According to Microsoft, starting with Windows 10, applications using shared-mode WASAPI can request buffer sizes smaller than 10ms (see https://msdn.microsoft.com/en-us/library/windows/hardware/mt298187%28v=vs.85%29.aspx).
According to the article, achieving such low latencies requires some driver updates, which I did. Using an exclusive-mode render and capture stream, I measured a total round-trip latency (using a hardware loopback cable) of around 13ms. This suggests to me that at least one of the endpoints successfully achieves a latency of < 10ms. (Is this assumption correct?)
The article mentions that applications can use the new IAudioClient3
interface to query the minimum buffer size supported by the Windows audio engine using IAudioClient3::GetSharedModeEnginePeriod()
. However, this function always returns 10ms on my system, and any attempt to initialize an audio stream using either IAudioClient::Initialize()
or IAudioClient3::InitializeSharedAudioStream()
with a period lower than 10ms always results in AUDCLNT_E_INVALID_DEVICE_PERIOD
.
Just to be sure, I also disabled any effects processing in the audio drivers. What am I missing? Is it even possible to get low latency from shared mode? See below for some sample code.
#include <windows.h>
#include <atlbase.h>
#include <mmdeviceapi.h>
#include <audioclient.h>
#include <iostream>
#define VERIFY(hr) do { \
auto temp = (hr); \
if(FAILED(temp)) { \
std::cout << "Error: " << #hr << ": " << temp << "\n"; \
goto error; \
} \
} while(0)
int main(int argc, char** argv) {
HRESULT hr;
CComPtr<IMMDevice> device;
AudioClientProperties props;
CComPtr<IAudioClient> client;
CComPtr<IAudioClient2> client2;
CComPtr<IAudioClient3> client3;
CComHeapPtr<WAVEFORMATEX> format;
CComPtr<IMMDeviceEnumerator> enumerator;
REFERENCE_TIME minTime, maxTime, engineTime;
UINT32 min, max, fundamental, default_, current;
VERIFY(CoInitializeEx(nullptr, COINIT_APARTMENTTHREADED));
VERIFY(enumerator.CoCreateInstance(__uuidof(MMDeviceEnumerator)));
VERIFY(enumerator->GetDefaultAudioEndpoint(eRender, eMultimedia, &device));
VERIFY(device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr, reinterpret_cast<void**>(&client)));
VERIFY(client->QueryInterface(&client2));
VERIFY(client->QueryInterface(&client3));
VERIFY(client3->GetCurrentSharedModeEnginePeriod(&format, ¤t));
// Always fails with AUDCLNT_E_OFFLOAD_MODE_ONLY.
hr = client2->GetBufferSizeLimits(format, TRUE, &minTime, &maxTime);
if(hr == AUDCLNT_E_OFFLOAD_MODE_ONLY)
std::cout << "GetBufferSizeLimits returned AUDCLNT_E_OFFLOAD_MODE_ONLY.\n";
else if(SUCCEEDED(hr))
std::cout << "hw min = " << (minTime / 10000.0) << " hw max = " << (maxTime / 10000.0) << "\n";
else
VERIFY(hr);
// Correctly? reports a minimum hardware period of 3ms and audio engine period of 10ms.
VERIFY(client->GetDevicePeriod(&engineTime, &minTime));
std::cout << "hw min = " << (minTime / 10000.0) << " engine = " << (engineTime / 10000.0) << "\n";
// All values are set to a number of frames corresponding to 10ms.
// This does not change if i change the device's sampling rate in the control panel.
VERIFY(client3->GetSharedModeEnginePeriod(format, &default_, &fundamental, &min, &max));
std::cout << "default = " << default_
<< " fundamental = " << fundamental
<< " min = " << min
<< " max = " << max
<< " current = " << current << "\n";
props.bIsOffload = FALSE;
props.cbSize = sizeof(props);
props.eCategory = AudioCategory_ForegroundOnlyMedia;
props.Options = AUDCLNT_STREAMOPTIONS_RAW | AUDCLNT_STREAMOPTIONS_MATCH_FORMAT;
// Doesn't seem to have any effect regardless of category/options values.
VERIFY(client2->SetClientProperties(&props));
format.Free();
VERIFY(client3->GetCurrentSharedModeEnginePeriod(&format, ¤t));
VERIFY(client3->GetSharedModeEnginePeriod(format, &default_, &fundamental, &min, &max));
std::cout << "default = " << default_
<< " fundamental = " << fundamental
<< " min = " << min
<< " max = " << max
<< " current = " << current << "\n";
error:
CoUninitialize();
return 0;
}