iOS Core Audio：在kAudioFormatFlagsCanonical和kAudioFormatFlagsAudioUnitCanonical之间转换 [英] iOS Core Audio : Converting between kAudioFormatFlagsCanonical and kAudioFormatFlagsAudioUnitCanonical
问题描述
我需要在以下格式之间进行转换：
I need to convert between this format :
format.mSampleRate = 44100.0;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kAudioFormatFlagsCanonical | kLinearPCMFormatFlagIsNonInterleaved;
format.mBytesPerPacket = sizeof(AudioUnitSampleType);
format.mFramesPerPacket = 1;
format.mBytesPerFrame = sizeof(AudioUnitSampleType);
format.mChannelsPerFrame = 2 ;
format.mBitsPerChannel = sizeof(AudioUnitSampleType)*8;
此格式
format.mSampleRate = 44100.0;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kAudioFormatFlagsAudioUnitCanonical;
format.mBytesPerPacket = sizeof(AudioUnitSampleType);
format.mFramesPerPacket = 1;
format.mBytesPerFrame = sizeof(AudioUnitSampleType);
format.mChannelsPerFrame = 2;
format.mBitsPerChannel = sizeof(AudioUnitSampleType)*8;
在音频渲染回调的范围内，其中有以下代码和缓冲区[]在第二种格式和array []需要第一种格式。
within the confines of a audio render callback where there is the following code and buffer[] is in the 2nd format and array[] requires the 1st format.
for (k = 0; k < channels; k++){
buffer = (AudioUnitSampleType *) ioData->mBuffers[k].mData;
for(j=0; j < samples; j++){
array[j] = buffer[j];
}
}
我知道你可以使用Apple转换器单元，但是在我的情况下我无法使用Apple Converter音频设备（这是有原因的）。
I know you can use the Apple converter unit, but I cannot use the Apple Converter audio unit in my situation (there's a reason).
基本上两种格式之间的唯一区别是format.mFormatFlags
（kAudioUnitSampleFractionBits<< kLinearPCMFormatFlagsSampleFractionShift）的以下标志。
Basically the only difference between the 2 formats the following flag for format.mFormatFlags (kAudioUnitSampleFractionBits << kLinearPCMFormatFlagsSampleFractionShift).
如何将buffer []（包含第二种格式的数据）转换为array []（包含第一种格式的数据），反之亦然？
How can I convert buffer[] (containing data in the 2nd format) to array[] (containing data in the 1st format) and vice-versa?
谢谢。
推荐答案
好吧，如果您参考 kAudioFormatFlagsAudioUnitCanonical 上的文档，那么你请参阅：
Well, if you refer to the docs on kAudioFormatFlagsAudioUnitCanonical, you see:
kAudioFormatFlagsAudioUnitCanonical The flags for the canonical audio unit sample
type. This matches AudioUnitSampleType.
和
The canonical audio sample type for audio units and other audio processing in
iPhone OS is noninterleaved linear PCM with 8.24-bit fixed-point samples.
因此， buffer [] 中的样本数组采用8.24位定点格式。这是什么意思？
So, the samples in buffer[] array are in 8.24-bit fixed-point format. What does it mean?
8.24位定点格式用于表示具有固定精度的浮点数 - 一个32位整数，其中前8位代表整个部分，最后一位代表整个部分24位表示小数部分（小数点后的数字）。 （进一步阅读）
8.24-bit fixed-point format is used to represent float numbers with fixed precision - a 32-bit integer where the first 8 bits represent the whole part, and the last 24 bits represent the fractional part (the numbers after the decimal). (Further reading)
在iOS音频单元中，存在细微差别 - 此浮点数（通常）的范围为[-1,1）（ [-1.000000000000，+ 0.999969482421875]确切地说是）。转换为16位PCM时，只会剪切此范围之外的值。在大多数情况下，您可以验证前8位是0x00还是0xff（在2的补码中为-1）。
In iOS Audio Units, there's a minor difference - this float number (usually) ranges in [-1, 1) ([-1.000000000000, +0.999969482421875] to be exact). Values outside of this range are simply clipped when converting to 16-bit PCM. You can validate that the first 8 bits would be 0x00 or 0xff (-1 in two's compliment) for most part.
要将此表示转换为16位数，使用此：
To convert this representation to a 16-bit number, use this:
SIGN((SInt8)(val >> 24)) * 32768 * (val & 0xFFFFFF)/(float)(1<<24)
即：从8 MSB中提取符号，从24 LSB中提取小数值，除以24位整数（2 ^ 24）的范围，得到0到1之间的浮点数，最后乘以32768得到所需范围的值。
That is: extract the sign from the 8 MSB, extract fractional value from 24 LSB and divide by range of 24-bit integer (2^24) resulting in a float between 0 and 1, and finally multiply this by 32768 to get a value in desired range.
我自己也没试过 - 你可能需要在这里和那里调整一些东西。
I haven't tried this myself though - you might have to adjust a few things here and there.
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