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I'm trying to write 4 uint32's of data into the flash memory of my STM32F767ZI so I've looked at some examples and in the reference manual but still I cannot do it. My goal is to write 4 uint32's into the flash and read them back and compare with the original data, and light different leds depending on the success of the comparison.
My code is as follows:
void flash_write(uint32_t offset, uint32_t *data, uint32_t size) {
FLASH_EraseInitTypeDef EraseInitStruct = {0};
uint32_t SectorError = 0;
HAL_FLASH_Unlock();
EraseInitStruct.TypeErase = FLASH_TYPEERASE_SECTORS;
EraseInitStruct.VoltageRange = FLASH_VOLTAGE_RANGE_3;
EraseInitStruct.Sector = FLASH_SECTOR_11;
EraseInitStruct.NbSectors = 1;
//EraseInitStruct.Banks = FLASH_BANK_1; // or FLASH_BANK_2 or FLASH_BANK_BOTH
st = HAL_FLASHEx_Erase(&EraseInitStruct, &SectorError);
if (st == HAL_OK) {
for (int i = 0; i < size; i += 4) {
st = HAL_FLASH_Program(FLASH_TYPEPROGRAM_WORD, FLASH_USER_START_ADDR + offset + i, *(data + i)); //This is what's giving me trouble
if (st != HAL_OK) {
// handle the error
break;
}
}
}else {
// handle the error
}
HAL_FLASH_Lock();
}
void flash_read(uint32_t offset, uint32_t *data, uint32_t size) {
for (int i = 0; i < size; i += 4) {
*(data + i) = *(__IO uint32_t*)(FLASH_USER_START_ADDR + offset + i);
}
}
int main(void) {
uint32_t data[] = {'a', 'b', 'c', 'd'};
uint32_t read_data[] = {0, 0, 0, 0};
HAL_Init();
SystemClock_Config();
MX_GPIO_Init();
flash_write(0, data, sizeof(data));
flash_read(0, read_data, sizeof(read_data));
if (compareArrays(data,read_data,4))
{
HAL_GPIO_WritePin(GPIOB, GPIO_PIN_7,SET);
}
else
{
HAL_GPIO_WritePin(GPIOB, GPIO_PIN_14,SET);
}
return 0;
}
The problem is that before writing data I must erase a sector, and when I do it with the HAL_FLASHEx_Erase(&EraseInitStruct, &SectorError), function, the program always crashes, and sometimes even corrupts my codespace forcing me to update firmware.
I've selected the sector farthest from the code space but still it crashes when i try to erase it.
I've read in the reference manual that
Any attempt to read the Flash memory while it is being written or erased, causes the bus to
stall. Read operations are processed correctly once the program operation has completed.
This means that code or data fetches cannot be performed while a write/erase operation is
ongoing.
which I believe means the code should ideally be run from RAM while we operate on the flash, but I've seen other people online not have this issue so I'm wondering if that's the only problem I have. With that in mind I wanted to confirm if this is my only issue, or if I'm doing something wrong?
In your loop, you are adding multiples of 4 to i, but then you are adding i to data. When you add to a pointer it is automatically multiplied by the size of the pointed type, so you are adding multiples of 16 bytes and reading past the end of your input buffer.
Also, make sure you initialize all members of EraseInitStruct. Uncomment that line and set the correct value!
I am trying to develop a webrtc simulator in C/C++. For media handling, I plan to use libav. I am thinking of below steps to realize media exchange between two webrtc simulator. Say I have two webrtc simulators A and B.
Read media at A from a input webm file using av_read_frame api.
I assume I will get the encoded media (audio / video) data, am I correct here?
Send the encoded media data to simulator B over a UDP socket.
Simulator B receives the media data in UDP socket as RTP packets.
Simulator B extracts audio/video data from just received RTP packet.
I assume the extracted media data at simulator B are the encoded data only (am I correct here). I do not want to decode it. I want to write it to a file. Later I will play the file to check if I have done everything right.
To simplify this problem lets take out UDP socket part. Then my question reduces to read data from a webm input file, get the encoded media, prepare the packet and write to a output file using av_interleaved_write_frame or any other appropriate api. All these things I want to do using libav.
Is there any example code I can refer.
Or can somebody please guide me to develop it.
I am trying with a test program. As a first step, my aim is to read from a file and write to an output file. I have below code, but it is not working properly.
//#define _AUDIO_WRITE_ENABLED_
#include "libavutil/imgutils.h"
#include "libavutil/samplefmt.h"
#include "libavformat/avformat.h"
static AVPacket pkt;
static AVFormatContext *fmt_ctx = NULL;
static AVFormatContext *av_format_context = NULL;
static AVOutputFormat *av_output_format = NULL;
static AVCodec *video_codec = NULL;
static AVStream *video_stream = NULL;
static AVCodec *audio_codec = NULL;
static AVStream *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *dst_filename = NULL;
int main (int argc, char **argv)
{
int ret = 0;
int index = 0;
if (argc != 3)
{
printf("Usage: ./webm input_video_file output_video_file \n");
exit(0);
}
src_filename = argv[1];
dst_filename = argv[2];
printf("Source file = %s , Destination file = %s\n", src_filename, dst_filename);
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0)
{
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0)
{
fprintf(stderr, "Could not find stream information\n");
exit(2);
}
av_output_format = av_guess_format(NULL, dst_filename, NULL);
if(!av_output_format)
{
fprintf(stderr, "Could not guess output file format\n");
exit(3);
}
av_output_format->audio_codec = AV_CODEC_ID_VORBIS;
av_output_format->video_codec = AV_CODEC_ID_VP8;
av_format_context = avformat_alloc_context();
if(!av_format_context)
{
fprintf(stderr, "Could not allocation av format context\n");
exit(4);
}
av_format_context->oformat = av_output_format;
strcpy(av_format_context->filename, dst_filename);
video_codec = avcodec_find_encoder(av_output_format->video_codec);
if (!video_codec)
{
fprintf(stderr, "Codec not found\n");
exit(5);
}
video_stream = avformat_new_stream(av_format_context, video_codec);
if (!video_stream)
{
fprintf(stderr, "Could not alloc stream\n");
exit(6);
}
avcodec_get_context_defaults3(video_stream->codec, video_codec);
video_stream->codec->codec_id = AV_CODEC_ID_VP8;
video_stream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
video_stream->time_base = (AVRational) {1, 30};
video_stream->codec->width = 640;
video_stream->codec->height = 480;
video_stream->codec->pix_fmt = PIX_FMT_YUV420P;
video_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
video_stream->codec->bit_rate = 400000;
video_stream->codec->gop_size = 10;
video_stream->codec->max_b_frames=1;
#ifdef _AUDIO_WRITE_ENABLED_
audio_codec = avcodec_find_encoder(av_output_format->audio_codec);
if (!audio_codec)
{
fprintf(stderr, "Codec not found audio codec\n");
exit(5);
}
audio_stream = avformat_new_stream(av_format_context, audio_codec);
if (!audio_stream)
{
fprintf(stderr, "Could not alloc stream for audio\n");
exit(6);
}
avcodec_get_context_defaults3(audio_stream->codec, audio_codec);
audio_stream->codec->codec_id = AV_CODEC_ID_VORBIS;
audio_stream->codec->codec_type = AVMEDIA_TYPE_AUDIO;
audio_stream->time_base = (AVRational) {1, 30};
audio_stream->codec->sample_rate = 8000;
audio_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
#endif
if(!(av_output_format->flags & AVFMT_NOFILE))
{
if (avio_open(&av_format_context->pb, dst_filename, AVIO_FLAG_WRITE) < 0)
{
fprintf(stderr, "Could not open '%s'\n", dst_filename);
}
}
/* Before avformat_write_header set the stream */
avformat_write_header(av_format_context, NULL);
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
pkt.stream_index = video_stream->index;
ret = av_read_frame(fmt_ctx, &pkt);
while (ret >= 0)
{
index++;
//pkt.stream_index = video_avstream->index;
if(pkt.stream_index == video_stream->index)
{
printf("Video: Read cycle %d, bytes read = %d, pkt stream index=%d\n", index, pkt.size, pkt.stream_index);
av_write_frame(av_format_context, &pkt);
}
#ifdef _AUDIO_WRITE_ENABLED_
else if(pkt.stream_index == audio_stream->index)
{
printf("Audio: Read cycle %d, bytes read = %d, pkt stream index=%d\n", index, pkt.size, pkt.stream_index);
av_write_frame(av_format_context, &pkt);
}
#endif
av_free_packet(&pkt);
ret = av_read_frame(fmt_ctx, &pkt);
}
av_write_trailer(av_format_context);
/** Exit procedure starts */
avformat_close_input(&fmt_ctx);
avformat_free_context(av_format_context);
return 0;
}
When I execute this program, it outputs "codec not found". Now sure whats going wrong, Can somebody help please.
Codec not found issue is resolved by separately building libvpx1.4 version. Still struggling to read from source file, and writing to a destination file.
EDIT 1: After code modification, only video stuff I am able to write to a file, though some more errors are still present.
EDIT 2: With modified code (2nd round), I see video frames are written properly. For audio frames I added the code under a macro _AUDIO_WRITE_ENABLED_ , but if I enable this macro program crashing. Can somebody guide whats wrong in audio write part (code under macro _AUDIO_WRITE_ENABLED_).
I am not fully answering your question, but I hope we will get to the final solution eventually. When I tried to run your code, I got this error "time base not set".
Time base and other header specs are part of codec. This is, how I have this thing specified for writing into file (vStream is of AVStream):
#if LIBAVCODEC_VER_AT_LEAST(53, 21)
avcodec_get_context_defaults3(rc->vStream->codec, AVMEDIA_TYPE_VIDEO);
#else
avcodec_get_context_defaults2(rc->vStream->codec, AVMEDIA_TYPE_VIDEO);
#endif
#if LIBAVCODEC_VER_AT_LEAST(54, 25)
vStream->codec->codec_id = AV_CODEC_ID_VP8;
#else
vStream->codec->codec_id = CODEC_ID_VP8;
#endif
vStream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
vStream->codec->time_base = (AVRational) {1, 30};
vStream->codec->width = 640;
vStream->codec->height = 480;
vStream->codec->pix_fmt = PIX_FMT_YUV420P;
EDIT: I ran your program in Valgrind and it segfaults on av_write_frame. Looks like its time_base and other specs for output are not set properly.
Add the specs before avformat_write_header(), before it is too late.
I am trying to do a simple playback from a file functionality and it appears that my callback function is never called. It doesn't really make sense because all of the OSStatuses come back 0 and other numbers all appear correct as well (like the output packets read pointer from AudioFileReadPackets).
Here is the setup:
OSStatus stat;
stat = AudioFileOpenURL(
(CFURLRef)urlpath, kAudioFileReadPermission, 0, &aStreamData->aFile
);
UInt32 dsze = 0;
stat = AudioFileGetPropertyInfo(
aStreamData->aFile, kAudioFilePropertyDataFormat, &dsze, 0
);
stat = AudioFileGetProperty(
aStreamData->aFile, kAudioFilePropertyDataFormat, &dsze, &aStreamData->aDescription
);
stat = AudioQueueNewOutput(
&aStreamData->aDescription, bufferCallback, aStreamData, NULL, NULL, 0, &aStreamData->aQueue
);
aStreamData->pOffset = 0;
for(int i = 0; i < NUM_BUFFERS; i++) {
stat = AudioQueueAllocateBuffer(
aStreamData->aQueue, aStreamData->aDescription.mBytesPerPacket, &aStreamData->aBuffer[i]
);
bufferCallback(aStreamData, aStreamData->aQueue, aStreamData->aBuffer[i]);
}
stat = AudioQueuePrime(aStreamData->aQueue, 0, NULL);
stat = AudioQueueStart(aStreamData->aQueue, NULL);
(Not shown is where I'm checking the value of stat in between the functions, it just comes back normal.)
And the callback function:
void bufferCallback(void *uData, AudioQueueRef queue, AudioQueueBufferRef buffer) {
UInt32 bread = 0;
UInt32 pread = buffer->mAudioDataBytesCapacity / player->aStreamData->aDescription.mBytesPerPacket;
OSStatus stat;
stat = AudioFileReadPackets(
player->aStreamData->aFile, false, &bread, NULL, player->aStreamData->pOffset, &pread, buffer->mAudioData
);
buffer->mAudioDataByteSize = bread;
stat = AudioQueueEnqueueBuffer(queue, buffer, 0, NULL);
player->aStreamData->pOffset += pread;
}
Where aStreamData is my user data struct (typedefed so I can use it as a class property) and player is a static instance of the controlling Objective-C class. If any other code is wanted please let me know. I am a bit at my wit's end. Printing any of the numbers involved here yields the correct result, including functions in bufferCallback when I call it myself in the allocate loop. It just never gets called thereafter. The start up method returns and nothing happens.
Also anecdotally, I am using a peripheral device (an MBox Pro 3) to play the sound which CoreAudio only boots up when it is about to output. IE if I start iTunes or something, the speakers pop faintly and there is an LED that goes from blinking to solid. The device boots up like it does so CA is definitely doing something. (Also I've of course tried it with the onboard Macbook sound sans the device.)
I've read other solutions to problems that sound similiar and they don't work. Stuff like using multiple buffers which I am doing now and doesn't appear to make any difference.
I basically assume I am doing something obviously wrong somehow but not sure what it could be. I've read the relevant documentation, looked at the available code examples and scoured the net a bit for answers and it appears that this is all I need to do and it should just go.
At the very least, is there anything else I can do to investigate?
My first answer was not good enough, so I compiled a minimal example that will play a 2 channel, 16 bit wave file.
The main difference from your code is that I made a property listener listening for play start and stop events.
As for your code, it seems legit at first glance. Two things I will point out, though:
1. Is seems you are allocating buffers with TOO SMALL a buffer size. I have noticed that AudioQueues won't play if the buffers are too small, which seems to fit your problem.
2. Have you verified the properties returned?
Back to my code example:
Everything is hard coded, so it is not exactly good coding practice, but it shows how you can do it.
AudioStreamTest.h
#import <Foundation/Foundation.h>
#import <AudioToolbox/AudioToolbox.h>
uint32_t bufferSizeInSamples;
AudioFileID file;
UInt32 currentPacket;
AudioQueueRef audioQueue;
AudioQueueBufferRef buffer[3];
AudioStreamBasicDescription audioStreamBasicDescription;
#interface AudioStreamTest : NSObject
- (void)start;
- (void)stop;
#end
AudioStreamTest.m
#import "AudioStreamTest.h"
#implementation AudioStreamTest
- (id)init
{
self = [super init];
if (self) {
bufferSizeInSamples = 441;
file = NULL;
currentPacket = 0;
audioStreamBasicDescription.mBitsPerChannel = 16;
audioStreamBasicDescription.mBytesPerFrame = 4;
audioStreamBasicDescription.mBytesPerPacket = 4;
audioStreamBasicDescription.mChannelsPerFrame = 2;
audioStreamBasicDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
audioStreamBasicDescription.mFramesPerPacket = 1;
audioStreamBasicDescription.mReserved = 0;
audioStreamBasicDescription.mSampleRate = 44100;
}
return self;
}
- (void)start {
AudioQueueNewOutput(&audioStreamBasicDescription, AudioEngineOutputBufferCallback, (__bridge void *)(self), NULL, NULL, 0, &audioQueue);
AudioQueueAddPropertyListener(audioQueue, kAudioQueueProperty_IsRunning, AudioEnginePropertyListenerProc, NULL);
AudioQueueStart(audioQueue, NULL);
}
- (void)stop {
AudioQueueStop(audioQueue, YES);
AudioQueueRemovePropertyListener(audioQueue, kAudioQueueProperty_IsRunning, AudioEnginePropertyListenerProc, NULL);
}
void AudioEngineOutputBufferCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
if (file == NULL) return;
UInt32 bytesRead = bufferSizeInSamples * 4;
UInt32 packetsRead = bufferSizeInSamples;
AudioFileReadPacketData(file, false, &bytesRead, NULL, currentPacket, &packetsRead, inBuffer->mAudioData);
inBuffer->mAudioDataByteSize = bytesRead;
currentPacket += packetsRead;
if (bytesRead == 0) {
AudioQueueStop(inAQ, false);
}
else {
AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
}
}
void AudioEnginePropertyListenerProc (void *inUserData, AudioQueueRef inAQ, AudioQueuePropertyID inID) {
//We are only interested in the property kAudioQueueProperty_IsRunning
if (inID != kAudioQueueProperty_IsRunning) return;
//Get the status of the property
UInt32 isRunning = false;
UInt32 size = sizeof(isRunning);
AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &isRunning, &size);
if (isRunning) {
currentPacket = 0;
NSString *fileName = #"/Users/roy/Documents/XCodeProjectsData/FUZZ/03.wav";
NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: fileName];
AudioFileOpenURL((__bridge CFURLRef) fileURL, kAudioFileReadPermission, 0, &file);
for (int i = 0; i < 3; i++){
AudioQueueAllocateBuffer(audioQueue, bufferSizeInSamples * 4, &buffer[i]);
UInt32 bytesRead = bufferSizeInSamples * 4;
UInt32 packetsRead = bufferSizeInSamples;
AudioFileReadPacketData(file, false, &bytesRead, NULL, currentPacket, &packetsRead, buffer[i]->mAudioData);
buffer[i]->mAudioDataByteSize = bytesRead;
currentPacket += packetsRead;
AudioQueueEnqueueBuffer(audioQueue, buffer[i], 0, NULL);
}
}
else {
if (file != NULL) {
AudioFileClose(file);
file = NULL;
for (int i = 0; i < 3; i++) {
AudioQueueFreeBuffer(audioQueue, buffer[i]);
buffer[i] = NULL;
}
}
}
}
-(void)dealloc {
[super dealloc];
AudioQueueDispose(audioQueue, true);
audioQueue = NULL;
}
#end
Lastly, I want to include some research I have done today to test the robustness of AudioQueues.
I have noticed that if you make too small AudioQueue buffers, it won't play at all. That made me play around a bit to see why it is not playing.
If I try buffer size that can hold only 150 samples, I get no sound at all.
If I try buffer size that can hold 175 samples, it plays the whole song through, but with A lot of distortion. 175 amounts to a tad less than 4 ms of audio.
AudioQueue keeps asking for new buffers as long as you keep supplying buffers. That is regardless of AudioQueue actually playing your buffers or not.
If you supply a buffer with size 0, the buffer will be lost and an error kAudioQueueErr_BufferEmpty is returned for that queue enqueue request. You will never see AudioQueue ask you to fill that buffer again. If this happened for the last queue you have posted, AudioQueue will stop asking you to fill any more buffers. In that case you will not hear any more audio for that session.
To see why AudioQueues is not playing anything with smaller buffer sizes, I made a test to see if my callback is called at all even when there is no sound. The answer is that the buffers gets called all the time as long as AudioQueues is playing and needs data.
So if you keep feeding buffers to the queue, no buffer is ever lost. It doesn't happen. Unless there is an error, of course.
So why is no sound playing?
I tested to see if 'AudioQueueEnqueueBuffer()' returned any errors. It did not. No other errors within my play routine either. The data returned from reading from file is also good.
Everything is normal, buffers are good, data re-enqueued is good, there is just no sound.
So my last test was to slowly increase buffer size till I could hear anything. I finally heard faint and sporadic distortion.
Then it came to me...
It seems that the problem lies with that the system tries to keep the stream in sync with time so if you enqueue audio, and the time for the audio you wanted to play has passed, it will just skip that part of the buffer. If the buffer size becomes too small, more and more data is dropped or skipped until the audio system is in sync again. Which is never if the buffer size is too small. (You can hear this as distortion if you chose a buffer size that is barely large enough to support continuous play.)
If you think about it, it is the only way the audio queue can work, but it is a good realisation when you are clueless like me and "discover" how it really works.
I decided to take a look at this again and was able to solve it by making the buffers larger. I've accepted the answer by #RoyGal since it was their suggestion but I wanted to provide the actual code that works since I guess others are having the same problem (question has a few favorites that aren't me at the moment).
One thing I tried was making the packet size larger:
aData->aDescription.mFramesPerPacket = 512; // or some other number
aData->aDescription.mBytesPerPacket = (
aData->aDescription.mFramesPerPacket * aData->aDescription.mBytesPerFrame
);
This does NOT work: it causes AudioQueuePrime to fail with an AudioConverterNew returned -50 message. I guess it wants mFramesPerPacket to be 1 for PCM.
(I also tried setting the kAudioQueueProperty_DecodeBufferSizeFrames property which didn't seem to do anything. Not sure what it's for.)
The solution seems to be to only allocate the buffer(s) with the specified size:
AudioQueueAllocateBuffer(
aData->aQueue,
aData->aDescription.mBytesPerPacket * N_BUFFER_PACKETS / N_BUFFERS,
&aData->aBuffer[i]
);
And the size has to be sufficiently large. I found the magic number is:
mBytesPerPacket * 1024 / N_BUFFERS
(Where N_BUFFERS is the number of buffers and should be > 1 or playback is choppy.)
Here is an MCVE demonstrating the issue and solution:
#import <Foundation/Foundation.h>
#import <AudioToolbox/AudioToolbox.h>
#import <AudioToolbox/AudioQueue.h>
#import <AudioToolbox/AudioFile.h>
#define N_BUFFERS 2
#define N_BUFFER_PACKETS 1024
typedef struct AStreamData {
AudioFileID aFile;
AudioQueueRef aQueue;
AudioQueueBufferRef aBuffer[N_BUFFERS];
AudioStreamBasicDescription aDescription;
SInt64 pOffset;
volatile BOOL isRunning;
} AStreamData;
void printASBD(AudioStreamBasicDescription* desc) {
printf("mSampleRate = %d\n", (int)desc->mSampleRate);
printf("mBytesPerPacket = %d\n", desc->mBytesPerPacket);
printf("mFramesPerPacket = %d\n", desc->mFramesPerPacket);
printf("mBytesPerFrame = %d\n", desc->mBytesPerFrame);
printf("mChannelsPerFrame = %d\n", desc->mChannelsPerFrame);
printf("mBitsPerChannel = %d\n", desc->mBitsPerChannel);
}
void bufferCallback(
void *vData, AudioQueueRef aQueue, AudioQueueBufferRef aBuffer
) {
AStreamData* aData = (AStreamData*)vData;
UInt32 bRead = 0;
UInt32 pRead = (
aBuffer->mAudioDataBytesCapacity / aData->aDescription.mBytesPerPacket
);
OSStatus stat;
stat = AudioFileReadPackets(
aData->aFile, false, &bRead, NULL, aData->pOffset, &pRead, aBuffer->mAudioData
);
if(stat != 0) {
printf("AudioFileReadPackets returned %d\n", stat);
}
if(pRead == 0) {
aData->isRunning = NO;
return;
}
aBuffer->mAudioDataByteSize = bRead;
stat = AudioQueueEnqueueBuffer(aQueue, aBuffer, 0, NULL);
if(stat != 0) {
printf("AudioQueueEnqueueBuffer returned %d\n", stat);
}
aData->pOffset += pRead;
}
AStreamData* beginPlayback(NSURL* path) {
static AStreamData* aData;
aData = malloc(sizeof(AStreamData));
OSStatus stat;
stat = AudioFileOpenURL(
(CFURLRef)path, kAudioFileReadPermission, 0, &aData->aFile
);
printf("AudioFileOpenURL returned %d\n", stat);
UInt32 dSize = 0;
stat = AudioFileGetPropertyInfo(
aData->aFile, kAudioFilePropertyDataFormat, &dSize, 0
);
printf("AudioFileGetPropertyInfo returned %d\n", stat);
stat = AudioFileGetProperty(
aData->aFile, kAudioFilePropertyDataFormat, &dSize, &aData->aDescription
);
printf("AudioFileGetProperty returned %d\n", stat);
printASBD(&aData->aDescription);
stat = AudioQueueNewOutput(
&aData->aDescription, bufferCallback, aData, NULL, NULL, 0, &aData->aQueue
);
printf("AudioQueueNewOutput returned %d\n", stat);
aData->pOffset = 0;
for(int i = 0; i < N_BUFFERS; i++) {
// change YES to NO for stale playback
if(YES) {
stat = AudioQueueAllocateBuffer(
aData->aQueue,
aData->aDescription.mBytesPerPacket * N_BUFFER_PACKETS / N_BUFFERS,
&aData->aBuffer[i]
);
} else {
stat = AudioQueueAllocateBuffer(
aData->aQueue,
aData->aDescription.mBytesPerPacket,
&aData->aBuffer[i]
);
}
printf(
"AudioQueueAllocateBuffer returned %d for aBuffer[%d] with capacity %d\n",
stat, i, aData->aBuffer[i]->mAudioDataBytesCapacity
);
bufferCallback(aData, aData->aQueue, aData->aBuffer[i]);
}
UInt32 numFramesPrepared = 0;
stat = AudioQueuePrime(aData->aQueue, 0, &numFramesPrepared);
printf("AudioQueuePrime returned %d with %d frames prepared\n", stat, numFramesPrepared);
stat = AudioQueueStart(aData->aQueue, NULL);
printf("AudioQueueStart returned %d\n", stat);
UInt32 pSize = sizeof(UInt32);
UInt32 isRunning;
stat = AudioQueueGetProperty(
aData->aQueue, kAudioQueueProperty_IsRunning, &isRunning, &pSize
);
printf("AudioQueueGetProperty returned %d\n", stat);
aData->isRunning = !!isRunning;
return aData;
}
void endPlayback(AStreamData* aData) {
OSStatus stat = AudioQueueStop(aData->aQueue, NO);
printf("AudioQueueStop returned %d\n", stat);
}
NSString* getPath() {
// change NO to YES and enter path to hard code
if(NO) {
return #"";
}
char input[512];
printf("Enter file path: ");
scanf("%[^\n]", input);
return [[NSString alloc] initWithCString:input encoding:NSASCIIStringEncoding];
}
int main(int argc, const char* argv[]) {
NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
NSURL* path = [NSURL fileURLWithPath:getPath()];
AStreamData* aData = beginPlayback(path);
if(aData->isRunning) {
do {
printf("Queue is running...\n");
[NSThread sleepForTimeInterval:1.0];
} while(aData->isRunning);
endPlayback(aData);
} else {
printf("Playback did not start\n");
}
[pool drain];
return 0;
}
I am currently working at a logger that uses a MSP430F2618 MCU and SanDisk 4GB SDHC Card.
Card initialization works as expected, I also can read MBR and FAT table.
The problem is that I can't write any data on it. I have checked if it is write protected by notch, but it's not. Windows 7 OS has no problem reading/writing to it.
Though, I have used a tool called "HxD" and I've tried to alter some sectors (under Windows). When I try to save the content to SD card, the tool pop up a windows telling me "Access denied!".
Then I came back to my code for writing to SD card:
uint8_t SdWriteBlock(uchar_t *blockData, const uint32_t address)
{
uint8_t result = OP_ERROR;
uint16_t count;
uchar_t dataResp;
uint8_t idx;
for (idx = RWTIMEOUT; idx > 0; idx--)
{
CS_LOW();
SdCommand(CMD24, address, 0xFF);
dataResp = SdResponse();
if (dataResp == 0x00)
{
break;
}
else
{
CS_HIGH();
SdWrite(0xFF);
}
}
if (0x00 == dataResp)
{
//send command success, now send data starting with DATA TOKEN = 0xFE
SdWrite(0xFE);
//send 512 bytes of data
for (count = 0; count < 512; count++)
{
SdWrite(*blockData++);
}
//now send tow CRC bytes ,through it is not used in the spi mode
//but it is still needed in transfer format
SdWrite(0xFF);
SdWrite(0xFF);
//now read in the DATA RESPONSE TOKEN
do
{
SdWrite(0xFF);
dataResp = SdRead();
}
while (dataResp == 0x00);
//following the DATA RESPONSE TOKEN are a number of BUSY bytes
//a zero byte indicates the SD/MMC is busy programing,
//a non_zero byte indicates SD/MMC is not busy
dataResp = dataResp & 0x0F;
if (0x05 == dataResp)
{
idx = RWTIMEOUT;
do
{
SdWrite(0xFF);
dataResp = SdRead();
if (0x0 == dataResp)
{
result = OP_OK;
break;
}
idx--;
}
while (idx != 0);
CS_HIGH();
SdWrite(0xFF);
}
else
{
CS_HIGH();
SdWrite(0xFF);
}
}
return result;
}
The problem seems to be when I am waiting for card status:
do
{
SdWrite(0xFF);
dataResp = SdRead();
}
while (dataResp == 0x00);
Here I am waiting for a response of type "X5"(hex value) where X is undefined.
But most of the cases the response is 0x00 (hex value) and I don't get out of the loop. Few cases are when the response is 0xFF (hex value).
I can't figure out what is the problem.
Can anyone help me? Thanks!
4GB SDHC
We need to see much more of your code. Many µC SPI codebases only support SD cards <= 2 GB, so using a smaller card might work.
You might check it yourself: SDHC needs a CMD 8 and an ACMD 41 after the CMD 0 (GO_IDLE_STATE) command, otherwise you cannot read or write data to it.
Thank you for your answers, but I solved my problem. It was a problem of timing. I had to put a delay at specific points.
I'd like to connect to IMAP and POP3 servers, for IMAP I'm currently using MailCore. Unfortunately I don't find a suitable POP3-framwork.
I tried with libetpan:
mailpop3 * pop3;
int r;
pop3 = mailpop3_new(0, NULL);
r = mailpop3_ssl_connect(pop3, "pop.gmail.com", 995);
check_error(r, "connect failed");
but I always get a connection refused error; and it's only C, I would prefer Objective-C. Even better would be a library which I could use for both; IMAP and POP3.
I haven't used OCMail, but it seems like it's what you're looking for. It claims to support "POP3, IMAP4, SMTP, POPS, IMAPS, SMTPS".
Edit: Build Error
Turns out, the solution is actually in the README file.
Once you've downloaded the ZIP from Github, open the Xcode project.
Build for Profiling (Product Menu > Build For > Profiling (Command-Shift-I)).
Open Xcode preferences and go to "Locations"
Under Derived Data, next to the Advanced button you'll see a file path (something like /Users/YourUserName/Library/Developer/Xcode/DerivedData). There'll be a little arrow next to the path; click the arrow to go to that location in Finder.
It'll take you to a folder with all of your Xcode projects. Find the folder whose name starts with OCMail (and has a bunch of gibberish after it).
In that folder, find Build > Products > Debug-iphoneos > libOCMail.a. That's the library file you'll want to add into your Xcode project. Just drag it into your Xcode project and you should be good to go.
I got a bunch of errors building the project. They came from a badly defined enum type. Here's a cleaned up file:
http://cl.ly/code/442x2x3X3Y2I
Just download and replace the existing MimeMessage.m file before you build.
I was working with libetpan in past and I was connecting to pop3 server without problems, so I checked if it still working. I used code from here: https://github.com/dinhviethoa/libetpan/blob/master/tests/pop-sample.c and adjusted it for iOS.
If You use it, You will see a lot of warnings and app will crash after fetching first message, but connecting is working (of course, You need to enter Your email login and password).
I'm not saying that libetpan is good solution. When I was developing app with mail support I also used mailcore for IMAP and eventually resigned from POP3 support. But if You run from options it could be useful.
static void check_error(int r, char * msg)
{
if (r == MAILPOP3_NO_ERROR)
return;
fprintf(stderr, "%s\n", msg);
exit(EXIT_FAILURE);
}
-(IBAction)testButtonClick:(id)sender
{
mailpop3 * pop3;
int r;
carray * list;
unsigned int i;
// if (argc < 3) {
// fprintf(stderr, "syntax: pop-sample [gmail-email-address] [gmail- password]\n");
// exit(EXIT_FAILURE);
// }
mkdir("download", 0700);
pop3 = mailpop3_new(0, NULL);
r = mailpop3_ssl_connect(pop3, "pop.gmail.com", 995);
check_error(r, "connect failed");
r = mailpop3_user(pop3, #"mail login".cString);
check_error(r, "user failed");
r = mailpop3_pass(pop3, #"mail password".cString);
check_error(r, "pass failed");
r = mailpop3_list(pop3, &list);
check_error(r, "list failed");
NSLog(#"carray_count(list_: %d", carray_count(list));
for(i = 0 ; i < carray_count(list) ; i ++) {
struct mailpop3_msg_info * info;
char * msg_content;
size_t msg_size;
FILE * f;
char filename[512];
struct stat stat_info;
info = (mailpop3_msg_info *) carray_get(list, i);
if (info->msg_uidl == NULL) {
continue;
}
snprintf(filename, sizeof(filename), "download/%s.eml", info->msg_uidl);
r = stat(filename, &stat_info);
if (r == 0) {
printf("already fetched %u %s\n", info->msg_index, info->msg_uidl);
continue;
}
if(msg_content != NULL)
NSLog(#"msg_content: %#", [NSString stringWithUTF8String:msg_content]);
r = mailpop3_retr(pop3, info->msg_index, &msg_content, &msg_size);
check_error(r, "get failed");
// f = fopen(filename, "w");
// fwrite(msg_content, 1, msg_size, f);
// fclose(f);
// mailpop3_retr_free(msg_content);
if (info->msg_uidl != NULL) {
printf("fetched %u %s\n", info->msg_index, info->msg_uidl);
}
else {
printf("fetched %u\n", info->msg_index);
}
}
mailpop3_quit(pop3);
mailpop3_free(pop3);
// exit(EXIT_SUCCESS);
}