我正在学习CoreAudio，我正在阅读Apple的文档中的一些示例，并弄清楚如何设置和不设置。到目前为止，我能够连接到默认连接的音频输入设备并将其输出到默认输出设备。我连接了一个2通道接口，并能够从中输出输入并输出它。
但是我正在搜索他们的API参考和示例，但是找不到任何可以从我的界面访问各个输入通道的实质内容。
我能够在我的渲染回调函数中劈开并从AudioBufferList中提取样本并以这种方式操作它，但我想知道是否有正确的方法或更正式的方式从每个输入通道访问数据
编辑：
这是我从我正在使用的示例中找到的用户数据：
typedef struct MyAUGraphPlayer
{
AudioStreamBasicDescription streamFormat;
AUGraph graph;
AudioUnit inputUnit;
AudioUnit outputUnit;
AudioBufferList * inputBuffer;
CARingBuffer * ringBuffer;
Float64 firstInputSampleTime;
Float64 firstOutputSampleTime;
Float64 inToOutSampleTimeOffset;
} MyAUGraphPlayer;
这就是我设置输入单元的方式：
void CreateInputUnit(MyAUGraphPlayer * player)
{
//Generates a description that matches audio HAL
AudioComponentDescription inputcd = {0};
inputcd.componentType = kAudioUnitType_Output;
inputcd.componentSubType = kAudioUnitSubType_HALOutput;
inputcd.componentManufacturer = kAudioUnitManufacturer_Apple;
UInt32 deviceCount = AudioComponentCount ( &inputcd );
printf("Found %d devices\n", deviceCount);
AudioComponent comp = AudioComponentFindNext(NULL, &inputcd);
if(comp == NULL) {
printf("Can't get output unit\n");
exit(1);
}
OSStatus status;
status = AudioComponentInstanceNew(comp, &player->inputUnit);
assert(status == noErr);
//Explicitly enable Input and disable output
UInt32 disableFlag = 0;
UInt32 enableFlag = 1;
AudioUnitScope outputBus = 0;
AudioUnitScope inputBus = 1;
status = AudioUnitSetProperty(player->inputUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
inputBus,
&enableFlag,
sizeof(enableFlag))
assert(status == noErr);
status = AudioUnitSetProperty(player->inputUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
outputBus,
&disableFlag,
sizeof(enableFlag));
assert(status == noErr);
printf("Finished enabling input and disabling output on an inputUnit\n");
//Get the default Audio input Device
AudioDeviceID defaultDevice = kAudioObjectUnknown;
UInt32 propertySize = sizeof(defaultDevice);
AudioObjectPropertyAddress defaultDeviceProperty;
defaultDeviceProperty.mSelector = kAudioHardwarePropertyDefaultInputDevice;
defaultDeviceProperty.mScope = kAudioObjectPropertyScopeGlobal;
defaultDeviceProperty.mElement = kAudioObjectPropertyElementMaster;
status = AudioObjectGetPropertyData(kAudioObjectSystemObject,
&defaultDeviceProperty,
0,
NULL,
&propertySize,
&defaultDevice);
assert(status == noErr);
//Set the current device property of the AUHAL
status = AudioUnitSetProperty(player->inputUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global,
outputBus,
&defaultDevice,
sizeof(defaultDevice));
assert(status == noErr);
//Get the AudioStreamBasicDescription from Input AUHAL
propertySize = sizeof(AudioStreamBasicDescription);
status = AudioUnitGetProperty(player->inputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
inputBus,
&player->streamFormat,
&propertySize);
assert(status == noErr);
//Adopt hardware input sample rate
AudioStreamBasicDescription deviceFormat;
status = AudioUnitGetProperty(player->inputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
inputBus,
&deviceFormat,
&propertySize);
assert(status == noErr);
player->streamFormat.mSampleRate = deviceFormat.mSampleRate;
printf("Sample Rate %f...\n", deviceFormat.mSampleRate);
propertySize = sizeof(AudioStreamBasicDescription);
status = AudioUnitSetProperty(player->inputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
inputBus,
&player->streamFormat,
propertySize);
assert(status == noErr);
//Calculating Capture buffer size for an I/O unit
UInt32 bufferSizeFrames = 0;
propertySize = sizeof(UInt32);
status = AudioUnitGetProperty(player->inputUnit,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&bufferSizeFrames,
&propertySize);
assert(status == noErr);
UInt32 bufferSizeBytes = bufferSizeFrames * sizeof(Float32);
//Create AudioBufferList to receive capture data
UInt32 propSize = offsetof(AudioBufferList, mBuffers[0]) +
(sizeof(AudioBuffer) * player->streamFormat.mChannelsPerFrame);
//Malloc buffer lists
player->inputBuffer = (AudioBufferList *) malloc(propSize);
player->inputBuffer->mNumberBuffers = player->streamFormat.mChannelsPerFrame;
//Pre malloc buffers for AudioBufferLists
for(UInt32 i = 0; i < player->inputBuffer->mNumberBuffers; i++){
player->inputBuffer->mBuffers[i].mNumberChannels = 1;
player->inputBuffer->mBuffers[i].mDataByteSize = bufferSizeBytes;
player->inputBuffer->mBuffers[i].mData = malloc(bufferSizeBytes);
}
//Create the ring buffer
player->ringBuffer = new CARingBuffer();
player->ringBuffer->Allocate(player->streamFormat.mChannelsPerFrame,
player->streamFormat.mBytesPerFrame,
bufferSizeFrames * 3);
printf("Number of channels: %d\n", player->streamFormat.mChannelsPerFrame);
printf("Number of buffers: %d\n", player->inputBuffer->mNumberBuffers);
//Set render proc to supply samples
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = InputRenderProc;
callbackStruct.inputProcRefCon = player;
status = AudioUnitSetProperty(player->inputUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
0,
&callbackStruct,
sizeof(callbackStruct);
assert(status == noErr);
status = AudioUnitInitialize(player->inputUnit);
assert(status == noErr);
player->firstInputSampleTime = -1;
player->inToOutSampleTimeOffset = -1;
printf("Finished CreateInputUnit()\n");
}
所以这是我的渲染回调函数，我正在访问各个缓冲区。 ：
OSStatus GraphRenderProc(void * inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
MyAUGraphPlayer * player = (MyAUGraphPlayer *) inRefCon;
if(player->firstOutputSampleTime < 0.0) {
player->firstOutputSampleTime = inTimeStamp->mSampleTime;
if((player->firstInputSampleTime > -1.0) &&
(player->inToOutSampleTimeOffset < 0.0)) {
player->inToOutSampleTimeOffset = player->firstInputSampleTime - player->firstOutputSampleTime;
}
}
//Copy samples out of ring buffer
OSStatus outputProcErr = noErr;
outputProcErr = player->ringBuffer->Fetch(ioData,
inNumberFrames,
inTimeStamp->mSampleTime + player->inToOutSampleTimeOffset);
//BUT THIS IS NOT HOW IT IS SUPPOSED TO WORK
Float32 * data = (Float32 *) ioData->mBuffers[0].mData;
Float32 * data2 = (Float32 *) ioData->mBuffers[1].mData;
for(int frame = 0; frame < inNumberFrames; frame++)
{
Float32 sample = data[frame] + data2[frame];
data[frame] = data2[frame] = sample;
}
return outputProcErr;
}
答案 0 :(得分：0)
虽然您的代码看似过于复杂，但我会尝试回答您的问题：
在回调中检索示例数据的概念没有任何问题。如果处理多声道音频设备，那将是不够的。设备有多少个通道，通过AudioStreamBasicDescription为给定设备查询的通道布局，格式等。此属性用于初始化您的处理链的其余部分。您在初始化时分配音频缓冲区，或让程序为您执行此操作（请阅读文档）。
如果您觉得使用额外的缓冲区更容易复制到数据处理和DSP，您可以在回调中管理它（简化代码）：
Float32 buf[streamFormat.mChanelsPerFrame][inNumberFrames];
for(int ch; ch < streamFormat.mChanelsPerFrame; ch++){
Float32 data = (Float32 *)ioData->mBuffers[ch].mData;
memcpy(buf[ch], data, inNumberFrames*sizeof(Float32));
}