2025-11-28 00:35:46 +09:00

590 lines
17 KiB
C++

// THIS CODE AND INFORMATION IS PROVIDED "AS IS" WITHOUT WARRANTY OF
// ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING BUT NOT LIMITED TO
// THE IMPLIED WARRANTIES OF MERCHANTABILITY AND/OR FITNESS FOR A
// PARTICULAR PURPOSE.
//
// Copyright (c) Microsoft Corporation. All rights reserved
//
#include "StdAfx.h"
#include <assert.h>
#include <avrt.h>
#include "WASAPIRenderer.h"
//
// A simple WASAPI Render client.
//
CWASAPIRenderer::CWASAPIRenderer(IMMDevice *Endpoint) :
_RefCount(1),
_Endpoint(Endpoint),
_AudioClient(NULL),
_RenderClient(NULL),
_RenderThread(NULL),
_ShutdownEvent(NULL),
_MixFormat(NULL),
_RenderBufferQueue(NULL),
_AudioSamplesReadyEvent(NULL)
{
_Endpoint->AddRef(); // Since we're holding a copy of the endpoint, take a reference to it. It'll be released in Shutdown();
}
//
// Empty destructor - everything should be released in the Shutdown() call.
//
CWASAPIRenderer::~CWASAPIRenderer(void)
{
}
//
// Initialize WASAPI in event driven mode, associate the audio client with our samples ready event handle, and retrieve
// a render client for the transport.
//
bool CWASAPIRenderer::InitializeAudioEngine()
{
REFERENCE_TIME bufferDuration = _EngineLatencyInMS*10000;
HRESULT hr = _AudioClient->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
bufferDuration,
bufferDuration ,
_MixFormat,
NULL);
//
// When rendering in exclusive mode event driven, the HDAudio specification requires that the buffers handed to the device must
// be aligned on a 128 byte boundary. When the buffer is initialized and the resulting buffer size would not be 128 byte aligned,
// we need to "swizzle" the periodicity of the engine to ensure that the buffers are properly aligned.
//
if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED)
{
UINT32 bufferSize;
printf("Buffers not aligned. Aligning the buffers... \n");
//
// Retrieve the buffer size for the audio client. The buffer size returned is aligned to the nearest 128 byte
// boundary given the input buffer duration.
//
hr = _AudioClient->GetBufferSize(&bufferSize);
if(FAILED(hr))
{
printf("Unable to get audio client buffer: %x. \n", hr);
return false;
}
//
// Release old AudioClient
//
SafeRelease(&_AudioClient);
//
// Calculate the new aligned periodicity. We do that by taking the buffer size returned (which is in frames),
// multiplying it by the frames/second in the render format (which gets us seconds per buffer), then converting the
// seconds/buffer calculation into a REFERENCE_TIME.
//
bufferDuration = (REFERENCE_TIME)(10000.0 * // (REFERENCE_TIME / ms) *
1000 * // (ms / s) *
bufferSize / // frames /
_MixFormat->nSamplesPerSec + // (frames / s)
0.5); // rounding
//
// Now reactivate an IAudioClient object on our preferred endpoint and reinitialize AudioClient
//
hr = _Endpoint->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL, reinterpret_cast<void **>(&_AudioClient));
if (FAILED(hr))
{
printf("Unable to activate audio client: %x.\n", hr);
return false;
}
hr = _AudioClient->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
bufferDuration,
bufferDuration,
_MixFormat,
NULL);
if (FAILED(hr))
{
printf("Unable to reinitialize audio client: %x \n", hr);
return false;
}
}
else 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);
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 get set event handle: %x.\n", hr);
return false;
}
//
// When rendering in event driven mode, we'll always have exactly a buffer's size worth of data
// available every time we wake up.
//
_BufferSizePerPeriod = _BufferSize;
hr = _AudioClient->GetService(IID_PPV_ARGS(&_RenderClient));
if (FAILED(hr))
{
printf("Unable to get new render client: %x.\n", hr);
return false;
}
return true;
}
//
// Retrieve the format we'll use to rendersamples.
//
// Start with the mix format and see if the endpoint can render that. If not, try
// the mix format converted to an integer form (most audio solutions don't support floating
// point rendering and the mix format is usually a floating point format).
//
bool CWASAPIRenderer::LoadFormat()
{
HRESULT hr = _AudioClient->GetMixFormat(&_MixFormat);
if (FAILED(hr))
{
printf("Unable to get mix format on audio client: %x.\n", hr);
return false;
}
assert(_MixFormat != NULL);
hr = _AudioClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE,_MixFormat, NULL);
if (hr == AUDCLNT_E_UNSUPPORTED_FORMAT)
{
printf("Device does not natively support the mix format, converting to PCM.\n");
//
// If the mix format is a float format, just try to convert the format to PCM.
//
if (_MixFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
{
_MixFormat->wFormatTag = WAVE_FORMAT_PCM;
_MixFormat->wBitsPerSample = 16;
_MixFormat->nBlockAlign = (_MixFormat->wBitsPerSample / 8) * _MixFormat->nChannels;
_MixFormat->nAvgBytesPerSec = _MixFormat->nSamplesPerSec*_MixFormat->nBlockAlign;
}
else if (_MixFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
reinterpret_cast<WAVEFORMATEXTENSIBLE *>(_MixFormat)->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
{
WAVEFORMATEXTENSIBLE *waveFormatExtensible = reinterpret_cast<WAVEFORMATEXTENSIBLE *>(_MixFormat);
waveFormatExtensible->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
waveFormatExtensible->Format.wBitsPerSample = 16;
waveFormatExtensible->Format.nBlockAlign = (_MixFormat->wBitsPerSample / 8) * _MixFormat->nChannels;
waveFormatExtensible->Format.nAvgBytesPerSec = waveFormatExtensible->Format.nSamplesPerSec*waveFormatExtensible->Format.nBlockAlign;
waveFormatExtensible->Samples.wValidBitsPerSample = 16;
}
else
{
printf("Mix format is not a floating point format.\n");
return false;
}
hr = _AudioClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE,_MixFormat,NULL);
if (FAILED(hr))
{
printf("Format is not supported \n");
return false;
}
}
_FrameSize = _MixFormat->nBlockAlign;
if (!CalculateMixFormatType())
{
return false;
}
return true;
}
//
// Crack open the mix format and determine what kind of samples are being rendered.
//
bool CWASAPIRenderer::CalculateMixFormatType()
{
if (_MixFormat->wFormatTag == WAVE_FORMAT_PCM ||
_MixFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
reinterpret_cast<WAVEFORMATEXTENSIBLE *>(_MixFormat)->SubFormat == KSDATAFORMAT_SUBTYPE_PCM)
{
if (_MixFormat->wBitsPerSample == 16)
{
_RenderSampleType = SampleType16BitPCM;
}
else
{
printf("Unknown PCM integer sample type\n");
return false;
}
}
else if (_MixFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
(_MixFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
reinterpret_cast<WAVEFORMATEXTENSIBLE *>(_MixFormat)->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))
{
_RenderSampleType = SampleTypeFloat;
}
else
{
printf("unrecognized device format.\n");
return false;
}
return true;
}
//
// Initialize the renderer.
//
bool CWASAPIRenderer::Initialize(UINT32 EngineLatency)
{
//
// Create our shutdown and samples ready events- we want auto reset events that start in the not-signaled state.
//
_ShutdownEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
if (_ShutdownEvent == NULL)
{
printf("Unable to create shutdown event: %d.\n", GetLastError());
return false;
}
_AudioSamplesReadyEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
if (_AudioSamplesReadyEvent == NULL)
{
printf("Unable to create samples ready event: %d.\n", GetLastError());
return false;
}
//
// Now activate an IAudioClient object on our preferred endpoint and retrieve the mix format for that endpoint.
//
HRESULT hr = _Endpoint->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL, reinterpret_cast<void **>(&_AudioClient));
if (FAILED(hr))
{
printf("Unable to activate audio client: %x.\n", hr);
return false;
}
//
// Load the MixFormat. This may differ depending on the shared mode used
//
if (!LoadFormat())
{
printf("Failed to load the mix format \n");
return false;
}
//
// Remember our configured latency in case we'll need it for a stream switch later.
//
_EngineLatencyInMS = EngineLatency;
if (!InitializeAudioEngine())
{
return false;
}
return true;
}
//
// Shut down the render code and free all the resources.
//
void CWASAPIRenderer::Shutdown()
{
if (_RenderThread)
{
SetEvent(_ShutdownEvent);
WaitForSingleObject(_RenderThread, INFINITE);
CloseHandle(_RenderThread);
_RenderThread = NULL;
}
if (_ShutdownEvent)
{
CloseHandle(_ShutdownEvent);
_ShutdownEvent = NULL;
}
if (_AudioSamplesReadyEvent)
{
CloseHandle(_AudioSamplesReadyEvent);
_AudioSamplesReadyEvent = NULL;
}
SafeRelease(&_Endpoint);
SafeRelease(&_AudioClient);
SafeRelease(&_RenderClient);
if (_MixFormat)
{
CoTaskMemFree(_MixFormat);
_MixFormat = NULL;
}
}
//
// Start rendering - Create the render thread and start rendering the buffer.
//
bool CWASAPIRenderer::Start(RenderBuffer *RenderBufferQueue)
{
HRESULT hr;
_RenderBufferQueue = RenderBufferQueue;
//
// Now create the thread which is going to drive the renderer.
//
_RenderThread = CreateThread(NULL, 0, WASAPIRenderThread, this, 0, NULL);
if (_RenderThread == NULL)
{
printf("Unable to create transport thread: %x.", GetLastError());
return false;
}
//
// We want to pre-roll the first buffer's worth of data into the pipeline. That way the audio engine won't glitch on startup.
//
{
BYTE *pData;
if (_RenderBufferQueue != NULL)
{
//
// Remove the buffer from the queue.
//
RenderBuffer *renderBuffer = _RenderBufferQueue;
_RenderBufferQueue = renderBuffer->_Next;
DWORD bufferLengthInFrames = renderBuffer->_BufferLength / _FrameSize;
hr = _RenderClient->GetBuffer(bufferLengthInFrames, &pData);
if (FAILED(hr))
{
printf("Failed to get buffer: %x.\n", hr);
return false;
}
CopyMemory(pData, renderBuffer->_Buffer, renderBuffer->_BufferLength);
hr = _RenderClient->ReleaseBuffer(bufferLengthInFrames, 0);
delete renderBuffer;
}
else
{
hr = _RenderClient->GetBuffer(_BufferSize, &pData);
if (FAILED(hr))
{
printf("Failed to get buffer: %x.\n", hr);
return false;
}
hr = _RenderClient->ReleaseBuffer(_BufferSize, AUDCLNT_BUFFERFLAGS_SILENT);
}
if (FAILED(hr))
{
printf("Failed to release buffer: %x.\n", hr);
return false;
}
}
//
// We're ready to go, start rendering!
//
hr = _AudioClient->Start();
if (FAILED(hr))
{
printf("Unable to start render client: %x.\n", hr);
return false;
}
return true;
}
//
// Stop the renderer.
//
void CWASAPIRenderer::Stop()
{
HRESULT hr;
//
// Tell the render thread to shut down, wait for the thread to complete then clean up all the stuff we
// allocated in Start().
//
if (_ShutdownEvent)
{
SetEvent(_ShutdownEvent);
}
hr = _AudioClient->Stop();
if (FAILED(hr))
{
printf("Unable to stop audio client: %x\n", hr);
}
if (_RenderThread)
{
WaitForSingleObject(_RenderThread, INFINITE);
CloseHandle(_RenderThread);
_RenderThread = NULL;
}
//
// Drain the buffers in the render buffer queue.
//
while (_RenderBufferQueue != NULL)
{
RenderBuffer *renderBuffer = _RenderBufferQueue;
_RenderBufferQueue = renderBuffer->_Next;
delete renderBuffer;
}
}
//
// Render thread - processes samples from the audio engine
//
DWORD CWASAPIRenderer::WASAPIRenderThread(LPVOID Context)
{
CWASAPIRenderer *renderer = static_cast<CWASAPIRenderer *>(Context);
return renderer->DoRenderThread();
}
DWORD CWASAPIRenderer::DoRenderThread()
{
bool stillPlaying = true;
HANDLE waitArray[2] = {_ShutdownEvent, _AudioSamplesReadyEvent};
HANDLE mmcssHandle = NULL;
DWORD mmcssTaskIndex = 0;
HRESULT hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
if (FAILED(hr))
{
printf("Unable to initialize COM in render thread: %x\n", hr);
return hr;
}
if (!DisableMMCSS)
{
mmcssHandle = AvSetMmThreadCharacteristics(L"Audio", &mmcssTaskIndex);
if (mmcssHandle == NULL)
{
printf("Unable to enable MMCSS on render thread: %d\n", GetLastError());
}
}
while (stillPlaying)
{
HRESULT hr;
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
switch (waitResult)
{
case WAIT_OBJECT_0 + 0: // _ShutdownEvent
stillPlaying = false; // We're done, exit the loop.
break;
case WAIT_OBJECT_0 + 1: // _AudioSamplesReadyEvent
//
// We need to provide the next buffer of samples to the audio renderer.
//
BYTE *pData;
//
// When rendering in event driven mode, every time we wake up, we'll have a buffer's worth of data available, so if we have
// data in our queue, render it.
//
if (_RenderBufferQueue == NULL)
{
stillPlaying = false;
}
else
{
RenderBuffer *renderBuffer = _RenderBufferQueue;
_RenderBufferQueue = renderBuffer->_Next;
UINT32 framesToWrite = renderBuffer->_BufferLength / _FrameSize;
hr = _RenderClient->GetBuffer(framesToWrite, &pData);
if (SUCCEEDED(hr))
{
//
// Copy data from the render buffer to the output buffer and bump our render pointer.
//
CopyMemory(pData, renderBuffer->_Buffer, framesToWrite*_FrameSize);
hr = _RenderClient->ReleaseBuffer(framesToWrite, 0);
if (!SUCCEEDED(hr))
{
printf("Unable to release buffer: %x\n", hr);
stillPlaying = false;
}
}
else
{
printf("Unable to get buffer: %x\n", hr);
stillPlaying = false;
}
//
// We're done with this set of samples, free it.
//
delete renderBuffer;
}
break;
}
}
if (!DisableMMCSS)
{
AvRevertMmThreadCharacteristics(mmcssHandle);
}
CoUninitialize();
return 0;
}
//
// IUnknown
//
HRESULT CWASAPIRenderer::QueryInterface(REFIID Iid, void **Object)
{
if (Object == NULL)
{
return E_POINTER;
}
*Object = NULL;
if (Iid == IID_IUnknown)
{
*Object = static_cast<IUnknown *>(this);
AddRef();
}
else
{
return E_NOINTERFACE;
}
return S_OK;
}
ULONG CWASAPIRenderer::AddRef()
{
return InterlockedIncrement(&_RefCount);
}
ULONG CWASAPIRenderer::Release()
{
ULONG returnValue = InterlockedDecrement(&_RefCount);
if (returnValue == 0)
{
delete this;
}
return returnValue;
}