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• x264 configure linking and building problems

  16 janvier 2014, par Pie

  I am trying to build x264 from source on Ubuntu 32bit in order to convert a sequence of jpg or png images into mp4 video : x264 site, sample images

  The downloaded binaries is able to convert the sequence into an mkv video (or few other formats) when I run this command :

  ./x264dist ~/Dev/x264emp/img/FLYOVER%4d.JPG -o abc.mkv

  x264dist is the renamed name of the binary I download from the site.

  However, when I grab the source and compile with simple configure :

  $ ./configure --enable-shared --enable-static --enable-pic

  platform:      X86
  
  system:        LINUX
  
  cli:           yes
  
  libx264:       internal
  
  shared:        yes
  
  static:        yes
  
  asm:           yes
  
  interlaced:    yes
  
  avs:           avxsynth
  
  lavf:          no
  
  ffms:          no
  
  mp4:           no
  
  gpl:           yes
  
  thread:        posix
  
  opencl:        yes
  
  filters:       crop select_every 
  
  debug:         no
  
  gprof:         no
  
  strip:         no
  
  PIC:           yes
  
  bit depth:     8
  
  chroma format: all

  then $ make. Then I use the binaries to run the exactly same command as above but there is this error :

  ./x264 ~/Dev/x264emp/img/FLYOVER%4d.JPG -o abc.mkv
  
  raw [error]: raw input requires a resolution.
  
  x264 [error]: could not open input file `/home/tmd/Dev/x264emp/img/FLYOVER%4d.JPG' via any method!

  It seems like it can’t read any input at all. But at least I am still able to run --help on that binaries.

  Then I realized that the downloaded binaries is 3.5Mb while my custom compilation results in 1.5Mb binaries.

  So I just want to know what are the build configurations used by the official build, and/or is there any dependency I am missing that leads to this problem.

  The reason I am trying to build myself because I want to port the x264 lib into Javascript using Emscripten. There has been a solution using FFmpeg but it seems like I don’t need the whole video processing library but only a simple H264 codec. So I need to solve the configure/compile/linking problem to port it rightly.

  Possibly similar How to configure X264 build before running make on OS X

• FFmpeg realtime volume changing with C API

  9 janvier 2019, par Tank2006

  My environment is FFmpeg 4.1(Prebuild package from an official site) on Windows 10/Visual Studio 2017.

  I want to change an audio volume as realtime according to a volume button operation from media player apps.

  Think simply, it will update if AVFilter parameters changes, but it seems doesn’t work.

  const char src[] = "C:\\sample.mp3";
  
  
  
  AVFilterGraph *graph = NULL;
  
  AVFilterContext *ctx_src, *ctx_sink;
  
  AVFilter *ctx_vol;
  
  
  
  int main()
  
  {
  
      int res;
  
      AVPacket *packet = av_packet_alloc();
  
      AVFrame *frame = av_frame_alloc();
  
      AVFrame *fframe = av_frame_alloc();
  
  
  
      AVCodec *codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      AVCodecParser *parser = av_parser_init(codec->id);
  
  
  
      AVCodecContext *cc = avcodec_alloc_context3(codec);
  
      avcodec_open2(cc, codec, NULL);
  
  
  
  
  
      FILE *fp; uint8_t buffer[1024]; int filterinit = 0;
  
      fopen_s(&fp, src, "rb");
  
      while (feof(fp) == 0) {
  
          int read = fread(buffer, 1, 1024, fp);
  
  
  
          res = av_parser_parse2(parser, cc, &packet->data, &packet->size,
  
              buffer, read, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
  
          if (packet->size) {
  
              res = avcodec_send_packet(cc, packet);
  
  
  
              while (res >= 0) {
  
                  res = avcodec_receive_frame(cc, frame);
  
  
  
                  if (res == AVERROR(EAGAIN) || res == AVERROR_EOF) {
  
                      break;
  
                  } else if (res < 0) {
  
                      fprintf(stderr, "Error during decoding\n");
  
                      exit(1);
  
                  }
  
  
  
                  if (filterinit == 0) {
  
                      // Create a volume filter, links and graph from AVCodecContext's parameter
  
                      if(init_filters(cc)) filterinit = 1;
  
                  }
  
  
  
                  if (filterinit == 1) {
  
                      res = av_buffersrc_add_frame(ctx_src, frame);
  
                      if(av_buffersink_get_frame(ctx_sink, frame) >= 0) {
  
  
  
                          // Change the volume realtime
  
                          av_opt_set(ctx_vol, "volume", AV_STRINGIFY(1.2), AV_OPT_SEARCH_CHILDREN);
  
  
  
  
  
                          int datasize = av_get_bytes_per_sample(cc->sample_fmt);
  
                          for (int i = 0; i < frame->nb_samples; i++) {
  
                              for (int ch = 0; ch < cc->channels; ch++) {
  
                                  //fwrite(frame->data[ch] + data_size * i, 1, data_size, outfile);
  
                              }
  
                          }
  
                      }
  
                  }
  
              }
  
          }
  
      }
  
  
  
      fclose(fp);
  
  
  
      avcodec_free_context(&cc);
  
      av_parser_close(parser);
  
      av_frame_free(&frame);
  
      av_frame_free(&fframe);
  
      av_packet_free(&packet);
  
  
  
      avfilter_graph_free(&graph);
  
  
  
      return 0;
  
  }

  Can I change the filter value in realtime with FFmpeg’s C API or I need to create a new filter link each time when it requires to update ?

• ffmpeg trim mp3 - determine precisely the start and end times of section to be trimmed

  4 février 2019, par Ahmed Khalil

  I have a long mp3 track of an audio book (more than 9 hours long) that I would like to trim using ffmpeg.

  The sample code below is used to trim an mp3 section by providing the start and end times. However, when I determine the start and end times, then checking the output file, it’s not as precisely as I want, sometimes several minutes ahead/before the desired point.

  import subprocess
  
  file = r'audio book.mp3'
  
  track_name = "trimmed section"
  
  output = r'D:\{0}'.format(track_name)
  
  start = '01:26:04'
  
  end = '01:33:17'
  
  
  
  d = subprocess.getoutput('ffmpeg -i "{0}" -ss {1} -to {2} -c copy {3}.mp3"'
  
                       .format(file, start, end, output))
  
  
  
  print(d)

  Is there a way to determine with accuracy the real start and end time of an mp3 audio track, to be given afterwards as inputs to the code...to trim the desired sections all at once, without the need to adjust/fine-tune the start and end time manually ??