#include <stdio.h>
#include <assert.h>
#include <stdlib.h>
#include <string.h>

#include "au_channel.h"
#include "aacplus.h"

int main(int argc, char *argv[])
{

  WavInfo inputInfo;
  FILE *inputFile = NULL;
  FILE *hADTSFile;

  int  error;
  int  bEncodeMono = 0;
  int frmCnt = 0;

   /*
   * parse command line arguments
   */
  if (argc != 5) {
    fprintf(stderr, "\nUsage:   %s <wav_file> <bitstream_file> <bitrate> <(m)ono/(s)tereo>\n", argv[0]);
    fprintf(stderr, "\nExample: %s input.wav out.aac 24000 s\n", argv[0]);
    return 0;
  }

  if ( strcmp (argv[4],"m") == 0 ) {
    bEncodeMono = 1;
  }
  else {
    if ( strcmp (argv[4],"s") != 0 ) {
      fprintf(stderr, "\nWrong mode %s, use either (m)ono or (s)tereo\n", argv[4]);
      return 0;
    }
  }
  fflush(stdout);

  inputFile = AuChannelOpen (argv[1], &inputInfo);

  if(inputFile == NULL){
    fprintf(stderr,"could not open %s\n",argv[1]);
    exit(10);
  }

  if (inputInfo.nChannels==1 && !bEncodeMono) {
	  fprintf(stderr,"Need stereo input for stereo coding mode !\n");
	  exit(10);
  }

  if (strcmp(argv[2],"-")==0)
   hADTSFile=stdout;
  else
   hADTSFile = fopen(argv[2], "wb");

  if(!hADTSFile) {
   fprintf(stderr, "\nFailed to create ADTS file\n") ;
      exit(10);
    }

  /*
    Be verbose
   */
  unsigned long inputSamples=0;
  unsigned long maxOutputBytes=0;
  aacplusEncHandle hEncoder = aacplusEncOpen(inputInfo.sampleRate,
		  	  inputInfo.nChannels,
		  	  &inputSamples,
		  	  &maxOutputBytes);

  aacplusEncConfiguration *cfg = aacplusEncGetCurrentConfiguration(hEncoder);
  cfg->bitRate = atoi(argv[3]);
  cfg->bandWidth = 0;
  cfg->outputFormat = 1;
  cfg->nChannelsOut = bEncodeMono ? 1 : inputInfo.nChannels;
  if(inputInfo.aFmt == WAV_FORMAT_FLOAT){
    cfg->inputFormat = AACPLUS_INPUT_FLOAT;
  }
	
  fprintf(stdout,"input file %s: \nsr = %d, nc = %d fmt = %d\n\n",
          argv[1], inputInfo.sampleRate, inputInfo.nChannels, inputInfo.aFmt);
  fprintf(stdout,"output file %s: \nbr = %d inputSamples = %lu  maxOutputBytes = %lu nc = %d m = %d\n\n",
            argv[2], cfg->bitRate, inputSamples, maxOutputBytes, cfg->nChannelsOut, bEncodeMono);
  fflush(stdout);

  int ret = 0;
  if((ret = aacplusEncSetConfiguration(hEncoder, cfg)) == 0) {
      fprintf(stdout,"setting cfg failed\n", ret);
      return -1;
  }

  uint8_t *outputBuffer = malloc(maxOutputBytes);
  int32_t *TimeDataPcm;
  if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
    TimeDataPcm = calloc(inputSamples, sizeof(float));
  } else {
    TimeDataPcm = calloc(inputSamples, sizeof(short));
  }

  int stopLoop = 0;
  int bytes = 0;
  do {
      int numSamplesRead = 0;
      if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
          if ( AuChannelReadFloat(inputFile, (float *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
                  stopLoop = 1;
                  break;
          }
      } else {
          if ( AuChannelReadShort(inputFile, (short *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
                  stopLoop = 1;
                  break;
          }
      }

      if(numSamplesRead < inputSamples) {
          stopLoop = 1;
          break;
      }

      bytes = aacplusEncEncode(hEncoder, (int32_t *) TimeDataPcm, numSamplesRead,
              outputBuffer,
              maxOutputBytes);

      if(bytes > 0) fwrite(outputBuffer, bytes, 1, hADTSFile);

      frmCnt++;
      fprintf(stderr,"[%d]\r",frmCnt); fflush(stderr);
  } while (!stopLoop && bytes >= 0);

  fprintf(stderr,"\n");
  fflush(stderr);

  printf("\nencoding finished\n");
  aacplusEncClose(hEncoder);
  fclose(hADTSFile);
  free(outputBuffer);
  free(TimeDataPcm);
  return 0;
}