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• PHP-FFMpeg not converting (error with ffmpeg)

  13 août 2016, par Magurean Dan Sergiu

  I have installed ffmpeg server side with all the dependencies, updated it. And then installed PHP-FFMpeg with Composer. Tested that ffmpeg is instaled with a ssh conexion.

  root@host [/opt/ffmpeg]# ffmpeg
  
  ffmpeg version N-81322-ge8b355a Copyright (c) 2000-2016 the FFmpeg developers
  
  built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
  
  configuration: --disable-yasm
  
  libavutil      55. 28.100 / 55. 28.100
  
  libavcodec     57. 51.100 / 57. 51.100
  
  libavformat    57. 46.100 / 57. 46.100
  
  libavdevice    57.  0.102 / 57.  0.102
  
  libavfilter     6. 51.100 /  6. 51.100
  
  libswscale      4.  1.100 /  4.  1.100
  
  libswresample   2.  1.100 /  2.  1.100
  
  Hyper fast Audio and Video encoder
  
  usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options]     outfile}...
  
  
  
  Use -h to get full help or, even better, run 'man ffmpeg'

  But when I try to convert a video file with a php script it throws me a very long error.
  PHP Code :

  <?PHP
  
  include($_SERVER['DOCUMENT_ROOT']."/vendors/install/vendor/autoload.php");
  
  $ffmpeg = FFMpeg\FFMpeg::create();
  
  $video = $ffmpeg->open($_SERVER['DOCUMENT_ROOT'].'/resources/uploads/96048/ffd3c5e349fa5745159deef163b5c253.wmv');
  
  $video
  
         ->filters()
  
         ->synchronize();
  
  $format = new FFMpeg\Format\Video\X264();
  
  $format->on('progress', function ($video, $format, $percentage) {
  
      echo "$percentage % transcoded";});
  
  $video
  
        ->save($format,$_SERVER['DOCUMENT_ROOT'].'/resources/uploads/96048/ffd3c5e349fa5745159deef163b5c253.mp4');
  
  ?>

  Error displayed :

  <b>Fatal error</b>:  Uncaught exception 'Alchemy\BinaryDriver\Exception\ExecutionFailureException' with message 'ffmpeg failed to execute command '/usr/local/bin/ffmpeg' '-y' '-i' '/home/user/working/resources/uploads/23097/2efeb41a1a7085a1dbd469c4895ac23e.wmv' '-async' '1' '-metadata:s:v:0' 'start_time=0' '-threads' '12' '-vcodec' 'libx264' '-acodec' 'libfaac' '-b:v' '1000k' '-refs' '6' '-coder' '1' '-sc_threshold' '40' '-flags' '+loop' '-me_range' '16' '-subq' '7' '-i_qfactor' '0.71' '-qcomp' '0.6' '-qdiff' '4' '-trellis' '1' '-b:a' '128k' '-pass' '1' '-passlogfile' '/tmp/ffmpeg-passes57ad320a1b685j794u/pass-57ad320a1b716' '/home/user/working/resources/uploads/23097/2efeb41a1a7085a1dbd469c4895ac23e.mp4'' in /home/user/working/vendors/install/vendor/alchemy/binary-driver/src/Alchemy/BinaryDriver/ProcessRunner.php:100↵Stack trace:↵#0 /home/user/working/vendors/install/vendor/alchemy/binary-driver/src/Alchemy/BinaryDriver/ProcessRunner.php(72): Alchemy\BinaryDriver\ProcessRunner-&amp;gt;doExecutionFailure(''/usr/bin/ffmpe...')↵#1 / in <b>/home/user/working/vendors/install/vendor/php-ffmpeg/php-ffmpeg/src/FFMpeg/Media/Video.php</b> on line <b>168</b><br />↵"

  Then I have took the basic code displayed upwards and ran it directly through ssh.

  '/usr/local/bin/ffmpeg' '-y' '-i' '/home/user/working/resources/uploads/23097/2efeb41a1a7085a1dbd469c4895ac23e.wmv' '-async' '1' '-metadata:s:v:0' 'start_time=0' '-threads' '12' '-vcodec' 'libx264' '-acodec' 'libfaac' '-b:v' '1000k' '-refs' '6' '-coder' '1' '-sc_threshold' '40' '-flags' '+loop' '-me_range' '16' '-subq' '7' '-i_qfactor' '0.71' '-qcomp' '0.6' '-qdiff' '4' '-trellis' '1' '-b:a' '128k' '-pass' '1' '-passlogfile' '/tmp/ffmpeg-passes57ad320a1b685j794u/pass-57ad320a1b716' '/home/user/working/resources/uploads/23097/2efeb41a1a7085a1dbd469c4895ac23e.mp4'

  And this is the response I get :

  ffmpeg version N-81322-ge8b355a Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
  
    configuration: --disable-yasm
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 51.100 / 57. 51.100
  
    libavformat    57. 46.100 / 57. 46.100
  
    libavdevice    57.  0.102 / 57.  0.102
  
    libavfilter     6. 51.100 /  6. 51.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, asf, from '/home/user/working/resources/uploads/23097/2efeb41a1a7085a1dbd469c4895ac23e.wmv':
  
    Metadata:
  
      SfOriginalFPS   : 299700
  
      WMFSDKVersion   : 11.0.6001.7000
  
      WMFSDKNeeded    : 0.0.0.0000
  
      comment         : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball
  
      title           : Wildlife in HD
  
      copyright       : © 2008 Microsoft Corporation
  
      IsVBR           : 0
  
      DeviceConformanceTemplate: AP@L3
  
    Duration: 00:00:30.09, start: 0.000000, bitrate: 6977 kb/s
  
      Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 2 channels, fltp, 192 kb/s
  
      Stream #0:1(eng): Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x720, 5942 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 1k tbc
  
  Unknown encoder 'libx264'

  My question is what to do next ? I have wasted 3 days to make this work and this is the furthest I have come. How can I test if libx264 is installed corectly ? And if it is not installed corectly how cand I reinstal or make it work ?

• How to open a .sw file using libav library in c language ?

  29 août 2018, par volatile

  I created a .sw(16 bit pcm) file by passing an audio file. Now I am trying to get back the original audio(.mp3) by passing the .sw file as an input to the following file.

  How can I read the .sw file content so that I can get back the mp3 file. Below is the code,

  #include 
  
  #include 
  
  #include 
  
  
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  
  
  #include <libavutil></libavutil>channel_layout.h>
  
  #include <libavutil></libavutil>common.h>
  
  #include <libavutil></libavutil>frame.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  
  
  /* check that a given sample format is supported by the encoder */
  
  static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
  
  {
  
      const enum AVSampleFormat *p = codec->sample_fmts;
  
  
  
      while (*p != AV_SAMPLE_FMT_NONE) {
  
          if (*p == sample_fmt)
  
              return 1; 
  
          p++;
  
      }
  
      return 0;
  
  }
  
  
  
  /* just pick the highest supported samplerate */
  
  static int select_sample_rate(const AVCodec *codec)
  
  {
  
      const int *p;
  
      int best_samplerate = 0;
  
  
  
      if (!codec->supported_samplerates)
  
          return 44100;
  
  
  
      p = codec->supported_samplerates;
  
      while (*p) {
  
          if (!best_samplerate || abs(44100 - *p) &lt; abs(44100 - best_samplerate))
  
              best_samplerate = *p;
  
          p++;
  
      }
  
      return best_samplerate;
  
  }
  
  
  
  /* select layout with the highest channel count */
  
  static int select_channel_layout(const AVCodec *codec)
  
  {
  
      const uint64_t *p;
  
      uint64_t best_ch_layout = 0;
  
      int best_nb_channels   = 0;
  
  
  
      if (!codec->channel_layouts)
  
          return AV_CH_LAYOUT_STEREO;
  
  
  
      p = codec->channel_layouts;
  
      while (*p) {
  
          int nb_channels = av_get_channel_layout_nb_channels(*p);
  
  
  
          if (nb_channels > best_nb_channels) {
  
              best_ch_layout    = *p;
  
              best_nb_channels = nb_channels;
  
          }
  
          p++;
  
      }
  
      return best_ch_layout;
  
  }
  
  
  
  static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
  
                     FILE *output)
  
  {
  
      int ret;
  
  
  
     /* send the frame for encoding */
  
     ret = avcodec_send_frame(ctx, frame);
  
     if (ret &lt; 0) {
  
         fprintf(stderr, "Error sending the frame to the encoder\n");
  
         exit(1);
  
     }
  
  
  
     /* read all the available output packets (in general there may be any
  
      * number of them */
  
     while (ret >= 0) {
  
         ret = avcodec_receive_packet(ctx, pkt);
  
         if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
  
             return;
  
         else if (ret &lt; 0) {
  
             fprintf(stderr, "Error encoding audio frame\n");
  
             exit(1);
  
         }
  
  
  
         fwrite(pkt->data, 1, pkt->size, output);
  
         av_packet_unref(pkt);
  
     }
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
     const char *filename;
  
     const AVCodec *codec;
  
     AVCodecContext *c= NULL;
  
     AVFrame *frame;
  
     AVPacket *pkt;
  
     int i, j, k, ret;
  
     FILE *f;
  
     uint16_t *samples;
  
     float t, tincr;
  
  
  
     av_register_all();
  
     avcodec_register_all();
  
  
  
     if (argc &lt;= 1) {
  
         fprintf(stderr, "Usage: %s <output file="file">\n", argv[0]);
  
         return 0;
  
     }
  
     filename = argv[1];
  
  
  
     /* find the MP2 encoder */
  
     codec = avcodec_find_encoder(AV_CODEC_ID_MP3);
  
     if (!codec) {
  
         fprintf(stderr, "Codec not found\n");
  
         exit(1);
  
     }
  
  
  
     c = avcodec_alloc_context3(codec);
  
     if (!c) {
  
         fprintf(stderr, "Could not allocate audio codec context\n");
  
         exit(1);
  
     }
  
  
  
     /* put sample parameters */
  
     c->bit_rate = 64000;
  
  
  
     /* check that the encoder supports s16 pcm input */
  
     c->sample_fmt = AV_SAMPLE_FMT_S16P;
  
     if (!check_sample_fmt(codec, c->sample_fmt)) {
  
         fprintf(stderr, "Encoder does not support sample format %s",
  
                 av_get_sample_fmt_name(c->sample_fmt));
  
         exit(1);
  
     }
  
  
  
     /* select other audio parameters supported by the encoder */
  
     c->sample_rate    = select_sample_rate(codec);
  
     c->channel_layout = select_channel_layout(codec);
  
     c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
  
  
  
     /* open it */
  
     if (avcodec_open2(c, codec, NULL) &lt; 0) {
  
         fprintf(stderr, "Could not open codec\n");
  
         exit(1);
  
     }
  
  
  
     f = fopen(filename, "wb");
  
     if (!f) {
  
         fprintf(stderr, "Could not open %s\n", filename);
  
         exit(1);
  
     }
  
  
  
     /* packet for holding encoded output */
  
     pkt = av_packet_alloc();
  
     if (!pkt) {
  
         fprintf(stderr, "could not allocate the packet\n");
  
         exit(1);
  
     }
  
  
  
     /* frame containing input raw audio */
  
     frame = av_frame_alloc();
  
     if (!frame) {
  
         fprintf(stderr, "Could not allocate audio frame\n");
  
         exit(1);
  
     }
  
  
  
     frame->nb_samples     = c->frame_size;
  
     frame->format         = c->sample_fmt;
  
     frame->channel_layout = c->channel_layout;
  
  
  
     /* allocate the data buffers */
  
     ret = av_frame_get_buffer(frame, 0);
  
     if (ret &lt; 0) {
  
         fprintf(stderr, "Could not allocate audio data buffers\n");
  
         exit(1);
  
     }
  
  
  
     /* encode a single tone sound */
  
     t = 0;
  
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
  
     for (i = 0; i &lt; 200; i++) {
  
         /* make sure the frame is writable -- makes a copy if the encoder
  
          * kept a reference internally */
  
         ret = av_frame_make_writable(frame);
  
         if (ret &lt; 0)
  
             exit(1);
  
         samples = (uint16_t*)frame->data[0];
  
  
  
         for (j = 0; j &lt; c->frame_size; j++) {
  
             samples[2*j] = (int)(sin(t) * 10000);
  
  
  
             for (k = 1; k &lt; c->channels; k++)
  
                 samples[2*j + k] = samples[2*j];
  
             t += tincr;
  
         }
  
         encode(c, frame, pkt, f);
  
     }
  
  
  
     /* flush the encoder */
  
     encode(c, NULL, pkt, f);
  
  
  
     fclose(f);
  
  
  
     av_frame_free(&amp;frame);
  
     av_packet_free(&amp;pkt);
  
     avcodec_free_context(&amp;c);
  
  
  
     return 0;
  
  }
  
  </output>

  I just want to know, where and how the .sw file is reading in the above audio encoding code ?

• libavcodec/libx265 : add user data unregistered SEI encoding

  12 juillet 2021, par Brad Hards

  libavcodec/libx265 : add user data unregistered SEI encoding
  MISB ST 0604 and ST 2101 require user data unregistered SEI messages
  
  (precision timestamps and sensor identifiers) to be included. That
  
  currently isn't supported for libx265. This patch adds support
  
  for user data unregistered SEI messages in accordance with
  
  ISO/IEC 23008-2:2020 Section D.2.7
  The design is based on nvenc, with support finished up at
  
  57de80673cb
  Signed-off-by : Derek Buitenhuis <derek.buitenhuis@gmail.com>
  

  • [DH] libavcodec/libx265.c