ios - Speex(在 iOS 中)音质真的很差(和机器人)

Question

我正在尝试使用 speex 进行编码/解码，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，我得到的音频质量非常差，而且声音很机械。

这是我的初始化音频方法:

#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, self), "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, 
                                      kAudioOutputUnitProperty_EnableIO, 
                                      kAudioUnitScope_Output, 
                                      kOutputBus,
                                      &flag, 
                                      sizeof(flag));
        XThrowIfError(status, "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, 
                                      kAudioUnitScope_Output, 
                                      kInputBus, 
                                      &audioFormat, 
                                      sizeof(audioFormat));
        XThrowIfError(status, "");

        status = AudioUnitSetProperty(rioUnit, 
                                      kAudioUnitProperty_StreamFormat, 
                                      kAudioUnitScope_Input, 
                                      kOutputBus, 
                                      &audioFormat, 
                                      sizeof(audioFormat));
        XThrowIfError(status, "");

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

        // Set output callback
        callbackStruct.inputProc = playingCallback;
        callbackStruct.inputProcRefCon = self;
        status = AudioUnitSetProperty(rioUnit, 
                                      kAudioUnitProperty_SetRenderCallback, 
                                      kAudioUnitScope_Global, 
                                      kOutputBus,
                                      &callbackStruct, 
                                      sizeof(callbackStruct));
        XThrowIfError(status, "");

        // Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
        flag = 0;
        status = AudioUnitSetProperty(rioUnit, 
                                      kAudioUnitProperty_ShouldAllocateBuffer,
                                      kAudioUnitScope_Output, 
                                      kInputBus,
                                      &flag, 
                                      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 编码和解码功能:

#define FRAME_SIZE 160
#define COMP_FRAME_SIZE 62
char* 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, buffer, encodedSize * FRAME_SIZE * *2);
    speex_decode_int(dec_state, &bits_out, outTemp);
    return outTemp;
}

最后，调用 speex 编码和解码的函数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);        
}

有人知道为什么我的质量这么差吗？根据网站上的 speex 示例，它不应该那样呈现...

Answer 1

我也遇到过这个问题。我通过检查我的缓冲区是否已正确填充来解决它。否则你会得到播放的空数据，导致机器人声音。