I have an Arduino with a WiFly shield, everything works perfectly!
The thing is, when I want to turn on an LED, I open in my
webbrowser:
192.168.1.120/ledon/
(I made a program which handles this URL).
But the thing is; when I make a request, I must wait 1-2 seconds before I can do another one.
So, it is very long, and if I want to control motors, it is just too long.
So, instead of using an HTTP request, I want to use something else which can be faster.
Something "super fast".
I just need to tell the Arduino:
- go direction 1
- go direction 2...
- turn on LED
- turn off LED
- tell me the light level (which return a int)
So it is just about a small amount of data.
Can you show me a way? (Telnet, UDP, OSC?)
For your arduino, have a look at just using sockets or even encoding the data in the URL requested.
You shouldnt get less than about 0.8 Seconds Lag maximum.
How big is your program for handling the Url /ledon/ ?
Using pure packets (usually TCP) from your computer to the arduino is faster sometimes..
But you may need to code a application to handle the packets on the pc.
There is the option of Javascript to parse data back and forth e.g. reading the light level and such.
Related
i know this forum dislikes "open" questions like this, nevertheless i'd like somebody to help untie the knot in my head, much appreciated.
The goal is simple:
read a stereo 32bit 44100 S/s I2S signal from 2 adafruit sph0645 mics
create a wav-header and store the data onto an SD-card
I've been at this for a few days now and i know that this will be much more complicated than i originally thought. Main reason: signal quality. Like most tutorials on this subject the simplest "hello world" for these mics is a looped polling for I2S-samples. Poll, fill buffer, output via serial or write to SD-card. This returns a choppy, noisy, sped up version of RL-audio. The filling of the internal DMA-buffers can be seen as constant, but the rest is mostly chaos, so
how to i sync these DMA-buffers with the rest of my code?
From experience with the STM32 HAL i'd imagine some register which can be set to throw an interrupt whenever a buffer is full, or an event which can be sent between tasks via queues. Examples on this subject either poll in a main loop with mono an abysmal sample-rate and bit depth or use pages of overkill code and never adress what it does, "just copy and it works", not good. Does the ESP32-Arduino framework provide some way to to this properly? The espressif-documentation isn't something to look forward to, since some of their I2S interface functions don't even work (if you are researching this topic as well, you too might have noticed that i2s_read only returns zeros). Just a hint into the right direction would help, i'm writing my own code anyway. Interrupts? Events? Timers? Polling for full buffers? Only you might know.
have a good one, thx
Thanks to https://github.com/atomic14/ i now have an answer for a syncing-method which works very well. This method has been tried by https://esp32.com/viewtopic.php?t=12546 who also didn't fully understand what was going on: the espressif i2s-interface offers a flag stored in an event which is triggererd every time one of the specified dma-buffers has received a full set of data, ergo, is full. It looks like this:
while(<your condition>){
i2s_event_t evt;
if (xQueueReceive(<your queue>, &evt, portMAX_DELAY) == pdPASS){
if (evt.type == I2S_EVENT_RX_DONE){
size_t bytesRead = 0;
do{
//read data via i2s_read or i2s_read_bytes
} while (bytesRead > 0);
No data is stored in this queue, but rather a flag which can then be used to synchronize dma-filling and further buffering/calculating/sending the read data.
HOWEVER this only works if you install the i2s driver in a specific setup. Instead of using
i2s_driver_install(I2S_NUM_0, &i2s_config, 0, NULL);
in your setup, you can activate the "affinity" for events by passing a queue-handle and a lenght:
i2s_driver_install(I2S_NUM_0, &i2s_config, 4, &<your queue>);
hope this helps getting started, it sure did help me.
I'm trying to filter a signal and then analyse the values of the filtered signal using Tone.js / Web-Audio API.
I'm expecting to get values of the filtered signal, but I only get -Infinity, meaning that my connections between the nodes are wrong. I've made a small fiddle demonstrating this, however in my use-case I do not want to send this node to the destination of the context - I only want to analyse it, not hear it.
osc.connect(filter)
filter.connect(gainNode)
gainNode.connect(meter)
console.log(meter.getLevel())
I guess you tested the code in Chrome because there is a problem with Chrome which causes it to not process anything until it is connected to the destination. When using Tone.js that means you need to call .toMaster() at the end of your chain. I updated you fiddle to make it work: https://jsfiddle.net/8f7abzoL/.
In Firefox calling .toMaster() is not necessary therefore the following works in Firefox as well: https://jsfiddle.net/yrjgfdtz/.
After some digging I've found out that I need to have a scriptProcessorNode - which is apparently no longer recommended - so looking into Audio Worklet Nodes
I have two xbee's series 1. I have them as endpoint devices working in API mode and talking to each other. The first xbee is attached at a raspberry pi, while the other is on my pc where I see the terminal tab of XCTU program. The baud rate I use is 125000.
From raspberry pi I try to send a jpg image which is 30Kbytes. I send data frames 100 byte long (the biggest as it is said in the xbee documentation). Inside a loop I create and send the packets, I have also a cout statement that prints the loop number. Everything is fine and all bytes are sent. When I comment out the cout statement not all bytes are sent.
From what I have understood the cout statement works as a delay between packets, but I still cannot understand why is this happening as it is supposed that I use the half speed ...
I hope I was clear and look forward for a reply.
UPDATE
Just to summarize, i changed baud rate to 250000 where there is the same behavior as in 125000. I also implemented hardware flow control by checking cts signal. When xbees are in transparent mode I need a delay between sending characters at around 150us. The same goes for api mode too. The difference with 125000 baud rate in api mode was that the delay needed, was enough to be betwween each data packet, but in 250000 the delay is needed between each byte that i send. If i do the above everything goes well.
The next thing i did was to plug both xbees in my pc in transparent mode. I went to terminal tab of xctu software where i chose assemble packet and sent at around 3000 bytes to the other xbee. The result was the same. The second xbee received at about 1500 bytes and then each time that i was sending one byte from the first to the second, the "lost bytes" were being received at packets of 1000. :/
So could anyone know what am I doing wrong?
You should connect the /CTS pin from the XBee module into the Raspberry Pi, and have your routine stop sending data when the XBee de-asserts it.
At higher baud rates, it's possible to stream data into the XBee module faster than it can send to the remote module. The local XBee module uses the /CTS pin to notify the host when its buffers are almost full and the host should stop sending. People refer to this as hardware flow control.
It may be necessary to modify the serial driver on the Raspberry Pi to make use of that signal -- it should pause the transmit buffer when de-asserted, and automatically resume sending when re-asserted.
here's what I'm trying to do:
I have 4 clocks and one PC on the network. I (PC) want to get the time from any of the clocks.
The idea:
Every clock uses UDP Broadcast (broadcast because I don't know the PC's IP). The PC gets a first dataset(time) and from then on only looks at datasets from this clock. (To not get confused by slight timedifferences between the clocks)
Clocks=Clients:
- socket
- bind to port 1234
- sendto(broadcast,'1234')
PC=Server
- socket
- bind to port 1234
- recvfrom(data,client_addr) //extract client_addr from first received data
- connect(client_addr) //to only receive data from this client
- recvfrom()
1: Is this the correct way to do this? Or is there a better option?
2: What can I do, if I want more than one application on the PC to get the time-data? Can I just copy the code and use it? My impression was, that port 1234 is now blocked from the first application that gets the bind.
(I guess SO_REUSEADDR won't work because only ONE application then gets the data AND I don't know which one it is going to be)
Regards
1: Is this the correct way to do this?
Yes.
Or is there a better option?
Can't think of one off-hand.
2: What can I do, if I want more than one application on the PC to get the time-data?
Use SO_REUSEADDR.
Can I just copy the code and use it?
Yes.
My impression was, that port 1234 is now blocked from the first application that gets the bind.
No.
(I guess SO_REUSEADDR won't work because only ONE application then gets the data AND I don't know which one it is going to be)
Wrong guess. Why guess about it at all? Why not try it? Much more reliable than guessing, and quicker than asking questions here too.
I'm writing a little VOIP app like Skype, which works quite good right now, but I've run into a very strange problem.
In one thread, I'm calling within a while(true) loop the winsock recv() function twice per run to get data from a socket.
The first call gets 2 bytes which will be casted into a (short) while the second call gets the rest of the message which looks like:
Complete Message: [2 Byte Header | Message, length determined by the 2Byte Header]
These packets are round about 49/sec which will be round about 3000bytes/sec.
The content of these packets is audio-data that gets converted into wave.
With ioctlsocket() I determine wether there is some data on the socket or not at each "message" I receive (2byte+data). If there's something on the socket right after I received a message within the while(true) loop of the thread, the message will be received, but thrown away to work against upstacking latency.
This concept works very well, but here's the problem:
While my VOIP program is running and when I parallely download (e.g. via browser) a file, there always gets too much data stacked on the socket, because while downloading, the recv() loop seems actually to slow down. This happens in every download/upload situation besides the actual voip up/download.
I don't know where this behaviour comes from, but when I actually cancel every up/download besides the voip traffic of my application, my apps works again perfectly.
If the program runs perfectly, the ioctlsocket() function writes 0 into the bytesLeft var, defined within the class where the receive function comes from.
Does somebody know where this comes from? I'll attach my receive function down below:
std::string D_SOCKETS::receive_message(){
recv(ClientSocket,(char*)&val,sizeof(val),MSG_WAITALL);
receivedBytes = recv(ClientSocket,buffer,val,MSG_WAITALL);
if (receivedBytes != val){
printf("SHORT: %d PAKET: %d ERROR: %d",val,receivedBytes,WSAGetLastError());
exit(128);
}
ioctlsocket(ClientSocket,FIONREAD,&bytesLeft);
cout<<"Bytes left on the Socket:"<<bytesLeft<<endl;
if(bytesLeft>20)
{
// message gets received, but ignored/thrown away to throw away
return std::string();
}
else
return std::string(buffer,receivedBytes);}
There is no need to use ioctlsocket() to discard data. That would indicate a bug in your protocol design. Assuming you are using TCP (you did not say), there should not be any left over data if your 2byte header is always accurate. After reading the 2byte header and then reading the specified number of bytes, the next bytes you receive after that constitute your next message and should not be discarded simply because it exists.
The fact that ioctlsocket() reports more bytes available means that you are receiving messages faster than you are reading them from the socket. Make your reading code run faster, don't throw away good data due to your slowness.
Your reading model is not efficient. Instead of reading 2 bytes, then X bytes, then 2 bytes, and so on, you should instead use a larger buffer to read more raw data from the socket at one time (use ioctlsocket() to know how many bytes are available, and then read at least that many bytes at one time and append them to the end of your buffer), and then parse as many complete messages are in the buffer before then reading more raw data from the socket again. The more data you can read at a time, the faster you can receive data.
To help speed up the code even more, don't process the messages inside the loop directly, either. Do the processing in another thread instead. Have the reading loop put complete messages in a queue and go back to reading, and then have a processing thread pull from the queue whenever messages are available for processing.