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• Extracting all frames of a video using ffmpeg

  4 décembre 2013, par Gonzalo Solera

  I'm trying to extract all the frames of a video using ffmpeg compiled statically for android. I want to extract all the frames with a lower quality (at jpg format) and then, select the specific frames I want with a higher resolution and extract them at png format.
  But the problem is that I need to know exactly at what time is placed the selected frame in order to be able to extract te same frame at higher resolution. When I calcule how much time are between frames (duration_of_the_video/total_frames_extracted) and I multiple it by the number of the frame I want to extract again, the result time isn't the exact time of the frame.

  How could I extract a specific number of frames of a video using ffmpeg ? I'm trying to extract all the frames but sometimes not all the frames are extracted. For example, I have a 1600 ms long video but when I use this command :

  ffmpeg -i file.mp4 -y %d.jpg

  it doesn't extract all the frames because it only extracts 45 frames but the frame rate is 30 fps (1600/45 = 35.5555 and 1000/30 = 33.33333).

  So, in order to be able to calculate at what time is placed the frame I want, I would need to extract ALL the frames of the video or extract a fixed number of frames (it doesn't matter if some frames are repeated if I can get the time of them).

  This is the output when I try to extract all the frames (there should be 48 but there are 45 instead so I can't calculate the exact time...)
  

  I'm not sure If I have explained correctly but I will appreciate any help. Thanks !

• Slideshow with ffmpeg

  15 août 2017, par Luca

  I want to make a video from many images. I tried using the concat demuxer as explained here but :

  1) I don’t know what the -vsync vfr parameter do and how to use it.

  2) Using the input.txt file I tell explicitly the duration of each image of the slideshow so I guess the input framerate (-framerate) is useless. Can I set the output framerate however with the -r or the -vf fps parameter ?

  3) The images are both .jpg and .png with different resolutions, dimensions etc... Do ffmpeg and the concat demuxer handle this ? Should I optimize something ?

• ffmpeg ; opus encoded sound in webm does not work with ffplay or YouTube, only VLC [on hold]

  2 août 2017, par Mockarutan

  I’m having trouble getting Opus encoded sound in the webm container to work. I’m using libopus in ffmpeg.

  The file does work in VLC. But not in ffplay or on YouTube. If I take the raw wav data in a wav file and then convert it to Opus/webm with the ffmpeg.exe that comes pre-compiled. It works in VLC, ffplay and YouTube.

  So ffmpeg can obviously do it correctly, I must be doing something wrong in my code.

  The file my code produces : https://drive.google.com/file/d/0B16rIXjPXJCqcU5HVllIYW1iODg/view?usp=sharing

  Edit, More details that I forgot in my frustration : The file can be opened by ffplay and uploaded to youtube (when I interlace it with VP9 video). But the sound is just "ticks", example : https://www.youtube.com/watch?v=j_ShBbuizeo&feature=youtu.be

  I have read though all example codes that I know of from ffmpeg, but all of them is in the old API, not the send/receive api, so a big part of the code does not apply anymore. This code works with all other Codes I’ve tested, including H.264+AAC in mp4, VP8+Opus in ogg and raw PCM F32LE in wav. I would have gone with VP8+Opus in ogg if the license was as straight forward as the webm license

  I’ve looked though the source for the ffmpeg.exe command line tool and coped everything applicable in to my code base.

  (Edit 3, reduced the code as much as I can)
  Here is my code : https://pastebin.com/HTuc0g8K

  Setup :

  int initialize(int sample_rate, int per_frame_audio_samples, int audio_bitrate, const char *filename)
  
      {
  
          int ret;
  
  
  
          avcodec_register_all();
  
          av_register_all();
  
  
  
          ret = avformat_alloc_output_context2(&outctx, NULL, "webm", filename);
  
  
  
          if (ret < 0)
  
              return ret;
  
  
  
          aud_codec = avcodec_find_encoder(aud_codec_id);
  
          avcodec_register(aud_codec);
  
  
  
          if (!aud_codec)
  
              return -1;
  
  
  
          // Setup Audio Stream
  
  
  
          aud_codec_context = avcodec_alloc_context3(aud_codec);
  
          if (!aud_codec_context)
  
              return -1;
  
  
  
          /* select other audio parameters supported by the encoder */
  
          aud_codec_context->bit_rate = audio_bitrate;
  
          aud_codec_context->sample_rate = sample_rate;
  
          aud_codec_context->sample_fmt = sample_fmt;
  
          aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
  
          aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
  
  
  
          aud_codec_context->codec = aud_codec;
  
          aud_codec_context->codec_id = aud_codec_id;
  
  
  
  
  
          AVRational time_base;
  
          time_base.num = per_frame_audio_samples;
  
          time_base.den = aud_codec_context->sample_rate;
  
          aud_codec_context->time_base = time_base;
  
  
  
          ret = avcodec_open2(aud_codec_context, aud_codec, NULL);
  
  
  
          if (ret < 0)
  
              return ret;
  
  
  
          outctx->audio_codec = aud_codec;
  
          outctx->audio_codec_id = aud_codec_id;
  
  
  
          audio_st = avformat_new_stream(outctx, aud_codec);
  
  
  
          avcodec_parameters_from_context(audio_st->codecpar, aud_codec_context);
  
  
  
          conv_time_base.num = aud_codec_context->frame_size;
  
          conv_time_base.den = aud_codec_context->sample_rate;
  
  
  
          // Setup audio frame
  
          aud_frame = av_frame_alloc();
  
          aud_frame->nb_samples = aud_codec_context->frame_size;
  
          aud_frame->format = aud_codec_context->sample_fmt;
  
          aud_frame->channel_layout = aud_codec_context->channel_layout;
  
          aud_frame->sample_rate = aud_codec_context->sample_rate;
  
  
  
          int buffer_size;
  
          if (aud_codec_context->frame_size == 0)
  
          {
  
              buffer_size = per_frame_audio_samples * 2 * 4;
  
              aud_frame->nb_samples = per_frame_audio_samples;
  
          }
  
          else
  
          {
  
              buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
  
                  aud_codec_context->sample_fmt, 0);
  
          }
  
  
  
          if (av_sample_fmt_is_planar(sample_fmt))
  
              ret = av_frame_get_buffer(aud_frame, buffer_size / 2);
  
          else
  
              ret = av_frame_get_buffer(aud_frame, buffer_size);
  
  
  
          if (!aud_frame || ret < 0)
  
              return ret;
  
  
  
          // Setup audio resampler
  
  
  
          audio_swr_ctx = swr_alloc();
  
          if (!audio_swr_ctx)
  
              return -1;
  
  
  
          /* set options */
  
          av_opt_set_int(audio_swr_ctx, "in_channel_layout", aud_codec_context->channel_layout, 0);
  
          av_opt_set_int(audio_swr_ctx, "in_sample_rate", sample_rate, 0);
  
          av_opt_set_int(audio_swr_ctx, "in_frame_size", per_frame_audio_samples, 0);
  
          av_opt_set_sample_fmt(audio_swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  
  
          av_opt_set_int(audio_swr_ctx, "out_channel_layout", aud_codec_context->channel_layout, 0);
  
          av_opt_set_int(audio_swr_ctx, "out_sample_rate", aud_codec_context->sample_rate, 0);
  
          av_opt_set_int(audio_swr_ctx, "out_frame_size", aud_codec_context->frame_size, 0);
  
          av_opt_set_sample_fmt(audio_swr_ctx, "out_sample_fmt", aud_codec_context->sample_fmt, 0);
  
  
  
          /* initialize the resampling context */
  
          if ((ret = swr_init(audio_swr_ctx)) < 0)
  
          {
  
              return ret;
  
          }
  
  
  
          dst_rate = aud_codec_context->sample_rate;
  
          src_rate = sample_rate;
  
  
  
          src_nb_samples = per_frame_audio_samples;
  
          dst_nb_samples = aud_codec_context->frame_size;
  
  
  
          max_dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  
  
  
          dst_nb_channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
  
  
  
          ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, dst_nb_samples, sample_fmt, 0);
  
  
  
          aud_frame_counter = 0;
  
  
  
          if (ret < 0)
  
              return ret;
  
  
  
          av_dump_format(outctx, 0, filename, 1);
  
  
  
          if (!(outctx->oformat->flags & AVFMT_NOFILE))
  
          {
  
              ret = avio_open(&outctx->pb, filename, AVIO_FLAG_WRITE);
  
              if (ret < 0)
  
              {
  
                  return ret;
  
              }
  
          }
  
  
  
          ret = avformat_write_header(outctx, NULL);
  
          if (ret < 0)
  
              return ret;
  
  
  
          return 0;
  
      }

  Encoding and ending :

  int process_encode_loop(AVFormatContext *local_outctx, AVCodecContext *codec_context, AVStream *stream, AVRational time_base, bool flush)
  
      {
  
          int ret;
  
  
  
          AVPacket pkt;
  
          av_init_packet(&pkt);
  
          pkt.data = NULL;
  
          pkt.size = 0;
  
  
  
          while (true)
  
          {
  
              ret = avcodec_receive_packet(codec_context, &pkt);
  
              if (!ret)
  
              {
  
                  pkt.stream_index = stream->index;
  
                  av_packet_rescale_ts(&pkt, time_base, stream->time_base);
  
                  av_interleaved_write_frame(local_outctx, &pkt);
  
  
  
                  av_packet_unref(&pkt);
  
              }
  
  
  
              if (ret == AVERROR(EAGAIN))
  
                  break;
  
              else if (ret == AVERROR_EOF)
  
                  break;
  
              else if (ret < 0)
  
                  return ret;
  
              else if (flush == false)
  
                  break;
  
          }
  
  
  
          return 0;
  
      }
  
  
  
      int write_audio_frame(float_t *aud_sample)
  
      {
  
          int ret;
  
          if (dst_nb_samples > max_dst_nb_samples)
  
          {
  
              av_free(&aud_frame->data[0]);
  
              ret = av_samples_alloc(aud_frame->data, &dst_linesize, dst_nb_channels, dst_nb_samples, sample_fmt, 1);
  
              if (ret < 0)
  
                  return ret;
  
  
  
              max_dst_nb_samples = dst_nb_samples;
  
          }
  
  
  
          ret = swr_convert(audio_swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)&aud_sample, src_nb_samples);
  
          if (ret < 0)
  
          {
  
              return ret;
  
          }
  
  
  
          aud_frame->data[0] = (uint8_t*)dst_data[0];
  
          aud_frame->extended_data[0] = (uint8_t*)dst_data[0];
  
  
  
          aud_frame->pts = aud_frame_counter++;
  
  
  
          ret = avcodec_send_frame(aud_codec_context, aud_frame);
  
  
  
          ret = process_encode_loop(outctx, aud_codec_context, audio_st, conv_time_base, false);
  
  
  
          if (ret < 0)
  
              return ret;
  
  
  
          return 0;
  
      }
  
  
  
      int finish_audio_encoding()
  
      {
  
          int ret = avcodec_send_frame(aud_codec_context, NULL);
  
          if (ret < 0)
  
              return ret;
  
  
  
          ret = process_encode_loop(outctx, aud_codec_context, audio_st, conv_time_base, true);
  
          if (ret < 0)
  
              return ret;
  
  
  
          av_write_trailer(outctx);
  
  
  
          return ret;
  
      }

  Main :

  void fill_samples(float_t *dst, int nb_samples, int nb_channels, int sample_rate, float_t *t)
  
      {
  
          int i, j;
  
          float_t tincr = 1.0 / sample_rate;
  
          const float_t c = 2 * M_PI * 440.0;
  
          /* generate sin tone with 440Hz frequency and duplicated channels */
  
          for (i = 0; i < nb_samples; i++) {
  
              *dst = sin(c * *t);
  
              for (j = 1; j < nb_channels; j++)
  
                  dst[j] = dst[0];
  
              dst += nb_channels;
  
              *t += tincr;
  
          }
  
      }
  
  
  
      int main()
  
      {
  
          int frame_rate = 30;
  
          int sec = 12;
  
          int bit_rate = 192000;
  
          float t = 0;
  
  
  
          int src_samples_linesize;
  
          int src_nb_samples = 1024;
  
          int src_channels = 2;
  
          int sample_rate = 48000;
  
  
  
          uint8_t **src_data = NULL;
  
  
  
          int ret;
  
  
  
          initialize(sample_rate, src_nb_samples, bit_rate, "sound_test.webm");
  
  
  
          ret = av_samples_alloc_array_and_samples(&src_data, &src_samples_linesize, src_channels,
  
              src_nb_samples, AV_SAMPLE_FMT_FLT, 0);
  
  
  
          for (size_t i = 0; i < frame_rate * sec; i++)
  
          {
  
              fill_samples((float *)src_data[0], src_nb_samples, src_channels, sample_rate, &t);
  
              write_audio_frame((float *)src_data[0]);
  
          }
  
          finish_audio_encoding();
  
  
  
          cleanup();
  
  
  
          return 0;
  
      }

  Edit 2, This code reproduces the issue exactly and is fully self contained, if you have the ffmpeg 3.3.x libraries. It’s tried with 3.3.1 and 3.3.2 is the same results.

  So what could I be missing ? I do not think something is wrong with the sample rates or any other specifications, else it would not work in VLC or an ogg file. I do think the audio stream itself if correct, just some part of the header or how the file is formatted (look further down for some EBML inspection) that is not correct.

  As explained earlier, the licence with VP9+Opus in webm is why I have these specifics. And the exact problem is that I want the audio stream produced to work well when I upload it to YouTube.

  Any suggestion is appreciated, thanks in Advance !

  Some other things I’ve tried :

  I’ve looked at the header with the "MediaInfo" app built in to MVKTool :
  https://i.gyazo.com/3b29b41629a28bd526bf7637ce3f2601.png
  It all looks fine to me.

  I’ve also inspected the raw EBML file with EBML-Viewer (https://code.google.com/archive/p/ebml-viewer/) and in there I can se some difference between the files ;

  My file : https://i.gyazo.com/6fa8c540a2698a8a4d3421d363aede0a.png
  File produced with ffmpeg.exe : https://i.gyazo.com/04d60e64ff3c3040ea83e98cdf507530.png

  In my file it’s "Cluster" -> "BlockGroup" -> "Block", " ?"
  In the other it’s just "Cluster" -> "SimpleBlock"
  And in the webm specs, it says both are supported (https://www.webmproject.org/docs/container/)

  But I do not know much about these specific things, just looking for anything.