Is it possible to recompile movie files to re-stream the audio so they all have the same volume level? We've got users submitting videos and to me it seems some are higher in volume whereas others are not, and they all have the same volume level on the controls, and i'd like to standardize this so all movie files have the same volume levels.
I was thinking of ffmpeg although I only have novice knowledge of this technology and haven't done my research in it yet.
Anyway if there's anything available I'd love to know.
Thanks!
FFmpeg has a loudnorm filter that will normalize the audio to meet EBU R128 recommendations. Basic syntax is
ffmpeg -i video.mp4 -c:v copy -af loudnorm out.mp4
A faster filter is dynaudnorm but this one may alter the audio shape a bit, or so I'm told.
ffmpeg -i video.mp4 -c:v copy -af dynaudnorm=r=0.5 out.mp4
Related
I would like to lower video resolution - usually from .mkv files - but to keep all possible audio tracks (might be only one, might be several) and subtitles (might be none, might be several) from the original one. I also would like to keep as many encoding parameters as I can from the original video file (especially those I do not understand).
I am still new to ffmpeg : at first the idea seemed simple, but after many attempts, it seems it is more complex than that. Do I have to use the -filter_complex option ? It seems to be an overkill (or overcomplex) for what I thought to be an easy conversion, but I might be wrong.
I tried to combine -vf scale=-1:720 with -c copy -map 0, which gave me an error that I now understand, but I am stuck with the next step.
Any lead on to achieve that ? Could it be done with ffmpeg only or would I need a script ?
Your try should've worked. For example,
ffmpeg -i input.mp4 -vf scale=-1:720 -map 0 -c:a copy -c:s copy output.mp4
grabs all the streams from the input, passes all video streams through the scale filter, and copies all audio and subtitle streams.
What was the error?
keep as many encoding parameters
This it cannot do. When you reencode, it's on you to pick the parameters to best match those possibly used for the input.
Merging two videos is easy, been answered couple of times. What I have is multiple overlapping videos. A video might have overlaps with video before it. Meaning if video 1 covers 1-5 timeline then video 2 may overlap 1, and cover 3 to 8. Merging them as is would result in 1-5|3-8, when i need 1-8 only.
Videos are alphabetically sorted.
My general idea of solution is...
grab last frame of the video
if it's first video continue
if it's not first video, ie. 2nd, search for frame saved in previous steps frame by frame
if it reaches to last frame of current video then there is no overlap continue
if it founds a frame then clip 2nd video up to that frame inclusive and then go to next frame
once all videos have been analyzed, merge them into one video.
I need to translate this to ffmpeg commands. Or opencv if that's a better tool.
If there is better way of doing that, I'm interested in that too.
for ffmepg you can use the script below. it tested it. But timing wise, you have to change of this STARTPTS+5 to +25 in your video. I put 5 here to test the merging is happening.
ffmpeg -i 2.mp4 -i 1.mp4 -filter_complex "[1]setpts=PTS-STARTPTS+5/TB[top];[0:0][top]overlay=enable='between(t\,10,15)'[out]" -shortest -map [out] -map 0:1 -pix_fmt yuv420p -c:a copy -c:v libx264 -crf 18 output1.mp4
Limitation
This one need the source to be long enough which means you need video canvas then use this script to add each video into the canvas.
And there is no fully autonomous way of use it in ffmpeg.
You are right. Opencv cant deal with audio. need 3rd party library support to run concurrently. Before then I have to use ROS to get both sound and vision to the robot system from a webcam. The sound is then process with NLP for natual language user interface and vision is used separately for locozlaiton and mapping.
There is some way to walk around.
First, you use OpenCV template matching or image difference on a local window batch. The smallest error position will give you the correct location A to insert. This should be accurate in terms of mili-second level. (if error is always big, then it means there is no overlap and return exception)
Second, based on the correct location obtained from opencv. call system.call to invoke the above script with A parameter as input to do auto merge.
Depends on your application, if you need to do it frequently, write opencv python script to automatic fuse. If just once every month, do it manually with ffmepg is good enough
I have a question relating to ffmpeg. First here is the scenario, I am working on a project where I need to have some audio with a presenter talking and then potentially some background music. I also have the requirement to normalize the audio. I would like to do this without presenting a bunch of options to the user.
For normalization I use something similar to this post:
How to normalize audio with ffmpeg.
In short, I get a volume adjustment which I then apply to ffmpeg like this:
ffmpeg -i <input> -af "volume=xxxdB" <output>
So far so good. Now let's consider the backing track, it doesn't want to be the same volume as the presenters voice, this would be really distracting, so I want to lower that by some percentage. I can also do this with ffmpeg, I could do it like this (example would set volume to 50%):
ffmpeg -i <input> -af "volume=0.5" <output>
Using these two commands back to back, I can get the desired result.
My question has two parts:
Is there a way to do this in one step?
Is there any benefit to doing it in one step?
Thanks for any help!
After testing some more, I actually think the answer was pretty straight forward, I just needed to do this.
ffmpeg -i <input> -af "volume=xxxdB,volume=0.5" <output>
Took me a while to realize it, I had to try with a view samples before I felt confident.
Earlier I wrote so:
ffmpeg -i input.mp4 -sameq output.mp3
...and thus receive audio from video file. Ffmpeg just taken out or converted audio to mp3 with an appropriate quality. All thanks to key: -sameq [use same quantizer as source]
Now in Ubuntu instead of ffmpeg we have libav and there (in man for avcomv) I see no -sameq key. Well, here is a question: what I have to do now?..
What I have to do now to get converted audio file with the same quality as in original?
PS. -sameq : Use same quantizer as source (implies VBR).
$ man ffmpeg | col -b > ./man_ffmpeg
this man_ffmpeg is there: http://pastebin.com/qYxz1M1E
FFMPEG(1)
NAME
ffmpeg - ffmpeg video converter
SYNOPSIS
ffmpeg [[infile options][-i infile]]... {[outfile options] outfile}...
...
...
...
-sameq
Use same quantizer as source (implies VBR).
...
...
...
SEE ALSO
avplay(1), avprobe(1), avserver(1) and the Libav HTML documentation
AUTHORS
The Libav developers
2014-02-06
FFMPEG(1)
You are correct, -sameq option has been deprecated and then removed from avconv, there were many reasons for it. Not the least of it being that there are different quantizers and it makes little sense talking about same quantizer parameters when reencoding between different codecs.
Majority of people, when reencoding are looking for quality, not quantizers. So they should use -qscale n where n is between 1 and 31 representing quality from best to worst.
In a way if you have gotten used to -sameq option, you have fallen victim to a tool that should have been there at best for testing purposes. It doesn't produce anything reasonable, and can be kinned to trying to put "same metadata" into the container that doesn't support it, or doing "copy stream" into an archaic file format (leading to things like AVI with vorbis audio, that can't even be played). You can hack something together that does all these things, but it has no place in a video encoding tool.
I suggest that if you are going to be doing much stress testing of different containers and codecs, then you install ffmpeg which has more tools allowing the creation of frankensteins. If you are reencoding for the purposes of actually keeping the files that you produce or distributing them, than you can create another question explaining your situation, and what is your desired outcome.
In short "How can i create reencoding process with exactly the same quantizer?" Can only be answered with "No".
From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong.
I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio tracks by a few seconds so that they are synchronized. Each file is one half of a conversation between a student and a teacher. For example teacher.wav might need to begin 3.3 seconds after student.wav. I can only figure out how to mix the files with SOX where both tracks begin at the same time.
My best attempt at this point is:
ffmpeg -y -i rtmp://server/appName/instance/student.flv -ac 1 student.wav
ffmpeg -y -i rtmp://server/appName/instance/teacher.flv -ac 1 teacher.wav
sox -m student.wav teacher.wav combined.mp3 splice 3.3
These tools (FFMEG/SoX) were chosen based on my best research, but are not required. Any working solution would allow an ASP.Net service to input the two FMS flvs and create a combined MP3 using open-source or free tools.
EDIT:
I was able to offset the files using the delay switch in SOX.
sox -M student.wav teacher.wav combined.mp3 delay 2.8
I'm leaving the question open in case someone has a better approach than the combined FFMPEG/SOX solution.
For what it's worth, this should be possible with a combination of -itsoffset and the amix filter, but a bug with -itsoffset prevents it. If it worked, the command would look something like this:
ffmpeg -i student.flv -itsoffset 3.3 -i teacher.flv -vn -filter_complex amix out.mp3
mixing can be pretty simple: how to mix two audio channels?
well i suggest you should use flash.
it may sounds weird, correct me if im wrong but with Flash's new multimedia abilities you can mix a couple tracks.
im not sure, but i'm just trying to help you,
theese 2 link can help you for your aim (specially second link i guess);
http://3d2f.com/programs/25-187-swf-to-mp3-converter-download.shtml
http://blog.debit.nl/2009/02/mp3-to-swf-converter-in-actionscript-3/