2012-02-26 211 views
9

我試圖在錄製過程中同時播放錄製的內容。目前我正在使用AVAudioRecorder進行錄製,並使用AVAudioPlayer進行播放。在iOS中同時錄製和播放音頻

當我試圖同時播放內容時沒有播放任何內容。請爲我正在做的事找到僞代碼。

如果我在停止錄音後做同樣的事情一切正常。

AVAudioRecorder *recorder; //Initializing the recorder properly. 
[recorder record]; 
NSError *error=nil; 
NSUrl recordingPathUrl;  //Contains the recording path. 
AVAudioPlayer *audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:recordingPathUrl 
                    error:&error]; 
[audioPlayer prepareToPlay]; 
[audioPlayer play]; 

請問您有人讓我知道您的想法或想法嗎?

+1

你正在記錄什麼類型的文件?如果您正在錄製MP4/MOV文件,那麼這是不可能的,因爲在記錄停止之前MOV原子不會被寫入文件。我不確定MP3。 – 2012-02-27 20:21:02

+0

[記錄和同時播放音頻]的可能重複(http://stackoverflow.com/questions/4215180/record-and-play-audio-simultaneously) – 2012-06-23 05:40:12

+0

您可以使用核心音頻去掉這個問題。安裝需要一些時間,但可以輕鬆完成。 – dubbeat 2012-07-10 10:26:10

回答

0

的RemoteIO音頻單元可用於同時錄製和播放。有很多使用RemoteIO(aurioTouch)錄製和使用RemoteIO進行錄製的示例。只需啓用單元輸入和單元輸出,並處理兩個緩衝區回調。看一個例子here

5

這是可以實現的,使用這些鏈接和下載: https://code.google.com/p/ios-coreaudio-example/downloads/detail?name=Aruts.zip&can=2&q=

此鏈接將從揚聲器播放聲音,但不會記錄它,我已經實現了創紀錄的功能,以及下面是完整的代碼說明..

在.h文件中

#import <Foundation/Foundation.h> 
#import <AudioToolbox/AudioToolbox.h> 

#ifndef max 
#define max(a, b) (((a) > (b)) ? (a) : (b)) 
#endif 

#ifndef min 
#define min(a, b) (((a) < (b)) ? (a) : (b)) 
#endif 


@interface IosAudioController : NSObject { 
    AudioComponentInstance audioUnit; 
    AudioBuffer tempBuffer; // this will hold the latest data from the microphone 
    ExtAudioFileRef    mAudioFileRef; 
} 
@property (readonly)ExtAudioFileRef  mAudioFileRef; 
@property (readonly) AudioComponentInstance audioUnit; 
@property (readonly) AudioBuffer tempBuffer; 

- (void) start; 
- (void) stop; 
- (void) processAudio: (AudioBufferList*) bufferList; 

@end 

// setup a global iosAudio variable, accessible everywhere 
extern IosAudioController* iosAudio; 

IN .M

#import "IosAudioController.h" 
#import <AudioToolbox/AudioToolbox.h> 
#import <AVFoundation/AVFoundation.h> 
#define kOutputBus 0 
#define kInputBus 1 

IosAudioController* iosAudio; 

void checkStatus(int status){ 
    if (status) { 
     printf("Status not 0! %d\n", status); 
//  exit(1); 
    } 
} 




static void printAudioUnitRenderActionFlags(AudioUnitRenderActionFlags * ioActionFlags) 
{ 
    if (*ioActionFlags == 0) { 

     printf("AudioUnitRenderActionFlags(%lu) ", *ioActionFlags); 
     return; 
    } 
    printf("AudioUnitRenderActionFlags(%lu): ", *ioActionFlags); 
    if (*ioActionFlags & kAudioUnitRenderAction_PreRender)    printf("kAudioUnitRenderAction_PreRender "); 
    if (*ioActionFlags & kAudioUnitRenderAction_PostRender)    printf("kAudioUnitRenderAction_PostRender "); 
    if (*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence)  printf("kAudioUnitRenderAction_OutputIsSilence "); 
    if (*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight)  printf("kAudioOfflineUnitRenderAction_Prefli ght "); 
    if (*ioActionFlags & kAudioOfflineUnitRenderAction_Render)   printf("kAudioOfflineUnitRenderAction_Render"); 
    if (*ioActionFlags & kAudioOfflineUnitRenderAction_Complete)  printf("kAudioOfflineUnitRenderAction_Complete "); 
    if (*ioActionFlags & kAudioUnitRenderAction_PostRenderError)  printf("kAudioUnitRenderAction_PostRenderError "); 
    if (*ioActionFlags & kAudioUnitRenderAction_DoNotCheckRenderArgs) printf("kAudioUnitRenderAction_DoNotCheckRenderArgs "); 
} 


/** 
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) { 

    double timeInSeconds = inTimeStamp->mSampleTime/44100.00; 

    printf("\n%fs inBusNumber: %lu inNumberFrames: %lu ", timeInSeconds, inBusNumber, inNumberFrames); 

    printAudioUnitRenderActionFlags(ioActionFlags); 

    // 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; 

    SInt16 samples[inNumberFrames]; // A large enough size to not have to worry about buffer overrun 
    memset (&samples, 0, sizeof (samples)); 



    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]; 


    // Now, we have the samples we just read sitting in buffers in bufferList 
     ExtAudioFileWriteAsync([iosAudio mAudioFileRef], inNumberFrames, &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, 
           AudioUnitRenderActionFlags *ioActionFlags, 
           const AudioTimeStamp *inTimeStamp, 
           UInt32 inBusNumber, 
           UInt32 inNumberFrames, 
           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,mAudioFileRef; 

/** 
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; 

    AVAudioSession *session = [AVAudioSession sharedInstance]; 
    NSLog(@"%f",session.preferredIOBufferDuration); 


    // 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); 

    // 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, 
            kAudioOutputUnitProperty_EnableIO, 
            kAudioUnitScope_Output, 
            kOutputBus, 
            &flag, 
            sizeof(flag)); 
    checkStatus(status); 

    // Describe format 
    AudioStreamBasicDescription audioFormat; 
    audioFormat.mSampleRate   = 44100.00; 
    audioFormat.mFormatID   = kAudioFormatLinearPCM; 
    audioFormat.mFormatFlags  = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; 
    audioFormat.mFramesPerPacket = 1; 
    audioFormat.mChannelsPerFrame = 1; 
    audioFormat.mBitsPerChannel  = 16; 
    audioFormat.mBytesPerPacket  = 2; 
    audioFormat.mBytesPerFrame  = 2; 

    // Apply format 
    status = AudioUnitSetProperty(audioUnit, 
            kAudioUnitProperty_StreamFormat, 
            kAudioUnitScope_Output, 
            kInputBus, 
            &audioFormat, 
            sizeof(audioFormat)); 
    checkStatus(status); 
    status = AudioUnitSetProperty(audioUnit, 
            kAudioUnitProperty_StreamFormat, 
            kAudioUnitScope_Input, 
            kOutputBus, 
            &audioFormat, 
            sizeof(audioFormat)); 
    checkStatus(status); 


    // Set input callback 
    AURenderCallbackStruct callbackStruct; 
    callbackStruct.inputProc = recordingCallback; 
    callbackStruct.inputProcRefCon = self; 
    status = AudioUnitSetProperty(audioUnit, 
            kAudioOutputUnitProperty_SetInputCallback, 
            kAudioUnitScope_Global, 
            kInputBus, 
            &callbackStruct, 
            sizeof(callbackStruct)); 
    checkStatus(status); 

    // Set output callback 
    callbackStruct.inputProc = playbackCallback; 
    callbackStruct.inputProcRefCon = self; 
    status = AudioUnitSetProperty(audioUnit, 
            kAudioUnitProperty_SetRenderCallback, 
            kAudioUnitScope_Global, 
            kOutputBus, 
            &callbackStruct, 
            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, 
            kAudioUnitScope_Output, 
            kInputBus, 
            &flag, 
            sizeof(flag)); 

    // set preferred buffer size 
    Float32 audioBufferSize = (0.023220); 
    UInt32 size = sizeof(audioBufferSize); 
    status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, 
            size, &audioBufferSize); 

    // 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); 





    NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); 
    NSString *documentsDirectory = [paths objectAtIndex:0]; 
    NSString *destinationFilePath = [[NSString alloc] initWithFormat: @"%@/output.caf", documentsDirectory]; 
    NSLog(@">>> %@\n", destinationFilePath); 

    CFURLRef destinationURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, (CFStringRef)destinationFilePath, kCFURLPOSIXPathStyle, false); 

    OSStatus setupErr = ExtAudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &audioFormat, NULL, kAudioFileFlags_EraseFile, &mAudioFileRef); 
    CFRelease(destinationURL); 

    NSAssert(setupErr == noErr, @"Couldn't create file for writing"); 


    setupErr = ExtAudioFileSetProperty(mAudioFileRef, kExtAudioFileProperty_ClientDataFormat, sizeof(AudioStreamBasicDescription), &audioFormat); 
    NSAssert(setupErr == noErr, @"Couldn't create file for format"); 


    setupErr = ExtAudioFileWriteAsync(mAudioFileRef, 0, NULL); 
    NSAssert(setupErr == noErr, @"Couldn't initialize write buffers for audio file"); 

    // Initialise 
    status = AudioUnitInitialize(audioUnit); 
    checkStatus(status); 

    // [NSTimer scheduledTimerWithTimeInterval:5 target:self selector:@selector(stopRecording:) userInfo:nil repeats:NO]; 

    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); 
    checkStatus(status); 
} 

/** 
Stop the audioUnit 
*/ 
- (void) stop { 
    OSStatus status = AudioOutputUnitStop(audioUnit); 
    checkStatus(status); 
    [self stopRecording:nil]; 
} 

/** 
Change this function 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); 
} 


- (void)stopRecording:(NSTimer*)theTimer 
{ 
    printf("\nstopRecording\n"); 
    OSStatus status = ExtAudioFileDispose(mAudioFileRef); 
    printf("OSStatus(ExtAudioFileDispose): %ld\n", status); 
} 

/** 
Clean up. 
*/ 
- (void) dealloc { 
    [super dealloc]; 
    AudioUnitUninitialize(audioUnit); 
    free(tempBuffer.mData); 
} 

這一定會幫助你的人..

這樣做的另一個最好的辦法是從https://github.com/tkzic/audiograph下載音頻的觸摸和看見這個應用程序,你講它重複聲音的回聲的功能,但它不記錄音頻等等添加錄音功能進去,如下所述:在這個類

-(void)Record{ 
    NSString *completeFileNameAndPath = [[NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject] stringByAppendingString:@"/Record.wav"]; 
    //create the url that the recording object needs to reference the file 
    CFURLRef audioFileURL = CFURLCreateFromFileSystemRepresentation (NULL, (const UInt8 *)[completeFileNameAndPath cStringUsingEncoding:[NSString defaultCStringEncoding]] , strlen([completeFileNameAndPath cStringUsingEncoding:[NSString defaultCStringEncoding]]), false); 
    AudioStreamBasicDescription dstFormat, clientFormat; 
    memset(&dstFormat, 0, sizeof(dstFormat)); 
    memset(&clientFormat, 0, sizeof(clientFormat)); 

    AudioFileTypeID fileTypeId = kAudioFileWAVEType; 
     UInt32 size = sizeof(dstFormat); 
    dstFormat.mFormatID = kAudioFormatLinearPCM; 

    // setup the output file format 
    dstFormat.mSampleRate = 44100.0; // set sample rate 

    // create a 16-bit 44100kHz Stereo format 
    dstFormat.mChannelsPerFrame = 2; 
    dstFormat.mBitsPerChannel = 16; 
    dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 4; 
    dstFormat.mFramesPerPacket = 1; 
    dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian 

    //get the client format directly from 
    UInt32 asbdSize = sizeof (AudioStreamBasicDescription); 
    AudioUnitGetProperty(mixerUnit, 
         kAudioUnitProperty_StreamFormat, 
         kAudioUnitScope_Input, 
         0, // input bus 
         &clientFormat, 
         &asbdSize); 

    ExtAudioFileCreateWithURL(audioFileURL, fileTypeId, &dstFormat, NULL, kAudioFileFlags_EraseFile, &mRecordFile); 


     printf("recording\n"); 
     ExtAudioFileSetProperty(mRecordFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat); 
     //call this once as this will alloc space on the first call 
     ExtAudioFileWriteAsync(mRecordFile, 0, NULL); 


} 



-(void)StopRecord{ 
    ExtAudioFileDispose(mRecordFile); 
} 



//In micLineInCallback function Add this line at last before return noErr; : 

    ExtAudioFileWriteAsync([THIS mRecordFile] , inNumberFrames, ioData); 

,並呼籲

IN MixerHostAudio.h 

@property (readwrite) ExtAudioFileRef mRecordFile; 
-(void)Record; 
-(void)StopRecord; 



IN MixerHostAudio.m 

//添加這兩個功能這些函數來自 - (IBAction)中的MixerHostViewController.m playOrStop:(id)發送方法

+0

給你一個問題。您是通過麥克風錄製聲音還是直接將音頻播放數據直接傳輸到文件,以免環境噪音不會通過麥克風錄製? – 2014-04-25 02:38:21

+0

它通過麥克風錄製聲音 – 2014-04-25 04:47:51