kingkevin
/
passkey_fw
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Read sample rate directly from control transfer to eliminate callback.

This commit is contained in:
HiFiPhile 2023-10-17 13:06:55 +02:00
parent 656e1e416a
commit 0a1d6cf4d0
3 changed files with 194 additions and 164 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
examples/device/audio_4_channel_mic/src/main.c
View File

@ -407,9 +407,8 @@ bool tud_audio_get_req_entity_cb(uint8_t rhport, tusb_control_request_t const *
{
case AUDIO_CS_REQ_CUR:
TU_LOG2(" Get Sample Freq.\r\n");
// Set sample rate for flow control
tud_audio_set_tx_flow_control(sampFreq);
return tud_control_xfer(rhport, p_request, &sampFreq, sizeof(sampFreq));
// Buffered control transfer is needed for IN flow control to work
return tud_audio_buffer_and_schedule_control_xfer(rhport, p_request, &sampFreq, sizeof(sampFreq));
case AUDIO_CS_REQ_RANGE:
TU_LOG2(" Get Sample Freq. range\r\n");

343
src/class/audio/audio_device.c
View File

@ -461,7 +461,8 @@ static inline uint8_t tu_desc_subtype(void const* desc)
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
static bool audiod_tx_calc_packet_size(const uint16_t* norminal_size, uint16_t data_size, uint16_t fifo_size, uint16_t* packet_size);
static bool audiod_calc_tx_packet_sz(audiod_function_t* audio);
static uint16_t audiod_tx_packet_size(const uint16_t* norminal_size, uint16_t data_count, uint16_t fifo_depth, uint16_t max_size);
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_OUT && CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
@ -832,58 +833,6 @@ uint16_t tud_audio_int_ctr_n_write(uint8_t func_id, uint8_t const* buffer, uint1
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
bool tud_audio_n_set_tx_flow_control(uint8_t func_id, uint32_t sample_rate)
{
TU_VERIFY(func_id < CFG_TUD_AUDIO && _audiod_fct[func_id].p_desc != NULL);
audiod_function_t* audio = &_audiod_fct[func_id];
TU_VERIFY(audio->format_type_tx == AUDIO_FORMAT_TYPE_I);
TU_VERIFY(audio->n_channels_tx);
TU_VERIFY(audio->n_bytes_per_sampe_tx);
TU_VERIFY(audio->interval_tx);
if (sample_rate == 0)
{
audio->packet_sz_tx[0] = 0;
audio->packet_sz_tx[1] = 0;
audio->packet_sz_tx[2] = 0;
return false;
}
const uint8_t interval = (tud_speed_get() == TUSB_SPEED_FULL) ? audio->interval_tx : 1 << (audio->interval_tx - 1);
const uint32_t sample_normimal = sample_rate * interval / ((tud_speed_get() == TUSB_SPEED_FULL) ? 1000 : 8000);
const uint32_t sample_reminder = sample_rate * interval % ((tud_speed_get() == TUSB_SPEED_FULL) ? 1000 : 8000);
const uint16_t packet_sz_tx_min = (sample_normimal - 1) * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
const uint16_t packet_sz_tx_norm = sample_normimal * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
const uint16_t packet_sz_tx_max = (sample_normimal + 1) * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
TU_ASSERT(packet_sz_tx_max <= audio->ep_in_sz);
// Frmt20.pdf 2.3.1.1 USB Packets
if (sample_reminder)
{
// All virtual frame packets must either contain INT(nav) audio slots (small VFP) or INT(nav)+1 (large VFP) audio slots
audio->packet_sz_tx[0] = packet_sz_tx_norm;
audio->packet_sz_tx[1] = packet_sz_tx_norm;
audio->packet_sz_tx[2] = packet_sz_tx_max;
} else
{
// In the case where nav = INT(nav), ni may vary between INT(nav)-1 (small VFP), INT(nav)
// (medium VFP) and INT(nav)+1 (large VFP).
audio->packet_sz_tx[0] = packet_sz_tx_min;
audio->packet_sz_tx[1] = packet_sz_tx_norm;
audio->packet_sz_tx[2] = packet_sz_tx_max;
}
return true;
}
#endif
// This function is called once a transmit of an audio packet was successfully completed. Here, we encode samples and place it in IN EP's buffer for next transmission.
// If you prefer your own (more efficient) implementation suiting your purpose set CFG_TUD_AUDIO_ENABLE_ENCODING = 0 and use tud_audio_n_write.
@ -949,12 +898,8 @@ static bool audiod_tx_done_cb(uint8_t rhport, audiod_function_t * audio)
#else
// No support FIFOs, if no linear buffer required schedule transmit, else put data into linear buffer and schedule
#if CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
uint16_t tgt_packet_sz;
// packet_sz_tx is based on total packet size, here we want size for each support buffer.
if (audiod_tx_calc_packet_size(audio->packet_sz_tx, tu_fifo_count(&audio->ep_in_ff), audio->ep_in_ff.depth, &tgt_packet_sz))
n_bytes_tx = tgt_packet_sz;
else
n_bytes_tx = tu_min16(tu_fifo_count(&audio->ep_in_ff), audio->ep_in_sz);
n_bytes_tx = audiod_tx_packet_size(audio->packet_sz_tx, tu_fifo_count(&audio->ep_in_ff), audio->ep_in_ff.depth, audio->ep_in_sz);
#else
n_bytes_tx = tu_min16(tu_fifo_count(&audio->ep_in_ff), audio->ep_in_sz); // Limit up to max packet size, more can not be done for ISO
#endif
@ -1056,7 +1001,6 @@ static uint16_t audiod_encode_type_I_pcm(uint8_t rhport, audiod_function_t* audi
// Determine amount of samples
uint8_t const n_ff_used = audio->n_ff_used_tx;
uint16_t const capPerFF = audio->ep_in_sz / n_ff_used; // Sample capacity per FIFO in bytes
uint16_t nBytesPerFFToSend = tu_fifo_count(&audio->tx_supp_ff[0]);
uint8_t cnt_ff;
@ -1070,19 +1014,18 @@ static uint16_t audiod_encode_type_I_pcm(uint8_t rhport, audiod_function_t* audi
}
#if CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
uint16_t tgt_packet_sz;
const uint16_t norm_packet_sz_tx[3] = {audio->packet_sz_tx[0] / n_ff_used,
audio->packet_sz_tx[1] / n_ff_used,
audio->packet_sz_tx[2] / n_ff_used};
// packet_sz_tx is based on total packet size, here we want size for each support buffer.
if (audiod_tx_calc_packet_size(audio->packet_sz_tx, nBytesPerFFToSend * n_ff_used, audio->tx_supp_ff[0].depth * n_ff_used, &tgt_packet_sz))
nBytesPerFFToSend = tgt_packet_sz / n_ff_used;
#endif
nBytesPerFFToSend = audiod_tx_packet_size(norm_packet_sz_tx, nBytesPerFFToSend, audio->tx_supp_ff[0].depth, audio->ep_in_sz / n_ff_used);
// Check if there is enough data
if (nBytesPerFFToSend == 0) return 0;
#else
// Check if there is enough data
if (nBytesPerFFToSend == 0) return 0;
// Limit to maximum sample number - THIS IS A POSSIBLE ERROR SOURCE IF TOO MANY SAMPLE WOULD NEED TO BE SENT BUT CAN NOT!
nBytesPerFFToSend = tu_min16(nBytesPerFFToSend, capPerFF);
#if !CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
nBytesPerFFToSend = tu_min16(nBytesPerFFToSend, audio->ep_in_sz / n_ff_used);
// Round to full number of samples (flooring)
uint16_t const nSlotSize = audio->n_channels_per_ff_tx * audio->n_bytes_per_sampe_tx;
nBytesPerFFToSend = (nBytesPerFFToSend / nSlotSize) * nSlotSize;
@ -1349,7 +1292,7 @@ void audiod_init(void)
#endif // CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_ENABLE_ENCODING
// Set encoding parameters for Type_I formats
#if CFG_TUD_AUDIO_ENABLE_TYPE_I_ENCODING
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_ENABLE_TYPE_I_ENCODING
switch (i)
{
#if CFG_TUD_AUDIO_FUNC_1_TX_SUPP_SW_FIFO_SZ > 0
@ -1529,84 +1472,114 @@ uint16_t audiod_open(uint8_t rhport, tusb_desc_interface_t const * itf_desc, uin
}
#if USE_ISO_EP_ALLOCATION
#if CFG_TUD_AUDIO_ENABLE_EP_IN
uint8_t ep_in = 0;
uint16_t ep_in_size = 0;
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_OUT
uint8_t ep_out = 0;
uint16_t ep_out_size = 0;
#endif
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
uint8_t ep_fb = 0;
#endif
uint8_t const *p_desc = _audiod_fct[i].p_desc;
uint8_t const *p_desc_end = p_desc + _audiod_fct[i].desc_length - TUD_AUDIO_DESC_IAD_LEN;
while (p_desc < p_desc_end)
{
if (tu_desc_type(p_desc) == TUSB_DESC_ENDPOINT)
{
tusb_desc_endpoint_t const *desc_ep = (tusb_desc_endpoint_t const *) p_desc;
if (desc_ep->bmAttributes.xfer == TUSB_XFER_ISOCHRONOUS)
{
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
// Explicit feedback EP
if (desc_ep->bmAttributes.usage == 1)
{
ep_fb = desc_ep->bEndpointAddress;
}
#endif
// Data EP
if (desc_ep->bmAttributes.usage == 0)
{
if (tu_edpt_dir(desc_ep->bEndpointAddress) == TUSB_DIR_IN)
{
#if CFG_TUD_AUDIO_ENABLE_EP_IN
ep_in = desc_ep->bEndpointAddress;
ep_in_size = TU_MAX(tu_edpt_packet_size(desc_ep), ep_in_size);
#if CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
_audiod_fct[i].interval_tx = desc_ep->bInterval;
#endif
uint8_t ep_in = 0;
uint16_t ep_in_size = 0;
#endif
} else
{
#if CFG_TUD_AUDIO_ENABLE_EP_OUT
ep_out = desc_ep->bEndpointAddress;
ep_out_size = TU_MAX(tu_edpt_packet_size(desc_ep), ep_out_size);
uint8_t ep_out = 0;
uint16_t ep_out_size = 0;
#endif
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
uint8_t ep_fb = 0;
#endif
uint8_t const *p_desc = _audiod_fct[i].p_desc;
uint8_t const *p_desc_end = p_desc + _audiod_fct[i].desc_length - TUD_AUDIO_DESC_IAD_LEN;
while (p_desc < p_desc_end)
{
if (tu_desc_type(p_desc) == TUSB_DESC_ENDPOINT)
{
tusb_desc_endpoint_t const *desc_ep = (tusb_desc_endpoint_t const *) p_desc;
if (desc_ep->bmAttributes.xfer == TUSB_XFER_ISOCHRONOUS)
{
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
// Explicit feedback EP
if (desc_ep->bmAttributes.usage == 1)
{
ep_fb = desc_ep->bEndpointAddress;
}
#endif
// Data EP
if (desc_ep->bmAttributes.usage == 0)
{
if (tu_edpt_dir(desc_ep->bEndpointAddress) == TUSB_DIR_IN)
{
#if CFG_TUD_AUDIO_ENABLE_EP_IN
ep_in = desc_ep->bEndpointAddress;
ep_in_size = TU_MAX(tu_edpt_packet_size(desc_ep), ep_in_size);
#endif
} else
{
#if CFG_TUD_AUDIO_ENABLE_EP_OUT
ep_out = desc_ep->bEndpointAddress;
ep_out_size = TU_MAX(tu_edpt_packet_size(desc_ep), ep_out_size);
#endif
}
}
}
}
p_desc = tu_desc_next(p_desc);
}
#if CFG_TUD_AUDIO_ENABLE_EP_IN
if (ep_in)
{
usbd_edpt_iso_alloc(rhport, ep_in, ep_in_size);
}
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_OUT
if (ep_out)
{
usbd_edpt_iso_alloc(rhport, ep_out, ep_out_size);
}
#endif
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
if (ep_fb)
{
usbd_edpt_iso_alloc(rhport, ep_fb, 4);
}
#endif
}
#endif // USE_ISO_EP_ALLOCATION
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
{
uint8_t const *p_desc = _audiod_fct[i].p_desc;
uint8_t const *p_desc_end = p_desc + _audiod_fct[i].desc_length - TUD_AUDIO_DESC_IAD_LEN;
while (p_desc < p_desc_end)
{
if (tu_desc_type(p_desc) == TUSB_DESC_ENDPOINT)
{
tusb_desc_endpoint_t const *desc_ep = (tusb_desc_endpoint_t const *) p_desc;
if (desc_ep->bmAttributes.xfer == TUSB_XFER_ISOCHRONOUS)
{
if (desc_ep->bmAttributes.usage == 0)
{
if (tu_edpt_dir(desc_ep->bEndpointAddress) == TUSB_DIR_IN)
{
_audiod_fct[i].interval_tx = desc_ep->bInterval;
}
}
}
} else
if (tu_desc_type(p_desc) == TUSB_DESC_CS_INTERFACE && tu_desc_subtype(p_desc) == AUDIO_CS_AC_INTERFACE_OUTPUT_TERMINAL)
{
if(tu_unaligned_read16(p_desc + 4) == AUDIO_TERM_TYPE_USB_STREAMING)
{
_audiod_fct[i].bclock_id_tx = p_desc[8];
}
}
p_desc = tu_desc_next(p_desc);
}
p_desc = tu_desc_next(p_desc);
}
#if CFG_TUD_AUDIO_ENABLE_EP_IN
if (ep_in)
{
usbd_edpt_iso_alloc(rhport, ep_in, ep_in_size);
}
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_OUT
if (ep_out)
{
usbd_edpt_iso_alloc(rhport, ep_out, ep_out_size);
}
#endif
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
if (ep_fb)
{
usbd_edpt_iso_alloc(rhport, ep_fb, 4);
}
#endif
#endif // USE_ISO_EP_ALLOCATION
#endif // CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
break;
}
@ -1910,6 +1883,10 @@ static bool audiod_set_interface(uint8_t rhport, tusb_control_request_t const *
if (disable) usbd_sof_enable(rhport, false);
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
audiod_calc_tx_packet_sz(audio);
#endif
tud_control_status(rhport, p_request);
return true;
@ -2352,6 +2329,19 @@ bool tud_audio_buffer_and_schedule_control_xfer(uint8_t rhport, tusb_control_req
// Copy into buffer
TU_VERIFY(0 == tu_memcpy_s(_audiod_fct[func_id].ctrl_buf, _audiod_fct[func_id].ctrl_buf_sz, data, (size_t)len));
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
// Find data for sampling_frequency_control
if (p_request->bmRequestType_bit.type == TUSB_REQ_TYPE_CLASS && p_request->bmRequestType_bit.recipient == TUSB_REQ_RCPT_INTERFACE)
{
uint8_t entityID = TU_U16_HIGH(p_request->wIndex);
uint8_t ctrlSel = TU_U16_HIGH(p_request->wValue);
if (_audiod_fct[func_id].bclock_id_tx == entityID && ctrlSel == AUDIO_CS_CTRL_SAM_FREQ && p_request->bRequest == AUDIO_CS_REQ_CUR)
{
_audiod_fct[func_id].sample_rate_tx = tu_unaligned_read32(_audiod_fct[func_id].ctrl_buf);
}
}
#endif
// Schedule transmit
return tud_control_xfer(rhport, p_request, (void*)_audiod_fct[func_id].ctrl_buf, len);
}
@ -2579,28 +2569,79 @@ static void audiod_parse_for_AS_params(audiod_function_t* audio, uint8_t const *
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
static bool audiod_tx_calc_packet_size(const uint16_t* norminal_size, uint16_t data_size, uint16_t fifo_size, uint16_t* packet_size)
static bool audiod_calc_tx_packet_sz(audiod_function_t* audio)
{
TU_VERIFY(norminal_size[1]);
TU_VERIFY(audio->format_type_tx == AUDIO_FORMAT_TYPE_I);
TU_VERIFY(audio->n_channels_tx);
TU_VERIFY(audio->n_bytes_per_sampe_tx);
TU_VERIFY(audio->interval_tx);
TU_VERIFY(audio->sample_rate_tx);
// This flow control method need a FIFO size of 4*Navg
TU_VERIFY(norminal_size[1] <= fifo_size * 4);
const uint8_t interval = (tud_speed_get() == TUSB_SPEED_FULL) ? audio->interval_tx : 1 << (audio->interval_tx - 1);
if (data_size < norminal_size[0])
*packet_size = 0;
else
const uint32_t sample_normimal = audio->sample_rate_tx * interval / ((tud_speed_get() == TUSB_SPEED_FULL) ? 1000 : 8000);
const uint32_t sample_reminder = audio->sample_rate_tx * interval % ((tud_speed_get() == TUSB_SPEED_FULL) ? 1000 : 8000);
const uint16_t packet_sz_tx_min = (sample_normimal - 1) * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
const uint16_t packet_sz_tx_norm = sample_normimal * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
const uint16_t packet_sz_tx_max = (sample_normimal + 1) * audio->n_channels_tx * audio->n_bytes_per_sampe_tx;
// Endpoint size must larger than packet size
TU_ASSERT(packet_sz_tx_max <= audio->ep_in_sz);
// Frmt20.pdf 2.3.1.1 USB Packets
if (sample_reminder)
{
uint16_t slot_size = norminal_size[2] - norminal_size[1];
if (data_size < fifo_size / 2 - slot_size)
*packet_size = norminal_size[0];
else if (data_size > fifo_size / 2 + slot_size)
*packet_size = norminal_size[2];
else
*packet_size = norminal_size[1];
// All virtual frame packets must either contain INT(nav) audio slots (small VFP) or INT(nav)+1 (large VFP) audio slots
audio->packet_sz_tx[0] = packet_sz_tx_norm;
audio->packet_sz_tx[1] = packet_sz_tx_norm;
audio->packet_sz_tx[2] = packet_sz_tx_max;
} else
{
// In the case where nav = INT(nav), ni may vary between INT(nav)-1 (small VFP), INT(nav)
// (medium VFP) and INT(nav)+1 (large VFP).
audio->packet_sz_tx[0] = packet_sz_tx_min;
audio->packet_sz_tx[1] = packet_sz_tx_norm;
audio->packet_sz_tx[2] = packet_sz_tx_max;
}
return true;
}
static uint16_t audiod_tx_packet_size(const uint16_t* norminal_size, uint16_t data_count, uint16_t fifo_depth, uint16_t max_depth)
{
// Flow control need a FIFO size of at least 4*Navg
if(norminal_size[1] && norminal_size[1] <= fifo_depth * 4)
{
uint16_t packet_size;
uint16_t slot_size = norminal_size[2] - norminal_size[1];
if (data_count < fifo_depth / 2 - slot_size)
{
if (data_count < norminal_size[0])
{
// If you get here frequently, then your I2S clock deviation is too big !
packet_size = 0;
} else
{
packet_size = norminal_size[0];
}
}
else if (data_count > fifo_depth / 2 + slot_size)
{
packet_size = norminal_size[2];
} else
{
packet_size = norminal_size[1];
}
// Normally this cap is not necessary
return tu_min16(packet_size, max_depth);
} else
{
return tu_min16(data_count, max_depth);
}
}
#endif
#if CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP

10
src/class/audio/audio_device.h
View File

@ -397,9 +397,6 @@ tu_fifo_t* tud_audio_n_get_tx_support_ff (uint8_t func_id, uint8_t ff_i
uint16_t tud_audio_int_ctr_n_write (uint8_t func_id, uint8_t const* buffer, uint16_t len);
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
bool tud_audio_n_set_tx_flow_control (uint8_t func_id, uint32_t sample_rate);
#endif
//--------------------------------------------------------------------+
// Application API (Interface0)
@ -679,13 +676,6 @@ static inline uint16_t tud_audio_int_ctr_write(uint8_t const* buffer, uint16_t l
}
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_IN && CFG_TUD_AUDIO_EP_IN_FLOW_CONTROL
static inline bool tud_audio_set_tx_flow_control(uint32_t sample_rate)
{
return tud_audio_n_set_tx_flow_control(0, sample_rate);
}
#endif
#if CFG_TUD_AUDIO_ENABLE_EP_OUT && CFG_TUD_AUDIO_ENABLE_FEEDBACK_EP
static inline bool tud_audio_fb_set(uint32_t feedback)