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• Creating video with multiple audios and one still image with ffmpeg [on hold]

  4 mai 2014, par user3601509

  Hi i want to create many videos with multiples audios and one single picture.

  For example i have song1.mp3, song2.mp3, song3.mp3 and one single picture. What i need to do is to create many videos and name that videos as : song1.mp4, song2.mp4, song3.mp4

  I want to create many videos at once. Because i have a huge amount of songs and i need a lot of videos with that songs, sorry for my english.

  I have this script for only one picture and one mp3, but i need to create bulk videos can you help me ? ffmpeg -loop 1 -shortest -r 0.1 -i online.mp3 -i imagen.png -ab 128k output.avi

  Many thanks

• how do i create a stereo mp3 file with latest version of ffmpeg ?

  17 juin 2016, par Sean

  I’m updating my code from the older version of ffmpeg (53) to the newer (54/55). Code that did work has now been deprecated or removed so i’m having problems updating it.

  Previously I could create a stereo MP3 file using a sample format called :

  SAMPLE_FMT_S16

  That matched up perfectly with my source stream. This has now been replace with

  AV_SAMPLE_FMT_S16

  Which works fine for mono recordings but when I try to create a stereo MP3 file it bugs out at avcodec_open2 with :

  "Specified sample_fmt is not supported."

  Through trial and error I’ve found that using

  AV_SAMPLE_FMT_S16P

  ...is accepted by avcodec_open2 but when I get through and create the MP3 file the sound is very distorted - it sounds about 2 octaves lower than usual with a massive hum in the background - here’s an example recording :

  http://hosting.ispyconnect.com/example.mp3

  I’ve been told by the ffmpeg guys that this is because I now need to manually deinterleave my byte stream before calling :

  avcodec_fill_audio_frame

  How do I do that ? I’ve tried using the swrescale library without success and i’ve tried manually feeding in L/R data into avcodec_fill_audio_frame but the results i’m getting are sounding exactly the same as without interleaving.

  Here is my code for encoding :

  void add_audio_sample( AudioWriterPrivateData^ data, BYTE* soundBuffer, int soundBufferSize)
  
  {
  
      libffmpeg::AVCodecContext* c = data->AudioStream->codec;
  
      memcpy(data->AudioBuffer + data->AudioBufferSizeCurrent,  soundBuffer, soundBufferSize);
  
      data->AudioBufferSizeCurrent += soundBufferSize;
  
      uint8_t* pSoundBuffer = (uint8_t *)data->AudioBuffer;
  
      DWORD nCurrentSize    = data->AudioBufferSizeCurrent;
  
  
  
      libffmpeg::AVFrame *frame;
  
  
  
      int got_packet;
  
      int ret;
  
      int size = libffmpeg::av_samples_get_buffer_size(NULL, c->channels,
  
                                                data->AudioInputSampleSize,
  
                                                c->sample_fmt, 1);
  
  
  
      while( nCurrentSize >= size)    {
  
  
  
          frame=libffmpeg::avcodec_alloc_frame();
  
          libffmpeg::avcodec_get_frame_defaults(frame);
  
  
  
          frame->nb_samples = data->AudioInputSampleSize;
  
  
  
          ret = libffmpeg::avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, pSoundBuffer, size, 1);
  
          if (ret<0)
  
          {
  
              throw gcnew System::IO::IOException("error filling audio");
  
          }
  
          //audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
  
  
  
          libffmpeg::AVPacket pkt = { 0 };
  
          libffmpeg::av_init_packet(&pkt);
  
  
  
          ret = libffmpeg::avcodec_encode_audio2(c, &pkt, frame, &got_packet);
  
  
  
          if (ret<0)
  
                  throw gcnew System::IO::IOException("error encoding audio");
  
          if (got_packet) {
  
              pkt.stream_index = data->AudioStream->index;
  
  
  
              if (pkt.pts != AV_NOPTS_VALUE)
  
                  pkt.pts = libffmpeg::av_rescale_q(pkt.pts, c->time_base, c->time_base);
  
              if (pkt.duration > 0)
  
                  pkt.duration = av_rescale_q(pkt.duration, c->time_base, c->time_base);
  
  
  
              pkt.flags |= AV_PKT_FLAG_KEY;
  
  
  
              if (libffmpeg::av_interleaved_write_frame(data->FormatContext, &pkt) != 0)
  
                      throw gcnew System::IO::IOException("unable to write audio frame.");
  
  
  
  
  
          }
  
          nCurrentSize -= size;  
  
          pSoundBuffer += size;   
  
      }
  
      memcpy(data->AudioBuffer, data->AudioBuffer + data->AudioBufferSizeCurrent - nCurrentSize, nCurrentSize);
  
      data->AudioBufferSizeCurrent = nCurrentSize; 
  
  
  
  }

  Would love to hear any ideas - I’ve been trying to get this working for 3 days now :(

• Building FFmpeg for Android

  13 juin 2013, par varevarao

  I've spent almost a week on this now, trying to get FFmpeg "Angel"/"Happiness" to build for Android.
  I've tried build scripts from all over the internet to no avail. I got closest was using this. As the author himself says the script doesn't work for newer versions of FFmpeg due to this bug, which has been dismissed on that ticket saying "I found a Makefile that does it." This was dis-heartening, being the only post on all of the vast Google world that was anywhere close to my problem.
  So, question time :

  Is there a way to get around the above bug ? I'm trying to use the newest ffmpeg API, and "Love" is just giving me "undefined reference" errors while trying to use av_encode_video2(), and av_free_frame(). The code I was working on the lines of is at the ffmpeg git repo, under /doc/examples/decoding_encoding.c (the function starting on line 338).

  Update : So they've done away with codec_names.sh in "Angel". Sorry didn't notice that before, but the problem persists in a different avatar now. With every build attempt the compiler throws a certain

  start ndk-building...
  
  /home/<user>/android-ndk/build/core/build-binary.mk:41: *** target file `clean&#39; has both : and :: entries.  Stop.
  
  </user>

  Say whatnow !?