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  • Adding audio to video without re encoding [migrated]

    4 mars 2015, par user1503606

    I am looking to add audio to a video without having to re encode the two pieces.

    I have looked at a few questions on stack overflow followed the answers and i still cant seemed to get it to work, if anyone can spot what i am missing please help.

    I am running this code.

    ffmpeg -i DJ_Mes_Rescue-Some_Day-Guesthouse_Music.mp3 -i output.mp4 -map 0:0 -map 1:0 -acodec copy -vcodec copy -shortest edit2.mp4

    Now from what i understand ffmpeg is mapping the stream over the top ffmpeg info for the audio track is.

    Input #0, mp3, from 'DJ_Mes_Rescue-Some_Day-Guesthouse_Music.mp3':
     Metadata:
       encoder         : LAME 64bits version 3.98.4 (http://www.mp3dev.org/)
       title           : Some Day
       artist          : DJ Mes, Rescue
       TLEN            : 378750
       genre           : House
       track           : 1/0
       date            : 2015
     Duration: 00:06:18.80, start: 0.025056, bitrate: 320 kb/s
       Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
       Metadata:
         encoder         : LAME3.98r
       Side data:
         replaygain: track gain - -10.100000, track peak - unknown, album gain - unknown, album peak - unknown,
       Stream #0:1: Video: png, rgb24, 225x225, 90k tbr, 90k tbn, 90k tbc
       Metadata:
         title           :
         comment         : Other

    And for the video it is.

    Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       encoder         : Lavf56.15.102
     Duration: 00:00:49.46, start: 0.046440, bitrate: 566 kb/s
       Stream #1:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 568x320, 462 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
       Metadata:
         rotate          : 90
         handler_name    : VideoHandler
       Side data:
         displaymatrix: rotation of -90.00 degrees
       Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 96 kb/s (default)
       Metadata:
         handler_name    : SoundHandler

    So i am mapping the stream 0:0 from the audio file to 1:0 from the video file, but all i seem to get is a video with no audio.

    Can someone help ?

    UPDATE whole output added

    ffmpeg -i Saison-Please_Don%27t_Go-Guesthouse_Music.mp3 -i output.mp4 -c copy -map 0:0 -map 1:0 -shortest mixed.mp4
    ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers
     built on Jan 12 2015 10:15:06 with Apple LLVM version 6.0 (clang-600.0.56) (based on LLVM 3.5svn)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/2.5.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libfreetype --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-aacenc --enable-libass --enable-ffplay --enable-libspeex --enable-libschroedinger --enable-libfdk-aac --enable-libopus --enable-frei0r --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.1_1/include/openjpeg-1.5 ' --enable-nonfree --enable-vda
     libavutil      54. 15.100 / 54. 15.100
     libavcodec     56. 13.100 / 56. 13.100
     libavformat    56. 15.102 / 56. 15.102
     libavdevice    56.  3.100 / 56.  3.100
     libavfilter     5.  2.103 /  5.  2.103
     libavresample   2.  1.  0 /  2.  1.  0
     libswscale      3.  1.101 /  3.  1.101
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  3.100 / 53.  3.100
    Input #0, mp3, from 'Saison-Please_Don%27t_Go-Guesthouse_Music.mp3':
     Metadata:
       encoder         : LAME 64bits version 3.98.4 (http://www.mp3dev.org/)
       title           : Please Donât Go
       artist          : Saison
       TLEN            : 408492
       genre           : House
       track           : 1/0
       date            : 2015
     Duration: 00:06:48.53, start: 0.025056, bitrate: 320 kb/s
       Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
       Metadata:
         encoder         : LAME3.98r
       Side data:
         replaygain: track gain - -8.400000, track peak - unknown, album gain - unknown, album peak - unknown,
       Stream #0:1: Video: png, rgb24, 225x225, 90k tbr, 90k tbn, 90k tbc
       Metadata:
         title           :
         comment         : Other
    Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       encoder         : Lavf56.15.102
     Duration: 00:00:49.46, start: 0.046440, bitrate: 566 kb/s
       Stream #1:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 568x320, 462 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
       Metadata:
         rotate          : 90
         handler_name    : VideoHandler
       Side data:
         displaymatrix: rotation of -90.00 degrees
       Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 96 kb/s (default)
       Metadata:
         handler_name    : SoundHandler
    Output #0, mp4, to 'mixed.mp4':
     Metadata:
       date            : 2015
       title           : Please Donât Go
       artist          : Saison
       TLEN            : 408492
       genre           : House
       track           : 1/0
       encoder         : Lavf56.15.102
       Stream #0:0: Audio: mp3 (i[0][0][0] / 0x0069), 44100 Hz, stereo, 320 kb/s
       Metadata:
         encoder         : LAME3.98r
       Side data:
         replaygain: track gain - -8.400000, track peak - unknown, album gain - unknown, album peak - unknown,
       Stream #0:1(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 568x320, q=2-31, 462 kb/s, 30 fps, 15360 tbn, 15360 tbc (default)
       Metadata:
         rotate          : 90
         handler_name    : VideoHandler
       Side data:
         displaymatrix: rotation of -90.00 degrees
    Stream mapping:
     Stream #0:0 -> #0:0 (copy)
     Stream #1:0 -> #0:1 (copy)
    Press [q] to stop, [?] for help
    frame= 1481 fps=0.0 q=-1.0 Lsize=    4759kB time=00:00:49.34 bitrate= 790.0kbits/s    
    video:2785kB audio:1929kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.960839%

    ANOTHER UPDATE :

    So i have tried the following

    ffmpeg -i 3.mp4 -i Saison-Please_Don%27t_Go-Guesthouse_Music.mp3 -i Saison-Please_Don%27t_Go-Guesthouse_Music.mp3 -map 0:0 -map 0:1 -map 1:0 -map 2:0 -c:v copy -c:a copy 23.mp4

    and if i run.

    ffmpeg -i 23.mp4

    You can see the streams/audio has been added but it is not playing ?

    Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       encoder         : Lavf56.15.102
     Duration: 00:06:48.53, start: 0.025057, bitrate: 713 kb/s
       Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 568x320, 462 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
       Metadata:
         rotate          : 90
         handler_name    : VideoHandler
       Side data:
         displaymatrix: rotation of -90.00 degrees
       Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 96 kb/s (default)
       Metadata:
         handler_name    : SoundHandler
       Stream #0:2(und): Audio: mp3 (mp4a / 0x6134706D), 44100 Hz, stereo, s16p, 319 kb/s
       Metadata:
         handler_name    : SoundHandler
       Stream #0:3(und): Audio: mp3 (mp4a / 0x6134706D), 44100 Hz, stereo, s16p, 319 kb/s
       Metadata:
         handler_name    : SoundHandler

    ONE MORE UPDATE THIS SEEMS TO WORK :

    ffmpeg -i 3.mp4 -i Saison-Please_Don%27t_Go-Guesthouse_Music.mp3 -map 0:0 -map 0:1 -map 1:0  -c:v copy -c:a copy 234.mp4 && ffmpeg -i 234.mp4 -map 0:0 -map 0:2 -acodec copy -vcodec copy new_file3.mp4
  • fate : gapless : remove useless tests

    22 avril 2015, par wm4
    fate : gapless : remove useless tests
    

    These could be kept, but they are not overly useful. The only thing they
    had over the remaining mp3 gapless test was seeking, which was incorrect
    in the toc test, and only by chance correct in the notoc test.

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] tests/fate/gapless.mak
    • [DH] tests/ref/fate/gapless-mp3-notoc
    • [DH] tests/ref/fate/gapless-mp3-toc
  • Is there a way to use ffmpeg audio filters to automatically synchronize 2 streams with similar content

    29 mai 2015, par user3741412

    I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a ’line in’ jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.

    The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.

    So I can do this today :

    • Take the good sound and the camera captured sound and load each into
      audacity and pretty quickly use the timeshift toot to perfectly fit
      the good audio to the questionable audio from the HDMI capture and
      cut the good audio to the exact size of the video. Then I can use
      ffmpeg or other video editing software to replace the questionable
      audio with the better audio.

    But while somewhat quick and easy, it always carries with it a bit of human error and time. I’d like to automate this if possible as this process is repeated at least weekly throughout the year.

    Does anyone have a suggestion if any of these ideas have merit or could suggest another approach ?

    1. I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn’t see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats ?

      • if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now— I’ve never tried either of these filters yet— just trying to optimize against blind alleys by asking for advice up front.
    2. If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I’d take it, if it turns out reasonably reliable

    3. Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum ? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible— but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.

    4. any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI ?

    Thanks in advance for any pointers or suggestinons !