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  • WMV will play audio but not show images (dark screen)

    - by GUI Junkie
    This happens on one computer. On the other computer I can see the video. I've installed GStreamer, FFMpeg and Ubuntu Restricted Extras. On the computer where it doesn't work, I've tried both Movie Player and VLC. There must be some difference between the two computers, but how can I obtain all the system information in such a way that a diff could be reasonably done? Or just any other suggestion on how to solve this would be appreciated. :-)

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  • HP dv7 Beats Audio Subwoofer on while headphones plugged in

    - by msilis
    I have looked for answers to this but have not found anything. People have similar problems but mostly with getting the subwoofer to work at all. I got the subwoofer to work by editing the /etc/modprobe.d/alsa-base.conf file and adding options snd-hda-intel model=ref. When I plug in my headphones, the main speakers turn off but the subwoofer still has sound coming through it. The output is selected automatically in sound preferences and is set to headphones when they are plugged in. I also have tried muting the main speakers, which in turn mutes the subwoofer, before plugging in the headphones as they seem to have two different volume controls, but as soon as the headphones are plugged in, the subwoofer has sound coming out of it. I am running Ubuntu 11.10 64 bit on HP dv7- 4285dx. I am able to just not have the subwoofer on but since I have gotten it to work I would like to keep it around and not have to change a config file every time I want to plug in my headphones. Any ideas or suggestions would be greatly appreciated. Thanks

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  • Audio not working for Gnome Screencast Ctrl + Alt + Shift + R

    - by Costa
    I can do screencasts by pressing ctrl+alt+shift+r, but I get no sound when I view the videos, I've check my built in mic and headset, they both work on skype and such, I just can't record sound with the gnome built in screencasting. Also, when I open the videos in movieplayer they are in mute by default and there's nothing the the preferences I can find to change that. Any help would be awesome!

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  • Mass audio encoder

    - by bessman
    I have a few thousand FLAC files which I would like to transcode to OGG Vorbis, but I can't find any suitable tools for the job. To name a few I have tried so far and why they are unsuitable: oggenc is single-threaded and would require me to automate it myself, mencoder requires the input to also contain video, and abcde assumes the input is a CD. The ideal tool should be multi-threaded, and support inputing multiple files located in different directories simultaneously. CLI or GUI makes no matter. Does such a tool exist?

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  • ALSA Mutes Incorrect Audio Device on Headphone Event

    - by Gardiner
    I know what the problem is, I just don't know how to prevent it. When I plug in my headphones, ALSA automatically mutes (and sets the volume to 0) the 'Speaker' entry in alsamixer. If I go in and manually unmute (and turn the volume up) I have sound from my headphones. Question is, how do I change this behavior? I want it to adjust the volume to half, but not mute or set the volume to 0. I'm using Ubuntu Gnome 13.04 Output of amixer scontents Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] This answer solved my issue: How can I get sound on headphones without switching back to 'Speakers' manually?

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  • USB audio device not detected [solved]

    - by user199052
    I've been having problems with USB sound for some time (it used to work about 6 months ago). I find that any USB sound device that I connect is not recognised, and is not listed by aplay -l I've tried disabling pulseaudio. I've tried purging and re-installing alsa-base, pulseaudio, and pavucontrol. To no avail. I'm using 12.04 LTS. I found the solution: 12.04 does not have the needed kernel module snd-usb-hiface this is introduced in the 3.11.1 kernel and installing that on 12.04 gives USB sound for my equipment. How to install the 3.11.1 kernel is given here:- installing 3.11.1 kernel

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  • How to send audio stream via UDP in java?

    - by Nob Venoda
    Hi to all :) I have a problem, i have set MediaLocator to microphone input, and then created Player. I need to grab that sound from the microphone, encode it to some lower quality stream, and send it as a datagram packet via UDP. Here's the code, i found most of it online and adapted it to my app: public class AudioSender extends Thread { private MediaLocator ml = new MediaLocator("javasound://44100"); private DatagramSocket socket; private boolean transmitting; private Player player; TargetDataLine mic; byte[] buffer; private AudioFormat format; private DatagramSocket datagramSocket(){ try { return new DatagramSocket(); } catch (SocketException ex) { return null; } } private void startMic() { try { format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 8000.0F, 16, 2, 4, 8000.0F, true); DataLine.Info info = new DataLine.Info(TargetDataLine.class, format); mic = (TargetDataLine) AudioSystem.getLine(info); mic.open(format); mic.start(); buffer = new byte[1024]; } catch (LineUnavailableException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } private Player createPlayer() { try { return Manager.createRealizedPlayer(ml); } catch (IOException ex) { return null; } catch (NoPlayerException ex) { return null; } catch (CannotRealizeException ex) { return null; } } private void send() { try { mic.read(buffer, 0, 1024); DatagramPacket packet = new DatagramPacket( buffer, buffer.length, InetAddress.getByName(Util.getRemoteIP()), 91); socket.send(packet); } catch (IOException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void run() { player = createPlayer(); player.start(); socket = datagramSocket(); transmitting = true; startMic(); while (transmitting) { send(); } } public static void main(String[] args) { AudioSender as = new AudioSender(); as.start(); } } And only thing that happens when I run the receiver class, is me hearing this Player from the sender class. And I cant seem to see the connection between TargetDataLine and Player. Basically, I need to get the sound form player, and somehow convert it to bytes[], therefore I can sent it as datagram. Any ideas? Everything is acceptable, as long as it works :)

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  • Streaming Audio from A URL in Android using MediaPlayer?

    - by Sena Gbeckor-Kove
    Hi, I've been trying to stream mp3's over http using Android's built in MediaPlayer class. The documentation would suggest to me that this should be as easy as : MediaPlayer mp = new MediaPlayer(); mp.setDataSource(URL_OF_FILE); mp.prepare(); mp.start(); However I am getting the following repeatedly. I have tried different URLs as well. Please don't tell me that streaming doesn't work on mp3's. E/PlayerDriver( 31): Command PLAYER_SET_DATA_SOURCE completed with an error or info PVMFErrNotSupported W/PlayerDriver( 31): PVMFInfoErrorHandlingComplete E/MediaPlayer( 198): error (1, -4) E/MediaPlayer( 198): start called in state 0 E/MediaPlayer( 198): error (-38, 0) E/MediaPlayer( 198): Error (1,-4) E/MediaPlayer( 198): Error (-38,0) Any help much appreciated, thanks S

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  • Is the Finch audio library for iPhone capable of doing this?

    - by mystify
    I need to: - start / stop sounds with lengths between 0.1 and 10 seconds - change the playback volume I want to / would like to / would be nice to have to: - change the playback speed - change the playback pitch / frequency - pause an sound and resume playing it later - play a sound backwards Is Finch my best friend here?

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  • how to get audio frequency data from a wave file?

    - by potlee
    I want to build a speech recognition engine in ruby. I know i'll never get there, doing it just for fun. I need to get data for the frequencies of the sound stored in a wav file to compare with data i already have of different sounds that i want to recognize. I will write the code in ruby but i dont think there are any libraries for this written in ruby, they would be too slow if there were any anyway. The good thing about ruby is I'll be able to use libraries for .net via IronRuby or Java via Jruby. How can i get the frequency data?

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  • How to route KVM virtual machine audio to Ubuntu 11.10 host using virt-manager?

    - by iGadget
    I've been using KVM in combination with Virt-Manager and Remmina at a fair success up until now. The issue I need to solve now is to get audio from a virtualized Windows XP and make it audible on the Ubuntu 11.10 host. Remmina / RDP works for 'simple' audio (system sounds and such), but when the source gets trickier (e.g. Flash audio), Remmina / RDP messes up. So I figured I'd just connect to the machine directly using Virt-Manager. Unfortunately, it seems that even though I have successfully configured the AC97 audio device on WinXP, it's unable to get it's output to the Ubuntu host. This is probably because Virt-Manager uses VNC (and AFAIK, VNC doesn't transport audio). Does anyone know if there is a solution to fix this? I've heard of Spice, but the installation required so much voodoo last time I checked, I figured I'd let that solution boil to maturity a little longer ;) But perhaps there are other options I haven't thought of yet (which don't require switching to VirtualBox / VMware)...

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  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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  • Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

    - by Ryan Thompson
    To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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  • Terrible noises from subwoofer of ACER Aspire 6930 with Realtek sound chip

    - by OneWorld
    After approximately 5-15 min of listening to music my subwoofer begins to make terrible noises. He's just "coughing". That began after 6 months I had this computer. Now I found out, that I can temporarily fix this problem by "restarting" the audio stream of the application that plays music. For example reloading last.fm page (reloads the flash file). Another way to reset the audio playback is switching the speaker configuration shown below in the screenshot. According to many posts on the internet like http://www.tomshardware.co.uk/forum/52918-20-acer-aspire-6935g-speaker-problem ACER support isn't any help Exchanging hardware doesn't fix the problem Even the later models have this problem Turning off the volume of the subwoofer is not an option to me. I still have warranty (I bought an extension of one year). I already tried about 15 versions of the Realtek driver with no success. I am not sure but MAYBE the problem did not occur on the original windows vista that was shipped with this computer. However, I removed the original windows for good reasons (english). What do you suggest me? Did anyone fix this problem? Maybe by writing a script which resets the audio streams every 5 minutes? Shall I take the effort to deal with the acer support until they give me another model? (I won't have a computer than for a longer time, will spend money on telephone hotlines (1,30 EUR / min)......) Here are additional infos, if they are any help: Windows 7 64 Bit (Original was Windows Vista Home Premium 32 Bit) All specs Audio driver version:

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