所以我在这里遇到一些麻烦,我的AudioUnit从iOS中的麦克风/线路输入数据.我能够将所有内容设置为我认为可以,并且它正在调用我的recordingCallback,但是我从缓冲区中获取的数据不正确.它总是返回完全相同的东西,主要是零和随机大数.有谁知道可能导致这种情况的原因.我的代码如下.
设置音频单元
OSStatus status;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBusNumber,
&flag,
sizeof(flag));
// Disable playback IO
flag = 0;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBusNumber,
&flag,
sizeof(flag));
// Describe format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked |kAudioFormatFlagIsNonInterleaved;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 32;
audioFormat.mBytesPerPacket = 4;
audioFormat.mBytesPerFrame = 4;
// Apply format
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBusNumber,
&audioFormat,
sizeof(audioFormat));
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = (__bridge void*)self;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBusNumber,
&callbackStruct,
sizeof(callbackStruct));
status = AudioUnitInitialize(audioUnit);
输入回调
static OSStatus recordingCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mDataByteSize = 4;
bufferList.mBuffers[0].mNumberChannels = 1;
bufferList.mBuffers[0].mData = malloc(sizeof(float)*inNumberFrames); //
InputAudio *input = (__bridge InputAudio*)inRefCon;
OSStatus status;
status = AudioUnitRender([input audioUnit],
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
&bufferList);
float* result = (float*)&bufferList.mBuffers[0].mData;
if (input->counter == 5) {
for (int i = 0;i<inNumberFrames;i++) {
printf("%f ",result[i]);
}
}
input->counter++;
return noErr;
}
任何人都遇到类似问题或在我的代码中看到明显错误的东西.在此先感谢您的帮助!
我将所有这些都从Michael Tysons Core Audio RemoteIO code开始
最佳答案 如果我没记错的话,你从回调中的音频缓冲区得到的样本不是浮点数,它们是SInt16.尝试像这样投射样本:
SInt16 *sn16AudioData= (SInt16 *)(bufferList.mBuffers[0].mData);
这些应该是最大值和最小值:
#define sn16_MAX_SAMPLE_VALUE 32767
#define sn16_MIN_SAMPLE_VALUE -32768