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• 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

• ImgFlip Gif Generator

  30 juillet 2013, par user1960364

  Reference : http://imgflip.com/gifgenerator

  So, I don't need the web UI and all the cusomizations. But I've been trying to figure out how to slice and convert a mp4 or mov to an animated gif automagically. I've looked at several ways of doing it with imagemagick/ffmpeg but none of them rendered results as good as imgflip. I always end up with some 24bit color-looking gif instead or some impractically large file.

  Any suggestions/ideas on how to do it as efficiently as imgflip ?

  Thanks :)

  — Edit —
  I found this on reddit :

  #!/bin/bash
  
  
  
  # Extracts each frame of the video as a single gif
  
  ffmpeg -i $1 out%04d.gif
  
  
  
  # Combines all the frames into one very nicely animated gif.
  
  convert -delay 4 out*.gif anim.gif
  
  
  
  # Optimizes the gif using imagemagick
  
  convert -layers Optimize anim.gif optimized_output.gif
  
  
  
  # Cleans up the leftovers
  
  rm out*
  
  rm anim.gif 

  Using that, I created (and yes, I'm on windows) :

  ffmpeg -i v.mp4 -r 10 -q:v 1 tmp/out%04d.jpg
  
  convert -delay 10 tmp/out*.jpg jpg-d10.gif
  
  echo y | del tmp

  And got decent quality, even this creates an 8MB file for a small 6sec 480x480 clip at 10fps. And because it's a video and not vector or series of images with very few colors, extracting jpg images does better compression than -layers Optimize. However, for some reason, -qscale/-q:v on the frame extraction does not effect the final file size, just the quality.

• Converting .3gp file into mp4 file in android using ffmpeg

  8 août 2013, par user2171513

  I want to convert .3gp file into .mp4 file with resolution modified in Android using ffmpeg.
  I want to increase the resolution of the video from its standard resolution to 1920x1080.

  So far I have been successful in
  1) extracting .h264 video file from .3gp file and increase its resolution
  2) extracting .aac audio file from .3gp file.

  Now I want to combine them back into .mp4 file. The commands that I have used to extract this .h264 and .aac files are :

  ./ffmpeg -i 1.3gp -vbsf h264_mp4toannexb -s 1920x1080 1.h264
  
  ./ffmpeg -i 1.3gp -ab 160k -ac 2 -ar 48000 -vn -strict -2  1.aac

  The command that I have tried to merge them back is

  ./ffmpeg -i 1.h264 -i 1.aac -map 0:0 -map 1:0 -strict -2 1.mp4

  The 1.mp4 that gets generated at the end basically has audio only at few sync frames of video. (Thats what I feel , because the audio is present at specific intervals within the video)

  Can anyone please help in figuring out what am I missing here.

  EDIT :
  So basically I want to concat 4 different videos of 4 different resolution and type.

  1)

  ./ffmpeg -i 1.mp4 
  
  Video: h264 (High), yuv420p, 1920x1080, 16959 kb/s, 29.85 fps, 90k tbr, 90k tbn, 180k tbc
  
  Audio: aac, 48000 Hz, stereo, s16, 106 kb/s

  2)

  ffmpeg -i 2.mp4
  
  Video: h264 (Constrained Baseline), yuv420p, 640x480, 3102 kb/s, 29.99 fps, 90k tbr, 90k tbn, 180k tbc
  
  Audio: aac, 48000 Hz, stereo, s16, 93 kb/s

  3)

  ffmpeg -i 3.3gp
  
  Video: h263, yuv420p, 1408x1152 [PAR 12:11 DAR 4:3], 2920 kb/s, 15 fps, 15 tbr, 15360 tbn, 29.97 tbc
  
  Audio: amrnb, 8000 Hz, 1 channels, flt, 12 kb/s

  4)

  ffmpeg -i 4.3gp
  
  Video: h264 (High), yuv420p, 352x288 [PAR 12:11 DAR 4:3], 216 kb/s, 24 fps, 24 tbr, 24 tbn, 48 tbc
  
  Audio: aac, 44100 Hz, stereo, s16, 92 kb/s

  So I am converting them to mpegts using following commands

  ./ffmpeg -i 1.mp4 -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 1.ts
  
  ./ffmpeg -i 2.mp4 -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 2.ts
  
  ./ffmpeg -i 3.3gp -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 3.ts
  
  ./ffmpeg -i 4.3gp -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 4.ts

  then concatenating the .ts files into f.ts and then creating a final .mp4 file from it using

   cat 1.ts 2.ts 3.ts 4.ts > f.ts
  
  ./ffmpeg -i f.ts -c copy -bsf:a aac_adtstoasc output.mp4

  But my f.ts also doesnt seem to play correctly in VLC on linux, it plays first 2 mp4's video + audio and it plays last .3gp's audio only.(Same for output.mp4 too) Could you please help me in figuring out what am I missing ?

  Thanks