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
Related
I am looking for a way to convert large number of MP3 files to videos, each using the same image. Efficient processing time is important.
I tried the following:
ffmpeg -i image.jpg -i audio.mp3 -vcodec libx264 video.mp4
VLC media player played the resulting video file with the correct sound, but a blank screen.
Microsoft Media Player played the sound and showed the intended image. I uploaded the video to YouTube and received the message:
"The video has failed to process. Please make sure you are uploading a supported file type."
How can I make this work?
Create video:
ffmpeg -framerate 6 -loop 1 -i input.jpg -c:v libx264 -vf format=yuv420p -t 00:10:00 video.mp4
The duration (-t) should be ≥ the MP3 with the longest duration.
Now stream copy the same video for each MP3:
ffmpeg -i video.mp4 -i audio.mp3 -map 0:v -map 1:a -c copy -movflags +faststart -shortest output.mp4
Some notes regarding compatibility:
MP3 in MP4 does not have universal support, but will be fine in YouTube. If your target players do not like it then add -c:a aac after -c copy to output AAC audio.
If your target player does not like it then increase the -framerate value or add the -r output option with an appropriate value, such as -r 15. Again, YouTube should be able to handle it.
i want convert video from any format to mp4. so i am using command:
ffmpeg -i ttt.mp4 -vcodec copy -acodec copy test.mp4
this is working perftectly but now i also add scale in this -s 320:240.
There also many other command for convert LIKE :
ffmpeg -i inputfile.avi -s 320x240 outputfile.avi
but after convert by this command video not play in html5 player
BUT this is not working so tell me in my command how i add scale;
So please provide me solution for this .
Thanks in advance.
You have several problems:
In your command, you have -vcodec copy you cannot scale video without reencoding.
In the command you randomly found on the Internet, they are using AVI, which is not HTML5-compatible.
What you should do is:
ffmpeg -i INPUT -s 320x240 -acodec copy OUT.mp4
Adding to Timothy_G:
Video copy will ignore the video filter chain of ffmpeg, so no scaling is available (man ffmpeg is a great source of information that you will not find on Google). Notice that once you start decoding-filtering-encoding (i.e., no copy) the process will be much slower (x100 time slower or even more). The libx264 is recommended if you want compatibility with all browsers.
$ ffmpeg -i INPUT -s 320x240 -threads 4 -c:a copy -c:v libx264 OUT.mp4
vp9 will provide nearly 50% extra bandwidth saving, but only for supported browsers (Firefox/Chrome), and the encoding will much slower compared to libx264 (that itself is much slower that v:c copy):
$ ffmpeg -i INPUT -s 320x240 -c:a copy -c:v vp9 OUT.webm
Notice that there is a set of formats (containers) accepted by browsers (most admit mp4, some also webm, ...) and for each format there is a set of audio/video codecs accepted. For example you can use mp3 or aac with an mp4 file (container), but not with webm files.
http://en.wikipedia.org/wiki/HTML5_video#Supported_video_formats
In my site, having upload video(only mp4 videos) functionality and then to combine. For the combining i used Mp4Box, If we want combine all the mp4 video, those videos have to same dimesions,bitrate,codecs,samplerate,etc, So while uploading the mp4 videos itself we set the constant dimension and other details like
ffmpeg -i test.mp4 -r 25 -s 640x360 -ar 48000 -acodec copy -f mp4 -vcodec libx264 -vpre default -async 1 -strict -2 -qscale 10 test.mp4
After using this command the video quality will loss fro the original video, Kindly suggest any solution?
Add
-qp 0
§ Lossless H.264
I am trying to support the recording of webcam video on our website, which I then need to transcode to MP4 and WebM to support HTML5 playback. I have ffmpeg 1.2 installed on our server, and have the whole process running fairly well.
The one problem I do have though is transcoding FLV to MP4. it is unacceptably slow, e.g. an 8 second FLV takes about 2.5 mins to transcode!
The ffmpeg command I am using is:
ffmpeg -y -i webcam.flv -c:a libfaac -ac 2 -b:a 64k -ar 44100 -c:v libx264 \
-b:v 350k webcam.mp4
There are so many ffmpeg params, I am a bit lost as to the best way forward with this issue. You can download a test flv from here:
dropbox.com/s/hhd6uhdiuhk800w/webcam.flv
By comparison, transcoding to WebM takes about 5 seconds:
ffmpeg -y -i webcam.flv -c:a libvorbis -ac 2 -b:a 64k -ar 44100 -c:v libvpx \
-b:v 350k -metadata:s:v:0 rotate=0 webcam.webm
ok i found the answer. i had a closer look at the ffmpeg output, and noticed:
[mp4 # 0xa0060c0] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
doh. so i added "-vsync 2" as the last parameter before the output file and it worked a charm, took transcoding time down to about 10 secs! very happy.
working out "generalised" ffmpeg settings for all types of a/v input still seems like black magic to me...
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.