Speex(在ios中)声音质量很差(和机器人)

概述我正在尝试使用speex进行编码/解码,但是当我不这样做时,音频响亮而清晰,但是当我编码/解码以测试音频质量时,我的音质非常差,机器人声音也很差. 这是我的init音频方法： #define AUDIO_QUALITY 10- (void) initAudio { try { //SPEEX CONFIG speex_bits_init(&bits_ 我正在尝试使用speex进行编码/解码,但是当我不这样做时,音频响亮而清晰,但是当我编码/解码以测试音频质量时,我的音质非常差,机器人声音也很差.

这是我的init音频方法：

#define AUdio_QUAliTY 10- (voID) initAudio {    try {           //SPEEX CONfig        speex_bits_init(&bits_in);        speex_bits_init(&bits_out);        enc_state = speex_encoder_init(&speex_nb_mode);        dec_state = speex_decoder_init(&speex_nb_mode);        int quality = AUdio_QUAliTY;        speex_encoder_ctl(enc_state,SPEEX_SET_QUAliTY,&quality);        int tmp=1;        speex_decoder_ctl(dec_state,SPEEX_SET_ENH,&tmp);        Osstatus status;        XThrowIfError(AudioSessionInitialize(NulL,NulL,rioInterruptionListener,self),"Couldn't initialize audio session");        float aBufferLength = 0.02; // In seconds        status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHarDWareIOBufferDuration,sizeof(aBufferLength),&aBufferLength);        XThrowIfError(status,"");        UInt32 audiocategory = kAudioSessioncategory_PlayAndRecord;        XThrowIfError(AudioSessionSetProperty(kAudioSessionProperty_Audiocategory,sizeof(audiocategory),&audiocategory),"Couldn't set audio category");        XThrowIfError(AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange,propListener,"Couldn't set property Listener");        // 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,&rIoUnit);        XThrowIfError(status,"1");        // Enable IO for recording        UInt32 flag = 1;        status = AudioUnitSetProperty(rIoUnit,kAudioOutputUnitProperty_EnableIO,kAudioUnitScope_input,kinputBus,&flag,sizeof(flag));        XThrowIfError(status,"2");        // Enable IO for playback        status = AudioUnitSetProperty(rIoUnit,kAudioUnitScope_Output,kOutputBus,"3");        // Describe format        AudioStreamBasicDescription audioFormat;        audioFormat.mSampleRate         = 8000.00;        audioFormat.mFormatID           = kAudioFormatlinearPCM;        audioFormat.mFormatFlags        =   kAudioFormatFlagIsSignedInteger |                                            kAudioFormatFlagsNativeEndian |                                            kAudioFormatFlagIsPacked;        audioFormat.mFramesPerPacket    = 1;        audioFormat.mChannelsPerFrame   = 1;        audioFormat.mBitsPerChannel     = 16;        audioFormat.mBytesPerPacket     = 2;        audioFormat.mBytesPerFrame      = 2;        // Apply format        status = AudioUnitSetProperty(rIoUnit,kAudioUnitProperty_StreamFormat,&audioFormat,sizeof(audioFormat));        XThrowIfError(status,"");        status = AudioUnitSetProperty(rIoUnit,"");        // Set input callback        AURenderCallbackStruct callbackStruct;        callbackStruct.inputProc = recordingCallback;        callbackStruct.inputProcRefCon = self;        status = AudioUnitSetProperty(rIoUnit,kAudioOutputUnitProperty_SetinputCallback,kAudioUnitScope_Global,&callbackStruct,sizeof(callbackStruct));        XThrowIfError(status,"");        // Set output callback        callbackStruct.inputProc = playingCallback;        callbackStruct.inputProcRefCon = self;        status = AudioUnitSetProperty(rIoUnit,kAudioUnitProperty_SetRenderCallback,"");        // disable buffer allocation for the recorder (optional - do this if we want to pass in our own)        flag = 0;        status = AudioUnitSetProperty(rIoUnit,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 = FRAME_SIZE * 2;        tempBuffer.mData = malloc( FRAME_SIZE * 2 );        XThrowIfError(AudioSessionSetActive(true),"Couldn't set audio session active\n");        // Initialise        status = AudioUnitinitialize(rIoUnit);        XThrowIfError(status,"");        status = AudioOutputUnitStart(rIoUnit);        XThrowIfError(status,"");    }    catch (CAXException &e) {        NSLog(@"CAXException...");    }    catch (...) {        fprintf(stderr,"An unkNown error occurred\n");    }}

我的speex编码&解码功能：

#define FRAME_SIZE 160#define COMP_FRAME_SIZE 62char* encodeSpeex(spx_int16_t *buffer,UInt32 inSize,int *encodedSize) {    char *outputBuffer = (char *)malloc(COMP_FRAME_SIZE);    speex_bits_reset(&bits_in);    speex_encode_int(enc_state,buffer,&bits_in);    *encodedSize = speex_bits_write(&bits_in,outputBuffer,FRAME_SIZE * 2);    return outputBuffer;}short* decodeSpeex(char* buffer,int encodedSize,int decodedSize) {    short *outTemp = (short *)calloc(1,FRAME_SIZE * 2);    speex_bits_read_from(&bits_out,encodedSize * FRAME_SIZE * *2);    speex_decode_int(dec_state,&bits_out,outTemp);    return outTemp;}

最后,调用speex编码的功能&解码并将其复制到将由特定回调播放的缓冲区：

- (voID) processAudio: (audiobufferlist*) bufferList{    AudioBuffer sourceBuffer = bufferList->mBuffers[0];    NSLog(@"Origin size: %lu",sourceBuffer.mDataByteSize);    int size = 0;    char *encodedAudio = encodeSpeex((spx_int16_t*) sourceBuffer.mData,sourceBuffer.mDataByteSize,&size);    NSLog(@"Encoded size: %i",size);    short* decodedAudio = decodeSpeex(encodedAudio,size,sourceBuffer.mDataByteSize);    free(encodedAudio);    memcpy(tempBuffer.mData,decodedAudio,FRAME_SIZE * 2);    free(decodedAudio);        }

任何人都知道我为什么会这么差的质量？根据网站上的speex示例,它不应该像那样……

解决方法 我也偶然发现了这个问题.我通过检查我的缓冲区是否实际填充正确来解决它.否则,您将获得播放的空数据,从而产生机器人声音. 总结

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