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C++ IAudioClient::Initialize方法代码示例

本文整理汇总了C++中IAudioClient::Initialize方法的典型用法代码示例。如果您正苦于以下问题:C++ IAudioClient::Initialize方法的具体用法?C++ IAudioClient::Initialize怎么用?C++ IAudioClient::Initialize使用的例子?那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在IAudioClient的用法示例。


在下文中一共展示了IAudioClient::Initialize方法的11个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。

示例1: BlankAudioPlayback

    BlankAudioPlayback(CTSTR lpDevice)
    {
        const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
        const IID IID_IMMDeviceEnumerator    = __uuidof(IMMDeviceEnumerator);
        const IID IID_IAudioClient           = __uuidof(IAudioClient);
        const IID IID_IAudioRenderClient     = __uuidof(IAudioRenderClient);

        HRESULT err;
        err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
        if(FAILED(err))
            CrashError(TEXT("Could not create IMMDeviceEnumerator: 0x%08lx"), err);

        if (scmpi(lpDevice, TEXT("Default")) == 0)
            err = mmEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &mmDevice);
        else
            err = mmEnumerator->GetDevice(lpDevice, &mmDevice);
        if(FAILED(err))
            CrashError(TEXT("Could not create IMMDevice"));

        err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
        if(FAILED(err))
            CrashError(TEXT("Could not create IAudioClient"));

        WAVEFORMATEX *pwfx;
        err = mmClient->GetMixFormat(&pwfx);
        if(FAILED(err))
            CrashError(TEXT("Could not get mix format from audio client"));

        UINT inputBlockSize = pwfx->nBlockAlign;

        err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, 0, ConvertMSTo100NanoSec(1000), 0, pwfx, NULL);
        if(FAILED(err))
            CrashError(TEXT("Could not initialize audio client, error = %08lX"), err);

        err = mmClient->GetService(IID_IAudioRenderClient, (void**)&mmRender);
        if(FAILED(err))
            CrashError(TEXT("Could not get audio render client"));

        //----------------------------------------------------------------

        UINT bufferFrameCount;
        err = mmClient->GetBufferSize(&bufferFrameCount);
        if(FAILED(err))
            CrashError(TEXT("Could not get audio buffer size"));

        BYTE *lpData;
        err = mmRender->GetBuffer(bufferFrameCount, &lpData);
        if(FAILED(err))
            CrashError(TEXT("Could not get audio buffer"));

        zero(lpData, bufferFrameCount*inputBlockSize);

        mmRender->ReleaseBuffer(bufferFrameCount, 0);//AUDCLNT_BUFFERFLAGS_SILENT); //probably better if it doesn't know

        if(FAILED(mmClient->Start()))
            CrashError(TEXT("Could not start audio source"));
    }
开发者ID:373137461,项目名称:OBS,代码行数:57,代码来源:BlankAudioPlayback.cpp

示例2: initialize_patch

HRESULT __stdcall initialize_patch(
    IAudioClient* this_, AUDCLNT_SHAREMODE ShareMode, DWORD StreamFlags, 
    REFERENCE_TIME hnsBufferDuration, REFERENCE_TIME hnsPeriodicity, 
    const WAVEFORMATEX* pFormat, LPCGUID AudioSessionGuid)
{
    // synchronize initializing so it doesn't happen while streams are being flushed
    HANDLE audio_router_mutex = OpenMutexW(SYNCHRONIZE, FALSE, L"Local\\audio-router-mutex");
    assert(audio_router_mutex != NULL);
    if(audio_router_mutex)
    {
        DWORD res = WaitForSingleObject(audio_router_mutex, INFINITE);
        assert(res == WAIT_OBJECT_0);
    }

    IAudioClient* proxy = get_duplicate(this_)->proxy;
    LPCGUID guid = ((GUID***)this_)[0][17];
    DWORD_PTR* old_vftptr = swap_vtable(this_);
    HRESULT hr = proxy->Initialize(
        ShareMode, 
        StreamFlags | 
        AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED | 
        AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED,
        hnsBufferDuration, 
        hnsPeriodicity, 
        pFormat, 
        guid);
    ((DWORD_PTR**)this_)[0] = old_vftptr;
    if(hr != S_OK)
        tell_error(hr);
    else
        *((WORD***)this_)[0][18] = pFormat->nBlockAlign;
    if(hr == S_OK)
    {
        for(iaudioclient_duplicate* next = get_duplicate(this_)->next;
            next != NULL; next = next->next)
        {
            HRESULT hr2 = next->proxy->Initialize(
                ShareMode,
                StreamFlags |
                AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM |
                AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY |
                AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED |
                AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED,
                hnsBufferDuration,
                hnsPeriodicity,
                pFormat,
                guid);
            if(hr2 != S_OK)
                tell_error(hr2);
        }
    }

    ReleaseMutex(audio_router_mutex);
    CloseHandle(audio_router_mutex);
    return hr;
}
开发者ID:fretelweb,项目名称:audio-router,代码行数:56,代码来源:patch_iaudioclient.cpp

示例3: InitializeAudioEngine

//
//  Initialize WASAPI in event driven mode, associate the audio client with our samples ready event handle, retrieve 
//  a capture client for the transport, create the capture thread and start the audio engine.
//
bool CWASAPICapture::InitializeAudioEngine()
{
    HRESULT hr = _AudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_NOPERSIST, _EngineLatencyInMS*10000, 0, MixFormat(), NULL);
    PersistentAssert(SUCCEEDED(hr), "_AudioClient->Initialize failed");
    
    //
    //  Retrieve the buffer size for the audio client.
    //
    hr = _AudioClient->GetBufferSize(&_BufferSize);
    PersistentAssert(SUCCEEDED(hr), "_AudioClient->GetBufferSize failed");

    hr = _AudioClient->GetService(IID_PPV_ARGS(&_CaptureClient));
    PersistentAssert(SUCCEEDED(hr), "_AudioClient->GetService failed");

    return true;
}
开发者ID:kbinani,项目名称:dxrip,代码行数:20,代码来源:AudioCapture.cpp

示例4: Initialize

bool MMDeviceAudioSource::Initialize(bool bMic, CTSTR lpID)
{
    const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
    const IID IID_IMMDeviceEnumerator    = __uuidof(IMMDeviceEnumerator);
    const IID IID_IAudioClient           = __uuidof(IAudioClient);
    const IID IID_IAudioCaptureClient    = __uuidof(IAudioCaptureClient);

    HRESULT err;
    err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDeviceEnumerator = %08lX"), (BOOL)bMic, err);
        return false;
    }

    bIsMic = bMic;

    if (bIsMic) {
        BOOL bMicSyncFixHack = GlobalConfig->GetInt(TEXT("Audio"), TEXT("UseMicSyncFixHack"));
        angerThreshold = bMicSyncFixHack ? 40 : 1000;
    }

    if (scmpi(lpID, TEXT("Default")) == 0)
        err = mmEnumerator->GetDefaultAudioEndpoint(bMic ? eCapture : eRender, bMic ? eCommunications : eConsole, &mmDevice);
    else
        err = mmEnumerator->GetDevice(lpID, &mmDevice);

    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDevice = %08lX"), (BOOL)bMic, err);
        return false;
    }

    err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IAudioClient = %08lX"), (BOOL)bMic, err);
        return false;
    }

    //-----------------------------------------------------------------
    // get name

    IPropertyStore *store;
    if(SUCCEEDED(mmDevice->OpenPropertyStore(STGM_READ, &store)))
    {
        PROPVARIANT varName;

        PropVariantInit(&varName);
        if(SUCCEEDED(store->GetValue(PKEY_Device_FriendlyName, &varName)))
        {
            CWSTR wstrName = varName.pwszVal;
            strDeviceName = wstrName;
        }

        store->Release();
    }

    if(bMic)
    {
        Log(TEXT("------------------------------------------"));
        Log(TEXT("Using auxilary audio input: %s"), GetDeviceName());

        bUseQPC = GlobalConfig->GetInt(TEXT("Audio"), TEXT("UseMicQPC")) != 0;
        if (bUseQPC)
            Log(TEXT("Using Mic QPC timestamps"));
    }
    else
    {
        Log(TEXT("------------------------------------------"));
        Log(TEXT("Using desktop audio input: %s"), GetDeviceName());

        bUseVideoTime = AppConfig->GetInt(TEXT("Audio"), TEXT("SyncToVideoTime")) != 0;
        SetTimeOffset(GlobalConfig->GetInt(TEXT("Audio"), TEXT("GlobalAudioTimeAdjust")));
    }

    //-----------------------------------------------------------------
    // get format

    WAVEFORMATEX *pwfx;
    err = mmClient->GetMixFormat(&pwfx);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get mix format from audio client = %08lX"), (BOOL)bMic, err);
        return false;
    }

    bool  bFloat;
    UINT  inputChannels;
    UINT  inputSamplesPerSec;
    UINT  inputBitsPerSample;
    UINT  inputBlockSize;
    DWORD inputChannelMask = 0;
    WAVEFORMATEXTENSIBLE *wfext = NULL;

    //the internal audio engine should always use floats (or so I read), but I suppose just to be safe better check
    if(pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
    {
        wfext = (WAVEFORMATEXTENSIBLE*)pwfx;
        inputChannelMask = wfext->dwChannelMask;
//.........这里部分代码省略.........
开发者ID:bradparks,项目名称:OBS,代码行数:101,代码来源:MMDeviceAudioSource.cpp

示例5: PlayAudio

void PlayAudio()
{
    REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;	// microseconds, so this is 1 seconds
    REFERENCE_TIME hnsActualDuration;

    HRESULT hr;

    IMMDeviceEnumerator *pEnumerator = NULL;
    IMMDevice *pDevice = NULL;
    IAudioClient *pAudioClient = NULL;
    IAudioRenderClient *pRenderClient = NULL;
    WAVEFORMATEX *pwfx = NULL;
    UINT32 bufferFrameCount;
    UINT32 numFramesAvailable;
    UINT32 numFramesPadding;
    BYTE *pData;
    DWORD flags = 0;


    hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&pEnumerator);
    EXIT_ON_ERROR(hr);

    hr = pEnumerator->GetDefaultAudioEndpoint(
        eRender, eConsole, &pDevice);
    EXIT_ON_ERROR(hr);

    hr = pDevice->Activate(
            IID_IAudioClient, CLSCTX_ALL,
            NULL, (void**)&pAudioClient);
    EXIT_ON_ERROR(hr);

    hr = pAudioClient->GetMixFormat(&pwfx);
    EXIT_ON_ERROR(hr);

    hr = pAudioClient->Initialize(
            AUDCLNT_SHAREMODE_SHARED,
            0,
            hnsRequestedDuration,
            0,
            pwfx,
            NULL);
    EXIT_ON_ERROR(hr);

    // Get the actual size of the allocated buffer.
    hr = pAudioClient->GetBufferSize(&bufferFrameCount);
    EXIT_ON_ERROR(hr);

    hr = pAudioClient->GetService(
                         IID_IAudioRenderClient,
                         (void**)&pRenderClient);
    EXIT_ON_ERROR(hr);

    // Grab the entire buffer for the initial fill operation.
    hr = pRenderClient->GetBuffer(bufferFrameCount, &pData);
    EXIT_ON_ERROR(hr);

    // load initial data
    hr = LoadAudioBuffer(bufferFrameCount, pData, pwfx, &flags);
    EXIT_ON_ERROR(hr);

    hr = pRenderClient->ReleaseBuffer(bufferFrameCount, flags);
    EXIT_ON_ERROR(hr);

    // Calculate the actual duration of the allocated buffer.
    hnsActualDuration = (REFERENCE_TIME)((double)REFTIMES_PER_SEC * bufferFrameCount / pwfx->nSamplesPerSec);

    hr = pAudioClient->Start();  // Start playing.
    EXIT_ON_ERROR(hr);

    // Each loop fills about half of the shared buffer.
    while (flags != AUDCLNT_BUFFERFLAGS_SILENT)
    {
                // Sleep for half the buffer duration.
        Sleep((DWORD)(hnsActualDuration/REFTIMES_PER_MILLISEC/2));

        // See how much buffer space is available.
        hr = pAudioClient->GetCurrentPadding(&numFramesPadding);
        EXIT_ON_ERROR(hr)

        numFramesAvailable = bufferFrameCount - numFramesPadding;

        // Grab all the available space in the shared buffer.
        hr = pRenderClient->GetBuffer(numFramesAvailable, &pData);
        EXIT_ON_ERROR(hr)

        // Get next 1/2-second of data from the audio source.
        hr = LoadAudioBuffer(numFramesAvailable, pData, pwfx, &flags);
        EXIT_ON_ERROR(hr)

        hr = pRenderClient->ReleaseBuffer(numFramesAvailable, flags);
        EXIT_ON_ERROR(hr)
    }

    // Wait for last data in buffer to play before stopping.
    Sleep((DWORD)(hnsActualDuration/REFTIMES_PER_MILLISEC/2));

    hr = pAudioClient->Stop();  // Stop playing.
    EXIT_ON_ERROR(hr);


//.........这里部分代码省略.........
开发者ID:noggs,项目名称:WASAPITest,代码行数:101,代码来源:WASAPI_Generation.cpp

示例6: Initialize

bool MMDeviceAudioSource::Initialize(bool bMic, CTSTR lpID)
{
    const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
    const IID IID_IMMDeviceEnumerator    = __uuidof(IMMDeviceEnumerator);
    const IID IID_IAudioClient           = __uuidof(IAudioClient);
    const IID IID_IAudioCaptureClient    = __uuidof(IAudioCaptureClient);

    HRESULT err;
    err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDeviceEnumerator = %08lX"), (BOOL)bMic, err);
        return false;
    }

    if(bMic)
        err = mmEnumerator->GetDevice(lpID, &mmDevice);
    else
        err = mmEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &mmDevice);

    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDevice = %08lX"), (BOOL)bMic, err);
        return false;
    }

    err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IAudioClient = %08lX"), (BOOL)bMic, err);
        return false;
    }

    WAVEFORMATEX *pwfx;
    err = mmClient->GetMixFormat(&pwfx);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get mix format from audio client = %08lX"), (BOOL)bMic, err);
        return false;
    }

    String strName = GetDeviceName();
    if(bMic)
    {
        Log(TEXT("------------------------------------------"));
        Log(TEXT("Using auxilary audio input: %s"), strName.Array());
    }

    //the internal audio engine should always use floats (or so I read), but I suppose just to be safe better check
    if(pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
    {
        WAVEFORMATEXTENSIBLE *wfext = (WAVEFORMATEXTENSIBLE*)pwfx;
        inputChannelMask = wfext->dwChannelMask;

        if(wfext->SubFormat != KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
        {
            AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
            return false;
        }
    }
    else if(pwfx->wFormatTag != WAVE_FORMAT_IEEE_FLOAT)
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
        return false;
    }

    inputChannels      = pwfx->nChannels;
    inputBitsPerSample = 32;
    inputBlockSize     = pwfx->nBlockAlign;
    inputSamplesPerSec = pwfx->nSamplesPerSec;

    DWORD flags = bMic ? 0 : AUDCLNT_STREAMFLAGS_LOOPBACK;
    err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, ConvertMSTo100NanoSec(5000), 0, pwfx, NULL);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not initialize audio client, result = %08lX"), (BOOL)bMic, err);
        return false;
    }

    err = mmClient->GetService(IID_IAudioCaptureClient, (void**)&mmCapture);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture client, result = %08lX"), (BOOL)bMic, err);
        return false;
    }

    err = mmClient->GetService(__uuidof(IAudioClock), (void**)&mmClock);
    if(FAILED(err))
    {
        AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture clock, result = %08lX"), (BOOL)bMic, err);
        return false;
    }

    CoTaskMemFree(pwfx);

    //-------------------------------------------------------------------------

    if(inputSamplesPerSec != 44100)
    {
        int errVal;
//.........这里部分代码省略.........
开发者ID:AndrewHolder,项目名称:OBS,代码行数:101,代码来源:MMDeviceAudioSource.cpp

示例7: LoopbackCaptureFor

void LoopbackCaptureFor(IMMDevice* mmDevice, std::string filename, int secs)
{
    // open new file
    MMIOINFO mi = { 0 };

    // some flags cause mmioOpen write to this buffer
    // but not any that we're using
    std::wstring wsFilename(filename.begin(), filename.end()); // mmioOpen wants a wstring
    HMMIO file = mmioOpen(const_cast<LPWSTR>(wsFilename.c_str()), &mi, MMIO_WRITE | MMIO_CREATE);

    time_t startTime = time(nullptr);

    // activate an IAudioClient
    IAudioClient* audioClient;
    HRESULT hr = mmDevice->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr, (void**)&audioClient);
    if (FAILED(hr))
    {
        fprintf(stderr, "IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
        return;
    }

    // get the default device periodicity
    REFERENCE_TIME hnsDefaultDevicePeriod;
    hr = audioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, nullptr);
    if (FAILED(hr))
    {
        fprintf(stderr, "IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
        audioClient->Release();
        return;
    }

    // get the default device format
    WAVEFORMATEX* waveform;
    hr = audioClient->GetMixFormat(&waveform);
    if (FAILED(hr))
    {
        fprintf(stderr, "IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
        CoTaskMemFree(waveform);
        audioClient->Release();
        return;
    }

    // coerce int-16 wave format
    // can do this in-place since we're not changing the size of the format
    // also, the engine will auto-convert from float to int for us
    switch (waveform->wFormatTag)
    {
        case WAVE_FORMAT_IEEE_FLOAT:
            waveform->wFormatTag = WAVE_FORMAT_PCM;
            waveform->wBitsPerSample = BITS_PER_SAMPLE;
            waveform->nBlockAlign = BLOCK_ALIGN;
            waveform->nAvgBytesPerSec = BYTE_RATE;
            break;
        case WAVE_FORMAT_EXTENSIBLE:
        {
            // naked scope for case-local variable
            PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(waveform);
            if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat))
            {
                pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
                pEx->Samples.wValidBitsPerSample = BITS_PER_SAMPLE;
                waveform->wBitsPerSample = BITS_PER_SAMPLE;
                waveform->nBlockAlign = waveform->nChannels * BYTE_PER_SAMPLE;
                waveform->nAvgBytesPerSec = waveform->nBlockAlign * waveform->nSamplesPerSec;
            }
            break;
        }
    }

    MMCKINFO ckRIFF = { 0 };
    MMCKINFO ckData = { 0 };
    hr = WriteWaveHeader(file, waveform, &ckRIFF, &ckData);

    // create a periodic waitable timer
    HANDLE hWakeUp = CreateWaitableTimer(nullptr, FALSE, nullptr);
    UINT32 nBlockAlign = waveform->nBlockAlign;

    // call IAudioClient::Initialize
    // note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
    // do not work together...
    // the "data ready" event never gets set
    // so we're going to do a timer-driven loop
    hr = audioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_LOOPBACK, 0, 0, waveform, 0);
    if (FAILED(hr))
    {
        fprintf(stderr, "IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
        CloseHandle(hWakeUp);
        audioClient->Release();
        return;
    }

    // free up waveform
    CoTaskMemFree(waveform);

    // activate an IAudioCaptureClient
    IAudioCaptureClient* audioCaptureClient;
    hr = audioClient->GetService(__uuidof(IAudioCaptureClient), (void**)&audioCaptureClient);

    // register with MMCSS
    DWORD nTaskIndex = 0;
//.........这里部分代码省略.........
开发者ID:Nespa32,项目名称:sm_project,代码行数:101,代码来源:sound_capture_loopback.cpp

示例8: __uuidof


//.........这里部分代码省略.........
        break;
        default:
            break;
        }
        pBuffer->SetAudioInfo(pwfx->nSamplesPerSec,pwfx->nBlockAlign,pwfx->nChannels,pwfx->wBitsPerSample,isFloat);
    }

    if (FAILED(hr)) {
        // WriteWaveHeader does its own logging
        CoTaskMemFree(pwfx);
        pAudioClient->Release();
        return hr;
    }

    // create a periodic waitable timer
    HANDLE hWakeUp = CreateWaitableTimer(NULL, FALSE, NULL);
    if (NULL == hWakeUp) {
        DWORD dwErr = GetLastError();
        printf("CreateWaitableTimer failed: last error = %u\n", dwErr);
        CoTaskMemFree(pwfx);
        pAudioClient->Release();
        return HRESULT_FROM_WIN32(dwErr);
    }

    UINT32 nBlockAlign = pwfx->nBlockAlign;
    UINT32 nChannels = pwfx->nChannels;
    nFrames = 0;

    // call IAudioClient::Initialize
    // note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
    // do not work together...
    // the "data ready" event never gets set
    // so we're going to do a timer-driven loop
    hr = pAudioClient->Initialize(
             AUDCLNT_SHAREMODE_SHARED,
             AUDCLNT_STREAMFLAGS_LOOPBACK,
             0, 0, pwfx, 0
         );
    if (FAILED(hr)) {
        printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
        CloseHandle(hWakeUp);
        pAudioClient->Release();
        return hr;
    }
    CoTaskMemFree(pwfx);

    // activate an IAudioCaptureClient
    IAudioCaptureClient *pAudioCaptureClient;
    hr = pAudioClient->GetService(
             __uuidof(IAudioCaptureClient),
             (void**)&pAudioCaptureClient
         );
    if (FAILED(hr)) {
        printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
        CloseHandle(hWakeUp);
        pAudioClient->Release();
        return hr;
    }

    // register with MMCSS
    DWORD nTaskIndex = 0;
    HANDLE hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
    if (NULL == hTask) {
        DWORD dwErr = GetLastError();
        printf("AvSetMmThreadCharacteristics failed: last error = %u\n", dwErr);
        pAudioCaptureClient->Release();
开发者ID:Harpseal,项目名称:sysAudioSpectrogram,代码行数:67,代码来源:loopback-capture.cpp

示例9: main

int main(int argc, char *argv[])
{
    CoInitialize(nullptr);

    listDevices();

    IAudioClient *pAudioClient;
    IMMDevice *device;

    getDefaultDevice(&device);

    HRESULT hr = device->Activate(__uuidof(IAudioClient),
        CLSCTX_ALL, nullptr, (void**)&pAudioClient);
    if (FAILED(hr)) {
        printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
        return hr;
    }

    REFERENCE_TIME hnsDefaultDevicePeriod;
    hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, nullptr);
    if (FAILED(hr)) {
        printf("IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
        pAudioClient->Release();
        return hr;
    }

    // get the default device format
    WAVEFORMATEX *pwfx;
    hr = pAudioClient->GetMixFormat(&pwfx);
    if (FAILED(hr)) {
        printf("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
        CoTaskMemFree(pwfx);
        pAudioClient->Release();
        return hr;
    }

    DVAR(pwfx->wFormatTag);
    DVAR(pwfx->wBitsPerSample);
    DVAR(pwfx->nBlockAlign);
    DVAR(pwfx->nAvgBytesPerSec);

    switch (pwfx->wFormatTag) {
        case WAVE_FORMAT_IEEE_FLOAT:
            pwfx->wFormatTag = WAVE_FORMAT_PCM;
            pwfx->wBitsPerSample = 16;
            pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
            pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
            break;

        case WAVE_FORMAT_EXTENSIBLE:
            {
                // naked scope for case-local variable
                PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
                if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
                    pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
                    pEx->Samples.wValidBitsPerSample = 16;
                    pwfx->wBitsPerSample = 16;
                    pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
                    pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
                } else {
                    printf("Don't know how to coerce mix format to int-16\n");
                    CoTaskMemFree(pwfx);
                    pAudioClient->Release();
                    return E_UNEXPECTED;
                }
            }
            break;

        default:
            printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", pwfx->wFormatTag);
            CoTaskMemFree(pwfx);
            pAudioClient->Release();
            return E_UNEXPECTED;
    }

    DVAR(pwfx->wFormatTag);
    DVAR(pwfx->wBitsPerSample);
    DVAR(pwfx->nBlockAlign);
    DVAR(pwfx->nAvgBytesPerSec);

    hr = pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_LOOPBACK, 0, 0, pwfx, 0 );
    if (FAILED(hr)) {
        printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
        pAudioClient->Release();
        return hr;
    }

    IAudioCaptureClient *pAudioCaptureClient;
    hr = pAudioClient->GetService(__uuidof(IAudioCaptureClient), (void**)&pAudioCaptureClient);
    if (FAILED(hr)) {
        printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
        pAudioClient->Release();
        return hr;
    }


    hr = pAudioClient->Start();
    if (FAILED(hr)) {
        printf("IAudioClient::Start failed: hr = 0x%08x\n", hr);
        pAudioCaptureClient->Release();
//.........这里部分代码省略.........
开发者ID:CltKitakami,项目名称:MyLib,代码行数:101,代码来源:TestAudio.cpp

示例10: _tmain

int _tmain(int argc, _TCHAR* argv[]) {
  IMMDeviceEnumerator *enumerator = 0;
  IMMDevice *device = 0;
  FILE *f;
    f=fopen("c:/1.wav","w");
  CoInitialize(0);
  CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), 
    (void**) &enumerator);
  enumerator->GetDefaultAudioEndpoint(eRender, eConsole, &device);

  HANDLE processOutWrite, processOutRead, processInWrite, processInRead;    

  /*wchar_t processCommand[2000];
  {
    FILE* commandFile;
    fopen_s(&commandFile, "command.txt", "r");
    char cmd[2000];
    fread(cmd, sizeof(char), 2000, commandFile);
    fclose(commandFile);
    size_t count;
    mbstowcs_s(&count, processCommand, cmd, 2000);
  }*/

  /*{	
    //create pipes for plink process
    SECURITY_ATTRIBUTES pipeAttributes = {0};
    pipeAttributes.nLength = sizeof(SECURITY_ATTRIBUTES);
    pipeAttributes.bInheritHandle = TRUE;
    pipeAttributes.lpSecurityDescriptor= NULL;
    CreatePipe(&processOutRead, &processOutWrite, &pipeAttributes, 0);
    CreatePipe(&processInRead,  &processInWrite,  &pipeAttributes, 0);
    STARTUPINFO startupInfo;
    ZeroMemory(&startupInfo, sizeof(STARTUPINFO));
    startupInfo.cb = sizeof(STARTUPINFO);
    startupInfo.hStdError = processOutWrite;
    startupInfo.hStdOutput = processOutWrite;
    startupInfo.hStdInput = processInRead;
    startupInfo.dwFlags |= STARTF_USESTDHANDLES;
    PROCESS_INFORMATION processInfo = {0};
    //launch process
    CreateProcess(NULL, processCommand, NULL, NULL, TRUE, 0, NULL, NULL, &startupInfo, &processInfo);

    //wait for plink to connect to minimze sound delay  (magic number)
    Sleep(2500);

  }*/

  HRESULT hr;
  // activate an IAudioClient
  IAudioClient *audioClient;
  hr = device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL, (void**) &audioClient); 
  if (FAILED(hr)) {
    printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
    return hr;
  }

  // get the default device format
  WAVEFORMATEX *waveFormat;
  hr = audioClient->GetMixFormat(&waveFormat);
  if (FAILED(hr)) {
    printf("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
    CoTaskMemFree(waveFormat);
    audioClient->Release();
    return hr;
  }

  // coerce int-16 wave format
  // can do this in-place since we're not changing the size of the format
  // also, the engine will auto-convert from float to int for us
  switch (waveFormat->wFormatTag) {
  case WAVE_FORMAT_IEEE_FLOAT:
    waveFormat->wFormatTag = WAVE_FORMAT_PCM;
    waveFormat->wBitsPerSample = 16;
    waveFormat->nBlockAlign = waveFormat->nChannels * waveFormat->wBitsPerSample / 8;
    waveFormat->nAvgBytesPerSec = waveFormat->nBlockAlign * waveFormat->nSamplesPerSec;
    break;

  case WAVE_FORMAT_EXTENSIBLE:
    {
      // naked scope for case-local variable
      PWAVEFORMATEXTENSIBLE waveFormatEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(waveFormat);
      if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, waveFormatEx->SubFormat)) {
        waveFormatEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
        waveFormatEx->Samples.wValidBitsPerSample = 16;
        waveFormat->wBitsPerSample = 16;
        waveFormat->nBlockAlign = waveFormat->nChannels * waveFormat->wBitsPerSample / 8;
        waveFormat->nAvgBytesPerSec = waveFormat->nBlockAlign * waveFormat->nSamplesPerSec;
      } else {
        printf("Don't know how to coerce mix format to int-16\n");
        CoTaskMemFree(waveFormat);
        audioClient->Release();
        return E_UNEXPECTED;
      }
    }
    break;

  default:
    printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", waveFormat->wFormatTag);
    CoTaskMemFree(waveFormat);
    audioClient->Release();
//.........这里部分代码省略.........
开发者ID:polyu,项目名称:Cloud_Game,代码行数:101,代码来源:SoundStreamer.cpp

示例11: LoopbackCapture


//.........这里部分代码省略.........
                        pAudioClient->Release();
                        return E_UNEXPECTED;
                    }
                }
                break;

            default:
                printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", pwfx->wFormatTag);
                CoTaskMemFree(pwfx);
                pAudioClient->Release();
                return E_UNEXPECTED;
        }
    }

    // create a periodic waitable timer
    HANDLE hWakeUp = CreateWaitableTimer(NULL, FALSE, NULL);
    if (NULL == hWakeUp) {
        DWORD dwErr = GetLastError();
        printf("CreateWaitableTimer failed: last error = %u\n", dwErr);
        CoTaskMemFree(pwfx);
        pAudioClient->Release();
        return HRESULT_FROM_WIN32(dwErr);
    }

    UINT32 nBlockAlign = pwfx->nBlockAlign;
    UINT32 nBufferSize;
    *pnFrames = 0;

    // call IAudioClient::Initialize
    // note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
    // do not work together...
    // the "data ready" event never gets set
    // so we're going to do a timer-driven loop
    hr = pAudioClient->Initialize(
        AUDCLNT_SHAREMODE_SHARED,
        AUDCLNT_STREAMFLAGS_LOOPBACK,
        0, 0, pwfx, 0
    );
    if (FAILED(hr)) {
        printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
        CloseHandle(hWakeUp);
        pAudioClient->Release();
        return hr;
    }
    CoTaskMemFree(pwfx);

    // Get the buffer size
    hr = pAudioClient->GetBufferSize(&nBufferSize);
    if (FAILED(hr)) {
        printf("IAudioClient::GetBufferSize failed: hr = 0x%08x\n", hr);
        CloseHandle(hWakeUp);
        pAudioClient->Release();
        return hr;
    }

    // Configure the server.  The buffer size returned is in frames
    // so assume stereo, 16 bits per sample to convert from frames to bytes
    server.configure(
        bMono,
        iSampleRateDivisor,
        nBufferSize * 2 * 2);

    // activate an IAudioCaptureClient
    IAudioCaptureClient *pAudioCaptureClient;
    hr = pAudioClient->GetService(
        __uuidof(IAudioCaptureClient),
开发者ID:lengvarsky,项目名称:SimpleProtocolServer,代码行数:67,代码来源:loopback-capture.cpp


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