iOS 将音频采样率从 16 kHz 转换为 8 kHz

Posted 2023-02-23

【中文标题】iOS 将音频采样率从 16 kHz 转换为 8 kHz

【英文标题】：iOS convert audio sample rate from 16 kHz to 8 kHz

【发布时间】：2017-03-23 09:41:44

【问题描述】：

我尝试将 PCM 音频从 16kHz 转换为 8kHz，只是采样率，没有格式更改，流程看起来很简单，但我一直通过调用 AudioConverterFillComplexBuffer 得到 kAudioConverterErr_InvalidInputSize ("insz")。我的输入音频样本大小是 320 字节，结果应该是 160 字节，但我的输出缓冲区中只有 144 字节。在过去的几个小时里一直在拉我的头发。是不是设置错了？

static AudioConverterRef PCM8kTo16kConverterRef;

- (instancetype)init 
    self = [super init];
    if (self) 
        [self initConverter];
    
    return self;


-(void)initConverter
    AudiostreamBasicDescription PCM8kDescription = 0;
    PCM8kDescription.mSampleRate = 8000.0;
    PCM8kDescription.mFormatID = kAudioFormatLinearPCM;
    PCM8kDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian;
    PCM8kDescription.mBitsPerChannel = 8 * sizeof(SInt16);
    PCM8kDescription.mChannelsPerFrame = 1;
    PCM8kDescription.mBytesPerFrame = sizeof(SInt16) * PCM8kDescription.mChannelsPerFrame;
    PCM8kDescription.mFramesPerPacket = 1;
    PCM8kDescription.mBytesPerPacket = PCM8kDescription.mBytesPerFrame * PCM8kDescription.mFramesPerPacket;

    AudioStreamBasicDescription PCM16kDescription = 0;
    PCM16kDescription.mSampleRate = 16000.0;
    PCM16kDescription.mFormatID = kAudioFormatLinearPCM;
    PCM16kDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian;
    PCM16kDescription.mBitsPerChannel = 8 * sizeof(SInt16);
    PCM16kDescription.mChannelsPerFrame = 1;
    PCM16kDescription.mBytesPerFrame = sizeof(SInt16) * PCM16kDescription.mChannelsPerFrame;
    PCM16kDescription.mFramesPerPacket = 1;
    PCM16kDescription.mBytesPerPacket = PCM16kDescription.mBytesPerFrame * PCM16kDescription.mFramesPerPacket;

    OSStatus status = AudioConverterNew(&PCM16kDescription, &PCM8kDescription, &converterRef);


OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)

    AudioBufferList audioBufferList = *(AudioBufferList *)inUserData;

    ioData->mBuffers[0].mData = audioBufferList.mBuffers[0].mData;
    ioData->mBuffers[0].mDataByteSize = audioBufferList.mBuffers[0].mDataByteSize;

    return  noErr;


- (NSData *)testSample:(NSData *)inAudio 

    NSMutableData *ddd = [inAudio mutableCopy];
    AudioBufferList inAudioBufferList = 0;
    inAudioBufferList.mNumberBuffers = 1;
    inAudioBufferList.mBuffers[0].mNumberChannels = 1;
    inAudioBufferList.mBuffers[0].mDataByteSize = (UInt32)[ddd length];
    inAudioBufferList.mBuffers[0].mData = [ddd mutableBytes];

    uint32_t bufferSize = (UInt32)[inAudio length] / 2;
    uint8_t *buffer = (uint8_t *)malloc(bufferSize);
    memset(buffer, 0, bufferSize);
    AudioBufferList outAudioBufferList;
    outAudioBufferList.mNumberBuffers = 1;
    outAudioBufferList.mBuffers[0].mNumberChannels = 1;
    outAudioBufferList.mBuffers[0].mDataByteSize = bufferSize;
    outAudioBufferList.mBuffers[0].mData = buffer;

    UInt32 ioOutputDataPacketSize = bufferSize;

    OSStatus ret = AudioConverterFillComplexBuffer(converterRef, inInputDataProc, &inAudioBufferList, &ioOutputDataPacketSize, &outAudioBufferList, NULL) ;

    NSData *data = [NSData dataWithBytes:outAudioBufferList.mBuffers[0].mData length:outAudioBufferList.mBuffers[0].mDataByteSize];
    free(buffer);
    return data;

【参考方案1】：

有两个问题：

你的AudioConverterComplexInputDataProc 没有设置ioNumberDataPackets：

*ioNumberDataPackets = audioBufferList.mBuffers[0].mDataByteSize/2;

ioOutputDataPacketSize 应该是 packets/frames 中的输出缓冲区容量，而不是字节，所以你不应该除以 2 吗？

【讨论】：

非常感谢！就是这样。没搞懂这些变量的正确含义，现在明白了XD再次感谢！

如何快速做到这一点？