I have two audio files one is 10 secs long and other is 17 secs long, I want to mix the files together so that the 17 sec file starts playing from the start, while the 10 sec file will start after 7 seconds into the 17seconds file.
How can I do this?
I followed this link, I also tried other commands mentioned in Sox FAQ, question number 7, but I am unable to mix two files by providing an offset, I also tried the command in command line and the error is same.
The error which I see is
option ` ' not recognized
and the command I used is
sox -m drums.wav "|sox beats.wav -p pad 1.5" out.wav
Edit: It seems to me that the pipe operator "|" is broken, how do I fix this?
My problem is exactly the same as mentioned in this forum
I think there's an issue with ".
Try
sox -m drums.wav '|sox beats.wav -p pad 1.5' out.wav
Related
I need a little bit help and hope to find that here.
I am using sox for tagging some music with voice tags on my server while user is uploading the file. This is my command which I was using. Everything is working fine.
sox -m {voice_tag_loop} {source_file} {output_file}
Now I want to change something, but don't know how to do that and find no solution.
So the {voice_tag_loop} will be uploaded by user and can have all length e.g. 30 seconds, 20s, 17s or 1 Minute. Don't know that before.
The {source_file} is the music file and can have also different length e.g. 3:13 Min, 4:20Min
How can I mix the {voice_tag_loop} with the {source_file} that the {output_file} has the length of {source_file} but has the {voice_tag_loop} is mixed and looped/ repeated into also with as long the length of the {source_file}
I hope I could explain that, that you can understand that.
Best regards
Just repeat until the source file is exhausted, e.g.:
sox -m "| sox {voice_tag_loop} -p repeat -" {source_file} trim 0 $(soxi -d {source_file})
NB, don't forget the trim bit, otherwise the repeat part will generate an infinite file.
OK I have now the answer for all you wants to mix short audio with long audio and the short one should repeated as long the long audio is.
In my case a small description. The short file will be changed to 44.1kHz and will be looped every 30 seconds. Max 100 times but as long as the long file is. And finally both files will be mixed. This all is one procedure.
sox {short_file} -r 44.1k -p pad 0 30 repeat 100 trim 0 $(sox --i -d {long_file}) | sox - -m {long_file} {output_file}
Regards
this did it for me
sox {short_file} -p repeat 100 trim 0 $(sox --i -d {long_file}) | sox - -m {long_file} {new_file}
I have a long list of audio files, and some of them are longer than an hour. I am using Python 3.6, Jupyter notebook by connecting to a remote machine and using TinyTag library to get a duration of audio. Ffmpeg version is 2.8.14-0ubuntu0.16.04.1.
My code below goes over the files and if a file is longer than an hour, it splits the file into one-hour long pieces, and a leftover piece less than an hour, and copies the pieces as fname_0, fname_1,fname_2, etc. Before chopped, each file is .m4a but during chopping, they are converted to a .wav file. However, after this chopping process, when reading the duration of pieces, I realized that all the pieces have 'None' duration. Something must be wrong in the command line but I can`t see what that is. Thanks in advance.
# fpaths is the list of filepaths
for i in range(0,len(fpaths)):
fpath=fpaths[i]
fname=os.path.basename(fpath)
fname0=os.path.splitext(fname)[0] #name without extension
tag = TinyTag.get(fname)
if tag.duration > 3600:
cmd2 = "ffmpeg -i %s -f segment -segment_time 3600 -c copy %s" %(fpath, fname0) + "_%d.wav"
os.system(cmd2)
os.remove(fpath)
When I change to the extension from .wav to .m4a in the cmd2 command line, it works. Writing here just in case if someone has the same problem.
I am attempting to convert a directory full of .bmp files into a .mp4 file (or similar format).
The bitmap files have the following name scheme:
output_N_1024.bmp
Where N is an integer in the range 0 to 1023. (No zero padding / fixed width.)
The command I am using is:
avconv -r 25 -i output_{0..1023}_1024.bmp outputfile.mp4
This appears to run okay, and takes about a minute to convert all 1024, 1024 by 1024 resolution - (confusing?) bitmap images into a new file, outputfile.mp4.
However, when I attempt to open this file with VLC, a black window briefly flashes up and then closes. VLC then goes back to its mode where it waits for you to tell it which file to open next. No error or warning messages appear from VLC, which seems kind of strange since it seems to be refusing to play.
What can I do to fix this? Perhaps my converting command is incorrect?
The problem most likely is that you haven't actually passed the command to encode these files to avconv. This has happened because your shell has expanded the filenames already.
The command i have just managed to get to work on my machine is:
avconv -r 2 -i "%d.bmp" -s 600x400 -an out.ogv
Also for whatever reason it didn't want to work without explicitely giving it the size, but i don't think this is your problem.
In here quotes tell your shell not to touch this string. %d means digits from 1 to whatever the last file is (if you would want them to be 0-padded this would look like %000d to have maximum of three naughts in front).
VLC has then opened and ran my file just fine.
I want to resample a bunch of wav files that I got on a folder.
My script is this:
for f in *.wav; do sox “$f” -r 48000 “${f%%%.wav}.wav”; done
The console give me this error: "sox FAIL formats: can't open input file `“90.wav”': No such file or directory" and so on with the 300 files that are placed on that folder.
How can I batch processing right this files? Why is it giving me this error?
Thanks a lot!
Solution:
for i in *wav; do echo $i; sox $i -r 48000 ${i%%.wav}r.wav; done
Summary: It is the quote symbols
The problem is with the double-quotes:
for f in *.wav; do sox “$f” -r 48000 “${f%%%.wav}.wav”; done
The double-quotes above are non-standard. For them to be properly processed by the shell, the standard ASCII quote symbol must be used:
for f in ./*.wav; do sox "$f" -r 48000 "${f%%%.wav}.wav"; done
As an aside, note that ${f%%%.wav} removes any occurrences of %.wav from the end of the input file name. ${f%%%.wav}.wav adds one .wav back on to the end after removing any %.wav suffixes. You likely want something else here.
Verification
Using the bad quote characters, as per the question, observe the error message:
$ for f in *.wav; do sox “$f” -r 48000 “${f%%%.wav}.wav”; done
sox FAIL formats: can't open input file `“90.wav”': No such file or directory
Note the file name in the error message is shown with two-sets of quotes around the file name. This is what you saw as per the error message that in the question. The outer single-quotes are supplied by sox. The inner double-quotes are the funny quote characters provided on the command line. Because they are non-standard characters, the shell left them in place and passed them to the sox command.
While the file 90.wav exists, no file by the name of “90.wav” exists. Hence, the error.
Conclusion
Stick to standard ASCII characters for shell commands.
This issue can easily happen if the shell commands are typed in using a fancy word-processing editor that substitutes in typographically-pretty but non-standard characters. As tripleee points out, it can also happen when copying-and-pasting from the websites with inappropriate typographical styling.
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.