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问题:
I need to convert audio files to mp3 using ffmpeg.
When i write the command as ffmpeg -i audio.ogg -acodec mp3 newfile.mp3
, I get the error:
FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration:
libavutil 49.15. 0 / 49.15. 0
libavcodec 52.20. 1 / 52.20. 1
libavformat 52.31. 0 / 52.31. 0
libavdevice 52. 1. 0 / 52. 1. 0
built on Jun 24 2010 14:56:20, gcc: 4.4.1
Input #0, mp3, from 'ZHRE.mp3':
Duration: 00:04:12.52, start: 0.000000, bitrate: 208 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 256 kb/s
Output #0, mp3, to 'audio.mp3':
Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, s16, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0
I also ran this command :
ffmpeg -formats | grep mp3
and got this in response:
FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration:
libavutil 49.15. 0 / 49.15. 0
libavcodec 52.20. 1 / 52.20. 1
libavformat 52.31. 0 / 52.31. 0
libavdevice 52. 1. 0 / 52. 1. 0
built on Jun 24 2010 14:56:20, gcc: 4.4.1
DE mp3 MPEG audio layer 3
D A mp3 MP3 (MPEG audio layer 3)
D A mp3adu ADU (Application Data Unit) MP3 (MPEG audio layer 3)
D A mp3on4 MP3onMP4
text2movsub remove_extra noise mov2textsub mp3decomp mp3comp mjpegadump imxdump h264_mp4toannexb dump_extra
I guess that the mp3 codec isnt installed. Am I right here ? Can anyone help me out here ?
回答1:
A very complete list of ffmpeg arguments can be found here:
http://www.catswhocode.com/blog/19-ffmpeg-commands-for-all-needs
So you could use this command:
ffmpeg -i input.wav -vn -ar 44100 -ac 2 -ab 192k -f mp3 output.mp3
Explanation of the used arguments in this example:
-i
- input file
-vn
- Disable video, to make sure no video is included if the source would be a video file
-ar
- Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
-ac
- Set the number of audio channels. For output streams it is set by default to the number of input audio channels. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options. So used here to make sure it is stereo (2 channels)
-ab
- actually seems to be changed, so should be replaced for newer ffmpeg version to -b:a 192k
Converts the audio bitrate to be exact 192kbit per second
-f
- Force input or output file format. The format is normally auto detected for input files and guessed from the file extension for output files, so this option is not needed in most cases.
回答2:
1) wav to mp3
ffmpeg -i audio.wav -acodec libmp3lame audio.mp3
2) ogg to mp3
ffmpeg -i audio.ogg -acodec libmp3lame audio.mp3
3) ac3 to mp3
ffmpeg -i audio.ac3 -acodec libmp3lame audio.mp3
4) aac to mp3
ffmpeg -i audio.aac -acodec libmp3lame audio.mp3
Get all ffmpeg conversion formats here.
回答3:
Never mind,
I am converting my audio files to mp2 by using the command:
ffmpeg -i input.wav -f mp2 output.mp3
This command works perfectly.
I know that this actually converts the files to mp2 format, but then the resulting file sizes are the same..
回答4:
I had to purge my ffmpeg and then install another one from a ppa:
sudo apt-get purge ffmpeg
sudo apt-add-repository -y ppa:jon-severinsson/ffmpeg
sudo apt-get update
sudo apt-get install ffmpeg
Then convert:
ffmpeg -i audio.ogg -f mp3 newfile.mp3
回答5:
For batch processing with files in folder aiming for 190 VBR and file extension = .mp3 instead of .ac3.mp3 you can use the following code
Change .ac3 to whatever the source audio format is.
ffmpeg mp3 settings
for f in *.ac3 ; do ffmpeg -i "$f" -acodec libmp3lame -q:a 2 "${f%.*}.mp3"; done
回答6:
For batch processing files in folder:
for i in *.wav; do ffmpeg -i "$i" -f mp3 "${i%}.mp3"; done
This script converts all "wav" files in folder to mp3 files and adds mp3 extension
ffmpeg have to be installed. (See other answers)
回答7:
https://trac.ffmpeg.org/wiki/Encode/MP3
VBR Encoding:
ffmpeg -vn -ar 44100 -ac 2 -q:a 1 -codec:a libmp3lame output.mp3
回答8:
Try FFmpeg Static Build Link
Documentation: https://www.johnvansickle.com/ffmpeg/
Host the static build on your server in same directory
$ffmpeg = dirname(__FILE__).'/ffmpeg';
$command = $ffmpeg.'ffmpeg -i audio.ogg -acodec libmp3lame audio.mp3';
shell_exec($command);
回答9:
If you have a folder and sub-folder full of wav's you want to convert, put below command in a file, save it in a .bat file in the root of the folder where you wan to convert, and then run the bat file
for /R %%g in (*.wav) do start /b /wait "" "C:\ffmpeg-4.0.1-win64-static\bin\ffmpeg" -threads 16 -i "%%g" -acodec libmp3lame "%%~dpng.mp3" && del "%%g"
回答10:
High quality for Mac OS works perfectly!
ffmpeg -i input.wma -q:a 0 output.mp3