avconv - a very fast video and audio converter

for audioFile in *wav; do avconv -i "$audioFile" -af aformat=s16:44100 "${audioFile%.*}.flac"; done

This encodes to 16 bits at 44100 hertz.

List available codecs :

avconv -codecs
V : video codec
A : audio codec
S : subtitle codec

This command returns a failure exit code as avconv expects an "output file" parameter (I found no better solution so far ). For this reason ("failure" exit code + output message sent to stderr rather than stdout), you have to redirect everything to stdout before processing it :

for a single file :

avconv -i file 2>&1 | grep 'Stream'

for multiple files :

streamType='Audio'; for mediaFile in *; do echo "$mediaFile"; avconv -i "$mediaFile" 2>&1 | grep -E "Stream.*$streamType"; done

1. "open" inputFile to do stuff with it
2. "select" data starting at startTime of inputFile
3. make this selection last for duration
4. and copy the selection into outputFile

avconv -i inputFile -ss startTime -t duration -codec copy outputFile

• startTime and duration are in either a number of seconds, or in the hh:mm:ss[.xxx] format.
• If no duration is specified, the process will continue until the end of the input file.

The command above may work pretty well. But, depending on the input file itself (I got this with a .avi file), it may end in a huge series of :

Re-encoding the audio track (here in mp3 format) fixed it :

During the tests to find the right start time (-ss value), save time by not processing the input until the end :

TESTING :
outputDir='./output'; mkdir -p "$outputDir"; for input in *avi; do echo "$input"; avconv -y -i "$input" -ss 15.5 -t 15 -c:v copy -c:a mp3 "$outputDir/$input"; done

REAL :
outputDir='./output'; mkdir -p "$outputDir"; for input in *avi; do echo "$input"; avconv -i "$input" -ss 15.5 -c:v copy -c:a mp3 "$outputDir/$input"; done

• one file : avconv -i movie.mp4 -codec:a libmp3lame audio.mp3
• more files :
  • for videoFile in *mp4; do avconv -i "$videoFile" -codec:a libmp3lame "$videoFile".mp3; done
  • outputDir='./audio'; mkdir -p "$outputDir"; for videoFile in *mp4; do avconv -i "$videoFile" -codec:a libmp3lame -b:a 256k "$outputDir/${videoFile%.*}.mp3"; done

• one file : avconv -i movie.mp4 -codec:a libvorbis audio.ogg
• more files : outputDir='./ogg'; mkdir -p "$outputDir"; for sourceFile in *mp4; do avconv -i "$sourceFile" -codec:a libvorbis "$outputDir/${sourceFile%.*}.ogg"; done

This method should be preferred whenever possible since it only extracts the audio data with no reconversion :

• preserves the original audio quality
• doesn't produce an artificial "high quality" audio file from a lower quality source (which is what websites offering to download music from YouTube as MP3 do)
• best possible audio quality vs file size ratio

1. identify the audio track :

   Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 389 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc (default)
   Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)

2. copy it into a new file :
   avconv -i movie.mp4 -map 0:1 -c copy audio.mp4
3. you can make sure this file as only 1 audio track :

   Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)

4. Rename the audio file based on the type of audio format :
   mv audio.{mp4,aac}

From any format to MPEG 1 :

avconv -i input.mp4 output.mpg

From any format to MPEG 2 :

avconv -i input.mp4 -f dvd output.mpg

From any format to H.264 inside an MP4 container :

avconv -i input.mp4 -strict experimental -vcodec libx264 -deinterlace -threads 0 output.mp4

-strict experimental was suggested by the software :

encoder 'aac' is experimental and might produce bad results.
Add '-strict experimental' if you want to use it.

From any format, convert video to H.264 + leave audio as-is, and save the result into an MP4 container :

avconv -i input.mp4 -threads auto -c:a copy -vcodec libx264 output.mp4

• Videos shot with a smartphone have a Rotation metadata parameter. It is used by the built-in player to orient the video properly (source).
• Looks like VLC ignores or misinterprets this parameter (?)
• Despite all my efforts, I've not been able to rotate a video made with an Android phone with avconv. Neither did others.
• OpenShot did it very well with a few clicks.
• Converting videos can be HARD on CPU, cause overheating (?) and make the system hang To workaround this, considering adjusting manually the CPU frequency.

90° :

https://stackoverflow.com/questions/3937387/rotating-videos-with-ffmpeg/9570992#9570992
	ffmpeg -i in.mov -vf "transpose=1" out.mov

For the transpose parameter you can pass:

0 = 90Counterclockwise and Vertical Flip (default)
1 = 90Clockwise
2 = 90Counterclockwise
3 = 90Clockwise and Vertical Flip

https://ffmpeg.org/ffmpeg-filters.html#transpose-1

180° (source) :

avconv -i inputFile -vf scale=iw*0.25:-1 -codec:a copy outputFile

-codec:a copy allows to stream copy the audio without re-encoding.