Tag Info

Hot answers tagged

12

Yes. MP3 is lossy so if you re-encode the new file to MP3 with Audacity/LAME, you will lose quality. But if you output the cropped file using a lossless codec such as FLAC, the quality should be no worse than the imported MP3. Some other tools allow you to crop MP3 tracks without re-encoding, such as mp3splt and mp3DirectCut. See How do I split an MP3 file ...


8

From a Unix-like (Linux, OSX, etc) commandline, ffmpeg can be used like this: for f in *.wav; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%wav/mp3}" -c:a libvorbis -q:a 4 "${f/%wav/ogg}"; done This will convert every WAV in a directory into one MP3 and one OGG; note that it's case-sensitive (the above command will convert every file ending in .wav, but ...


6

Download the GSnap plug-in and choose "Download GSnap / Grymmjack skin (Size 310 KB, last modified 02 May 2007)" . You will have a folder containing two icons, GSnap-GJ.dll and GVSTLicense. Open My Computer -> Local Disk -> Program Files -> Audacity -> Plug-ins and copy GSnap-GJ.dll there (you should have Audacity 1.3 Beta (Unicode)). Open Audacity and open ...


6

In Windows Vista, I had to do the following: Go to Control Panel > Sound > Manage Audio Devices Select the Recording tab Right click an item or blank area on the list of devices and enable "Show Disabled Devices". This made the Stereo Mix device appear. Right click the Stereo Mix device and select "Enable" "Stereo Mix" was then available as an input ...


5

There are different types of gain control: Peak normalization will search for the loudest part of the file, amplify it to the loudest level possible and then amplify the rest accordingly. It's a pretty "stupid" technology in the sense of not making use of any psychoacoustics. ReplayGain is an algorithm that analyzes the files on a psychoacoustic basis and ...


5

Using Audacity 1.3.13-beta: Create new file Menu > Generate > Tone Pick your fundamental frequency (f = 1000Hz), and any other options you'd like (amplitude, length, etc.) Hit Ok - It will generate a 1000Hz tone. Menu > Tracks > Add New > Audio Track Ensure the new track is selected, Menu > Generate > Tone Pick your first harmonic frequency (2f, 3f, etc.) ...


5

I would suggest using sox for this kind of thing. Drop right channel with: sox in.wav out.wav remix 1 To reduce noise, you need to get a noise profile from a silent part of the file, i.e. something like this: sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered See the below for details on noiseprof and noisered. So the sequence for each file ...


4

This is actually possible using Audacity's Beat Finder analysis plug-in when combined with a bandpass filter. Duplicate your mono audio track. Find the tone's exact frequency. Select the tone you want Audacity to detect and choose Analyze → Plot Spectrum. Set the size to 16384. Move the mouse pointer over the highest peak. Look for the "peak" ...


4

There are heaps of products that can grab the speakers, but few that do direct digital capture. Some free products that are potential solutions (but which I don't know well enough to recommend): jackaudio VB-Audio CABLE (donationware, a similar free product on same web-page is Hi-Fi cable) Virtual Audio Pipeline (open-source) The best-known product is ...


4

I would convert it to WAV so Audacity can process it. An article on audio files - here. It says: There is one major uncompressed audio format, PCM, which is usually stored in a .wav file on Windows or in a .aiff file on Mac OS (So, a WMA/Windows Media Audio file is a lossy compressed wav.)


3

In Audacity, you can set keyboard shortcut to Export command (this will save you time opening file menu and hitting Export...) Also you can deselect the option "Show Metadata Editor prior to export step" in Preferences. This will make it a bit less painful (sorry for my sarcasm)


3

I have an idea which could work, but before I write out the whole thing, have a try: Get yourself an FFMPEG build: http://ffmpeg.zeranoe.com/builds/ Extract Run FFMPEG with the following parameters: ffmpeg -i somefile.mp3 -ac 1 somefile.wav This will load somefile.mp3, limit the audio channels to one, hopefully removing the correct audio channel (the ...


3

Your mixer has an on-board USB sound card, so make sure you set up Audacity to record directly from the Mixer's USB interface. Do not route the audio through your Sound Blaster for recording (Edit -> Preferences -> Devices -> Recording). You can, of course, still use the Sound Blaster as your sound output device (Edit -> Preferences -> Devices -> Playback). ...


3

You essentially want to make a mono track. Click on the Title drop down of the track ("Livin by the" in your screen shot) Choose Split Stereo to Mono. Delete the right track (Click the X in the upper right corner.) Export as necessary. Done.


3

Have you tried Audacity's Noise Removal feature(s)? This may be more of a 'click' than a 'noise' (looks like it), so perhaps you'll have better luck with the Audacity "Click Removal" features? Example of what's on the page: Sometimes an even better result can be obtained by zooming in (CTRL + 1) to near sample level and either silencing the click (Edit ...


2

Don't know if you're using a PC, or Mac, or which version of operating system, so I'll pretend it's a PC. Go to START/All Programs/Accessories/Entertainment/Volume Control. That's the legacy Windows volume controls, not the generic ones that come with the soundcard driver. Make sure everything's on full there. Then check up everything's on full in your ...


2

First: Since CDDA is a 16 bit format is there any advantage to archive storage using flac with 24 bits instead of 16? I wouldn't think so but I'm new to this If not - I'm curious as to who uses the 24 bit option? No. You don't get any improvement by storing more bits than the source data offers. Second: If flac is "lossless" why does it ...


2

I decided to roll my own script in autohotkey (it is a bit unstable, but can get through a few files at a time): SetTitleMatchMode 2 #p::Pause #x::Exit #a:: direc = C:\Documents and Settings\Test\My Documents\myaups\ FileList = ; Initialize to be blank. Loop, %direc%*.aup FileList = %FileList%%A_LoopFileName%`n Loop, parse, FileList, `n { Sleep ...


2

Well there is a bug in audacity so that's why this is happening. In order to change the sample rate you need to use a different program.


2

Once a file has been loaded into Audacity it's converted into the format used in the Audacity workspace. When you open an MP3 you may notice the loading dialog actually says "Importing", so (depending on the input file) it's converted to something like a "Stereo, 44100Hz 32-bit float" 'track'. At this point the file's bitrate is no longer involved, so ...


2

A standard .WAV file can contain multiple channels, but there is no universal mapping of channels to speakers. So you will need to know something about the playback system to target the rear speakers only. If your target system is 5.1, you might get away with creating a 6 channel file, and only putting your data in the 5th and 6th channels. See this page ...


2

Looks like you can use oggenc to convert WAV into OGG, and you can use lame to convert WAV into MP3.


2

You could use foobar2000 with encoders for ogg and mp3. I believe you can find encoders at rarewares.


2

I would convert it to FLAC. It is lossless and audacity can process it


2

Or is there a lightweight tool better suited to such tasks, on either Windows or Linux? On Windows, there is WaveShop. It's similar to Audacity and the save process is shorter with 3 clicks: Close WaveShop via X Prompt to save pops up Click Yes Prompt for MP3 quality settings pops up (don't change anything) Click yes WaveShop automatically saves ...


2

I think you're trying to over complicate things. I believe the problem you're trying to solve here is to record Skype conversations. To do that, you need: Audio stream from your sound card Audio stream from your notebook's mic (or external mic) Video stream of the chat session You can get 3 by using any screen recording software. I prefer Debut by NCH ...


1

This is a complementary answer to sum up the others into the solution that I was asking for. So, after getting this answer and trying to apply the multi-track part by using MixPad, when I press the record button for the individual tracks after selecting a recording device, I get an error message unless that device is set as the default. As the recording ...


1

When you feed it MP3-encoded files, it will decode the files into an uncompressed format that is used for internal representation. Working on encoded files directly is not easy since audio samples may be interleaved or dependent on others. An audio sample is represented in the percetpual hearing domain – so, what you see in the file is not actual audio: you ...


1

It's decompressing and converting them to a raw (or a proprietary) format so that you can easily edit them in a non-linear fashion. The MP3 and the WMA are more than likely compressed, where as the WAV more than likely isn't, or as least isn't compressed nearly as much. This decompression and conversion is what you're waiting for when you import them (note ...



Only top voted, non community-wiki answers of a minimum length are eligible