Homebrew Noise enters at the end when using ndsp.

[Togetoge]

Noise flows instantaneously after I finish playing wav file using ndsp.
What should I do?

#include <3ds.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>

int main(int argc, char* argv[])
{
    gfxInitDefault();
    
    consoleInit(GFX_TOP, nullptr);
    
    // The dsp channel number
    constexpr int channel = 1;
    
    u32 sampleRate;
    u32 dataSize;
    u16 channels;
    u16 bitsPerSample;
    
    // Initialize ndsp
    ndspInit();
    
    ndspSetOutputMode(NDSP_OUTPUT_STEREO);
    ndspSetOutputCount(1); // Num of buffers
    
    // Reading wav file
    FILE* fp = fopen(".wav", "rb");
    
    if(!fp)
    {
        printf("Could not open the example.wav file.\n");
        return 1;
    }
    
    char signature[4];
    
    fread(signature, 1, 4, fp);
    
    if( signature[0] != 'R' &&
        signature[1] != 'I' &&
        signature[2] != 'F' &&
        signature[3] != 'F')
    {
        printf("Wrong file format.\n");
        fclose(fp);
        return 1;
    }
    
    fseek(fp,0,SEEK_END);
    dataSize = ftell(fp);
    fseek(fp, 22, SEEK_SET);
    fread(&channels, 2, 1, fp);
    fseek(fp, 24, SEEK_SET);
    fread(&sampleRate, 4, 1, fp);
    fseek(fp, 34, SEEK_SET);
    fread(&bitsPerSample, 2, 1, fp);
    
    if(dataSize == 0 || (channels != 1 && channels != 2) || (bitsPerSample != 8 && bitsPerSample != 16))
    {
        printf("Corrupted wav file.\n");
        fclose(fp);
        return 1;
    }
    
    // Allocating and reading samples
    u8* data = static_cast<u8*>(linearAlloc(dataSize));
    
    if(!data)
    {
        fclose(fp);
        return 1;
    }
    
    fseek(fp, 44, SEEK_SET);
    fread(data, 1, dataSize, fp);
    fclose(fp);
    
    // Find the right format
    u16 ndspFormat;
    
    if(bitsPerSample == 8)
    {
        ndspFormat = (channels == 1) ?
            NDSP_FORMAT_MONO_PCM8 :
            NDSP_FORMAT_STEREO_PCM8;
    }
    else
    {
        ndspFormat = (channels == 1) ?
            NDSP_FORMAT_MONO_PCM16 :
            NDSP_FORMAT_STEREO_PCM16;
    }
    
    ndspChnReset(channel);
    ndspChnSetInterp(channel, NDSP_INTERP_NONE);
    ndspChnSetRate(channel, float(sampleRate));
    ndspChnSetFormat(channel, ndspFormat);
    
    // Create and play a wav buffer
    ndspWaveBuf waveBuf;
    std::memset(&waveBuf, 0, sizeof(ndspWaveBuf));
    
    //waveBuf.data_vaddr = reinterpret_cast<u32>(data);
    waveBuf.data_vaddr = (const void*)data;
    waveBuf.nsamples = (dataSize / (bitsPerSample >> 3);
    waveBuf.looping = false; // Loop enabled
    waveBuf.status = NDSP_WBUF_FREE;
    
    DSP_FlushDataCache(data, dataSize);
    
    while(aptMainLoop())
    {
        hidScanInput();
        
        u32 keys = hidKeysDown();
        ndspChnWaveBufAdd(channel, &waveBuf);
        
        if(keys & KEY_START) break;
        
        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
    
    ndspChnWaveBufClear(channel);
    
    linearFree(data);
    
    gfxExit();
    ndspExit();
    
    return 0;
}

 

[elhobbs]

Noise flows instantaneously after I finish playing wav file using ndsp.
What should I do?

#include <3ds.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>

int main(int argc, char* argv[])
{
    gfxInitDefault();
   
    consoleInit(GFX_TOP, nullptr);
   
    // The dsp channel number
    constexpr int channel = 1;
   
    u32 sampleRate;
    u32 dataSize;
    u16 channels;
    u16 bitsPerSample;
   
    // Initialize ndsp
    ndspInit();
   
    ndspSetOutputMode(NDSP_OUTPUT_STEREO);
    ndspSetOutputCount(1); // Num of buffers
   
    // Reading wav file
    FILE* fp = fopen(".wav", "rb");
   
    if(!fp)
    {
        printf("Could not open the example.wav file.\n");
        return 1;
    }
   
    char signature[4];
   
    fread(signature, 1, 4, fp);
   
    if( signature[0] != 'R' &&
        signature[1] != 'I' &&
        signature[2] != 'F' &&
        signature[3] != 'F')
    {
        printf("Wrong file format.\n");
        fclose(fp);
        return 1;
    }
   
    fseek(fp,0,SEEK_END);
    dataSize = ftell(fp);
    fseek(fp, 22, SEEK_SET);
    fread(&channels, 2, 1, fp);
    fseek(fp, 24, SEEK_SET);
    fread(&sampleRate, 4, 1, fp);
    fseek(fp, 34, SEEK_SET);
    fread(&bitsPerSample, 2, 1, fp);
   
    if(dataSize == 0 || (channels != 1 && channels != 2) || (bitsPerSample != 8 && bitsPerSample != 16))
    {
        printf("Corrupted wav file.\n");
        fclose(fp);
        return 1;
    }
   
    // Allocating and reading samples
    u8* data = static_cast<u8*>(linearAlloc(dataSize));
   
    if(!data)
    {
        fclose(fp);
        return 1;
    }
   
    fseek(fp, 44, SEEK_SET);
    fread(data, 1, dataSize, fp);
    fclose(fp);
   
    // Find the right format
    u16 ndspFormat;
   
    if(bitsPerSample == 8)
    {
        ndspFormat = (channels == 1) ?
            NDSP_FORMAT_MONO_PCM8 :
            NDSP_FORMAT_STEREO_PCM8;
    }
    else
    {
        ndspFormat = (channels == 1) ?
            NDSP_FORMAT_MONO_PCM16 :
            NDSP_FORMAT_STEREO_PCM16;
    }
   
    ndspChnReset(channel);
    ndspChnSetInterp(channel, NDSP_INTERP_NONE);
    ndspChnSetRate(channel, float(sampleRate));
    ndspChnSetFormat(channel, ndspFormat);
   
    // Create and play a wav buffer
    ndspWaveBuf waveBuf;
    std::memset(&waveBuf, 0, sizeof(ndspWaveBuf));
   
    //waveBuf.data_vaddr = reinterpret_cast<u32>(data);
    waveBuf.data_vaddr = (const void*)data;
    waveBuf.nsamples = (dataSize / (bitsPerSample >> 3);
    waveBuf.looping = false; // Loop enabled
    waveBuf.status = NDSP_WBUF_FREE;
   
    DSP_FlushDataCache(data, dataSize);
   
    while(aptMainLoop())
    {
        hidScanInput();
       
        u32 keys = hidKeysDown();
        ndspChnWaveBufAdd(channel, &waveBuf);
       
        if(keys & KEY_START) break;
       
        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
   
    ndspChnWaveBufClear(channel);
   
    linearFree(data);
   
    gfxExit();
    ndspExit();
   
    return 0;
}

You are using the file length for dataSize and neglecting to remove the size of the headers. So you are playing past the end of the wav samples you read from the file.