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• the "non-monotonic DTS in output stream " error while concatenation even after reencoded input files

  19 juin 2024, par nish

  hi I am trying to write a Python program that is trying to first edit different videos separately under the heading "intro" "story" and "byte" with the help of FFmpeg and Python subprocess module and then concatenate them in the function named "bind". At first, I encountered this "Non-monotonic DTS in output stream" then i reencoded the input files before concatenating them but i am still getting this error

  


  # function to assemble all the videos 


  


  def bind() :

  


  bunch = ["final_intro.mp4","final_story.mp4","finalbite.mp4"]
new_bunch =[]
for video in bunch:
    name = f"re_{video}"
    re_command =[
        "ffmpeg", "-y",
        "-i", video,
        "-c:v", "libx264",
        "-c:a", "aac",
        "-strict", "experimental",
        "-b:a", "192k",
        name
    ]
    subprocess.run(re_command)
    new_bunch.append(name)
    

with open("concat_list_final.txt", "w") as f:
    for video in new_bunch:
        f.write(f"file './{video}'\n")
  
command_final = [ 
    "ffmpeg", "-y",
    "-f", "concat",
    "-safe", "0",
    "-i", "concat_list_final.txt",
    "-c", "copy",
    "done.mp4"
]
subprocess.run(command_final)
for video in new_bunch:
    os.remove(video)
os.remove("final.mp4")
os.remove("updated_final.mp4")
os.remove("concat_list_final.txt")


return "done.mp4"


  


  for reference this is the error messages :-

  


  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x14cf16040] Auto-inserting h264_mp4toannexb bitstream filter [mp4 @ 0x12ce06880] Non-monotonic DTS in output stream 0:0 ; previous : 152933, current : 127760 ; changing to 152934. This may result in incorrect timestamps in the output file. [mp4 @ 0x12ce06880] Non-monotonic DTS in output stream 0:0 ; previous : 152934, current : 128272 ; changing to 152935. This may result in incorrect timestamps in the output file..........

  


• FFmpeg error with ffmpeg.FS("readfile", "output.mp4"). trying to get ffmpeg to work in the react app

  21 juin 2024, par Paul Tham

   const stackVideos = useCallback(
    async (video1) => {
      try {
        console.log("Fetching video2 from storage...");
        const video2Ref = ref(storage, "video2.mp4");
        const video2Url = await getDownloadURL(video2Ref);
        const video2Blob = await (await fetch(video2Url)).blob();

        console.log("Writing video1 to FFmpeg FS...");
        await ffmpeg.FS("writeFile", "video1.mp4", await fetchFile(video1));

        console.log("Writing video2 to FFmpeg FS...");
        await ffmpeg.FS("writeFile", "video2.mp4", await fetchFile(video2Blob));

        console.log("Files in FFmpeg FS after write:");
        const files = await ffmpeg.FS("readdir", "/");
        console.log(files);

        const { start, end } = inputs[0];
        const startSeconds = new Date(`1970-01-01T${start}Z`).getTime() / 1000;
        const endSeconds = new Date(`1970-01-01T${end}Z`).getTime() / 1000;
        const duration = endSeconds - startSeconds;

        console.log("Running FFmpeg command...");
        await ffmpeg.run(
          "-i",
          "video1.mp4",
          "-ss",
          startSeconds.toString(),
          "-t",
          duration.toString(),
          "-i",
          "video2.mp4",
          "-filter_complex",
          "[0:v]scale=1080:-1[v1];[1:v]scale=-1:1920/2[v2scaled];[v2scaled]crop=1080:1920/2[v2cropped];[v1][v2cropped]vstack=inputs=2,scale=1080:1920[vid]",
          "-map",
          "[vid]",
          "-map",
          "0:a",
          "-c:v",
          "libx264",
          "-crf",
          "23",
          "-preset",
          "veryfast",
          "-shortest",
          "output1.mp4"
        );

        console.log("Files in FFmpeg FS after run:");
        const filesAfterRun = await ffmpeg.FS("readdir", "/");
        console.log(filesAfterRun);

        console.log("Reading output1.mp4 from FFmpeg FS...");
        const data = await ffmpeg.FS("readfile", "output1.mp4");
        console.log("after the FS readfile");
        const url = URL.createObjectURL(
          new Blob([data.buffer], { type: "video/mp4" })
        );
        setStackedVideo(url);
        setOutputFileReady(true); // Mark output file as ready
      } catch (err) {
        console.error("FFmpeg error output:", err);
        setError(`FFmpeg run error: ${err.message}`);
        setIsProcessing(false);
      }
    },
    [inputs]
  );


  


  My error seems to be stemming from this line :

  


   const data = await ffmpeg.FS("readfile", "output1.mp4");


  


  Seeing the ffmpeg.wasm documentation i thought the functions for some of the functions had changed, but when I changed it, it seemed like they did not recognise the new functions. Sometimes this will also give me some other errors like worker.js which I dont understand enough to debug this myself.

  


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• Alexa developer console gives me "The audio is not of a supported MPEG version" Error

  20 mars, par Boban BoBo Banjevic

  I am trying to make an Alexa skill, and I am using audio mp3 files stored is S3, lambda function has access to my DynamoDb and ultimately S3. But I have an issue with my audio files. I keep on getting "The audio is not of a supported MPEG version" when I test alexa skill. I tried multiple ffmpeg -i file conversions.

  


  When I type ffprobe -v quiet -print_format json -show_format -show_streams "Piano.mf.A1.audio10.mp3" this is what i have for my file down below :

  


  
Input #0, mp3, from 'Piano.mf.A1.audio10.mp3':
  Metadata:
    encoder         : Lavf61.9.107
  Duration: 00:00:07.00, start: 0.025057, bitrate: 129 kb/s
  Stream #0:0: Audio: mp3 (mp3float), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc61.33
PS C:\Users\boban\OneDrive\Desktop\newBob2> ffprobe -v quiet -print_format json -show_format -show_streams "Piano.mf.A1.audio10.mp3"
{
    "streams": [
        {
            "index": 0,
            "codec_name": "mp3",
            "codec_long_name": "MP3 (MPEG audio layer 3)",
            "codec_type": "audio",
            "codec_tag_string": "[0][0][0][0]",
            "codec_tag": "0x0000",
            "sample_fmt": "fltp",
            "sample_rate": "44100",
            "channels": 2,
            "channel_layout": "stereo",
            "bits_per_sample": 0,
            "initial_padding": 0,
            "r_frame_rate": "0/0",
            "avg_frame_rate": "0/0",
            "time_base": "1/14112000",
            "start_pts": 353600,
            "start_time": "0.025057",
            "duration_ts": 98795520,
            "duration": "7.000816",
            "bit_rate": "128000",
            "disposition": {
                "default": 0,
                "dub": 0,
                "original": 0,
                "comment": 0,
                "lyrics": 0,
                "karaoke": 0,
                "forced": 0,
                "hearing_impaired": 0,
                "visual_impaired": 0,
                "clean_effects": 0,
                "attached_pic": 0,
                "timed_thumbnails": 0,
                "non_diegetic": 0,
                "captions": 0,
                "descriptions": 0,
                "metadata": 0,
                "dependent": 0,
                "still_image": 0,
                "multilayer": 0
            },
            "tags": {
                "encoder": "Lavc61.33"
            }
        }
    ],
    "format": {
        "filename": "Piano.mf.A1.audio10.mp3",
        "nb_streams": 1,
        "nb_programs": 0,
        "nb_stream_groups": 0,
        "format_name": "mp3",
        "format_long_name": "MP2/3 (MPEG audio layer 2/3)",
        "start_time": "0.025057",
        "duration": "7.000816",
        "size": "112892",
        "bit_rate": "129004",
        "probe_score": 51,
        "tags": {
            "encoder": "Lavf61.9.107"
        }
    }
}


  


  It should be everything that is required, am I doing something wrong, this is the only thing stopping me from finishing the skill. Asking ChatGPT said it could be because of the mp3float, and after many conversions it still doesn't work.