Would someone kindly recommend a command line utility that can play any section of an audio file specified in milliseconds e.g.
player -start-time=0.1234 end-time=5.6789 audio.wav
None of the audio players that I've come across seem to have this functionality. vlc supports start and end times but in seconds only, while Audacity does not appear to have much in the way of command line options.
sox
You can use sox play with the trim effect:
play audio.wav trim START =END
Which in your case would become:
play audio.wav trim 0.1234 =5.6789
Note that the end can also be specified as a length:
play audio.wav trim 0.1234 2
Which starts playing at 0.1234 and plays 2 seconds of the file.
Also note that the offsets can be specified as number of samples by appending an s to the number.
mplayer
mplayer also supports this:
mplayer -ss START -endpos END audio.wav
ffplay from ffmpeg
ffplay uses similar input parameters but doesn't support absolute end times, so some minor arithmetic is needed:
ffplay -ss START -t $(( END - START )) audio.wav
Related
I want to remove completely silence parts from wav files with ffmpeg.
Input wav can be like :
I am using the following ffmpeg command to remove silence part ffmpeg -i input.wav -af silenceremove=stop_periods=-1:stop_duration=0.2:stop_threshold=-45dB output.wav because I understand from the doc that it will remove all silence parts longer than 0.2 s (silence being below -45dB).
But I get that where silence part has only been reduced to around 0.1 wheras I want it to be 0 (no remaining silence).
In Audacity I will use "truncate audio" filter and choose the above parameters to detect silence and in the action part I will choose to truncate to 0: .
This will yield to what I want (ie an audio with no silence part remaining):
Searching on the internet only lead me to what I already do.
So how can I reproduce the output I get from Audacity "Truncate Silence" filter with ffmpeg and remove all silence parts from audio ?
Edit: The output from silencedetect filter is correct: ffmpeg -i input.wav -af silencedetect=0.2:n=-45dB -f null - detects exactly what audacity detects.
Thanks in advance for your help
It looks like the equivalent command to AUdacity's truncate silence behaviour is the following (in bold what is added):
ffmpeg -i input.wav -af silenceremove=start_periods=1:stop_periods=-1:stop_duration=0.2:start_threshold=-45dB:stop_threshold=-45dB output.wav
I am not sure why adding those 2 parameters leads to the expected behaviour but it works although for some files silenceremove can remove more parts than Audacity / silencedetect detect.
I'm encoding mpd for DashJS using ffmpeg. I'm trying to encode a video that has audio starting 26 seconds after video. When I play it use DashJS, the video refuses to show until the audio starts. What's the best way to stretch the audio to to the beginning of the video using the ffmpeg cli using silence?
I've tried several filters but with no success.
Here's my command:
"ffmpeg" "-i" "static/uploads/tmp.10370ba20dc842f7af87037c8e25e895.mkv" "-c:v" "libx264" "-x264opts" "keyint=25:min-keyint=25:no-scenecut" "-pix_fmt" "yuv420p" "-profile:v" "baseline" "-level" "3" "-coder" "1" "-g" "250" "-c:a" "libfdk_aac" "-b:a" "192k" "-ar" "44100" "-ac" "2" "-map" "0:0" "-map" "0:1" "-map" "0:2" "-map" "0:3" "-metadata" "title=Video" "-f" "dash" "-use_template" "1" "-use_timeline" "1" "-seg_duration" "5" "-init_seg_name" "$RepresentationID$-init.m4s" "-media_seg_name" "$RepresentationID$-$Time$.m4s" "-max_muxing_queue_size" "9999" "-shortest" "-avoid_negative_ts" "make_non_negative" "static/uploads/abea93fe8fe645f291beb6223398e025/index.out"
I tried amixing evalsrc=0 with source 0:1, the stream that starts at 26 seconds but it still starts at 26 seconds.
I found out how to do this. Read the section on first_pts on the aresample page: https://ffmpeg.org/ffmpeg-resampler.html
In my case I added -filter_complex "[0:1] aresample=first_pts=0 [s1]"
I want to use ffmpeg to trim some mp3s without re-encoding. The command I used was
ffmpeg -i "inputfile.mp3" -t 00:00:12.414 -c copy out.mp3
However, out.mp3 has a length of 12.460s, and when I load the file in Audacity I can see that it was cut at the wrong spot, and not at 12.414s.
Why is this? I googled a bit and tried some other commands like ffmpeg -i "inputfile.mp3" -ss 0 -to 00:00:12.414 -c copy out.mp3 (which interestingly results in a different length of 12.434s) but could never get the milliseconds to be cut right.
PS. I wasn't sure whether SO was the right place to ask since it isn't technically programming related, however most of the stuff I found on ffmpeg for trimming audio files were stackoverflow questions, e. g. ffmpeg trimming videos with millisecond precision
You can't trim MP3 (nor most lossy codec output) with that level of precision. An MP3 frame or so of padding is added during encoding. (See also: https://wiki.hydrogenaud.io/index.php?title=Gapless, and all the hacks required to make this work.)
If you need precision timing, use something uncompressed like PCM in WAV, or a lossless compression like FLAC.
On Linux you can use mp3splt:
mp3splt -f mp3file.mp3 from to -o output file format
Example:
mp3splt -f "/home/audio folder/test.mp3" 0.11.89 3.25.48 -o #f_trimmed
this will create a "/home/audio folder/test_trimmed.mp3"
For more info to the parameters, check the mp3splt man page here
On Windows you can use mp3DirectCut
mp3DirectCut has a GUI, but it also have command line support
I have a single intro video. I want to add the intro using ffmpeg or a similar program in the beggining of the users uploaded video (and yes I do need to merge them in one file, so it would be possible to download it later)
I`ve been searching internet and it suggests to convert both (intro and the other video) in to .mpg format.
OK, so far so good, but now when I try to join them together I get
[mpeg4 # 0x5547c60]Invalid and inefficient vfw-avi packed B frames detected
So I`m guessing it is because of something being different in both videos, like frame rate or size.
The worst thing is users are allowed to upload videos in almost any formats, also 240p-720p quality, so there is not one default size to convert the intro video into.
How could this be done?
Your intro video should match the resolution of the user videos, you should have as many intro-videos in as many resolutions as the user videos. Or convert all the user videos to a single resolution to match that of the intro video.Are you doing intro.mpg + user.mpg to combine the videos? Is this giving the above error?
Use ffmpeg:
ffmpeg -i 'concat:input1|input2' -codec copy output
or
ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] concat=n=3:v=1:a=1 [v] [a]' -map '[v]' -map '[a]' output.mkv
or
$ cat mylist.txt
file '/path/to/file1'
file '/path/to/file2'
file '/path/to/file3'
$ ffmpeg -f concat -i mylist.txt -c copy output
Source: Concatenate two mp4 files using ffmpeg
I have been trying to get lots of wav files delayed by 2 seconds at the start using ffmpeg. And so far, even though I have read the manual, I was not able to get it working. Here is my command:
for %%A in (*.wav) do (
ffmpeg -i "%%A" -itsoffset 00:00:02 "%%~NA"1.wav )
And nothing is being changed. Files are simply getting copied. I also tried the same with mp3 files. I also tried mkv and avi (to make sure it was not a container writing issue), but it gives the same result also.
Command is same here and here, but it does not work. Please, help.
You must put -itsoffset BEFORE you specify input. So:
ffmpeg -itsoffset 00:00:02 -i "%%A" "%%~NA"1.wav
Changing the input time offset like that isn't going to do anything noticeable for a single stream, it's meant for fixing out-of-sync issues between audio and video streams.
Do you want to tack on two seconds of silence at the start? If so, one simple way that'd work (although it may feel a bit hackish) is to simply tack on a 2 second WAV full of silence, before the actual input. This would be accomplished by simply adding another -i option before the actual file:
ffmpeg -i 2secsilence.wav -i "%%A" "%%~NA"1.wav
I know this question is over 9 months old, but I came across it and wanted to add some more information about '-itsoffset'. From the ffmpeg trouble ticket pages (https://ffmpeg.org/trac/ffmpeg/ticket/1349):
This command should display file1 content one second earlier than file2 content:
ffmpeg -itsoffset -1 -i file1.ts -i file2.ts -vcodec copy -acodec copy -map 0:0 -map 1:1 out.ts
1) What I see is that -itsoffset adds or subtracts from all the timestamps (both the video and audio streams) in a file. So this option is only going to be useful when remuxing from separate input files.
2) outfile has expected playback behavior with .ts and .mkv containers.
3) It does not work with .avi (no timestamps, so not a surprise)
4) It does not work with .mp4 container (a bug?)
And that is where this issue stands as of today.