रिकॉर्ड और iOS

Question

में एक साथ ऑडियो चला रहा है, जबकि रिकॉर्डिंग रिकॉर्ड किए गए सामग्री एक साथ खेलने के लिए कोशिश कर रहा हूँ। वर्तमान में मैं AVAudioRecorder रिकॉर्डिंग के लिए और AVAudioPlayer खेलने के लिए उपयोग कर रहा हूं।

जब मैं सामग्री एक साथ कुछ भी नहीं खेल रहा है खेलने के लिए कोशिश कर रहा था। मैं जो कर रहा हूं उसके लिए कृपया छद्म कोड पाएं।

मैं रोक के बाद एक ही सामान करते हैं रिकॉर्डिंग सब कुछ ठीक काम करता है।

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]; 

क्या आप कृपया मुझे अपने विचार या विचारों को बताने दे सकते हैं?

Answer 1

यदि आपको अपने ऑडियो इनपुट की वास्तविक समय की निगरानी चाहते AudioUnits का उपयोग करना होगा।

• Apple's Audio Unit Hosting Guide
• Tutorial on configuring the Remote I/O Audio Unit

Answer 2

RemoteIO ऑडियो यूनिट एक साथ रिकॉर्ड के लिए इस्तेमाल किया और खेलने जा सकता है। रिमोटियो (ऑरियो टच) का उपयोग करके रिकॉर्डिंग के कई उदाहरण हैं और रिमोटियो का उपयोग कर खेल रहे हैं। बस यूनिट इनपुट और यूनिट आउटपुट दोनों को सक्षम करें, और दोनों बफर कॉलबैक को संभाल लें। एक उदाहरण देखें here

Answer 3

यह प्राप्त है, इन लिंक का उपयोग करें और इसे डाउनलोड: https://code.google.com/p/ios-coreaudio-example/downloads/detail?name=Aruts.zip&can=2&q=

यह लिंक स्पीकर से ध्वनि खेलेंगे लेकिन यह रिकॉर्ड नहीं करेगा, मैं रिकॉर्ड कार्यक्षमता को लागू किया है और साथ ही नीचे पूर्ण कोड वर्णन है ..

ज फ़ाइल में

#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; 

मीटर में

#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); 
} 

.. यह निश्चित रूप से आप लोगों की मदद करेंगे

ऐसा करने का एक और सबसे अच्छा तरीका https://github.com/tkzic/audiograph से ऑडियो टच डाउनलोड करने और इस आवेदन में यह आवाज को दोहराने के रूप में आप से बात की इको समारोह को देखने के लिए है, लेकिन यह इतना ऑडियो रिकॉर्ड नहीं करता है इसे में रिकॉर्डिंग समारोह जोड़ें, के रूप में नीचे वर्णित:

IN MixerHostAudio.h 

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



IN MixerHostAudio.m 

// इस वर्ग

-(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); 

और कॉल में इन दोनों समारोह जोड़ें में MixerHostViewController.m से इन समारोह - (IBAction) playOrStop: (आईडी) इस विधि