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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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  • Bluetooth refuses to connect since update to Ubuntu Gnome 13.10

    - by Niklas Berg
    I can no longer connect to my bluetooth speakers since since upgrading to Ubuntu Gnome 13.10 and then Gnome shell to 3.10, which never were a problem with Ubuntu Gnome 13.04. My bluetooth-dongle seems to working fine and I can even detect and add the speakers (Creative D100) but when I try to slide the button from off to on in the bluetooth settings it just slides back to off. The "bluetooth-B" in the upper right corner is also gone. I actually managed to connect after I added "Enable=Socket" under "[general]" /etc/bluetooth/audio.conf and the indicator on the speakers confirms the connection, but I cannot find the speakers in the audio settings even then. I've tried to solve this for several days, reading tons of other possibly related questions here on ask ubuntu and elsewhere but am unable to find a solution. Any ideas?

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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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  • 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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  • 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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  • Audio on beagleboard xM

    - by Francesco
    I am running Ubuntu 12.04 Precise for omap on a beagleboard xM. I am trying to setup audio. No soundcards are listed in /proc/asound/cards. Alsamixer fails with cannot open mixer: No such file or directory Under /dev/snd/ I have only: seq timer Driver's name should be omap3beagle - twl4030. I am using alsa 1.0.24 that is installed by default with Ubuntu 12.04. I've googled a lot but I have not found anything yet. Thanks

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  • Where can i get the openal sdk for c++?

    - by Peter Short
    The OpenAL site I'm looking at is a crappy outdated and broken sharepoint portal and the SDK in the downloads section give me a 500 html code when i request it. http://connect.creativelabs.com/openal/Downloads/OpenAL11CoreSDK.zip I found an OpenAL SDK on a softpedia and it has headers but not alu.h or alut.h which the tutorials I'm looking at apparently require for loading wavs etc. What am I missing? Is OpenAL dead or something?

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  • Converting from mp4 to Xvid avi using avconv?

    - by Ricardo Gladwell
    I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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  • Media keys play/pause globally worked in 12.10, not in 13.10

    - by Stéphane Gourichon
    Laptop media keys On Asus n55sf laptop, there are a dedicated keys for volume up, volume down, mute, [play/pause], stop, launch (plus a dozen Fn-key combinations). In 12.10 most worked. (Overall is seems unrelated to desktop environment used, stating it for the sake of completeness.) On Ubuntu 12.10 under XFCE they just worked. That is: when a player like rhythmbox or totem was started, it would alternate between play and pause. Interestingly, if several were started, they would alternate independently. E.g. use mouse to pause rhythmbox, launch totem, and one hit on [play/pause] key would pause one and resume the other. Keys Next,Previous and Stop worked as expected in any program. In 13.10 most still work, but play/skip related ignored. On Xubuntu 13.10 (XFCE too) the volume keys work but the [play/pause], stop, next and prev are ignored. Not tried regular Ubuntu 13.10 (Unity). Search before you ask Here are a few facts: https://wiki.ubuntu.com/Hotkeys/Architecture is ummutable and mentions Ubuntu 9.10. https://wiki.ubuntu.com/Hotkeys/Troubleshooting is also outdated as it mentions /usr/share/doc/udev/README.keymap.txt which no longer exists. On 12.10 and 13.10 versions, at XFCE level (as visible by xfconf-query or using xfce4-settings-manager) there are a couple of shortcut for keys like XF86Calculator or XF86TouchpadToggle but nothing related to volume prev/next/play/stop, which is okay. XF86Audio substring doesn't appear in /etc (which is normal) Kernel-level test: "showkey -s" on console shows that keys Next,Play/Pause,Previous,Stop are keycodes 163,164,165,166. Nothing relevant in /etc about that. Reports https://bugs.launchpad.net/ubuntu/+source/udev/+bug/1072371 and https://bugs.launchpad.net/ubuntu/+source/systemd/+bug/1012365 suggest to adjust at udev level. Alas, the udev tutorials I found ( e.g. https://wiki.debian.org/udev ) don't even mention keyboard. A thread in french seems to deal with a similar issue: https://forum.ubuntu-fr.org/viewtopic.php?id=1395051. @sudo evtest /dev/input/event3@, in X as well as on plain console, reports events on key pressed and repeats, but nothing when pressing those media keys. Is udev a dead end ? Questions How did it work in 12.10 ? Through udev ? Something else ? Any other hint ?

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  • Make alsamixer changes permanent

    - by Mauricio Andrés
    I have installed ubuntu GNOME 14.04 in my macbook Pro 13" mid 2012, and something is wrong with the headphones jack, a red led is on and it just can be turned off with the alsamixer command and muting a special section (I'm sorry for not remember the setting, I'm on OS X for work reasons) I already know how to fix it, but I want to know how to save the setting in the alsamixer, because with every reboot the red led is back. Thanks for the help.

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  • Record 8 separate Line IN Channels from M-Audio Delta 1010 Card

    - by Peter Hoffmann
    I want to record the 8 separate Line IN Channels from my M-Audio Delta 1010 Card. The card is recogniced nicely and a can record a single channel via arecord -d 10 -f cd -t wav -D channel1 out2.wav. I've set up the different channels in ~/.asoundrc. Now if I want to record a second channel in parallel (arecord -d 10 -f cd -t wav -D channel2 out2.wav) I get the error arecord: main:564: audio open error: Device or resource busy As I understand the delta 1010 is a single Access Card, so only one application can access it at a time. Is this correct? The next step was to configure a dual channel input in .asoundrc # envy24 channel 1+2 only pcm.test { type plug ttable.0.0 1 ttable.0.1 1 slave.pcm ice1712 } Which works ok when I do a arecord -d 10 -f cd -t wav -D test -c 2 out.wav (BTW can anyone point me to a tool to split a multi channel wav into a file per channel?) But when I want to record the channels separately with (-I option) arecord -d 10 -f cd -t wav -D test -c 2 -I channel1.wav channel2.wav I get no recordings. Did I miss something with the configuration or what are my options to record all 8 channels via arecord. I've no experience with jackd. Is it an option to install jackd and record the line ins via jackd?

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  • Are there plans to use the empty space in the SoundMenu?

    - by Kyle Clarke
    There seems to be roughly 4 lines of space next to the album art. However only 3 are used. Song Title Artist Album If nothing is planned for the 4th line. I propose that it is used for track time/length. This way you can tell how far along a song is without the need of a scrub bar. Unrelated, but I feel that the play-lists section should display how many songs are in that play list. Some of my play lists have no songs, and without realising this, it seems like a bug that the songs wont play.

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  • Linux based audio prodcuction tutorials

    - by thelinuxer
    I have been searching for a while for Linux based audio production tutorials. All I can find is tool based tutorials. For example I found tutorials on how to use jack, ardour, lmms ..etc. What I need is tutorials that teaches professional audio production with opensource/free tools, like those already available for protools and likes. If any one can guide me to any videos/articles available it would be highly appreciated. Thanks.

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  • Audio 2 dj soundcard configuration

    - by Jaroslav
    I have an http://www.native-instruments.com/#/en/products/dj/audio-2-dj/ The problem in settings it only sees one outpout, when there should be two(I need that for mixxx etc.) Also I want to be able set the sample rate to one of these 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

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  • Support for HRT Music Streamer II

    - by Duckers
    I'm considering a HRT Music Streamer II DAC (digital analog converter) for my HTPC. I run Ubuntu 10.10, but considering a upgrade to 11.04. Will this DAC be P'n'P, or at least easy to set up? http://www.highresolutiontechnologies.com/support/ indicates its supported on Mac and Windows. Linux is not mentioned. All my searches seems to indicate that its a bit troublesome, but those are posts from 2009 and older.

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