I have watches a way of doing this with audacity by using sound finder option. But since audacity is only gui it cannot be used with terminal commands. So is there a program that does the same work but in command interface like sox for example.
sox input.wav slice.wav silence 1 1.0 2% 1 3.0 2% : newfile : restart
Currently, Audacity does have experimental support for scripting.
The scripting module is an experimental GUI plug-in that allows Audacity to be driven from an external Perl or Python script. Commands are sent to Audacity over a named pipe. Any scripting language that supports named pipes can be used. External scripting is one of several ways to extend the functionality of Audacity.
Scripting support is currently considered experimental and is mainly intended for use by developers for the time being.
Feel free to try it out, but don't be too surprised if there are problems, or if the details of commands change between versions of Audacity.
Currently Windows or Linux are recommended. Mac requires more work to get anything useful at all.
There is a fuller list of limitations at the foot of this page.
Specifically, try out the ‘Menu Command’ option:
https://manual.audacityteam.org/man/scripting.html#MenuCommand
I am working on FFMPEG, I read that http://dranger.com/ffmpeg/ article which I understand that FFMPEG doesn't download the file before processing, FFMPEG play the file through ffmplayer or any other player, I want to exactly make sure about FFMPEG, that how it works?
1) It can download the file first and then make instance
OR
2) The file play and during play through FFMPEG Player make instance or conversion
Which point is correct?
If someone knows that, it will be very helpful for others and also me .. :) Thanks in Advance
FFmpeg is a media processing utility. Like most Unix tools, you give it an input to produce an output. It does not grab sources on its own so, no, it will not download anything by itself.
Read the man page for more information about on ffmpeg.
Alternatively, run man ffmpeg!
I am trying to look for a way to play an AIFF file using Erlang.
I have found this tutorial, but it seems to be only about reading the content of the file and not actually playing it.
I suggest using the linux command "play" with "os:cmd" or "ports". It is quite ad-hoc but it is not a very uncommon command and it may do the trick.
I am using SoX to create slow but pitch corrected audio files. The resulting files sound pretty good, but often have a very hard "S" sound that I would like to filter out. Many desktop programs include a "De-Essing" filter that works well, but I would like to have a filter that works on the server side.
What SoX filter and parameters should I use to De-Ess an audio file?
Edit: I should add that this needs to work on Linux.
There is a LADSPA DeEsser plugin that can be used from SoX. You need to have tap plugins installed and properly configured on your system. On Archlinux this can be easily achieved with
pacman -S tap-plugins
You can specify threshold and frequency as first and second arguments. I succesfully used a variant of the following command
# -30: threshold (dB)
# 6200: hiss frequency (Hz)
sox from.wav to.wav ladspa tap_deesser tap_deesser -30 6200
The filter has a fistful of other options I did not analyzed. More details can be found here.
While far from perfect, you may be able to get sufficient results by a suitable low-pass filter. That should not affect other parts of a speech signal too much.
You could use a de-esser VST such as spitfish and a command-line VST host such as MissWatson. Sox has very limited plugin support, so if you need something more specific, you're better off going the VST route.
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I've got many, many mp3 files that I would like to merge into a single file. I've used the command line method
copy /b 1.mp3+2.mp3 3.mp3
but it's a pain when there's a lot of them and their namings are inconsistent. The time never seems to come out right either.
David's answer is correct that just concatenating the files will leave ID3 tags scattered inside (although this doesn't normally affect playback, so you can do "copy /b" or on UNIX "cat a.mp3 b.mp3 > combined.mp3" in a pinch).
However, mp3wrap isn't exactly the right tool to just combine multiple MP3s into one "clean" file. Rather than using ID3, it actually inserts its own custom data format in amongst the MP3 frames (the "wrap" part), which causes issues with playback, particularly on iTunes and iPods. Although the file will play back fine if you just let them run from start to finish (because players will skip these is arbitrary non-MPEG bytes) the file duration and bitrate will be reported incorrectly, which breaks seeking. Also, mp3wrap will wipe out all your ID3 metadata, including cover art, and fail to update the VBR header with the correct file length.
mp3cat on its own will produce a good concatenated data file (so, better than mp3wrap), but it also strips ID3 tags and fails to update the VBR header with the correct length of the joined file.
Here's a good explanation of these issues and method (two actually) to combine MP3 files and produce a "clean" final result with original metadata intact -- it's command-line so works on Mac/Linux/BSD etc. It uses:
mp3cat to combine the MPEG data frames only into a continuous file, then
id3cp to copy all metadata over to the combined file, and finally
VBRFix to update the VBR header.
For a Windows GUI tool, take a look at Merge MP3 -- it takes care of everything. (VBRFix also comes in GUI form, but it doesn't do the joining.)
As Thomas Owens pointed out, simply concatenating the files will leave multiple ID3 headers scattered throughout the resulting concatenated file - so the time/bitrate info will be wildly wrong.
You're going to need to use a tool which can combine the audio data for you.
mp3wrap would be ideal for this - it's designed to join together MP3 files, without needing to decode + re-encode the data (which would result in a loss of audio quality) and will also deal with the ID3 tags intelligently.
The resulting file can also be split back into its component parts using the mp3splt tool - mp3wrap adds information to the IDv3 comment to allow this.
Use ffmpeg or a similar tool to convert all of your MP3s into a consistent format, e.g.
ffmpeg -i originalA.mp3 -f mp3 -ab 128kb -ar 44100 -ac 2 intermediateA.mp3
ffmpeg -i originalB.mp3 -f mp3 -ab 128kb -ar 44100 -ac 2 intermediateB.mp3
Then, at runtime, concat your files together:
cat intermediateA.mp3 intermediateB.mp3 > output.mp3
Finally, run them through the tool MP3Val to fix any stream errors without forcing a full re-encode:
mp3val output.mp3 -f -nb
The time problem has to do with the ID3 headers of the MP3 files, which is something your method isn't taking into account as the entire file is copied.
Do you have a language of choice that you want to use or doesn't it matter? That will affect what libraries are available that support the operations you want.
MP3 files have headers you need to respect.
You could ether use a library like Open Source Audio Library Project and write a tool around it.
Or you can use a tool that understands mp3 files like Audacity.
What I really wanted was a GUI to reorder them and output them as one file
Playlist Producer does exactly that, decoding and reencoding them into a combined MP3. It's designed for creating mix tapes or simple podcasts, but you might find it useful.
(Disclosure: I wrote the software, and I profit if you buy the Pro Edition. The Lite edition is a free version with a few limitations).
As David says, mp3wrap is the way to go. However, I found that it didn't fix the audio length header, so iTunes refused to play the whole file even though all the data was there. (I merged three 7-minute files, but it only saw up to the first 7 minutes.)
I dug up this blog post, which explains how to fix this and also how to copy the ID3 tags over from the original files (on its own, mp3wrap deletes your ID3 tags). Or to just copy the tags (using id3cp from id3lib), do:
id3cp original.mp3 new.mp3
I would use Winamp to do this. Create a playlist of files you want to merge into one, select Disk Writer output plugin, choose filename and you're done. The file you will get will be correct MP3 file and you can set bitrate etc.
I'd not heard of mp3wrap before. Looks great. I'm guessing someone's made it into a gui as well somewhere. But, just to respond to the original post, I've written a gui that does the COPY /b method. So, under the covers, nothing new under the sun, but the program is all about making the process less painful if you have a lot of files to merge...AND you don't want to re-encode AND each set of files to merge are the same bitrate. If you have that (and you're on Windows), check out Mp3Merge at: http://www.leighweb.com/david/mp3merge and see if that's what you're looking for.
If you want something free with a simple user interface that makes a completely clean mp3 I recommend MP3 Joiner.
Features:
Strips ID3 data (both ID3v1 and ID3v2.x) and doesn't add it's own (unlike mp3wrap)
Lossless joining (doesn't decode and re-encode the .mp3s). No codecs required.
Simple UI (see below)
Low memory usage (uses streams)
Very fast (compared to mp3wrap)
I wrote it :) - so you can request features and I'll add them.
Links:
MP3 Joiner website: Here
Latest installer: Here
Personally I would use something like mplayer with the audio pass though option eg -oac copy
Instead of using the command line to do
copy /b 1.mp3+2.mp3 3.mp3
you could instead use "The Rename" to rename all the MP3 fragments into a series of names that are in order based on some kind of counter. Then you could just use the same command line format but change it a little to:
copy /b *.mp3 output_name.mp3
That is assuming you ripped all of these fragment MP3's at the same time and they have the same audio settings. Worked great for me when I was converting an Audio book I had in .aa to a single .mp3. I had to burn all the .aa files to 9 CD's then rip all 9 CD's and then I was left with about 90 mp3's. Really a pain in the a55.