I'm using ffmpeg -i video.mp4 -i audio.mp3 -c:a aac -shortest output.mp4 to join a part of an mp3 file to a video while converting to AAC, but the last 3 centiseconds of the audio are silent. Is there a way to fix this?
Log: https://jpst.it/2aIIh - This one had 10 ms of silence at the end.
Related
everybody here! So basically this is what I want to achieve:
I have a muted video about 3 minutes long.
I have a list of audio tracks in mp3 format (40 songs in a folder with duration 2 to 6 mins each one)
I want this video to play cycled automatically taking songs from playlist and injecting them to the video one by one. Every time a song finishes the next one from the list should start playing at the moment. Video continues playing and doesn't care duration of tracks.
I consider it as the first step on the way to broadcast radio with a video background on youtube in 24/7 mode with ability to put additional tracks to playlist without need to stop translation.
My problem is that I'm new in FFmpeg and I would appreciate any suggestions regarding which FFMpeg topic to start investigate with in order to achieve my goal
Use the concat demuxer
You can do live updates to the playlist for the concat demuxer, but each audio file must have the same attributes, the same number of streams, and all be the same format.
Create input.txt containing:
ffconcat version 1.0
file 'audio1.mp3'
file 'audio2.mp3'
file 'audio3.mp3'
file 'audio40.mp3'
All file names must be "safe" or it will fail with Unsafe file name. Basically no special characters in file names and only use absolute paths. See concat demuxer for more info.
Run ffmpeg to stream to YouTube:
ffmpeg -re -framerate 10 -loop 1 -i image.jpg -re -f concat -i input.txt -map 0:v -map 1:a -c:v libx264 -tune stillimage -vf format=yuv420p -c:a aac -g 20 -b:v 2000k -maxrate 2000k -bufsize 8000k -f flv rtmp://youtube
When you are ready to add new songs make temp.txt containing:
ffconcat version 1.0
file 'audio41.mp3'
file 'audio42.mp3'
file 'audio43.mp3'
Replace input.txt atomically:
mv temp.txt input.txt
See FFmpeg Wiki: Concatenate for lots more info.
If your audio files are not the same
The files listed in input.txt must all have the same:
Format (AAC, MP3, etc, but not mixed)
Sample rate (48000, 44100, etc)
Number of channels (mono, stereo, etc).
If they vary then you will have to pre-process them before adding them to the playlist. Bash example conforming each audio to stereo (-ac 2) with 44100 sample rate (-ar 44100) and save as AAC format in M4A container:
mkdir conformed
for f in *.mp3; do ffmpeg -i "$f" -map 0:a -ac 2 -ar 44100 -c:a aac "conformed/${f%.*}.m4a"; done
Outputting to AAC is recommended for streaming to YouTube.
If you do this then you can avoid re-encoding the audio in the ffmpeg command to YouTube. Just change -c:a aac to -c:a copy in step #2: Run ffmpeg to stream to YouTube.
I am using ffmpeg to extract the audio from a video. Below code downlaods the audio from a video file. I'm not sure how efficient this program is but I do know that it downloaods it in 48KHZ.
How do I use this program to extract audio from a video in 8Khz because the file is getting too big.
ffmpeg -i video_link -vn output.wav
Use -ar option to change frequency rate
ffmpeg -i video_link -vn -ar 8000 output.wav
If you want to try different formats of audio check the available formats in ffmpeg using ffmpeg -formats and available codecs using ffmpeg -codecs
Here's an example to extract to mp3 file
ffmpeg -i video_link -vn -ar 8000 -f mp3 output.mp3
Edit: as #llogan pointed out, -f option is not needed, ffmpeg automatically mux mp3 file.
ffmpeg -i video_link -vn -ar 8000 output.mp3
I'm trying to calculate the audio + visual difference between a harshly compressed video file and one that hasn't been.
I'm using pipes because ultimately I wish this to take src from a camera stream.
I've managed to get the video results that I'm looking for, but I'm struggling with the audio.
I've added a line to invert the phase of the compressed audio, so that when they add up in the blend they should almost cancel each other out, but that doesn't happen.
ffmpeg -i input.avi -f avi -c:v libxvid -qscale:v 30 -c:a wmav1 - | \
ffmpeg -i - -f avi -af "aeval='-val(0)':c=same" - | \
ffmpeg -i input.avi -i - -filter_complex "blend=all_mode=difference" -c:v libx264 -crf 18 -f avi - | \
ffplay -
I can still hear all the audio, when what I should be hearing are solely compression artifacts. thx
To preface, I'm not sure your method would identify audio compression 'artifacts'
Your command doesn't perform any audio comparison, it only inverts a single channel. Also, the audio and video are compressed twice and the codecs the last ffmpeg command receives are the default AVI codecs of mpeg4 and mp3.
Use
ffmpeg -i input.avi -f matroska -c:v libxvid -qscale:v 30 -c:a wmav1 - |\
ffmpeg -i input.avi -i - -filter_complex "[0][1]blend=all_mode=difference;[1]aselect=gt(n\,0),asetpts=PTS-STARTPTS[1a];[0][1a]amerge,aeval=val(0)-val(1):c=mono" -c:v rawvideo -c:a pcm_s16le -f matroska - |\
ffplay -
I assume your audio is mono. If your audio has N channels, your aeval will need N expressions where the Mth expression is val(M-1)-val(N+M-1)
I also trim out the first encoded audio frame in order to mitigate encoder delay that Paul mentioned, and it seems to work here.
There might be some delay introduced with encoded audio samples. Also your command is incorrect.
I'm using avconv for merging an audio file with a video file. I'm using a script in which the video length will vary from file to file. So, I use -shortest flag to stop audio abruptly at the end of the video. I want to fade out audio alone for 2 seconds at the end of the video. But, I don't know the exact length of a particular video.
avconv -i inputvideo.mp4 -i inputaudio.mp3 -c:v copy -c:a aac -strict experimental -shortest videowithfadedaudio.mp4
What should I do?
I have some video files that I need to re-encode due to compatibility issues. They are currently mkv files with h.264 video and ac3-a52 audio. I want to keep the h.264 video, convert the container to m4v and create two audio tracks, one with the original ac3-a52 and one copied from that but in aac stereo.
I assume there has to be some sort of audio stream mapping command but I don't know how to map and re-encode at the same time. What command should I enter into ffmpeg to achieve this?
Also, what is the difference between ac3 and ac3-a52? Will an apple TV still be able to pass through ac3-a52 or does that have to be converted to ac3?
this works for me:
ffmpeg -y -i Source.mkv -map 0:v -c:v copy -map 0:a -c:a copy -map 0:a -strict -2 -c:a aac out.mkv
-y – A global option to overwrite the output file if it already exists.
-map 0:v – Designate the video stream(s) from the first input as a source for the output file.
-c:v copy – Stream copy the video. This just muxes the input to the output. No re-encoding occurs.
-map 0:a – Designate the audio stream(s) from the first input as a source for the output file.
-c:a copy – Stream copy the audio. This just muxes the input to the output. No re-encoding occurs.
-strict -2 -c:a aac – Use the native FFmpeg AAC audio encoder. -strict -2 is required as a way that you acknowledge that the encoder is designated as experimental. It is not a great encoder, but it is not too bad at higher bitrates.
According to wikipedia, there is no difference between AC3 and ATSC A/52: the 1st one is the name of the codec, the 2nd is the name of the standard specifying the AC3 codec. Maybe someone have more knowledge about it?
I'm doing the same as the OP, but with an m4v container. I'm using the MacPorts "nonfree" variant of ffmpeg so that I can use libfaac, which gives better audio quality than the built-in AAC encoder and also had the same issue as #dkam. The command line I ended using is like this:
ffmpeg -i input.m4v -map 0:v -c:v copy -map 0:a -c:a:0 copy -map 0:a -c:a:1 libfaac output.m4v
(The videos are for playback on an iPad, which doesn't seem to be able to handle ac3.)
This command will take a video with 1 audio stream, and downmix to stereo and convert the audio stream and add it as a 2nd audio stream. It will be in AAC 384k.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:1 aac -b:a 384k -ac 2 OUTPUT.mkv
Explanation of the command
ffmpeg -i INPUT.mkv The application and input file
-strict -2 Enable downmixing support
-map 0 Tell ffmpeg read all Video, Audio, and Subtitle streams for the following arguments
-c copy Copy everything
-map 0:a:0 Tell ffmpeg to read the first audio stream for the following arguments
-c:a:1 aac Output the audio to a 2nd audio channel (0 = first channel) in aac format. Important! You must change the output channel to a higher number if there are multiple audio streams to prevent overwriting them.
-b:a 384k 384k bitrate (I don't know what's good for aac stereo but this is really high since it's for 5.1 aac)
-ac 2 Downmix to stereo
OUTPUT.mkv Output file
More examples
A video with two audio streams. Creating a third audio stream by encoding the first.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv
Again a video with two audio streams, but you want to encode the second one
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:1 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv