c++ - 为 OpenAL 加载 .WAV 文件

Question

我正在尝试加载要用 OpenAL 播放的 .WAV 文件。我正在按照我在互联网上找到的示例进行操作，但它的行为很奇怪。这是代码:

struct RIFF_Header {
    char chunkID[4];
    long chunkSize;//size not including chunkSize or chunkID
    char format[4];
};

/*
 * Struct to hold fmt subchunk data for WAVE files.
 */
struct WAVE_Format {
    char subChunkID[4];
    long subChunkSize;
    short audioFormat;
    short numChannels;
    long sampleRate;
    long byteRate;
    short blockAlign;
    short bitsPerSample;
};

/*
* Struct to hold the data of the wave file
*/
struct WAVE_Data {
    char subChunkID[4]; //should contain the word data
    long subChunk2Size; //Stores the size of the data block
};

bool loadWavFile(std::string filename, ALuint* buffer,
                 ALsizei* size, ALsizei* frequency,
                 ALenum* format) {
    //Local Declarations
    FILE* soundFile = NULL;
    WAVE_Format wave_format;
    RIFF_Header riff_header;
    WAVE_Data wave_data;
    unsigned char* data;

    *size = wave_data.subChunk2Size;
    *frequency = wave_format.sampleRate;
    if (wave_format.numChannels == 1) {
        if (wave_format.bitsPerSample == 8 )
            *format = AL_FORMAT_MONO8;
        else if (wave_format.bitsPerSample == 16)
            *format = AL_FORMAT_MONO16;
    } else if (wave_format.numChannels == 2) {
        if (wave_format.bitsPerSample == 8 )
            *format = AL_FORMAT_STEREO8;
        else if (wave_format.bitsPerSample == 16)
            *format = AL_FORMAT_STEREO16;
    }

    try {
        soundFile = fopen(filename.c_str(), "rb");
        if (!soundFile)
            throw (filename);

        // Read in the first chunk into the struct
        fread(&riff_header, sizeof(RIFF_Header), 1, soundFile);

        //check for RIFF and WAVE tag in memeory
        if ((riff_header.chunkID[0] != 'R' ||
             riff_header.chunkID[1] != 'I' ||
             riff_header.chunkID[2] != 'F' ||
             riff_header.chunkID[3] != 'F') ||
            (riff_header.format[0] != 'W' ||
             riff_header.format[1] != 'A' ||
             riff_header.format[2] != 'V' ||
             riff_header.format[3] != 'E'))
            throw ("Invalid RIFF or WAVE Header");

        //Read in the 2nd chunk for the wave info
        fread(&wave_format, sizeof(WAVE_Format), 1, soundFile);
        //check for fmt tag in memory
        if (wave_format.subChunkID[0] != 'f' ||
            wave_format.subChunkID[1] != 'm' ||
            wave_format.subChunkID[2] != 't' ||
            wave_format.subChunkID[3] != ' ')
            throw ("Invalid Wave Format");

        //check for extra parameters;
        if (wave_format.subChunkSize > 16)
            fseek(soundFile, sizeof(short), SEEK_CUR);

        //Read in the the last byte of data before the sound file
        fread(&wave_data, sizeof(WAVE_Data), 1, soundFile);

        //check for data tag in memory
        if (wave_data.subChunkID[0] != 'd' ||
            wave_data.subChunkID[1] != 'a' ||
            wave_data.subChunkID[2] != 't' ||
            wave_data.subChunkID[3] != 'a')
            throw ("Invalid data header");

        //Allocate memory for data
        data = new unsigned char[wave_data.subChunk2Size];

        // Read in the sound data into the soundData variable
        if (!fread(data, wave_data.subChunk2Size, 1, soundFile))
            throw ("error loading WAVE data into struct!");

        //Now we set the variables that we passed in with the
        //data from the structs
        *size = wave_data.subChunk2Size;
        *frequency = wave_format.sampleRate;
        //The format is worked out by looking at the number of
        //channels and the bits per sample.
        if (wave_format.numChannels == 1) {
            if (wave_format.bitsPerSample == 8 )
                *format = AL_FORMAT_MONO8;
            else if (wave_format.bitsPerSample == 16)
                *format = AL_FORMAT_MONO16;
        } else if (wave_format.numChannels == 2) {
            if (wave_format.bitsPerSample == 8 )
                *format = AL_FORMAT_STEREO8;
            else if (wave_format.bitsPerSample == 16)
                *format = AL_FORMAT_STEREO16;
        }
        //create our openAL buffer and check for success
        alGenBuffers(2, buffer);
        if(alGetError() != AL_NO_ERROR) {
            std::cerr << alGetError() << std::endl;
        }
        //now we put our data into the openAL buffer and
        //check for success
        alBufferData(*buffer, *format, (void*)data,
                     *size, *frequency);
        if(alGetError() != AL_NO_ERROR) {
            std::cerr << alGetError() << std::endl;
        }
        //clean up and return true if successful
        fclose(soundFile);
        delete data;
        return true;
    } catch(const char* error) {
        //our catch statement for if we throw a string
        std::cerr << error << " : trying to load "
        << filename << std::endl;
        //clean up memory if wave loading fails
        if (soundFile != NULL)
            fclose(soundFile);
        //return false to indicate the failure to load wave
        delete data;
        return false;
    }
}

int main() {
    ALuint buffer, source;
    ALint state;
    ALsizei size;
    ALsizei frequency;
    ALenum format;

    ALCcontext *context;
    ALCdevice *device;

    device = alcOpenDevice(nullptr);
    if (device == NULL)
    {
        cerr << "Error finding default Audio Output Device" << endl;
    }

    context = alcCreateContext(device,NULL);

    alcMakeContextCurrent(context);

    alGetError();

    loadWavFile("test.wav", &buffer, &size, &frequency, &format);

    alGenSources(1, &source);
    alSourcei(source, AL_BUFFER, buffer);

    // Play
    alSourcePlay(source);

    // Wait for the song to complete
    do {
        alGetSourcei(source, AL_SOURCE_STATE, &state);
    } while (state == AL_PLAYING);

    // Clean up sources and buffers
    alDeleteSources(1, &source);
    alDeleteBuffers(1, &buffer);
    return 0;
}

我有几个大约 50kb 的 WAV 文件。他们加载和播放都很好。但是，当我尝试加载整首歌曲时(是的，我使用 VLC Media Player 和 MusicBee 验证了文件格式正确)它返回“无效数据 header :尝试加载 test.wav”，这是由此处的这个 block 引起的:

if (wave_data.subChunkID[0] != 'd' ||
wave_data.subChunkID[1] != 'a' ||
wave_data.subChunkID[2] != 't' ||
wave_data.subChunkID[3] != 'a')
throw ("Invalid data header");

我怀疑它是一些与大小相关的东西导致了那个标题，因为它似乎只有不到 1000kb 的东西才能工作(还没有完全测试过，很难找到在我的电脑上漂浮的完美大小的声音文件并且在网上)。但这只是一个猜测，我真的不知道发生了什么。感谢您的帮助!

Answer 1

我知道这个问题已经存在很长时间了，但我找到了一个教程并对其进行了测试。有用。试试这个:

//http://www.youtube.com/user/thecplusplusguy
//Playing 3D sound with OpenAL, and loading a wav file manually
#include <iostream>
#include <fstream>
#include <cstring>
#include <al.h>
#include <alc.h>

bool isBigEndian()
{
    int a = 1;
    return !((char*)&a)[0];
}

int convertToInt(char* buffer, int len)
{
    int a = 0;
    if (!isBigEndian())
        for (int i = 0; i<len; i++)
            ((char*)&a)[i] = buffer[i];
    else
        for (int i = 0; i<len; i++)
            ((char*)&a)[3 - i] = buffer[i];
    return a;
}

char* loadWAV(const char* fn, int& chan, int& samplerate, int& bps, int& size)
{
    char buffer[4];
    std::ifstream in(fn, std::ios::binary);
    in.read(buffer, 4);
    if (strncmp(buffer, "RIFF", 4) != 0)
    {
        std::cout << "this is not a valid WAVE file" << std::endl;
        return NULL;
    }
    in.read(buffer, 4);
    in.read(buffer, 4);      //WAVE
    in.read(buffer, 4);      //fmt
    in.read(buffer, 4);      //16
    in.read(buffer, 2);      //1
    in.read(buffer, 2);
    chan = convertToInt(buffer, 2);
    in.read(buffer, 4);
    samplerate = convertToInt(buffer, 4);
    in.read(buffer, 4);
    in.read(buffer, 2);
    in.read(buffer, 2);
    bps = convertToInt(buffer, 2);
    in.read(buffer, 4);      //data
    in.read(buffer, 4);
    size = convertToInt(buffer, 4);
    char* data = new char[size];
    in.read(data, size);
    return data;
}

int main(int argc, char** argv)
{
    int channel, sampleRate, bps, size;
    char* data = loadWAV("C:/Users/Gizego/Desktop/Youtube/Músicas/TheFatRat+-+Time+Lapse.wav", channel, sampleRate, bps, size);
    ALCdevice* device = alcOpenDevice(NULL);
    if (device == NULL)
    {
        std::cout << "cannot open sound card" << std::endl;
        return 0;
    }
    ALCcontext* context = alcCreateContext(device, NULL);
    if (context == NULL)
    {
        std::cout << "cannot open context" << std::endl;
        return 0;
    }
    alcMakeContextCurrent(context);

    unsigned int bufferid, format;
    alGenBuffers(1, &bufferid);
    if (channel == 1)
    {
        if (bps == 8)
        {
            format = AL_FORMAT_MONO8;
        }
        else {
            format = AL_FORMAT_MONO16;
        }
    }
    else {
        if (bps == 8)
        {
            format = AL_FORMAT_STEREO8;
        }
        else {
            format = AL_FORMAT_STEREO16;
        }
    }
    alBufferData(bufferid, format, data, size, sampleRate);
    unsigned int sourceid;
    alGenSources(1, &sourceid);
    alSourcei(sourceid, AL_BUFFER, bufferid);
    alSourcePlay(sourceid);

    while (true)
    {

    }

    alDeleteSources(1, &sourceid);
    alDeleteBuffers(1, &bufferid);

    alcDestroyContext(context);
    alcCloseDevice(device);
    delete[] data;
    return 0;
}