首页 \ 问答 \ 如何使用外部音频接口访问具有核心音频的单个通道(How to access individual channels with core audio from an external audio interface)

如何使用外部音频接口访问具有核心音频的单个通道(How to access individual channels with core audio from an external audio interface)

 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。  
 我能够从我的渲染回调函数中的AudioBufferList中删除并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;
}

I am learning CoreAudio and I am just going through some of the examples on Apple's documentation and figuring out how to set things up and what not. So far I am able to connect to the default connected audio input device and output it to the default output device. I connected a 2 channel interface and was able to output the input from it and output it as well.  
However I was searching through their API references and examples, but could not find any thing substantial to access the individual input channels from my interface.  
I was able to hack away and extract the samples from the AudioBufferList in my Render Callback function and manipulate it that way but I am wondering if there is a correct way or a more official way of accessing the data from each individual input channels. 
EDIT: 
This is the user data that I found from an example that I am using: 
typedef struct MyAUGraphPlayer
{
    AudioStreamBasicDescription streamFormat;

    AUGraph graph;
    AudioUnit inputUnit;
    AudioUnit outputUnit;

    AudioBufferList * inputBuffer;
    CARingBuffer * ringBuffer;

    Float64 firstInputSampleTime;
    Float64 firstOutputSampleTime;
    Float64 inToOutSampleTimeOffset;

} MyAUGraphPlayer;
 
This is how i set up the input unit: 
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");

}
 
So this is my render callback function where I am accessing the individual buffers. :  
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;
}

原文：https://stackoverflow.com/questions/37606550

更新时间：2021-12-15 12:12

最满意答案

 您应该将代码拆分为更多行以处理这些错误或特殊情况。 mysql_query将返回0到n行，如果发生则返回错误。 因此，返回的资源仅在非错误查询时才为真。 这可用于处理如下情况。  
 首先构建并执行查询，然后处理资源 。  
$query="SELECT COUNT(user_id) FROM users WHERE username = ".$username;
$result = mysql_query($query);  
 
 如果出现错误，您可以使用以下内容确定发生了什么：  
if(!$result) die("SELECT failed: ".mysql_error());
 
 或者这些想法来处理这个问题  
if (!$result=mysql_query($query)) {
        return false; // or similar operation
    }

    if (mysql_num_rows($result)!=1){
        return false;
    }else{
        return true;
    }

You should split your code in some more lines to handle those errors or special cases. mysql_query will return zero to n rows or an error if it occurs. The returned resource will therefore only be true on non-error queries. This can be used to handle such situations like follows. 
At first build and execute query, next process the resource. 
$query="SELECT COUNT(user_id) FROM users WHERE username = ".$username;
$result = mysql_query($query);  
 
u may use the following to determine what is going on in case of an error: 
if(!$result) die("SELECT failed: ".mysql_error());
 
or these idea to handle the problem 
if (!$result=mysql_query($query)) {
        return false; // or similar operation
    }

    if (mysql_num_rows($result)!=1){
        return false;
    }else{
        return true;
    }

如何使用外部音频接口访问具有核心音频的单个通道(How to access individual channels with core audio from an external audio interface)

最满意答案

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