java - 最佳实践 : Where to resample PCM and which tool?

Question

我开发了一个内核模块 (Android)，它为我提供:

PCM
16-bit
48000 Hz
2 channel

我想用 java 将其流式传输到 Apple 的机场快线 (AEX)。

AEX 需要 44.1 kHz PCM，所以我必须重新采样 PCM 流。

我有以下几种可能性，但哪种是最好的？

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 中重新采样和流式传输(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

我本质上是一个 java 人，但是这个任务(尤其是在 cpu 受限的设备上，比如手持设备上)需要 C。我建议简单地使用 libsamplerate。它有一个简单的 API，即使您是 C 的新手，您也可以通过谷歌搜索找到大量示例。

当然，基于 java 的解决方案可以而且会起作用，只是因为您是 C 的新手而耗尽电池对用户来说似乎并不礼貌 :)

编辑:我可能有点自相矛盾，但即使性能是一个严重的问题，我也会避免在内核空间中做任何，除非我真正非常了解内核和硬件.鉴于此，我将使用链接到 libsamplerate 的用户空间程序。经过一番谷歌搜索后，我找到了这个示例(请注意，输出是插孔接口(interface)，显然它对您来说必须不同)

#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);
}

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

这个例子看起来很简单，希望你能使用它。