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  • Get duration through FFMPEG 2014 gcc 4.86

    19 février 2014, par M.Yazdian

    I like to use ffmpeg in my web project to convert video files and get some information of metadata's.

    So I download FFMPEG and use this codes in asp.net:

    For view my code please follow this link http://forums.asp.net/t/1969247.aspx?Problem+width+FFMPEG+2014+gcc+4+86

    This Codes worked currently when I use FFMPEG 2013 gcc 4.2 But when I use FFMPEG 2014 gcc 4.8.2 it's not working...!!!

    In FFMPEG 2014 , I can Convert video formats but I cann't get duration and In FFMPEG 2013 gcc 4.2 I can get duration but I cann't convert video formats (I mean specialy MKV).

    In the FFMPEG 2014 Standard Error I cann't find Duration to parse(HOUR , MINUTE , SECOND).

    In the FFMPEG 2013 Standard Error I can find Duration to parse(HOUR , MINUTE , SECOND) but I cann't connvert video formats.

    Thanks in advance

  • Error when use command of ffmpeg on android

    19 février 2014, par user2830969

    I get source code from https://github.com/JayH5/android-ffmpeg-cmdline and then I edit code to be simple than.

    public ProcessRunnable create() {
            if (inputPath == null || outputPath == null) {
                throw new IllegalStateException("Need an input and output filepath!");
            }   
    
            final List cmd = new LinkedList();
    
            cmd.add(mFfmpegPath);
            cmd.add("-i");
            cmd.add(inputPath);
            cmd.add("-s 240x160");
    
            cmd.add(outputPath);
            Log.w("Command", cmd.toString());
            final ProcessBuilder pb = new ProcessBuilder(cmd);
            return new ProcessRunnable(pb);
        }
    

    When I run code, the result is *.mp file 0.00 kb. Please help me fix about this. thank so much

  • Reduce Bandwidth requirement for Video Streaming

    19 février 2014, par user3239282

    Please suggest some points on, how to reduce the bandwidth requirement for a video stream.

    We are actually streaming the desktop across internet, Remote Access Application.

    Encoder : libx264.lib

    Decoder : Ffmpeg

    Codec : H264

    Following is how am configuring h264 for encoding:

        x264_param_default_preset(&this->m_x264Settings, "veryfast", "zerolatency");
        this->m_x264Settings.i_threads = 1;
        this->m_x264Settings.i_width = width / 2;
        this->m_x264Settings.i_height = height / 2;
        this->m_x264Settings.i_keyint_max = 10;//fps;
        this->m_x264Settings.b_intra_refresh = 1;
        this->m_x264Settings.i_fps_num = 10;
        this->m_x264Settings.i_fps_den = 1;
        // Intra refres:
        this->m_x264Settings.i_keyint_max = 25;
        this->m_x264Settings.b_intra_refresh = 1;
        //Rate control:
        this->m_x264Settings.rc.i_rc_method = X264_RC_CRF;
        this->m_x264Settings.rc.f_rf_constant = 25;
        this->m_x264Settings.rc.f_rf_constant_max = 35;
    
        //For streaming:
        this->m_x264Settings.b_repeat_headers = 1;
        this->m_x264Settings.b_annexb = 1;
        x264_param_apply_profile(&this->m_x264Settings, "baseline");
    
        this->m_width = width;
        this->m_height = height;
        this->m_encoder = x264_encoder_open(&this->m_x264Settings);
        x264_picture_alloc(&this->m_picIn,X264_CSP_I420,(width / 2),(height / 2));
        int sws_flags = SWS_BICUBIC;
        m_imgConverCtx = sws_getContext(width, height, PIX_FMT_RGB32, (width/2), (height/2), PIX_FMT_YUV420P,sws_flags, NULL, NULL, NULL);
    

    Thanks in advance, Paul.

  • ffmpeg cannot open a simple microsoft wav file exported with Audacity

    18 février 2014, par sebpiq

    I have exported a sound file to microsoft wav using Audacity. I am trying to open this file with ffmpeg :

    ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg
    

    and here's the ouput I get :

    fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
      built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
      configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
      libavutil      52. 18.100 / 52. 18.100
      libavcodec     54. 92.100 / 54. 92.100
      libavformat    54. 63.104 / 54. 63.104
      libavdevice    54.  3.103 / 54.  3.103
      libavfilter     3. 42.103 /  3. 42.103
      libswscale      2.  2.100 /  2.  2.100
      libswresample   0. 17.102 /  0. 17.102
      libpostproc    52.  2.100 / 52.  2.100
    [dca @ 0x7fd30c013600] Not a valid DCA frame
    
    ... SNIP ...
    
    [dca @ 0x7fd5bc013600] Invalid bit allocation index
    [dca @ 0x7fd5bc013600] error decoding block
        Last message repeated 3 times
    [dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
    [dca @ 0x7fd5bc013600] error decoding block
    [wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
    [wav @ 0x7fd5bc013000] decoding for stream 0 failed
    [wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
    Consider increasing the value for the 'analyzeduration' and 'probesize' options
    steps-stereo-16b-44khz.wav: could not find codec parameters
    

    If I export the same file to .ogg or .aiff, no problem, the following works fine :

    ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg
    

    Any idea what could be wrong?

    A link to my wav file so you can try to reproduce.

    NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

    EDIT

    Getting file info with another program like sox, works well :

    sox --info steps-stereo-16b-44khz.wav
    
    Input File     : 'steps-stereo-16b-44khz.wav'
    Channels       : 2
    Sample Rate    : 44100
    Precision      : 16-bit
    Duration       : 00:00:02.10 = 92608 samples = 157.497 CDDA sectors
    File Size      : 370k
    Bit Rate       : 1.41M
    Sample Encoding: 16-bit Signed Integer PCM
    
  • command line closes too fast inside python using ffmpeg

    18 février 2014, par user3012914

    I am using python to create a Macro Module inside MevisLab environment. I wanted to use ffmpeg in command prompt inside Python. The Problem is the command prompt closes too fast that i couldnt read while executing ffmpeg.

    My code

      t = 'ffmpeg -r'+str(ctx.field("FrameRate").value)+'-f image2 -pix_fmt yuv44p -s 1920x1080 -i'+path+'%0d.png -vodec libx264 -crf 15'+path+'.mp4'
    
    os.system(t)