Search Results

Search found 86 results on 4 pages for 'audacity'.

Page 3/4 | < Previous Page | 1 2 3 4  | Next Page >

• Unable to record from USB sound card line-in connector

  - by tete

  I've been googling and struggling with this for a while but haven't been able to make this work. I bought a USB sound card (Encore ENMAB 8-CM) to record from its line-in connector. I'm not sure which input to pick in Audicity, but I'me sure I've tested them all. These are all the available inputs I have in Audacity. But there's no sound being recorded at all. I've already checked the device is working, so I really don't know if I'm doing something wrong. Any help will be very appreciated. Cheers.

  Read the article

• How do I output my audio input?

  - by VarLogRant

  A few iterations ago, I think this was Jaunty but could've been before, I would plug a 1/8" audio cable from the line-out of a Windows netbook to the line-in of my Ubuntu machine, so I would have all the sound from both machines without having to plug both into a mixer which I don't have. I didn't do this much, as I was pretty-much happy with Banshee at the time. But with Karmic, and still with Lucid, I can only get the output if I'm recording with Audacity. Which I'm not going to do from my web-development and systems programming workstation. I can tell by plugging in headphones that my netbook has audio out working. I can see Sound Preferences that the Ubuntu machine is receiving them. I just want the old behavior back. Help?

  Read the article

• How to revert to 10.04

  - by Keith Mastin

  Since "upgrading" to 12.10, the multitude of problems and slowness has wondering me , if I'm running windows, so I want to take it back to 10.04. Just some of the problems that we never had in 10.4: Can't play YouTube and chat at same time; Can't open more than 5 photos in GIMP without constant grayouts; Can't easily close apps or programs on desktop; Can't Use Avidimux and Audacity at same time, CPU load stays at 100%; New Gnome is not nearly as intuitive as classic, focus is all over the place, have to constantly switch to have the focus on right window of same program (either browser), etc. Do I need to wipe my system partition and start over, or is there an easier way to downgrade?

  Read the article

• Password Authentication Problems

  - by Bobby Hathorn

  I am new to Ubuntu, am extremely delighted with the performance and speed, as compared to Windows 7-However, I messed up, I think...when I booted my USB disc, I set a password, as directed, and when Ubuntu booted up I tried to reset my password via User Accounts to "None". Now, the Password Authentication window prevents me from downloading software, (Audacity and my Ubuntu updates. Also, I've tried to boot into GRUB and the Recovery Console, as directed; however, the PC bypasses GRUB and boots into Ubuntu instead. Also, when attempting to use the terminal as directed to change the password, I'm given a password prompt there also. If the problem is on my end, could you email/reset my password? My PC is an emachines EL1358G. I am otherwise happy with Ubuntu!

  Read the article

• iPhone Game Developers - What does your toolchain look like?

  - by slf

  For example: source control: git + adobe drive 3d: google sketchup - *.dae - blender - *.obj 2d: photoshop/illustrator - *.png audio: audacity - *.caf code: ArgoUML, Xcode, Textmate test: OCUnit build: rake, Xcode Feel free to mention any other tools that you think are awesome :) Changed to Community Wiki

  Read the article

• Encoding MP3 and adding VBR or Xing headers (with lame or another method)

  - by J. Pablo Fernández

  I'm writing a program that converts wavs to mp3s, so far, by using lame. It's generating a command line more or less like this: "c:\Program Files (x86)\Lame for Audacity\lame.exe" --preset fast medium in.wav out.mp3 The problem I'm having is that no VBR or Xing headers are written to the MP3. How can I make lame.exe write those headers? Should I use another program to write those headers (platform is Windows, .Net 3.5)? Should I use another program for MP3 encoding?

  Read the article

• How to process audio in real time?

  - by user1756648

  I am giving some audio input through microphone. I recorded it in Audacity, it looks something like as shown below. I want to process this audio in real time. I mainly want to do this. 1) see real time audio amplitude vs time graph 2) perform some actions based on some thing (like if a specific type of hike is seen in audio, then do something, else do something else) Is there any python module or C library that can allow me to do this ?

  Read the article

• How to pass custom options to configure when building a package with debuild?

  - by TestUser16418

  Short background: I'm using Debian Sid. Currently the audacity package is conflicting with the pidgin package, because gstreamer0.10-plugins-bad are outdated. I'm trying to rebuild it, but one of the unit tests is failing as one plugin I don't need is causing a segfault. I need to disable these tests, and there's a configure option for that, but I don't know how to pass it. So, how can I run configure with custom options? Either by passing them to debuild, or by editing some file in the debian directory? I only worked with Gentoo ebuilds so far, which are extremely simple compared to the Debian control files, which I still find completely undecipherable.

  Read the article

• Combine Multiple Audio Files into a single higher-quality audio File

  - by namenlos

  BACKGROUND My team gave a demo to a large audience - we recorded the audio of the demo in multiple locations in the room (3) the audio was recorded using cheap laptop microphones I was not involved in the recording of the audio or the demo Both audio files suck in some form the first one is of a recording near the speaker - which clearly gets his voice but the the audience is audience is muffled - also this one is slightly noisy The second recording was done in the middle of the audience - it gets the audience questions clearly but actually gets the speaker rather sometimes well and sometimes poorly (not all the speakers spoke loudly enough to be heard) MY QUESTION Is there any techinque or software which can be used to merge these audio files in such a way that the best qualities of each are preserved. I am NOT asking now to simply merge them together in one track - I've already done that in Audacity and it is certainly better - what I am looking for could be considered closer to how HDR images are created - multiple exposures combined into an enhanced new version which is not simply an average of the inputs. NOTE Am not an "Audio" guy - just a normal user

  Read the article

• What is a good alternative to Camtasia Studio (screen recording not required)?

  - by tnorthcutt

  I'm looking for a good alternative to Camtasia Studio - really just the video editing part. My current workflow looks like this: Record audio (with Audacity) Save screenshots of the subject matter (websites) Import audio and screenshots into Camtasia Use the zoom and callout tools in Camtasia to go along with the spoken audio and highlight certain parts of the sites Given that Camtasia costs $300 (I've been using a trial version), I'd like to find something else that can do this well. I suspect that a lot of the pricetag for Camtasia is because of the screencasting features. I've tried Sony Vegas Movie Studio, but it was really a lot more than I need. I'd like something relatively simple (so that new users can pick it up relatively quickly), with the zoom and callout functionality that Camtasia has. Any suggestions? Edit for clarity: I'm looking for a program that will let me combine the screenshots and audio, and (most importantly) that has features equal to or better than Camtasia for text overlays, highlighting areas on the screen, etc.

  Read the article

• Setting up podcasting for a non-tech user

  - by Force Flow

  I have a user who wants to start making podcasts, but they only have basic skills when it comes to technology. So, I was trying to get a process together that would be easy for them to follow. To upload files (the mp3's and rss feed files), I have an explorer shortcut for their FTP space. To record the podcast, I was going to either use audacity or PodProducer. For the RSS feed, I was looking for a podcast RSS generator of some sort. In my search for this, I've come across a lot of dead links and a lot of paid tools, so I haven't come up with anything too useful. Is there a free, reliable webservice or windows-based tool available that folks like to use?

  Read the article

• How can I split a stereo audio track of a movie into two separate audio tracks?

  - by pesche

  I often record TV shows with a hard disk recorder/DVD writer, burn them as VRO file and convert to MP4 with Handbrake. The shows are bilingual broadcasts with two mono audio channels instead of a stereo one: dubbed voice on the left, original voice on the right. The TV set and VLC are both perfectly capable to play only the left or the right channel, but other video players may just offer to select between different stereo audio tracks (like they are present on many DVDs). I'd like to have an easy process to create MP4 or MKV files of these shows where the two audio channels are split into two separate audio tracks. The only way that I know of is to extract the audio track (e.g. using MPEG Streamclip), split it into two tracks using an audio tool like Audacity and then merge the audio tracks back (using a DVD authoring software, don't remember all details). Clearly not a thing to repeat regularly. Preferably a solution should run on Mac OS X, but Linux or Windows solutions are very welcome, too.

  Read the article

• ffmpeg volume parameter format

  - by tanon

  ffmpeg's -vol parameter is confusing me. 256 => normal (i guess meaning same as input volume, no change) 512 => (double the volume - read this somewhere). So what to do for 3 times the volume? 1.5 times the volume? Basically, lets say I have the max sound amplitudes (audacity levels) in 3 files as: 0.8 0.6 0.9 I want to amplify in the first two files, so that max=0.9 in all files. What parameters of -vol I would use?

  Read the article

• Audio splitting and noise removal on Windows

  - by pts

  My mother has about 100 hours of audio in a mix of MP3 and WAV files, the digitized versions of her vinyl records. Each file contains about 5 songs with a few seconds of (noisy) pause between them. My mother needs software for Windows XP with which she can listen to the files, find the gaps manually, split the files at the gaps found, reduce noise on each song, and export the songs to individual MP3 files. My mother has very limited software user skills and affinity, and she doesn't speak English. The simpler the software, the better for her, even if noise reduction is worse than with a more sophisticated, but more complicated software. I'd prefer free software, freeware or shareware (which can do all above). Please recommend something much simpler than Audacity. The software should guide the user through the process, always showing the next few available steps, and being intuitive in the sense that there are only a few allowed actions and it's obvious what they are and how to activate them. Which software would you recommend?

  Read the article

• On Windows 7, how can I tell if a recording is multi-channel without third party tools? [migrated]

  - by engineerchuan

  A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

  Read the article

• Record the audio stream from HDMI monitor

  - by Nick

  I am trying to record sound playing on my comupter with Audacity but am running into some troubles. I have the stereo mix set to be the default audio recorder but it doesn't pick up the audio that is being played through my HDMI monitors speakers: Playback Recording When I plug in headphones the stereo mix will pick up the audio stream and I can record but not when playing through the HDMI. I have installed the latest audio drivers and have tried all the different record options to no avail. How can I capture the Audio stream going through the HDMI?

  Read the article

• On Windows 7, how can I tell if a recording is multi-channel without third party tools?

  - by engineerchuan

  A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

  Read the article

• Stereo Mix does not work

  - by rfw

  Stereo Mix no longer seems to work on my computer -- it did once, but now I am unable to receive any sort of audio from it. When trying to record it with programs such as Audacity, it reports that there is an error opening the sound device. Additionally, I can make no changes whatsoever to Stereo Mix, such as with regards to default format, where it simply reports Format not supported by device. I am sure that Stereo Mix had definitely worked in the past, so does anyone have any idea as to why Stereo Mix would suddenly break? (I have no applications taking exclusive control of Stereo Mix)

  Read the article

• DAW with realtime pitchshifting

  - by monov

  I'm very happy with Renoise but it has a problem, it can't do realtime pitchshifting. For example, consider a breakbreat which I want to trigger at different pitches sometimes. Like I keep 'Q' pressed for a 'C' pitch, but then I press 'B' for a lower pitch. In Renoise the resulting beat is not only lower pitch, but also longer/slower. I want it to be the same speed, just a different pitch. I've been doing this externally in Audacity, then keeping a couple of different pitched versions in Renoise but that's tedious. Or consider a melodic segment snipped from a song into a sample. Say I want to play it simultaneously in its real pitch and 5 semitones higher, so it forms a sort of chord effect. Again, no easy way to do this in renoise. Is there a DAW app that does that kind of thing? I've heard of ableton live, it has 'live' in the name so maybe it can do it?

  Read the article

• How to negotiate with software vendors who do not follow HL7 standards

  - by Peter Turner

  Take, for instance the "", I'd hope that anyone who has spent any time in dealing with HL7 messages knows that the "" signifies that something should be deleted. "" is not an empty string, it's not a filler etc... But occasionally, one may meet a vendor who persists in sending "" instead of just sending nothing at all. Since, I work for a small business and have an extremely flexible HL7 interface, I can ignore ""'s in received messages. But these things are adding up. Some vendors like to send custom formatted fields with psuedo-components that they leave others to interpret themselves. Some vendors send all their information in note segments and assume you're going to only show users the information they send in a monospace font. Some vendors even have the audacity to send Carriage Return Line Feeds at the end of each line of a file interface. Some vendors absolutely refuse to send decimal numbers and in-so-doing refuse to send any numbers. So, with all this crippling humanity against the simple plastic software man, how does one bend without breaking*? Or better yet, how does one fight back and still make money? *my answer is usually to create an interface for the interface and keep the HL7 processing pure, but I don't think this is the best solution

  Read the article

• After upgrading to trusty, ALSA midi connection (aconnect) doesn't seem to work right

  - by SougonNaTakumi

  Previously in kubuntu 13.10 I was able to open vmpk or plug in a midi keyboard, and provided that TiMidity was running in server mode, I could run aconnect [keyboard port (129:0 for vmpk)] 14:0 aconnect 14:0 128:0 and I could play the keyboard and get sound. But now, a while after upgrading to trusty, I tried to do that, and didn't get any sound. TiMidity itself still plays files fine, but if I try to play them with aplaymidi, I still just get silence. Oddly, the midi files are clearly being read. When I ran (where 130:0 was vmpk's input port) aplaymidi -p 130:0 ~/path/to/midi.mid vmpk was highlighting notes on the piano as if it were playing the midi. One time I tried this, TiMidity (?) very briefly played a fraction of a second of the first chord of my song before everything went silent and vmpk just highlighted the first voice on the keyboard as usual. Now the weirdest part of this is that probably about 40% of the time, when I've played at least one note with either aplaymidi or vmpk, when I run aconnect -x I get a sudden burst of a note or chord from my speakers (that is, if I played one note, I get a note; if I played multiple sequential notes, they turn into a chord), as if the notes were being queued up but not being played and that somehow liberated them. I have no idea what's going on there. A little while ago I remember having a problem with Audacity playing wav files sped up and also locking up if I tried to pause it, which it stopped doing when I set the audio devices to the actual audio devices rather than pulse. But now when I checked again, it's doing the opposite: it won't play audio at all and/or acts weirdly if I don't set the audio devices to pulse, and either way will very occasionally randomly do the speeding up thing regardless. Oddly in the midst of what's looking like a pretty screwed up sound system, sound in VLC and Firefox has been working fine and if I play a wav file with aplay ~/path/to/sound.wav that works fine too. Any idea what I could do to figure out what's wrong with ALSA and/or fix it?

  Read the article

• Solutions for cheaply replacing poorly-supported onboard ATI card with discreet graphics on desktop machine?

  - by echo-flow

  I have put Ubuntu on my mum's desktop computer. Unfortunately, the open source radeon driver does not work well with the onboard ATI graphics, and ATI's proprietary driver no longer supports the hardware at all. In order to use the ATI proprietary driver with this hardware, it is necessary to use an older version of Xorg, which is now only available in versions of Ubuntu older than 8.10. Unfortunately, the open source radeon driver seems to be causing X to lock up intermittently when my mum uses Audacity. I'm willing to accept that some hardware is not well-supported on Ubuntu, and so, because this is a desktop computer with a couple of free PCI slots, I think a better solution might simply be to plug in a new graphics card that might have better driver support, and to disable the onboard ATI card in the BIOS. The requirements for this card are that it be inexpensive and have robust (preferably open source) driver support in Ubuntu 10.04. Heavy-duty graphics processing power is not a requirement. A second-hand card on Ebay would also be fine. Can anyone make some recommendations?

  Read the article

• Manipulating multi-track ogg files programatically

  - by Chad Birch

  I'm planning to create a program for manipulating multi-track OGG files, but I don't have any experience with the relevant libraries, so I'm looking for recommendations about which language/library to use for this. I don't really have any preference for the language, I'll happily code it in C, C#, Python, whatever makes things the easiest (or even possible). Perhaps it's even a possibility to automate Audacity somehow? In terms of requirements, I'm not looking for anything particularly fancy. It will probably be a command-line program, I don't need to be able to play the audio, draw image representations of the waveforms, etc. The program will basically be used as a converter, but I need to do some processing before outputting. That is, I need the ability to programatically remove some tracks, set panning per-track, change track volumes, etc. Nothing too complex, just some basic processing, and then output the result in either MP3 or a format easily converted to MP3, such as WAV. Any suggestions or general information would be appreciated, thanks.

  Read the article

• How to stream a WAV file?

  - by jonasb

  I'm writing an app where I record audio and upload the audio file over the web. In order to speed up the upload I want to start uploading before I've finished recording. The file I'm creating is a WAV file. My plan was to use multiple data chunks. So instead of the normal encoding (RIFF, fmt , data) I’m using (RIFF, fmt , data, data, ..., data). The first issue is that the RIFF header wants the total length of the whole file, but that is of course not known when streaming the audio (I’m now using an arbitrary number). The other problem is that I'm not sure if it's valid since Audacity doesn't recognise the file, and Windows Media Player opens the file but plays only a very small part. I've been reading WAV specs but haven’t found an answer. Any suggestions?

  Read the article

• Workaround for UnsupportedAudioFileException ?

  - by tschan

  I'm in a very early stage of writing a small music/rhythm game in Java (via Slick framework, which in turns uses OpenAL, but that's probably irrelevant here). The game needs to read (and playback) several sound files in WAV format, but some of the files are throwing [javax.sound.sampled.UnsupportedAudioFileException] exceptions. at javax.sound.sampled.AudioSystem.getAudioInputStream(AudioSystem.java:1102) at org.newdawn.slick.openal.WaveData.create(WaveData.java:123) at org.newdawn.slick.openal.SoundStore.getWAV(SoundStore.java:713) at org.newdawn.slick.openal.SoundStore.getWAV(SoundStore.java:683) at org.newdawn.slick.Sound.<init>(Sound.java:33) The files can be played back just fine in Winamp or Foobar2000, so this means Java just don't recognize some variants of the file format. What are my options at this point? Note: The files in question are user-supplied, so i cannot just convert them beforehand (using something like audacity). Any conversion steps must be done at runtime.

  Read the article

< Previous Page | 1 2 3 4  | Next Page >