在Cocoa中加载OpenAL的.wav文件

我需要将声音文件加载到基于Cocoa的OpenAL应用程序.

进展:

OpenAL实用程序函数alutLoadWAVFile已被弃用; Mac OS X SDK中不再包含alut标头.根据TechNotes,实际代码仍然存在二进制兼容性.但是,如果我尝试为函数添加声明,代码将编译但链接器将中止,抱怨无法找到alutLoadWAVFile的符号.(我正在链接到OpenAL.framework).

但是,Apple OpenAL示例代码仍然使用此符号.当我清理示例代码项目时,它编译和链接就好了.然而,没有声明可以找到该功能.(旁边的问题:如何构建和链接呢？)

所以,我在Apple发现了George Warner的一些代码,其中包含alutCreateBufferFromFile和alutLoadMemoryFromFile的替换函数.虽然能够直接从大多数任何类型的音频文件创建OpenAL缓冲区,但代码似乎仅支持8位单声道声音文件.16位立体声或单声道44khz文件导致令人讨厌的嘶嘶声和剪辑.(文件还可以; Quicktime播放它们就好了.)

因此,我的问题:有人可以指点一些.wav加载Cocoa/Carbon的代码/帮助,适合与OpenAL Buffer一起使用吗？谢谢.

我相信你已经解决了这个问题,但是对于那些通过Google找到这个问题的人来说,这里有一些经过测试的WAV加载代码.它可以工作,但你最好仔细检查内存泄漏和什么之前使用真正的东西.

static bool LoadWAVFile(const char* filename, ALenum* format, ALvoid** data, ALsizei* size, ALsizei* freq, Float64* estimatedDurationOut)
{
    CFStringRef filenameStr = CFStringCreateWithCString( NULL, filename, kCFStringEncodingUTF8 );
    CFURLRef url = CFURLCreateWithFileSystemPath( NULL, filenameStr, kCFURLPOSIXPathStyle, false );
    CFRelease( filenameStr );

    AudioFileID audioFile;
    OSStatus error = AudioFileOpenURL( url, kAudioFileReadPermission, kAudioFileWAVEType, &audioFile );
    CFRelease( url );

    if ( error != noErr )
    {
        fprintf( stderr, "Error opening audio file. %d\n", error );
        return false;
    }

    AudioStreamBasicDescription basicDescription;
    UInt32 propertySize = sizeof(basicDescription);
    error = AudioFileGetProperty( audioFile, kAudioFilePropertyDataFormat, &propertySize, &basicDescription );

    if ( error != noErr )
    {
        fprintf( stderr, "Error reading audio file basic description. %d\n", error );
        AudioFileClose( audioFile );
        return false;
    }

    if ( basicDescription.mFormatID != kAudioFormatLinearPCM )
    {
        // Need PCM for Open AL. WAVs are (I believe) by definition PCM, so this check isn't necessary. It's just here
        // in case I ever use this with another audio format.
        fprintf( stderr, "Audio file is not linear-PCM. %d\n", basicDescription.mFormatID );
        AudioFileClose( audioFile );
        return false;
    }

    UInt64 audioDataByteCount = 0;
    propertySize = sizeof(audioDataByteCount);
    error = AudioFileGetProperty( audioFile, kAudioFilePropertyAudioDataByteCount, &propertySize, &audioDataByteCount );
    if ( error != noErr )
    {
        fprintf( stderr, "Error reading audio file byte count. %d\n", error );
        AudioFileClose( audioFile );
        return false;
    }

    Float64 estimatedDuration = 0;
    propertySize = sizeof(estimatedDuration);
    error = AudioFileGetProperty( audioFile, kAudioFilePropertyEstimatedDuration, &propertySize, &estimatedDuration );
    if ( error != noErr )
    {
        fprintf( stderr, "Error reading estimated duration of audio file. %d\n", error );
        AudioFileClose( audioFile );
        return false;
    }

    ALenum alFormat = 0;

    if ( basicDescription.mChannelsPerFrame == 1 )
    {
        if ( basicDescription.mBitsPerChannel == 8 )
            alFormat = AL_FORMAT_MONO8;
        else if ( basicDescription.mBitsPerChannel == 16 )
            alFormat = AL_FORMAT_MONO16;
        else
        {
            fprintf( stderr, "Expected 8 or 16 bits for the mono channel but got %d\n", basicDescription.mBitsPerChannel );
            AudioFileClose( audioFile );
            return false;
        }

    }
    else if ( basicDescription.mChannelsPerFrame == 2 )
    {
        if ( basicDescription.mBitsPerChannel == 8 )
            alFormat = AL_FORMAT_STEREO8;
        else if ( basicDescription.mBitsPerChannel == 16 )
            alFormat = AL_FORMAT_STEREO16;
        else
        {
            fprintf( stderr, "Expected 8 or 16 bits per channel but got %d\n", basicDescription.mBitsPerChannel );
            AudioFileClose( audioFile );
            return false;
        }
    }
    else
    {
        fprintf( stderr, "Expected 1 or 2 channels in audio file but got %d\n", basicDescription.mChannelsPerFrame );
        AudioFileClose( audioFile );
        return false;
    }

    UInt32 numBytesToRead = audioDataByteCount;
    void* buffer = malloc( numBytesToRead );

    if ( buffer == NULL )
    {
        fprintf( stderr, "Error allocating buffer for audio data of size %u\n", numBytesToRead );
        return false;
    }

    error = AudioFileReadBytes( audioFile, false, 0, &numBytesToRead, buffer );
    AudioFileClose( audioFile );

    if ( error != noErr )
    {
        fprintf( stderr, "Error reading audio bytes. %d\n", error );
        free(buffer);
        return false;
    }

    if ( numBytesToRead != audioDataByteCount )
    {
        fprintf( stderr, "Tried to read %lld bytes from the audio file but only got %d bytes\n", audioDataByteCount, numBytesToRead );
        free(buffer);
        return false;
    }

    *freq = basicDescription.mSampleRate;
    *size = audioDataByteCount;
    *format = alFormat;
    *data = buffer;
    *estimatedDurationOut = estimatedDuration;

    return true;
}