我正在使用CoreAudio示例来连接各种不同的音频接口。我采用了CAPlayThrough示例（https://developer.apple.com/library/content/samplecode/CAPlayThrough/Introduction/Intro.html）并且一直在修改一些小东西。我想要完成的主要工作之一是在播放期间同时访问我的4通道音频接口的所有输入通道的所有输入流。在我的特定设置中，我有一个behringer uphoria 404接口，它有4个物理输入通道，我想同时访问。即如果我将麦克风/乐器插入外部接口的通道1,2,3和4中，我希望能够听到所有4个通道的输出，就像在DAW中那样，例如专业工具，逻辑等。
所以我从这个文档（https://developer.apple.com/library/content/technotes/tn2091/_index.html）中取了示例，然后继续创建我自己的频道图。
所以我做的就是写这个并为我的输入AudioUnit
设置频道图//Create channel Map
SInt32 *channelMap = NULL;
UInt32 numOfChannels = this->streamFormat.mChannelsPerFrame;
UInt32 mapSize = numOfChannels *sizeof(SInt32);
channelMap = (SInt32 *)malloc(mapSize);
for(UInt32 i=0;i<numOfChannels;i++)
{
channelMap[i]=-1;
}
channelMap[0] = 0;
channelMap[1] = 1;
channelMap[2] = 0;
channelMap[3] = 1;
AudioUnitSetProperty(mInputUnit,
kAudioOutputUnitProperty_ChannelMap,
kAudioUnitScope_Output,
1,
channelMap,
mapSize);
free(channelMap);
结果是我只能从我的界面访问来自通道1和2的数据。但是，当我将频道地图更改为：
channelMap[0] = 2;
channelMap[1] = 3;
我可以从音频接口访问第3和第4频道的数据，但不能从第1和第2频道访问数据。
所以我想我必须为我的输出AudioUnit设置频道映射。所以我将频道映射设置为我的输出AudioUnit：
SInt32 *channelMap = NULL;
UInt32 numOfChannels = this->streamFormat.mChannelsPerFrame;
UInt32 mapSize = numOfChannels *sizeof(SInt32);
channelMap = (SInt32 *)malloc(mapSize);
for(UInt32 i=0;i<numOfChannels;i++)
{
channelMap[i]=-1;
}
channelMap[0] = 0;
channelMap[1] = 1;
channelMap[2] = 0;
channelMap[3] = 1;
AudioUnitSetProperty(this->outputUnit,
kAudioOutputUnitProperty_ChannelMap,
kAudioUnitScope_Output,
1,
channelMap,
mapSize);
free(channelMap);
结果仍然相同。
设置频道地图的正确方法是什么？我已经通过Apple的文档网站进行了梳理，并试图查看他们的文档，但找不到比上面发布的文章更有帮助的内容。
更新：
除了修改频道地图之外，这些是我迄今为止尝试过的一些事情：
我想也许我的4通道接口上的每个物理输入都需要一个AudioUnit来表示它，并选择从通道映射获取输入流的通道。但是当为输入设置多个AudioUnit时，最后设置的AudioUnit最终充当默认输入。此外，AUGraph只允许我添加1个输出节点。
使用AudioUnit类型kAudioUnitSubType_StereoMixer将多个流与多个输出渲染回调组合在一起。我能够设置渲染回调并将其正确链接到立体声混音器AudioUnit，但不确定如何将输入流连接到输出。
更新2：
我之前应该添加这个。这来自上面列出的CAPlayThrough示例，用于设置kAudioUnitProperty_StreamFormat：
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev1_in);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &asbd_dev1_in, &propertySize);
checkErr(err);
//printf("=====Input DEVICE stream format\n" );
//asbd_dev1_in.Print();
//Get the Stream Format (client side)
propertySize = sizeof(asbd);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, &propertySize);
checkErr(err);
//printf("=====current Input (Client) stream format\n");
//asbd.Print();
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev2_out);
err = AudioUnitGetProperty(mOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd_dev2_out, &propertySize);
checkErr(err);
//printf("=====Output (Device) stream format\n");
//asbd_dev2_out.Print();
//////////////////////////////////////
//Set the format of all the AUs to the input/output devices channel count
//For a simple case, you want to set this to the lower of count of the channels
//in the input device vs output device
//////////////////////////////////////
asbd.mChannelsPerFrame =((asbd_dev1_in.mChannelsPerFrame < asbd_dev2_out.mChannelsPerFrame) ?asbd_dev1_in.mChannelsPerFrame :asbd_dev2_out.mChannelsPerFrame) ;
// We must get the sample rate of the input device and set it to the stream format of AUHAL
propertySize = sizeof(Float64);
AudioObjectPropertyAddress theAddress = { kAudioDevicePropertyNominalSampleRate,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
err = AudioObjectGetPropertyData(mInputDevice.mID, &theAddress, 0, NULL, &propertySize, &rate);
checkErr(err);
asbd.mSampleRate =rate;
propertySize = sizeof(asbd);
//Set the new formats to the AUs...
err = AudioUnitSetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mVarispeedUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
这是示例如何设置InputCall返回和输出回调
InputCallBack设置：
OSStatus err = noErr;
AURenderCallbackStruct input;
input.inputProc = InputProc;
input.inputProcRefCon = this;
//Setup the input callback.
err = AudioUnitSetProperty(mInputUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
0,
&input,
sizeof(input));
输入回叫：
OSStatus CAPlayThrough::InputProc(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
OSStatus err = noErr;
CAPlayThrough *This = (CAPlayThrough *)inRefCon;
if (This->mFirstInputTime < 0.)
This->mFirstInputTime = inTimeStamp->mSampleTime;
//Get the new audio data
err = AudioUnitRender(This->mInputUnit,
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames, //# of frames requested
This->mInputBuffer);// Audio Buffer List to hold data
checkErr(err);
if(!err) {
err = This->mBuffer->Store(This->mInputBuffer, Float64(inNumberFrames), SInt64(inTimeStamp->mSampleTime));
}
return err;
}
这是设置输出回调的方式：
output.inputProc = OutputProc;
output.inputProcRefCon = this;
err = AudioUnitSetProperty(mVarispeedUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
0,
&output,
sizeof(output));
checkErr(err);
这是输出过程
OSStatus CAPlayThrough::OutputProc(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *TimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
OSStatus err = noErr;
CAPlayThrough *This = (CAPlayThrough *)inRefCon;
Float64 rate = 0.0;
AudioTimeStamp inTS, outTS;
if (This->mFirstInputTime < 0.) {
// input hasn't run yet -> silence
MakeBufferSilent (ioData);
return noErr;
}
//use the varispeed playback rate to offset small discrepancies in sample rate
//first find the rate scalars of the input and output devices
err = AudioDeviceGetCurrentTime(This->mInputDevice.mID, &inTS);
// this callback may still be called a few times after the device has been stopped
if (err)
{
MakeBufferSilent (ioData);
return noErr;
}
err = AudioDeviceGetCurrentTime(This->mOutputDevice.mID, &outTS);
checkErr(err);
rate = inTS.mRateScalar / outTS.mRateScalar;
err = AudioUnitSetParameter(This->mVarispeedUnit,kVarispeedParam_PlaybackRate,kAudioUnitScope_Global,0, rate,0);
checkErr(err);
//get Delta between the devices and add it to the offset
if (This->mFirstOutputTime < 0.) {
This->mFirstOutputTime = TimeStamp->mSampleTime;
Float64 delta = (This->mFirstInputTime - This->mFirstOutputTime);
This->ComputeThruOffset();
//changed: 3865519 11/10/04
if (delta < 0.0)
This->mInToOutSampleOffset -= delta;
else
This->mInToOutSampleOffset = -delta + This->mInToOutSampleOffset;
MakeBufferSilent (ioData);
return noErr;
}
//copy the data from the buffers
err = This->mBuffer->Fetch(ioData, inNumberFrames, SInt64(TimeStamp->mSampleTime - This->mInToOutSampleOffset));
if(err != kCARingBufferError_OK)
{
MakeBufferSilent (ioData);
SInt64 bufferStartTime, bufferEndTime;
This->mBuffer->GetTimeBounds(bufferStartTime, bufferEndTime);
This->mInToOutSampleOffset = TimeStamp->mSampleTime - bufferStartTime;
}
return noErr;
}