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authorChris Robinson <[email protected]>2023-09-21 13:04:46 -0700
committerChris Robinson <[email protected]>2023-09-21 13:04:46 -0700
commit1b8c63910fc36a292ae591f5bedc8654c0173be9 (patch)
tree770bb67022ee1b2ac1016e50ee2ced65f8b749db /alc
parentbf30af18960ec1443bd166bf145e632570bd9072 (diff)
Try to get the device period for spatial audio streams
Diffstat (limited to 'alc')
-rw-r--r--alc/backends/wasapi.cpp82
1 files changed, 52 insertions, 30 deletions
diff --git a/alc/backends/wasapi.cpp b/alc/backends/wasapi.cpp
index 9c2553d1..d06d7401 100644
--- a/alc/backends/wasapi.cpp
+++ b/alc/backends/wasapi.cpp
@@ -176,10 +176,11 @@ overloaded(Ts...) -> overloaded<Ts...>;
/* Scales the given reftime value, rounding the result. */
-inline uint RefTime2Samples(const ReferenceTime &val, uint srate)
+template<typename T>
+constexpr uint RefTime2Samples(const ReferenceTime &val, T srate) noexcept
{
const auto retval = (val*srate + ReferenceTime{seconds{1}}/2) / seconds{1};
- return static_cast<uint>(mini64(retval, std::numeric_limits<uint>::max()));
+ return static_cast<uint>(std::min<decltype(retval)>(retval, std::numeric_limits<uint>::max()));
}
@@ -1637,21 +1638,22 @@ HRESULT WasapiPlayback::resetProxy()
if(FAILED(hr))
ERR("Failed to get max frames: 0x%08lx\n", hr);
else
- TRACE("Max frames: %u\n", maxFrames);
+ TRACE("Max sample frames: %u\n", maxFrames);
for(UINT32 i{1};i < fmtcount;++i)
{
- WAVEFORMATEX *other{};
- hr = fmtenum->GetFormat(i, &other);
+ WAVEFORMATEX *otherFormat{};
+ hr = fmtenum->GetFormat(i, &otherFormat);
if(FAILED(hr))
ERR("Failed to format %u: 0x%08lx\n", i+1, hr);
else
{
- TraceFormat("Other mix format", other);
- hr = audio.mClient->GetMaxFrameCount(other, &maxFrames);
+ TraceFormat("Other mix format", otherFormat);
+ UINT32 otherMaxFrames{};
+ hr = audio.mClient->GetMaxFrameCount(otherFormat, &otherMaxFrames);
if(FAILED(hr))
ERR("Failed to get max frames: 0x%08lx\n", hr);
else
- TRACE("Max frames: %u\n", maxFrames);
+ TRACE("Max sample frames: %u\n", otherMaxFrames);
}
}
@@ -1716,16 +1718,16 @@ HRESULT WasapiPlayback::resetProxy()
{
switch(chans)
{
- case DevFmtMono: return ChannelMask_Mono;
- case DevFmtStereo: return ChannelMask_Stereo;
- case DevFmtQuad: return ChannelMask_Quad;
- case DevFmtX51: return isRear51 ? ChannelMask_X51Rear : ChannelMask_X51;
- case DevFmtX61: return ChannelMask_X61;
- case DevFmtX3D71:
- case DevFmtX71: return ChannelMask_X71;
- case DevFmtX714: return ChannelMask_X714;
- case DevFmtAmbi3D:
- break;
+ case DevFmtMono: return ChannelMask_Mono;
+ case DevFmtStereo: return ChannelMask_Stereo;
+ case DevFmtQuad: return ChannelMask_Quad;
+ case DevFmtX51: return isRear51 ? ChannelMask_X51Rear : ChannelMask_X51;
+ case DevFmtX61: return ChannelMask_X61;
+ case DevFmtX3D71:
+ case DevFmtX71: return ChannelMask_X71;
+ case DevFmtX714: return ChannelMask_X714;
+ case DevFmtAmbi3D:
+ break;
}
return ChannelMask_Stereo;
};
@@ -1761,21 +1763,41 @@ HRESULT WasapiPlayback::resetProxy()
setDefaultWFXChannelOrder();
- /* TODO: Get the real update and buffer size. Does
- * ISpatialAudioClient::GetMaxFrameCount give the buffer size, update
- * size, or neither? According to MSDN, it
- *
- * "Gets the maximum possible frame count per processing pass."
+ /* FIXME: Get the real update and buffer size. Presumably the actual
+ * device is configured once ActivateSpatialAudioStream succeeds, and
+ * an IAudioClient from the same IMMDevice accesses the same device
+ * configuration. This isn't obviously correct, but for now assume
+ * IAudioClient::GetDevicePeriod returns the current device period time
+ * that ISpatialAudioObjectRenderStream will try to wake up at.
*
- * If it tries to keep a full buffer, the max possible could be a full
- * buffer for the first update or underrun. Though if it always does a
- * period at a time, it doesn't make sense for a pass to be less.
- *
- * Perhaps activating a normal IAudioClient to get the period size is
- * the proper thing to do (still won't get us the buffer size though).
+ * Unfortunately this won't get the buffer size of the
+ * ISpatialAudioObjectRenderStream, so we only assume there's two
+ * periods.
*/
- mOrigBufferSize = mDevice->BufferSize;
mOrigUpdateSize = mDevice->UpdateSize;
+ mOrigBufferSize = mOrigUpdateSize*2;
+ ReferenceTime per_time{ReferenceTime{seconds{mDevice->UpdateSize}} / mDevice->Frequency};
+
+ ComPtr<IAudioClient> tmpClient;
+ hr = sDeviceHelper->activateAudioClient(mMMDev, __uuidof(IAudioClient),
+ al::out_ptr(tmpClient));
+ if(FAILED(hr))
+ ERR("Failed to activate audio client: 0x%08lx\n", hr);
+ else
+ {
+ hr = tmpClient->GetDevicePeriod(&reinterpret_cast<REFERENCE_TIME&>(per_time), nullptr);
+ if(FAILED(hr))
+ ERR("Failed to get device period: 0x%08lx\n", hr);
+ else
+ {
+ mOrigUpdateSize = RefTime2Samples(per_time, mFormat.Format.nSamplesPerSec);
+ mOrigBufferSize = mOrigUpdateSize*2;
+ }
+ }
+ tmpClient = nullptr;
+
+ mDevice->UpdateSize = RefTime2Samples(per_time, mDevice->Frequency);
+ mDevice->BufferSize = mDevice->UpdateSize*2;
mResampler = nullptr;
mResampleBuffer = nullptr;