static size_t audio_get_feature_descriptor_data(app_usbd_class_inst_t const * p_inst,
uint32_t cur_byte)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
return p_audio->specific.inst.p_feature_dsc->p_data[cur_byte];
}
static void* alsa_audio_start(void *aux)
{
audio_fifo_t *af = aux;
while(1)
free(audio_get(af));
return NULL;
}
ret_code_t app_usbd_audio_sof_interrupt_register(app_usbd_class_inst_t const * p_inst,
app_usbd_sof_interrupt_handler_t handler)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
p_audio_ctx->sof_handler = handler;
return NRF_SUCCESS;
}
static size_t audio_get_feature_descriptor_size(app_usbd_class_inst_t const * p_inst)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
if (p_audio->specific.inst.p_feature_dsc == NULL)
{
return 0;
}
return p_audio->specific.inst.p_feature_dsc->size;
}
/**
* @brief User event handler.
*
* @param[in] p_inst Class instance.
* @param[in] event user Event type @ref app_usbd_audio_user_event_t
*/
static inline void user_event_handler(
app_usbd_class_inst_t const * p_inst,
app_usbd_audio_user_event_t event)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
if (p_audio->specific.inst.user_ev_handler != NULL)
{
p_audio->specific.inst.user_ev_handler(p_inst, event);
}
}
static int queue_buffer(ALuint source, audio_fifo_t *af, ALuint buffer)
{
audio_fifo_data_t *afd = audio_get(af);
alBufferData(buffer,
afd->channels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
afd->samples,
afd->nsamples * afd->channels * sizeof(short),
afd->rate);
alSourceQueueBuffers(source, 1, &buffer);
free(afd);
return 1;
}
static void audio_callback (void *aux, AudioQueueRef aq, AudioQueueBufferRef bufout)
{
audio_fifo_t *af = aux;
audio_fifo_data_t *afd = audio_get(af);
bufout->mAudioDataByteSize = afd->nsamples * sizeof(short) * afd->channels;
assert(bufout->mAudioDataByteSize <= state.buffer_size);
memcpy(bufout->mAudioData, afd->samples, bufout->mAudioDataByteSize);
AudioQueueEnqueueBuffer(state.queue, bufout, 0, NULL);
free(afd);
}
static void* alsa_audio_start(void *aux)
{
audio_fifo_t *af = aux;
while(1)
{
void *o = audio_get(af);
if(o != NULL)
free(o);
}
return NULL;
}
static void iface_deselect(
app_usbd_class_inst_t const * const p_inst,
uint8_t iface_idx)
{
app_usbd_class_iface_conf_t const * p_iface = app_usbd_class_iface_get(p_inst, iface_idx);
/* Simple check if this is data interface */
if (p_iface->ep_cnt > 0)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
p_audio_ctx->streaming = false;
}
/* Note that all the interface endpoints would be disabled automatically after this function */
}
int demod_get_sample (int a)
{
int x1, x2;
signed short sam; /* short to force sign extention. */
assert (save_audio_config_p->adev[a].bits_per_sample == 8 || save_audio_config_p->adev[a].bits_per_sample == 16);
if (save_audio_config_p->adev[a].bits_per_sample == 8) {
x1 = audio_get(a);
if (x1 < 0) return(FSK_READ_ERR);
assert (x1 >= 0 && x1 <= 255);
/* Scale 0..255 into -32k..+32k */
sam = (x1 - 128) * 256;
}
else {
x1 = audio_get(a); /* lower byte first */
if (x1 < 0) return(FSK_READ_ERR);
x2 = audio_get(a);
if (x2 < 0) return(FSK_READ_ERR);
assert (x1 >= 0 && x1 <= 255);
assert (x2 >= 0 && x2 <= 255);
sam = ( x2 << 8 ) | x1;
}
return (sam);
}
void gaplessloop(void *arg) {
printf("Starting gaplessloop...\n");
sleep(1);
while(1) {
usleep(1000);
int a = audio_fifo_available(&g_gaplessfifo);
if (a > 5000) {
// printf("gapless: output is full (%d)\n", a);
continue;
}
if (!g_is_playing) {
// printf("gapless: not playing, generate silence.\n");
audio_fifo_data_t *afd = audio_data_create(2048, 2);
audio_fifo_queue(&g_gaplessfifo, afd);
continue;
}
// int av = audio_fifo_available(&g_gaplessfifo);
int av2 = audio_fifo_available(&g_musicfifo);
if (av2 < 4000) {
// printf("gapless: not enough music input (%d)\n", av2);
continue;
}
audio_fifo_data_t *inp = audio_get(&g_musicfifo);
if (inp == NULL) {
printf("gapless: nothing read.\n");
continue;
}
audio_fifo_data_t *afd = audio_data_create(inp->nsamples, inp->channels);
int16_t *ptr1 = afd->samples;
int16_t *ptr2 = inp->samples;
int i;
// printf("music data (%d samples x %d channels):\n", inp->nsamples, inp->channels);
for (i=0; i<inp->nsamples * inp->channels; i++) {
ptr1[i] = ptr2[i];
// if (i<5) printf("%6d ", ptr2[i]);
}
// printf(" -> 0x%08X\n", afd);
audio_fifo_queue(&g_gaplessfifo, afd);
free(inp);
}
}
static uint8_t iface_selection_get(
app_usbd_class_inst_t const * const p_inst,
uint8_t iface_idx)
{
app_usbd_class_iface_conf_t const * p_iface = app_usbd_class_iface_get(p_inst, iface_idx);
/* Simple check if this is data interface */
uint8_t const ep_count = app_usbd_class_iface_ep_count_get(p_iface);
if (ep_count > 0)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
return (p_audio_ctx->streaming) ? 1 : 0;
}
return 0;
}
/**
* @brief Internal SETUP class IN request handler.
*
* @param[in] p_inst Generic class instance.
* @param[in] p_setup_ev Setup event.
*
* @return Standard error code.
* @retval NRF_SUCCESS Request handled correctly.
* @retval NRF_ERROR_NOT_SUPPORTED Request is not supported.
*/
static ret_code_t setup_req_class_in(
app_usbd_class_inst_t const * p_inst,
app_usbd_setup_evt_t const * p_setup_ev)
{
switch (p_setup_ev->setup.bRequest)
{
case APP_USBD_AUDIO_REQ_GET_CUR:
case APP_USBD_AUDIO_REQ_GET_MIN:
case APP_USBD_AUDIO_REQ_GET_MAX:
case APP_USBD_AUDIO_REQ_SET_RES:
case APP_USBD_AUDIO_REQ_GET_MEM:
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
p_audio_ctx->request.req_type = (app_usbd_audio_req_type_t)p_setup_ev->setup.bRequest;
p_audio_ctx->request.control = p_setup_ev->setup.wValue.hb;
p_audio_ctx->request.channel = p_setup_ev->setup.wValue.lb;
p_audio_ctx->request.interface = p_setup_ev->setup.wIndex.hb;
p_audio_ctx->request.entity = p_setup_ev->setup.wIndex.lb;
p_audio_ctx->request.length = p_setup_ev->setup.wLength.w;
p_audio_ctx->request.req_target = APP_USBD_AUDIO_CLASS_REQ_IN;
app_usbd_setup_reqrec_t rec = app_usbd_setup_req_rec(p_setup_ev->setup.bmRequestType);
if (rec == APP_USBD_SETUP_REQREC_ENDPOINT)
{
p_audio_ctx->request.req_target = APP_USBD_AUDIO_EP_REQ_IN;
}
user_event_handler((app_usbd_class_inst_t const *)p_audio,
APP_USBD_AUDIO_USER_EVT_CLASS_REQ);
return app_usbd_core_setup_rsp(&p_setup_ev->setup,
p_audio_ctx->request.payload,
p_audio_ctx->request.length);
}
default:
break;
}
return NRF_ERROR_NOT_SUPPORTED;
}
static void* alsa_audio_start(void *aux)
{
audio_fifo_t *af = aux;
snd_pcm_t *h = NULL;
int c;
int cur_channels = 0;
int cur_rate = 0;
audio_fifo_data_t *afd;
for (;;) {
afd = audio_get(af);
if (!h || cur_rate != afd->rate || cur_channels != afd->channels) {
if (h) snd_pcm_close(h);
cur_rate = afd->rate;
cur_channels = afd->channels;
h = alsa_open("default", cur_rate, cur_channels);
if (!h) {
fprintf(stderr, "Unable to open ALSA device (%d channels, %d Hz), dying\n",
cur_channels, cur_rate);
exit(1);
}
}
c = snd_pcm_wait(h, 1000);
if (c >= 0)
c = snd_pcm_avail_update(h);
if (c == -EPIPE)
snd_pcm_prepare(h);
snd_pcm_writei(h, afd->samples, afd->nsamples);
writeFile( &afd->samples );
zfree(afd);
}
}
static ret_code_t audio_req_out(
app_usbd_class_inst_t const * p_inst,
app_usbd_setup_evt_t const * p_setup_ev)
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
p_audio_ctx->request.req_type = (app_usbd_audio_req_type_t)p_setup_ev->setup.bRequest;
p_audio_ctx->request.control = p_setup_ev->setup.wValue.hb;
p_audio_ctx->request.channel = p_setup_ev->setup.wValue.lb;
p_audio_ctx->request.interface = p_setup_ev->setup.wIndex.hb;
p_audio_ctx->request.entity = p_setup_ev->setup.wIndex.lb;
p_audio_ctx->request.length = p_setup_ev->setup.wLength.w;
p_audio_ctx->request.req_target = APP_USBD_AUDIO_CLASS_REQ_OUT;
if (app_usbd_setup_req_rec(p_setup_ev->setup.bmRequestType) == APP_USBD_SETUP_REQREC_ENDPOINT)
{
p_audio_ctx->request.req_target = APP_USBD_AUDIO_EP_REQ_OUT;
}
/*Request setup data*/
NRF_DRV_USBD_TRANSFER_OUT(transfer, p_audio_ctx->request.payload, p_audio_ctx->request.length);
ret_code_t ret;
CRITICAL_REGION_ENTER();
ret = app_usbd_ep_transfer(NRF_DRV_USBD_EPOUT0, &transfer);
if (ret == NRF_SUCCESS)
{
app_usbd_core_setup_data_handler_desc_t desc = {
.handler = audio_req_out_data_cb,
.p_context = (void *)p_audio
};
ret = app_usbd_core_setup_data_handler_set(NRF_DRV_USBD_EPOUT0, &desc);
}
CRITICAL_REGION_EXIT();
return ret;
}
void static_generate(STATICSTATE statics, audio_fifo_t *inputfifo, audio_fifo_t *outputfifo, bool comped) {
staticstate_private *state = (staticstate_private *)statics;
audio_fifo_t *af = outputfifo;
int i, o, x;
audio_fifo_data_t *infd = audio_get(inputfifo);
if (infd == NULL)
{
printf("static input ready.\n");
return;
}
// printf("Generating noise; samples=%d, channels=%d\n", infd->nsamples, infd->channels);
audio_fifo_data_t *afd = audio_data_create(infd->nsamples, infd->channels);
o = state->offset;
for(i=0; i<afd->nsamples * afd->channels; i+=2) {
x = infd->samples[i];
if (comped) {
x -= (x * state->vol) / 100000;
x += (state->buf[o] * state->vol) / 60000;
}
else {
x += (state->buf[o] * state->vol) / 32768;
}
if (x<-32767) x=-32767;
if (x>32767) x=32767;
afd->samples[i] = x;
afd->samples[i+1] = x;
o ++;
if (o > state->length) o=0;
if (state->vol < state->targetvol) state->vol++;
if (state->vol > state->targetvol) state->vol--;
}
state->offset = o;
free(infd);
audio_fifo_queue(outputfifo, afd);
}
/**
* @brief Select interface.
*
* @param[in,out] p_inst Instance of the class.
* @param[in] iface_idx Index of the interface inside class structure.
* @param[in] alternate Alternate setting that should be selected.
*/
static ret_code_t iface_select(
app_usbd_class_inst_t const * const p_inst,
uint8_t iface_idx,
uint8_t alternate)
{
app_usbd_class_iface_conf_t const * p_iface = app_usbd_class_iface_get(p_inst, iface_idx);
/* Simple check if this is data interface */
uint8_t const ep_count = app_usbd_class_iface_ep_count_get(p_iface);
if (ep_count > 0)
{
if (alternate > 1)
{
return NRF_ERROR_INVALID_PARAM;
}
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
p_audio_ctx->streaming = (alternate != 0);
uint8_t i;
for (i = 0; i < ep_count; ++i)
{
nrf_drv_usbd_ep_t ep_addr =
app_usbd_class_ep_address_get(app_usbd_class_iface_ep_get(p_iface, i));
if (alternate)
{
app_usbd_ep_enable(ep_addr);
}
else
{
app_usbd_ep_disable(ep_addr);
}
}
return NRF_SUCCESS;
}
return NRF_ERROR_NOT_SUPPORTED;
}
static void* audio_start(void *aux)
{
audio_fifo_t *af = aux;
audio_fifo_data_t *afd;
ALCdevice *device = NULL;
ALCcontext *context = NULL;
ALuint buffers[NUM_BUFFERS];
ALint processed;
ALenum error;
ALint rate;
ALint channels;
device = alcOpenDevice(NULL); /* Use the default device */
if (!device) error_exit("failed to open device");
context = alcCreateContext(device, NULL);
alcMakeContextCurrent(context);
alListenerf(AL_GAIN, 1.0f);
alDistanceModel(AL_NONE);
alGenBuffers((ALsizei)NUM_BUFFERS, buffers);
alGenSources(1, &source);
/* First prebuffer some audio */
queue_buffer(source, af, buffers[0]);
queue_buffer(source, af, buffers[1]);
queue_buffer(source, af, buffers[2]);
for (;;) {
alSourcePlay(source);
for (;;) {
/* Wait for some audio to play */
alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
if (processed <= 0)
{
usleep(200);
continue;
}
/* Remove old audio from the queue.. */
ALuint buffer;
alSourceUnqueueBuffers(source, 1, &buffer);
/* and queue some more audio */
afd = audio_get(af);
alGetBufferi(buffer, AL_FREQUENCY, &rate);
alGetBufferi(buffer, AL_CHANNELS, &channels);
if (afd->rate != rate || afd->channels != channels)
{
log_debug("openal","audio_start","rate or channel count changed, resetting");
break;
}
alBufferData(buffer,
afd->channels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
afd->samples,
afd->nsamples * afd->channels * sizeof(int16_t),
afd->rate);
free(afd);
ALenum error = alGetError();
if (error != AL_NO_ERROR)
{
log_error("openal","audio_start","Error buffering: %s", alGetString(error));
return NULL;
}
alSourceQueueBuffers(source, 1, &buffer);
error = alGetError();
if (alGetError() != AL_NO_ERROR)
{
log_error("openal","audio_start","Error queing buffering: %s", alGetString(error));
return NULL;
}
alGetSourcei(source, AL_SOURCE_STATE, &processed);
if (processed != AL_PLAYING)
{
// Resume playing
alSourcePlay(source);
}
if ((error = alcGetError(device)) != AL_NO_ERROR) {
log_error("openal","audio_start","Error queing buffering: %s", alGetString(error));
exit(1);
}
}
/* Format or rate changed, so we need to reset all buffers */
alSourcei(source, AL_BUFFER, 0);
alSourceStop(source);
/* Make sure we don't lose the audio packet that caused the change */
alBufferData(buffers[0],
afd->channels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
afd->samples,
afd->nsamples * afd->channels * sizeof(short),
afd->rate);
free(afd);
alSourceQueueBuffers(source, 1, &buffers[0]);
queue_buffer(source, af, buffers[1]);
queue_buffer(source, af, buffers[2]);
}
}
static bool audio_feed_descriptors(app_usbd_class_descriptor_ctx_t * p_ctx,
app_usbd_class_inst_t const * p_inst,
uint8_t * p_buff,
size_t max_size)
{
static uint8_t ifaces = 0;
ifaces = app_usbd_class_iface_count_get(p_inst);
ASSERT(ifaces == 2);
app_usbd_audio_t const * p_audio = audio_get(p_inst);
APP_USBD_CLASS_DESCRIPTOR_BEGIN(p_ctx, p_buff, max_size);
/* CONTROL INTERFACE DESCRIPTOR */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x09); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_DESCRIPTOR_INTERFACE); // bDescriptorType = Interface
static app_usbd_class_iface_conf_t const * p_cur_iface = NULL;
p_cur_iface = app_usbd_class_iface_get(p_inst, 0);
APP_USBD_CLASS_DESCRIPTOR_WRITE(app_usbd_class_iface_number_get(p_cur_iface)); // bInterfaceNumber
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bAlternateSetting
APP_USBD_CLASS_DESCRIPTOR_WRITE(app_usbd_class_iface_ep_count_get(p_cur_iface)); // bNumEndpoints
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS); // bInterfaceClass = Audio
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_SUBCLASS_AUDIOCONTROL); // bInterfaceSubclass (Audio Control)
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS_PROTOCOL_UNDEFINED); // bInterfaceProtocol
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // iInterface
/* HEADER INTERFACE */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x09); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_DESCRIPTOR_INTERFACE); // bDescriptorType = Audio Interfaces
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_AC_IFACE_SUBTYPE_HEADER); // bDescriptorSubtype = Header
APP_USBD_CLASS_DESCRIPTOR_WRITE(LSB_16(0x0100)); // bcdADC LSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(MSB_16(0x0100)); // bcdADC MSB
static uint16_t header_desc_len = 0;
header_desc_len = 9 + audio_get_feature_descriptor_size(p_inst) +
audio_get_input_descriptor_size(p_inst) + audio_get_output_descriptor_size(
p_inst);
APP_USBD_CLASS_DESCRIPTOR_WRITE(LSB_16(header_desc_len)); // wTotalLength LSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(MSB_16(header_desc_len)); // wTotalLength MSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x01); // bInCollection
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_control_interface_number(p_inst) + 1); // baInterfaceNr(1)
/* INPUT TERMINAL DESCRIPTOR */
static uint32_t cur_byte = 0;
static size_t input_desc_size = 0;
input_desc_size = audio_get_input_descriptor_size(p_inst);
for (cur_byte = 0; cur_byte < input_desc_size; cur_byte++)
{
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_input_descriptor_data(p_inst, cur_byte));
}
/* FEATURE UNIT DESCRIPTOR */
static size_t feature_desc_size = 0;
feature_desc_size = audio_get_feature_descriptor_size(p_inst);
for (cur_byte = 0; cur_byte < feature_desc_size; cur_byte++)
{
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_feature_descriptor_data(p_inst, cur_byte));
}
/* OUTPUT TERMINAL DESCRIPTOR */
static size_t output_desc_size = 0;
output_desc_size = audio_get_output_descriptor_size(p_inst);
for (cur_byte = 0; cur_byte < output_desc_size; cur_byte++)
{
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_output_descriptor_data(p_inst, cur_byte));
}
p_cur_iface++;
/* STREAM INTERFACE DESCRIPTOR ALT 0 */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x09); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_DESCRIPTOR_INTERFACE); // bDescriptorType = Interface
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_control_interface_number(p_inst) + 1); // bInterfaceNumber
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bAlternateSetting
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bNumEndpoints
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS); // bInterfaceClass = Audio
APP_USBD_CLASS_DESCRIPTOR_WRITE(p_audio->specific.inst.type_streaming); // bInterfaceSubclass (Audio Control)
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS_PROTOCOL_UNDEFINED); // bInterfaceProtocol
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // iInterface
/* STREAM INTERFACE DESCRIPTOR ALT 1 */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x09); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_DESCRIPTOR_INTERFACE); // bDescriptorType = Interface
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_control_interface_number(p_inst) + 1); // bInterfaceNumber
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x01); // bAlternateSetting
APP_USBD_CLASS_DESCRIPTOR_WRITE(app_usbd_class_iface_ep_count_get(p_cur_iface)); // bNumEndpoints
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS); // bInterfaceClass = Audio
APP_USBD_CLASS_DESCRIPTOR_WRITE(p_audio->specific.inst.type_streaming); // bInterfaceSubclass (Audio Control)
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_CLASS_PROTOCOL_UNDEFINED); // bInterfaceProtocol
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // iInterface
/* AudioStreaming GENERAL DESCRIPTOR */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x07); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_DESCRIPTOR_INTERFACE); // bDescriptorType = Audio Interface
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_AS_IFACE_SUBTYPE_GENERAL); // bDescriptorSubtype = General
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_control_interface_number(p_inst) + 1); // bTerminalLink
APP_USBD_CLASS_DESCRIPTOR_WRITE(p_audio->specific.inst.delay); // bDelay
APP_USBD_CLASS_DESCRIPTOR_WRITE(LSB_16(p_audio->specific.inst.format)); // wFormatTag LSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(MSB_16(p_audio->specific.inst.format)); // wFormatTag MSB
/* FORMAT DESCRIPTOR */
static size_t format_desc_size = 0;
format_desc_size = audio_get_format_descriptor_size(p_inst);
for (cur_byte = 0; cur_byte < format_desc_size; cur_byte++)
{
APP_USBD_CLASS_DESCRIPTOR_WRITE(audio_get_format_descriptor_data(p_inst, cur_byte));
}
/* ENDPOINT GENERAL DESCRIPTOR */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x07); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_DESCRIPTOR_ENDPOINT); // bDescriptorType = Audio Descriptor
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_AUDIO_EP_SUBTYPE_GENERAL); // bDescriptorSubtype = EP General
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bmAttributes
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bLockDelayUnits
APP_USBD_CLASS_DESCRIPTOR_WRITE(LSB_16(0x0000)); // wLockDelay LSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(MSB_16(0x0000)); // wLockDelay MSB
/* ENDPOINT ISO DESCRIPTOR */
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x09); // bLength
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_DESCRIPTOR_ENDPOINT); // bDescriptorType = Endpoint
static app_usbd_class_ep_conf_t const * p_cur_ep = NULL;
p_cur_ep = app_usbd_class_iface_ep_get(p_cur_iface, 0);
APP_USBD_CLASS_DESCRIPTOR_WRITE(app_usbd_class_ep_address_get(p_cur_ep)); // bEndpointAddress
APP_USBD_CLASS_DESCRIPTOR_WRITE(APP_USBD_DESCRIPTOR_EP_ATTR_TYPE_ISOCHRONOUS); // bmAttributes
APP_USBD_CLASS_DESCRIPTOR_WRITE(LSB_16(p_audio->specific.inst.ep_size)); // wMaxPacketSize LSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(MSB_16(p_audio->specific.inst.ep_size)); // wMaxPacketSize MSB
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x01); // bInterval
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bRefresh
APP_USBD_CLASS_DESCRIPTOR_WRITE(0x00); // bSynchAddress
APP_USBD_CLASS_DESCRIPTOR_END();
}
int process_get(request * req)
{
int bytes_written;
volatile int bytes_to_write;
bytes_to_write = req->filesize - req->filepos;
if (bytes_to_write > SOCKETBUF_SIZE)
bytes_to_write = SOCKETBUF_SIZE;
if (sigsetjmp(env, 1) == 0) {
handle_sigbus = 1;
#ifdef SERVER_SSL
if(req->ssl == NULL){
#endif /*SERVER_SSL*/
bytes_written = write(req->fd, req->data_mem + req->filepos,
bytes_to_write);
#ifdef SERVER_SSL
}else{
bytes_written = SSL_write(req->ssl, req->data_mem + req->filepos, bytes_to_write);
#if 0
printf("SSL_write\n");
#endif /*0*/
}
#endif /*SERVER_SSL*/
handle_sigbus = 0;
/* OK, SIGBUS **after** this point is very bad! */
} else {
/* sigbus! */
log_error_doc(req);
/* sending an error here is inappropriate
* if we are here, the file is mmapped, and thus,
* a content-length has been sent. If we send fewer bytes
* the client knows there has been a problem.
* We run the risk of accidentally sending the right number
* of bytes (or a few too many) and the client
* won't be the wiser.
*/
req->status = DEAD;
fprintf(stderr, "%sGot SIGBUS in write(2)!\n", get_commonlog_time());
return 0;
}
if (bytes_written < 0) {
if (errno == EWOULDBLOCK || errno == EAGAIN)
return -1;
/* request blocked at the pipe level, but keep going */
else {
if (errno != EPIPE) {
log_error_doc(req);
/* Can generate lots of log entries, */
perror("write");
/* OK to disable if your logs get too big */
}
req->status = DEAD;
return 0;
}
}
req->filepos += bytes_written;
req->busy_flag = BUSY_FLAG_AUDIO|BUSY_FLAG_VIDEO;
if (req->filepos == req->filesize) { /* EOF */
#ifdef DAVINCI_IPCAM
if (req->http_stream == URI_STREAM_MJPEG) {
#if 1
#if 0
while (audio_get(req) > 0);
if (req->audio_length >= AUDIO_SEND_SIZE) {
audio_send(req);
return 1;
}
#else
if (audio_get(req, FMT_MJPEG) > 0)
return 1;
#endif
#else
req->busy_flag &= ~BUSY_FLAG_AUDIO;
#endif
if (req->serial_lock) {
GetAVData(AV_OP_UNLOCK_MJPEG, req->serial_lock, NULL);
req->serial_lock = 0;
}
GetAVData(AV_OP_GET_MJPEG_SERIAL, -1, &req->av_data);
if (req->av_data.serial < req->serial_book) {
req->busy_flag &= ~BUSY_FLAG_VIDEO;
return 1;
}
GetAVData(AV_OP_LOCK_MJPEG, req->av_data.serial, &req->av_data );
req->data_mem = req->av_data.ptr;
req->filesize = req->av_data.size+16;
req->filepos = 0;
req->serial_lock = req->av_data.serial;
req->serial_book = req->av_data.serial+1;
reset_output_buffer(req);
req_write(req, "\r\n");
print_mjpeg_headers(req);
return 1;
}
if (req->http_stream == URI_STREAM_MPEG4 || req->http_stream == URI_STREAM_AVC) {
int ret;
#if 1
#if 0
while (audio_get(req) > 0);
if (req->audio_length > AUDIO_SEND_SIZE) {
req->busy_flag |= BUSY_FLAG_AUDIO;
audio_send(req);
return 1;
}
#else
if (audio_get(req, FMT_MPEG4) > 0)
return 1;
#endif
#else
req->busy_flag &= ~BUSY_FLAG_AUDIO;
#endif
ret = GetAVData(AV_OP_LOCK_MP4, req->serial_book, &req->av_data);
if (ret == RET_SUCCESS) {
GetAVData(AV_OP_UNLOCK_MP4, req->serial_lock, NULL);
req->data_mem = req->av_data.ptr;
req->filesize = req->av_data.size+16;
req->filepos = 0;
req->serial_lock = req->av_data.serial;
req->serial_book = req->av_data.serial+1;
reset_output_buffer(req);
req_write(req, "\r\n");
if (req->http_stream == URI_STREAM_AVC)
{ print_avc_headers(req); }
else
{ print_mpeg4_headers(req); }
return 1;
}
else if (ret == RET_NO_VALID_DATA) {
req->busy_flag &= ~BUSY_FLAG_VIDEO;
return 1;
}
else {
GetAVData(AV_OP_GET_MPEG4_SERIAL, -1, &req->av_data );
req->serial_book = req->av_data.serial;
dbg("lock error ret=%d\n", ret);
return 1;
}
}
if (req->http_stream == URI_STREAM_MPEG4CIF || req->http_stream == URI_STREAM_AVCCIF) {
int ret;
#if 1
#if 0
while (audio_get(req) > 0);
if (req->audio_length > AUDIO_SEND_SIZE) {
audio_send(req);
return 1;
}
#else
if (audio_get(req, FMT_MPEG4_EXT) > 0)
return 1;
#endif
#else
req->busy_flag &= ~BUSY_FLAG_AUDIO;
#endif
ret = GetAVData(AV_OP_LOCK_MP4_CIF, req->serial_book, &req->av_data);
if (ret == RET_SUCCESS) {
GetAVData(AV_OP_UNLOCK_MP4_CIF, req->serial_lock, NULL);
req->data_mem = req->av_data.ptr;
req->filesize = req->av_data.size+16;
req->filepos = 0;
req->serial_lock = req->av_data.serial;
req->serial_book = req->av_data.serial+1;
reset_output_buffer(req);
req_write(req, "\r\n");
if (req->http_stream == URI_STREAM_AVCCIF)
{ print_avc_headers(req); }
else
{ print_mpeg4_headers(req); }
return 1;
}
else if (ret == RET_NO_VALID_DATA) {
req->busy_flag &= ~BUSY_FLAG_VIDEO;
return 1;
}
else {
GetAVData(AV_OP_GET_MPEG4_CIF_SERIAL, -1, &req->av_data );
req->serial_book = req->av_data.serial;
dbg("lock error ret=%d\n", ret);
return 1;
}
}
#endif // DAVINCI_IPCAM
return 0;
} else
return 1; /* more to do */
}
static void* audio_start(void *aux)
{
audio_fifo_t *af = aux;
audio_fifo_data_t *afd;
unsigned int frame = 0;
ALCdevice *device = NULL;
ALCcontext *context = NULL;
ALuint buffers[NUM_BUFFERS];
ALuint source;
ALint processed;
ALenum error;
ALint rate;
ALint channels;
device = alcOpenDevice(NULL); /* Use the default device */
if (!device) error_exit("failed to open device");
context = alcCreateContext(device, NULL);
alcMakeContextCurrent(context);
alListenerf(AL_GAIN, 1.0f);
alDistanceModel(AL_NONE);
alGenBuffers((ALsizei)NUM_BUFFERS, buffers);
alGenSources(1, &source);
/* First prebuffer some audio */
queue_buffer(source, af, buffers[0]);
queue_buffer(source, af, buffers[1]);
queue_buffer(source, af, buffers[2]);
for (;;) {
alSourcePlay(source);
for (;;) {
/* Wait for some audio to play */
do {
alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
usleep(100);
} while (!processed);
/* Remove old audio from the queue.. */
alSourceUnqueueBuffers(source, 1, &buffers[frame % 3]);
/* and queue some more audio */
afd = audio_get(af);
alGetBufferi(buffers[frame % 3], AL_FREQUENCY, &rate);
alGetBufferi(buffers[frame % 3], AL_CHANNELS, &channels);
if (afd->rate != rate || afd->channels != channels) {
printf("rate or channel count changed, resetting\n");
free(afd);
break;
}
alBufferData(buffers[frame % 3],
afd->channels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
afd->samples,
afd->nsamples * afd->channels * sizeof(short),
afd->rate);
free(afd);
alSourceQueueBuffers(source, 1, &buffers[frame % 3]);
if ((error = alcGetError(device)) != AL_NO_ERROR) {
printf("openal al error: %d\n", error);
exit(1);
}
frame++;
}
/* Format or rate changed, so we need to reset all buffers */
alSourcei(source, AL_BUFFER, 0);
alSourceStop(source);
/* Make sure we don't lose the audio packet that caused the change */
alBufferData(buffers[0],
afd->channels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
afd->samples,
afd->nsamples * afd->channels * sizeof(short),
afd->rate);
alSourceQueueBuffers(source, 1, &buffers[0]);
queue_buffer(source, af, buffers[1]);
queue_buffer(source, af, buffers[2]);
frame = 0;
}
}
/**
* @brief @ref app_usbd_class_methods_t::event_handler
*/
static ret_code_t audio_event_handler(
app_usbd_class_inst_t const * p_inst,
app_usbd_complex_evt_t const * p_event)
{
ASSERT(p_inst != NULL);
ASSERT(p_event != NULL);
ret_code_t ret = NRF_SUCCESS;
switch (p_event->app_evt.type)
{
case APP_USBD_EVT_DRV_RESET:
break;
case APP_USBD_EVT_DRV_SETUP:
ret = setup_event_handler(p_inst, (app_usbd_setup_evt_t const *)p_event);
break;
case APP_USBD_EVT_DRV_EPTRANSFER:
if (NRF_USBD_EPIN_CHECK(p_event->drv_evt.data.eptransfer.ep))
{
ret = endpoint_in_event_handler(p_inst);
}
else
{
ret = endpoint_out_event_handler(p_inst);
}
break;
case APP_USBD_EVT_DRV_SUSPEND:
break;
case APP_USBD_EVT_DRV_RESUME:
break;
case APP_USBD_EVT_INST_APPEND:
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
if(p_audio_ctx->sof_handler != NULL)
{
ret = app_usbd_class_sof_interrupt_register(p_inst, p_audio_ctx->sof_handler);
APP_ERROR_CHECK(ret);
}
break;
}
case APP_USBD_EVT_INST_REMOVE:
{
app_usbd_audio_t const * p_audio = audio_get(p_inst);
app_usbd_audio_ctx_t * p_audio_ctx = audio_ctx_get(p_audio);
if(p_audio_ctx->sof_handler != NULL)
{
ret = app_usbd_class_sof_interrupt_unregister(p_inst);
APP_ERROR_CHECK(ret);
}
break;
}
case APP_USBD_EVT_STARTED:
break;
case APP_USBD_EVT_STOPPED:
break;
default:
ret = NRF_ERROR_NOT_SUPPORTED;
break;
}
return ret;
}
