问题描述
我正在尝试同时录制和播放音频。
当前输出：-音频的输入和输出相同，即内置麦克风或AirPods。
需要的输出：-音频的输入和输出应该不同，即音频输入应该来自内置麦克风，音频输出应该来自连接的AirPods。
参考应用程序：-https://apps.apple.com/gb/app/chatable-hear-better/id1494968908
#import <AudioToolBox/AudioToolBox.h>
#define kOutputBus 0
#define kInputBus 1
IosAudioController* iosAudio;
void checkStatus(int status){
if (status) {
printf("Status not 0! %d\n",status);
// exit(1);
}
}
/**
This callback is called when new audio data from the microphone is
available.
*/
static Osstatus recordingCallback(void *inRefCon,AudioUnitRenderActionFlags *ioActionFlags,const AudioTimeStamp *inTimeStamp,UInt32 inBusNumber,UInt32 inNumberFrames,audiobufferlist *ioData) {
// Because of the way our audio format (setup below) is chosen:
// we only need 1 buffer,since it is mono
// Samples are 16 bits = 2 bytes.
// 1 frame includes only 1 sample
AudioBuffer buffer;
buffer.mNumberChannels = 1;
buffer.mDataByteSize = inNumberFrames * 2;
buffer.mData = malloc( inNumberFrames * 2 );
// Put buffer in a audiobufferlist
audiobufferlist bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// Then:
// Obtain recorded samples
Osstatus status;
status = AudioUnitRender([iosAudio audioUnit],ioActionFlags,inTimeStamp,inBusNumber,inNumberFrames,&bufferList);
checkStatus(status);
// Now,we have the samples we just read sitting in buffers in bufferList
// Process the new data
[iosAudio processAudio:&bufferList];
// release the malloc'ed data in the buffer we created earlier
free(bufferList.mBuffers[0].mData);
return noErr;
}
/**
This callback is called when the audioUnit needs new data to play through the
speakers. If you don't have any,just don't write anything in the buffers
*/
static Osstatus playbackCallback(void *inRefCon,audiobufferlist *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
for (int i=0; i < ioData->mNumberBuffers; i++) { // in practice we will only ever have 1 buffer,since audio format is mono
AudioBuffer buffer = ioData->mBuffers[i];
// NSLog(@" Buffer %d has %d channels and wants %d bytes of data.",i,buffer.mNumberChannels,buffer.mDataByteSize);
// copy temporary buffer data to output buffer
UInt32 size = min(buffer.mDataByteSize,[iosAudio tempBuffer].mDataByteSize); // dont copy more data then we have,or then fits
memcpy(buffer.mData,[iosAudio tempBuffer].mData,size);
buffer.mDataByteSize = size; // indicate how much data we wrote in the buffer
// uncomment to hear random noise
/*
UInt16 *frameBuffer = buffer.mData;
for (int j = 0; j < inNumberFrames; j++) {
frameBuffer[j] = rand();
}
*/
}
return noErr;
}
@implementation IosAudioController
@synthesize audioUnit,tempBuffer;
/**
Initialize the audioUnit and allocate our own temporary buffer.
The temporary buffer will hold the latest data coming in from the microphone,and will be copied to the output when this is requested.
*/
- (id) init {
self = [super init];
Osstatus status;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL,&desc);
// Get audio units
status = AudioComponentInstanceNew(inputComponent,&audioUnit);
checkStatus(status);
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,kAudioOutputUnitProperty_EnableIO,kAudioUnitScope_Input,kInputBus,&flag,sizeof(flag));
checkStatus(status);
// Enable IO for playback
status = AudioUnitSetProperty(audioUnit,kAudioUnitScope_Output,kOutputBus,sizeof(flag));
checkStatus(status);
// Describe format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;//kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
// Apply format
status = AudioUnitSetProperty(audioUnit,kAudioUnitProperty_StreamFormat,&audioFormat,sizeof(audioFormat));
checkStatus(status);
status = AudioUnitSetProperty(audioUnit,sizeof(audioFormat));
checkStatus(status);
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit,kAudioOutputUnitProperty_SetInputCallback,kAudioUnitScope_Global,&callbackStruct,sizeof(callbackStruct));
checkStatus(status);
// Set output callback
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit,kAudioUnitProperty_SetRenderCallback,sizeof(callbackStruct));
checkStatus(status);
// disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,kAudioUnitProperty_ShouldAllocateBuffer,sizeof(flag));
// Allocate our own buffers (1 channel,16 bits per sample,thus 16 bits per frame,thus 2 bytes per frame).
// Practice learns the buffers used contain 512 frames,if this changes it will be fixed in processAudio.
tempBuffer.mNumberChannels = 1;
tempBuffer.mDataByteSize = 512 * 2;
tempBuffer.mData = malloc( 512 * 2 );
// Initialise
status = AudioUnitinitialize(audioUnit);
checkStatus(status);
return self;
}
/**
Start the audioUnit. This means data will be provided from
the microphone,and requested for Feeding to the speakers,by
use of the provided callbacks.
*/
- (void) start {
Osstatus status = AudioOutputUnitStart(audioUnit);
// float volume = 0.5;
// status = AudioUnitSetProperty(audioUnit,kMultiChannelmixerParam_Volume,&volume,sizeof(volume));
// NSLog(@"status = %d",(int)status);
UInt32 sessionCategory = kAudioSessionCategory_PlayAndRecord;
AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,sizeof(sessionCategory),&sessionCategory);
// Speaker Playback
UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker;
AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryMixWithOthers,sizeof(audioRouteOverride),&audioRouteOverride);
AudioSessionSetActive(true);
// UInt32 enabled = true;
//
// UInt32 category = kAudioSessionCategory_PlayAndRecord;
// status = AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryEnableBluetoothInput,sizeof(UInt32),&category);
//
// status = AudioSessionSetActive(true);
//
// status = AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryDefaultToSpeaker,&enabled);
checkStatus(status);
}
/**
Stop the audioUnit
*/
- (void) stop {
Osstatus status = AudioOutputUnitStop(audioUnit);
checkStatus(status);
}
/**
Change this funtion to decide what is done with incoming
audio data from the microphone.
Right Now we copy it to our own temporary buffer.
*/
- (void) processAudio: (audiobufferlist*) bufferList{
AudioBuffer sourceBuffer = bufferList->mBuffers[0];
// fix tempBuffer size if it's the wrong size
if (tempBuffer.mDataByteSize != sourceBuffer.mDataByteSize) {
free(tempBuffer.mData);
tempBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
tempBuffer.mData = malloc(sourceBuffer.mDataByteSize);
}
// copy incoming audio data to temporary buffer
memcpy(tempBuffer.mData,bufferList->mBuffers[0].mData,bufferList->mBuffers[0].mDataByteSize);
}
/**
Clean up.
*/
- (void) dealloc {
[super dealloc];
AudioUnitUninitialize(audioUnit);
free(tempBuffer.mData);
}
解决方法
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