C++ - XAudio2 - Cracking sound when trying to play a continuous sine sound

Question

Edit: Today I found out that I only encounter this problem when I use my headphones with a cord. It's not the headphones that's the problem, because the same headphones can also be used wireless, and then the problem is gone, similar to when I use my other wireless headphones. I would prefer working with a cord though, as it has a smaller delay. So I hope someone can help me with this mystery having provided this additional information.

I have this code to play a sine wave sound. I'm constantly hearing clicking sounds when I try to play this. I'm pretty sure it's because the playback of the buffers is not going perfectly, because when you change the value of l to a larger value (for instance 44100) the clicks are further apart. I think I have followed as accurate as possible the explanation of how to use the callbacks on the Microsoft website. I create three source voices that take turns playing: one is playing while the next is ready and the one after that is being made. I use a total time (tt) to put into the sin() function, so the first byte of the next buffer should align perfectly with the last byte of the current one.

Does anyone know what's going wrong?

P.S.: The many similar questions did not answer my question. In short: I'm not modifying the playing buffer (I don't think so at least); there should be no discontinuity at the border of one buffer to another; I'm not adjusting the frequency either during playback. So I don't think this is a duplicate.

#include <xaudio2.h>
#include <iostream>

#define PI 3.14159265358979323846f
#define l 4410 //0.1 seconds

IXAudio2MasteringVoice* pMasterVoice;
IXAudio2* pXAudio2;
IXAudio2SourceVoice* pSourceVoice[3];
XAUDIO2_BUFFER buffer;
WAVEFORMATEX wfx;
XAUDIO2_VOICE_STATE state;
BYTE pDataBuffer[2*l];
BYTE bytw[2];

int pow16[2];
float w[l];
int i, p;
float tt, ampl;

class VoiceCallback : public IXAudio2VoiceCallback {
public:
    HANDLE hBufferEndEvent;
    VoiceCallback() : hBufferEndEvent(CreateEvent(NULL, FALSE, FALSE, NULL)) {}
    ~VoiceCallback() { CloseHandle(hBufferEndEvent); }

    //Called when the voice has just finished playing a contiguous audio stream.
    void STDMETHODCALLTYPE OnStreamEnd() {SetEvent(hBufferEndEvent);}

    //Unused methods are stubs
    void STDMETHODCALLTYPE OnVoiceProcessingPassEnd() {}
    void STDMETHODCALLTYPE OnVoiceProcessingPassStart(UINT32 SamplesRequired) {}
    void STDMETHODCALLTYPE OnBufferEnd(void * pBufferContext) {}
    void STDMETHODCALLTYPE OnBufferStart(void * pBufferContext) {}
    void STDMETHODCALLTYPE OnLoopEnd(void * pBufferContext) {}
    void STDMETHODCALLTYPE OnVoiceError(void * pBufferContext, HRESULT Error) {}
};

VoiceCallback voiceCallback[3];

int main() {
    CoInitializeEx(nullptr, COINIT_MULTITHREADED);

    pXAudio2 = nullptr;
    XAudio2Create(&pXAudio2, 0, XAUDIO2_DEFAULT_PROCESSOR);

    pMasterVoice = nullptr;
    pXAudio2->CreateMasteringVoice(&pMasterVoice);

    tt = 0, p = 660, ampl = 2000;

    pow16[0] = 16;
    pow16[1] = 4096;

    wfx = {0};
    wfx.wFormatTag = WAVE_FORMAT_PCM;
    wfx.nChannels = (WORD)1; //mono
    wfx.nSamplesPerSec = (DWORD)44100; //samplerate
    wfx.wBitsPerSample = (WORD)16; //16 bit (signed)
    wfx.nBlockAlign = (WORD)2; //2 bytes per sample
    wfx.nAvgBytesPerSec = (DWORD)88200; //samplerate*blockalign
    wfx.cbSize = (WORD)0;

    i = 0;
    while (true) {
        for (int t = 0; t < l; t++) {
            tt = (float)(t + i*l); //total time

            w[t] = sin(2.f*PI*tt/p)*ampl;

            int intw = (int)w[t];
            if (intw < 0) {
                intw += 65535;
            }

            bytw[0] = 0; bytw[1] = 0;
            for (int k = 1; k >= 0; k--) {
                //turn integer into a little endian byte array
                bytw[k] += (BYTE)(16*(intw/pow16[k]));
                intw -= bytw[k]*(pow16[k]/16);
                bytw[k] += (BYTE)(intw/(pow16[k]/16));
                intw -= (intw/(pow16[k]/16))*pow16[k]/16;
            }

            pDataBuffer[2*t] = bytw[0];
            pDataBuffer[2*t + 1] = bytw[1];
        }

        buffer.AudioBytes = 2*l; //number of bytes per buffer
        buffer.pAudioData = pDataBuffer;
        buffer.Flags = XAUDIO2_END_OF_STREAM;

        if (i > 2) {
            pSourceVoice[i%3]->DestroyVoice();
        }
        pSourceVoice[i%3] = nullptr;

        pXAudio2->CreateSourceVoice(&pSourceVoice[i%3], &wfx, 0, XAUDIO2_DEFAULT_FREQ_RATIO, &voiceCallback[i%3], NULL, NULL);
        pSourceVoice[i%3]->SubmitSourceBuffer(&buffer);

        if (i > 1) {
            //wait until the current one is done playing
            while (pSourceVoice[(i - 2)%3]->GetState(&state), state.BuffersQueued > 0) {
                WaitForSingleObjectEx(voiceCallback[(i - 2)%3].hBufferEndEvent, INFINITE, TRUE);
            }
        }
        if (i > 0) {
            //play the next one while you're writing the one after that in the next iteration
            pSourceVoice[(i - 1)%3]->Start(0);
        }
        i++;
    }
}

Answer 1

If you want the sound to be 'looping', then submit multiple data-packets to the same source voice -or- set a loop value so it automatically restarts the existing audio packet. If you allow a source voice to run out of data, then you are going to hear the break in between depending upon the latency of your audio output system.

Furthermore, creating and destroy source voices is a relatively expensive operation, so doing it in a loop like this is not particular efficient.

See DirectX Tool Kit for Audio for a complete example of XAudio2 usage, as well as the latest version of the XAudio2 samples from the legacy DirectX SDK on GitHub.