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    2 mars 2010, par

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Sur d’autres sites (3433)

  • ffmpeg : How can a MOV with transparent background be created ?

    25 mars 2017, par Mat

    I’m trying - with no success at all - to convert the green pixels of a background into transparent ones and output the result as clip with ffmpeg. N.b. I do not want to lay the clip over anything ; I’m not having a problem with that. What I want is a clip with transparent background for the OpenShot video editor (the chromakey filter of which doesn’t work satisfyingly).

    What I have tried (amongst 1 zillion other things over the last 15 hrs.) was

    ffmpeg.exe -i in.mov -vf chromakey=0x008001:0.115:0.0 -c:v qtrle out.mov

    but the pixels simply would not be transparent. Seemingly, nothing happens. I reckon the filter is ok, because it works fine in a complex chain (overlaying a background image).

    The output of ffprompt -show_stream -show_format of out.mov is as follows :

    [STREAM]
    
index=0
    
codec_name=qtrle
    
codec_long_name=QuickTime Animation (RLE) video
    
profile=unknown
    
codec_type=video
    
codec_time_base=1/30
    
codec_tag_string=rle
    
codec_tag=0x20656c72
    
width=1920
    
height=1080
    
coded_width=1920
    
coded_height=1080
    
has_b_frames=0
    
sample_aspect_ratio=1:1
    
display_aspect_ratio=16:9
    
pix_fmt=bgra
    
level=-99
    
color_range=N/A
    
color_space=unknown
    
color_transfer=unknown
    
color_primaries=unknown
    
chroma_location=unspecified
    
field_order=progressive
    
timecode=N/A
    
refs=1
    
id=N/A
    
r_frame_rate=30/1
    
avg_frame_rate=30/1
    
time_base=1/15360
    
start_pts=0
    
start_time=0.000000
    
duration_ts=54789
    
duration=3.566992
    
bit_rate=822383192
    
max_bit_rate=N/A
    
bits_per_raw_sample=N/A
    
nb_frames=107
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
TAG:language=eng
    
TAG:handler_name=DataHandler
    
TAG:encoder=Lavc57.64.101 qtrle
    
[/STREAM]
    
[STREAM]
    
index=1
    
codec_name=aac
    
codec_long_name=AAC (Advanced Audio Coding)
    
profile=LC
    
codec_type=audio
    
codec_time_base=1/44100
    
codec_tag_string=mp4a
    
codec_tag=0x6134706d
    
sample_fmt=fltp
    
sample_rate=44100
    
channels=2
    
channel_layout=stereo
    
bits_per_sample=0
    
id=N/A
    
r_frame_rate=0/0
    
avg_frame_rate=0/0
    
time_base=1/44100
    
start_pts=926
    
start_time=0.020998
    
duration_ts=157481
    
duration=3.570998
    
bit_rate=132103
    
max_bit_rate=132103
    
bits_per_raw_sample=N/A
    
nb_frames=153
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
TAG:language=eng
    
TAG:handler_name=DataHandler
    
[/STREAM]
    
[FORMAT]
    
filename=out.mov
    
nb_streams=2
    
nb_programs=0
    
format_name=mov,mp4,m4a,3gp,3g2,mj2
    
format_long_name=QuickTime / MOV
    
start_time=0.000000
    
duration=3.567000
    
size=366708874
    
bit_rate=822447712
    
probe_score=100
    
TAG:major_brand=qt
    
TAG:minor_version=512
    
TAG:compatible_brands=qt
    
TAG:encoder=Lavf57.56.101
    
[/FORMAT]

    I have a "sample" clip which shows the behaviour I want, with the following stream and information :

    [STREAM]
    
index=0
    
codec_name=qtrle
    
codec_long_name=QuickTime Animation (RLE) video
    
profile=unknown
    
codec_type=video
    
codec_time_base=1/24
    
codec_tag_string=rle
    
codec_tag=0x20656c72
    
width=1920
    
height=1080
    
coded_width=1920
    
coded_height=1080
    
has_b_frames=0
    
sample_aspect_ratio=0:1
    
display_aspect_ratio=0:1
    
pix_fmt=bgra
    
level=-99
    
color_range=N/A
    
color_space=unknown
    
color_transfer=unknown
    
color_primaries=unknown
    
chroma_location=unspecified
    
field_order=progressive
    
timecode=N/A
    
refs=1
    
id=N/A
    
r_frame_rate=24/1
    
avg_frame_rate=24/1
    
time_base=1/12288
    
start_pts=0
    
start_time=0.000000
    
duration_ts=74760
    
duration=6.083984
    
bit_rate=49226848
    
max_bit_rate=N/A
    
bits_per_raw_sample=N/A
    
nb_frames=146
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
TAG:language=eng
    
TAG:handler_name=DataHandler
    
TAG:encoder=Lavc57.24.102 qtrle
    
[/STREAM]
    
[STREAM]
    
index=1
    
codec_name=aac
    
codec_long_name=AAC (Advanced Audio Coding)
    
profile=LC
    
codec_type=audio
    
codec_time_base=1/48000
    
codec_tag_string=mp4a
    
codec_tag=0x6134706d
    
sample_fmt=fltp
    
sample_rate=48000
    
channels=2
    
channel_layout=stereo
    
bits_per_sample=0
    
id=N/A
    
r_frame_rate=0/0
    
avg_frame_rate=0/0
    
time_base=1/48000
    
start_pts=0
    
start_time=0.000000
    
duration_ts=293856
    
duration=6.122000
    
bit_rate=53537
    
max_bit_rate=128000
    
bits_per_raw_sample=N/A
    
nb_frames=288
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
TAG:language=eng
    
TAG:handler_name=DataHandler
    
[/STREAM]
    
[FORMAT]
    
filename=templateOK.mov
    
nb_streams=2
    
nb_programs=0
    
format_name=mov,mp4,m4a,3gp,3g2,mj2
    
format_long_name=QuickTime / MOV
    
start_time=0.000000
    
duration=6.144000
    
size=37478506
    
bit_rate=48800138
    
probe_score=100
    
TAG:major_brand=qt
    
TAG:minor_version=512
    
TAG:compatible_brands=qt
    
TAG:encoder=Lavf57.25.100
    
[/FORMAT]

    and I simply am not able to spot the relevant difference.

    The input, output and the working template can be found here.

    (The security issue you might see when clicking the link comes from the server certificate being self-signed. You can accept a temporal exception. Btw : The ridiculous file size of the output file will be the next nut to crack. Probably something about compression.)

  • fluent ffmpeg size output option not working

    19 janvier 2017, par Ashbury

    Summary : I’m trying to limit output to 3mb, .outputOptions('-fs 3000000') isn’t working for me, the file is coming back with a size of 119260428 or 119mb.

    Here is the code to try for yourself, all you need is a test.mp3 large enough that the resulting testoutput.ogg is > 3mb :

    var ffmpeg = require("fluent-ffmpeg");
    
var command = ffmpeg();
    

    
var convertToOGG = function(){
    
  var fileName = 'test.mp3'
    

    
  ffmpeg.ffprobe(fileName, function(err, metadata) {
    
    command
    
      .input(fileName)
    
      .inputFormat("mp3")
    
      .audioChannels(1)
    
      .outputOptions('-fs', 3000000)
    
      .output('testoutput.ogg')
    
      .on("progress", function(progress) {
    
        console.log("Processing: " + progress.timemark);
    
      })
    
      .on("error", function(err, stdout, stderr) {
    
        console.log("Cannot process video: " + err.message);
    
      })
    
      .on("end", function(stdout, stderr) {
    
        ffmpeg.ffprobe('testoutput.ogg', function(err,metadata){
    
          if(metadata.format.size >= 3000000){
    

    
            console.log("didn't work")
    
          } 
    
        })
    
      })
    
    .run();
    
  });
    
};
    

    
convertToOGG();

    Per the fluent-ffmpeg documentation you should be able to use a ffmpeg command in an output option : outputOption()

    This method allows passing any output-related option to ffmpeg. You can call it with a single argument to pass a single option, optionnaly
    with a space-separated parameter :

    /* Single option */
    
ffmpeg('/path/to/file.avi').outputOptions('-someOption');

    and in FFMPEG’s documentation :

    -fs limit_size (output) Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The
    size of the output file is slightly more than the requested file size.

    It’s giving me no errors, just seemingly ignoring the file size limit of 99mb and outputting a 119.3mb file.

    Edit - Looks like -fs 3000000 is working for mp3 to wav, but still wont do mp3 to ogg. This is the output from running the command in terminal :

    ✗ ffmpeg -i test.mp3 -fs 3000000 testoutput.ogg
    
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
    
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
    
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
    
  libavutil      55. 34.100 / 55. 34.100
    
  libavcodec     57. 64.101 / 57. 64.101
    
  libavformat    57. 56.100 / 57. 56.100
    
  libavdevice    57.  1.100 / 57.  1.100
    
  libavfilter     6. 65.100 /  6. 65.100
    
  libavresample   3.  1.  0 /  3.  1.  0
    
  libswscale      4.  2.100 /  4.  2.100
    
  libswresample   2.  3.100 /  2.  3.100
    
  libpostproc    54.  1.100 / 54.  1.100
    
[mp3 @ 0x7fc6a4000000] Estimating duration from bitrate, this may be inaccurate
    
Input #0, mp3, from 'test.mp3':
    
  Metadata:
    
    lyrics-eng      : xxx
    
    title           : xxx
    
    artist          : xxx
    
    album_artist    : xxx
    
    album           : xxx
    
    genre           : xxx
    
  Duration: 03:27:28.74, start: 0.000000, bitrate: 128 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, mono, s16p, 128 kb/s
    
    Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 540x360, 90k tbr, 90k tbn, 90k tbc
    
    Metadata:
    
      title           : Array
    
      comment         : Cover (front)
    
[swscaler @ 0x7fc6a4808800] deprecated pixel format used, make sure you did set range correctly
    
[ogg @ 0x7fc6a3815800] Frame rate very high for a muxer not efficiently supporting it.
    
Please consider specifying a lower framerate, a different muxer or -vsync 2
    
Output #0, ogg, to 'testoutput.ogg':
    
  Metadata:
    
    lyrics-eng      : xxx
    
    title           : xxx
    
    artist          : xxx
    
    album_artist    : xxx
    
    album           : xxx
    
    genre           : xxx
    
    encoder         : Lavf57.56.100
    
    Stream #0:0: Video: theora (libtheora), yuv444p, 540x360, q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
    
    Metadata:
    
      title           : Array
    
      DESCRIPTION     : Cover (front)
    
      encoder         : Lavc57.64.101 libtheora
    
      lyrics-eng      : xxx
    
      artist          : xxx
    
      ALBUMARTIST     : xxx
    
      album           : xxx
    
      genre           : xxx
    
    Stream #0:1: Audio: vorbis (libvorbis), 44100 Hz, mono, fltp
    
    Metadata:
    
      encoder         : Lavc57.64.101 libvorbis
    
      lyrics-eng      : xxx
    
      title           : xxx
    
      artist          : xxx
    
      ALBUMARTIST     : xxx
    
      album           : xxx
    
      genre           : xxx
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (mjpeg (native) -> theora (libtheora))
    
  Stream #0:0 -> #0:1 (mp3 (native) -> vorbis (libvorbis))
    
Press [q] to stop, [?] for help
    
frame=    1 fps=0.0 q=-0.0 Lsize=  116465kB time=03:27:28.71 bitrate=  76.6kbits/s speed=61.2x
    
video:9kB audio:114907kB subtitle:0kB other streams:0kB global headers:6kB muxing overhead: 1.347787%

  • Audio recorded with MediaRecorder on Chrome missing duration

    27 octobre 2016, par suppp111

    I am recording audio (oga/vorbis) files with MediaRecorder. When I record these file through Chrome I get problems : I cannot edit the files on ffmpeg and when I try to play them on Firefox it says they are corrupt (they do play fine on Chrome though).

    Looking at their metadata on ffmpeg I get this :

    Input #0, matroska,webm, from '91.oga':
    
  Metadata:
    
    encoder         : Chrome
    
  Duration: N/A, start: 0.000000, bitrate: N/A
    
    Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
    
[STREAM]
    
index=0
    
codec_name=opus
    
codec_long_name=Opus (Opus Interactive Audio Codec)
    
profile=unknown
    
codec_type=audio
    
codec_time_base=1/48000
    
codec_tag_string=[0][0][0][0]
    
codec_tag=0x0000
    
sample_fmt=fltp
    
sample_rate=48000
    
channels=1
    
channel_layout=mono
    
bits_per_sample=0
    
id=N/A
    
r_frame_rate=0/0
    
avg_frame_rate=0/0
    
time_base=1/1000
    
start_pts=0
    
start_time=0.000000
    
duration_ts=N/A
    
duration=N/A
    
bit_rate=N/A
    
max_bit_rate=N/A
    
bits_per_raw_sample=N/A
    
nb_frames=N/A
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
TAG:language=eng
    
[/STREAM]
    
[FORMAT]
    
filename=91.oga
    
nb_streams=1
    
nb_programs=0
    
format_name=matroska,webm
    
format_long_name=Matroska / WebM
    
start_time=0.000000
    
duration=N/A
    
size=7195
    
bit_rate=N/A
    
probe_score=100
    
TAG:encoder=Chrome

    As you can see there are problems with the duration. I have looked at posts like this :
    How can I add predefined length to audio recorded from MediaRecorder in Chrome ?

    But even trying that, I got errors when trying to chop and merge files.For example when running :

    ffmpeg -f concat  -i 89_inputs.txt -c copy final.oga

    I get a lot of this :

    [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57612, current: 1980; changing to 57613. This may result in incorrect timestamps in the output file.
    
[oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57613, current: 2041; changing to 57614. This may result in incorrect timestamps in the output file.
    
DTS -442721849179034176, next:42521 st:0 invalid dropping
    
PTS -442721849179034176, next:42521 invalid dropping st:0
    
[oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57614, current: 2041; changing to 57615. This may result in incorrect timestamps in the output file.
    
[oga @ 00000000006789c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
    
DTS -442721849179031296, next:42521 st:0 invalid dropping
    
PTS -442721849179031296, next:42521 invalid dropping st:0

    Does anyone know what we need to do to audio files recorded from Chrome for them to be useful ? Or is there a problem with my setup ?

    Recorder js :

    if (navigator.getUserMedia) {
    
  console.log('getUserMedia supported.');
    

    
  var constraints = { audio: true };
    
  var chunks = [];
    

    
  var onSuccess = function(stream) {
    
    var mediaRecorder = new MediaRecorder(stream);
    

    
    record.onclick = function() {
    
      mediaRecorder.start();
    
      console.log(mediaRecorder.state);
    
      console.log("recorder started");
    
      record.style.background = "red";
    

    
      stop.disabled = false;
    
      record.disabled = true;
    

    
      var aud = document.getElementById("audioClip");
    
      start = aud.currentTime;
    
    }
    

    
    stop.onclick = function() {
    
      console.log(mediaRecorder.state);
    
      console.log("Recording request sent.");
    
      mediaRecorder.stop();
    
    }
    

    
    mediaRecorder.onstop = function(e) {
    
      console.log("data available after MediaRecorder.stop() called.");
    

    
      var audio = document.createElement('audio');
    
      audio.setAttribute('controls', '');
    
      audio.setAttribute('id', 'audioClip');
    

    
      audio.controls = true;
    
      var blob = new Blob(chunks, { 'type' : 'audio/ogg; codecs="vorbis"' });
    
      chunks = [];
    
      var audioURL = window.URL.createObjectURL(blob);
    
      audio.src = audioURL;
    

    
      sendRecToPost(blob);   // this just send the audio blob to the server by post
    
      console.log("recorder stopped");
    

    
    }

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