Distance between 2 arduino's using rf links - communication

I currently have a setup where I send a char using a Tx of 434MHz and an Uno to a Mega with a Rx. The Mega counts how many times it receives the char and then if it falls below a certain number it triggers an alarm. Is this a viable way to measure the distance between two microcontrollers while indoors or is there a better way.
Transmitter (Mega)
#include <SoftwareSerial.h>
int rxPin=2; //Goes to the Receiver Pin
int txPin=5; //Make sure it is set to pin 5 going to input of receiver
SoftwareSerial txSerial = SoftwareSerial(rxPin, txPin);
SoftwareSerial rxSerial = SoftwareSerial(txPin, rxPin);
char sendChar ='H';
void setup() {
pinMode(rxPin, INPUT);
pinMode(txPin,OUTPUT);
txSerial.begin(2400);
rxSerial.begin(2400);
}
void loop() {
txSerial.println(sendChar);
Serial.print(sendChar);
}
Receiver
#include <SoftwareSerial.h>
//Make sure it is set to pin 5 going to the data input of the transmitter
int rxPin=5;
int txPin=3; //Don't need to make connections
int LED=13;
int BUZZ=9;
int t=0;
char incomingChar = 0;
int counter = 0;
SoftwareSerial rxSerial = SoftwareSerial(rxPin, txPin);
void setup() {
pinMode(rxPin, INPUT); //initilize rxpin as input
pinMode(BUZZ, OUTPUT); //initilize buzz for output
pinMode(LED, OUTPUT); //initilize led for output
rxSerial.begin(2400); //set baud rate for transmission
Serial.begin(2400); //see above
}
void loop() {
for(int i=0; i<200; i++) {
incomingChar = rxSerial.read(); //read incoming msg from tx
if (incomingChar =='H') {
counter++; //if we get bit "h" count it
}
delay(5); //delay of 10 secs
}
Serial.println(incomingChar);
Serial.println(counter); //prints the the bits we recieved
if(counter<55) {
//if we receive less than 100 bits than print out of range triggers alarm
Serial.println("out of range");
tone(BUZZ,5000,500);digitalWrite(LED,HIGH);
}
else {
noTone(BUZZ);digitalWrite(LED, LOW);
//if we get more than 100 bits than we are within range turn off alarm
Serial.println("in range");
}
counter = 0;
incomingChar=0;
}

In theory you could achieve distance measuring by making the uno send a message which the mega would echo back. That would give the uno a round-trip time for message propagation between the arduinos. You would have to approximate the processing delays. After that it is basic physics. That is basically the same as how radar works. The actual delay would be something like
troundtrip = tuno send + 2*tpropagation + tmega receive + tmega send + tuno receive
I am guessing the distance you are trying to achieve is in the order of meters. Required resolution is going to be an issue, because s = vt => t = s/v, where s is the distance between your arduinos and v = c in case of radio waves. As the transmission delays should stay constant, you have to be able to measure differences in the order of 1/c second intervals, basically. I am not very familiar with arduinos, so I do not know if they are capable of this kind of measurements.

I would suggest you use an ultrasonic range finder like the Maxbotix HRLV-EZ4 sold by Sparkfun.
It is within your price range and it should be able to measure distances up to 5m/195 inches with 1mm resolution.

It is actually possible to do it, I have seen it be done with other microcontrollers. Therefore using arduino you would have to solve the equations,fit in arduino language and make a lot of measurements to value discrepancies over communication itself. Do not forget about atmospheric attenuation wich need to be known and fit in the equations. Humidity may deviate electromagnetic waves.

Related

STM32 - Reading I2S to record a .WAV file. Audio choppy, what is causing it?

I'm using an STM32 (STM32F446RE) to receive audio from two INMP441 mems microphone in an stereo setup via I2S protocol and record it into a .WAV on a micro SD card, using the HAL library.
I wrote the firmware that records audio into a .WAV with FreeRTOS. But the audio files that I record sound like Darth Vader. Here is a screenshot of the audio in audacity:
if you zoom in you can see a constant noise being inserted in between the real audio data:
I don't know what is causing this.
I have tried increasing the MessageQueue, but that doesnt seem to be the problem, the queue is kept at 0 most of the time. I've tried different frame sizes and sampling rates, changing the number of channels, using only one inmp441. All this without any success.
I proceed explaining the firmware.
Here is a block diagram of the architecture for the RTOS that I have implemented:
It consists of three tasks. The first one receives a command via UART (with interrupts) that signals to start or stop recording. the second one is simply an state machine that walks through the steps to write a .WAV.
Here the code for the WriteWavFileTask:
switch(audio_state)
{
case STATE_START_RECORDING:
sprintf(filename, "%saud_%03d.wav", SDPath, count++);
do
{
res = f_open(&file_ptr, filename, FA_CREATE_ALWAYS|FA_WRITE);
}
while(res != FR_OK);
res = fwrite_wav_header(&file_ptr, I2S_SAMPLE_FREQUENCY, I2S_FRAME, 2);
HAL_I2S_Receive_DMA(&hi2s2, aud_buf, READ_SIZE);
audio_state = STATE_RECORDING;
break;
case STATE_RECORDING:
osDelay(50);
break;
case STATE_STOP:
HAL_I2S_DMAStop(&hi2s2);
while(osMessageQueueGetCount(AudioQueueHandle)) osDelay(1000);
filesize = f_size(&file_ptr);
data_len = filesize - 44;
total_len = filesize - 8;
f_lseek(&file_ptr, 4);
f_write(&file_ptr, (uint8_t*)&total_len, 4, bw);
f_lseek(&file_ptr, 40);
f_write(&file_ptr, (uint8_t*)&data_len, 4, bw);
f_close(&file_ptr);
audio_state = STATE_IDLE;
break;
case STATE_IDLE:
osThreadSuspend(WAVHandle);
audio_state = STATE_START_RECORDING;
break;
default:
osDelay(50);
break;
Here are the macros used in the code for readability:
#define I2S_DATA_WORD_LENGTH (24) // industry-standard 24-bit I2S
#define I2S_FRAME (32) // bits per sample
#define READ_SIZE (128) // samples to read from I2S
#define WRITE_SIZE (READ_SIZE*I2S_FRAME/16) // half words to write
#define WRITE_SIZE_BYTES (WRITE_SIZE*2) // bytes to write
#define I2S_SAMPLE_FREQUENCY (16000) // sample frequency
The last task is the responsible for processing the buffer received via I2S. Here is the code:
void convert_endianness(uint32_t *array, uint16_t Size) {
for (int i = 0; i < Size; i++) {
array[i] = __REV(array[i]);
}
}
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
convert_endianness((uint32_t *)aud_buf, READ_SIZE);
osMessageQueuePut(AudioQueueHandle, aud_buf, 0L, 0);
HAL_I2S_Receive_DMA(hi2s, aud_buf, READ_SIZE);
}
void pvrWriteAudioTask(void *argument)
{
/* USER CODE BEGIN pvrWriteAudioTask */
static UINT *bw;
static uint16_t aud_ptr[WRITE_SIZE];
/* Infinite loop */
for(;;)
{
osMessageQueueGet(AudioQueueHandle, aud_ptr, 0L, osWaitForever);
res = f_write(&file_ptr, aud_ptr, WRITE_SIZE_BYTES, bw);
}
/* USER CODE END pvrWriteAudioTask */
}
This tasks reads from a queue an array of 256 uint16_t elements containing the raw audio data in PCM. f_write takes the Size parameter in number of bytes to write to the SD card, so 512 bytes. The I2S Receives 128 frames (for a 32 bit frame, 128 words).
The following is the configuration for the I2S and clocks:
Any help would be much appreciated!
Solution
As pmacfarlane pointed out, the problem was with the method used for buffering the audio data. The solution consisted of easing the overhead on the ISR and implementing a circular DMA for double buffering. Here is the code:
#define I2S_DATA_WORD_LENGTH (24) // industry-standard 24-bit I2S
#define I2S_FRAME (32) // bits per sample
#define READ_SIZE (128) // samples to read from I2S
#define BUFFER_SIZE (READ_SIZE*I2S_FRAME/16) // number of uint16_t elements expected
#define WRITE_SIZE_BYTES (BUFFER_SIZE*2) // bytes to write
#define I2S_SAMPLE_FREQUENCY (16000) // sample frequency
uint16_t aud_buf[2*BUFFER_SIZE]; // Double buffering
static volatile int16_t *BufPtr;
void convert_endianness(uint32_t *array, uint16_t Size) {
for (int i = 0; i < Size; i++) {
array[i] = __REV(array[i]);
}
}
void HAL_I2S_RxHalfCpltCallback(I2S_HandleTypeDef *hi2s)
{
BufPtr = aud_buf;
osSemaphoreRelease(RxAudioSemHandle);
}
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
BufPtr = &aud_buf[BUFFER_SIZE];
osSemaphoreRelease(RxAudioSemHandle);
}
void pvrWriteAudioTask(void *argument)
{
/* USER CODE BEGIN pvrWriteAudioTask */
static UINT *bw;
/* Infinite loop */
for(;;)
{
osSemaphoreAcquire(RxAudioSemHandle, osWaitForever);
convert_endianness((uint32_t *)BufPtr, READ_SIZE);
res = f_write(&file_ptr, BufPtr, WRITE_SIZE_BYTES, bw);
}
/* USER CODE END pvrWriteAudioTask */
}
Problems
I think the problem is your method of buffering the audio data - mainly in this function:
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
convert_endianness((uint32_t *)aud_buf, READ_SIZE);
osMessageQueuePut(AudioQueueHandle, aud_buf, 0L, 0);
HAL_I2S_Receive_DMA(hi2s, aud_buf, READ_SIZE);
}
The main problem is that you are re-using the same buffer each time. You have queued a message to save aud_buf to the SD-card, but you've also instructed the I2S to start DMAing data into that same buffer, before it has been saved. You'll end up saving some kind of mish-mash of "old" data and "new" data.
#Flexz pointed out that the message queue takes a copy of the data, so there is no issue about the I2S writing over the data that is being written to the SD-card. However, taking the copy (in an ISR) adds overhead, and delays the start of the new I2S DMA.
Another problem is that you are doing the endian conversion in this function (that is called from an ISR). This will block any other (lower priority) interrupts from being serviced while this happens, which is a bad thing in an embedded system. You should do the endian conversion in the task that reads from the queue. ISRs should be very short and do the minimum possible work (often just setting a flag, giving a semaphore, or adding something to a queue).
Lastly, while you are doing the endian conversion, what is happening to audio samples? The previous DMA has completed, and you haven't started a new one, so they will just be dropped on the floor.
Possible solution
You probably want to allocate a suitably big buffer, and configure your DMA to work in circular buffer mode. This means that once started, the DMA will continue forever (until you stop it), so you'll never drop any samples. There won't be any gap between one DMA finishing and a new one starting, since you never need to start a new one.
The DMA provides a "half-complete" interrupt, to say when it has filled half the buffer. So start the DMA, and when you get the half-complete interrupt, queue up the first half of the buffer to be saved. When you get the fully-complete interrupt, queue up the second half of the buffer to be saved. Rinse and repeat.
You might want to add some logic to detect if the interrupt happens before the previous save has completed, since the data will be overrun and possibly corrupted. Depending on the speed of the SD-card (and the sample rate), this may or may not be a problem.

Servo Moves Upon Plugin. Sometimes to 90 and stops. Sometimes to 180 and back to 0

Newbie here. When I plug the USB into my Arduino to power the project, my servo wants to move around right away. Sometimes it goes to 90 degrees and stops until it is triggered by the ultrasonic sensor. In this case, it starts at 90, goes back to 0, to 180 back to 0...
The other case is that upon powering it up, sometimes the servo goes to 180 and then back to 0. Neither of these options are good because I am making a lock on a box and I can't have it unlock as soon as I power up the Arduino. I need the servo to remain at 0 until I trigger the sensor.
(More mods have to be made to the code so that the lock will remain open until I trigger it again but that is another discussion).
Your help is much appreciated!
#include<Servo.h>
int trig=8;
int echo=9;
int dt=10;
Servo servo;
//int distance,duration;
void setup() {
// put your setup code here, to run once:
pinMode(trig,OUTPUT);
pinMode(echo,INPUT);
Serial.begin(9600);
servo.attach(11);
}
void loop() {
// put your main code here, to run repeatedly:
if (calc_dis()<5)
{
for (int i=0;i<=540;i++)
{
servo.write(i);
delay(1);
}
delay(100);
for (int i=540;i>=0;i--)
{
servo.write(i);
delay(1);
}
delay(100);
}
}
//This code is written to calculate the DISTANCE using ULTRASONIC SENSOR
int calc_dis()
{
int duration,distance;
digitalWrite(trig,HIGH);
delay(dt);
digitalWrite(trig,LOW);
duration=pulseIn(echo,HIGH);
distance = (duration/2) / 29.1;
return distance;
}

detect pulse objective c

At the moment i'm creating an iOS app which is visualizing port status of an arduino. Therefor the iPad is receiving information via Serial Cable from Arduino.
The Arduino sends every 100ms a package with it's current port status. This status is visualized on the iPad.
The Ports are input Ports. I've recognized that the device i'm reading is pulsing the ports so the Arduino reads high low level alternating. That creates flickering in the visualization.
My question is now how to detect if the level is up or the input is flickering.
The port is high for x seconds get low for y seconds and after that it repeats. If the port is low for z seconds i need to set the port as low in the visualization. Otherwise it is high.
- (void) readBytesAvailable:(UInt32)numBytes {
int bytesRead = [manager read:rxBuffer Length:numBytes];
if(rxBuffer[i]==48){
[self setButtonRed];
}else if(rxBuffer[i]==49){
[self setButtonWhite]
}
}
https://www.dropbox.com/s/bhy5lbm8lkdhnoy/3wire.png?dl=0
If I understood well the scenario is this: you want to determine if the output is alternating its state or it is stuck at ground. You didn't specify the period/up-low time nor the number of pins, so I assume that you have four buttons connected to arduino pins 1,2,3,5 and I'll use literals.
You'll have to set CHECK_PERIOD to an appropriate sampling period so that you can sample the input 4/5 times for every state, and CHECK_ITERATIONS so that you can also miss some points.
For instance if the normal wave is 100ms high and 100ms low, I'd set CHECK_PERIOD to 20 and CHECK_ITERATIONS to, let's say, 3 or 4.
long previousInputCheck;
#define NUM_INPUTS 4
const int inputPins[] = { 1, 2, 3, 5}
unsigned char inputCounter[NUM_INPUTS];
unsigned char inputStates[NUM_INPUTS];
... THEN, INTO THE MAIN ...
if ((millis() - previousInputCheck) >= CHECK_PERIOD)
{
previousInputCheck += CHECK_PERIOD;
unsigned char i;
for (i = 0; i < NUM_INPUTS; i++)
{
if (digitalRead(inputPins[i]) == LOW)
{
if (inputCounter[i] <= CHECK_ITERATIONS)
inputCounter[i]++;
if (inputCounter[i] == CHECK_ITERATIONS)
{
inputStates[i] = LOW;
}
}
else
{ // HIGH
inputCounter[i] = 0;
inputStates[i] = HIGH;
}
}
}

Varispeed with Libsndfile, Libsamplerate and Portaudio in C

I'm working on an audio visualizer in C with OpenGL, Libsamplerate, portaudio, and libsndfile. I'm having difficulty using src_process correctly within my whole paradigm. My goal is to use src_process to achieve Vinyl Like varispeed in real time within the visualizer. Right now my implementation changes the pitch of the audio without changing the speed. It does so with lots of distortion due to what sounds like missing frames as when I lower the speed with the src_ratio it almost sounds granular like chopped up samples. Any help would be appreciated, I keep experimenting with my buffering chunks however 9 times out of 10 I get a libsamplerate error saying my input and output arrays are overlapping. I've also been looking at the speed change example that came with libsamplerate and I can't find where I went wrong. Any help would be appreciated.
Here's the code I believe is relevant. Thanks and let me know if I can be more specific, this semester was my first experience in C and programming.
#define FRAMES_PER_BUFFER 1024
#define ITEMS_PER_BUFFER (FRAMES_PER_BUFFER * 2)
float src_inBuffer[ITEMS_PER_BUFFER];
float src_outBuffer[ITEMS_PER_BUFFER];
void initialize_SRC_DATA()
{
data.src_ratio = 1; //Sets Default Playback Speed
/*---------------*/
data.src_data.data_in = data.src_inBuffer; //Point to SRC inBuffer
data.src_data.data_out = data.src_outBuffer; //Point to SRC OutBuffer
data.src_data.input_frames = 0; //Start with Zero to Force Load
data.src_data.output_frames = ITEMS_PER_BUFFER
/ data.sfinfo1.channels; //Number of Frames to Write Out
data.src_data.src_ratio = data.src_ratio; //Sets Default Playback Speed
}
/* Open audio stream */
err = Pa_OpenStream( &g_stream,
NULL,
&outputParameters,
data.sfinfo1.samplerate,
FRAMES_PER_BUFFER,
paNoFlag,
paCallback,
&data );
/* Read FramesPerBuffer Amount of Data from inFile into buffer[] */
numberOfFrames = sf_readf_float(data->inFile, data->src_inBuffer, framesPerBuffer);
/* Looping of inFile if EOF is Reached */
if (numberOfFrames < framesPerBuffer)
{
sf_seek(data->inFile, 0, SEEK_SET);
numberOfFrames = sf_readf_float(data->inFile,
data->src_inBuffer+(numberOfFrames*data->sfinfo1.channels),
framesPerBuffer-numberOfFrames);
}
/* Inform SRC Data How Many Input Frames To Process */
data->src_data.end_of_input = 0;
data->src_data.input_frames = numberOfFrames;
/* Perform SRC Modulation, Processed Samples are in src_outBuffer[] */
if ((data->src_error = src_process (data->src_state, &data->src_data))) {
printf ("\nError : %s\n\n", src_strerror (data->src_error)) ;
exit (1);
}
* Write Processed SRC Data to Audio Out and Visual Out */
for (i = 0; i < framesPerBuffer * data->sfinfo1.channels; i++)
{
// gl_audioBuffer[i] = data->src_outBuffer[i] * data->amplitude;
out[i] = data->src_outBuffer[i] * data->amplitude;
}
I figured out a solution that works well enough for me and am just going to explain it best I can for anyone else with a similar issue. So to get the Varispeed to work, the way the API works is you give it a certain number of frames, and it spits out a certain number of frames. So for a SRC ratio of 0.5, if you process 512 frames per loop you are feeding in 512/0.5 frames = 1024 frames. That way when the API runs its src_process function, it compresses those 1024 frames into 512, speeding up the samples. So I dont fully understand why it solved my issue, but the problem was if the ratio is say 0.7, you end up with a float number which doesn't work with the arrays indexed int values. Therefore there's missing samples unless the src ratio is eqaully divisble by the framesperbuffer potentially at the end of each block. So what I did was add +2 frames to be read if the framesperbuffer%src.ratio != 0 and it seemed to fix 99% of the glitches.
/* This if Statement Ensures Smooth VariSpeed Output */
if (fmod((double)framesPerBuffer, data->src_data.src_ratio) == 0)
{
numInFrames = framesPerBuffer;
}
else
numInFrames = (framesPerBuffer/data->src_data.src_ratio) + 2;
/* Read FramesPerBuffer Amount of Data from inFile into buffer[] */
numberOfFrames = sf_readf_float(data->inFile, data->src_inBuffer, numInFrames);

ios audio queue - how to meter audio level in buffer?

I'm working on an app that should do some audio signal processing. I need to measure the audio level in each one of the buffers I get (through the Callback function). I've been searching the web for some time, and I found that there is a build-in property called Current level metering:
AudioQueueGetProperty(recordState->queue,kAudioQueueProperty_CurrentLevelMeter,meters,&dlen);
This property gets me the average or peak audio level, but it's not synchronised to the current buffer.
I figured out I need to calculate the audio level from the buffer data by myself, so I had this:
double calcAudioRMS (SInt16 * audioData, int numOfSamples)
{
double RMS, adPercent;
RMS = 0;
for (int i=0; i<numOfSamples; i++)
{
adPercent=audioData[i]/32768.0f;
RMS += adPercent*adPercent;
}
RMS = sqrt(RMS / numOfSamples);
return RMS;
}
This function gets the audio data (casted into Sint16) and the number of samples in the current buffer. The numbers I get are indeed between 0 and 1, but they seem to be rather random and low comparing to the numbers I got from the built-in audio level metering.
The recording audio format is:
format->mSampleRate = 8000.0;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1;
format->mChannelsPerFrame = 1;
format->mBytesPerFrame = 2;
format->mBytesPerPacket = 2;
format->mBitsPerChannel = 16;
format->mReserved = 0;
format->mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |kLinearPCMFormatFlagIsPacked;
My question is how to get the right values from the buffer? Is there a built-in function \ property for this? Or should I calculate the audio level myself, and how to do it?
Thanks in advance.
Your calculation for RMS power is correct. I'd be inclined to say that you have a fewer number of samples than Apple does, or something similar, and that would explain the difference. You can check by inputting a loud sine wave, and checking that Apple (and you) calculate RMS power at 1/sqrt(2).
Unless there's a good reason, I would use Apple's power calculations. I've used them, and they seem good to me. Additionally, generally you don't want RMS power, you want RMS power as decibels, or use the kAudioQueueProperty_CurrentLevelMeterDB constant. (This depends on if you're trying to build an audio meter, or truly display the audio power)

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