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Connect filters with commas. A linear sequence of one or more filters makes a filterchain. Connect filterchains with semicolons to create a filtergraph. See Filtering Introduction and Filtergraph Description for more detailed info. Example: -filter_complex \ "[0:v]filter_0,filter_1,filter_2[chain1]; \ [1:v]filter_3,filter_4,filter_5[chain2]; \ ...


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Looks like you can just download the zip for "Snow Leopard and Above 64bit", unzip the file once it's finished downloading (by double clicking it or using the commandline tool unzip), and then you have the binaries. Then you just have to decide where to put the binaries. Since you don't want an admin password, it's probably best to keep them in your home ...


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Your y-offset is set to 24. Adding this value to your input video size results in a value that is larger than your screen size (745 + 24 = 769). A segmentation fault seems odd because I believe ffmpeg should provide an unintuitive error message. Something like: X Error of failed request: BadMatch (invalid parameter attributes) Major opcode of failed ...


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The article HTML 5 and iPad-friendly Video from your own Web Site, last updated Nov 12, 2014, has this information : The article recommends using MP4 as a good solution with a recent enough version of ffmpeg, using H.264 encoding with AAC. I suggest reading the article with attention to the details : It contains an example HTML file that will work on all ...


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No, ffprobe cannot do this for you (as far as I know). I had experimented with something similar in the past and was unable to find a solution using only ffprobe. As a possible work around you may want to perform post-processing on the output.


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Notice; on Windows if you are using concat with a textfile the paths in the textfile need to be relative to the output path. E.g. ffmpeg -f concat file.txt "C:\foo\file.mp4 If your files are located in the directory C:\temp\bar.mp4 your textfile looks like this: file ..\temp\bar.mp4 ...


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The quickest 1-liner would be: ls Movie\ Part\ * | while read line; do echo file \'$line\'; done | ffmpeg -f concat -i - -c copy output.mp4


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You don't need to create the overlay mov as a separate step. First create the overlay like a movie in this way: -loop 1 -i watermark.png Then use a fade filter to fade in for 4 seconds (that is 100 frames): fade=in:0:100 And then delay it by 10 seconds to start later in this way: setpts=PTS-STARTPTS+10/TB Your command then becomes: ffmpeg ...


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That mp4 is only 6 seconds long, and the audio stream is a different length than the video, I was able to sync the two and get rid of the DTS errors by doing the following though: ffmpeg -i op.mp4 -ss 00:00:00.11 -t 00:00:06.00 -vn -y op.wav ffmpeg -i op.mp4 -ss 0.010000 -t 00:00:06 -vcodec rawvideo -an -y op.avi ffmpeg -i op.avi -i op.wav -b:v 590k -b:a ...


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Windows Media Player does not seem to like MP3 audio with sample rates less than 32 KHz in MP4 container format. I tested WMP 12 in Windows 7. I'm unsure what the MP4 container specifications define regarding this issue, but you have several options to deal with this limitation: Use a different player VLC works and is a great player. This option has the ...


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You want to move the moov atom to the beginning. While encoding you can use: ffmpeg -i inputfile [other parameters] -movflags +faststart outputfile.mp4 If they have been encoded already, you can simply copy streams without re-encoding: ffmpeg -i input.mp4 -codec copy -map 0 -movflags +faststart output.mp4 You can also use the qt-fasstart tool. ...


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As Nifle commented, you can use sed like this. Without sed, if your output is: $ ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi -loglevel debug here is a line without any brackets and stuff, it should display too [dca @ 0x7fe86c80f000] leave this stuff here [dca @ 0x7fe86c80f000] and this [dca @ 0x7fe86c80f000] this stuff too another line that ...


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There might be a single command to do all of this but because you're trying to trim and combine clips the simplest method I could find was to break the process into three parts. Start by using ffmpeg to trim the guide clip into 3 sections like this: $ ffmpeg -i guide.mp4 -t 5 guide-out-1.mp4 $ ffmpeg -i guide.mp4 -ss 00:09 -t 9 guide-out-2.mp4 $ ffmpeg -i ...


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You need to install the development package for x11. To do so, type the following. sudo apt-get install libx11-dev If you ever run across a similar situation, you can check to see if there is a development package by typing a command similar to the following. apt-cache search libx11 | grep dev Once it provides a list, pick the one that matches what ...


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this is a very common issue with all of the "external" input sources, for example with network inputs, which can stop for a while due to network connectivity issues. It seems that the -shortest output option is the most viable option in this case - and you have a cronjob that constantly checks it process died and restarts if required. Hopefully some better ...


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I used avconv instead, which got much closer to what I wanted: avconv -r 24 -i src.mov -an -vf fps=fps=12 output.mov


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Get ffmpeg: Ubuntu won’t provide ffmpeg until 15.04. So until then you'll have to compile, or download, or use a PPA. Compile FFcast: Here is how you can compile FFcast from source: sudo apt-get install autoconf automake build-essential checkinstall git libx11-dev x11-utils git clone --recursive https://github.com/lolilolicon/FFcast.git cd FFcast ...


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Stream selection By default ffmpeg stream selection will only map one stream per stream type based upon the following criteria: video – the stream with the highest resolution audio – the stream with the most channels subtitles – the first subtitle stream In the case where several streams of the same type rate equally the stream with the lowest index is ...


-1

You can use the stream specifiers to select a stream type and a particular stream. More info here... http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1


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Video only You can use one filtergraph for all filtering. ffmpeg -loop 1 -i image0.png -i video1.m4v -i video2.m4v -filter_complex \ "[0:v]scale=1280:-2[bg]; \ [1:v]scale=140:-1,setpts=PTS-STARTPTS[fg1]; \ [2:v]scale=100:-1,setpts=PTS-STARTPTS[fg2]; \ [bg][fg1]overlay=70:70:shortest=1[ol]; \ [ol][fg2]overlay=300:70,format=yuv420p[v]" \ -map "[v]" ...


1

The "\" on the linked web page is not really part of the command line. On Linux a "\" at the end of a line is an information to the shell that the command line wraps over two lines and that the shell shall remove the "\" and append the next line to the current line. So remove the "\" and just write ffmpeg -i front_left.wav -i front_right.wav -i ...


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Red5 only records what is received from a client; there is no "extra" created by the server, that includes silent audio data. I would assume based on what I know about the server that your publishing client is sending more data after you believe it has stopped.


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I believe HDMV Presentation Graphic Stream subtitles (hdmv_pgs_subtitle) are image based, not text based, so they can't be converted to text because ffmpeg does not currently support OCR conversions. Hardsubs You will be able to create hardsubs, but that means you would have to re-encode the video: ffmpeg -i in.mkv -filter_complex "[0:v][0:s]overlay[v]" ...


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You should read the compilation guide (this one is for Ubuntu and derivates, but there are others too). You only need libmp3lame installed, so you could simply run apt-get install libmp3lame-dev. Then, run: ./configure --enable-libmp3lame As for your other requirements: For AAC encoding, use -c:a aac -strict experimental and set an appropriate bitrate. ...


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Your command has several issues: -f .mkv should be -f matroska. See ffmpeg -formats for a list of format names. Your second pass command is missing ffmpeg. The hqdn3d filter accepts four optional values, but yours has six. You do not need to encode audio in the first pass. Your command should look something like: ffmpeg -y -i input -c:v libx264 -preset ...


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I took the liberty of using your ffmpeg command and this is what I got: yt "My Youtube File.mp4" A file yt.bat placed in system32 containing: @echo off ffmpeg -i %1 -vn -c:a copy _%1 rm %1 mv _%1 "%~n1.m4a" *renames automatically to m4a, which can be changed in the code. **it needs to be through temp file and then using rm, mv, otherwise the file and ...


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The Windows Command Prompt equivalent is: for %i in (*.mp3) do ffmpeg -f image2 -loop 1 -i picture.png -framerate 60 ^ -i "%i" -c:v libx264 -tune stillimage -c:a copy "%~ni.mp4" Note: This will work when typed in interactively but if you are running it as a script you will need to prefix the variables with %% instead of %.


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Here are some examples for taking an audio file, running it through ffmpeg, and have a video created based on some of the filters available in ffmpeg. Examples: spectogram: ffmpeg -i song.mp3 -filter_complex showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt -y -acodec copy video.mp4 avectorscope: ffmpeg -i song.mp3 -filter_complex ...


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You are essentially talking about separate encodes for separate chunks of video, which can be achieved as separate runs- as in the case of your command ffmpeg -i mtb.mp4 -ss 05 -t 5 -c:v libx264 -c:a aac -strict -2 transcoded3.mp4. To use the Trim filter to slice a chunk from say 5 seconds to 10 seconds, use this (this does use setpts and asetpts): ...


1

Download yasm source code from http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz Unpack tar xvzf yasm-1.2.0.tar.gz cd yasm-1.2.0 Configure and build: ./configure && make -j 4 && sudo make install


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for /f "tokens=5" %a in ('find /i "max_volume" ffmpeg.log') do set max_volume=%a echo %max_volume% it case-insensitively searches the file, determines the line, and then gets its 5th element (token) delimited by whitespace by default (absent delims option) saving the value in the max_volume variable


0

You'll need to split the string. Check this link to see how to do it right: http://stackoverflow.com/questions/1707058/how-to-split-a-string-in-a-windows-batch-file


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Audio visualization with ffmpeg ffmpeg -i input.mp3 -filter_complex \ "[0:a]avectorscope=s=640x518,pad=1280:720[vs]; \ [0:a]showspectrum=mode=separate:color=intensity:scale=cbrt:s=640x518[ss]; \ [0:a]showwaves=s=1280x202:mode=line[sw]; \ [vs][ss]overlay=w[bg]; \ ...


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It does this by rescaling timestamp values from the input timebase (i.e. FPS as a fraction, e.g. 24fps would become 1/24) to the output timebase. First the timebase is set based on the requested FPS: link->time_base = av_inv_q(s->framerate); When filtering, the number of output frames is calculated based on the number of input frames in the ...


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http://sonnati.wordpress.com/2012/07/02/ffmpeg-the-swiss-army-knife-of-internet-streaming-part-v/ "5. Record a stream endlessly rotating target file" That handles your circular buffer. You might have to do a little experimenting to see what works for you to get the previous 30 seconds. There are a number of ways to do that so it'll be up to the ...


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Here are the steps: First open your video file in QuickTime. You can either fire up QuickTime first, go to “File” and then down to “Open File”. Or you could right-click the file itself, choose “Open With” and then choose QuickTime. Once the video is open click “Edit” and you’ll then find the rotate and flip options straight below Once you’ve locked the ...


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The page you linked to is very imprecise in what it states. The actual bitrate of a signal may indeed not be the same as the amount of information encoded therein, but the cutoff of the frequency spectrum is not directly related to the bitrate. See, I can choose to encode a pristine original file with full spectral resolution at 192 kBit/s with an MP3 ...


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As far as your question of whether or not ffmpeg can detect the bitrate of an audio file, the answer is yes. Here is an example ffmpeg output for an mp3 file: Duration: 00:04:12.73, start: 0.025056, bitrate: 332 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s Additionally, ffmpeg makes it very easy for you to encode with predefined ...



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