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
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.
This has been discussed before, but my question is a bit different.
Lets say have a video file with 1+2 as stereo.
Now I want to add 3-8 with silent audio - most likely using the anullsrc.
I need to map that generator to only affect channel 3-8.
Anyone has a solution for that?
No need for a source filter. Just need pan.
ffmpeg -i input -af "pan=8C|c0=c0|c1=c1" -c:v copy out
The first two output channels have their maps set to the two input channels. Since the other 6 are omitted, they will be mute.
I know how to change tempo with atempo, but the audio file becomes distorted a bit, and I can't find a reliable way to change pitch. (say, increase tempo and pitch together 140%)
Sox has a speed option, but truncates the volume AND isn't as widely available as ffmpeg. mplayer has a speed option which works perfectly, but I can't output without additional libraries.
I seem to understand ffmpeg doesn't have a way to change pitch (maybe it does recently?) but is there a way to change frequency or some other flags to emulate changing pitch? Looked quite far and can't find a decent solution.
Edit: asetrate:48k*1.4 (assuming originally 48k) doesn't seem to work, still distortion and pitch doesn't really change much.
Edit2: https://superuser.com/a/1076762 this answer sort of works, but the quality is so much lower than sox speed 1.4 option
ffmpeg -i <input file name> -filter:a "asetrate=<new frequency>" -y <output file name> seems to be working for me. I checked the properties of both input and output files with ffprobe and there doesn't seem to be any differences that could affect its quality. Although it's true that I've run it a few times and the resulting file on some of those had some artifacts, even if the line of code was the same, so it may be caused by some ffmpeg bug; try to run it again if you aren't satisfied with the quality.
As of 2022 (though contributed in 2015), FFmpeg has a rubberband filter that works out of the box without any aforementioned ugly, allegedly slow and poor quality and unintuitive workarounds.
To change the pitch using the rubber band filter, you will have to specify the pitch using the frequency ratio of a semi-tone. This is based on using the formula (2^x/12), where x represents the number of semitones you would like to transpose.
For example, to transpose up by one semitone you would use the following command:
ffmpeg -i my.mp3 -filter:a "rubberband=pitch=1.059463094352953" -acodec copy my-up.mp3
To transpose down, simply use a negative number for x.
To alter both properties simultaneously, specify tempo and pitch values. The tempo value is specified as a multiple of the original speed.
The following command transposes down by one semitone and bumps the speed up 4x:
ffmpeg -i slow.mp3 -filter:a "rubberband=pitch=0.9438743126816935, rubberband=tempo=4" -acodec copy fast.mp3
Quality degradation is imperceptible unless measured statistically.
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.
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/