오디오 선형 변환 pcm을 mp3로 변환 (LAME 사용) 낮음, 중간, 높음 오디오 품질 설정

Question

iOS에서 MP3로 LinearPCM을 인코딩합니다. AudioToolbox 프레임 워크 및 Lame을 사용하여 마이크에서 MP3로 원시 PCM 데이터를 인코딩하려고합니다. . 그리고 비록 내가 녹음 오디오를 녹음하면 잘 작동하는 것처럼 보입니다. 녹음 된 mp3 오디오 파일을 재생하는 동안 절름발이 인코딩 개념을 사용하여 mp3로 변환됩니다. 저역, 중음, 저역과 같은 오디오 (절름발이 사용) 고품질 mp3 파일. 난, 정확한 설정 (샘플 레이트, 비트 깊이, 비트 레이트, chennal, 품질)이 시도하는 동안 절름발이 변환 과정

void AQRecorder::MyInputBufferHandler( void *        inUserData, 
           AudioQueueRef      inAQ, 
           AudioQueueBufferRef     inBuffer, 
           const AudioTimeStamp *    inStartTime, 
           UInt32        inNumPackets, 
           const AudioStreamPacketDescription* inPacketDesc) 
    { 
    AQRecorder *aqr = (AQRecorder *)inUserData; 
    // NSLog(@"%f",inStartTime->mSampleTime); 
    try 
    { 
    if (inNumPackets > 0) 
    { 
     AudioFileWritePackets(aqr->mRecordFile, FALSE, inBuffer->mAudioDataByteSize, inPacketDesc, aqr->mRecordPacket, &inNumPackets, inBuffer->mAudioData); 

     aqr->mRecordPacket += inNumPackets; 

     int MP3_SIZE =inBuffer->mAudioDataByteSize * 4; 
     unsigned char mp3_buffer[MP3_SIZE]; 
     lame_t lame = lame_init(); 
     lame_set_in_samplerate(lame, 44100); 
     lame_set_VBR(lame, vbr_default); 
     lame_init_params(lame); 

    //    int encodedBytes=lame_encode_buffer_interleaved(lame, (short int *)inBuffer->mAudioData , inNumPackets, mp3_buffer, MP3_SIZE); 


     int encodedBytes = lame_encode_buffer(lame, (short*)inBuffer->mAudioData, (short*)inBuffer->mAudioData, inNumPackets, mp3_buffer, MP3_SIZE); 

     [delegate.mp3AudioData appendBytes:mp3_buffer length:encodedBytes]; 

     if (inBuffer->mAudioDataByteSize != 0) { 
     } 
     else 
     { 
      int encode=lame_encode_flush(lame, mp3_buffer, MP3_SIZE); 
      [delegate.mp3AudioData appendBytes:mp3_buffer length:encode]; 
     } 
     lame_close(lame); 
    } 

    if (aqr->IsRunning()) 
    { 
     AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL); 
    } 
    } catch (CAXException e) 
{ 
char buf[256]; 
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf)); 
} 
} 

Answer 1

를 모르는

낮은 품질 :

AppDelegate *appDelegate = (AppDelegate *)[[UIApplication sharedApplication]delegate]; 
     NSMutableDictionary *dictAudioQuality =[[NSMutableDictionary alloc]init]; 
     [dictAudioQuality setValue:@"Low" forKey:@"audioquality"]; 
     [dictAudioQuality setValue:@"11025" forKey:@"samplerate"]; 
     [dictAudioQuality setValue:@"16" forKey:@"bitdepth"]; 
     [dictAudioQuality setValue:@"120" forKey:@"bitrate"]; 
     [dictAudioQuality setValue:@"1" forKey:@"channel"]; 

중간 품질 :

AppDelegate *appDelegate = (AppDelegate *)[[UIApplication sharedApplication]delegate]; 
     NSMutableDictionary *dictAudioQuality =[[NSMutableDictionary alloc]init]; 
     [dictAudioQuality setValue:@"Medium" forKey:@"audioquality"]; 
     [dictAudioQuality setValue:@"22050" forKey:@"samplerate"]; 
     [dictAudioQuality setValue:@"16" forKey:@"bitdepth"]; 
     [dictAudioQuality setValue:@"240" forKey:@"bitrate"]; 
     [dictAudioQuality setValue:@"1" forKey:@"channel"]; 

고품질 :

AppDelegate *appDelegate = (AppDelegate *)[[UIApplication sharedApplication]delegate]; 
     NSMutableDictionary *dictAudioQuality =[[NSMutableDictionary alloc]init]; 
     [dictAudioQuality setValue:@"High" forKey:@"audioquality"]; 
     [dictAudioQuality setValue:@"44100" forKey:@"samplerate"]; 
     [dictAudioQuality setValue:@"24" forKey:@"bitdepth"]; 
     [dictAudioQuality setValue:@"320" forKey:@"bitrate"]; 
     [dictAudioQuality setValue:@"2" forKey:@"channel"]; 

AQRecorder.m 녹음 시작

void AQRecorder::StartRecord(CFStringRef inRecordFile) 
{ 
int i, bufferByteSize; 
UInt32 size; 

delegate =[[UIApplication sharedApplication]delegate]; 
nSampleRate =[[delegate.dictMP3Quality valueForKey:@"samplerate"] intValue]; 
nBitRate =[[delegate.dictMP3Quality valueForKey:@"bitrate"] intValue]; 
nChannel =[[delegate.dictMP3Quality valueForKey:@"channel"] intValue]; 

try { 

    UInt32 category = kAudioSessionCategory_RecordAudio; 

    OSStatus error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); 
    if (error) printf("couldn't set audio category!"); 

    // specify the recording format 
    SetupAudioFormat(kAudioFormatLinearPCM); 

    // create the queue 
    XThrowIfError(AudioQueueNewInput(
            &mRecordFormat, 
            MyInputBufferHandler, 
            this /* userData */, 
            NULL /* run loop */, NULL /* run loop mode */, 
            0 /* flags */, &mQueue), "AudioQueueNewInput failed"); 

    // get the record format back from the queue's audio converter -- 
    // the file may require a more specific stream description than was necessary to create the encoder. 
    mRecordPacket = 0; 

    size = sizeof(mRecordFormat); 
    XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_StreamDescription, 
             &mRecordFormat, &size), "couldn't get queue's format"); 

    // copy the cookie first to give the file object as much info as we can about the data going in 
    // not necessary for pcm, but required for some compressed audio 
    CopyEncoderCookieToFile(); 

    // allocate and enqueue buffers 
    bufferByteSize = ComputeRecordBufferSize(&mRecordFormat, kBufferDurationSeconds); // enough bytes for half a second 
    for (i = 0; i < kNumberRecordBuffers; ++i) { 
     XThrowIfError(AudioQueueAllocateBuffer(mQueue, bufferByteSize, &mBuffers[i]), 
         "AudioQueueAllocateBuffer failed"); 
     XThrowIfError(AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL), 
         "AudioQueueEnqueueBuffer failed"); 
    } 
    // start the queue 
    mIsRunning = true; 
    XThrowIfError(AudioQueueStart(mQueue, NULL), "AudioQueueStart failed"); 

    lame = lame_init(); 
    lame_set_in_samplerate(lame, mRecordFormat.mSampleRate); 
    lame_set_out_samplerate(lame, nSampleRate); 
    lame_set_num_channels(lame, nChannel); 
    // lame_set_brate(lame, nBitRate); 
    lame_set_VBR(lame, vbr_default); 
    lame_init_params(lame); 
} 
catch (CAXException e) { 
    char buf[256]; 
    fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf)); 
} 
catch (...) { 
    fprintf(stderr, "An unknown error occurred\n");; 
} 
} 

AQRecorder :: MyInputBufferHandler

void AQRecorder::MyInputBufferHandler( void *        inUserData, 
            AudioQueueRef      inAQ, 
            AudioQueueBufferRef     inBuffer, 
            const AudioTimeStamp *    inStartTime, 
            UInt32        inNumPackets, 
            const AudioStreamPacketDescription* inPacketDesc) 
{ 
AQRecorder *aqr = (AQRecorder *)inUserData; 
try 
{ 

    if (inNumPackets > 0) 
    { 
     AudioFileWritePackets(aqr->mRecordFile, FALSE, inBuffer->mAudioDataByteSize, inPacketDesc, aqr->mRecordPacket, &inNumPackets, inBuffer->mAudioData); 

     aqr->mRecordPacket += inNumPackets; 

     int MP3_SIZE =inNumPackets * 4; 
     unsigned char mp3_buffer[MP3_SIZE]; 

     memset(mp3_buffer, 0, sizeof(mp3_buffer)); 
    // int encodedBytes=lame_encode_buffer_interleaved(lame, (short int*)inBuffer->mAudioData , inNumPackets, mp3_buffer, MP3_SIZE); 

     int encodedBytes = lame_encode_buffer(aqr->lame, (short int*)inBuffer->mAudioData, (short int*)inBuffer->mAudioData, inNumPackets, mp3_buffer, MP3_SIZE); 

     [aqr->delegate.mp3AudioData appendBytes:mp3_buffer length:encodedBytes]; 

     if (inBuffer->mAudioDataByteSize != 0) { 
     } 
     else 
     { 
      int encode=lame_encode_flush(aqr->lame, mp3_buffer, MP3_SIZE); 
      [aqr->delegate.mp3AudioData appendBytes:mp3_buffer length:encode]; 
     } 

     { 
      NSLog(@"------------"); 
      NSLog(@"%d",encodedBytes); 
      NSLog(@"%lu",inNumPackets); 
      NSLog(@"%d",MP3_SIZE); 
      NSLog(@"------------"); 
     } 
    } 

    if (aqr->IsRunning()) 
    { 
     AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL); 
    } 
} catch (CAXException e) 
{ 
    char buf[256]; 
    fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf)); 
} 
}