I found that in the Microsoft sample example:
void D3D12HelloTriangle::OnRender()
{
// Record all the commands we need to render the scene into the command list.
PopulateCommandList();
// Execute the command list.
ID3D12CommandList* ppCommandLists[] = { m_commandList.Get() };
m_commandQueue->ExecuteCommandLists(_countof(ppCommandLists), ppCommandLists);
// Present the frame.
ThrowIfFailed(m_swapChain->Present(1, 0));
WaitForPreviousFrame();
}
How does is actually work ? ExecuteCommandLists is a asynchronous function call, so it means the code execution will continue and it hits Present function.
What will happen after Present call ? Let's say, GPU is still drawing, working and present is called. Is Present sychronous call ? It cannot present buffer when gpu is still drawing. Is that correct ? Could someone explain what's happening here ?
Present is also an asynchronous command that tells the GPU to start scanning out (displaying) from the next buffer in the swap chain. You don't have to worry about the GPU not having finished executing all previously issued work (on the graphics command queue) before the 'Flip' takes place.
Related
I am trying to control a robot using a template-based controller class written in c++. Essentially I have a UDP connection setup with the robot to receive the state of the robot and send new torque commands to the robot. I receive new observations at a higher frequency (say 2000Hz) and my controller takes about 1ms (1000Hz) to calculate new torque commands to send to the robot. The problem I am facing is that I don't want my main code to wait to send the old torque commands while my controller is still calculating new commands to send. From what I understand I can use Ubuntu with RT-Linux kernel, multi-thread the code so that my getTorques() method runs in a different thread, set priorities for the process, and use mutexes and locks to avoid data race between the 2 threads, but I was hoping to learn what the best strategies to write hard-realtime code for such a problem are.
// main.cpp
#include "CONTROLLER.h"
#include "llapi.h"
void main{
...
CONTROLLERclass obj;
...
double new_observation;
double u;
...
while(communicating){
get_newObs(new_observation); // Get new state of the robot (2000Hz)
obj.getTorques(new_observation, u); // Takes about 1ms to calculate new torques
send_newCommands(u); // Send the new torque commands to the robot
}
...
}
Thanks in advance!
Okay, so first of all, it sounds to me like you need to deal with the fact that you receive input at 2 KHz, but can only compute results at about 1 KHz.
Based on that, you're apparently going to have to discard roughly half the inputs, or else somehow (in a way that makes sense for your application) quickly combine the inputs that have arrived since the last time you processed the inputs.
But as the code is structured right now, you're going to fetch and process older and older inputs, so even though you're producing outputs at ~1 KHz, those outputs are constantly being based on older and older data.
For the moment, let's assume you want to receive inputs as fast as you can, and when you're ready to do so, you process the most recent input you've received, produce an output based on that input, and repeat.
In that case, you'd probably end up with something on this general order (using C++ threads and atomics for the moment):
std::atomic<double> new_observation;
std::thread receiver = [&] {
double d;
get_newObs(d);
new_observation = d;
};
std::thread sender = [&] {
auto input = new_observation;
auto u = get_torques(input);
send_newCommands(u);
};
I've assumed that you'll always receive input faster than you can consume it, so the processing thread can always process whatever input is waiting, without receiving anything to indicate that the input has been updated since it was last processed. If that's wrong, things get a little more complex, but I'm not going to try to deal with that right now, since it sounds like it's unnecessary.
As far as the code itself goes, the only thing that may not be obvious is that instead of passing a reference to new_input to either of the existing functions, I've read new_input into variable local to the thread, then passed a reference to that.
I've read several articles on the CPU-GPU (using fences) and GPU-GPU (using semaphores) synchronization mechanisms, but still got trouble to understand how I should implement a simple render-loop.
Please take a look at the simple render() function below. If I got it right, the minimal requirement is that we ensure the GPU-GPU synchronization between vkAcquireNextImageKHR, vkQueueSubmit and vkQueuePresentKHR by a single set of semaphores image_available and rendering_finished as I've done in the example code below.
However, is this really safe? All operations are asynchronous. So, is it really safe to "reuse" the image_available semaphore in a subsequent call of render() again even though the signal request from the previous call hasn't fired yet? I would think it's not, but, on the other hand, we're using the same queues (don't know if it matters where the graphics and presentation queue are actually the same) and operations inside a queue should be sequentially consumed ... But if I got it right, they might not be consumed "as a whole" and could be reordered ...
The second thing is that (again, unless I'm missing something) I clearly should use one fence per swap chain image to ensure that the operation on the image corresponding to the image_index of the call to render() has finished. But does that mean that I necessarily need to do a
if (vkWaitForFences(device(), 1, &fence[image_index_of_last_call], VK_FALSE, std::numeric_limits<std::uint64_t>::max()) != VK_SUCCESS)
throw std::runtime_error("vkWaitForFences");
vkResetFences(device(), 1, &fence[image_index_of_last_call]);
before my call to vkAcquireNextImageKHR? And do I then need dedicated image_available and rendering_finished semaphores per swap chain image? Or maybe per frame? Or maybe per command buffer/pool? I'm really confused ...
void render()
{
std::uint32_t image_index;
switch (vkAcquireNextImageKHR(device(), swap_chain().handle(),
std::numeric_limits<std::uint64_t>::max(), m_image_available, VK_NULL_HANDLE, &image_index))
{
case VK_SUBOPTIMAL_KHR:
case VK_SUCCESS:
break;
case VK_ERROR_OUT_OF_DATE_KHR:
on_resized();
return;
default:
throw std::runtime_error("vkAcquireNextImageKHR");
}
static VkPipelineStageFlags constexpr wait_destination_stage_mask = VK_PIPELINE_STAGE_COLOR_ATTACHMENT_OUTPUT_BIT;
VkSubmitInfo submit_info{};
submit_info.sType = VK_STRUCTURE_TYPE_SUBMIT_INFO;
submit_info.waitSemaphoreCount = 1;
submit_info.pWaitSemaphores = &m_image_available;
submit_info.signalSemaphoreCount = 1;
submit_info.pSignalSemaphores = &m_rendering_finished;
submit_info.pWaitDstStageMask = &wait_destination_stage_mask;
if (vkQueueSubmit(graphics_queue().handle, 1, &submit_info, VK_NULL_HANDLE) != VK_SUCCESS)
throw std::runtime_error("vkQueueSubmit");
VkPresentInfoKHR present_info{};
present_info.sType = VK_STRUCTURE_TYPE_PRESENT_INFO_KHR;
present_info.waitSemaphoreCount = 1;
present_info.pWaitSemaphores = &m_rendering_finished;
present_info.swapchainCount = 1;
present_info.pSwapchains = &swap_chain().handle();
present_info.pImageIndices = &image_index;
switch (vkQueuePresentKHR(presentation_queue().handle, &present_info))
{
case VK_SUCCESS:
break;
case VK_ERROR_OUT_OF_DATE_KHR:
case VK_SUBOPTIMAL_KHR:
on_resized();
return;
default:
throw std::runtime_error("vkQueuePresentKHR");
}
}
EDIT: As suggested in the answers below, assume we have k "frames in flight" and hence k instances of the semaphores and the fence used in the code above, which I will denote by m_image_available[i], m_rendering_finished[i] and m_fence[i] for i = 0, ..., k - 1. Let i denote the current index of the frame in flight, which is increased by 1 after each invocation of render(), and j denote the number of invocations of render(), starting from j = 0.
Now, assume the swap chain contains three images.
If j = 0, then i = 0 and the first frame in flight is using swap chain image 0
In the same way, if j = a, then i = a and the ath frame in flight is using swap chain image a, for a= 2, 3
Now, if j = 3, then i = 3, but since the swap chain image only has three images, the fourth frame in flight is using swap chain image 0 again. I wonder whether this is problematic or not. I guess it's not, since the wait/signal semaphores m_image_available[3]/m_rendering_finished[3], used in the calls of vkAcquireNextImageKHR, vkQueueSubmit and vkQueuePresentKHR in this invocation of render(), are dedicated to this particular frame in flight.
If we reach j = k, then i = 0 again, since there are only k frames in flight. Now we potentially wait at the beginning of render(), if the call to vkQueuePresentKHR from the first invocation (i = 0) of render() hasn't signaled m_fence[0] yet.
So, besides my doubts described in the third bullet point above, the only question which remains is why I shouldn't take k as large as possible? What I theoretically could imagine is that if we are submitting work to the GPU in a quicker fashion than the GPU is able to consume, the used queue(s) might continually grow and eventually overflow (is there some kind of "max commands in queue" limit?).
If I got it right, the minimal requirement is that we ensure the GPU-GPU synchronization between vkAcquireNextImageKHR, vkQueueSubmit and vkQueuePresentKHR by a single set of semaphores image_available and rendering_finished as I've done in the example code below.
Yes, you got it right. You submit the desire to get a new image to render into via vkAcquireNextImageKHR. The presentation engine will signal the m_image_available semaphore as soon as an image to render into has become available. But you have already submitted the instruction.
Next, you submit some commands to the graphics queue via submit_info. I.e. they are also already submitted to the GPU and wait there until the m_image_available semaphore receives its signal.
Furthermore, a presentation instruction is submitted to the presentation engine that expresses the dependency that it needs to wait until the submit_info-commands have completed by waiting on the m_rendering_finished semaphore.
I.e. everything has been submitted. If nothing has been signalled yet, everything just sits there in some GPU buffers and waits for signals.
Now, if your code loops right back into the render() function and re-uses the same m_image_available and m_rendering_finished semaphores, it will only work if you are very lucky, namely if all the semaphores have already been signalled before you use them again.
The specifications says the following for vkAcquireNextImageKHR:
If semaphore is not VK_NULL_HANDLE it must not have any uncompleted signal or wait operations pending
and furthermore, it says under 7.4.2. Semaphore Waiting
the act of waiting for a binary semaphore also unsignals that semaphore.
I.e. indeed, you need to wait on the CPU until you know for sure that the previous vkAcquireNextImageKHR that uses the same m_image_available semaphore has completed.
And yes, you already got it right: You need to use a fence for that which you pass to vkQueueSubmit. If you do not synchronize on the CPU, you'll shovel ever more work to the GPU (which is a problem) and the semaphores that you are re-using might not get properly unsignalled in time (which is a problem).
What is often done is that the semaphores and fences are multiplied, e.g. to 3 each, and these sets of synchronization objects are used in sequence, so that more work can be parallelized on the GPU. The Vulkan Tutorial describes this quite nicely in its Rendering and presentation chapter. It is also explained with animation in this lecture starting at 7:59.
So first of all, as you mentioned correctly, semaphores are strictly for GPU-GPU synchronization, e.g. to make sure that one batch of commands (one submit) has finished before another one starts. This is here used to synchronize the rendering commands with the present command such that the presenting engine knows when to present the rendered image.
Fences are the main utility for CPU-GPU synchronization. You place a fence in a queue submit and then on the CPU side wait for it before you want to proceed. This is usually done here such that we do not queue any new rendering/present commands while the previous frame hasn't finished.
But does that mean that I necessarily need to do a
if (vkWaitForFences(device(), 1, &fence[image_index_of_last_call], VK_FALSE, std::numeric_limits<std::uint64_t>::max()) != VK_SUCCESS)
throw std::runtime_error("vkWaitForFences");
vkResetFences(device(), 1, &fence[image_index_of_last_call]);
before my call to vkAcquireNextImageKHR?
Yes, you definitely need this in your code, otherwise your semaphores would not be safe and you would probably get validation errors.
In general, if you want your CPU to wait until your GPU has finished rendering of the previous frame, you would have only a single fence and a single pair of semaphores. You could also replace the fence by a waitIdle command of the queue or device.
However, in practice you do not want to stall the CPU and in the meantime record commands for the next frame. This is done via frames in flight. This simply means that for every frame in flight (i.e. number of frames that can be recorded in parallel to the execution on the GPU), you have one fence and one pair of semaphores which synchronize that particular frame.
So in essence for your render loop to work properly you need a pair of semaphores + fence per frame in flight, independent of the number of swapchain images. However, do note that the current frame index (frame in flight) and image index (swapchain) will generally not be the same except you use the same amount of swapchain images as frames in flight. This is because the presenting engine might give you swapchain images out of order depending on your presenting mode.
I have a problem with Microsoft's WaveOut API:
edit1: Added Link to sample project:
edit2: removed link, its not representative of the issue
After playing some audio, when I want to terminate a given playback stream, I call the function:
waveOutClose(hWaveOut_);
However, even after waveOutClose() is called, sometimes the library will still access memory previously passed to it by waveOutWrite(), causing an invalid memory access.
I then tried to ensure all the buffers are marked as done before freeing the buffer:
PcmPlayback::~PcmPlayback()
{
if(hWaveOut_ == nullptr)
return;
waveOutReset(hWaveOut_); // infinite-loops, never returns
for(auto it = buffers_.begin(); it != buffers_.end(); ++it)
waveOutUnprepareHeader(hWaveOut_, &it->wavehdr_, sizeof(WAVEHDR));
while( buffers_.empty() == false ) // infinite loops
removeCompletedBuffers();
waveOutClose(hWaveOut_);
//Unhandled exception at 0x75629E80 (msvcrt.dll) in app.exe:
// 0xC0000005: Access violation reading location 0xFEEEFEEE.
}
void PcmPlayback::removeCompletedBuffers()
{
for(auto it = buffers_.begin(); it != buffers_.end();)
{
if( it->wavehdr_.dwFlags & WHDR_DONE )
{
waveOutUnprepareHeader(hWaveOut_, &it->wavehdr_, sizeof(WAVEHDR));
it = buffers_.erase(it);
}
else
++it;
}
}
However, this situation never happens - the buffer never becomes empty. There will be 4-5 blocks remaining with wavehdr_.dwFlags == 18 (I believe this means the blocks are still marked as in playback)
How can I resolve this issue?
# Martin Schlott ("Can you provide the loop where you write the buffer to waveOutWrite?")
Its not quite a loop, instead I have a function that is called whenever I receive an audio packet over the network:
void PcmPlayback::addData(const std::vector<short> &rhs)
{
removeCompletedBuffers();
if(rhs.empty())
return;
// add new data
buffers_.push_back(Buffer());
Buffer & buffer = buffers_.back();
buffer.data_ = rhs;
ZeroMemory(&buffers_.back().wavehdr_, sizeof(WAVEHDR));
buffer.wavehdr_.dwBufferLength = buffer.data_.size() * sizeof(short);
buffer.wavehdr_.lpData = (char *)(buffer.data_.data());
waveOutPrepareHeader(hWaveOut_, &buffer.wavehdr_, sizeof(WAVEHDR)); // prepare block for playback
waveOutWrite(hWaveOut_, &buffer.wavehdr_, sizeof(WAVEHDR));
}
The described behavior can happen if you do not call
waveOutUnprepareHeader
to every buffer you used before you use
waveOutClose
The flagfield _dwFlags seems to indicate that the buffers are still enqueued (WHDR_INQUEUE | WHDR_PREPARED) try:
waveOutReset
before unprepare buffers.
After analyses your code, I found two problems/bugs which are not related to waveOut (funny, you use C++11 but the oldest media interface). You use a vector as buffer. During some calling operations, the vector is copied! One bug I found is:
typedef std::function<void(std::vector<short>)> CALLBACK_FN;
instead of:
typedef std::function<void(std::vector<short>&)> CALLBACK_FN;
which forces a copy of the vector.
Try to avoid using vectors if you expect to use it mostly as rawbuffer. Better use std::unique_pointer as buffer pointer.
Your callback in the recorder is not monitored by a mutex, nor does it check if a destructor was already called. The destructing happens during the callback (mostly) which leads to an exception.
For your test program, go back and use raw pointer and static callbacks before blaming waveOut. Your code is not bad, but the first bug already shows, that a small bug will lead to unpredictical errors. As you also organize your buffers in a std::array, I would search for bugs there. I guess, you make a unintentional copy of your whole buffer array, unpreparing the wrong buffers.
I did not have the time to dig deeper, but I guess those are the problems.
I managed to find my problem in the end, it was caused by multiple bugs and a deadlock. I will document what happened here so people can learn from this in the future
I was clued in to what was happening when I fixed the bugs in the sample:
call waveInStop() before waveInClose() in ~Recorder.cpp
wait for all buffers to have the WHDR_DONE flag before calling waveOutClose() in ~PcmPlayback.
After doing this, the sample worked fine and did not display the behavior of the WHDR_DONE flag never being marked.
In my main program, that behavior was caused by a deadlock that occurs in the following situation:
I have a vector of objects representing each peer I am streaming audio with
Each Object owns a Playback class
This vector is protected by a mutex
Recorder callback:
mutex.lock()
send audio packet to each peer.
Remove Peer:
mutex.lock()
~PcmPlayback
wait for WHDR_DONE flags to be marked
A deadlock occurs when I remove a peer, locking the mutex and the recorder callback tries to acquire a lock too.
Note that this will happen often because the playback buffer is usually (~4 * 20ms) while the recorder has a cadence of 20ms.
In ~PcmPlayback, the buffers will never be marked as WHDR_DONE and any calls to the WaveOut API will never return because the WaveOut API is waiting for the Recorder callback to complete, which is in turn waiting on mutex.lock(), causing a deadlock.
I am trying to implement streaming audio and I've run into a problem where OpenAL is giving me an error codes seems impossible given the information in the documentation.
int buffersProcessed = 0;
alGetSourcei(m_Source, AL_BUFFERS_PROCESSED, &buffersProcessed);
PrintALError();
int toAddBufferIndex;
// Remove the first buffer from the queue and move it to
//the end after buffering new data.
if (buffersProcessed > 0)
{
ALuint unqueued;
alSourceUnqueueBuffers(m_Source, 1, &unqueued);
/////////////////////////////////
PrintALError(); // Prints AL_INVALID_OPERATION //
/////////////////////////////////
toAddBufferIndex = firstBufferIndex;
}
According to the documentation [PDF], AL_INVALID_OPERATION means: "There is no current context." This seems like it can't be true because OpenAL has been, and continues to play other audio just fine!
Just to be sure, I called ALCcontext* temp = alcGetCurrentContext( ); here and it returned a valid context.
Is there some other error condition that's possible here that's not mentioned in the docs?
More details: The sound source is playing when this code is being called, but the impression I got from reading the spec is you can safely unqueue processed buffers while the source is playing. PrintALError is just a wrapper for alGetError that prints if there is any error.
I am on a Mac (OS 10.8.3), in case it matters.
So far what I've gathered is that it seems this OpenAL implementation incorrectly throws an error if you unqueue a buffer while the source is playing. The spec says that you should be able to unqueue a buffer that has been marked as processing while the source is playing:
Removal of a given queue entry is not possible unless either the source is stopped (in which case then entire queue is considered processed), or if the queue entry has already been processed (AL_PLAYING or AL_PAUSED source).
On that basis I'm gonna say this is probably a bug in my OpenAL implementation. I'm gonna leave the question open in case someone can give a more concrete answer though.
To handle condition for multiple buffers use a loop.
Following works on iOS and linux :
// UN queue used buffers
ALint buffers_processed = 0;
alGetSourcei(streaming_source, AL_BUFFERS_PROCESSED, & buffers_processed); // get source parameter num used buffs
while (buffers_processed > 0) { // we have a consumed buffer so we need to replenish
ALuint unqueued_buffer;
alSourceUnqueueBuffers(streaming_source, 1, & unqueued_buffer);
available_AL_buffer_array_curr_index--;
available_AL_buffer_array[available_AL_buffer_array_curr_index] = unqueued_buffer;
buffers_processed--;
}
I am continuing work on my DirectShow application and am just putting the finishing touches on it. What the program is doing is going through a video file in 1 second intervals and capturing from the samplegrabber the current buffer and processing it before moving on. However, I was noticing that I was getting duplicate images in my tests to which I found out that DirectShow has not incremented through the video in that 1 second interval fast enough. My question is if there is a way to check when DirectShow is ready for me to call the samplegrabber to get the current frame and to process it. At the moment I call sleep for 1 second but there has to be a better method. Thank you in advance for the help.
EDIT
I just tried running a check to see if the video's position is equal to the next position that I would like to grab and process. That decreased the number of duplicate frames but I still see them showing up in chunks.
I always let the DS framework handle the processing rate:
in the main application thread, configure the sample grabber callback and then when the callback is triggered, you get the media sample as well as sample time: at this point you can process the sample if the appropriate interval i.e. 1 second has elapsed.
What do you mean you call sleep for a second and from where (which thread) do you call it?
If you're doing this from inside the callback you are effectively blocking the DirectShow pipeline? Perhaps if you could explain your setup in more detail I could be more helpful.
/// Callback that is triggered per media sample
/// Note this all happens in the DirectShow streaming thread!
STDMETHODIMP SampleCB( double dSampleTime, IMediaSample * pSample )
{
// check timestamp and if one second has elapsed process sample accordingly
// else do nothing
...
// Get byte pointer
BYTE* pbData(NULL);
HRESULT hr = pSample->GetPointer(&pbData);
if (FAILED(hr))
{
return hr;
}
...
}
P.S if you want to process the samples as fast as possible, you can set the sample timestamp to NULL in your callback.
// set time to NULL to allow for fast rendering since the
// the video renderer controls the rendering rate according
// to the timestamps
pSample->SetTime(NULL, NULL);
Try setting your graph timer to NULL. It will allow to:
process the file as fast as possible
will relieve you of the issues you have.
Of course, it won't work if you are rendering the file to screen at the same time.