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  • How do I restore the Gnome Panel volume control in Ubuntu 10.04?

    - by Neil
    I use alsa, and I don't have a volume control applet on my Gnome Panel. When I right click and select "add to panel", there is nothing that has to do with "sound", "audio" or "volume" in the list, and the "Indicator Applet" or "Indicator Applet Session" things have no volume controls, or properties that would let you enable any sort of volume control. How can I get a volume control in Ubuntu, so I don't have to run aumix in a terminal or something? I've been using Linux since Redhat 5, it's beyond me why these sorts of problems are still around. Someone should just put a damn "Volume Control" element in the list of things to add to the panel, even if it doesn't work, perhaps showing an error message.

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  • linux mint VIA sound issue

    - by user2699451
    So I installed linux Mint 15 "Olivia" 64 bit on my Mecer W550EU laptop I have HD Audio with a VIA chipset charles-W55xEU charles # lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_via 51018 1 snd_hda_intel 39619 5 snd_hda_codec 136453 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13602 1 snd_hda_codec snd_pcm 97451 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 1 snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 19 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_via,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd And my sound card charles-W55xEU charles # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1802 Analog [VT1802 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1802 HP [VT1802 HP] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 and my audio device charles-W55xEU charles # lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High `Definition Audio Controller (rev 04)` Subsystem: CLEVO/KAPOK Computer Device 0550 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at f7c10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Sometimes when I boot up, soundworks, other times it doenst, it is completely random, so far, no-one on xchat linux help or linux mint forums was able to help me, I have always had issues with sound on VIA chipsets I have: sudo apt-get upgrade && apt-get install mint-meta-cinnamon it seemed to help but after 2-3 reboots, the problem came back, btw, everytime I checked, pulse audio is selected to Duplex Audio Input & Output and alsa mixer is always unmuted!

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  • Strange sound behavior

    - by caarlos0
    First of all, sorry for the tittle. English isn't my native language, and I can't find a word to describe the strange behavior I'm getting here. In the most simple explanation, is like sound keeps going down and up again... Think in a kid with that old radios that have a circled volume button. Think like these kid keeps "turning" the volume button. That is the behavior I'm getting here. At first, I believed that it was a pulseaudio issue, but, it isn't. I followed the wiki part I think that should be my problem, but it didn't work. After that, as I'm using XFCE, I didn't really need pulseaudio, so I removed it and stays with a clean alsa, hopping that will fix my problem. Sweet mistake. It really looks like a kid looking for trouble. I believed it worked, and, suddenly, here is the same issue again. BTW: I have a full-upgraded testing system (yeah, I upgraded to testing hopping for new pulseaudio version which fix the issue), no pulseaudio at all, just xfce, started with startxfce. What can I do to fix this? It's extremely annoying... sometimes I just want to throw my laptop in the wall because of this. Any extra info you need, please, tell me. Thanks in advance -- EDIT: My alsamixer is like this: And here is a video with the sound behavior.

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  • Real Time Progress Indicator

    - by jignesh
    I am trying to get the real time progress indicator. In the code snippet below,i am trying to set the width of DIV on the basis of the Number of Items retrieve from the Web Service Call Back Function. for (i = 0; i < data.length; i++) { table += '<tr>'; table += '<td>' + data[i].ItemId + '</td>'; table += '<td></td>'; table += '<td>' + data[i].Name + '</td>'; table += '<td>' + data[i].Unit + '</td>'; table += '<td><input type=text value=' + data[i].Quantity + '></td>'; table += '<td>' + data[i].Brands + '</td>'; table += '<td><img border=0 src=../images/Delete_icon.gif></td>'; table += '</tr>'; $('#divProgressIndicator').width((100 * (i / data.length)) + '%'); } As shown in the code above,after the for loop is executed i only see the divProgressIndicator filled with the background color and I am not able to see the progress. Can I achieve this in a for loop or i need to use some thing else. Regards

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  • Computer making strange sound when turned on, ever since power outage

    - by Dot NET
    Recently we experienced a power outage, and the PC was off. However, once the power came back, I switched on the PC and heard a strange noise - almost as if the hard disk or fans were struggling to work. I can't really describe the sound, but it's a laboured, loud sound almost like a jack-hammer. This has been persisting ever since the power outage, however the noise stops after around 10 minutes or so, and doesn't start again until the computer is turned off and on again. At first I thought it had something to do with the HDD, but all my files are intact, chkdsk did not report any issues and performance is 100% unchanged, even in games (so the gfx card is fine, and so is the HDD most likely). My PC setup basically has around 3 cooling fans, but I'm not sure if it's one of these either as the noise actually stops after 10 minutes or so, and if I leave the PC on for 4 hours (for example) the noise never starts again. It's there solely when turning on the PC. I haven't got a UPS, and it's important to note that the computer was not on when the power went out - it was merely plugged in. I then promptly unplugged the PC once the power was out, and only plugged it in again when the power came back. Could it be the power supply? Unfortunately I can't open my tower as I would void the warranty. Are there any tests which I could carry out without voiding the warranty?

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  • Reset sound volume in Windows 8

    - by Svish
    There seems to be a bug in Windows 8 causing the maximum volume to become lower than it really should be. I'm now at a stage where I put the volume up to max and the sound is still very low. I have a couple of Logitech Z-10 speakers with a display on them and when I touch the increase volume button on that it shows me the volume (but not able to increase it) is actually below middle even though Windows claims it to be maxed out. Is there any way I can reset the volume in Windows 8 so that I can get it up fully max again? A registry setting or something? Really don't want to have to reinstall windows or drivers or anything like that cause if it is a bug it'll probably happen again and I really don't want to have to do that every time this happens :p Any ideas? Manged to fix it by unplugging the usb connection to my speakers, turning the volume down on my computer and up on the speakers, and finally reconnecting the usb connection. Seems to have been an issue with the speakers and not Windows this time. BUT, I'm still curious how you can adjust/reset the Windows 8 sound volume without using the volume controls. Like, where is the value of the current volume setting(s) really stored? And can you manually adjust them?

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  • Default soundcard per application in Vista?

    - by Liquid5n0w
    I use a USB headset during the day, then at night I use my speakers to watch video. I constantly have to use the windows playback list to change the default soundcard before I run the program I want. I did some google research but it doesn't seem to help much, mostly because of false positives. I tried to use "Sound Card Manager" but it didn't seem to work properly. Has anyone had any sucess dealing with the Vista sound system?

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  • how to disable all system sounds - permanently

    - by studiohack
    I prefer the sound scheme of Windows 7 HP to be set to "No Sounds" ALL the time. The problem is, when I switch themes, the themes switch sound schemes also. I know I can go in and disable/change to "No Sounds" every time by hand, but is there a way to turn it off and keep it off, regardless of the theme? Thanks!

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  • Why is the disk making my motherboard beep?

    - by Mark Ransom
    Whenever I let my PC do heavy disk accesses for a long time, the speaker on the motherboard starts making a continuous chirping sound. Thankfully it doesn't happen often, but it drives me nuts when it does. Anybody know where this sound might be coming from, or have any hints as to how to track it down? Edit: The problem appears to be with the processor, the correlation with disk access was coincidental. Thanks for all the answers.

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  • Convert file from VOC to MP3

    - by Thomas
    I would like to convert a sound file (from a digital voice recorder) with the extension .voc to an .mp3 file or some other common sound files. I am on Windows 7 64 bit. I have tried the program voc2wav but it gives me an error message saying that the program isn't 64 bit. The program has to be free and able to run without installing. (The voice recorder did come with a program that I could install, but I would like to avoid that).

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  • Ubuntu Touch Porting - Audio

    - by user205695
    I'm currenty trying to port Ubuntu Touch to the Galaxy s4 International LTE (GI9505/ jfltexx). I've come to the point where I need to create a UCM directory but I don't know where and how I should call it. By "looking at /usr/share/alsa/ucm/apq8064-tabla-snd-card/" is the local Ubuntu PC directory or a directory on the downloaded CM meant? Same thing for "/proc/asound/cards" which should give a hint about what the directory should be called. 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xfb200000 irq 51 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfb080000 irq 17 I dont think the directory should be called anything like this. Thanks for the help Robin Kertels

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  • Google Talk Plugin in GMail on MacBook 2,1

    - by jrc03c
    I'd like to use the chat section in GMail to make phone calls. I've downloaded and installed the Google Talk plugin, and it acts like it knows what it's doing. But when I try to make calls, the internal laptop mic doesn't work at all (i.e., no one on the other end can hear me). In the GMail chat settings, I've tried selecting "Default Device" for the microphone, as well as "Internal Audio Analog Stereo." No matter which setting I try, none seem to work. As I said at the top, this is only a problem in Ubuntu; it works just fine in OSX and Windows (which means that yes, my Google Voice account is properly configured). Here are my tech specs: Ubuntu 10.10 Kernel Linux 2.6.35-24-generic Gnome 2.32.0 Google Chrome 8.0.552.237 Google Talk Plugin (google-talkplugin) 1.8.0.0-1 MacBook (2,1) w/ internal microphone Any help will be greatly appreciated! Thanks!

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  • How do I use my headphones and microphone?

    - by Pavan Kumar
    Hello, The headphones and microphone on my Ubuntu 10.10 work fine. But when I start an audio conversation using empathy or pidgin, my computer hangs, the microphone doesn't work, and I can't record anything. I have tried sudo alsa force-reload, I have installed pavucontrol, but nothing works. I can't increase the volume of the headphones and master channel using alsamixer though they are unmuted. (Everything works fine in Fedora and Windows XP.) How do I fix this? Thank you.

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • Ardour wont start Jack problem

    - by Drew S
    I downloaded Ardour yesterday, it worked, edited an audio file done. Come back today it wont start I get this: Ardour could not start JACK There are several possible reasons: 1) You requested audio parameters that are not supported.. 2) JACK is running as another user. Please consider the possibilities, and perhaps try different parameters. So I try and look at qjackctl to see what happening there. When I try to start JACK I get D-BUS: JACK server could not be started. then Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. and this is the message box in JACK. 15:22:12.927 Patchbay deactivated. 15:22:12.927 Statistics reset. 15:22:12.944 ALSA connection change. 15:22:12.951 D-BUS: Service is available (org.jackaudio.service aka jackdbus). Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started 15:22:12.959 ALSA connection graph change. 15:22:45.850 ALSA connection graph change. 15:22:46.021 ALSA connection change. 15:22:56.492 ALSA connection graph change. 15:22:56.624 ALSA connection change. 15:23:42.340 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Wed Oct 23 15:23:42 2013: Starting jack server... Wed Oct 23 15:23:42 2013: JACK server starting in realtime mode with priority 10 Wed Oct 23 15:23:42 2013: ERROR: Cannot lock down 82274202 byte memory area (Cannot allocate memory) Wed Oct 23 15:23:42 2013: Acquired audio card Audio0 Wed Oct 23 15:23:42 2013: creating alsa driver ... hw:0|hw:0|1024|2|44100|0|0|nomon|swmeter|-|32bit Wed Oct 23 15:23:42 2013: ERROR: ATTENTION: The playback device "hw:0" is already in use. The following applications are using your soundcard(s) so you should check them and stop them as necessary before trying to start JACK again: pulseaudio (process ID 2553) Wed Oct 23 15:23:42 2013: ERROR: Cannot initialize driver Wed Oct 23 15:23:42 2013: ERROR: JackServer::Open failed with -1 Wed Oct 23 15:23:42 2013: ERROR: Failed to open server Wed Oct 23 15:23:43 2013: Saving settings to "/home/drew/.config/jack/conf.xml" ... 15:26:41.669 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started 15:26:49.006 D-BUS: JACK server could not be started. Sorry Wed Oct 23 15:26:48 2013: Starting jack server... Wed Oct 23 15:26:48 2013: JACK server starting in non-realtime mode Wed Oct 23 15:26:48 2013: ERROR: Cannot lock down 82274202 byte memory area (Cannot allocate memory) Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Wed Oct 23 15:26:48 2013: ERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:26:48 2013: ERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:26:48 2013: ERROR: Audio device hw:0 cannot be acquired... Wed Oct 23 15:26:48 2013: ERROR: Cannot initialize driver Wed Oct 23 15:26:48 2013: ERROR: JackServer::Open failed with -1 Wed Oct 23 15:26:48 2013: ERROR: Failed to open server Wed Oct 23 15:26:50 2013: Saving settings to "/home/drew/.config/jack/conf.xml" ... 15:26:52.441 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started 15:26:55.997 D-BUS: JACK server could not be started. Sorry Wed Oct 23 15:26:55 2013: Starting jack server... Wed Oct 23 15:26:55 2013: JACK server starting in non-realtime mode Wed Oct 23 15:26:55 2013: ERROR: Cannot lock down 82274202 byte memory area (Cannot allocate memory) Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Wed Oct 23 15:26:55 2013: ERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:26:55 2013: ERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:26:55 2013: ERROR: Audio device hw:0 cannot be acquired... Wed Oct 23 15:26:55 2013: ERROR: Cannot initialize driver Wed Oct 23 15:26:55 2013: ERROR: JackServer::Open failed with -1 Wed Oct 23 15:26:55 2013: ERROR: Failed to open server Wed Oct 23 15:26:57 2013: Saving settings to "/home/drew/.config/jack/conf.xml" ... 15:26:59.054 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started 15:29:24.624 ALSA connection graph change. 15:29:24.641 ALSA connection change. 15:33:11.760 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Wed Oct 23 15:33:11 2013: Starting jack server... Wed Oct 23 15:33:11 2013: JACK server starting in non-realtime mode Wed Oct 23 15:33:11 2013: ERROR: Cannot lock down 82274202 byte memory area (Cannot allocate memory) Wed Oct 23 15:33:11 2013: ERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:33:11 2013: ERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0 Wed Oct 23 15:33:11 2013: ERROR: Audio device hw:0 cannot be acquired... Wed Oct 23 15:33:11 2013: ERROR: Cannot initialize driver Wed Oct 23 15:33:11 2013: ERROR: JackServer::Open failed with -1 Wed Oct 23 15:33:11 2013: ERROR: Failed to open server Wed Oct 23 15:33:12 2013: Saving settings to "/home/drew/.config/jack/conf.xml" ... 15:34:09.439 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started

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  • Static background noise while using new headset Ubuntu 13.04

    - by ThundLayr
    Today I bought a new gaming headset (Gx-Gaming Lychas), and when I tried to record some gameplay-comentary I noticed that there always is a static background noise, I just recorded an example so you guys can listen it (no downloaded needed): http://www47.zippyshare.com/v/65167832/file.html I'm using Kubuntu 13.04 and Kernel version is 3.8.0-19, my laptop is an Acer Travelmate 5760Z, I tried tons of configurations on Alsamixer and none of them made result, I really need to get this working so any kind of help will be very aprecciated. cat /proc/asound/cards: 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc6400000 irq 44 cat /proc/asound/card0/codec#0 Codec: Conexant CX20588 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506c Subsystem Id: 0x10250574 Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x4a 0x4a] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Control: name="Headphone Jack", index=0, device=0 Pincap 0x0000001c: OUT HP Detect Pin Default 0x04214040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x90a70130: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Control: name="Mic Jack", index=0, device=0 Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x0: Pin Default 0x04a19020: [Jack] Mic at Ext Right Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Control: name="Speaker Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono

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  • CMUS Error: opening audio device: No such device

    - by clamp
    I cant seem to play any audio with CMUS because it always gives the above error the output of lspci -v | grep -A7 -i "audio" gives 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 02) Subsystem: ASRock Incorporation Device c892 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at dff00000 (64-bit, non-prefetchable) [size=16K] Capabilities: Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation NM10/ICH7 Family PCI Express Port 1 (rev 02) (prog-if 00 [Normal decode]) what could be the problem?

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  • Ubuntu 11.10 - How can i stop self-feedback-loop, coming directly from my microphone to speaker?

    - by YumYumYum
    I have microphone audio, which comes instantly to my speaker. I am using default pulseaudio and alsa from the package. I have tried to setup: 1) PA /etc/pulse/default.pa /etc/asound.conf $ ls analog-input-aux.conf analog-input-fm.conf analog-input-mic.conf analog-input-tvtuner.conf analog-output-desktop-speaker.conf analog-output-mono.conf analog-input.conf analog-input-front-mic.conf analog-input-mic.conf.common analog-input-video.conf analog-output-headphones-2.conf analog-output-speaker.conf analog-input.conf.common analog-input-internal-mic.conf analog-input-mic-line.conf analog-output.conf analog-output-headphones.conf iec958-stereo-output.conf analog-input-dock-mic.conf analog-input-linein.conf analog-input-rear-mic.conf analog-output.conf.common analog-output-lfe-on-mono.conf 2) ALSA in lsmod to make sure no loopback modules are loaded etc but none is resolving it. And there are many less information available on this. Has anyone similar problem solution in Ubuntu 11.10? (this problem i have resolved in Ubuntu 11.04 by replacing the default pulseaudio version to latest source from git, but while trying the same in Ubuntu 11.10 does not worked). Any tips please?

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  • How do I change which audio jacks are used for input and output?

    - by yamaha1996
    I'm using a Realtek HD audio card built-in my motherboard. The Windows driver comes with a control panel that allows me to select which back panel jacks are used for what. So for example I can make both the blue jack and green jack for output and only the red one for mic-in. (Whereas by default, the blue jack is for line in, which I never need.) How can I do the same under Linux? If possible, please don't suggest something that involves PulseAudio or JACK; I'd like to do it the plain way, e.g. by editing ALSA configuration files, if possible. The way I understand it, my problem should have nothing to do with software servers redirecting streams, just instructing the driver to treat this jack as so and so because it's hardware supported. Thank you very much!

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  • No audio flash movies

    - by Valerio Giuffrida
    I've installed natty a few days ago and I'm experience some issues with the audio. Basically, I'm not able to listen anything coming from flash movies (such as youtube). This happen with both chromium and firefox (I use only the first one) and the errore I get in the stdout is: ALSA lib pcm.c:2109:(snd_pcm_open_conf) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_pulse.so I don't know how to fix it. I found somewhere to install native x64 flash plugins, but somebody says it is not advisable. I didn't get this kind of issues with 10.10 I've gotten before of natty. Hence, what am I supposed to do? thank you very much

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  • No rear audio when front jack is connected

    - by Shanoop
    I have Ubuntu 14.04 64bit dual booted. When I connect something on front audio jack then rear audio is not working. I have tried changing analolog-output-headphones.conf file. After changing that alsamixer showing that both centre and surround not muted with full volum. Unfortunately no audio. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0

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  • having trouble getting audio drivers

    - by barry
    i'm trying to install the audio driver for my laptop and having problems. everything works great except for no audio. i locate the file, download it, try to open it, and this is what i get: Archive: /home/barry/Downloads/Audio_Conexant_v.6.14.10.575_XPx86/HXFSetup.exe [/home/barry/Downloads/Audio_Conexant_v.6.14.10.575_XPx86/HXFSetup.exe] End-of-central-directory signature not found. Either this file is not a zipfile, or it constitutes one disk of a multi-part archive. In the latter case the central directory and zipfile comment will be found on the last disk(s) of this archive. zipinfo: cannot find zipfile directory in one of /home/barry/Downloads/Audio_Conexant_v.6.14.10.575_XPx86/HXFSetup.exe or /home/barry/Downloads/Audio_Conexant_v.6.14.10.575_XPx86/HXFSetup.exe.zip, and cannot find /home/barry/Downloads/Audio_Conexant_v.6.14.10.575_XPx86/HXFSetup.exe.ZIP, period. can anyone help me out here? thanks

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  • How do I fix a noisy input device (Internal Mic)? snd_hda_intel - debug included

    - by hazrpg
    As the title says, I'm having trouble with a very noisy audio input device - the internal mic or any plugged in mics. So far I've narrowed it down to an problem in ALSA since my debug info is showing a lot of "null" values. Can anyone help? Debug Info: http://www.alsa-project.org/db/?f=e0c6fb7e10624bf7691aa2b405cf0d3968e56c63 Exert from the debug: model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null)

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  • SPDIF Input not working

    - by BiggJJ
    One of my motherboard has two SPDIF sockets. One input and one output. Under windows these work fine and I am able to achieve what I'm trying here. I want to set Ubuntu up so it will play the Digital input out the Analogue output(normal headphone jack). Under my input devices there is no digital options, and under configuration there is no option for Digital Stereo input I only have options for Digital output and Analogue input, when really I want it the other way around. Can anyone shed any light? error when starting gnome-alsamixer:

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