This is a question I've been trying to get an answer for some time.
Is there any limit to the number of RFNoC blocks that can be added to a x310 USRP?
To cite the source code you'd need to be modifying to add your own blocks:
https://github.com/EttusResearch/fpga/blob/maint/usrp3/top/x300/rfnoc_ce_auto_inst_x310.v#L1:
localparam NUM_CE = 10; // Must be no more than 10 (6 ports taken by transport and IO connected CEs)
You can't have more than ten blocks.
Related
Title may be wildly incorrect for what I'm trying to work out.
I'm trying to interpret packets I am recieving from a racing game in a way that I understand, but I honestly don't really know what I'm looking at, or what to search to understand it.
Information on the packets I am recieving here:
https://forums.codemasters.com/topic/54423-f1%C2%AE-2020-udp-specification/?tab=comments#comment-532560
I'm using python to print the packets, here's a snippet of the output, which I don't understand how to interpret.
received message: b'\xe4\x07\x01\x03\x01\x07O\x90.\xea\xc2!7\x16\xa5\xbb\x02C\xda\n\x00\x00\x00\xff\x01\x00\x03:\x00\x00\x00 A\x00\x00\xdcB\xb5+\xc1#\xc82\xcc\x10\t\x00\xd9\x00\x00\x00\x00\x00\x12\x10\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00$tJ\x03\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x01
I'm very new to coding, and not sure what my next step is, so a nudge in the right direction will help loads, thanks.
This is the python code:
import socket
UDP_IP = "127.0.0.1"
UDP_PORT = 20777
sock = socket.socket(socket.AF_INET, # Internet
socket.SOCK_DGRAM) # UDP
sock.bind((UDP_IP, UDP_PORT))
while True:
data, addr = sock.recvfrom(4096)
print ("received message:", data)
The website you link to is describing the data format. All data represented as a series of 1's and 0's. A byte is a series of 8 1's and 0's. However, just because you have a series of bytes doesn't mean you know how to interpret them. Do they represent a character? An integer? Can that integer be negative? All of that is defined by whoever crafted the data in the first place.
The type descriptions you see at the top are telling you how to actually interpret that series of 1's and 0's. When you see "unit8", that is an "unsigned integer that is 8 bits (1 byte) long". In other words, a positive number between 0 and 255. An "int8" on the other hand is an "8-bit integer", or a number that can be positive or negative (so the range is -128 to 127). The same basic idea applies to the *16 and *64 variants, just with 16 bits or 64 bits. A float represent a floating point number (a number with a fractional part, such as 1.2345), generally 4 bytes long. Additionally, you need to know the order to interpret the bytes within a word (left-to-right or right-to-left). This is referred to as the endianness, and every computer architecture has a native endianness (big-endian or little-endian).
Given all of that, you can interpret the PacketHeader. The easiest way is probably to use the struct package in Python. Details can be found here:
https://docs.python.org/3/library/struct.html
As a proof of concept, the following will interpret the first 24 bytes:
import struct
data = b'\xe4\x07\x01\x03\x01\x07O\x90.\xea\xc2!7\x16\xa5\xbb\x02C\xda\n\x00\x00\x00\xff\x01\x00\x03:\x00\x00\x00 A\x00\x00\xdcB\xb5+\xc1#\xc82\xcc\x10\t\x00\xd9\x00\x00\x00\x00\x00\x12\x10\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00$tJ\x03\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x01'
#Note that I am only taking the first 24 bytes. You must pass data that is
#the appropriate length to the unpack function. We don't know what everything
#else is until after we parse out the header
header = struct.unpack('<HBBBBQfIBB', data[:24])
print(header)
You basically want to read the first 24 bytes to get the header of the message. From there, you need to use the m_packetId field to determine what the rest of the message is. As an example, this particular packet has a packetId of 7, which is a "Car Status" packet. So you would look at the packing format for the struct CarStatus further down on that page to figure out how to interpret the rest of the message. Rinse and repeat as data arrives.
Update: In the format string, the < tells you to interpret the bytes as little-endian with no alignment (based on the fact that the documentation says it is little-endian and packed). I would recommend reading through the entire section on Format Characters in the documentation above to fully understand what all is happening regarding alignment, but in a nutshell it will try to align those bytes with their representation in memory, which may not match exactly the format you specify. In this case, HBBBBQ takes up 2 bytes more than you'd expect. This is because your computer will try to pack structs in memory so that they are word-aligned. Your computer architecture determines the word alignment (on a 64-bit computer, words are 64-bits, or 8 bytes, long). A Q takes a full word, so the packer will try to align everything before the Q to a word. However, HBBBB only requires 6 bytes; so, Python will, by default, pad an extra 2 bytes to make sure everything lines up. Using < at the front both ensures that the bytes will be interpreted in the correct order, and that it won't try to align the bytes.
Just for information if someone else is looking for this. In python there is the library f1-2019-telemetry existing. On the documentation, there is a missing part about the "how to use" so here is a snippet:
from f1_2020_telemetry.packets import *
...
udp_socket = socket.socket(family=socket.AF_INET, type=socket.SOCK_DGRAM)
udp_socket.bind((host, port))
while True:
udp_packet = udp_socket.recv(2048)
packet = unpack_udp_packet(udp_packet)
if isinstance(packet, PacketSessionData_V1): # refer to doc for classes / attribute
print(packet.trackTemperature) # for example
if isinstance(packet, PacketParticipantsData_V1):
for i, participant in enumerate(packet.participants):
print(DriverIDs[participant.driverId]) # the library has some mapping for pilot name / track name / ...
Regards,
Nicolas
I have a project where I send data to an Android phone. I get data via the serial port to the rfduino and then send this data to the phone.
First I had 20 byte data chunks. I use "Serial Event" to set a flag and then read and send the data within the main loop. The serial baud rate is 9600 as recommended by the rfduino staff.
Now I have 40 bytes of data. The Rfduino reads 40 bytes of serial data and sends it in two parts (see code). The first byte of each part is an identifier used to distunguish between the different packets.
After some successful transmissions the data becomes corrupt. Usually, once the transmission starts being faulty, I have to reinitiate the connection to get correct data.
This is a basic idea of the code:
void loop() {
if (mData)
{
mData = false;
Serial.readBytes(data, 40);
while(!RFduinoBLE.send(&data[0], 20));
while(!RFduinoBLE.send(&data[20], 20));
}
}
void serialEvent(void) {
mData = true;
}
I checked the serial data with a logic analyzer and there is nothing wrong with it. I also used wireshark to check the ble transmissions. Both 20 Byte chunks are send within the same connection interval. The data sent by the RFduino becomes corrupt after some time.
When I omit the second packet and just send 20 Bytes, the problem does not occur.
I assume, that the radio interferes with the reading of the serial port. I tried to use while(RFduinoBLE.radioActive); before the Serial.read, but I think the data is buffered before that. So this did not change anything.
Additionally I tried to lower the sending power to minimum, without any improvements.
I also tried different connection intervals. I have new data every 128ms. This limits the max. connection interval. All changes did not help at all.
I've read that the BLE radio priority takes about 5-6ms and that there is a 6 byte buffer for data. Using 9600 baud this buffer overflows.
I haven't looked for sources to those statements yet, but lowering the baud rate to 4800 seems to improve the issue with the faulty data.
Still this is no satisfying data rate and the transmission faults still occur. Not that often anymore, but still.
I've run out of ideas how to fix this and would appreciate any thought!
I mean it must be possible to send 40 bytes every 128ms... that's not that much.
I already posted this question in the RFduino Forum - without answer until now - and I wonder if anyone experienced similar issues with the RFduino.
RFduino Forum
I'm experimenting with 2 Gluster 3.7 servers in 1x2 configuration. Servers are connected over 1 Gbit network. I'm using Debian Jessie.
My use case is as follows: open file -> append 64 bytes -> close file and do this in a loop for about 5000 different files. Execution time for such loop is roughly 10 seconds if I access files through mounted glusterfs drive. If I use libgfsapi directly, execution time is about 5 seconds (2 times faster).
However, the same loop executes in 50ms on plain ext4 disk.
There is huge performance difference between Gluster 3.7 end earlier versions which is, I believe, due to the cluster.eager-lock setting.
My target is to execute the loop in less than 1 second.
I've tried to experiment with lots of Gluster settings but without success. dd tests with various bsize values behave like that TCP no-delay option is not set, although from Gluster source code it seems that no-delay is default.
Any idea how to improve the performance?
Edit:
I've found a solution that works in my case so I'd like to share it in case anyone else faces the same issue.
The root cause of the problem is the number of roundtrips between client and Gluster server during execution of open/write/close sequence. I don't know exactly what is happening behind but timing measurements shows exactly that pattern. Now, the obvious idea would be to "pack" open/write/close sequence into a single write function. Roughly, the C prototype of such function would be:
int write(const char* fname, const void *buf, size_t nbyte, off_t offset)
But, there is already such API function glfs_h_anonymous_write in libgfapi (thanks goes to Suomya from Gluster mailing group). Kind of hidden thing there is the file identifier which is not plain file name, but something of type struct glfs_object. Clients obtain an instance of such object through API calls glfs_h_lookupat/glfs_h_creat. The point here is that glfs_object representing filename is "stateless" in a sense that corresponding inode is left intact (not ref counted). One should think of glfs_object as plain filename identifier and use it as you would use filename (actually, glfs_object stores plain pointer to corresponding inode without ref counting it).
Finally, we should use glfs_h_lookupat/glfs_h_creat once and write many times to the file using glfs_h_anonymous_write.
That way I was able to append 64 bytes to 5000 files in 0.5 seconds, which is 20 times faster than using mounted volume and open//write/close sequence.
I need a step by step walkthrough on how to use audioConverterFillComplexBuffer and its callback. No, don't tell me to read the Apple docs. I do everything they say and the conversion always fails. No, don't tell me to go look for examples of audioConverterFillComplexBuffer and its callback in use - I've duplicated about a dozen such examples both line for line and modified and the conversion always fails. No, there isn't any problem with the input data. No, it isn't an endian issue. No, the problem isn't my version of OS X.
The problem is that I don't understand how audioConverterFillComplexBuffer works, so I don't know what I'm doing wrong. And nothing out there is helping me understand, because it seems like nobody on Earth really understands how audioConverterFillComplexBuffer works, either. From the people who actually use it(I spy cargo cult programming in their code) to even the authors of Learning Core Audio and/or Apple itself(http://stackoverflow.com/questions/13604612/core-audio-how-can-one-packet-one-byte-when-clearly-one-packet-4-bytes).
This isn't just a problem for me, it's a problem for anybody who wants to program high-performance audio on the Mac platform. Threadbare documentation that's apparently wrong and examples that don't work are no fun.
Once again, to be clear: I NEED A STEP BY STEP WALKTHROUGH ON HOW TO USE audioConverterFillComplexBuffer plus its callback and so does the entire Mac developer community.
This is a very old question but I think is still relevant. I've spent a few days fighting this and have finally achieved a successful conversion. I'm certainly no expert but I'll outline my understanding of how it works. Note I'm using Swift, which I'm also just learning.
Here are the main function arguments:
inAudioConverter: AudioConverterRef: This one is simple enough, just pass in a previously created AudioConverterRef.
inInputDataProc: AudioConverterComplexInputDataProc: The very complex callback. We'll come back to this.
inInputDataProcUserData, UnsafeMutableRawPointer?: This is a reference to whatever data you may need to be provided to the callback function. Important because even in swift the callback can't inherit context. E.g. you may need to access an AudioFileID or keep track of the number of packets read so far.
ioOutputDataPacketSize: UnsafeMutablePointer<UInt32>: This one is a little misleading. The name implies it's the packet size but reading the documentation we learn it's the total number of packets expected for the output format. You can calculate this as outPacketCount = frameCount / outStreamDescription.mFramesPerPacket.
outOutputData: UnsafeMutablePointer<AudioBufferList>: This is an audio buffer list which you need to have already initialized with enough space to hold the expected output data. The size can be calculated as byteSize = outPacketCount * outMaxPacketSize.
outPacketDescription: UnsafeMutablePointer<AudioStreamPacketDescription>?: This is optional. If you need packet descriptions, pass in a block of memory the size of outPacketCount * sizeof(AudioStreamPacketDescription).
As the converter runs it will repeatedly call the callback function to request more data to convert. The main job of the callback is simply to read the requested number packets from the source data. The converter will then convert the packets to the output format and fill the output buffer. Here are the arguments for the callback:
inAudioConverter: AudioConverterRef: The audio converter again. You probably won't need to use this.
ioNumberDataPackets: UnsafeMutablePointer<UInt32>: The number of packets to read. After reading, you must set this to the number of packets actually read (which may be less than the number requested if we reached the end).
ioData: UnsafeMutablePointer<AudioBufferList>: An AudioBufferList which is already configured except for the actual data. You need to initialise ioData.mBuffers.mData with enough capacity to hold the expected number of packets, i.e. ioNumberDataPackets * inMaxPacketSize. Set the value of ioData.mBuffers.mDataByteSize to match.
outDataPacketDescription: UnsafeMutablePointer<UnsafeMutablePointer<AudioStreamPacketDescription>?>?: Depending on the formats used, the converter may need to keep track of packet descriptions. You need to initialise this with enough capacity to hold the expected number of packet descriptions.
inUserData: UnsafeMutableRawPointer?: The user data that you provided to the converter.
So, to start you need to:
Have sufficient information about your input and output data, namely the number of frames and maximum packet sizes.
Initialise an AudioBufferList with sufficient capacity to hold the output data.
Call AudioConverterFillComplexBuffer.
And on each run of the callback you need to:
Initialise ioData with sufficient capacity to store ioNumberDataPackets of source data.
Initialise outDataPacketDescription with sufficient capacity to store ioNumberDataPackets of AudioStreamPacketDescriptions.
Fill the buffer with source packets.
Write the packet descriptions.
Set ioNumberDataPackets to the number of packets actually read.
return noErr if successful.
Here's an example where I read the data from an AudioFileID:
var converter: AudioConverterRef?
// User data holds an AudioFileID, input max packet size, and a count of packets read
var uData = (fRef, maxPacketSize, UnsafeMutablePointer<Int64>.allocate(capacity: 1))
err = AudioConverterNew(&inStreamDesc, &outStreamDesc, &converter)
err = AudioConverterFillComplexBuffer(converter!, { _, ioNumberDataPackets, ioData, outDataPacketDescription, inUserData in
let uData = inUserData!.load(as: (AudioFileID, UInt32, UnsafeMutablePointer<Int64>).self)
ioData.pointee.mBuffers.mDataByteSize = uData.1
ioData.pointee.mBuffers.mData = UnsafeMutableRawPointer.allocate(byteCount: Int(uData.1), alignment: 1)
outDataPacketDescription?.pointee = UnsafeMutablePointer<AudioStreamPacketDescription>.allocate(capacity: Int(ioNumberDataPackets.pointee))
let err = AudioFileReadPacketData(uData.0, false, &ioData.pointee.mBuffers.mDataByteSize, outDataPacketDescription?.pointee, uData.2.pointee, ioNumberDataPackets, ioData.pointee.mBuffers.mData)
uData.2.pointee += Int64(ioNumberDataPackets.pointee)
return err
}, &uData, &numPackets, &bufferList, nil)
Again, I'm no expert, this is just what I've learned by trial and error.
HI all,
I am doing serial port communication program. How do I achieve the following.
Need to know number bytes available for reading.
Flushing
Note: I am creating File with Overlapped option.
thanks in advance
~ Johnnie
You are trying to query the number of bytes available first, and then read them. The standard way would be to just allocate a buffer (say 1000 chars), then call ReadComm() which tells you how many bytes were actually used (e.g. less than or equal to 1000).
You can flush the buffer of serial io using FlushFileBuffers() (http://msdn.microsoft.com/en-us/library/aa364439%28VS.85%29.aspx) but since you want asynchronous IO, you probably only want to do that when you have written to a file and then want to move the file (certainly not on every call to WriteComm()).
More info:
http://msdn.microsoft.com/en-us/library/ms810467.aspx