (2) /* Bits 2-3: Left DAC digital sidetone source */
#define WM8904_ADC_TO_DACL_MASK (3 << WM8904_ADC_TO_DACL_SHIFT)
# define WM8904_ADC_TO_DACL_NONE (0 << WM8904_ADC_TO_DACL_SHIFT) /* No sidetone */
# define WM8904_ADC_TO_DACL_LEFT (1 << WM8904_ADC_TO_DACL_SHIFT) /* Left ADC */
# define WM8904_ADC_TO_DACL_RIGHT (2 << WM8904_ADC_TO_DACL_SHIFT) /* Right ADC */
#define WM8904_ADC_TO_DACR_SHIFT (0) /* Bits 0-1: Right DAC digital sidetone source */
#define WM8904_ADC_TO_DACR_MASK (3 << WM8904_ADC_TO_DACR_SHIFT)
# define WM8904_ADC_TO_DACR_NONE (0 << WM8904_ADC_TO_DACR_SHIFT) /* No sidetone */
# define WM8904_ADC_TO_DACR_LEFT (1 << WM8904_ADC_TO_DACR_SHIFT) /* Left ADC */
# define WM8904_ADC_TO_DACR_RIGHT (2 << WM8904_ADC_TO_DACR_SHIFT) /* Right ADC */
@ -518,6 +527,7 @@
# define WM8904_DEEMPH_48KHZ (3 << WM8904_DEEMPH_SHIFT) /* 48kHz sample rate */
/* 0x24 ADC Digital Volume Left */
/* 0x25 ADC Digital Volume Right */
#define WM8904_ADC_VU (1 << 8) /* Bit 8: ADC Volume Update */
@ -529,10 +539,12 @@
#define WM8904_ADC_HPF_CUT_SHIFT (5) /* Bits 5-6: ADC digital high pass filter cut-off */
#define WM8904_ADC_HPF_CUT_MASK (3 << WM8904_ADC_HPF_CUT_SHIFT)
# define WM8904_ADC_HPF_CUT_HIFI (0 << WM8904_ADC_HPF_CUT_SHIFT) /* Hi-fi mode */
# define WM8904_ADC_HPF_CUT_VOICE1 (1 << WM8904_ADC_HPF_CUT_SHIFT) /* Voice mode 1 */
# define WM8904_ADC_HPF_CUT_VOICE2 (2 << WM8904_ADC_HPF_CUT_SHIFT) /* Voice mode 2 */
# define WM8904_ADC_HPF_CUT_VOICE3 (3 << WM8904_ADC_HPF_CUT_SHIFT) /* Voice mode 3 */
#define WM8904_ADC_HPF (1 << 4) /* Bit 4: ADC digital high pass filter enable */
#define WM8904_ADCL_DATINV (1 << 1) /* Bit 1: Left ADC invert */
#define WM8904_ADCR_DATINV (1 << 0) /* Bit 0: Right ADC invert */
@ -548,10 +560,12 @@
#define WM8904_DRC_DAC_PATH (1 << 14) /* Bit 14: DRC path select */
#define WM8904_DRC_GS_HYST_LVL_SHIFT (11) /* Bits 11-12: Gain smoothing hysteresis threshold */
#define WM8904_DRC_GS_HYST_LVL_MASK (3 << WM8904_DRC_GS_HYST_LVL_SHIFT)
# define WM8904_DRC_GS_HYST_LVL(n) ((uint16_t)(n) << WM8904_DRC_GS_HYST_LVL_SHIFT)
# define WM8904_DRC_GS_HYST_LOW (0 << WM8904_DRC_GS_HYST_LVL_SHIFT) /* Low */
# define WM8904_DRC_GS_HYST_MEDIUM (1 << WM8904_DRC_GS_HYST_LVL_SHIFT) /* Medium */
# define WM8904_DRC_GS_HYST_HIGH (2 << WM8904_DRC_GS_HYST_LVL_SHIFT) /* High */
#define WM8904_DRC_STARTUP_GAIN_SHIFT (6) /* Bits 6-10: Initial gain at DRC startup */
#define WM8904_DRC_STARTUP_GAIN_MASK (31 << WM8904_DRC_STARTUP_GAIN_SHIFT)
# define WM8904_DRC_STARTUP_GAIN(n) ((uint16_t)(n) << WM8904_DRC_STARTUP_GAIN_SHIFT)
@ -586,14 +600,17 @@
#define WM8904_DRC_HI_COMP_SHIFT (3) /* Bits 3-5: Compressor slope (upper region) */
#define WM8904_DRC_HI_COMP_MASK (7 << WM8904_DRC_HI_COMP_SHIFT)
# define WM8904_DRC_HI_COMP_DIV1 (0 << WM8904_DRC_HI_COMP_SHIFT) /* 1 (no compression) */
# define WM8904_DRC_HI_COMP_DIV2 (1 << WM8904_DRC_HI_COMP_SHIFT) /* 1/2 */
# define WM8904_DRC_HI_COMP_DIV4 (2 << WM8904_DRC_HI_COMP_SHIFT) /* 1/4 */
# define WM8904_DRC_HI_COMP_DIV8 (3 << WM8904_DRC_HI_COMP_SHIFT) /* 1/8 */
# define WM8904_DRC_HI_COMP_DIV16 (4 << WM8904_DRC_HI_COMP_SHIFT) /* 1/16 */
# define WM8904_DRC_HI_COMP_ZERO (5 << WM8904_DRC_HI_COMP_SHIFT) /* 0 */
#define WM8904_DRC_LO_COMP_SHIFT (0) /* Bits 0-2: Compressor slope (lower region)*/
#define WM8904_DRC_LO_COMP_MASK (7 << WM8904_DRC_LO_COMP_SHIFT)
# define WM8904_DRC_LO_COMP(n) ((uint16_t)(n) << WM8904_DRC_LO_COMP_SHIFT)
# define WM8904_DRC_LO_COMP_DIV1 (0 << WM8904_DRC_HI_COMP_SHIFT) /* 1 (no compression) */
# define WM8904_DRC_LO_COMP_DIV2 (1 << WM8904_DRC_HI_COMP_SHIFT) /* 1/2 */
@ -612,6 +629,7 @@
# define WM8904_DRC_KNEE_OP(n) ((uint16_t)(n) << WM8904_DRC_KNEE_OP_SHIFT)
/* 0x2c Analogue Left Input 0 */
/* 0x2d Analogue Right Input 0 */
#define WM8904_INMUTE (1 << 7) /* Bit 7: Input PGA mute */
@ -620,6 +638,7 @@
# define WM8904_IN_VOL(n) ((uint16_t)(n) << WM8904_IN_VOL_SHIFT)
/* 0x2e Analogue Left Input 1 */
/* 0x2f Analogue Right Input 1 */
#define WM8904_IN_CM_ENA (1 << 6) /* Bit 6: Input PGA common mode rejection enable */
@ -642,6 +661,7 @@
# define WM8904_MODE_DIFFMIC (2 << WM8904_MODE_SHIFT) /* Differential MIC */
/* 0x39 Analogue OUT1 Left */
/* 0x3a Analogue OUT1 Right */
#define WM8904_HPOUT_MUTE (1 << 8) /* Bit 8: Headphone output mute */
@ -652,6 +672,7 @@
# define WM8904_HPOUT_VOL(n) ((uint16_t)(n) << WM8904_HPOUT_VOL_SHIFT)
/* 0x3b Analogue OUT2 Left */
/* 0x3c Analogue OUT2 Right */
#define WM8904_LINEOUT_MUTE (1 << 8) /* Bit 8: Headphone output mute */
@ -846,6 +867,7 @@
# define WM8904_FLL_FRATIO_DIV16 (4 << WM8904_FLL_FRATIO_SHIFT) /* Divide by 16 */
/* 0x76 FLL Control 3, Bits 0-15=Fractional multiply for FREF */
/* 0x77 FLL Control 4 */
#define WM8904_FLL_N_SHIFT (5) /* Bits 5-14: Integer multiply for FREF */
@ -871,6 +893,7 @@
# define WM8904_FLL_CLK_REF_DIV2 (1 << WM8904_FLL_CLK_REF_DIV_SHIFT) /* MCLK / 2 */
# define WM8904_FLL_CLK_REF_DIV4 (2 << WM8904_FLL_CLK_REF_DIV_SHIFT) /* MCLK / 4 */
# define WM8904_FLL_CLK_REF_DIV8 (3 << WM8904_FLL_CLK_REF_DIV_SHIFT) /* MCLK / 8 */
#define WM8904_FLL_CLK_REF_SRC_SHIFT (0) /* Bits 0-2: FLL clock source */
#define WM8904_FLL_CLK_REF_SRC_MASK (3 << WM8904_FLL_CLK_REF_SRC_SHIFT)
# define WM8904_FLL_CLK_REF_SRC_MCLK (0 << WM8904_FLL_CLK_REF_SRC_SHIFT)
@ -958,9 +981,13 @@
#define WM8904_BCLK_PD (1 << 0) /* Bit 0: BCLK pull-down resistor enable */
/* Common interrupt bits */
/* 0x7f Interrupt Status */
/* 0x80 Interrupt Status Mask */
/* 0x81 Interrupt Polarity */
/* 0x82 Interrupt Debounce */
#define WM8904_GPIO_BCLK_INT (1 << 9) /* Bit 9: GPIO4 interrupt */
@ -985,6 +1012,7 @@
#define WM8904_EQ_ENA (1 << 0) /* Bit 0: EQ enable */
/* 0x87-0x8b EQ2-EQ6: 5 bit equalizer value */
/* 0x8c-0x9d EQ7-EQ24: 16-bit equalizer value */
/* 0xa1 Control Interface Test 1 */
@ -1012,7 +1040,8 @@
#define WM8904FLL_FRC_NCO_MASK (0x3f << WM8904FLL_FRC_NCO_SHIFT)
# define WM8904FLL_FRC_NCO_VAL(n) ((uint16_t)(n) << WM8904FLL_FRC_NCO_SHIFT)
/* FLL Configuration *********************************************************/
/* FLL Configuration ********************************************************/
/* Default FLL configuration */
#define WM8904_DEFAULT_SAMPRATE 11025 /* Initial sample rate */
@ -1056,10 +1085,11 @@ struct wm8904_dev_s
/* We are an audio lower half driver (We are also the upper "half" of
* the WM8904 driver with respect to the board lower half driver).
*
* Terminology: Our "lower" half audio instances will be called dev for the
* publicly visible version and "priv" for the version that only this driver
* knows. From the point of view of this driver, it is the board lower
* "half" that is referred to as "lower".
* Terminology:
* Our "lower" half audio instances will be called dev for the publicly
* visible version and "priv" for the version that only this driver knows.
* From the point of view of this driver, it is the board lower "half"
* that is referred to as "lower".
*/
struct audio_lowerhalf_s dev; /* WM8904 audio lower half (this device) */
@ -1107,7 +1137,7 @@ struct wm8904_dev_s
****************************************************************************/
#ifdef CONFIG_WM8904_CLKDEBUG
extern const uint8_t g_sysclk_scaleb1[WM8904_BCLK_MAXDIV+1];
extern const uint8_t g_sysclk_scaleb1[WM8904_BCLK_MAXDIV + 1];
extern const uint8_t g_fllratio[WM8904_NFLLRATIO];
#endif

12
drivers/audio/wm8904_debug.c
View file

@ -337,9 +337,12 @@ void wm8904_clock_analysis(FAR struct audio_lowerhalf_s *dev,
syslog(LOG_INFO, " Fref: %lu Hz (after divider)\n", fref);
regval = wm8904_readreg(priv, WM8904_FLL_CTRL2);
frndx = (regval & WM8904_FLL_FRATIO_MASK) >> WM8904_FLL_FRATIO_SHIFT;
tmp = (regval & WM8904_FLL_CTRL_RATE_MASK) >> WM8904_FLL_CTRL_RATE_SHIFT;
outdiv = ((regval & WM8904_FLL_OUTDIV_MASK) >> WM8904_FLL_OUTDIV_SHIFT) + 1;
frndx = (regval & WM8904_FLL_FRATIO_MASK) >>
WM8904_FLL_FRATIO_SHIFT;
tmp = (regval & WM8904_FLL_CTRL_RATE_MASK) >>
WM8904_FLL_CTRL_RATE_SHIFT;
outdiv = ((regval & WM8904_FLL_OUTDIV_MASK) >>
WM8904_FLL_OUTDIV_SHIFT) + 1;
syslog(LOG_INFO, " FLL_CTRL_RATE: Fvco / %u\n", tmp + 1);
@ -397,7 +400,8 @@ void wm8904_clock_analysis(FAR struct audio_lowerhalf_s *dev,
sysclk >>= 1;
}
syslog(LOG_INFO, " SYSCLK: %lu (after divider)\n", (unsigned long)sysclk);
syslog(LOG_INFO, " SYSCLK: %lu (after divider)\n",
(unsigned long)sysclk);
regval = wm8904_readreg(priv, WM8904_CLKRATE2);

125
include/nuttx/audio/audio.h
View file

@ -43,10 +43,11 @@
*
* The Audio driver is split into two parts:
*
* 1) An "upper half", generic driver that provides the common Audio interface
* to application level code, and
* 2) A "lower half", platform-specific driver that implements the low-level
* controls to configure and communicate with the audio device(s).
* 1) An "upper half", generic driver that provides the common Audio
* interface to application level code, and
* 2) A "lower half", platform-specific driver that implements the
* low-level controls to configure and communicate with the audio
* device(s).
*/
/****************************************************************************
@ -67,14 +68,17 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Configuration ************************************************************/
/* CONFIG_AUDIO - Enables Audio driver support
* CONFIG_DEBUG_AUDIO - If enabled (with CONFIG_DEBUG_FEATURES and, optionally,
* CONFIG_DEBUG_INFO), this will generate output that can be used to
* debug Audio drivers.
* CONFIG_DEBUG_AUDIO - If enabled (with CONFIG_DEBUG_FEATURES and,
* optionally, CONFIG_DEBUG_INFO), this will generate output that can
* be used to debug Audio drivers.
*/
/* IOCTL Commands ***********************************************************/
/* The Audio module uses a standard character driver framework. However, a
* lot of the Audio driver functionality is configured via a device control
* interface, such as sampling rate, volume, data format, etc.
@ -126,6 +130,7 @@
#define AUDIOIOC_SETBUFFERINFO _AUDIOIOC(17)
/* Audio Device Types *******************************************************/
/* The NuttX audio interface support different types of audio devices for
* input, output, synthesis, and manipulation of audio data. A given driver/
* device could support a combination of these device type. The following
@ -143,6 +148,7 @@
#define AUDIO_TYPE_EXTENSION 0x80
/* Audio Format Types *******************************************************/
/* The following defines the audio data format types in NuttX. During a
* format query, these will be converted to bit positions within the
* ac_format field, meaning we currently only support up to 16 formats. To
@ -232,7 +238,7 @@
#define AUDIO_SUBSAMPLE_MIN AUDIO_SUBSAMPLE_2X
#define AUDIO_SUBSAMPLE_MAX AUDIO_SUBSAMPLE_16X
/* Supported Bit Rates *************************************************/
/* Supported Bit Rates ******************************************************/
#define AUDIO_BIT_RATE_22K 0x01
#define AUDIO_BIT_RATE_44K 0x02
@ -352,15 +358,22 @@ struct audio_caps_s
uint8_t ac_subtype; /* Capabilities sub-type, if needed */
uint8_t ac_channels; /* Number of channels (1, 2, 5, 7) */
union /* Audio data format(s) for this device */
/* Audio data format(s) for this device */
union
{
uint8_t b[2];
uint16_t hw;
} ac_format;
union /* Device specific controls. For AUDIO_DEVICE_QUERY, */
{ /* this field reports the device type supported */
uint8_t b[4]; /* by this lower-half driver. */
/* Specific controls for AUDIO_DEVICE_QUERY
* this field reports the device type supported
* by this lower-half driver.
*/
union
{
uint8_t b[4];
uint16_t hw[2];
uint32_t w;
#ifdef CONFIG_HAVE_LONG_LONG
@ -384,7 +397,9 @@ struct audio_info_s
uint8_t samplerate; /* Sample Rate of the audio data */
uint8_t channels; /* Number of channels (1, 2, 5, 7) */
uint8_t format; /* Audio data format */
uint8_t subformat; /* Audio subformat (maybe should be combined with format? */
uint8_t subformat; /* Audio subformat
* (maybe should be combined with format?
*/
};
/* This structure describes the preferred number and size of
@ -430,10 +445,10 @@ struct audio_msg_s
#ifdef CONFIG_AUDIO_MULTI_SESSION
FAR void *session; /* Associated channel */
#endif
uint16_t msgId; /* Message ID */
uint16_t msg_id; /* Message ID */
union
{
FAR void *pPtr; /* Buffer being dequeued */
FAR void *ptr; /* Buffer being dequeued */
uint32_t data; /* Message data */
} u;
};
@ -490,14 +505,15 @@ struct audio_ops_s
{
/* This method is called to retrieve the lower-half device capabilities.
* It will be called with device type AUDIO_TYPE_QUERY to request the
* overall capabilities, such as to determine the types of devices supported
* audio formats supported, etc. Then it may be called once or more with
* reported supported device types to determine the specific capabilities
* of that device type (such as MP3 encoder, WMA encoder, PCM output, etc.).
* overall capabilities, such as to determine the types of devices
* supported audio formats supported, etc.
* Then it may be called once or more with reported supported device types
* to determine the specific capabilities of that device type
* (such as MP3 encoder, WMA encoder, PCM output, etc.).
*/
CODE int (*getcaps)(FAR struct audio_lowerhalf_s *dev, int type,
FAR struct audio_caps_s *pCaps);
FAR struct audio_caps_s *caps);
/* This method is called to bind the lower-level driver to the upper-level
* driver and to configure the driver for a specific mode of
@ -509,10 +525,10 @@ struct audio_ops_s
#ifdef CONFIG_AUDIO_MULTI_SESSION
CODE int (*configure)(FAR struct audio_lowerhalf_s *dev,
FAR void *session, FAR const struct audio_caps_s *pCaps);
FAR void *session, FAR const struct audio_caps_s *caps);
#else
CODE int (*configure)(FAR struct audio_lowerhalf_s *dev,
FAR const struct audio_caps_s *pCaps);
FAR const struct audio_caps_s *caps);
#endif
/* This method is called when the driver is closed. The lower half driver
@ -520,16 +536,16 @@ struct audio_ops_s
* output generation. It should also disable the audio hardware and put
* it into the lowest possible power usage state.
*
* Any enqueued Audio Pipeline Buffers that have not been processed / dequeued
* should be dequeued by this function.
* Any enqueued Audio Pipeline Buffers that have not been
* processed / dequeued should be dequeued by this function.
*/
CODE int (*shutdown)(FAR struct audio_lowerhalf_s *dev);
/* Start audio streaming in the configured mode. For input and synthesis
* devices, this means it should begin sending streaming audio data. For output
* or processing type device, it means it should begin processing of any enqueued
* Audio Pipeline Buffers.
* devices, this means it should begin sending streaming audio data.
* For output or processing type device, it means it should begin
* processing of any enqueued Audio Pipeline Buffers.
*/
#ifdef CONFIG_AUDIO_MULTI_SESSION
@ -538,7 +554,9 @@ struct audio_ops_s
CODE int (*start)(FAR struct audio_lowerhalf_s *dev);
#endif
/* Stop audio streaming and/or processing of enqueued Audio Pipeline Buffers */
/* Stop audio streaming and/or processing of enqueued
* Audio Pipeline Buffers
*/
#ifndef CONFIG_AUDIO_EXCLUDE_STOP
#ifdef CONFIG_AUDIO_MULTI_SESSION
@ -548,8 +566,9 @@ struct audio_ops_s
#endif
#endif
/* Pause the audio stream. Should keep current playback context active
* in case a resume is issued. Could be called and then followed by a stop.
/* Pause the audio stream.
* Should keep current playback context active in case a resume is issued.
* Could be called and then followed by a stop.
*/
#ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME
@ -587,16 +606,18 @@ struct audio_ops_s
CODE int (*freebuffer)(FAR struct audio_lowerhalf_s *dev,
FAR struct audio_buf_desc_s *apb);
/* Enqueue a buffer for processing. This is a non-blocking enqueue operation.
* If the lower-half driver's buffer queue is full, then it should return an
* error code of -ENOMEM, and the upper-half driver can decide to either block
* the calling thread or deal with it in a non-blocking manner.
/* Enqueue a buffer for processing.
* This is a non-blocking enqueue operation.
* If the lower-half driver's buffer queue is full, then it should return
* an error code of -ENOMEM, and the upper-half driver can decide to either
* block the calling thread or deal with it in a non-blocking manner.
* For each call to enqueuebuffer, the lower-half driver must call
* audio_dequeuebuffer when it is finished processing the bufferr, passing the
* previously enqueued apb and a dequeue status so that the upper-half driver
* can decide if a waiting thread needs to be release, if the dequeued buffer
* should be passed to the next block in the Audio Pipeline, etc.
* audio_dequeuebuffer when it is finished processing the bufferr, passing
* the previously enqueued apb and a dequeue status so that the upper-half
* driver can decide if a waiting thread needs to be release, if the
* dequeued buffer should be passed to the next block in the
* Audio Pipeline, etc.
*/
CODE int (*enqueuebuffer)(FAR struct audio_lowerhalf_s *dev,
@ -630,7 +651,8 @@ struct audio_ops_s
*/
#ifdef CONFIG_AUDIO_MULTI_SESSION
CODE int (*reserve)(FAR struct audio_lowerhalf_s *dev, FAR void **psession);
CODE int (*reserve)(FAR struct audio_lowerhalf_s *dev,
FAR void **psession);
#else
CODE int (*reserve)(FAR struct audio_lowerhalf_s *dev);
#endif
@ -638,7 +660,8 @@ struct audio_ops_s
/* Release a session. */
#ifdef CONFIG_AUDIO_MULTI_SESSION
CODE int (*release)(FAR struct audio_lowerhalf_s *dev, FAR void *session);
CODE int (*release)(FAR struct audio_lowerhalf_s *dev,
FAR void *session);
#else
CODE int (*release)(FAR struct audio_lowerhalf_s *dev);
#endif
@ -670,7 +693,9 @@ struct audio_lowerhalf_s
FAR audio_callback_t upper;
/* The private opaque pointer to be passed to upper-layer during callbacks */
/* The private opaque pointer to be passed to upper-layer during
* callbacks
*/
FAR void *priv;
@ -698,6 +723,7 @@ extern "C"
/****************************************************************************
* "Upper-Half" Audio Driver Interfaces
****************************************************************************/
/****************************************************************************
* Name: audio_register
*
@ -711,12 +737,12 @@ extern "C"
*
* Input Parameters:
* name - The name of the audio device. This name will be used to generate
* a full path to the driver in the format "/dev/audio/[name]" in the NuttX
* filesystem (i.e. the path "/dev/audio" will be prepended to the supplied
* device name. The recommended convention is to name Audio drivers
* based on the type of functionality they provide, such as "/dev/audio/pcm0",
* "/dev/audio/midi0", "/dev/audio/mp30, etc.
* dev - A pointer to an instance of lower half audio driver. This instance
* a full path to the driver in the format "/dev/audio/[name]" in the
* NuttX filesystem (i.e. the path "/dev/audio" will be prepended to the
* supplied device name. The recommended convention is to name Audio
* drivers based on the type of functionality they provide, such as
* "/dev/audio/pcm0", "/dev/audio/midi0", "/dev/audio/mp30, etc.
* dev - A pointer to an instance of lower half audio driver. This instance
* is bound to the Audio driver and must persists as long as the driver
* persists.
*
@ -731,8 +757,9 @@ int audio_register(FAR const char *name, FAR struct audio_lowerhalf_s *dev);
* Name: abp_alloc
*
* Description:
* Allocated an AP Buffer and prepares it for use. This allocates a dynamically
* allocated buffer that has no special DMA capabilities.
* Allocated an AP Buffer and prepares it for use.
* This allocates a dynamically allocated buffer that has no special
* DMA capabilities.
*
* Input Parameters:
* bufdesc: Pointer to a buffer descriptor

1
include/nuttx/audio/audio_null.h
View file

@ -51,6 +51,7 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Configuration ************************************************************
*
* CONFIG_AUDIO_NULL - Enabled NULL audio device support

18
include/nuttx/audio/cs43l22.h
View file

@ -54,14 +54,16 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Configuration ************************************************************
*
* CONFIG_AUDIO_CS43L22 - Enables CS43L22 support
* CONFIG_CS43L22_INITVOLUME - The initial volume level in the range {0..1000}
* CONFIG_CS43L22_INFLIGHT - Maximum number of buffers that the CS43L22 driver
* will send to the I2S driver before any have completed.
* CONFIG_CS43L22_MSG_PRIO - Priority of messages sent to the CS43L22 worker
* thread.
* CONFIG_CS43L22_INITVOLUME - The initial volume level
* in the range {0..1000}
* CONFIG_CS43L22_INFLIGHT - Maximum number of buffers that the CS43L22
* driver will send to the I2S driver before any have completed.
* CONFIG_CS43L22_MSG_PRIO - Priority of messages sent to the CS43L22
* worker thread.
* CONFIG_CS43L22_BUFFER_SIZE - Preferred buffer size
* CONFIG_CS43L22_NUM_BUFFERS - Preferred number of buffers
* CONFIG_CS43L22_WORKER_STACKSIZE - Stack size to use when creating the the
@ -130,6 +132,7 @@
/****************************************************************************
* Public Types
****************************************************************************/
/* This is the type of the CS43L22 interrupt handler. The lower level code
* will intercept the interrupt and provide the upper level with the private
* data that was provided when the interrupt was attached.
@ -137,8 +140,9 @@
struct cs43l22_lower_s; /* Forward reference. Defined below */
typedef CODE int (*cs43l22_handler_t)(FAR const struct cs43l22_lower_s *lower,
FAR void *arg);
typedef CODE int (*cs43l22_handler_t)
(FAR const struct cs43l22_lower_s *lower,
FAR void *arg);
/* A reference to a structure of this type must be passed to the CS43L22
* driver. This structure provides information about the configuration

35
include/nuttx/audio/i2s.h
View file

@ -53,6 +53,7 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Access macros ************************************************************/
/****************************************************************************
@ -261,6 +262,7 @@
/****************************************************************************
* Public Types
****************************************************************************/
/* Transfer complete callbacks */
struct i2s_dev_s;
@ -273,27 +275,36 @@ struct i2s_ops_s
{
/* Receiver methods */
CODE int (*i2s_rxchannels)(FAR struct i2s_dev_s *dev, uint8_t channels);
CODE uint32_t (*i2s_rxsamplerate)(FAR struct i2s_dev_s *dev, uint32_t rate);
CODE uint32_t (*i2s_rxdatawidth)(FAR struct i2s_dev_s *dev, int bits);
CODE int (*i2s_rxchannels)(FAR struct i2s_dev_s *dev,
uint8_t channels);
CODE uint32_t (*i2s_rxsamplerate)(FAR struct i2s_dev_s *dev,
uint32_t rate);
CODE uint32_t (*i2s_rxdatawidth)(FAR struct i2s_dev_s *dev,
int bits);
CODE int (*i2s_receive)(FAR struct i2s_dev_s *dev,
FAR struct ap_buffer_s *apb, i2s_callback_t callback,
FAR void *arg, uint32_t timeout);
FAR struct ap_buffer_s *apb,
i2s_callback_t callback,
FAR void *arg,
uint32_t timeout);
/* Transmitter methods */
CODE int (*i2s_txchannels)(FAR struct i2s_dev_s *dev, uint8_t channels);
CODE uint32_t (*i2s_txsamplerate)(FAR struct i2s_dev_s *dev, uint32_t rate);
CODE uint32_t (*i2s_txdatawidth)(FAR struct i2s_dev_s *dev, int bits);
CODE int (*i2s_txchannels)(FAR struct i2s_dev_s *dev,
uint8_t channels);
CODE uint32_t (*i2s_txsamplerate)(FAR struct i2s_dev_s *dev,
uint32_t rate);
CODE uint32_t (*i2s_txdatawidth)(FAR struct i2s_dev_s *dev,
int bits);
CODE int (*i2s_send)(FAR struct i2s_dev_s *dev,
FAR struct ap_buffer_s *apb, i2s_callback_t callback,
FAR void *arg, uint32_t timeout);
FAR struct ap_buffer_s *apb,
i2s_callback_t callback,
FAR void *arg,
uint32_t timeout);
/* Ioctl */
CODE int (*i2s_ioctl)(FAR struct i2s_dev_s *dev,
int cmd, unsigned long arg);
};
/* I2S private data. This structure only defines the initial fields of the
@ -320,7 +331,7 @@ extern "C"
#endif
/****************************************************************************
* Public Functions
* Public Functions Prototypes
****************************************************************************/
/****************************************************************************

5
include/nuttx/audio/pcm.h
View file

@ -50,6 +50,7 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Configuration ************************************************************
*
* CONFIG_AUDIO_FORMAT_PCM - Enabled PCM support
@ -64,6 +65,7 @@
/* Default configuration values */
/* WAVE Header Definitions **************************************************/
/* All values are little 32-bit or 16-bit endian */
#define WAV_HDR_CHUNKID 0x46464952 /* "RIFF" */
@ -78,6 +80,7 @@
/****************************************************************************
* Public Types
****************************************************************************/
/* The standard WAV header consist of three chunks
*
* 1. A WAV header chunk,
@ -119,7 +122,7 @@ struct wav_header_s
struct wav_datachunk_s data;
};
/****************************************************************************
/****************************************************************************
* Public Data
****************************************************************************/

2
include/nuttx/audio/wm8776.h
View file

@ -61,7 +61,6 @@ struct wm8776_lower_s
uint8_t address; /* 7-bit I2C address (only bits 0-6 used) */
};
/****************************************************************************
* Public Data
****************************************************************************/
@ -87,7 +86,6 @@ FAR struct audio_lowerhalf_s *
FAR struct i2s_dev_s *i2s,
FAR const struct wm8776_lower_s *lower);
#undef EXTERN
#ifdef __cplusplus
}

2
include/nuttx/audio/wm8904.h
View file

@ -54,6 +54,7 @@
/****************************************************************************
* Pre-processor Definitions
****************************************************************************/
/* Configuration ************************************************************
*
* CONFIG_AUDIO_WM8904 - Enables WM8904 support
@ -129,6 +130,7 @@
/****************************************************************************
* Public Types
****************************************************************************/
/* This is the type of the WM8904 interrupt handler. The lower level code
* will intercept the interrupt and provide the upper level with the private
* data that was provided when the interrupt was attached.