I have an android phone. The voice recorder records files in 3gp audio only format. I can play these on my computer with the standard gnome player and with vlc. However, audacity won't open it up. There is an error that says that FFmpeg should import it but it didn't understand the format. I need to edit some of these audio files for use. Is there a way to convert these files to mp3 or flac so I can edit them? Searches turn up w32 ware and some arcane mencoder commands but they have to do with converting video.
I have installed this program ok but I am new to command lines in terminal.
I want to convert some wav files to wma files. I have the wav files currently in a folder called Test to make it easy. So I have entered the following command line:
ajpearson@ajpearson-laptop:~/Desktop/pacpl-4.0.5$ pacpl --to wma home/ajpearson/Desktop/Test and the error message I get is:
error: the following is not a file or directory: home/ajpearson/Desktop/Test
It does not matter what directory I use I get the same error. I am sure the answer is obvious - but not t me.
DSS and DSS Pro file formats are used by professional voice recorders such as the Olympus DS-4000, Olympus DS-5000 and Philips LFH-9600. Is there any known program running under Linux to play and/or convert these files into something more compatible?
a movie is encoded with AC3 in 6 channel audio, what I get out is all of the sounds except for voices, which in 5.1 would be sent to the center channel. What I usually do is fire up avidemux and convert the audio to mp3 stereo, as converting to a 5.1 format usually ends up with a very odd sound (like running everything through an echo chamber). What I'd like to do is run a script to batch-convert these files from AC3 to MP3. The video format may vary, but they are usually XVID. I am comfortable at the command line, but I am not well-versed in audio/video tool terms. I don't need anything extravagant, I just want something that works. Heck, even if it is done one at a time, having a shell script that I can use to simply type:
I want to convert some audio files, to mp3 files. I have only k3b but it converts into ogg or wav. Is there any program to convert a track in mp3? r a k3b add-on?
All such conversions doesnt produce any *.flac file. It seems flac doesnt accept minus sign for the standard input although flac manual allows to use it.
So my question is how I can use the standard input in order to decode audio data with flac?
i am trying to use Audacity to record internally the sound playing on my computer, and im finding the program quite confusing. I have found that by connecting the line out to the line in i am able to record, but the quality is rubbish, it will be a lot smoother recording it internally but I cant find a way of doing this, does anyone know how to record that way or know where i'm going wrong?
I got a mixer today and did everything correctly there and I am trying to put the audio through the microphone input which is fine. But when I try to record through Audacity it's as though the microphone and mixer aren't plugged in. I've played around with the settings in both qJackctl and Audacity only to result worse off than before. Can anybody tell me what I'm doing wrong?
I've installed a tv card but I can't hear the sound through my built-in CMedia soundcard when I watch tv with TVTime. The tv card's audio output is externally connected to my soundcard's line in input. In the 'Input' tab of the 'Sound Preferences' dialog, I can see the input level meter moving.
Audacity can record the tv sound, and I can hear it when I have the 'pass-through' option enabled, or when I play the recording.
I can play media files without a problem, and I can hear the system sounds.
But for some reason I can't get the tv sound to go through to the soundcard's output.
Obviously I've got the audio configured wrong, but I don't know what to do to fix it.
I'm trying to capture audio from a stream, NOT from the mic. This is what I'm doing: Start Audacity, push its record button, play a song. It shows that it is recording but when I play back the results, all background noise, conversation, etc, is in the recording as well as the audio from the song. If the mic is unplugged, I cannot record what I hear coming from the sound card (as in what you hear from the sound card is what you get). I got directions on how to do what I want here ( Looking (maybe) for audio mixer for use with Pulse Audio - Page 2 Posts # 4 and 5) but it's not permanent. I got the instructions in post #11 from that same page to make such a change permanent. I tried using the instructions to make the change permanent (I backed up my default.pa just in case something went wrong) and these are the changes I had made:
Code:
### Load audio drivers statically (it's probably better to not load ### these drivers manually, but instead use module-hal-detect -- ### see below -- for doing this automatically) #load-module module-alsa-sink
I'm trying to record some audio from 4-track cassette tapes using Audacity and a Sound Blaster mp3+ external usb audio card.I'm using Karmic.I have fiddled with levels on the sound card using alsamixer, but the only I way I can detect any sound from the tapes when recording is by turning the levels all the way up in alsamixer, and in doing this, I can faintly make out the audio beneath a large wall of static. If the levels are not maxed out, I only get static when recording in Audacity.
Im trying to create some screencast videos on Ubuntu. As its a well known fact we dont have a very good screencasting too like Camtasia(only windows) on linux.
After googling extensively I zeroed on Xvidcap for screen capturing and Audacity for audio capture.
According to tutorial [url] first we need to run Xvidcap (do the screen capturing recording first) and then again use Audacity for audio recording .. then mix both video and audio.
I guess this is a pain staking process (running video and audio recording seperately), I thought of running Xvidcap and Audacity simultaneously, if i do this xvicap works perfectly (video capturing happens) but audio capture through audacity does not work.
Note: I even try to run the above tools from seperate terminals/consoles (assuming they are seperate processes)
I even disabled 'audio' in xvidcap and tried to run both at the same time but audacity refuses to work.
I think both the tools are trying to use the same system resources ... and audacity is unable to get hold of the required resources.
I am trying to record in Audacity in Ubuntu 8.10 64 bit on a Dell Studio 1535 laptop.The problem is not that its not recording, per se.Rather, the problem is its not recording the way that I want it to.
Its recording from the built-in mic, and I want to use the mic jack instead as a line in from a guitar (well, technically, an acoustic-electric resonator, but its close enough to a guitar to call it a guitar).
I have tried using various devices for recording, and changing settings in both the Volume Control and Sound Preferences, but no matter what I do, Audacity still records from my built-in microphones, instead of from the line-in.
So I asked around and to combine audio files apparently I want to use Audacity.
It seems easy enough to LAYER audio files, but how do you string them together in sequence? I have an audiobook divided into 5 minute chapters and I want to combine them into longer chapters....my MP3 player likes to play them out of order which makes them difficult to listen to. I want to make it 1 long tract, or maybe 6 medium length tracks.
How do I combine tracks with Audacity...Not layering them but putting them in order into 1 long track?
I have Ubuntu 9.04 and just installed Sound Converter. I am trying to convert a bunch of .ogg files to mp3 to play on my iPod and it's not working so well. In the Sound Converter options I have is set to convert to high quality mp3. I choose the folder that the files are in and after a moment (slow laptop) Sound Converter populates, I hit 'convert' and it shows that the conversion completes in two seconds. All that it did was create the new folder structure of artist/album but there is nothing in there. Not sure what I am missing. I used Sound Converter before and it worked fine.
I'm trying to use convert, I have installed the imagemagick. I use this line:convert *.jpg test.pdf but I'm only able to convert to pdf 1 single jpg file, not multiple files at once. When there's more than one file, I get the following error: Segmentation fault
I have a lot of .flac files downloaded from several sites. Most of them come with a .cue file, and the .jpg with the cover, etc. It seems it is the intention of the uploader that one rebuilds the original CDDA. However, if I had a stand-alone CD/DVD player with flac I would hardly see the point of converting the flac to cdda. Furthermore, I could even play the flacs with a software player although, in this case, the audio quality would not be so good due to the noise picked up by the signal from the PC digital circuits.
when i had windows xp i used to convert almost any videos video into audio (the whole video into audio) by going to [url], downloading vdownloader and just inserting the url in the box provided; waiting for a couple of minutes and having an mp3 format file at the location that i want it to be in the desktop. when i searched for something similar for linux, because vdownloader was an .exe file and obviously wasn't going to work on ubuntu.
i found that i could get a video downloaded from videos but not an audio - i wait for the video to finish buffering; click on places; computer; file system; tmp; then rename the video folder and copy/relocate it to where ever on the desktop. ending with this video format "flash video (video/x-flv)". my question is: is there a linux program where i get to convert the entire video format to an audio format, possibly mp3?
I'm trying to extract audio and then convert to mp4 format a bunch of flv files downloaded from internet. There are three files I intend to use ffmpeg in the following options:
ffmpeg -i input.flv -acodec copy output.mp3 and ffmpeg -i "input.flv" -f mp4 -vcodec libxvid -s 640x360 -b 768kb -r 25 -aspect 16:9 -acodec libfaac -ab 96kb -ar 44100 -ac 2 "output.mp4" So I started writing a script like this: #!/bin/bash -x cd /home/koli/exp
So I did a short 30 second video of Recordmydesktop in ogv, and was wanting to convert it to mkv. So when I did the video was awesome but the sound kept skipping. Here is the conversion command i did
Code: ffmpeg -i test.ogv -vcodec libx264 -vpre medium -crf 24 -threads 0 -acodec copy test.mkv video was great, audio not.
I see lots of threads on converting your CDs to MP3s, but I want to do the opposite. I want to burn MP3s into CDs that will play on older CD players that dont have MP3 support.
So how do I do it? I have Mint on this particular computer, which is like having Medibuntu already I think.
I'm using banshee/rhythmbox/amaroK on Fedora 12 (will be moving to debian lenny soon) to organize my music collection. As I don't have much space available on my MP3 player (an 8gb Creative zen) I've always used realplayer on windows to automatically convert EVERY file to 128kb MP3, as most of my files are stored as higher bitrates, which is kind of wasted on a player with 7 headphones...
My question is, how do I tell banshee/amaroK/rhythmox to convert EVERY file that gets transferred, not just the ones the player can't play? There's no checkbox anywhere like there was in reaplayer. I guess there's a script to tweak somewhere, but I have no idea where to start looking.
I have a number of uncompressed audio files recorded off of an analog (POTS) telephone line of fax transmissions. Is there a Linux utility or library I could use to convert these files into images of the fax they contain? I'm not looking to send/receive a fax via a modem, but just to "replay" the communications tones and parse out the fax message.I'm guessing this may not be possible due to duplex issues and not knowing which end of the conversation is sending what,but thought I'd ask to see if anyone knew of something.
