IAudioClient over RDP lags on event notification

Question

Consider this code that captures audio from the default endpoint and displays how many frames are passed each second.

This works OK if it's not a RDP session (Remote Audio). If it is a RDP session, this code only works without the event.
If I use an event, then the audio lags. Each second only about half the samples are returned.

Is this a bug of the Remote Audio interface when using an event notification?

#include "stdafx.h"
int main()
{
	int BS = 512;

	CoInitialize(0);
	CComPtr<IMMDeviceEnumerator> deviceEnumerator = 0;
	auto hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_INPROC_SERVER, __uuidof(IMMDeviceEnumerator), (LPVOID*)&deviceEnumerator);
	if (!deviceEnumerator)
		return 0;
	CComPtr<IMMDevice> d;
	deviceEnumerator->GetDefaultAudioEndpoint(eCapture, eMultimedia, &d);
	if (!d)
		return 0;

	CComPtr<IAudioClient> ac;
	hr = d->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL, (LPVOID*)&ac);
	auto Has = CreateEvent(0, 0, 0, 0);


	WAVEFORMATEX* wf = 0;
	ac->GetMixFormat(&wf);

	unsigned long long a = BS;
	a *= 1000;
	a *= 10000;
	a /= wf->nSamplesPerSec;
	REFERENCE_TIME rt = a;

	bool Ev = true;

	DWORD flg = AUDCLNT_STREAMFLAGS_NOPERSIST;
	flg |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
	if (Ev)
		flg |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
	hr = ac->Initialize(AUDCLNT_SHAREMODE_SHARED, flg, rt, rt, (WAVEFORMATEX*)wf, 0);
	CComPtr<IAudioCaptureClient> cap;
	ac->GetService(__uuidof(IAudioCaptureClient), (void**)&cap);
	ac->SetEventHandle(Has);

	ac->Start();

	UINT32 sz = 0;
	ac->GetBufferSize(&sz);

	auto t = GetTickCount64();
	unsigned long long tfr = 0;
	for (;;)
	{
		BYTE* pData = 0;
		UINT32 framesAvailable = 0;
		DWORD  flags = 0;
		if (Ev)
			WaitForSingleObject(Has, INFINITE);
		auto t2 = GetTickCount64();

		hr = cap->GetBuffer(&pData, &framesAvailable, &flags, NULL, NULL);
		if (FAILED(hr))
			break;
		if (!framesAvailable)
			continue;
		if (framesAvailable)
			tfr += framesAvailable;
		if ((t2 - t) >= 1000)
		{
			printf("%llu - %lli\r\n", (unsigned long long)(t2 - t), tfr);
			tfr = 0;
			t = t2;
		}
		cap->ReleaseBuffer(framesAvailable);
	}

}

Michael

Answer 1

Your code assumes there is only a single packet in the buffer when the event is signaled. You need to fully drain the buffer whenever WASAPI wakes you up.

See https://matthewvaneerde.wordpress.com/2014/11/05/draining-the-wasapi-capture-buffer-fully/ and https://docs.microsoft.com/en-us/samples/microsoft/windows-universal-samples/windowsaudiosession/

---

Matthew van Eerde