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• Encode image to video using ffmpeg (sws_scale)

  26 septembre 2012, par bahar_p

  I'm trying to encode an image to video using ffmpeg library.
  I have these global params :

  //Global params
  
  AVCodec         *codec;
  
  AVCodecContext  *codecCtx;
  
  uint8_t         *output_buffer;
  
  int             output_buffer_size;

  I divided the encoding to 3 methods :
  Initialize the encoder :

  jint Java_com_camera_simpledoublewebcams2_CameraPreview_initencoder(JNIEnv* env,jobject thiz){
  
  avcodec_register_all();
  
  avcodec_init();
  
  av_register_all();
  
  
  
  int fps = 30;
  
  
  
  /* find the H263 video encoder */
  
  codec = avcodec_find_encoder(CODEC_ID_H263);
  
  if (!codec) {
  
      LOGI("avcodec_find_encoder() run fail.");
  
      return -5;
  
  }
  
  
  
  //allocate context
  
  codecCtx = avcodec_alloc_context();
  
  
  
  /* put sample parameters */
  
  codecCtx->bit_rate = 400000;
  
  /* resolution must be a multiple of two */
  
  codecCtx->width = 176;
  
  codecCtx->height = 144;
  
  /* frames per second */
  
  codecCtx->time_base = (AVRational){1,fps};
  
  codecCtx->pix_fmt = PIX_FMT_YUV420P;
  
  codecCtx->codec_id = CODEC_ID_H263;
  
  codecCtx->codec_type = AVMEDIA_TYPE_VIDEO;
  
  
  
  /* open it */
  
  if (avcodec_open(codecCtx, codec) < 0) {
  
      LOGI("avcodec_open() run fail.");
  
      return -10;
  
  }
  
  
  
  //init buffer
  
  output_buffer_size = 500000;
  
  output_buffer = malloc(output_buffer_size);
  
  
  
  return 0;

  }

  Encoding the image :

  jint Java_com_camera_simpledoublewebcams2_CameraPreview_encodejpeg(JNIEnv* env,jobject thiz,jchar* cImage, jint imageSize){
  
  int             out_size;
  
  AVFrame         *picture;
  
  AVFrame         *outpic;
  
  uint8_t         *outbuffer;
  
  
  
  //allocate frame    
  
  picture = avcodec_alloc_frame();    
  
  outpic = avcodec_alloc_frame();
  
  
  
  int nbytes = avpicture_get_size(PIX_FMT_YUV420P, codecCtx->width, codecCtx->height);
  
  outbuffer = (uint8_t*)av_malloc(nbytes);
  
  outpic->pts = 0;
  
  
  
  //fill picture with image
  
  avpicture_fill((AVPicture*)picture, (uint8_t*)cImage, PIX_FMT_RGBA, codecCtx->width, codecCtx->height);
  
  //fill outpic with empty image
  
  avpicture_fill((AVPicture*)outpic, outbuffer, PIX_FMT_YUV420P, codecCtx->width, codecCtx->height);
  
  
  
  //rescale the image 
  
  struct SwsContext* fooContext = sws_getContext(codecCtx->width, codecCtx->height, 
  
                                                         PIX_FMT_RGBA, 
  
                                                         codecCtx->width, codecCtx->height, 
  
                                                         PIX_FMT_YUV420P, 
  
                                                         SWS_FAST_BILINEAR, NULL, NULL, NULL);
  
  sws_scale(fooContext, picture->data, picture->linesize, 0, codecCtx->height, outpic->data, outpic->linesize);   
  
  
  
  //encode the image
  
  out_size = avcodec_encode_video(codecCtx, output_buffer, output_buffer_size, outpic);
  
  out_size += avcodec_encode_video(codecCtx, output_buffer, output_buffer_size, outpic);
  
  
  
  //release pictures
  
  av_free(outbuffer);
  
  av_free(picture);
  
  av_free(outpic);
  
  
  
  return out_size;

  }

  And closing the encoder :

  void Java_com_camera_simpledoublewebcams2_CameraPreview_closeencoder(JNIEnv* env,jobject thiz){
  
  free(output_buffer);
  
  avcodec_close(codecCtx);
  
  av_free(codecCtx);

  }

  When I send the first image, I get a result from the encoder. When I try to send another image the program crashes.
  I tried calling init once and then the images, then the close - didn't work.
  I tried calling the init and the close for every image - didn't work.

  Any suggestions ?

  Thanks !

  EDIT : After further research I found that the problem is at sws_scale method.
  Still don't know what is causing this issue...

• using pocketsphinx_continuous with a .wav file

  3 avril 2013, par user2242131

  I am attempting to write an application that will allow a user to speak a small set of commands from a remote system and have them executed on my server. Using pocketsphinx to parse the spoken text. When run locally with the microphone, pocketsphinx_continuous works perfectly no matter how I slur the words. But when importing the audio file and using ffmpeg to downsample the audio to a single channel, 16 bit PCM file, it will parse the first word without difficulty. Then it will skip everything else and treat it as . I am confident that the problem is in the file format and not in the pocketsphinx configuration.

  Using command line
  ffmpeg -y -i Sound\AddSheet.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav
  in a batch file.

  The bottom of the output I get is :

  INFO: fsg_search.c(1407): Start node ADD.0:5:47
  
  INFO: fsg_search.c(1407): Start node <sil>.0:2:49
  
  INFO: fsg_search.c(1446): End node <sil>.126:128:305 (-486)
  
  INFO: fsg_search.c(1662): lattice start node <s>.0 end node <sil>.126
  
  INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:126:305) = -175371
  
  INFO: ps_lattice.c(1390): Joint P(O,S) = -176076 P(S|O) = -705
  
  000000000: ADD USER
  
  </sil></sil></s></sil></sil>

  Which is not the audio in the file. The words spoken in the file are "ADD SPREADSHEET", which works perfectly from the same microphone without the intervening .wav file.

  I have tried increasing the audio volume and decreasing the background noise using sox :

  sox -v 3.0 Sound\%1 Sound\%1-loud.wav ffmpeg -i Sound\%1-loud.wav -vn -ss 00:00:00 -t 00:00:01 -y Sound\%1-noiseaud.wav 
  
  sox Sound\%1-noiseaud.wav -n noiseprof Sound\%1-noise.prof 
  
  sox Sound\%1 Sound\%1-clean.wav noisered sound\noise.prof 0.21 
  
  ffmpeg -y -i Sound\%1-clean.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav

  with no noticeable effect on the final results.

  If you look at the output you will notice that fsg_search.c has found ADD as the start node, then silence for the remainder. Please help on this.

• AAC's converted from CAFs for HTTP Live Streaming

  13 mars 2014, par user2901994

  I have several AAC files that were converted from CAF files, for use in HTTP Live Streaming. The stream works, however there is a small gap between each AAC file. It is my understanding that this gap is caused by the "Priming" and "Remainder" frames that are attached to AAC files when they are transcoded from CAFs.

  My question is, is there any way to remove this gap ? Or use FFMpeg to wrap the files, (possibly in m4a ?) so that audio players (VLC, JWPlayer) will understand to skip the gap ?