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• Failed to convert web-saved .wemb audio to .wav by using php "shell_exec" and javascript

  30 mai 2022, par Anirbasgnaw

  I'm working on an online experimenter which could record participants' audio from the browser. The audio data I get has an extension of .wemb, so I plan to use ffmpeg to convert it to .wav while I save the data.

  


  I tried to use PHP's shell_exec but nothing happens when I run the scripts. Then I found that my echo and print_r also did not work. I'm new to PHP and javascript, so I''m really confused now.

  


  Below are the relevant codes, I really appreciate it if you could help !

  


  write_data.php :

  


  <?php
  $post_data = json_decode(file_get_contents('php://input'), true); 
  // the directory "data" must be writable by the server
  $name = "../".$post_data['filename'];
  $data = $post_data['filedata'];
   // write the file to disk
  file_put_contents($name, $data);
  
  $INPUT = trim($name) . ".webm";
  $OUTPUT = trim($name) . ".wav";
  echo "start converting...";

  // check if ffmprg is available
  $ffmpeg = trim(shell_exec('which ffmpeg'));
  print_r($ffmpeg);
  // call ffmpeg
  shell_exec("ffmpeg -i '$INPUT' -ac 1 -f wav '$OUTPUT' 2>&1 ");
?>


  


  javascript :

  


    saveData: function(fileName,format){
    // save  as json by default
    if (!format){ format = 'json';}
    // add extension to filename
    fileName = `${fileName}.${format}`
    // create saveData object using fetch
    let saveData = [];
    if (format == 'json') {
        saveData = {
          type: 'call-function',
          async: true,
          func: async function(done) {
            let data = jsPsych.data.get().json();
            const response = await fetch("../write_data.php", {
              method: "POST",
              headers: {
                "content-type": "application/json"
              },
              body: JSON.stringify({ filename: fileName, filedata: data })
            });
            if (response.ok) {
              const responseBody = await response.text();
              done(responseBody);
            }
          }
        }
    } else {
        saveData = {
          type: 'call-function',
          async: true,
          func: async function(done) {
            let data = jsPsych.data.get().csv();
            const response = await fetch("../write_data.php", {
              method: "POST",
              headers: {
                "content-type": "application/json"
              },
              body: JSON.stringify({ filename: fileName, filedata: data })
            });
            if (response.ok) {
              const responseBody = await response.text();
              done(responseBody);
            }
          }
        }
    }
    return saveData;
  },


  


• I have a ffmpeg command to concatenate 300+ videos of different formats. What is the proper syntax for the concat complex filter ?

  25 avril 2022, par jokoon

  I plan to concatenate a large amount of video files of different formats and resolution, some without sound, and add a short black screen "pause" of about 0.5s between each.

  


  I wrote a python script to generate such command.

  


  I created a 0.5s video file using ffmpeg.exe -t 0.5 -f lavfi -i color=c=black:s=640x480 -c:v libx264 -tune stillimage -pix_fmt yuv420p blank500ms.mp4.

  


  I then added a silent audio to it with -f lavfi -i anullsrc -c:v copy -c:a aac -shortest

  


  I now have the problem of adding a blank audio track for streams without one, but I don't want to generate new file, I want to add it to my complex filter.

  


  This is my complex script and generate command.

  


  The command (there are line returns, because I send this with the python subprocess module)

  


  ffmpeg.exe
-i
input0.mp4
-i
input1.mp4
-i
input2.mp4
-i
input3.mp4
-i
input4.mp4
-i
input5.mp4
-i
input6.mp4
-i
input7.mp4
-i
input8.mp4
-i
input9.mp4
-i
input10.mp4
-f
lavfi
-i
anullsrc
-filter_complex_script
C:/filter_complex_script.txt
-map
"[final_video]"
-map
"[final_audio]"
output.mp4


  


  The complex_filter_script :

  


  [0]fps=24[fps0];
[fps0]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled0];
[1]fps=24[fps1];
[fps1]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled1];
[2]fps=24[fps2];
[fps2]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled2];
[3]fps=24[fps3];
[fps3]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled3];
[4]fps=24[fps4];
[fps4]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled4];
[5]fps=24[fps5];
[fps5]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled5];
[6]fps=24[fps6];
[fps6]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled6];
[7]fps=24[fps7];
[fps7]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled7];
[8]fps=24[fps8];
[fps8]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled8];
[9]fps=24[fps9];
[fps9]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled9];
[10]fps=24[fps10];
[fps10]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled10];
[10]split=10[blank0][blank1][blank2][blank3][blank4][blank5][blank6][blank7][blank8][blank9];
[rescaled0:v][0:a][blank0][rescaled1:v][1:a][blank1][rescaled2:v][2:a][blank2][rescaled3:v][3:a][blank3][rescaled4:v][4:a][blank4][rescaled5:v][5:a][blank5][rescaled6:v][11:a][blank6][rescaled7:v][11:a][blank7][rescaled8:v][11:a][blank8][rescaled9:v][11:a][blank9]concat=n=22:v=1:a=1[final_video][final_audio]


  


  As you can see, some video use [11:a], because it's a silent audio stream.

  


  input10.mp4, mapped to [10] and then split (or "cloned") into blanked0 to 9, is a short pause separator.

  


  ffmpeg tells me the error

  


  [Parsed_split_55 @ 000001591c33b280] Media type mismatch between the 'Parsed_split_55' filter output pad 1 (video) and the 'Parsed_concat_56' filter input pad 5 (audio)
[AVFilterGraph @ 000001591bf1e6c0] Cannot create the link split:1 -> concat:5
Error initializing complex filters.
Invalid argument


  


  I'm a bit lost when it comes to using the [X:Y:Z] syntax, and how the order matter in the concat argument list.

  


  I'm open to any other suggestion to solve my problem. I would rather do this in a single command, without intermediate file.

  


  EDIT :

  


  For details, I already wrote a large concat+xstack filter that worked well with 8GB of memory.

  


  In this case, there are a lot of inputs, but those inputs are small, most of them are between 1 and 10MB, so it would probably not generate out-of-memory problems, although I'm not certain.

  


• FFMPEG stereo track stops capturing at random times during a capture session

  26 mai 2022, par mrwassen

  I am currently working on building a workflow to capture and archive a large stash of family and friends PAL and NTSC VHS tapes. The hardware setup is as follows :

  


  

  • JVC HR-7860S VCR

  


  • s-video / RCA audio >

  


  • ADVC-3000 converter

  


  • SDI / BNC cable >

  


  • Blackmagic Decklink Mini Recorder 4K PCIe card

  


  • installed in a fairly hi-spec windows machine : AMD Ryzen 9 5900X 3.7 Ghz base 12 core, GEFORCE RTX 3060 12 gB, 32 gB ram

  


  


  The plan is to capture to lossless AVI, then drop into an NLE (Vegas Pro v.16) to do a minimal amount of cleanup / trimming, then render to a more compressed video format (TBD) for upload to AWS S3 accessible through a family website.

  


  The issue I am having is that when I run the capture using ffmpeg/directshow e.g. for a perfectly fine 90 min. PAL tape, at some random point of time during the capture one of the 2 stereo channels just stops capturing. This has happened with all of the tapes I have tested so far, and it happens at different times during the same video. I have examined the frames surrounding points in time when this happens, and it doesn't correlate to any transitions or jitter, but often just randomly in the middle of a perfectly smooth scene. Once the one channel stops capturing it never starts back up again during that capture session.

  


  The ADVC-3000 and the VCR are both showing both stereo channels playing normally throughout the capture. The windows machine running the capture hardly breaks a sweat at any time, and the transfer easily keeps up constantly showing a speed = 1x which I assume means nothing lagging. Also there are no video/audio sync issues at any point in time even towards the end of long tapes e.g. 90 mins.

  


  I am fairly new at ffmpeg, so I have spent extensive amounts of time reading up on forum posts and experimenting and have ended up with the following syntax :

  


  ffmpeg -y -f dshow -rtbufsize 2000M -i video="Blackmagic WDM Capture":audio="Blackmagic WDM Capture" -codec:v v210 -pix_fmt yuv422p -codec:a pcm_s16le -b:a 128k -t 02:00:00 -r 25 -threads 4 -maxrate 2500k -filter:a "volume=1.5" output_v210_audio.avi


  


  The capture runs without a single dropped frame, the only error I am getting when launching (and perhaps this is a smoking gun ?) is :

  


  


  "Non-monotonous DTS in output stream 0:1 ; previous : 0, current : -30 ;
changing to 1. This may result in incorrect timestamps in the output
file."

  


  


  I have tried to troubleshoot this in the hopes that it is tied to my issue but so far without luck.

  


  Hoping somebody can help correct or modify my command line or perhaps other ideas to help resolve the issue.