The syntax is sox input output trim <start> <duration> e.g. sox input.wav output.wav trim 0 00:35 will output the first 35 seconds into output.wav. (you can know what the length is using sox input -n stat) From the SoX documentation on the trim command: Cuts portions out of the audio. Any number of positions may … Read more
There is a better way: soxi -D out.wav
The wav container just adds a simple header to the raw PCM data. The header includes the format, sample rate, and number of channels. Since the raw PCM data does not include this information, you will need to specify it on the command line. Options are specified before the file they apply to, so options … Read more
You will have heard, that audio signals are represented by some kind of wave. If you have ever seen this wave diagrams with a line going up and down — that’s basically what’s inside those files. Take a look at this file picture from http://en.wikipedia.org/wiki/Sampling_rate You see your audio wave (the gray line). The current … Read more
Physically, as you probably know, audio is a vibration. Typically, we’re talking about vibrations of air between approximitely 20Hz and 20,000Hz. That means the air is moving back and forth 20 to 20,000 times per second. If you measure that vibration and convert it to an electrical signal (say, using a microphone), you’ll get an … Read more
In Swift 3 Add framework AVFoundation **In info.plist add key value Key = Privacy – Microphone Usage Description and Value = For using microphone (the apps will crash if you don’t provide the value – description why you are asking for the permission)** Import AVFoundation & AVAudioRecorderDelegate, AVAudioPlayerDelegate import AVFoundation class RecordVC: UIViewController , AVAudioRecorderDelegate, … Read more
Just as there are printer drivers that do not connect to a printer at all but rather write to a PDF file, analogously there are virtual audio drivers available that do not connect to a physical microphone at all but can pipe input from other sources such as files or other programs. I hope I’m … Read more
All the previous answers are correct if you want a technically accurate or scientifically valuable answer. But if you just want a general estimation of comparative loudness, like if you want to check whether the dog is barking or whether a baby is crying and you want to specify the threshold in dB, then it’s … Read more
For me both ffmpeg -ss 10 -t 6 -i input.mp3 output.mp3 or ffmpeg -ss 10 -i input.mp3 -t 6 output.mp3 work OK, just 6 seconds of audio. Here’s the mplayer output (last line): A: 5.8 (05.7) of 6.0 (06.0) 0.5% Also ffmpeg -ss 10 -to 16 -i input.mp3 output.mp3 work the same way. I use … Read more
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 … Read more