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• FFMpeg generating screenshots drop frames

  6 octobre 2011, par Morgan

  I'm using FFMpeg to extract frames from a video file, and for most videos it works flawlessly. But certain videos it seems to have to seek to the position that I want to take the screenshot from, and drops all the frames on the way there taking quite a while if the position i'm generating a screenshot from is near the end of the video. Here is the command I'm using to extract my screenshot.

  ffmpeg -ss 60 -i input.mkv -vframes 1 -f image2 -an -r 1/5 output.jpg

  and the output I receive

  ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Sep 20 2011 00:46:47 with clang 2.1 (tags/Apple/clang-163.7.1)
  
    configuration: --prefix=/Users/Morgan/Downloads/installed/ffmpeg --extra-libs=/Users/Morgan/Downloads/installed/lib/libfaac.a --enable-gpl --extra-cflags=-I/Users/Morgan/Downloads/installed/include --disable-ffplay --disable-ffserver --disable-shared --disable-debug --extra-ldflags=-L/Users/Morgan/Downloads/installed/lib --enable-libmp3lame --enable-libfaac --enable-libx264 --enable-nonfree --cc=clang
  
    libavutil    51.  9. 1 / 51.  9. 1
  
    libavcodec   53.  7. 0 / 53.  7. 0
  
    libavformat  53.  4. 0 / 53.  4. 0
  
    libavdevice  53.  1. 1 / 53.  1. 1
  
    libavfilter   2. 23. 0 /  2. 23. 0
  
    libswscale    2.  0. 0 /  2.  0. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  [matroska,webm @ 0x7f9dd3007c00] Estimating duration from bitrate, this may be inaccurate
  
  
  
  Seems stream 0 codec frame rate differs from container frame rate: 47.95 (500000/10427) -> 23.98 (24000/1001)
  
  Input #0, matroska,webm, from 'input.mkv':
  
    Duration: 01:01:36.16, start: 0.000000, bitrate: 576 kb/s
  
      Stream #0.0(eng): Video: h264 (High), yuv420p, 1280x720, PAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
  
      Metadata:
  
        title           : Input Video
  
      Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s (default)
  
      Metadata:
  
        title           : English Dolby Digital 5.1 AC3
  
      Stream #0.2(eng): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s
  
      Metadata:
  
        title           : Commentary
  
      Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 (default)
  
      Metadata:
  
        title           : English
  
  Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p'
  
  [buffer @ 0x106c10c00] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param:
  
  [buffersink @ 0x106c10e40] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
  
  [scale @ 0x106c11140] w:1280 h:720 fmt:yuv420p -> w:1280 h:720 fmt:yuvj420p flags:0x4
  
  Output #0, image2, to 'output.jpg':
  
    Metadata:
  
      encoder         : Lavf53.4.0
  
      Stream #0.0(eng): Video: mjpeg, yuvj420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 0.20 tbc (default)
  
      Metadata:
  
        title           : Input Video
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop, [?] for help
  
  frame=    0 fps=  0 q=0.0 Lsize=      -0kB time=00:00:00.00 bitrate=   0.0kbits/s dup=0 drop=275    
  
  video:0kB audio:0kB global headers:0kB muxing overhead -inf%

• FFMPEG - Read Audio File on html player while writing ?

  23 février 2018, par Nathan Ferreyrolles

  I have the project to record a streams of french radios and after read files on my website (record every weeks) I have segmented by hours and for 7 days.
  But i have 2 questions :
  When i play the mp3 file who writing at the time on my website i have a bug like -02:30:33s remain ? What’s wrong for you ?

  This is my ffmpeg code : ./ffmpeg -y -i http://broadcast.infomaniak.net/radioscoop-bourg-128.mp3 -f segment -strftime 1 -segment_atclocktime 1 -segment_time 3600 -reconnect 1 -reconnect_at_eof 1 -reconnect_delay_max 2 -timeout 200000 -acodec copy /var/www/mysite.fr/public_html/rsbourg/%A/%H.mp3

  And i would like to know a code to reconnect the record if i have a problem, because this morning the programm has stopped for nothing reasons ? I use Putty and the programm Screen for record many radios at same time :)

  Best regards :)
  (sorry for my bad english i’m french :/ )
  Nathan

• QT paly audio by QAudioDevidce can't connect to PulseAudioService

  30 mars 2022, par XMingG

  Like title, i'm trying to use Qt and FFmpeg to play audio. My code like this :

  


  QAudioOutput *audio_output;

    QIODevice *stream_out;

    QAudioFormat audio_fmt;
    audio_fmt.setSampleRate(44100);
    audio_fmt.setChannelCount(2);
    audio_fmt.setSampleSize(16);
    audio_fmt.setCodec("audio/pcm");
    audio_fmt.setByteOrder(QAudioFormat::LittleEndian);
    audio_fmt.setSampleType(QAudioFormat::SignedInt);
    
    QAudioDeviceInfo info = QAudioDeviceInfo::defaultOutputDevice();
    if(!info.isFormatSupported(audio_fmt))
    {
        audio_fmt = info.nearestFormat(audio_fmt);
    }
    
    audio_output = new QAudioOutput(audio_fmt);


  


  When i use QAudioDeviceInfo info = QAudioDeviceInfo::defaultOutputDevice()
i get PulseAudioService : pa_context_connect() failed error.
So how can i fix it ?
By the way, i'm using Ubuntu 16.04 and Qt 5.14.2, and i have add 'mutilmedia' to Qt pro file

  


  I checked my Qt file ,and i have audio dir in plugins, it's not lib problem. Also, i read this post ,but i don't know how to fix it, anybody have idea ? Thank you guys,and my English is bad, wish you can understand what do i say.