最佳实践:在哪里重新采样PCM和哪种工具？

Question

我开发了一个内核模块(Android),它为我提供:

PCM
16-bit
48000 Hz
2 channel

我想把它流式传输到Java的Apple机场快线(AEX).

AEX需要44.1 kHz PCM,因此我必须重新采样PCM流.

我有以下可能性,但哪个最好？

1.使用C程序"raop_play"(raop-play的一部分)

advantages: 
            high-performant due to native C
            already uses libsamplerate to resample wav, mp3, ogg, flac, aac, pls
            openssl as static library
            usable via command-line from my java-program via Runtime.exec()

disadvantages:
            I am relative new to C
            overloaded: I don't need wav, mp3.. only PCM
            many dependencies with GPL-libraries which I have to compile for Android
            only supports PCM already with 44.1 kHz, no resampling for PCM implemented yet 
            -> have to implement resampling for PCM

2.在java中重新采样和流(使用libresample JNI-bridge)

advantages: 
            I CAN java :)
            middle-performant due to resamling in C , but streaming in java
            just one dependency to LGPL-library
            no Runtime.exec() needed

disadvantages:
            needs [bouncycastle][3] for AES which is a bit larger than openssl
            less performant than solution #1 (but maybe fast enough)

3.已在内核模块中重新采样

advantages: 
            most performant
            no resampling at higher level

disadvantages:
            I am relative new to C
            Is it possible to use libsamplerate or libresample in kernel-space?!

Answer 1

我是一个内心的java人,但是这个任务(尤其是在cpu受限设备上,比如手持设备)正在为C而哭.我建议只使用libsamplerate.它有一个简单的API,即使你是C的新手,你也可以通过Google搜索找到很多例子.

当然基于java的解决方案可以并且可以工作,它只是因为你的新C到用户吃电池似乎不礼貌:)

编辑:我可能会自相矛盾了一点,但即使性能是一个严重的问题,我会避免做任何事情在内核空间,除非我知道内核和硬件确实很好.鉴于此,我将使用链接到libsamplerate的用户空间程序.经过一段谷歌搜索后,我发现了这个例子(注意输出是插孔界面,显然它必须与你不同)

#include <jack/jack.h>
#include <samplerate.h>

int channels;
float data_samplerate;


/////////////////////////////////////////////////////
/////////////////////////////////////////////////////
void getDasData(float **dst,int num_frames){
/* Provide sound data here, and only here. */
}
/////////////////////////////////////////////////////
/////////////////////////////////////////////////////



long getDasResampledData_callback(void *cb_data, float **data){
  static float ret[1024];
  static float ret3[1024];
  static float *ret2[2]={&ret[0],&ret[512]};
  getDasData(ret2,512);
  for(int i=0;i<512;i++){
    ret3[i*2]=ret2[0][i];
    ret3[i*2+1]=ret2[1][i];
  }
  *data=&ret3[0];
  return 512;
}

void getDasResampledData(float **dst,int num_frames){
  double ratio=samplerate/getSourceRate();
  float outsound[num_frames*2];
  long read=src_callback_read(dassrc_state,ratio,num_frames,outsound);
  //fprintf(stderr,"read: %d, num_frames: %d\n",read,num_frames);
  for(int i=0;i<read;i++){
      dst[0][i]=outsound[i*2];
      dst[1][i]=outsound[i*2+1];
  }
  if(read<num_frames){
    float *newdst[2]={dst[0]+read,dst[1]+read};
    getDasResampledData(newdst,num_frames-read);
  }
}


static int process (jack_nframes_t nframes, void *arg){
  int ch;
  sample_t *out[channels];

  for(ch=0;ch<channels;ch++){
    out[ch]=(sample_t*)jack_port_get_buffer(ports[ch],nframes);
  }

  if( (fabs(data_samplerate - jack_samplerate)) > 0.1)
    getDasResampledData(out,numSamples);
  else
    getDasData(outputChannelData,numSamples);
  return;

  audioCallback(NULL,0,out,channels,nframes);
}

int main(){
  dassrc_state=src_callback_new(getDasResampledData_callback,SRC_QUALITY,2,NULL,NULL);
  jack_set_process_callback(client, process,NULL);
}

来自http://old.nabble.com/Example-of-using-libresample-with-jack-td8795847.html

这个例子看起来很简单,我希望你能用它.