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Sur d’autres sites (11950)

  • Android - JavaCV concat videos sound out of sync

    30 mars 2016, par Luis Reja

    Im developing an Android app using Android Studio. The app let you to record 4 videos (steps) from the device camera.

    Once the final step is recorded, the app concat all 4 videos using javacv (with ffmpeg).

    All working fine, I made a procedure to concat the videos, I concat step1 and step 2 to output12, then concat step3 and step4 to output34. Finally I concat output12 and output34. The video looks fine, but the sound is not sync.

    If I reproduce step1,step2,step3,step4,output12 and output34, the sound is sync.

    If I reproduce the final video the sounds is not sync.

    For your information, I reduce the quality video for test purposes.

    Any Clues ? I have been looking for information, but no results.

    This is the procedure to concat videos.

    protected void ConcatTwoVideos(String step1,String step2,String finalvideo){

    FrameGrabber grabber1 = new FFmpegFrameGrabber(Environment.getExternalStorageDirectory() + "/"+step1+".mp4");
    try {
         grabber1.start();
    } catch (FrameGrabber.Exception e) {
         e.printStackTrace();
    }

    FrameGrabber grabber2 = new FFmpegFrameGrabber(Environment.getExternalStorageDirectory() + "/"+step2+".mp4");
    try {
         grabber2.start();
    } catch (FrameGrabber.Exception e) {
         e.printStackTrace();
    }

    FrameRecorder recorder2 = new FFmpegFrameRecorder(Environment.getExternalStorageDirectory() + "/"+finalvideo+".mp4", grabber1.getImageWidth(), grabber1.getImageHeight(), grabber1.getAudioChannels());
    recorder2.setFrameRate(grabber1.getFrameRate());
    recorder2.setSampleFormat(grabber1.getSampleFormat());
    recorder2.setSampleRate(grabber1.getSampleRate());
    recorder2.setVideoOption("preset", "ultrafast");
    recorder2.setVideoCodec(avcodec.AV_CODEC_ID_H264);

    try {
       recorder2.start();
    } catch (FrameRecorder.Exception e) {
       e.printStackTrace();
    }
    videopaso++;
    publishProgress();
    Frame frame;
    int j = 0;
    try {
       while ((frame = grabber1.grabFrame()) != null) {
           j++;
           recorder2.record(frame);

       }
    } catch (FrameGrabber.Exception e) {
       e.printStackTrace();
    } catch (FrameRecorder.Exception e) {
       e.printStackTrace();
    }
    videopaso++;
    publishProgress();
    try {
       while ((frame = grabber2.grabFrame()) != null) {
           recorder2.record(frame);

       }
    } catch (FrameGrabber.Exception e) {
       e.printStackTrace();
    } catch (FrameRecorder.Exception e) {
       e.printStackTrace();
    }

    try {
       recorder2.stop();
       grabber2.stop();
       grabber1.stop();
    } catch (FrameRecorder.Exception e) {
       e.printStackTrace();
    } catch (FrameGrabber.Exception e) {
       e.printStackTrace();
    }
    }
  • How to play FFMPEG sound sample with OpenAL ?

    22 mars 2016, par Asim

    I am using FFMPEG to load Audio Video from File. It works with video, but I don’t know how to play audio samples.

    Here is my code to get audio samples :

    m_AdotimeBase = (int64_t(m_Adocdec_ctx->time_base.num) * AV_TIME_BASE) / int64_t(m_Adocdec_ctx->time_base.den);  
    if(!m_Adofmt_ctx)
    {
       //AfxMessageBox(L"m_timeBase");
       return FALSE ;
    }
    int64_t seekAdoTarget = int64_t(m_currFrame) * m_AdotimeBase;  


    if(av_seek_frame(m_Adofmt_ctx, -1, seekAdoTarget, AVSEEK_FLAG_ANY) < 0)
    {
       /*CString st;
       st.Format(L"%d",m_currFrame);
       AfxMessageBox(L"av_seek_frame "+st);*/
       m_currFrame = m_totalFrames-1;
       return FALSE ;
    }

    if ((ret = av_read_frame(m_Adofmt_ctx, &packet)) < 0)
           return FALSE;
    if (packet.stream_index == 0)      
    {
       ret = avcodec_decode_audio4(m_Adocdec_ctx, &in_AdeoFrame, &got_frame, &packet);
       if (ret < 0)
       {
           av_free_packet(&packet);
           return FALSE;
       }
    }

    My problem is I want to listen that sample using OPENAL.

    I would appreciate any tutorials or references on the subject.

  • FFMpeg - add sound to video that already contain sound

    24 mars 2016, par jacky brown

    Here is what I have : input1.avi - video that contain sounds. input2.avi - video that doesn’t contain sounds. music.mp3 - audio file.

    I want to add background music(music.mp3 file) to the video.

    C :\input1.avi -i C :\music.mp3 -shortest -c:v copy -c:a copy C :\output1.avi
    then output1.avi is the same as input1 - movie with sounds but without the background music (music.mp3)

    when I try to use the other file (video without sounds) :

    C :\input2.avi -i C :\music.mp3 -shortest -c:v copy -c:a copy C :\output2.avi
    then output2.avi is the same as input2 + it have the background music.

    I tried to execute this too :

    C:\ffmpeg\bin>ffmpeg -i C:\input.avi -i C:\music.mp3 -shortest -c:v copy -filter_ complex "[1]volume=1.5[1a];[0][1a]amerge[a]" -map 0:v -map "[a]" -ac 2 C:\output1.avi

    but got the next error messsage :

    ffmpeg version N-78949-g6f5048f Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 5.3.0 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
    enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
    ibilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopenc
    ore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --ena
    ble-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable
    -libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --ena
    ble-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx
    264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable
    -lzma --enable-decklink --enable-zlib
     libavutil      55. 19.100 / 55. 19.100
     libavcodec     57. 27.101 / 57. 27.101
     libavformat    57. 28.100 / 57. 28.100
     libavdevice    57.  0.101 / 57.  0.101
     libavfilter     6. 39.100 /  6. 39.100
     libswscale      4.  0.100 /  4.  0.100
     libswresample   2.  0.101 /  2.  0.101
     libpostproc    54.  0.100 / 54.  0.100
    Input #0, avi, from 'C:\output1.avi':
     Metadata:
       encoder         : Lavf57.28.100
     Duration: 00:02:05.76, start: 0.000000, bitrate: 450 kb/s
       Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658), yuv420p, 720
    x480 [SAR 1:1 DAR 3:2], 440 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
       Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16p, 128 k
    b/s
    [mp3 @ 00000000005abc20] Skipping 0 bytes of junk at 32370.
    Input #1, mp3, from 'C:\music.mp3':
     Metadata:
       title           : Broadcast News Package - News Intro
       artist          : After Effects News Template
     Duration: 00:01:57.89, start: 0.025057, bitrate: 194 kb/s
       Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
       Metadata:
         encoder         : Lavc56.26
    [Parsed_amerge_1 @ 0000000000610200] No channel layout for input 1
    [Parsed_amerge_1 @ 0000000000610200] No channel layout for input 2
    [AVFilterGraph @ 00000000005ddfe0] The following filters could not choose their
    formats: Parsed_amerge_1
    Consider inserting the (a)format filter near their input or output.
    Error configuring complex filters.
    I/O error

    So why input1 does not contain the background music ? and how can I decrease or increase the volume of music.mp3 file ?