I am trying to implement a UVM Driver for a simple pipelined model using semaphores, fork-join & get()-put() methods in the run_phase of the driver.
The driver part is doing the job fine if only I code the sequence in a particular way. From what I know the body task is coded as below
Code1:
pkt = packet::type_id::create("pkt"); // Factory create the sequence item
for(int i=0;i<num_trans;i++) // Repeat as required
begin
assert(pkt.randomize()); // Randomize the sequence item
start_item(pkt); //Send the request to Driver.
finish_item(pkt); //Wait for the driver to finish the current item
Above style, there's no pipelining achieved and moreover the data beat corresponding to the first transaction packet is lost. When the randomization is invoked after start_item, the test bench works as expected.
Code2:
pkt = packet::type_id::create("pkt");
for(int i=0;i<num_trans;i++)
begin
start_item(pkt);
assert(pkt.randomize());
finish_item(pkt);
I'd like to know what is the difference between coding style 1 and 2
This might be happening because on the task start_item() task we are waiting for the following.
sequencer.wait_for_grant(this, set_priority);
so we are waiting for the sequencer to grant the sequence and then sequence_item will be taken, but if you do like the following.
assert(pkt.randomize()); // Randomize the sequence item
start_item(pkt); //Send the request to Driver.
that randomization lost because that start_item might be waiting for the sequencer to be free and till that time we lost the randomization.
further you can read the following article, that might help https://verificationacademy.com/forums/uvm/startitem/finishitem-versus-uvmdo-macros
Related
I am a newbie in CANOPEN. I wrote a program that read actual position via PDO1 (default is statusword + actual position).
void canopen_init() {
// code1 setup PDO mapping
nmtPreOperation();
disablePDO(PDO_TX1_CONFIG_COMM);
setTransmissionTypePDO(PDO_TX1_CONFIG_COMM, 1);
setInhibitTimePDO(PDO_TX1_CONFIG_COMM, 0);
setEventTimePDO(PDO_TX1_CONFIG_COMM, 0);
enablePDO(PDO_TX1_CONFIG_COMM);
setCyclePeriod(1000);
setSyncWindow(100);
//code 2: enable OPeration
readyToSwitchOn();
switchOn();
enableOperation();
motionStart();
// code 3
nmtActiveNode();
}
int main (void) {
canopen_init();
while {
delay_ms(1);
send_sync();
}
}
If I remove "code 2" (the servo is in Switch_on_disable status), i can read position each time sync send. But if i use "code 2", the driver has error "sync frame timeout". I dont know driver has problem or my code has problem. Does my code has problem? thank you!
I don't know what protocol stack this is or how it works, but these:
setCyclePeriod(1000);
setSyncWindow(100);
likely correspond to these OD entries :
Object 1006h: Communication cycle period (CiA 301 7.5.2.6)
Object 1007h: Synchronous window length (CiA 301 7.5.2.7)
They set the SYNC interval and time window for synchronous PDOs respectively. The latter is described by the standard as:
If the synchronous window length expires all synchronous TPDOs may be discarded and an EMCY message may be transmitted; all synchronous RPDOs may be discarded until the next SYNC message is received. Synchronous RPDO processing is resumed with the next SYNC message.
Now if you set this sync time window to 100us but have a sloppy busy-wait delay delay_ms(1), then that doesn't add up. If you write zero to Object 1007h, you disable the sync window feature. I suppose setSyncWindow(0); might do that. You can try to do that to see if that's the issue. If so, you have to drop your busy-wait in favour for proper hardware timers, one for the SYNC period and one for PDO timeout (if you must use that feature).
Problem fixed. Due to much EMI from servo, that make my controller didn't work properly. After isolating, it worked very well :)!
I want to create a realtime sine generator using apples core audio framework. I want do do it low level so I can learn and understand the fundamentals.
I know that using PortAudio or Jack would probably be easier and I will use them at some point but I would like to get this to work first so I can be confident to understand the fundamentals.
I literally searched for days now on this topic but no one seems to have ever created a real time wave generator using core audio trying to optain low latency while using C and not Swift or Objective-C.
For this I use a project I set up a while ago. It was first designed to be a game. So after the Application starts up, it will enter a run loop. I thought this would perfectly fit as I can use the main loop to copy samples into the audio buffer and process rendering and input handling as well.
So far I get sound. Sometimes it works for a while then starts to glitch, sometimes it glitches right away.
This is my code. I tried to simplify if and only present the important parts.
I got multiple questions. They are located in the bottom section of this post.
Applications main run loop. This is where it all starts after the window is created and buffers and memory is initialized:
while (OSXIsGameRunning())
{
OSXProcessPendingMessages(&GameData);
[GlobalGLContext makeCurrentContext];
CGRect WindowFrame = [window frame];
CGRect ContentViewFrame = [[window contentView] frame];
CGPoint MouseLocationInScreen = [NSEvent mouseLocation];
BOOL MouseInWindowFlag = NSPointInRect(MouseLocationInScreen, WindowFrame);
CGPoint MouseLocationInView = {};
if (MouseInWindowFlag)
{
NSRect RectInWindow = [window convertRectFromScreen:NSMakeRect(MouseLocationInScreen.x, MouseLocationInScreen.y, 1, 1)];
NSPoint PointInWindow = RectInWindow.origin;
MouseLocationInView= [[window contentView] convertPoint:PointInWindow fromView:nil];
}
u32 MouseButtonMask = [NSEvent pressedMouseButtons];
OSXProcessFrameAndRunGameLogic(&GameData, ContentViewFrame,
MouseInWindowFlag, MouseLocationInView,
MouseButtonMask);
#if ENGINE_USE_VSYNC
[GlobalGLContext flushBuffer];
#else
glFlush();
#endif
}
Through using VSYNC I can throttle the loop down to 60 FPS. The timing is not super tight but it is quite steady. I also have some code to throttle it manually using mach timing which is even more imprecise. I left it out for readability.
Not using VSYNC or using mach timing to get 60 iterations a second also makes the audio glitch.
Timing log:
CyclesElapsed: 8154360866, TimeElapsed: 0.016624, FPS: 60.155666
CyclesElapsed: 8174382119, TimeElapsed: 0.020021, FPS: 49.946926
CyclesElapsed: 8189041370, TimeElapsed: 0.014659, FPS: 68.216309
CyclesElapsed: 8204363633, TimeElapsed: 0.015322, FPS: 65.264511
CyclesElapsed: 8221230959, TimeElapsed: 0.016867, FPS: 59.286217
CyclesElapsed: 8237971921, TimeElapsed: 0.016741, FPS: 59.733719
CyclesElapsed: 8254861722, TimeElapsed: 0.016890, FPS: 59.207333
CyclesElapsed: 8271667520, TimeElapsed: 0.016806, FPS: 59.503273
CyclesElapsed: 8292434135, TimeElapsed: 0.020767, FPS: 48.154209
What is important here is the function OSXProcessFrameAndRunGameLogic. It is called 60 times a second and it is passed a struct containing basic information like a buffer for rendering, keyboard state and a sound buffer which looks like this:
typedef struct osx_sound_output
{
game_sound_output_buffer SoundBuffer;
u32 SoundBufferSize;
s16* CoreAudioBuffer;
s16* ReadCursor;
s16* WriteCursor;
AudioStreamBasicDescription AudioDescriptor;
AudioUnit AudioUnit;
} osx_sound_output;
Where game_sound_output_buffer is:
typedef struct game_sound_output_buffer
{
real32 tSine;
int SamplesPerSecond;
int SampleCount;
int16 *Samples;
} game_sound_output_buffer;
These get set up before the application enters its run loop.
The size for the SoundBuffer itself is SamplesPerSecond * sizeof(uint16) * 2 where SamplesPerSecond = 48000.
So inside OSXProcessFrameAndRunGameLogic is the sound generation:
void OSXProcessFrameAndRunGameLogic(osx_game_data *GameData, CGRect WindowFrame,
b32 MouseInWindowFlag, CGPoint MouseLocation,
int MouseButtonMask)
{
GameData->SoundOutput.SoundBuffer.SampleCount = GameData->SoundOutput.SoundBuffer.SamplesPerSecond / GameData->TargetFramesPerSecond;
// Oszi 1
OutputTestSineWave(GameData, &GameData->SoundOutput.SoundBuffer, GameData->SynthesizerState.ToneHz);
int16* CurrentSample = GameData->SoundOutput.SoundBuffer.Samples;
for (int i = 0; i < GameData->SoundOutput.SoundBuffer.SampleCount; ++i)
{
*GameData->SoundOutput.WriteCursor++ = *CurrentSample++;
*GameData->SoundOutput.WriteCursor++ = *CurrentSample++;
if ((char*)GameData->SoundOutput.WriteCursor >= ((char*)GameData->SoundOutput.CoreAudioBuffer + GameData->SoundOutput.SoundBufferSize))
{
//printf("Write cursor wrapped!\n");
GameData->SoundOutput.WriteCursor = GameData->SoundOutput.CoreAudioBuffer;
}
}
}
Where OutputTestSineWave is the part where the buffer is actually filled with data:
void OutputTestSineWave(osx_game_data *GameData, game_sound_output_buffer *SoundBuffer, int ToneHz)
{
int16 ToneVolume = 3000;
int WavePeriod = SoundBuffer->SamplesPerSecond/ToneHz;
int16 *SampleOut = SoundBuffer->Samples;
for(int SampleIndex = 0;
SampleIndex < SoundBuffer->SampleCount;
++SampleIndex)
{
real32 SineValue = sinf(SoundBuffer->tSine);
int16 SampleValue = (int16)(SineValue * ToneVolume);
*SampleOut++ = SampleValue;
*SampleOut++ = SampleValue;
SoundBuffer->tSine += Tau32*1.0f/(real32)WavePeriod;
if(SoundBuffer->tSine > Tau32)
{
SoundBuffer->tSine -= Tau32;
}
}
}
So when the Buffers are created at start up also Core audio is initialized which I do like this:
void OSXInitCoreAudio(osx_sound_output* SoundOutput)
{
AudioComponentDescription acd;
acd.componentType = kAudioUnitType_Output;
acd.componentSubType = kAudioUnitSubType_DefaultOutput;
acd.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponent outputComponent = AudioComponentFindNext(NULL, &acd);
AudioComponentInstanceNew(outputComponent, &SoundOutput->AudioUnit);
AudioUnitInitialize(SoundOutput->AudioUnit);
// uint16
//AudioStreamBasicDescription asbd;
SoundOutput->AudioDescriptor.mSampleRate = SoundOutput->SoundBuffer.SamplesPerSecond;
SoundOutput->AudioDescriptor.mFormatID = kAudioFormatLinearPCM;
SoundOutput->AudioDescriptor.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsNonInterleaved | kAudioFormatFlagIsPacked;
SoundOutput->AudioDescriptor.mFramesPerPacket = 1;
SoundOutput->AudioDescriptor.mChannelsPerFrame = 2; // Stereo
SoundOutput->AudioDescriptor.mBitsPerChannel = sizeof(int16) * 8;
SoundOutput->AudioDescriptor.mBytesPerFrame = sizeof(int16); // don't multiply by channel count with non-interleaved!
SoundOutput->AudioDescriptor.mBytesPerPacket = SoundOutput->AudioDescriptor.mFramesPerPacket * SoundOutput->AudioDescriptor.mBytesPerFrame;
AudioUnitSetProperty(SoundOutput->AudioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&SoundOutput->AudioDescriptor,
sizeof(SoundOutput->AudioDescriptor));
AURenderCallbackStruct cb;
cb.inputProc = OSXAudioUnitCallback;
cb.inputProcRefCon = SoundOutput;
AudioUnitSetProperty(SoundOutput->AudioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
0,
&cb,
sizeof(cb));
AudioOutputUnitStart(SoundOutput->AudioUnit);
}
The initialization code for core audio sets the render callback to OSXAudioUnitCallback
OSStatus OSXAudioUnitCallback(void * inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
#pragma unused(ioActionFlags)
#pragma unused(inTimeStamp)
#pragma unused(inBusNumber)
//double currentPhase = *((double*)inRefCon);
osx_sound_output* SoundOutput = ((osx_sound_output*)inRefCon);
if (SoundOutput->ReadCursor == SoundOutput->WriteCursor)
{
SoundOutput->SoundBuffer.SampleCount = 0;
//printf("AudioCallback: No Samples Yet!\n");
}
//printf("AudioCallback: SampleCount = %d\n", SoundOutput->SoundBuffer.SampleCount);
int SampleCount = inNumberFrames;
if (SoundOutput->SoundBuffer.SampleCount < inNumberFrames)
{
SampleCount = SoundOutput->SoundBuffer.SampleCount;
}
int16* outputBufferL = (int16 *)ioData->mBuffers[0].mData;
int16* outputBufferR = (int16 *)ioData->mBuffers[1].mData;
for (UInt32 i = 0; i < SampleCount; ++i)
{
outputBufferL[i] = *SoundOutput->ReadCursor++;
outputBufferR[i] = *SoundOutput->ReadCursor++;
if ((char*)SoundOutput->ReadCursor >= (char*)((char*)SoundOutput->CoreAudioBuffer + SoundOutput->SoundBufferSize))
{
//printf("Callback: Read cursor wrapped!\n");
SoundOutput->ReadCursor = SoundOutput->CoreAudioBuffer;
}
}
for (UInt32 i = SampleCount; i < inNumberFrames; ++i)
{
outputBufferL[i] = 0.0;
outputBufferR[i] = 0.0;
}
return noErr;
}
This is mostly all there is to it. This is quite long but I did not see a way to present all needed information in a more compact way. I wanted to show all because I am by no means a professional programmer. If there is something you feel is missing, pleas tell me.
My feeling tells me that there is something wrong with the timing. I feel the function OSXProcessFrameAndRunGameLogic sometimes needs more time so that the core audio callback is already pulling samples out of the buffer before it is fully written by OutputTestSineWave.
There is actually more stuff going on in OSXProcessFrameAndRunGameLogic which I did not show here. I am "software rendering" very basic stuff into a framebuffer which is then displayed by OpenGL and I also do keypress checks in there because yeah, its the main function of functionality. In the future this is the place where I would like to handle the controls for multiple oscillators, filters and stuff.
Anyway even if I stop the Rendering and Input handling from being called every iteration I still get audio glitches.
I tried pulling all the sound processing in OSXProcessFrameAndRunGameLogic into an own function void* RunSound(void *GameData) and changed it to:
pthread_t soundThread;
pthread_create(&soundThread, NULL, RunSound, GameData);
pthread_join(soundThread, NULL);
However I got mixed results and was not even sure if multithreading is done like that. Creating and destroying threads 60 times a second didn't seem the way to go.
I also had the idea to let sound processing happen on a completely different thread before the application actually runs into the main loop. Something like two simultaneously running while loops where the first processes audio and the latter UI and input.
Questions:
I get glitchy audio. Rendering and input seem to work correctly but audio sometimes glitches, sometimes it doesn't. From the code I provided, can you maybe see me doing something wrong?
Am I using the core audio technology in a wrong way in order to achieve real time low latency signal generation?
Should I do sound processing in a separate thread like I talked about above? How would threading in this context be done correctly? It would make sense to have a thread only dedicated for sound am I right?
Am I right that the basic audio processing should not be done in the render callback of core audio? Is this function only for outputting the provided sound buffer?
And if sound processing should be done right here, how can I access information like the keyboard state from inside the callback?
Are there any resources you could point me to that I maybe missed?
This is the only place I know where I can get help with this project. I would really appreciate your help.
And if something is not clear to you please let me know.
Thank you :)
In general when dealing with low-latency audio you want to achieve the most deterministic behaviour possible.
This, for example, translates to:
Don't hold any locks on the audio thread (priority inversion)
No memory allocation on the audio thread (takes often too much time)
No file/network IO on the audio thread (takes often too much time)
Question 1:
There are indeed some problems with your code for when you want to achieve continuous, realtime, non-glitching audio.
1. Two different clock domains.
You are providing audio data from a (what I call) a different clock domain than the clock domain asking for data. Clock domain 1 in this case is defined by your TargetFramesPerSecond value, clock domain 2 defined by Core Audio. However, due too how scheduling works you have no guarantee that you thread is finishing in time and on time. You try to target your rendering to n frames per second, but what happens when you don't make it time wise? As far as I can see you don't compensate for the deviation a render cycle took compared to the ideal timing.
The way threading works is that ultimately the OS scheduler decides when your thread is active. There are never guarantees and this causes you render cycles to be not very precise (in term of precision you need for audio rendering).
2. There is no synchronisation between the render thread and the Core Audio rendercallback thread.
The thread where the OSXAudioUnitCallback runs is not the same as where your OSXProcessFrameAndRunGameLogic and thus OutputTestSineWave run. You are providing data from your main thread, and data is being read from the Core Audio render thread. Normally you would use some mutexes to protect you data, but in this case that's not possible because you would run into the problem of priority inversion.
A way of dealing with race conditions is to use a buffer which uses atomic variables to store the usage and pointers of the buffer and let only 1 producer and 1 consumer use this buffer.
Good examples of such buffers are:
https://github.com/michaeltyson/TPCircularBuffer
https://github.com/andrewrk/libsoundio/blob/master/src/ring_buffer.h
3. There are a lot of calls in you audio render thread which prevent deterministic behaviour.
As you wrote you are doing a lot more inside the same audio render thread. Changes are quite high that there will be stuff going on (under the hood) which prevents your thread from being on time. Generally, you should avoid calls which take either too much time or are not deterministic. With all the OpenGL/keypres/framebuffer rendering there is no way to be certain you thread will "arrive on time".
Below are some resources worth looking into.
Question 2:
AFAICT generally speaking, you are using the Core Audio technology correctly. The only problem I think you have is on the providing side.
Question 3:
Yes. Definitely! Although, there are multiple ways of doing this.
In your case you have a normal-priority thread running to do the rendering and a high-performance, realtime thread on which the audio render callback is being called. Looking at your code I would suggest putting the generation of the sine wave inside the render callback function (or call OutputTestSineWave from the render callback). This way you have the audio generation running inside a reliable high prio thread, there is no other rendering interfering with the timing precision and there is no need for a ringbuffer.
In other cases where you need to do "non-realtime" processing to get audiodata ready (think of reading from a file, reading from a network or even from another physical audio device) you cannot run this logic inside the Core Audio thread. A way to solve this is to start a separate, dedicated thread to do this processing. To pass the data to the realtime audio thread you would then make use of the earlier mentioned ringbuffer.
It basically boiles down to two simple goals: for the realtime thread it is necessary to have the audio data available at all times (all render calls), if this failes you will end up sending invalid (or better zeroed) audio data.
The main goal for the secondary thread is to fill up the ringbuffer as fast as possible and to keep the ringbuffer as full as possible. So, whenever there is room to put new audio data into the ringbuffer the thread should do just that.
The size of the ringbuffer in this case will dicate how much tolerance there will be for delay. The size of the ringbuffer will be a balance between certainty (bigger buffer) and latency (smaller buffer).
BTW. I'm quite certain Core Audio has all the facilities to do all this for you.
Question 4:
There are multiple ways of achieving you goal, and rendering the stuff inside the render callback from Core Audio is definitely one of them. The one thing you should keep in mind is that you have to make sure the function returns in time.
For changing parameters to manipulate the audio rendering you'll have to find a way of passing messages which enables the reader (audio renderer function) to get messages without locking and waiting. The way I have done this is to create a second ringbuffer which hold messages from which the audio renderer can consume. This can be as simple as a ringbuffer which hold structs with data (or even pointers to data). As long as you stick to the rules of no locking.
Question 5:
I don't know what resources you are aware of but here are some must-reads:
http://atastypixel.com/blog/four-common-mistakes-in-audio-development/
http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-for-nothing
https://developer.apple.com/library/archive/qa/qa1467/_index.html
You basic problem is that you are trying to push audio from your game loop instead of letting the audio system pull it; e.g. instead of always having (or quickly being able to create *) enough audio samples ready for the amount requested by the audio callback to be pulled by the audio callback. The "always" has to account for enough slop to cover timing jitter (being called late or early or too few times) in your game loop.
(* with no locks, semaphores, memory allocation or Objective C messages)
I have an assignment with mpi.
In the assignment a slave must share some data to all the other slaves.
Let tasks = {1,2,3}
Task 1 broadcasts a message to all other tasks using Ibcast with root set to 1.
At the same time, task 2 broadcasts a message to all other slaves using Ibcast with root set to 2.
After a while, all the tasks do
For (i : tasks | where i != myRank)
Use Ibcast to recive message from root I
I Have a code written in c doing the example above.
The code works only if 1 task broadcasts a message,
but if two tasks are broadcasting messages at the same time (before, any other task has received any message)
I’m getting that the request never completes ( using mpi test function on the request handler from ibcast - in the foreach loop).
I’m not sure if I’m doing something wrong or mpi Ibcast does not provide expected behavior in this case.
Thanks
When trying to implement a simple echo server with concurrent support on linux.
Following approaches are used:
Use pthread functions to create a pool of thread, and maintained in a linked list. It's created on process start, and destroy on process termination.
Main thread will accept request, and use a POSIX message queue to store accepted socket file descriptor.
Threads in pool loop to read from message queue, and handle request it gets, when there is no request, it will block.
The program seems working now.
The questions are:
Is it suitable to use message queue in the middle, is it efficient enough?
What is the general approach to accomplish a thread tool that needs to handle concurrent request from multiple clients?
If it's not proper to make threads in pool loop & block to retrieve msg from message queue, then how to deliver requests to threads?
This seems unneccesarily complicated to me. The usual approach for a multithreaded server is:
Create a listen-socket in a thread process
Accept the client-connections in a thread
For each accepted client connection, create a new threads, which receives the corresponding file descriptor and does the work
The worker thread closes the client connection, when it is fully handled
I do not see much benefit in prepopulating a thread-pool here.
If you really want a threadpool:
I would just use a linked list for accepted connections and a pthread_mutex to synchronize access to it:
The listener-process enqueues client fds at the tail of the list.
The clients dequeue it at the head.
If the queue is empty, the thread can wait on a variable (pthread_cond_wait) and are notified by the listener process (pthread_cond_signal) when connections are available.
Another alternative
Depending on the complexity of handling requests, it might be an option to make the server single-threaded, i.e. handle all connections in one thread. This eliminates context-switches altogether and can thus be very performant.
One drawback is, that only one CPU-core is used. To improve that, a hybrid-model can be used:
Create one worker-thread per core.
Each thread handles simultaneously n connections.
You would however have to implement mechanisms to distribute the work fairly amongst the workers.
In addition to using pthread_mutex, you will want to use pthread_cond_t (pthread condition), this will allow you to put the threads in the thread pool to sleep while they are not actually doing work. Otherwise, you will be wasting compute cycles if they are sitting there in a loop checking for something in the work queue.
I would definitely consider using C++ instead of just pure C. The reason I suggest it is that in C++ you are able to use templates. Using a pure virtual base class (lets call it: "vtask"), you can create templated derived classes that accept arguments and insert the arguments when the overloaded operator() is called, allowing for much, much more functionality in your tasks:
//============================================================================//
void* thread_pool::execute_thread()
{
vtask* task = NULL;
while(true)
{
//--------------------------------------------------------------------//
// Try to pick a task
m_task_lock.lock();
//--------------------------------------------------------------------//
// We need to put condition.wait() in a loop for two reasons:
// 1. There can be spurious wake-ups (due to signal/ENITR)
// 2. When mutex is released for waiting, another thread can be waken up
// from a signal/broadcast and that thread can mess up the condition.
// So when the current thread wakes up the condition may no longer be
// actually true!
while ((m_pool_state != state::STOPPED) && (m_main_tasks.empty()))
{
// Wait until there is a task in the queue
// Unlock mutex while wait, then lock it back when signaled
m_task_cond.wait(m_task_lock.base_mutex_ptr());
}
// If the thread was waked to notify process shutdown, return from here
if (m_pool_state == state::STOPPED)
{
//m_has_exited.
m_task_lock.unlock();
//----------------------------------------------------------------//
if(mad::details::allocator_list_tl::get_allocator_list_if_exists() &&
tids.find(CORETHREADSELF()) != tids.end())
mad::details::allocator_list_tl::get_allocator_list()
->Destroy(tids.find(CORETHREADSELF())->second, 1);
//----------------------------------------------------------------//
CORETHREADEXIT(NULL);
}
task = m_main_tasks.front();
m_main_tasks.pop_front();
//--------------------------------------------------------------------//
//run(task);
// Unlock
m_task_lock.unlock();
//--------------------------------------------------------------------//
// execute the task
run(task);
m_task_count -= 1;
m_join_lock.lock();
m_join_cond.signal();
m_join_lock.unlock();
//--------------------------------------------------------------------//
}
return NULL;
}
//============================================================================//
int thread_pool::add_task(vtask* task)
{
#ifndef ENABLE_THREADING
run(task);
return 0;
#endif
if(!is_alive_flag)
{
run(task);
return 0;
}
// do outside of lock because is thread-safe and needs to be updated as
// soon as possible
m_task_count += 1;
m_task_lock.lock();
// if the thread pool hasn't been initialize, initialize it
if(m_pool_state == state::NONINIT)
initialize_threadpool();
// TODO: put a limit on how many tasks can be added at most
m_main_tasks.push_back(task);
// wake up one thread that is waiting for a task to be available
m_task_cond.signal();
m_task_lock.unlock();
return 0;
}
//============================================================================//
void thread_pool::run(vtask*& task)
{
(*task)();
if(task->force_delete())
{
delete task;
task = 0;
} else {
if(task->get() && !task->is_stored_elsewhere())
save_task(task);
else if(!task->is_stored_elsewhere())
{
delete task;
task = 0;
}
}
}
In the above, each created thread runs execute_thread() until the m_pool_state is set to state::STOPPED. You lock the m_task_lock, and if the state is not STOPPED and the list is empty, you pass the m_task_lock to your condition, which puts the thread to sleep and frees the lock. You create the tasks (not shown), add the task (m_task_count is an atomic, by the way, that is why it is thread safe). During the add task, the condition is signaled to wake up a thread, from which the thread proceeds from the m_task_cond.wait(m_task_lock.base_mutex_ptr()) section of execute_thread() after m_task_lock has been acquired and locked.
NOTE: this is a highly customized implementation that wraps most of the pthread functions/objects into C++ classes so copy-and-pasting will not work whatsoever... Sorry. And w.r.t. the thread_pool::run(), unless you are worrying about return values, the (*task)() line is all you need.
I hope this helps.
EDIT: the m_join_* references is for checking whether all the tasks have been completed. The main thread sits in a similar conditioned wait that checks whether all the tasks have been completed as this is necessary for the applications I use this implementation in before proceeding.
I'm trying to acquire data from an MCU, save them to a file and plot them. The code functions properly for some time, then just hangs randomly (sometimes after 1 sec, sometimes after 1 minute ...!). Also the serialport timeouts are not respected, i.e. I'm not receiving any timeout exceptions. I'm using an FTDI232RL chip. The only time I get a timeout exception is when I unplug it while the program is running.
Code:
private: System::Void START_Click(System::Object^ sender, System::EventArgs^ e) {
seconds=0;
minutes=0;
hours=0;
days=0;
t=0;
if((this->comboBox4->Text == String::Empty)||(this->textBox2->Text == String::Empty)||(this->textBox3->Text == String::Empty)){
this->textBox1->Text="please select port, save file directory and logging interval";
timer1->Enabled=false;
}
else{ // start assigning
w=Convert::ToDouble(this->textBox3->Text);
double q=fmod(w*1000,10);
if(q!=0){
MessageBox::Show("The logging interval must be a multiple of 0.01s");
}
else{
period=static_cast<int>(w*1000);
this->interval->Interval = period;
try{ // first make sure port isn't busy/open
if(!this->serialPort1->IsOpen){
// select the port whose name is in comboBox4 (select port)
this->serialPort1->PortName=this->comboBox4->Text;
//open the port
this->serialPort1->Open();
this->serialPort1->ReadTimeout = period+1;
this->serialPort1->WriteTimeout = period+1;
String^ name_ = this->serialPort1->PortName;
START=gcnew String("S");
this->textBox1->Text="Logging started";
timer1->Enabled=true;
interval->Enabled=true;
myStream=new ofstream(directory,ios::out);
*myStream<<"time(ms);ADC1;ADC2;ADC3;ADC4;ADC5;ADC6;ADC7;ADC8;";
*myStream<<endl;
chart1->Series["ADC1"]->Points->Clear();
chart1->Series["ADC2"]->Points->Clear();
chart1->Series["ADC3"]->Points->Clear();
chart1->Series["ADC4"]->Points->Clear();
chart1->Series["ADC5"]->Points->Clear();
chart1->Series["ADC6"]->Points->Clear();
chart1->Series["ADC7"]->Points->Clear();
chart1->Series["ADC8"]->Points->Clear();
backgroundWorker1->RunWorkerAsync();
}
else
{
this->textBox1->Text="Warning: port is busy or isn't open";
timer1->Enabled=false;
interval->Enabled=false;
}
}
catch(UnauthorizedAccessException^)
{
this->textBox1->Text="Unauthorized access";
timer1->Enabled=false;
interval->Enabled=false;
}
}
}
}
private: System::Void backgroundWorker1_DoWork(System::Object^ sender, System::ComponentModel::DoWorkEventArgs^ e) {
while(!backgroundWorker1->CancellationPending){
if(backgroundWorker1->CancellationPending){
e->Cancel=true;
return;
}
t+=period;
if(t<10*period){
this->chart1->ChartAreas["ChartArea1"]->AxisX->Minimum=0;
this->chart1->ChartAreas["ChartArea1"]->AxisX->Maximum=t+10*period;
}
else {
this->chart1->ChartAreas["ChartArea1"]->AxisX->Minimum=t-10*period;
this->chart1->ChartAreas["ChartArea1"]->AxisX->Maximum=t+10*period;
}
*myStream<<t<<";";
for (int n=0;n<8;n++){
adc_array[n]= this->serialPort1->ReadByte();
}
Array::Copy(adc_array,ADC,8);
for(int f=0; f<8; f++){
*myStream<<ADC[f]<<";";
}
*myStream<<endl;
backgroundWorker1->ReportProgress(t);
}
}
private: System::Void backgroundWorker1_ProgressChanged(System::Object^ sender, System::ComponentModel::ProgressChangedEventArgs^ e) {
chart1->Series["ADC1"]->Points->AddXY(t,ADC[0]);
chart1->Series["ADC2"]->Points->AddXY(t,ADC[1]);
chart1->Series["ADC3"]->Points->AddXY(t,ADC[2]);
chart1->Series["ADC4"]->Points->AddXY(t,ADC[3]);
chart1->Series["ADC5"]->Points->AddXY(t,ADC[4]);
chart1->Series["ADC6"]->Points->AddXY(t,ADC[5]);
chart1->Series["ADC7"]->Points->AddXY(t,ADC[6]);
chart1->Series["ADC8"]->Points->AddXY(t,ADC[7]);
}
the user is allowed to define intervals in seconds for data acquisition (in the code this interval is w after conversion to double). In this case, the program sends a pulse to the MCU requesting a new data transmission. So far, I have been testing this for 1 second intervals (note, during each interval the MCU sends 8 frames, each representing an ADC). However, I need to get this to run for 10ms intervals at some point. Will this be possible? Any idea on how to solve the few problems I mentioned at the beginning?
Thanks in advance
UPDATE
Just to give you an idea of what's happening:
I commented the charting part and ran the program for about 5 minutes, with a reading interval of 1s. So I expected to get around 5x60=300 values in the output file, but I only got 39 (i.e. starting from 1s till 39s). The program was still running, but the data were not getting stored anymore.
Testing was done in release mode and not debug mode. In debug mode, setting a break point under serialport->readbyte(), does not reproduce the problem. My guess is it's a timing issue between program and MCU.
You are making several standard mistakes. First off, do NOT unplug the cable when the port is opened. Many USB emulators don't know how to deal with that, the FTDI driver is particularly notorious about that. They just make the port disappear while it is in use, this invariably gives code that uses the port a severe heart attack. An uncatchable exception is common.
Secondly, you are accessing properties of a class that is not thread-safe in a worker thread. The Chart control was made to be used only in a UI thread, accessing the ChartAreas property in a worker is going to buy you a lot of misery. Getting an InvalidOperationException is pretty typical when you violate threading requirements, it is however not consistently implemented. Nastiness includes random AccessViolationExceptions, corrupted data and deadlock.
Third, you are setting completely unrealistic goals. Pursuing an update every 10 milliseconds is pointless, the human eye cannot perceive that. Anything past 50 milliseconds just turns into a blur. Something that is taken advantage of when you watch a movie in the cinema, it displays at 24 frames per second. The failure mode for that is unpleasant as well, you'll eventually reach a point where you are pummeling the UI thread (or the Chart control) with more updates than it can process. The side effect is that the UI stops painting itself, it is too busy trying to keep up with the deluge of invoke requests. And the amount of memory your program consumes keeps building, the update queue grows without bounds. That does eventually end with an OOM exception, it takes a while to consume 2 jiggabytes however. You will need to prevent this from happening, you need to throttle the rate at which you invoke. A simple thread-safe counter can take care of that.
Forth, you are accessing the data you gather in more than one thread without taking care of thread-safety. The ADC array content is being changed by the worker while the UI thread is reading it. Various amounts of misery from that, bad data at a minimum. A simply workaround is to pass a copy of the data to the ReportProgress method. In general, address these kind of threading problems by using pull instead of push. Get rid of the fire-hose problem by having the UI thread pace the requests instead of trying to have the UI thread keep up.