使用 WASAPI 捕获蓝牙音频数据

Question

I'm working on a small project where I need to mix incoming data from my internal microphone and a bluetooth headset (Bose). I thought I'd use WASAPI for it. However, whilst I can easily read out my internal microphone, this is not the case for my Bluetooth headphones. I followed the example given by the Docs almost to the letter, however I made a small change to be able to choose my own "input device" (internal microphone or headset) using th ID they have been given.

Tho method is the following:

HRESULT RecordAudioStreamBLE(MyAudioSink *pMySink, LPWSTR pwszID)
{

    HRESULT hr;
    REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;
    REFERENCE_TIME hnsActualDuration;
    UINT32 bufferFrameCount;
    UINT32 numFramesAvailable;
    IMMDeviceEnumerator *pEnumerator = NULL;
    IMMDevice *pDevice = NULL;
    IAudioClient *pAudioClient = NULL;
    IAudioCaptureClient *pCaptureClient = NULL;
    WAVEFORMATEX *pwfx = NULL;
    UINT32 packetLength = 0;
    BOOL bDone = FALSE;
    BYTE *pData;
    DWORD flags;

    hr = CoInitialize(0);

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

    hr = pEnumerator->GetDevice(pwszID, &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 size of the allocated buffer.
    hr = pAudioClient->GetBufferSize(&bufferFrameCount);
    EXIT_ON_ERROR(hr)

    hr = pAudioClient->GetService(IID_IAudioCaptureClient,
                                  (void**)&pCaptureClient);
    EXIT_ON_ERROR(hr)

    // Notify the audio sink which format to use.
    hr = pMySink->SetFormat(pwfx);
    EXIT_ON_ERROR(hr)

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

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

    // Each loop fills about half of the shared buffer.
    while (bDone == FALSE)
    {
        // Sleep for half the buffer duration.
        Sleep(hnsActualDuration/REFTIMES_PER_MILLISEC/2);

        hr = pCaptureClient->GetNextPacketSize(&packetLength);
        EXIT_ON_ERROR(hr)

        printf("packet size = %d\n", packetLength);
        while (packetLength != 0)
        {
            // Get the available data in the shared buffer.
            hr = pCaptureClient->GetBuffer( &pData,
                                            &numFramesAvailable,
                                            &flags, NULL, NULL);
            EXIT_ON_ERROR(hr)

            if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
            {
                pData = NULL;  // Tell CopyData to write silence.
            }

            // Copy the available capture data to the audio sink.
            hr = pMySink->CopyData(pData, numFramesAvailable, &bDone);
            EXIT_ON_ERROR(hr)

            hr = pCaptureClient->ReleaseBuffer(numFramesAvailable);
            EXIT_ON_ERROR(hr)

            hr = pCaptureClient->GetNextPacketSize(&packetLength);
            EXIT_ON_ERROR(hr)
        }
    }

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

Exit:
    printf("%s\n", hr);
    CoTaskMemFree(pwfx);
    SAFE_RELEASE(pEnumerator)
    SAFE_RELEASE(pDevice)
    SAFE_RELEASE(pAudioClient)
    SAFE_RELEASE(pCaptureClient)

    return hr;
}

When doing this with the ID for the internal microphone, everything works normally. However, if the ID of the bluetooth device is used I get the following output: (I first list all the active devices and choose the ID in the terminal:

Endpoint 0: "S24D330 (Intel(R) Display Audio)" ({0.0.0.00000000}.{0f483f83-6e29-482e-94b5-fb9cc257a03d})
Endpoint 1: "Hoofdtelefoon (LE-My headphone Hands-Free AG Audio)" ({0.0.0.00000000}.{75c8e0c3-2e44-4538-940d-f8c2ae6424ca})
Endpoint 2: "Hoofdtelefoon (My headphone Stereo)" ({0.0.0.00000000}.{849b4cc2-ed72-4d40-9725-d1ce6b4abfa0})
Endpoint 3: "Luidsprekers (2- High Definition Audio Device)" ({0.0.0.00000000}.{cb8d7625-257e-4fd0-84b8-26de6aeb1e1b})
Endpoint 4: "Microfoon (2- High Definition Audio Device)" ({0.0.1.00000000}.{5edec961-7a46-4554-bdcd-43fb7d9a9d9a})
Endpoint 5: "Hoofdtelefoon (LE-My headphone Hands-Free AG Audio)" ({0.0.1.00000000}.{f937a0fa-1475-495b-81be-7aec0c1c7ea5})
Which input would you like?5
samples per second 16000
packet size = 0
packet size = 0
packet size = 0
packet size = 0
packet size = 0
packet size = 0
packet size = 0
packet size = 0
packet size = 0
^C

The packet size is printed in the loop in the method shown above. Shows that the packet size each time is just 0. Does anybody know how to fix this, and just get "regular" data out of it? Do I maybe need to use a different API? Speed is key however.

With kind regards

Answer 1

Many Bluetooth headsets support both the A2DP profile for stereo playback and the Hands-Free protocol for bidirectional mono communication, but not at the same time.

If one of these protocols is active, the other will stall.

I suspect the problem in your case is that something is playing to the A2DP endpoint "Hoofdtelefoon (My headphone Stereo)" and that is causing all activity on the two HF endpoints to stall.