为OpenAL加载.WAV文件

Nat*_* G. 5 c++ audio openal wav

我正在尝试加载.WAV文件以与OpenAL一起播放。我正在跟踪我在互联网上找到的示例,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;
}
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我有几个大约50kb的WAV文件。他们加载并播放正常。但是,当我尝试加载整首歌曲时(是的,我已验证文件已使用VLC Media Player和MusicBee正确格式化),它返回“无效的数据标头:尝试加载test.wav”,这是由于此块导致的:

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");
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我怀疑这是与大小有关的东西,它摆脱了标题,因为似乎只有1000kb以下的东西才能工作(还没有完全测试过,很难找到大小合适的声音文件在我的计算机和Internet上浮动)。不过,这只是一个猜测,我真的不知道发生了什么。感谢帮助!

Raz*_*hel 1

这个问题有点老了,但还没有答案。我碰巧写了一个 WAV 文件加载器,并且我偶然发现了与您完全相同的问题。

事实上,“数据”部分并不能保证存在于您期望的位置。还可以指定其他块,例如我的例子中的“提示”块。这是一种真正隐藏的信息,我花了很多时间试图找到它: https: //sites.google.com/site/musicgapi/technical-documents/wav-file-format#cue

就我而言,然后我只需检查是否存在“提示”部分并忽略其数据。我(还)没有检查其他块类型,因为我没有任何测试材料。

在 C++ 代码中,这样做是这样的:

// std::ifstream file("...", std::ios_base::in | std::ios_base::binary);
// std::array<uint8_t, 4> bytes {};

file.read(reinterpret_cast<char*>(bytes.data()), 4); // Supposed to be "data"

// A "cue " field can be specified; if so, the given amount of bytes will be ignored
if (bytes[0] == 'c' && bytes[1] == 'u' && bytes[2] == 'e' && bytes[3] == ' ') {
  file.read(reinterpret_cast<char*>(bytes.data()), 4); // Cue data size

  const uint32_t cueDataSize = fromLittleEndian(bytes);
  file.ignore(cueDataSize);

  file.read(reinterpret_cast<char*>(bytes.data()), 4); // "data"
}

// A LIST segment may be specified; see the edit below

// "data" is now properly expected
if (bytes[0] != 'd' && bytes[1] != 'a' && bytes[2] != 't' && bytes[3] != 'a')
  return false;
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编辑:我现在也有一个“LIST”标签,这是可以预料的。这可以像提示一样被忽略:

if (bytes[0] == 'L' && bytes[1] == 'I' && bytes[2] == 'S' && bytes[3] == 'T') {
  file.read(reinterpret_cast<char*>(bytes.data()), 4); // List data size

  const uint32_t listDataSize = fromLittleEndian(bytes);
  file.ignore(listDataSize);

  file.read(reinterpret_cast<char*>(bytes.data()), 4);
}
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