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• How to correctly calculate which segments are ready to be downloaded using MPEG-DASH

  24 avril 2019, par igal k

  What i’m trying to do ?

  Write a simple MPEG-DASH client using the SegmentTemplate pattern to calculate which segments are ready to be downloaded for a live source.

  A picture taken using chrome’s debugging tools at a moment X showing an mpd request(8af651fd747.....mpd) and the actual segments fetched respectfully to that request.

  

  Given the following MPD

  <mpd availabilitystarttime="2019-04-24T06:43:32Z" maxsegmentduration="PT4.096S" minbuffertime="PT4.096S" minimumupdateperiod="PT15.835S" profiles="urn:mpeg:dash:profile:isoff-live:2011" publishtime="2019-04-24T11:14:01Z" suggestedpresentationdelay="PT11.878S" timeshiftbufferdepth="PT65.536S" type="dynamic" xmlns="urn:mpeg:dash:schema:mpd:2011">
  
    <location>https://content-aaps1.uplynk.com/channel/8af651fd7473474f86a05ffb0a1c8972.mpd?rmt=wv&amp;amp;cid=8af651fd7473474f86a05ffb0a1c8972&amp;amp;oid=600e5c27541344a1bf3818617ad712ce&amp;amp;prettydash=1&amp;amp;exp=1556091088&amp;amp;rn=4138683939&amp;amp;tc=1&amp;amp;ct=c&amp;amp;sig=5fb7f0c18f3f1d2ad4fdee53c02c1e1ed904bc5e8474f4ebf886d209ff7f21c9&amp;amp;pbs=05b6594bcf4b4728ac1094976a80194d</location>
  
    <period start="PT2826.240S">
  
      <adaptationset maxframerate="30" maxheight="720" maxwidth="1280" mimetype="video/mp4" segmentalignment="true" startwithsap="1">
  
        <representation bandwidth="2604473" codecs="avc1.64001e" framerate="30" height="360" scantype="progressive" width="640">
  
          <baseurl>https://x-default-stgec.uplynk.com/aapm/slices/8c1/600e5c27541344a1bf3818617ad712ce/8c1027496a964b049f1bd5895f8f0412/</baseurl>
  
          <segmenttemplate duration="368640" initialization="https://x-default-stgec.uplynk.com/aapm/slices/8c1/600e5c27541344a1bf3818617ad712ce/8c1027496a964b049f1bd5895f8f0412/$RepresentationID$_init.mp4?pbs=05b6594bcf4b4728ac1094976a80194d&amp;amp;_jt=l&amp;amp;chid=8af651fd7473474f86a05ffb0a1c8972" media="$RepresentationID$$Number%08d$.m4f?pbs=05b6594bcf4b4728ac1094976a80194d&amp;amp;_jt=l&amp;amp;chid=8af651fd7473474f86a05ffb0a1c8972" presentationtimeoffset="254361599" startnumber="690" timescale="90000"></segmenttemplate>
  
        </representation>
  
      </adaptationset>
  
    </period>
  
    <utctiming schemeiduri="urn:mpeg:dash:utc:http-iso:2014" value="https://content-aaps1.uplynk.com/misc/utcservertime"></utctiming>
  
  </mpd>

  I see that the next segment request should be #3955

  What i have tried so far

      period.end = 1556104456;
  
      period.start = 2826;
  
      availability_start_time = 1556088212;
  
      max_segment_duration = 4;
  
      time_shift_buffer_depth = 65

  So, first of all, i read DASH-IF-IOP 4.3 section 4.3.4.2 page #82 and implemented the following code :

     int k1 = 1;
  
  int period_duration = period.end - (period.start + data_.availability_start_time);
  
  int k2 = ceil((float)period_duration / (float)data_.max_segment_duration);
  
  double duration = ((float)representation.duration / (float)representation.timeScale);
  
  size_t live_edge = std::min(
  
      (int)floor((float)((data_.publish_time - data_.availability_start_time - period.start) / duration)), k2);
  
  
  
  size_t oldest = std::max(k1, (int)floor((float)((data_.publish_time - data_.availability_start_time - period.start -
  
                                                      data_.time_shift_buffer_depth) /
  
                                                  duration)));

  after calculating everything : k1=1, k2=3355, live_edge=3272 and oldest=3256

  Also tried using ffmpeg’s dashdec.c

  for min_segment :

  if (c->is_live && pls->fragment_duration)
  
      {
  
          num = pls->first_seq_no + (((get_current_time_in_sec() - c->availability_start_time) - c->time_shift_buffer_depth) * pls->fragment_timescale) / pls->fragment_duration;
  
      }

  for max_segment :

  num = pls->first_seq_no + (((get_current_time_in_sec() - c->availability_start_time)) * pls->fragment_timescale) / pls->fragment_duration;

  after a small modification :

  size_t pmax = (((data_.publish_time - data_.availability_start_time))) / duration;
  
  size_t pmin = ((data_.publish_time - data_.availability_start_time) - data_.time_shift_buffer_depth) / duration;

  pmin=3946 pmax=3961

  in the ffmpeg example, i had to manually remove the first_seq_no variable because it looked like i doubled added the SegmentTemplate@StartNumber.

  even after succeeding in this task, how do i exactly build the request list of Segment(NOW) ----> Segment(LIVE_EDGE)

• FFMPEG PHP enter command lines ?

  3 mai 2019, par Robert

  You will have to excuse me I have been spending the past 2 days reading through old FFMPEG posts for an answer but did little but confuse myself.
  It seems from what I read FFMPEG-PHP wrappers aren’t supported anymore ??? and to be honest they don’t seem like the proper way of learning how to incorporate it with PHP as there is a whole lot more help for command line FFMPEG usage and the FFMPEG-PHP wrapper usage looks nothing like the command line as far as I can tell.

  So I have 2 questions on using ffmpeg with PHP. So we are on the same page I posted a little info below.

  I downloaded FFMPEG static for windows 64bit.
  I then ran (in composer)

        $ composer require php-ffmpeg/php-ffmpeg

  in my vendor folder i have the following path.

      vendor\ffmpeg-20190429-ac551c5-win64-static\bin

  if I open the command prompt in that folder and type FFMPEG I get.

   ffmpeg version N-93710-gac551c54b1 Copyright (c) 2000-2019 the FFmpeg 
  
   developers
  
  built with gcc 8.3.1 (GCC) 20190414
  
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable- 
  
   fontconfig --enable-gnutls --enable-iconv --enable-libass --enable- 
  
   libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame -- 
  
   enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
  
   --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr -- 
  
   enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
  
   --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 -- 
  
   enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable- 
  
   libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa 
  
   --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx -- 
  
   enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable- 
  
   nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
   libavutil      56. 26.100 / 56. 26.100
  
   libavcodec     58. 52.100 / 58. 52.100
  
   libavformat    58. 27.103 / 58. 27.103
  
   libavdevice    58.  7.100 / 58.  7.100
  
   libavfilter     7. 50.100 /  7. 50.100
  
   libswscale      5.  4.100 /  5.  4.100
  
   libswresample   3.  4.100 /  3.  4.100
  
   libpostproc    55.  4.100 / 55.  4.100
  
   Hyper fast Audio and Video encoder
  
   usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...

  So I’m pretty sure FFMPEG is installed which was my goal.

  Now on my PHP side here is where I am at, and I’m not sure how this works.

  1ST QUESTION. Do I need to save my $_POST files to the drive before manipulating them ? Or can I use the $file and $filep as is ? I don’t really want to store those files, only the output.

  My SLIM code.

   $app->post('/telestrator', function(Request $request, Response $response) 
  
   {
  
  
  
   $response = array();
  
  
  
   if (isset ($_POST['ID']) && ($_POST['position']) && $_FILES['video'] 
  
   ['error'] === UPLOAD_ERR_OK && $_FILES['image']['error'] === 
  
   UPLOAD_ERR_OK) {
  
  
  
   $file = $_FILES['video']['tmp_name'];
  
   $filep = $_FILES['image']['tmp_name'];
  
   $time = $_POST['position'];
  
   $position = msToTime($time);
  
   $filetime = round(microtime(true) * 1000);
  
   $outputfolder = 'teletemp/';
  
  
  
   $ID = $_POST['ID'];
  
  
  
   $tempvid = $ID . 'tempvid' . 'mp4';
  
   $finalvid = $ID  . $filetime . 'mp4';
  
  
  
  
  
   $ffmpegpath = "public/ffmpeg.exe";
  
  
  
  echo "Starting ffmpeg...\n\n";
  
  echo shell_exec("$ffmpegpath -loop 1 -i $filep -c:v libx264 -t 3 -pix_fmt 
  
  yuv420p \"$outputfolder.\" $tempvid /");
  
   echo shell_exec("$ffmpegpath -i $file -t $position -c copy 
  
  \"$outputfolder\" 
  
   small-1.mp4 -ss $position -codec copy \"$outputfolder\" small-2.mp4 ");
  
   echo shell_exec("$ffmpegpath -i small-1.mp4 -i $tempvid.mp4 -i small- 
  
  2.mp4 \ 
  
   -filter_complex \"[0:v:0][1:v:0][2:v:0]concat=n=3:v=1:a=1[outv]\" \ 
  
   -map \"[outv]\" $finalvid" );
  
   echo "Done.\n";
  
  
  
   $upload = new videouploads();
  
  
  
          $desc = 'telestrated video for ' . $ID . $filetime;
  
          $ID = $_POST['ID'];  
  
  
  
          if ($upload->saveVideoFile($finalvid, getFileExtension($finalvid), 
  
   $desc, $ID)) {
  
              $response['error'] = false;
  
              $response['message'] = 'File Uploaded Successfullly';
  
          }
  
       else {
  
          $response['error'] = true;
  
          $response['message'] = 'Required parameters are not available';
  
      }
  
      echo json_encode($response);
  
    } 
  
   });
  
  
  
   function getFileExtension($file)
  
   {
  
   $path_parts = pathinfo($file);
  
   return $path_parts['extension'];
  
   }
  
  
  
   function msToTime($duration) {
  
   $seconds = floor($duration / 1000);
  
   $minutes = floor($seconds / 60);
  
   $hours = floor($minutes / 60);
  
   $milliseconds = $duration % 1000;
  
   $seconds = $seconds % 60;
  
   $minutes = $minutes % 60;
  
  
  
   $format = '%02u:%02u:%02u.%03u';
  
   $time = sprintf($format, $hours, $minutes, $seconds, $milliseconds);
  
   return rtrim($time, '0');
  
  
  
   }

  so it’s not running here is my postman. How do I get it to actually run the FFMPEG.

   Starting ffmpeg...
  
  
  
   Done.
  
  
  
   <br />
  
   <b>Notice</b>:  Undefined index: extension in
  
   <b>C:\xampp\htdocs\Pathways\public\index.php</b> on line
  
   <b>14741</b>
  
   <br />
  
   {"error":false,"message":"File Uploaded Successfullly"}

• Why is audio stream extraction slow ? (compared to AviSynth)

  6 mai 2019, par LLL

  I want to extract the audio stream of an avi file as a wav file, it works but it is really slow ( 4-5fps) although I just want to copy the stream.

  Here is the type of stream I want to extract (ffprobe info) :
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

  Going through AviSynth does it about 100 times faster, but I would prefer a pure FFmpeg solution. Why such a speed difference ? It looks like FFmpeg is reading and processing through the whole file whereas AviSynth can just extract the data without reading it.

  Example :
  ffmpeg -i file.avi -vn -ac 2 -c:a copy audio.wav
  or
  ffmpeg -i file.avi -map 0:a -ac 2 -c:a copy audio.wav
  both work fine but take time.

  Using an AviSynth script as input :
  ffmpeg -i script.avs -map 0:a -ac 2 -c:a copy audio.wav
  with script.avs containing just :
  AviSource("file.avi")
  does the same but almost instantaneously !

  Any idea why AviSynth is so much faster and if there is a way to get the same speed in FFmpeg ?

  Edit : adding logs
  Using FFmpeg directly :

  E:\>ffmpeg -i "file.avi" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  [avi @ 0000018d3c38a680] non-interleaved AVI
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avi, from 'file.avi':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: 534682 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 1280x720, 533183 kb/s, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.12 bitrate=1411.2kbits/s speed=4.77x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=1.188s stime=50.766s rtime=234.254s
  
  bench: maxrss=17468kB

  Using AviSynth :

  E:\>ffmpeg -i "soundout.avs" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avisynth, from 'soundout.avs':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (BGR[24] / 0x18524742), bgr24, 1280x720, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.11 bitrate=1411.2kbits/s speed= 155x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=0.234s stime=1.047s rtime=7.236s
  
  bench: maxrss=23792kB