hnsbufferduration parameter in iaudioclient:: initialize method

Question

In the iaudioclient:: initialize method, the third parameter is hnsbufferduration. In the explanation of Microsoft documents, this parameter contains the buffer size requested by the caller. If the iaudioclient:: initialize method is called successfully, the system will allocate a buffer of at least this size.（https://learn.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient-initialize）

For capture and render, I have the following two questions：

1. For the rendering device, when calling the iaudioclient:: initialize method, set the hnsbufferduration to 100 * 10000 (i.e. the buffer size is 100ms), and set the hnsperiodicity parameter to 0 (which means that the default period is 10ms). Does that mean that I must fill the 100ms buffer to play the sound from the speaker? Or does the system play the data in the rendering buffer from the speaker device every 10ms?Because I will receive the system event notification every 10ms. At this time, I will call the iaudiorenderclient:: getbuffer method to insert 10ms of data into the buffer. So I want to know when the speaker will really play sound? Do you need to fill all the buffers set by hnsbufferduration (or reach a certain amount) before playing through the speaker device.
2. For the capture device, when calling the iaudioclient:: initialize method, set the hnsbufferduration to 100 * 10000 (i.e. the buffer size is 100ms), and set the hnsperiodicity parameter to 0 (which means that the default period is 10ms). Does it mean that this buffer can help me cache the next 100ms of data at most？For example, under normal circumstances, I should have received the system event notification for each 10ms, then called IAudioCaptureClient:: GetBuffer to get the data collected by the microphone. If my thread gets stuck during this period and calls iaudiocaptureclient:: getbuffer after 60ms, does the capture buffer help me save all the data in the past 60ms?

Look forward to your professional answer, thank you

Answer 1

Hello,
Welcome to Microsoft Q&A!

Sorry to keep you waiting. Since I didn't have a rendering device with me, I could only do capture tests.
When I did my tests, I found that the buffer could read incomplete data by CopyMemory, but for C++ memory safety, I recommend using WaitForSingleObject(AudioSamplesReadyEvent, INFINITE) in the thread to prevent memory problems.

//InitializeAudioEngine  
	hr = _AudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST, 100* 10000, 0, _MixFormat, NULL);  
  
	if (FAILED(hr))  
	{  
		printf("Unable to initialize audio client: %x.\n", hr);  
		return false;  
	}  
  
	//  
	//  Retrieve the buffer size for the audio client.  
	//  
	hr = _AudioClient->GetBufferSize(&_BufferSize);  
	//printf(" get audio client buffer: %d \n", _BufferSize);  
	if (FAILED(hr))  
	{  
		printf("Unable to get audio client buffer: %x. \n", hr);  
		return false;  
	}  
	hr = _AudioClient->SetEventHandle(_AudioSamplesReadyEvent);  
	if (FAILED(hr))  
	{  
		printf("Unable to set ready event: %x.\n", hr);  
		return false;  
	}  
  
	hr = _AudioClient->GetService(IID_PPV_ARGS(&_CaptureClient));  
	if (FAILED(hr))  
	{  
		printf("Unable to get new capture client: %x.\n", hr);  
		return false;  
	}  
  
	//start capture  
  
	hr = _AudioClient->Start();  
	if (FAILED(hr))  
	{  
		printf("Unable to get new capture client: %x.\n", hr);  
		return false;  
	}  
  
	bool stillPlaying = true;  
	while (stillPlaying)  
	{  
		DWORD waitResult = WaitForSingleObject(_AudioSamplesReadyEvent, INFINITE);  
  
		BYTE* pData, * pBuffer;  
		INT nBufferLenght;  
		UINT32 framesAvailable;  
		DWORD  flags;  
		pBuffer = new BYTE[MAX_AUDIO_FRAME_SIZE];  
		hr = _CaptureClient->GetBuffer(&pData, &framesAvailable, &flags, NULL, NULL);  
		if (SUCCEEDED(hr))  
		{  
			if (framesAvailable != 0)  
			{  
				if (flags & AUDCLNT_BUFFERFLAGS_SILENT)  
				{  
					//  
					//  Fill 0s from the capture buffer to the output buffer.  
					//  
				}  
				else  
				{  
					//  
					//  Copy data from the audio engine buffer to the output buffer.  
					//  
  
					CopyMemory(pBuffer, pData, framesAvailable * _FrameSize);  
					printf("get capture frames: %d!\n", framesAvailable);  
				}  
			}  
			delete[] pBuffer;  
			hr = _CaptureClient->ReleaseBuffer(framesAvailable);  
			if (FAILED(hr))  
			{  
				printf("Unable to release capture buffer: %x!\n", hr);  
			}  
		}  
	}  

Thank you.

---

If the answer is the right solution, please click "Accept Answer" and kindly upvote it. If you have extra questions about this answer, please click "Comment".
Note: Please follow the steps in our documentation to enable e-mail notifications if you want to receive the related email notification for this thread.