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  • Extracting MP3 Data from Generic Movie Files using FFMPEG

    21 février 2017, par MoonKnight

    I am using FFMPEG via MediaToolkit to extract MP3 data (to .mp3 file) from different movie files (merely .mp4 and .mov for now).

    MediaToolkit is just a C# wrapper for FFMPEG which calls FFMPEG via Process.Start, so to do this I am using the method

    public static string GenerateMp3FromVideoFile(string filePath)
{
    string mp3Path = String.Empty;
    using (var engine = new Engine())
    {
        mp3Path = Path.GetFileNameWithoutExtension(filePath);
        mp3Path = Path.Combine(Utils.GetBuildRoamingAppDataDirectory(), mp3Path);
        mp3Path = Path.ChangeExtension(mp3Path, ".mp3");
        string paramString = String.Format(
            //"-i \"{0}\" -q:a 0 -map a \"{1}\"",
            //"-i \"{0}\" -ar 320K \"{1}\"",
            //"-i \"{0}\" -acodec libmp3lame -ar 44100 -b:a 192k -id3v2_version 3 -write_id3v1 1 \"{1}\"",
            //"-i \"{0}\" -vn -ar 44100 -ac 2 -ab 192k -f mp3 \"{1}\"",
            "-i \"{0}\" -acodec libmp3lame \"{1}\"",
            filePath,
            mp3Path);
        engine.CustomCommand(paramString);
    }
    return mp3Path;
}

    All of the options I have tried above for the command line arguments passed to FFMPEG have worked for .mp4 video files and create the desired .mp3 output. However, for the .mov file I have I am getting the following System.Exception

    69: video:0kB audio:1059kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.030074%Conversion failed!

    I have tried a number of different methods to perform this extraction with varing control via the arguments passed to FFMPEG, but with no luck for the .mov file.

    Looking at https://linuxconfig.org/ffmpeg-audio-format-conversions it seems as though there is not conversion to MP3 from .mov files, so I have looked at first converting the .mov to .mp4 via 

    ffmpeg -i mymovie.mov -vcodec copy -acodec copy out.mp4

    and then extracting the audio, but this is expensive for large files. Is there a way of extracting the MP3 data directly from the .mov file?

    Thanks for your time.

    When the above command is run from FFMPEG.exe, the output is:

    [aac @ 00000000021b00a0] Inconsistent channel configuration. [aac @ 00000000021b00a0] get_buffer() failed Error while decoding stream #0:1: Invalid argument [aac @ 00000000021b00a0] Reserved bit set. [aac @ 00000000021b00a0] Number of bands (6) exceeds limit (5). Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Number of bands (16) exceeds limit (13). Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] channel element 3.7 is not allocated Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] channel element 3.8 is not allocated Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Reserved bit set. [aac @ 00000000021b00a0] TNS filter order 28 is greater than maximum 12. Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Number of bands (26) exceeds limit (18). Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 00000000021b00a0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git . If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 00000000021b00a0] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and cont act the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org) Error while decoding stream #0:1: Not yet implemented in FFmpeg, patches welcome [aac @ 00000000021b00a0] Reserved bit set. [aac @ 00000000021b00a0] Prediction is not allowed in AAC-LC. Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Reserved bit set. [aac @ 00000000021b00a0] ms_present = 3 is reserved. Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Number of bands (3) exceeds limit (2). Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] SBR was found before the first channel element. [aac @ 00000000021b00a0] channel element 3.14 is not allocated Error while decoding stream #0:1: Invalid data found when processing input [aac @ 00000000021b00a0] Sample rate index in program config element does not match the sample rate index configured by the container.

  • mid-roll ads in HLS or mpeg-dash

    21 février 2017, par evan

    I'm encoding my videos with ffmpeg and transcoding them into HLS videos. I want to add mid-roll ads to my converted videos.

    I was wondering what is the best way to do it? should I just add the ads to my video and make a big video stream and then do the HLS thing? or I should break the HLs video and put the ads in the middle while playing?

    My content is live streaming and I want to add mid-roll ads in the middle of my video while playing.

    ffmpeg -i main720.MTS -movflags faststart -s 640*360 -r 24 -vcodec libx264 -preset slow -tune zerolatency -tune fastdecode -b:v 300k -maxrate 300k -bufsize 400k -g 96 -an hls3/main720_300.mp4 -s 1280*720 -r 24 -vcodec libx264 -preset slow -tune zerolatency -tune fastdecode -b:v 700k -maxrate 700k -bufsize 1000k -g 96 -an hls3/main720_700.mp4 -hls_time 4 -hls_segment_filename 'hls3/file%03d.ts' -hls_list_size 0 hls3/out.m3u8

  • ffmpeg - set variable video duration

    21 février 2017, par Vee

    I use this command to convert files in batch plus crop and re-scale:

    for i in $( ls *.mp4 ); do
    ffmpeg -y -i "$i" -acodec libfaac -ab 128k -ar 44100 -vcodec libx264 -b 850k -threads 0 -vf [in]crop=in_w-112:in_h-63:56:0,scale=1280:720[out] "../../archive/${i/.mp4/}.mp4"
done

    this command will start at second 15, and makes video 30 seconds long:

    for i in $( ls *.mp4 ); do
    ffmpeg -ss 00:00:15 -t 30 -y -i "$i" -acodec libfaac -ab 128k -ar 44100 -vcodec libx264 -b 850k -threads 0 -vf [in]crop=in_w-112:in_h-63:56:0,scale=1280:720[out] "${i/.mp4/}_test.mp4"
done

    what I would like is a command that cuts off 15s from the beginning and 15s from the end of EACH video from the BATCH ... the trick is that each video has different duration, so "how long it is" must be a variable (duration minus 15s or minus 30s if I count 15s from the start as well)

    video duration examples:

    video 1 - 00:25:19
video 2 - 00:15:34
video 3 - 00:19:21
video 4 - 00:22:49
etc.

  • FFMPEG replace video audio with filter_complex

    21 février 2017, par Chintan7027

    I wants such a output video where audio of output is created using ffmpeg -filter_complex mechanism,

    /usr/local/Cellar/ffmpeg/3.2.2/bin/ffmpeg 
-i /uploads/videos/1487684390-lg9htt0RW2.mov 
-i /uploads/audios/1487664761-SCPbo6Tkac.mp3 
-filter_complex "
[0:a]atrim=0:8.70824980736,asetpts=PTS-STARTPTS[aud1]; 
[1:a]atrim=0:12.9567301273,asetpts=PTS-STARTPTS[aud2];
[0:a]volume=0.3,atrim=start=8.70824980736:21.6649799347,asetpts=PTS-STARTPTS[slow_aud]; 
[aud2][slow_aud] amerge=inputs=2[a_merged];
[0:a]atrim=start=21.6649799347:31.6410098076 [remaining_audio]; 
[aud1][a_merged][remaining_audio]concat=n=3:v=0:a=1[aout]" 
-map 0:v -map "[aout]" -c:v copy -acodec mp3 
/uploads/output/1487684390-lg9htt0RW2.mov

    Original Audio Recorded Based On UTC timestamp Vs Original Video Recorded Based on UTC timestamp

                13:00-------- Original Event Audio -------- 13:20
       12:50------------- Event Video Recorded --------------13:30

    This is my requirement

    So the audio of the output video should contains

    1. First 10 seconds(12:50 - 13:00) are Audio of Event Video Recorded
    2. Next 20 seconds (13:00 -13:20) are merged audio(Original Audio+ Original Video where Original video's audio volume is .3)
    3. Remaining 10 seconds(13:21-13:30) of video will play remaing audio of video

    What I am getting by above commands

    1. First 10 seconds(12:50 - 13:00) are Audio of Event Video Recorded Achieved
    2. Next 20 seconds (13:00 -13:20) are merged audio(Original Audio+ Original Video where Original video's audio volume is .3) Achieved
    3. Remaining 10 seconds(13:21-13:30) of video will play remaining audio of video Not Achieved

  • ffmpeg webm encode for low powered devices

    21 février 2017, par Max Tkachenko

    I want to play transparent video into my app using built-in player over phone's camera capture. I try to encode my video with alpha channel for android device:

    ffmpeg -i "Comp.avi" -c:v libvpx -pix_fmt yuva420p -metadata:s:v:0 alpha_mode="1" output.webm

    The result is pretty good, but I have lags (freezing video from time to time) while playing it on my android phone. Is it any options to improve decode performance?

    Some console output: 

    D:\SOFT\ffmpeg-20160207-git-9ee4c89-win64-static\bin>ffmpeg -i "d:\temp\cherti\Comp 1.avi" -c:v libvpx -pix_fmt yuva420p -metadata:s:v:0 alpha_mode="1" d:\temp\cherti\output.webm
ffmpeg version N-80386-g5f5a97d Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 24.100 / 55. 24.100
  libavcodec     57. 46.100 / 57. 46.100
  libavformat    57. 38.100 / 57. 38.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 46.101 /  6. 46.101
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Input #0, avi, from 'd:\temp\cherti\Comp 1.avi':
  Metadata:
    date            : 2017-02-18T14:10:42.00916
    encoder         : Adobe After Effects CC 2015 (Windows)
  Duration: 00:00:05.00, start: 0.000000, bitrate: 1592542 kb/s
    Stream #0:0: Video: rawvideo, bgra, 1080x1920, 1605907 kb/s, 24 fps, 24 tbr, 24 tbn, 24 tbc
File 'd:\temp\cherti\output.webm' already exists. Overwrite ? [y/N] y
[libvpx @ 0000000002593640] v1.5.0
[webm @ 00000000025a54e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, webm, to 'd:\temp\cherti\output.webm':
  Metadata:
    date            : 2017-02-18T14:10:42.00916
    encoder         : Lavf57.38.100
    Stream #0:0: Video: vp8 (libvpx), yuva420p, 1080x1920, q=-1--1, 200 kb/s, 24 fps, 1k tbn, 24 tbc
    Metadata:
      alpha_mode      : 1
      encoder         : Lavc57.46.100 libvpx
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> vp8 (libvpx))

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