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• Convert Videos with FFMPEG to PowerPoint 2016 compatible video format [closed]

  11 septembre 2020, par Sebastian S.

  I am trying to convert a bunch of videos to a video format that is natively supported by PowerPoint 2013/2016 on a Windows 7 system.

  


  Microsoft recommends on their website mp4 with h264 and aac.

  


  Video and audio file formats supported in PowerPoint

  


  


  In PowerPoint 2013 and later, and in PowerPoint 2016 for Mac, for the best video playback experience, we recommend that you use .mp4 files encoded with H.264 video (a.k.a. MPEG-4 AVC) and AAC audio. In PowerPoint 2010, we recommend that you use .wmv files.

  


  


  


  We recommend using .m4a files encoded with AAC audio. In PowerPoint 2010, we recommend that you use .wav or .wma files.

  


  


  Audio is not important for me.
I tried to convert my videos with ffmpeg using the following options :

  


  ffmpeg -i Input.avi -c:v libx264 -preset slow -crf 22 -c:a copy Output.mp4


  


  However I cannot import the video to PowerPoint 2016 (32 or 64bit, I tried both). I always get a missing codec error.

  


  PPT Error when including video files

  


  Has anyone successfully encoded videos to a natively supported PowerPoint video format (on Windows) ?

  


• not able to convert a specific .wav file to mp3 or m4a with sox, avconv

  25 juillet 2017, par astrograph

  At the office we have a project where we apply IoT technologies to a real bee hive.

  One of the features is to detect specific sounds the bees make when a new queen hatches. We have a special microphone in place, the algorithm is also implemented. For now we get a lot of false positives, and want to quickly be able to identify them, by listening to the audio files in the browser. Therefore I want to convert the .wav files to either .mp3 or .m4a

  The .wav file format seems to be quite strange, as I was not able to convert it to mp3 using avconv, sox or even audacity. The funny thing is, the Microsoft media player can play the .wav file fine.

  Here is the information soxi gives about the wav file :

  pi@raspberrypi:~ $ soxi Channel1.wav
  
  soxi WARN wav: wave header missing extended part of fmt chunk
  
  
  
  Input File     : 'Channel1.wav'
  
  Channels       : 1
  
  Sample Rate    : 6250
  
  Precision      : 24-bit
  
  Duration       : 00:01:21.00 = 506250 samples ~ 6075 CDDA sectors
  
  File Size      : 2.03M
  
  Bit Rate       : 200k
  
  Sample Encoding: 32-bit Floating Point PCM

  This is the avconv command I am trying to use :

  avconv -y -v quiet -i Channel1.wav -strict experimental -ar 44100 -ab 160k Channel1.m4a

  I also tried with sox :

  sox  -v 0.60 Channel1.wav -r 22050 Channel1.m4a

  but the output is mostly silent, with some random noise.

  The question is how can a wav file like this : https://drive.google.com/open?id=0B9YVh-jkOMLsQThERlI2emN2QWM be converted to an audio format using a raspberry pi that can be played in the browser ?

• FFMPEG ALSA xrun crash

  13 décembre 2017, par Liam Martens

  I’m running a YouTube RTMP stream using FFMPEG with x11grab and an alsa loopback device but sometimes after let’s say 20 hours there is an ALSA xrun and then the ffmpeg command crashes, but I’m not sure why or how this happens. (mind you the ffmpeg command does not run continuously it gets restarted automatically every so often, but the xrun makes the command crash causing the stream to go offline sometimes because a crash restart is not fast enough)

  I’m using thread_queue_size and I’ve even manually compiled ffmpeg with a higher ALSA BUFFER SIZE, but the issue appears to persist still. Besides this I’ve also scoured many posts with people having similar issues but these never really seem to end up resolved.

  This is the stream command

  ffmpeg -loglevel verbose -f alsa -thread_queue_size 12288 -ac 2 -i hw:Loopback,1,0 \
  
           -probesize 10M -f x11grab -field_order tt -thread_queue_size 12288 -video_size 1280x720 -r 30 -i :1.1 \
  
          -c:v libx264 -c:a libmp3lame -shortest -tune fastdecode -tune zerolatency \
  
          -crf 26 -pix_fmt yuv420p -threads 0 -maxrate 2500k -bufsize 2500k -pass 1 -af aresample=async=1 \
  
          -movflags +faststart -flags +global_header -preset ultrafast -r 30 -g 60 -b:v 2000k -b:a 192k -ar 44100 \
  
          -f flv -rtmp_live live rtmp://a.rtmp.youtube.com/live2/{KEY}

  Log excerpt

  ffmpeg version N-89463-gc7a5e80 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
  
    configuration: --prefix=/usr --enable-avresample --enable-avfilter --enable-gpl --enable-libmp3lame --enable-librtmp --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libtheora --enable-postproc --enable-pic --enable-pthreads --enable-shared --disable-stripping --disable-static --enable-vaapi --enable-libopus --enable-libfreetype --enable-libfontconfig --enable-libpulse --disable-debug
  
    libavutil      56.  5.100 / 56.  5.100
  
    libavcodec     58.  6.103 / 58.  6.103
  
    libavformat    58.  3.100 / 58.  3.100
  
    libavdevice    58.  0.100 / 58.  0.100
  
    libavfilter     7.  7.100 /  7.  7.100
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, alsa, from 'hw:Loopback,1,0':
  
    Duration: N/A, start: 1513163617.594224, bitrate: 1536 kb/s
  
      Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
  
  Input #1, x11grab, from ':1.1':
  
    Duration: N/A, start: 1513163617.632434, bitrate: N/A
  
      Stream #1:0: Video: rawvideo, 1 reference frame (BGR[0] / 0x524742), bgr0(top first), 854x480, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
  
  Parsing...
  
  Parsed protocol: 0
  
  Parsed host    : a.rtmp.youtube.com
  
  Parsed app     : live2
  
  RTMP_Connect1, ... connected, handshaking
  
  HandShake: Type Answer   : 03
  
  HandShake: Server Uptime : 0
  
  HandShake: FMS Version   : 4.0.0.1
  
  HandShake: Handshaking finished....
  
  RTMP_Connect1, handshaked
  
  Invoking connect
  
  HandleServerBW: server BW = 2500000
  
  HandleClientBW: client BW = 10000000 2
  
  HandleChangeChunkSize, received: chunk size change to 256
  
  RTMP_ClientPacket, received: invoke 240 bytes
  
  (object begin)
  
  Property: 
  
  Property: 
  
  Property: 
  
  (object begin)
  
  Property: 3,5,3,824>
  
  Property: 
  
  Property: 
  
  (object end)
  
  Property: 
  
  (object begin)
  
  Property: 
  
  Property: 
  
  Property: 
  
  Property: 
  
  Property: 
  
  (object begin)
  
  Property: 
  
  (object end)
  
  (object end)
  
  (object end)
  
  HandleInvoke, server invoking <_result>
  
  HandleInvoke, received result for method call <connect>
  
  Invoking releaseStream
  
  Invoking FCPublish
  
  Invoking createStream
  
  RTMP_ClientPacket, received: invoke 21 bytes
  
  (object begin)
  
  Property: 
  
  Property: 
  
  Property: NULL
  
  (object end)
  
  HandleInvoke, server invoking <onbwdone>
  
  Invoking _checkbw
  
  RTMP_ClientPacket, received: invoke 29 bytes
  
  (object begin)
  
  Property: 
  
  Property: 
  
  Property: NULL
  
  Property: 
  
  (object end)
  
  HandleInvoke, server invoking &lt;_result>
  
  HandleInvoke, received result for method call <createstream>
  
  Invoking publish
  
  RTMP_ClientPacket, received: invoke 73 bytes
  
  (object begin)
  
  Property: 
  
  Property: 
  
  Property: NULL
  
  Property: 
  
  (object begin)
  
  Property: 
  
  Property: 
  
  (object end)
  
  (object end)
  
  HandleInvoke, server invoking <onstatus>
  
  HandleInvoke, onStatus: NetStream.Publish.Start
  
  Stream mapping:
  
    Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  
    Stream #0:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [graph 0 input from stream 1:0 @ 0x5607d087e060] w:854 h:480 pixfmt:bgr0 tb:1/30 fr:30/1 sar:0/1 sws_param:flags=2
  
  [auto_scaler_0 @ 0x5607d087d800] w:iw h:ih flags:'bicubic' interl:0
  
  [format @ 0x5607d087ed40] auto-inserting filter 'auto_scaler_0' between the filter 'Parsed_null_0' and the filter 'format'
  
  [auto_scaler_0 @ 0x5607d087d800] w:854 h:480 fmt:bgr0 sar:0/1 -> w:854 h:480 fmt:yuv420p sar:0/1 flags:0x4
  
  [swscaler @ 0x5607d0880260] Warning: data is not aligned! This can lead to a speed loss
  
  [libx264 @ 0x5607d08684e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x5607d08684e0] profile Constrained Baseline, level 3.1
  
  [libx264 @ 0x5607d08684e0] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=60 keyint_min=6 scenecut=0 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=26.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1500 vbv_bufsize=1500 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=0
  
  [graph_1_in_0_0 @ 0x5607d091c840] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3
  
  [Parsed_aresample_0 @ 0x5607d0916b40] ch:2 chl:stereo fmt:s16 r:48000Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
  
  Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/{KEY}':
  
    Metadata:
  
      encoder         : Lavf58.3.100
  
      Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(top coded first (swapped)), 854x480, q=-1--1, 1000 kb/s, 30 fps, 1k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc58.6.103 libx264
  
      Side data:
  
        cpb: bitrate max/min/avg: 1500000/0/1000000 buffer size: 1500000 vbv_delay: -1
  
      Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, s16p, delay 1105, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc58.6.103 libmp3lame
  
  frame=   29 fps=0.0 q=17.0 size=     146kB time=00:00:00.94 bitrate=1267.3kbits/s speed=1.86x    
  
  frame=   44 fps= 44 q=18.0 size=     168kB time=00:00:01.46 bitrate= 942.4kbits/s speed=1.45x    
  
  frame=   60 fps= 40 q=16.0 size=     191kB time=00:00:01.96 bitrate= 794.8kbits/s speed= 1.3x    
  
  ...
  
  frame= 2740 fps= 30 q=17.0 size=    7993kB time=00:01:31.32 bitrate= 717.0kbits/s speed=   1x    
  
  frame= 2755 fps= 30 q=18.0 size=    8013kB time=00:01:31.82 bitrate= 714.9kbits/s speed=   1x    
  
  [alsa @ 0x5607d084d7e0] ALSA buffer xrun.
  
  </onstatus></createstream></onbwdone></connect>