How can I decode a FLV's audio if it's recorded from a live stream using Flash Media Server and uses NellyMoser codec?
I'm writing a script that process several FLVs, using FFmpeg, so I need a command line solution.
Any ideas?
This should work for you, since NellyMoser is supported by FFmpeg.
1. Using mp3
ffmpeg -i yourinput.flv -vn -acodec libmp3lame output.flv
2. Using AAC (switch aac with libfaac depending on which you have loaded)
ffmpeg -i yourinput.flv -vn -acodec libfaac output.mp4
I'm assuming of course you dont care about video.
Related
If I convert an audio file with an album cover thank's to:
ffmpeg -i sample1.flac -ar 48000 -vn -c:a libvorbis -b:a 320k sample1.ogg
My sample1.ogg file doesn't have any album cover. Is there a way to ask explicitly to ffmpeg to keep the cover ?
instead "-vn" write "-c:v libtheora -q:a 10".
-vn - means no video, picture is frame of video in ffmpeg
Sum up of answers I found: Remove the option -vn which means: no video, because thumbnail are frame of video. Use libtheora instead of libvorbis. If you get a bitrate error, remove option -b:a 320k. At the end we get:
ffmpeg -i file.flac -c:v libtheora -q:v 10 -c:a libvorbis file.ogg
This gives you a file with an audio and a video content.
If you prefer to extract thumbnail in a separate file (to re-add it later for example), use:
ffmpeg -i file.flac -an -vcodec copy thumbnail.jpg
Thank's to Баяр Гончикжапов for his help and answers!
I've been using ffmpeg convert audio from one format to another and to change audio's bitrate. When I try to convert aac audio to mp3 audio using the command:
ffmpeg -i SomeAudio.aac -c:a mp3 -b:a 128k SomeOutputPath.mp3
everything works correctly and output audio is of the same length as the input audio (6 minutes, 15 seconds).
However, when I try converting it to aac audio using a similar command:
ffmpeg -i SomeAudio.aac -c:a aac -b:a 128k SomeOutputPath.aac
it makes the output audio longer (around 10 minutes). I have tried specifying output length but that still makes the video longer, it just cuts of part of the audio:
ffmpeg -i SomeAudio.aac -c:a aac -b:a 128k -t 00:06:15 SomeOutputPath.aac
Here is a link to the screenshot:
My suspicion is that message "Estimating duration from bitrate, this may be innacurate" (the one in the screenshot) is the root of my problem but I just haven't been able to find any useful information about it on the web.
Thanks a lot for any help in advance :)
The duration shown for raw AAC is a guess because it does not contain duration info. You can find the actual duration with:
ffmpeg -i input.aac -f null -
Or a faster, "close enough" method:
ffmpeg -i input.aac -c copy -f null -
Workaround is to remux to M4A:
ffmpeg -i input.aac -c copy output.m4a
I have actually encoded an audio file from ac3 to aac using ffmpeg native aac encoder but the issue is that the file is not playing correctly , more specifically i have played that file in different media player but most of them start from 19 seconds and in vlc it is not even starting till I seek to more than 19 seconds duration.
command i have used is :-
ffmpeg -i source.mkv -map 0:a:0 -c:a aac audio.mp4.
That is the proper way.
Don't know if this will make a difference, try -b:a 400k and -strict experimental.
If you want audio only, convert to m4a or aac.
ffmpeg -i input.mkv -y -c:a aac -b:a 400k -map 0:a:0? -strict experimental output.mp4
Other encoders, may require compiling ffmpeg with use flags:
http://trac.ffmpeg.org/wiki/Encode/AAC
libfdk_aac
libfaac
I have a program generating a bunch of raw H264 frames and would like to place that into a mp4 container for streaming.
Anyone know how to do that?
I was thinking I'd use ffmpeg however, this needs to be used commercially and it seems like ffmpeg can only do this through it's x264 library... which uses a GPL license.
Thank you!
If you're looking for the FFMPEG command line to do that, then try the following:
ffmpeg -i "source.h264" -c:v copy -f mp4 "myOutputFile.mp4"
If you have a separate audio file you can add it too:
ffmpeg -i "source.h264" -i "myAudio" -c:v copy -c:a copy -f mp4 "myOutputFile.mp4"
If your audio needs to be encoded as well (for instance codec AAC-LC, bitrate 256kbps):
ffmpeg -i "source.h264" -i "myAudio" -c:v copy -c:a aac -b:a 256k -strict -2 -f mp4 "myOutputFile.mp4"
libmp4v2 is under the MPL and can be used as part of a larger work commercially. It is much lighter than libavformat also.
I'm trying to decode an FLV's audio to a playable format. I attempted to use this SO post: FMS FLV to mp3.. as an example, but my FLV is encoded in Speex.
I have compiled ffmpeg with --enable-libspeex on a Fedora 15 machine.
I believe this can be done with ffmpeg but I'm having a hard time figuring out how to do it.
Any thoughts? Thanks
Your ffmpeg needs to be configured with --enable-libspeex to support Speex decoding. Since you did not provide your OS I can not give any more specific instructions. Once you have a build of ffmpeg that can decode speex the most simple command would be:
ffmpeg -i input.flv output.wav
while reencoding flv file (speex to mp3) if you get sample rate error try this:
ffmpeg -i c:\in.flv -acodec libmp3lame -ar 44100 -vcodec copy c:\out.flv
It does not matter what your input. As long as you have the decoder and encoder enabled in your ffmpeg it will do it.
ffmpeg -i inputfile.flv -acodec libmp3lame any_other_parameters_you_want -vcodec copy out.flv
will do the trick.
run ffmpeg -codecs to see the codecs supported and ffmpeg -formats to see the formats supported in your install.