I am getting audio from a microphone with PortAudio (PA). I need to then resample this audio to 44,100KHz. I'm attempting to do this with FFmpeg. Currently, the mic I'm testing with has a sample rate of 48,000KHz, but this won't always be the case when the application is used. Anyway, whenever I attempt to resample with swr_convert, I get a segmentation fault. I am initializing the SwrContext with
this->swr_ctx = swr_alloc_set_opts(
nullptr, // No current context
num_channels, // The number of channls I'm getting from PA
AV_SAMPLE_FMT_S16, // 16 bit Signed, should correspond to paInt16
FINAL_SAMPLE_RATE, // 44100
num_channels, // The number of channls I'm getting from PA
AV_SAMPLE_FMT_S16, // 16 bit Signed, should correspond to paInt16
this->source_sample_rate, // Mic I'm testing with currently is 44800, but depends on source
0, // Logging offset (0 is what examples use, so I did too)
nullptr // "parent logging context, can be NULL"
);
I know PA is working right, as the project works if I hard-code the sample rate in other aspects of this project. The callback looks like this
auto paCallback( const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData ) -> int {
// Calls class's callback handler
return ((Audio*)userData)->classPaCallback((const uint8_t **)inputBuffer);
}
// Class callback handler
auto Audio::classPaCallback(const uint8_t **inputBuffer) -> int {
// This line throws SIGSEGV
int out_count = swr_convert(swr_ctx, this->resample_buffer, BUFFER_CHUNK_SIZE, inputBuffer, this->source_buffer_size);
if (out_count < 0)
throw std::runtime_error("Error resampling audio");
}
// Add data to buffers to handle outside of callback context (This is a special context according to PA docs)
return 0;
}
Playing around with the swr_convert line, I changed the out_count and in_count parameters (BUFFER_CHUNK_SIZE and this->source_buffer_size) to 0 to make sure that the code would at least run, that worked. I then changed one of them to 1, and left the other at 0, to test which buffer access was throwing the SIGSEGV, and it was thrown when the in_count (buffer from PA) was not 0. What am I doing wrong when passing the audio from PA to FFMpeg?
I do know that the PA audio is "interleaved" (I.E. input[0] is the first sample from channel 0, input[1] is the first sample from channel 1, ect.). Is this also the format that FFMpeg uses, or should I create a different SwrContext for each channel?
In case this is helpful, this->resample_buffer is successfully initialized with
av_samples_alloc_array_and_samples(
&this->resample_buffer, // Buffer
nullptr, // "linesize", not used from what I could tell
this->num_channels, // number of channels expected
BUFFER_CHUNK_SIZE, // Number of frames to be stored per channel
AV_SAMPLE_FMT_S16, // 16 bit Signed, should correspond to paInt16
0 // For alignment, not needed here
);
Since resampling directly after receiving the data from PortAudio (PA) was not working, and no one said anything right away, I knew I wasn't making any "dumb" mistakes. After a bit of a break, I decided to try again, but instead of resampling right away, first adding it to the appropriate buffer (based on which channel it was from PA). Then, when removing the data from the buffer, I applied the resample. Doing it this way allowed me to do resampling on one channel, therefore not having to wonder how to include multiple channels for FFMpeg. For anyone who comes across this in the future, the code ended up looking like this
auto *converted_sample = (uint16_t *) malloc(sizeof(uint16_t) * this->converted_sample_max);
if (converted_sample == nullptr) {
throw std::runtime_error("Failed to allocate memory for converted sample");
}
uint16_t *sample = buffer.pop(); // Gets data from circular buffer
if (sample == nullptr) {
free(converted_sample); // IMPORTANT to not incur a memory leak
return return_data{nullptr, 0};
}
if (swr_ctx == nullptr) {
free(converted_sample); // IMPORTANT to not incur a memory leak
throw std::runtime_error("swr_ctx is not initialized");
}
int frames = swr_convert(swr_ctx, (uint8_t **)&converted_sample, this->converted_sample_max, (const uint8_t **)&sample, BUFFER_CHUNK_SIZE);
free(sample); // In my case, it is the job of whoever takes the data out of the circular buffer to free() it
if (frames < 0) {
free(converted_sample); // Prevent memory leak
throw std::runtime_error("no frames converted");
}
return return_data{converted_sample, frames}; // A struct I made
}
this->converted_sample_max is initialized with
this->converted_sample_max = av_rescale_rnd(BUFFER_CHUNK_SIZE, FINAL_SAMPLE_RATE, source_sample_rate, AV_ROUND_UP);
Related
I created an application a couple of years ago that allowed me to process audio by downmixing a 6 channel or 8 channel a.k.a 5.1 as 7.1 as matrixed stereo encoded for that purpose I used the portaudio library with great results this is an example of the open stream function and callback to downmix a 7.1 signal
Pa_OpenStream(&Flujo, &inputParameters, &outParameters, SAMPLE_RATE, 1, paClipOff, ptrFunction, NULL);
Notice the use of framesPerBuffer value of just one (1), this is my callback function
int downmixed8channels(const void *input, void *output, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo * info, PaStreamCallbackFlags state, void * userData)
{
(void)userData;
(void)info;
(void)state;
(void)framesBuffer;
float *ptrInput = (float*)input;
float *ptrOutput = (float*)ouput;
/*This is a struct to identify samples*/
AudioSamples->L = ptrInput[0];
AudioSamples->R = ptrInput[1];
AudioSamples->C = ptrInput[2];
AudioSamples->LFE = ptrInput[3];
AudioSamples->RL = ptrInput[4];
AudioSamples->RR = ptrInput[5];
AudioSamples->SL = ptrInput[6];
AudioSamples->SR = ptrInput[7];
Encoder->8channels(AudioSamples->L,
AudioSamples->R,
AudioSamples->C,
AudioSamples->LFE,
MuestrasdeAudio->SL,
MuestrasdeAudio->SR,
MuestrasdeAudio->RL,
MuestrasdeAudio->RR,);
ptrOutput[0] = Encoder->gtLT();
ptrOutput[1] = Encoder->gtRT();
return paContinue;
}
As you can see the order set by the index in the output and input buffer correspond to a discrete channel
in the case of the output 0 = Left channel, 1 = right Channel. This used to work well, until entered Windows 10 2004, since I updated my system to this new version my audio glitch and I get artifacts like those
Those are captures from the sound of the channel test window under the audio device panel of windows. By the images is clear my program is dropping frames, so the first try to solve this was to use a larger buffer than one to hold samples process them and send then, the reason I did not use a buffer size larger than one in the first place was that the program would drop frames.
But before implementing a I did a proof of concept, would not include audio processing at all, of simple passing of data from input to ouput, for that I set the oputput channelCount parameters to 8 just like the input, resulting in something as simple as this.
for (int i = 0; i < FramesPerBuffer /*1000*/; i++)
{
ptrOutput[i] = ptrOutput[i];
}
but still the program is still dropping samples.
Next I used two callbacks one for writing to a buffer and a second one to read it and send it to output
(void)info;
(void)userData;
(void)state;
(void)output;
float* ptrInput = (float*)input;
for (int i = 0; i < FRAME_SIZE; i++)
{
buffer_input[i] = ptrInput[i];
}
return paContinue;
Callback to store.
(void)info;
(void)userData;
(void)state;
(void)output;
float* ptrOutput = (float*)output;
for (int i = 0; i < FRAME_SIZE; i++)
{
AudioSamples->L = (buffer_input[i] );
AudioSamples->R = (buffer_input[i++]);
AudioSamples->C = (buffer_input[i++] );
AudioSamples->LFE = (buffer_input[i++]);
AudioSamples->SL = (buffer_input[i++] );
AudioSamples->SR = (buffer_input[i++]);
Encoder->Encoder(AudioSamples->L, AudioSamples->R, AudioSamples->C, AudioSamples->LFE,
AudioSamples->SL, AudioSamples->SR);
bufferTransformed[w] = (Encoder->getLT() );
bufferTransformed[w++] = (Encoder->getRT() );
}
w = 0;
for (int i = 0; i < FRAME_REDUCED; i++)
{
ptrOutput[i] = buffer_Transformed[i];
}
return paContinue;
Callback for processing
The processing callback use a reduced frames per buffer since 2 channel is less than eight since it seems in portaudio a frame is composed of a sample for each audio channel.
This also did not work, since the first problem, is how to syncronize the two callback?, after all of this, what recommendation or advice, can you give me to solve this issue,
Notes: the samplerate must be same for both devices, I implemeted logic in the program to prevent this, the bitdepth is also the same I am using paFloat32,
.The portaudio is the modified one use by audacity, since I wanted to use their implementation of WASAPI
loopback
Thank very much in advance!.
At the end of the day it I did not have to change my callbacks functions in any way, what solved it, was changing or increasing the parameter ".suggestedLatency" of the input and output parameters, to 1.0, even the devices defaultLowOutputLatency or defaultHighOutputLatency values where causing to much glitching, I test it until 1.0 was de sweepspot, higher values did not seen to improve.
TL;DR Increased the suggestedLatency until the glitching is gone.
I am saving settings to the flash memory and reading them back again. 2 of the values always comes back empty. However, the data IS written to flash, since after a reset the values read are the new saved values and not empty.
I started experiencing this problem after I did some code-refactoring after taking the code over from another company.
Saving and reading the settings back works when you actually do the following (old inefficient way):
save setting 0 - read setting 0
save setting 1 - read setting 1
...
save setting 13 read setting 13
This is EXTREMELY inefficient and slow since the same page with all the settings are read from flash, the whole block of flash cleared, the new setting put into the read buffer and then the whole block (with only 1 changed setting) are written to flash. And this happens for all 14 settings!! But it works ...
unsigned char Save_One_Setting(unsigned char Setting_Number, unsigned char* value, unsigned char length)
{
/* Program the user Flash area word by word
(area defined by FLASH_USER_START_ADDR and FLASH_USER_END_ADDR) ***********/
unsigned int a;
Address = FLASH_USER_START_ADDR;
a = 0;
while (Address < FLASH_USER_END_ADDR)
{
buf[a++] = *(__IO uint32_t *)Address;
Address = Address + 4;
}
memset(&buf[Setting_Number * 60], 0, 60); // Clear setting value
memcpy(&buf[Setting_Number * 60], &value[0], length); // Set setting value
Erase_User_Flash_Memory();
HAL_FLASH_Unlock();
Address = FLASH_USER_START_ADDR;
a = 0;
while (Address < FLASH_USER_END_ADDR)
{
if (HAL_FLASH_Program(FLASH_TYPEPROGRAM_WORD, Address, buf[a++]) == HAL_OK)
{
Address = Address + 4;
}
else
{
/* Error occurred while writing data in Flash memory.
User can add here some code to deal with this error */
while (1)
{
/* Make LED1 blink (100ms on, 2s off) to indicate error in Write operation */
BSP_LED_On(LED1);
HAL_Delay(100);
BSP_LED_Off(LED1);
HAL_Delay(2000);
}
}
}
/* Lock the Flash to disable the flash control register access (recommended
to protect the FLASH memory against possible unwanted operation) *********/
HAL_FLASH_Lock();
}
I changed this by actually, after reading the settings from the flash into a buffer, update all the changed settings in the buffer, then erase the flash block and write the buffer back to flash. Downside: my first and 4th values always comes back as NULL after saving this buffer to flash.
However, after a system reset the correct values are read from flash.
unsigned char Save_Settings(Save_Settings_struct* newSettings)
{
/* Program the user Flash area word by word
(area defined by FLASH_USER_START_ADDR and FLASH_USER_END_ADDR) ***********/
unsigned int a;
unsigned char readBack[60];
Address = FLASH_USER_START_ADDR;
a = 0;
while (Address < FLASH_USER_END_ADDR)
{
buf[a++] = *(__IO uint32_t *)Address;
Address = Address + 4;
}
a = 0;
while (a < S_MAXSETTING)
{
if (newSettings[a].settingNumber < S_MAXSETTING)
{
memset(&buf[a * 60], 0, 60); // Clear setting value
memcpy(&buf[a * 60], newSettings[a].settingValue, newSettings[a].settingLength); // Set setting value
}
++a;
}
Erase_User_Flash_Memory();
HAL_FLASH_Unlock();
Address = FLASH_USER_START_ADDR;
a = 0;
while (Address < FLASH_USER_END_ADDR)
{
if (HAL_FLASH_Program(FLASH_TYPEPROGRAM_WORD, Address, buf[a++]) == HAL_OK)
{
Address = Address + 4;
}
else
{
/* Error occurred while writing data in Flash memory.
User can add here some code to deal with this error */
while (1)
{
/* Make LED1 blink (100ms on, 2s off) to indicate error in Write operation */
BSP_LED_On(LED1);
HAL_Delay(100);
BSP_LED_Off(LED1);
HAL_Delay(2000);
}
}
}
/* Lock the Flash to disable the flash control register access (recommended
to protect the FLASH memory against possible unwanted operation) *********/
HAL_FLASH_Lock();
}
I started playing around with cleaning and invalidating the data cache. At least the 2 values are not NULL anymore, however, they are still the old values. All other values are the new, saved values. Do a reset, and all values are correct.
Anybody ever had some similar problem? Or maybe an idea of what I can try to get rid of this problem?
I'm trying to write a image via UART on Beaglebone Black. But when I use the write() function in library .
int write(int handle, void *buffer, int nbyte);
Regardless of the agurment nbyte has int type, I can not transfer 70kB at once. I displayed the number of byte which is transfered, and the result is the number of byte = 4111.
length = write(fd,body.c_str(),strlen(body.c_str())); //
cout<<length<<endl; // result length = 4111;
cout<<strlen(body.c_str())<<endl; // result strlen(body.c_str()) = 72255;
I hope to hear from you!
The write call does not assure that you can write the amount of data supplied, that's why it as an integer as its return, not a Boolean. The behavior you see is actually common among different operating systems, it may be due to the underline device might does not have sufficient buffer or storage for you to write 70kb. What you need is to write in a loop, each write will write the amount that is left unwritten:
int total = body.length(); // or strlen(body.c_str())
char *buffer = body.c_str();
int written = 0;
int ret;
while (written < total) {
ret = write(fd, buffer + written, total - written);
if (ret < 0) {
// error
break;
}
written += ret;
}
I'm trying for several months to figure out how it works. I have a program that I'm developing, I have an mp3 file in and out I have the pcm that goes to "alsa" for playback. Using the library mpg123 where the main code is this:
while (mpg123_read (mh, buffer, buffer_size, & done) == MPG123_OK)
sendoutput (dev, buffer, done);
Now, my attempts have been based on the use of library avutil/avcodec on the buffer for reducing/increase the number of samples in one second. The result is awful and isn't audibly. In a previous question someone advised me to increase my PC performance but if a simple program like "VLC" can do this on old computers why I can't?
And for the problem of position in the audio file how can I achieve this?
Edit
I Add some piece of code to try to explain.
SampleConversion.c
#define LENGTH_MS 1000 // how many milliseconds of speech to store 0,5s:x=1:44100 x=22050 sample da memorizzare
#define RATE 44100 // the sampling rate (input)
struct AVResampleContext* audio_cntx = 0;
//(LENGTH_MS*RATE*16*CHANNELS)/8000
void inizializeResample(int inRate, int outRate)
{
audio_cntx = av_resample_init( outRate, //out rate
inRate, //in rate
16, //filter length
10, //phase count
0, //linear FIR filter
0.8 ); //cutoff frequency
assert( audio_cntx && "Failed to create resampling context!");
}
void resample(char dataIn[],char dataOut[],int nsamples)
{
int samples_consumed;
int samples_output = av_resample( audio_cntx, //resample context
(short*)dataOut, //buffout
(short*)dataIn, //buffin
&samples_consumed, //&consumed
nsamples, //nb_samples
sizeof(dataOut)/2,//lenout sizeof(out_buffer)/2 (Right?)
0);//is_last
assert( samples_output > 0 && "Error calling av_resample()!" );
}
void endResample()
{
av_resample_close( audio_cntx );
}
My edited play function (Mpg123.c)
if (isPaused==0 && mpg123_read(mh, buffer, buffer_size, &done) == MPG123_OK)
{
int i=0; char * resBuffer=malloc(sizeof(buffer));
//resBuffer=&buffer[0];
resample(buffer,resBuffer,44100);
if((ao_play(dev, (char*)resBuffer, done)==0)){
return 1;
}
}
Both codes are made by me so I can not ask anybody ever suggested improvements as in the previous question (although I do not know if they are right, sigh)
Edit2: Updated with changes
In the call to av_resample, samples_consumed is never read, so any unconsumed frames are skipped.
Furthermore, nsamples is the constant value 44100 instead of the actual number of frames read (done from mpg123_read).
sizeof(dataOut) is wrong; it's the size of a pointer.
is_last is wrong at the end of the input.
In the play function, sizeof(buffer) is likely to be wrong, depending on the definition of buffer.
I'm currently working on replacing a blocking busy-wait implementation of an SD card driver over SSP with a non-blocking DMA implementation. However, there are no bytes actually written, even though everything seems to go according to plan (no error conditions are ever found).
First some code (C++):
(Disclaimer: I'm still a beginner in embedded programming so code is probably subpar)
namespace SD {
bool initialize() {
//Setup SSP and detect SD card
//... (removed since not relevant for question)
//Setup DMA
LPC_SC->PCONP |= (1UL << 29);
LPC_GPDMA->Config = 0x01;
//Enable DMA interrupts
NVIC_EnableIRQ(DMA_IRQn);
NVIC_SetPriority(DMA_IRQn, 4);
//enable SSP interrupts
NVIC_EnableIRQ(SSP2_IRQn);
NVIC_SetPriority(SSP2_IRQn, 4);
}
bool write (size_t block, uint8_t const * data, size_t blocks) {
//TODO: support more than one block
ASSERT(blocks == 1);
printf("Request sd semaphore (write)\n");
sd_semaphore.take();
printf("Writing to block " ANSI_BLUE "%d" ANSI_RESET "\n", block);
memcpy(SD::write_buffer, data, BLOCKSIZE);
//Start the write
uint8_t argument[4];
reset_argument(argument);
pack_argument(argument, block);
if (!send_command(CMD::WRITE_BLOCK, CMD_RESPONSE_SIZE::WRITE_BLOCK, response, argument)){
return fail();
}
assert_cs();
//needs 8 clock cycles
delay8(1);
//reset pending interrupts
LPC_GPDMA->IntTCClear = 0x01 << SD_DMACH_NR;
LPC_GPDMA->IntErrClr = 0x01 << SD_DMACH_NR;
LPC_GPDMA->SoftSReq = SD_DMA_REQUEST_LINES;
//Prepare channel
SD_DMACH->CSrcAddr = (uint32_t)SD::write_buffer;
SD_DMACH->CDestAddr = (uint32_t)&SD_SSP->DR;
SD_DMACH->CLLI = 0;
SD_DMACH->CControl = (uint32_t)BLOCKSIZE
| 0x01 << 26 //source increment
| 0x01 << 31; //Terminal count interrupt
SD_SSP->DMACR = 0x02; //Enable ssp write dma
SD_DMACH->CConfig = 0x1 //enable
| SD_DMA_DEST_PERIPHERAL << 6
| 0x1 << 11 //mem to peripheral
| 0x1 << 14 //enable error interrupt
| 0x1 << 15; //enable terminal count interrupt
return true;
}
}
extern "C" __attribute__ ((interrupt)) void DMA_IRQHandler(void) {
printf("dma irq\n");
uint8_t channelBit = 1 << SD_DMACH_NR;
if (LPC_GPDMA->IntStat & channelBit) {
if (LPC_GPDMA->IntTCStat & channelBit) {
printf(ANSI_GREEN "terminal count interrupt\n" ANSI_RESET);
LPC_GPDMA->IntTCClear = channelBit;
}
if (LPC_GPDMA->IntErrStat & channelBit) {
printf(ANSI_RED "error interrupt\n" ANSI_RESET);
LPC_GPDMA->IntErrClr = channelBit;
}
SD_DMACH->CConfig = 0;
SD_SSP->IMSC = (1 << 3);
}
}
extern "C" __attribute__ ((interrupt)) void SSP2_IRQHandler(void) {
if (SD_SSP->MIS & (1 << 3)) {
SD_SSP->IMSC &= ~(1 << 3);
printf("waiting until idle\n");
while(SD_SSP->SR & (1UL << 4));
//Stop transfer token
//I'm not sure if the part below up until deassert_cs is necessary.
//Adding or removing it made no difference.
SPI::send(0xFD);
{
uint8_t response;
unsigned int timeout = 4096;
do {
response = SPI::receive();
} while(response != 0x00 && --timeout);
if (timeout == 0){
deassert_cs();
printf("fail");
return;
}
}
//Now wait until the device isn't busy anymore
{
uint8_t response;
unsigned int timeout = 4096;
do {
response = SPI::receive();
} while(response != 0xFF && --timeout);
if (timeout == 0){
deassert_cs();
printf("fail");
return;
}
}
deassert_cs();
printf("idle\n");
SD::sd_semaphore.give_from_isr();
}
}
A few remarks about the code and setup:
Written for the lpc4088 with FreeRTOS
All SD_xxx defines are conditional defines to select the right pins (I need to use SSP2 in my dev setup, SSP0 for the final product)
All external function that are not defined in this snippet (e.g. pack_argument, send_command, semaphore.take() etc.) are known to be working correctly (most of these come from the working busy-wait SD implementation. I can't of course guarantee 100% that they are bugless, but they seem to be working right.).
Since I'm in the process of debugging this there are a lot of printfs and hardcoded SSP2 variables. These are of course temporarily.
I mostly used this as example code.
Now I have already tried the following things:
Write without DMA using busy-wait over SSP. As mentioned before I started with a working implementation of this, so I know the problem has to be in the DMA implementation and not somewhere else.
Write from mem->mem instead of mem->sd to eliminate the SSP peripheral. mem->mem worked fine, so the problem must be in the SSP part of the DMA setup.
Checked if the ISRs are called. They are: first the DMA IRS is called for the terminal count interrupt, and then the SSP2 IRS is called. So the IRSs are (probably) setup correctly.
Made a binary dump of the entire sd content to see if it the content might have been written to the wrong location. Result: the content send over DMA was not present anywhere on the SD card (I did this with any change I made to the code. None of it got the data on the SD card).
Added a long (~1-2 seconds) timeout in the SSP IRS by repeatedly requesting bytes from the SD card to make sure that there wasn't a timeout issue (e.g. that I tried to read the bytes before the SD card had the chance to process everything). This didn't change the outcome at all.
Unfortunately due to lack of hardware tools I haven't been able yet to verify if the bytes are actually send over the data lines.
What is wrong with my code, or where can I look to find the cause of this problem? After spending way more hours on this then I'd like to admit I really have no idea how to get this working and any help is appreciated!
UPDATE: I did a lot more testing, and thus I got some more results. The results below I got by writing 4 blocks of 512 bytes. Each block contains constantly increasing numbers module 256. Thus each block contains 2 sequences going from 0 to 255. Results:
Data is actually written to the SD card. However, it seems that the first block written is lost. I suppose there is some setup done in the write function that needs to be done earlier.
The bytes are put in a very weird (and wrong) order: I basically get alternating all even numbers followed by all odd numbers. Thus I first get even numbers 0x00 - 0xFE and then all odd numbers 0x01 - 0xFF (total number of written bytes seems to be correct, with the exception of the missing first block). However, there's even one exception in this sequence: each block contains 2 of these sequences (sequence is 256 bytes, block is 512), but the first sequence in each block has 0xfe and 0xff "swapped". That is, 0xFF is the end of the even numbers and 0xFE is the end of the odd series. I have no idea what kind of black magic is going on here. Just in case I've done something dumb here's the snippet that writes the bytes:
uint8_t block[512];
for (int i = 0; i < 512; i++) {
block[i] = (uint8_t)(i % 256);
}
if (!SD::write(10240, block, 1)) { //this one isn't actually written
WARN("noWrite", proc);
}
if (!SD::write(10241, block, 1)) {
WARN("noWrite", proc);
}
if (!SD::write(10242, block, 1)) {
WARN("noWrite", proc);
}
if (!SD::write(10243, block, 1)) {
WARN("noWrite", proc);
}
And here is the raw binary dump. Note that this exact pattern is fully reproducible: so far each time I tried this I got this exact same pattern.
Update2: Not sure if it's relevant, but I use sdram for memory.
When I finally got my hands on a logic analyzer I got a lot more information and was able to solve these problems.
There were a few small bugs in my code, but the bug that caused this behaviour was that I didn't send the "start block" token (0xFE) before the block and I didn't send the 16 bit (dummy) crc after the block. When I added these to the transfer buffer everything was written successfully!
So this fix was as followed:
bool write (size_t block, uint8_t const * data, size_t blocks) {
//TODO: support more than one block
ASSERT(blocks == 1);
printf("Request sd semaphore (write)\n");
sd_semaphore.take();
printf("Writing to block " ANSI_BLUE "%d" ANSI_RESET "\n", block);
SD::write_buffer[0] = 0xFE; //start block
memcpy(&SD::write_buffer[1], data, BLOCKSIZE);
SD::write_buffer[BLOCKSIZE + 1] = 0; //dummy crc
SD::write_buffer[BLOCKSIZE + 2] = 0;
//...
}
As a side note, the reason why the first block wasn't written was simply because I didn't wait until the device was ready before sending the first block. Doing so fixed the problem.