What we need to do is either, extract the audio from a group or singular Vob video file and convert it to a .Wav format. Now the process of actually extracting the audio and converting it can be done from a third party application however this application has to be activated from a batch or c# script. The media we are retrieving the vob files from always comes from the CD drive and will always be placed in the same folder location once the conversion is complete.
In shot I need a program or script that can be activated remotely to extract and convert vob files to .WAV.
Any help will be greatly appreciated.
I only install ffmpeg on Windows and put in cmd this command:
ffmpeg -i [VOB_file_route].VOB [WAV_file_output_route].wav
And that's all, just works.
To do somthing like this you will need to use VLC and cmd they work togeather well and will allow you to strip video and audio from a file easily
Related
So I am wanting to create spectrograms using the FFMPEG for thousands of FLAC files in batch.
I am using the following for just one file.
ffmpeg -i audio-in.wav -lavfi showspectrumpic image-out.png
However, I would like to do this for all the files in a certain folder (\Desktop\FLACfiles) and don't want to keep changing the file name and the image output name.
I would like to somehow create a batch script in Windows 10 that automatically creates a spectrogram based on the filename.
I was trying to make it work but I don't have much experience via command line or programming in general. Not sure how to do achieve this.
Simply put, would like to use commands from a working directory containing FLAC files and create a spectrogram for each file matching the filename.
This worked for me using mp3 files but it wasn't fast. Someone may have a better solution. Try this as a bat file.
for %%a in ("*.mp3") do ffmpeg -i "%%a" -lavfi showspectrumpic "%%~na .png"
So I have a lot of mp4 files on my computer and I read that QT-FastStart is for moving the metadata from the end of the files to the beginning but how do I use or run it ?
Every time I drag and drop a file into qt-faststart, nothing seems to happen?
I downloaded the 64bit version from here:
https://web.archive.org/web/20140206214912/http://ffmpeg.zeranoe.com/blog/?p=59
Do I need a batch file or something or a specific command line parameter to make it run?
Note, QT-FastStart is described here https://manpages.debian.org/stretch/ffmpeg/qt-faststart.1.en.html
qt-faststart is a utility that rearranges a Quicktime file such that
the moov atom is in front of the data, thus facilitating network
streaming.
It can be used (perhaps among other purposes), for making a sample file when demonstrating an issue. One can take a large file, fix it with QT-FastStart, then use dd to cut a sample. And the sample should play. Whereas if you did dd without doing that then it wouldn't or may not play.
See Neil's answer qt-faststart infile.mp4 outfile.mp4
However, QT-FastStart has now been integrated into ffmpeg.
ffmpeg -i original.3gp -codec copy -movflags +faststart fixed.3gp
List item
simple. in CMD prompt run qt-faststart infile.mp4 outfile.mp4
Is there an ActionScript library for MP3 encoding than Shine-MP3-Encoder? (https://github.com/kikko/Shine-MP3-Encoder-on-AS3-Alchemy)
This library works fine, but I wanted to find a similar solution (on ActionScript, not C) to optimize it for specific needs.
Thanks.
Dont wait for it. A pure AS3 library is impossible as the speed of AS3 is too slow to do MP3 encoding. You will need C++ via AIR Native Extensions (ANE) or Alchemy for this. ANEs work with Adobe AIR so you can publish your content as an app for desktop and mobile. I dont see any issue with it.
See these for ANE-based Mic recording:
ANE-RecordAudio
AirMicrophone
AirMicrophone-ANE
I recorded the mic into a WAV and then use ffmpeg -i d:/path/input.wav d:/path/output.mp3 to convert it instantly into an MP3. I delete the WAV when FFMPEG completes. FFMPEG is a cool little EXE you can just add to your project in any dir. I used Thibault's class to record into WAV.
I am trying to play audio as i used to do but it seems not working now. These are the codes I tried:
local birdSound = audio.loadSound("bird.mp3")
audio.play(birdSound)
It gives an error like that:
WARNING: Failed to create audio sound
Can you help me out? Thanks.
Don't use .mp3. .wav works for both iphone and android.
Make sure the .wav file is in our folder.
Sometimes some sound file can be played on computer, but not works in simulator. In that case, use other files instead.
If you really like that .mp3 file and can not find .wav, find some free software to convert it.
Changing the bit rate(increasing) of my files helped me to solve the problem. Thanks to SatheeshJM!
In my application there are mp3 files located in the bundle (nothing from the web). Some of the mp3 files are original files and some I had edited using simple sound editing software (the ones where you insert a file, cut a slice of it and save it as a new and shorter mp3 file).
I'm using the AVAudioPlayer [initWithData] method.
All the original files (the ones that I hadn't edited and inserted to the bundle as is) are working perfectly and all the ones that were edited are not working at all.
I used 2 different editing software and the outcome is the same.
Anyone had ever encountered that or have any idea what may I done wrong?
Thanks,
Ohad
Converting the mp3 to caf worked for me
as specified here.
see the following:
How do I convert an audio file to the preferred format for iPhone OS?
The preferred full-quality audio format for iPhone OS is 16-bit, little-endian, linear PCM packaged as a CAF file. To convert an audio file to this format, use the afconvert tool at the command line in Mac OS X, as shown in Listing 5.
Listing 5 Converting an audio file to the preferred format for iPhone OS
/usr/bin/afconvert -f caff -d LEI16 {INPUT} {OUTPUT}
To see all the options available for the afconvert tool, enter afconvert -h at the command line.