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• From URL of Video GetThumbnail Using Nreco

  8 février 2016, par Muhammad Abid Fahad

  I working on a sharepoint project in which i have to upload the videos in the document library as videoset. after creating a video set i have have to upload the video and fetch the thumbnail from the video and upload it. video is uploaded succesfully using

  spfile = item.Folder.Files.Add(fuUpload.FileName, fuUpload.PostedFile.InputStream, true);

  I am using using Nreco to get thumbnail from the video. However my code works fine on local machine but its giving error "http://mysite/Download/abc/abc.mp4 : Server returned 401 Unauthorized (authorization failed) (exit code : 1)" when i am using my application from other pc browsers.

  ffMpeg.GetVideoThumbnail(videoPath, ms, 10) ; the error line.

  here is the code i am using

  private MemoryStream SaveThumbnail(string videoPath) 
  
      {
  
          MemoryStream ms; 
  
          try 
  
          {
  
              videoPath = "http://mysitehttp/Download/abc/abc.mp4"
  
              ms = new MemoryStream();
  
              SPSecurity.RunWithElevatedPrivileges(delegate() {
  
  
  
                  var ffMpeg = new NReco.VideoConverter.FFMpegConverter();
  
                  ffMpeg.GetVideoThumbnail(videoPath, ms, 10); 
  
              });
  
  
  
  
  
          }
  
          catch(Exception ex)
  
          {
  
              throw ex;
  
          }
  
  
  
          return ms;
  
      }

• FFmpeg watermark duration

  4 mai 2016, par user2364292

  I am developing an Android app, which will work with FFmpeg library which only applies multiple watermarks on some test.mp4 video. For that case I am using the Android java library for FFmpeg binary from that source : [http://writingminds.github.io/ffmpeg-android-java/] which is working fine for some examples, but I just can not use commands like these :

  -i /storage/emulated/0/Download/test.mp4 -vf "movie=/storage/emulated/0/Download/test.mp4 [logo]; [in][logo] overlay=W-w-10:H-h-10, fade=in:0:20 [out]" /storage/emulated/0/Download/test.mp4

  Please note these quotes after -vf flag and these [in], [logo] flags. These just throw me an errors like

  [NULL @ 0xb5bf3200] Unable to find a suitable output format for '[logo];'
  
  [logo];: Invalid argument

  and so on. The quotes in the command are also the problem which throws me an error like :

  [AVFilterGraph @ 0xb5c092c0] No such filter: 'overlay=main_w-50:main_h-50,overlay=0:0'
  
  Error initializing complex filters.
  
  Invalid argument".

  Am I even using proper library for that case ? I also want to show the watermark in specific duration of the video. For example if video is 10 seconds long, I only want watermark to be shown starting from 3rd second of the video to 7th second. I can show the overlay images like this :

  -i /storage/emulated/0/Download/test.mp4 -i /storage/emulated/0/Download/test.jpg -i /storage/emulated/0/Download/test.jpg -filter_complex overlay=main_w-50:main_h-50,overlay=0:0 -codec:a copy /storage/emulated/0/Download/test_edited.mp4

  but they’re shown for the whole video duration. Where can I set the duration start/end for the specific overlay ?

  Thank you very much !

• C++ FFmpeg pitch issue

  24 janvier 2016, par David Andrei Norgren

  I’m using swr_convert to lower/raise the pitch of incoming audio and store it in a .mp3. To change the pitch, I’m dividing the out sample rate by a factor. However, the resulting audio is slightly distorted when this factor is anything other than 1. Here’s my conversion code :

  ...
  
  
  
  // Set up resample context
  
  swrContext = swr_alloc();
  
  if (!swrContext)
  
      throw -15;
  
  
  
  av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
  
  av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
  
  
  
  av_opt_set_int(swrContext, "out_channel_count", STREAM_AUDIO_CHANNELS, 0);
  
  av_opt_set_int(swrContext, "out_channel_layout", STREAM_AUDIO_CHANNEL_LAYOUT, 0);
  
  av_opt_set_int(swrContext, "out_sample_rate", STREAM_AUDIO_SAMPLE_RATE / pitch, 0);
  
  av_opt_set_sample_fmt(swrContext, "out_sample_fmt", STREAM_AUDIO_SAMPLE_FORMAT_GM, 0);
  
  
  
  if (swr_init(swrContext))
  
      throw -16;
  
  
  
  // Allocate re-usable frame
  
  frameDecoded = av_frame_alloc();
  
  if (!frameDecoded)
  
      throw -17;
  
  
  
  frameDecoded->format = codecContext->sample_fmt;
  
  frameDecoded->channel_layout = codecContext->channel_layout;
  
  frameDecoded->channels = codecContext->channels;
  
  frameDecoded->sample_rate = codecContext->sample_rate;
  
  
  
  // Load frames
  
  inPacket.data = NULL;
  
  inPacket.size = 0;
  
  
  
  int gotFrame, samples = 0;
  
  
  
  while (av_read_frame(formatContext, &inPacket) >= 0) {
  
  
  
      if (inPacket.stream_index != streamId) 
  
          continue;
  
  
  
      if (avcodec_decode_audio4(codecContext, frameDecoded, &gotFrame, &inPacket) < 0)
  
          throw -18;
  
  
  
      if (!gotFrame)
  
          continue;
  
  
  
      // Begin conversion
  
      if (swr_convert(swrContext, NULL, 0, (const uint8_t **)frameDecoded->data, frameDecoded->nb_samples) < 0)
  
          throw -19;
  
  
  
      while (swr_get_out_samples(swrContext, 0) >= RAW_AUDIO_FRAME_SIZE) {
  
  
  
          // Allocate data
  
          uint8_t **convertedData = NULL;
  
          if (av_samples_alloc_array_and_samples(&convertedData, NULL, STREAM_AUDIO_CHANNELS, RAW_AUDIO_FRAME_SIZE, STREAM_AUDIO_SAMPLE_FORMAT_GM, 0) < 0)
  
              throw -20;
  
  
  
          // Convert
  
          if (swr_convert(swrContext, convertedData, RAW_AUDIO_FRAME_SIZE, NULL, 0) < 0)
  
              throw -21;
  
  
  
          // Calculate buffer size
  
          size_t bufferSize = av_samples_get_buffer_size(NULL, STREAM_AUDIO_CHANNELS, RAW_AUDIO_FRAME_SIZE, STREAM_AUDIO_SAMPLE_FORMAT_GM, 0);
  
          if (bufferSize < 0)
  
              throw -22;
  
  
  
          fwrite(convertedData[0], 1, bufferSize, outStream);
  
          av_free(convertedData);
  
      }
  
  }
  
  
  
  ...

  STREAM_AUDIO_SAMPLE_RATE is defined as 44100.
  Here’s the entire program if it helps : http://pastebin.com/5akEwNg4

  The program generates a .mp3 with 25 notes that decrease in pitch.
  Here’s an example of the distortion : http://www.stuffbydavid.com/dl/30256478.mp3

  Can you spot anything incorrect about my conversion, or is my method of changing the pitch incorrect ? Is there another way ?