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Hi i am working on a compression audio stuff and i would like to ask you about the audio format the most adequate for human voice that can concerve the same quality of my files while trasfering to the server ? Thanks
With standard audio formats, there's not much of a difference between music and speech compression. MP3, for example, is designed to only lose information that is largely imperceptible to the human ear, especially at high bit rates. MP3 is nice because can choose a bit rate that meets your data needs. If you need more extreme compression you'll definitely lose a noticeable amount of quality.
You will not be able to tune the flac codec, and it's seams overkill to use it for voice recording.
Even if mp3 is not supported natively with java, you should take a look at "lame" which is a CLI mp3 codec, very easy to use with Java (create a Process object, with the parameters you wants...)
usage:
lame.exe -V2 file.wav file.mp3
or from a wav buffer (if your application records the voice itself)
lame.exe -V2 - file.mp3
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I'm developing a gstreamer plugin using the rust programming language. It's a source element and gets a text as a parameter and returns the correspoding speech of the text using some kinds of TTS providers(Google, Amazon, WellSaid, etc). Some providers return an MP3 file and some WAV. So what is the best approach to send the received sound file to the src pad of the element?
Decode MP3 and return PCM for both MP3 and WAV files. (I don't know whether it's possible to decode it inside the plugin or not)
Make the source dynamic to have an MP3 pad or WAV pad.
I'm new to gstreamer and I don't know which approach is better.
Output whatever you receive and let the next elements worry about decoding. That way you're not adding unnecessary complexity to your element, and applications can decide which MP3 decoder they want to use or if they want to directly forward the MP3 elsewhere without re-encoding.
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I want to know about Linux audio, i spent a lot of time on reading but i didn't understand(clearly). Can anybody give a brief information on various Linux audio sub systems(Like OSS, ALSA, JACK, Gstreamer, Phonon, Xine)?.
Any help, Thanks in advance.
I once wrote a famous blog post about the jungle of Linux audio output formats. You can find it here.
Regrettably, the picture is no longer there, here's a copy:
It's a bit old (dating from 2007), but I hope it gives you the general idea. OSS and ALSA are the layers closest to the actual audio hardware. All the other libraries and frameworks simply talk to those lower layers. And as you can see, some of these libs and frameworks actually have wrappers around other libs and frameworks.
Which layer you want to call upon depends largely on what you wish to accomplish.
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Does anyone know of any linux software to reduce the size of mp3 files while having a minimal effect on quality?
I just used pngcrush to do this to all the pngs in our asset library with good results, and would love to be able to do the same to our mp3s. Even if there is a 10% reduction in file size that would be a decent win for me.
Any suggestions? It needs to be something i can call from a shell script.
cheers, max
You can use lame with the --mp3input option.
For example you can do a downsampling of the mp3 quality, specifing the a new lower bitrate using the -b option. For example if your starting mp3 has a quality of 256kbs you can lower it's bitrate to 128kbs:
lame --mp3input -b 128 input.mp3 output.mp3
Depending from the input file bitrate, the output.mp3 file has lower quality and so a file size reduction.
From the lame manuale page:
--mp3input
Assume the input file is a MP3 file.
Useful for downsampling from one mp3 to another. As an example, it can be useful for streaming through an IceCast
server.
It depends on what you want to achieve. avconv (formerly ffmpeg) is a good tool, but there's no one size fits all solution pertaining the parameters, since it depends heavily on the type of audio data (music, speech, etc) and the original bitrate, etc. Try the different settings, maybe convert it to mono, see what produces still acceptable results for you.
As a general rule of thumb: Speech compresses really well, downsampling it to 11kHz may still give acceptable results, but music can also be downsized by decreasing the bitrate or switching to mono.
Try to use Audacity. It's a free audio editing program
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I want to build a simple Audio Converter (between major file formats) using C#.NET, so I need to know the basic steps to do so.
Thanks.
Step 1: find a good third-party component(s) that do(es) conversion between file formats.
Step 2: use this component in your app.
If your intent is to write all the raw conversion code yourself, get ready for some pain. The WAV file format (aka Linear PCM) is easy enough to deal with, as long as the file is just a header plus the sample data. Often, however, WAV files are a lot more complicated than this and require much more elaborate code to locate the various RIFF chunks and parse the file.
And that's just for a very straightforward file format that (usually) does not do any encoding of any sort. The MP3 format is vastly more complex, and requires a good knowledge of FFT (Fast Fourier Transform) and the like.
Update: Alvas.Audio is one third-party C# component that may do what you need. NAudio is another.
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Does anyone know a great audio and video file compressor?
I currenting compressing wmv to 3gp with AVS software.
when i compress the file its out 250kb and that only about 2mins and 30 sec of video.
i would like to compress the sound and video smaller so that i can add more time to my video.
FFmpeg is probably the best choice. It has a wide-range of formats, flexibility and codec support. There's little comparison. There are a variety of desktop apps that are built on it too, if you need a UI.
if you are looking for an application super (c) is very good at all sort of video compressions tho it has a pretty weird UI
I'm a huge fan of ffmpeg. Find out what codec and resolution your mobile device wants. If you're lucky, H.264 will be supported. That codec seems to produce excellent compression these days.
Crazy thing I found the other day. I used QuickTime to File->Export->iPhone my 250MB .avi file and it output the file into four new/different files. Three were in .m4v format and varied in size from 62.3MB to 6.8MB. However the fourth file was a .mov file that was only 381 bytes in size. You read that right. It compressed the video from 250MB to 381b or <1KB. If my math serves me right that is over 600K times smaller than the original file.
Just fyi. Hope this helps.