音频输出到 .wav 文件是有效的，但是当使用 Vorbis 编码时，它会记录静音吗？

Question

过去三天我一直在寻找错误，但我有点放弃了。我已经研究了 OpenAL SDK 和 Vorbis 示例中的所有示例，但无济于事，所以我希望有人可以帮助我。

问题： 我使用 OpenAL 录制音频，出于调试原因，我将其输出到 C:/out.wav，然后我可以使用选择的任何音频播放器播放它，并播放我录制的任何内容。

我从 openAL 获得的缓冲区与我输入 libvorbisenc 的缓冲区完全相同
（我使用 vorbis_analysis_buffer 请求一个缓冲区并在其上运行 alcCaptureSamples，然后让 vorbis 完成它的工作。）

重点是：为什么 vorbis 将静音返回到我的输出文件以及如何在我的文件中获取有效的压缩音频“C:/out.ogg”？

不要担心一些缺失或多余的括号，它们在复制+粘贴和删除注释中丢失了
代码运行，但它的输出无效。

相关定义等：

//due to strange formatting constraints of this site the's are omitted  
define CHANNELS                 1  
define HERTZ                22050  
define BITSPERSAMPLE                16  
define BYTESPERSAMPLE               2  
define SAMPLES              4410  
define  SAMPLESIZE          2  
define ALIGN                (CHANNELS*BITSPERSAMPLE/8)  
define BUFFERSIZE           (SAMPLES*SAMPLESIZE)

typedef struct
{
    char            szRIFF[4];  
    long            lRIFFSize;  
    char            szWave[4];  
    char            szFmt[4];  
    long            lFmtSize;  
    WAVEFORMATEX    wfex;  
    char            szData[4];  
    long            lDataSize;  
} WAVEHEADER;  

class vorbispacker{  
public:  
    vorbispacker();  
    ~vorbispacker();  
    void consume();  

    muxer * mux;  
        // the needed ogg_stream_state is found at this->mux->state;  


    ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */  
    ogg_packet       op; /* one raw packet of data for decode */  

    vorbis_info      vi; /* struct that stores all the static vorbis bitstream  
                         settings */  
    vorbis_comment   vc; /* struct that stores all the user comments */   

    vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */  
    vorbis_block     vb; /* local working space for packet->PCM decode */  
};  

vorbis setup

void vorbispacker::setup(){

    this->mux = new muxer;

    vorbis_info_init(&this->vi);
    int ret = vorbis_encode_init_vbr(&this->vi,CHANNELS,HERTZ,0.7);
    vorbis_comment_init(&this->vc);
    vorbis_comment_add_tag(&this->vc,"ENCODER","Test");
    ret =vorbis_analysis_init(&this->vd,&this->vi);
    ret= vorbis_block_init(&this->vd,&this->vb);

    iSize=0;
    iDataSize=0;
    //TEST//

    out = fopen("C:/out.wav","wb");
    // Prepare a WAVE file header for the captured data
    sprintf(swaveheader.szRIFF, "RIFF");
    swaveheader.lRIFFSize = 0;
    sprintf(swaveheader.szWave, "WAVE");
    sprintf(swaveheader.szFmt, "fmt ");
    swaveheader.lFmtSize = sizeof(WAVEFORMATEX);        
    swaveheader.wfex.nChannels = CHANNELS;
    swaveheader.wfex.wBitsPerSample = BITSPERSAMPLE;
    swaveheader.wfex.wFormatTag = WAVE_FORMAT_PCM;
    swaveheader.wfex.nSamplesPerSec = HERTZ;
    swaveheader.wfex.nBlockAlign = swaveheader.wfex.nChannels * swaveheader.wfex.wBitsPerSample / 8;
    swaveheader.wfex.nAvgBytesPerSec = swaveheader.wfex.nSamplesPerSec * swaveheader.wfex.nBlockAlign;
    swaveheader.wfex.cbSize = 0;
    sprintf(swaveheader.szData, "data");
    swaveheader.lDataSize = 0;

    fwrite(&swaveheader, sizeof(WAVEHEADER), 1, out);




    srand(time(NULL));
    ret = ogg_stream_init(&this->mux->state,rand());


    this->eos=0;

    ogg_packet header;
    ogg_packet header_comm;
    ogg_packet header_code;



    vorbis_analysis_headerout(&this->vd,&this->vc,&header,&header_comm,&header_code);
    ret = ogg_stream_packetin(&this->mux->state,&header); 
    ret =ogg_stream_packetin(&this->mux->state,&header_comm);
    ret =ogg_stream_packetin(&this->mux->state,&header_code);



    while(1){
        int res = ogg_stream_flush(&this->mux->state,&this->og);
        if(!res)break;
        this->mux->write(this->og);

    }


      // this code works great, the headers are correct , and are output to out.ogg


};

有问题的代码：

* // set up buffer*

float ** vorbisbuffer =  vorbis_analysis_buffer(&this->vd,SAMPLES);  
*// retrieve audio samples ( MONO 16 bit)*  
alcCaptureSamples(mic->pdevice,vorbisbuffer[0],SAMPLES);    


    //this goes to debug .wav file -> the exact same buffer that goes into vorbis  
    fwrite(vorbisbuffer[0], BUFFERSIZE, 1, out);  
    iDataSize +=BUFFERSIZE;  

    int eos =0;  




      /* tell the library how much we actually submitted */    

    //SAMPLES is what is inserted into openAL , hence what we put in vorbis  
      vorbis_analysis_wrote(&vd,SAMPLES);    




    while(vorbis_analysis_blockout(&this->vd,&this->vb)==1){

      /* analysis, assume we want to use bitrate management */
      vorbis_analysis(&this->vb,NULL);
      vorbis_bitrate_addblock(&this->vb);

      while(vorbis_bitrate_flushpacket(&this->vd,&this->op)){

        /* weld the packet into the bitstream */
        ogg_stream_packetin(&this->mux->state,&op);

        /* write out pages (if any) */
        while(!eos){
          //if result > 0 there are more packets available
          int result=ogg_stream_pageout(&this->mux->state,&this->og);
          if(result==0)break;
          //write ogg page to stream 
          // since that function outputs the ogg headers OK, i suppose there's no error there.
          this->mux->write(this->og);

          /* this could be set above, but for illustrative purposes, I do
             it here (to show that vorbis does know where the stream ends) */

          if(ogg_page_eos(&og))eos=1;
        }
      }
    }

需要消化很多内容，但我真的希望有人可以帮助我。
提前致谢。

Answer 1

错误在于将 16 位样本传递到浮点（32 位！！！）vorbis 缓冲区，其中我假设浮点也是 16 位（doh），

因此分配缓冲区

char * data[SAMPLES*2]; // for 16 bit mono samples

并将内容移至浮点 vorbis 缓冲区解决了整个问题

vorbisbuffer =  vorbis_analysis_buffer(&this->vd,SAMPLES);//still 4410 samples
alcCaptureSamples(mic->pdevice,data,SAMPLES); // move the data to the  "data"buffer

for(int i =0;i<SAMPLES;i++){
    // make floats out of 16 bit samples
    vorbisbuffer[0][i]=((data[i*2+1]<<8)|(0x00ff&(int)data[i*2]))/32768.f; 
}

所以对于人类来说，如下所示：

• 获取样本的右侧字节，在 32 位缓冲区中将其向左移动 8 位
• 获取样本的左侧字节，然后对其运行 AND 运算符，并将缓冲区
• 移位整个内容 16左边的位（/32768）（最低有效字节在前）

可能听起来过于复杂，但是当去隔行立体声时它非常有意义，看起来像这样（是的，我从 SDK 中偷了这个，但我根本没有当我读到它时得到它）

vorbisbuffer =  vorbis_analysis_buffer(&this->vd,SAMPLES*CHANNELS);
alcCaptureSamples(mic->pdevice,data,SAMPLES); // open al init'ed to stereo16
//where the first dimension is the audio channel and the second the sample index
for(int i =0;i<SAMPLES;i++){  
    vorbisbuffer[0][i]=((data[i*4+1]<<8)|
              (0x00ff&(int)data[i*4]))/32768.f;
    vorbisbuffer[1][i]=((data[i*4+3]<<8)|
              (0x00ff&(int)data[i*4+2]))/32768.f;  
}