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  • No sound after updating from 11.10 to 12.04

    - by shaneo
    Hi I updated a friends computer from 11.10 to 12.04 and the sound stopped working in the built-in speakers. The computer is a Sony Vaio PCG-2J1L. I tried using the guide found here: https://help.ubuntu.com/community/HdaIntelSoundHowto and still no luck. $cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC889 $cat /proc/asound/card0/pcm0c/info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: ALC889 Analog name: ALC889 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 cat /proc/asound/card0/codec* output can be found here http://paste.ubuntu.com/1043188/ If anymore information is needed just let me know and ill post it here. Thank you in advance for any help with this issue.

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  • Static noise in headphones

    - by John Murdoch
    I have a Asus P6T based system. I was using the on-board sound (plugging in Logitech X-230 2.1 analog speakers in the green "front speakers" 3.5mm analog output, then plugging in my headphones in that). I was quite happy with the sound quality (didn't hear any static noise if volume was turned down to my normal listening level). Then about a week ago I started having terrible static noise from the left channel, and no normal audio on that left channel. Right channel had more static noise than usual but did have a bit of sound. I tried using the AC'97 in front of my case but that seemed to have no signal. I decided my on-board sound card has gone bad and bought an internal sound card to replace it (Startech 7.1Ch PCI). This fixed the "no sound from left channel problem", but I had much more audible static noise. I decided the card was low quality and/or it had interference from all the other things happening inside the computer case, and bought a Sweex SC016 external USB sound card. But even with that I have static noise in headphones. Positioning the USB sound card differently doesn't seem to help. Trying the other analog outputs (e.g., surround) doesn't help. The static noise in all cases is proportional to the volume. I have tried different headphones, but the situation is situation though perhaps the flavour of the static noise changes slightly. So what are my options? a) Get another, more expensive, external USB sound card hoping the quality will improve? b) Get another, more expensive, internal sound card (PCIe 1x perhaps) hoping the quality will improve? c) Get a dedicated DAC box? d) Get some Hi-Fi earphones? Suggestions? tl;dr - Two different sound cards both still have static noise in headphones.

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  • Windows 7 Media Center: sometimes no sound on a TV channel

    - by torbengb
    Problem Sometimes, recorded TV shows have no sound. This is of course very annoying because the recorded show is worthless. Also, when switching channels while watching live TV, sometimes a channel has no sound. This is always solved by switching channel again and then back (most often, once will do the trick). Media Center doesn't do this trick by itself when recording of course, so that's the bigger issue - but the cause is certainly the same. Details The computer is running a newly installed Windows 7 and Windows Media Center. It has 2 different tuner-cards installed. Both are installed with signed and up-to-date Win7 drivers and appear OK in Device Manager. Both tuners get the same antenna signal, from a split cable from the wall. The cable delivers analog cable TV (40+ channels) and digital cable TV (4 channels) at the same time. Both tuners have been configured to receive both analog and digital channels. This only happens with analog channels. How can I fix the no-sound problem? Update: I've now spent some time with the computer to try and pinpoint the problem, but I've had little success so far: I flipped through the channels until one didn't have sound, then I disconnected the antenna cable from one of the tuners. It was the right one because then the video also went away. I flipped lots more channels to see if the other card also would come up mute once but I never had a channel without sound. It might still be possible, I don't know. Then I disconnected the "good" tuner and connected the "bad" tuner and again flipped lots of channels but again I never had a channel without sound. It seems to me that the problem is erratic. It happens on any channel, and I haven't ruled out yet that it only happens on one tuner.

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  • How do I forward a request to a different url in python

    - by tax
    import SimpleHTTPServer import SocketServer class myHandler(SimpleHTTPServer.SimpleHTTPRequestHandler): def do_GET(self): print self.path if self.path == '/analog': return "http://someserver.com/analog" return SimpleHTTPServer.SimpleHTTPRequestHandler.do_GET(self) theport = 1234 Handler = myHandler pywebserver = SocketServer.TCPServer(("", theport), Handler) print "Python based web server. Serving at port", theport pywebserver.serve_forever()

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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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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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]

    - by martin
    Possible Duplicate: How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab This is all done in MatLab 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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  • Digital audio input on Macbook?

    - by Ken
    I have: a Macbook (not Pro), don't know the exact model but it's a Core 2 Duo 2.0GHz and probably what Wikipedia calls the "Late 2006" or "Mid 2007" model a DVD player, region-free, that has "Coax and TosLink optical digital audio outputs" I want to make an MP3 of the audio track of some DVDs (for learning a new language), and I can't use the Macbook's built-in DVD drive because it's a different region (ugh!). I'm sure I can connect the DVD player to the Macbook with an analog audio cable. However, if it's possible I'd prefer to keep the signal digital. I'm not even positive if my old Macbook has digital audio in, and if so what I need to connect to it. (I've done plenty of home audio geeking, but always in analog!) Will a "Toslink cable" plus a "Toslink Female to Mini-Plug Male Adapter" (found on Amazon) let me connect my things together? It looks like the pieces will fit but I'd like to hear someone confidently knowledgeable on the matter before I buy something. Thanks!

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  • Help find correct alsa model for onboard sound (alc887?) that will work with jack and have correct mixer setup

    - by Jazz
    I have a Gigabyte GA-MA74GMT-S2 motherboard. I am using Jack for sound - connected to ALSA. I am running Ubuntu 12.04. aplay -l reports card 0: SB [HDA ATI SB], device 0: ALC887 Analog [ALC887 Analog] The problem is that the default setup, that alsa decides to use, causes stuttering and xruns no matter how generous I set the frames/period or periods/buffer etc. Also, Jack works fine if I plug in an external USB sound system and use that. My processor is an AMD phenom x4 945, and I have 8GB ram, and Video card is Geforce GTX550 Ti, all of which should be quite capable enough. I also tried Pulseaudio and that works fine, but I need to use Jack At first I thought it might be an interrupt conflict, but I have found that adding "options snd-hda-intel model=generic" to /etc/modprobe.d/alsa-base.conf causes it to play correctly, but the limited mixer setup lacks controls I need - so this setup isn't good enough. Still, it seems to prove it isn't a hardware conflict. I have tried many other models, such as 3stack, 6stack, auto and even basic, and they all suffer from the stuttering. I eventually found "options snd-hda-intel model=3stack-6ch-intel" works without stuttering, and mixer is much closer to what it needs to be. Can anyone help on how to get a correct and accurate model for ALSA to use? More info on the hardware that might help... *-multimedia description: Audio device product: SBx00 Azalia (Intel HDA) vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 14.2 bus info: pci@0000:00:14.2 version: 00 width: 64 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=snd_hda_intel latency=32 resources: irq:16 memory:fe024000-fe027fff

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  • Audio not working

    - by user3215
    Anybody could help me in troubleshooting audio problem on ubutnu 9.04 desktop edition?. For some reason I've to keep this os not upgraded and I'm trying to fix the audio problem on this for months. It works well on upgraded version(9.10,10.04) but not on jaunty. aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod | grep snd: snd_hda_intel 436148 7 snd_pcm_oss 46336 0 snd_mixer_oss 22656 1 snd_pcm_oss snd_pcm 83076 4 snd_hda_intel,snd_pcm_oss snd_seq_dummy 10756 0 snd_seq_oss 37760 0 snd_seq_midi 14336 0 snd_rawmidi 29696 1 snd_seq_midi snd_seq_midi_event 15104 2 snd_seq_oss,snd_seq_midi snd_seq 56880 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 29704 2 snd_pcm,snd_seq snd_seq_device 14988 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 62756 21 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15200 1 snd snd_page_alloc 16904 2 snd_hda_intel,snd_pcm cat /proc/asound/cards: 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xe1280000 irq 16 cat /proc/asound/version: Advanced Linux Sound Architecture Driver Version 1.0.18rc3. vim /etc/modules: # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with "#" are ignored. lp Audio Settings:

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  • System starts without sound in Lubuntu

    - by cipricus
    When I log in the sound is muted. Why is that? This image is from the Xfce mixer, while the lxde volume does not show that the sound is off. So, I can see the sound muted only in Xfce Mixer. But I can also start the sound in another application, Pulse Audio Volume Control, if I change a setting in Output Devices from Analog Mono Output to LFE on Mono Separate Devices: The Analog Mono Output option here sets in fact the Xfce Mixer in the mute position from the first image. Could this be related to managing sessions (and therefore to these question)? This happens in LXDE session as well as in Lubuntu session. Starting sound and logging out and in between any of these two sessions would mute the volume as shown. Starting the volume in one of these and logging out and then logging in into the Gnome session, the sound would be off; but then, after starting sound, logging out of Gnome and then logging in into any session, the sound would not be muted. But after restart it would be so no matter the session.

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  • 12.04.1 no audio through HDMI

    - by JoJo
    having a bit of an issue with getting audio to go through HDMI. Here are the base specs: OS: Ubuntu Desktop 12.04.1 x64 CPU: AMD A10-5800K 3.8G 4M FM2 R Mobo: MSI FM2-A75MA-E35 OS: Ubuntu 12.04.1 LTS Vid Card: (integrated on CPU) AMD Radeon HD 7660D HDMI sound works fine under Win7 (after mobo and vid drivers are installed), so it's not physically broken. Audio through the normal headphone jacks works fine under Ubuntu. Looking at the audio panel, there is no HDMI output at all. aplay -l also reports only: card 0: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 In additional drivers there are two versions: ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER (post-release update) The first installs, but problem persists. I do get more resolutions to pick from. Second version does not, reporting it failed installation and to find details at: /var/log/jockey.log Looked at the log, and it's insanely long, if necessary I can get it to you guys. Did more research and some had luck by manually installing the drivers, so tried to give that a shot by following this: https://help.ubuntu.com/community/BinaryDriverHowto/ATI#Manually_installing_Catalyst_12.6 starting at 3.1 Manually installing Catalyst 12.6. I immediately had 2 issues, (1) the AMD website does not provide any drivers for Linux, and (2) the following command did not work: sudo sh amd-driver-installer-12-6-x86.x86_64.run --buildpkg Ubuntu/precise sh: 0: Can't open amd-driver-installer-12-6-x86.x86_64.run Some other posts stated to update "alsa-drivers", but that also did not work as install command for the new version of them did not work. I forget the exact issue, but similar to above, cannot open / cannot find. Any help would be greatly appreciated!

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  • 12.04 No Sound - ALC888 / Radeon 3200HD

    - by Ross
    Evening all. I have a MSI U230 netbook, MV40 processor, 4Gb RAM with integrated ATI Radeon 3200HD grahics & an ALC888 codec integrated soundcard. It has HDMI out as well. I've tried a few distro's and have been around linux for a short time. I reckon I've settled on Ubuntu 12.04 (32bit) due to it doing pretty much everything I want it to. I'm working with a fresh install right now. I recently re-installed when I was going in circles trying to solve my problem before. I install Ubuntu and it works, except for the sound. I have tried things like reinstalling Alsa, editing my asound.conf file, installing HDA Verb and a few other things. Its at the point where I need to ask for help... Some outputs: ross@ross:~$ aplay -l ** List of PLAYBACK Hardware Devices ** card 0: SB [HDA ATI SB], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ross@ross:~$ uname -r 3.2.0-34-generic-pae ross@ross:~$ lspci | grep VGA 01:05.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI RS780M/RS780MN [Mobility Radeon HD 3200 Graphics] ross@ross:~$ lspci | grep Audio 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) 01:05.1 Audio device: Advanced Micro Devices [AMD] nee ATI RS780 HDMI Audio [Radeon HD 3000-3300 Series] Added options snd-hda-intel position_fix=1 to alsa-base.conf file Unmuted all in alsamixer Can anyone suggest anything more?

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  • Good 2D Platformer Physics

    - by Joe Wreschnig
    I have a basic character controller set up for a 2D platformer with Box2D, and I'm starting to tweak it to try to make it feel good. Physics engines have a lot of knobs to tweak, and it's not clear to me, writing with a physics engine for the first time, which ones I should use. Should jumping apply a force for several ticks? An impulse? Directly set velocity? How do I stop the avatar from sticking to walls without taking away all its friction (or do I take away all the friction, but only in the air)? Should I model the character as a capsule? A box with rounded corners? A box with two wheels? Just one big wheel? I feel like someone must have done this before! There seem to be very few resources available on the web that are not "baby's first physics", which all cut off where I'm hoping someone has already solved the issues. Most examples of physics engines for platformers have floaty-feeling controls, or in-air jumps, or easily exploitable behavior when temporary penetration is too high, etc. Some examples of what I mean: A short tap of jump jumps a short distance; a long tap jumps higher. Short skidding when stopping or reversing directions at high velocity. Standing stably on inclines (but maybe sliding down them when ducking). Analog speed when using an analog controller. All the other things that separate good platformers from bad platformers. Dare I suggest, stable moving platforms? I'm not really looking for "hey, do this." Obviously, the right thing to do is dependent on what I want in the game. But I'm hoping someone somewhere has gone through the possibilities and said "well technique A does feature X well, technique B does Y well, but that doesn't work with C", or has some worked examples beyond "if (key == space) character.impulse(0, 1)"

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  • Digital HD Transition

    - by Bill Evjen
    The HD Experience Roughly 53% of the viewing public has HD capable devices in their home 24% think they are watching HD while they have no subscription to any HD content Today’s HD Considerations Choices abound: format resolution – 720p, 1080i/p frame rates compression and wrapping audio compression and delivery metadata packaging, delivery, and usage content delivery protocols Metadata is going to be a part of the overall experience With emerging technologies: Super Hi-Vision (SHV, UHDTV 4320p), 3D HEVC/H.265, WEBM/VP8 HDBaseT, P2PTV Dolby Pulse/HE-AAC Industry standardization Metadata registration, packaging, and delivery standards Improved picture and sound quality is a logical next step but we need to also think about the end to end viewing experience including; 3D video and audio content Mixed-mode viewing to bring interactive and immersive experiences Content Transportability both on-to and off-of the aircraft High Definition Standardization Analog switch off around the world DTV transition completed: 17 countries DTV transition in progress: 45 countries The EU has mandated the end of 2012 as the final date for Analog Switch Off D-Cinema was standardized by SMPTE in 2006 Airlines are installing HD displays today Passengers are bringing their own devices now HD TV on airlines are getting bigger and bigger – bigger than SD was – now up to 23” Gray scale data input for color – 6 to 8 bit Contrast – 400 to 700 Backlit – LED Encryption – can it be the same for HD? PPV in the cabin?

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  • Audio comes out of both headphone and speaker at the same time.. Ubuntu 12.04LTS [closed]

    - by pst007x
    I have the same issue on an Aspire. Ubuntu 12.04LTS 64bit realtek audio sound chip onboard If I plug in a headset, audio does not switch from internal speaker to headset, instead plays out of both at the same time. I have looked at the alsamixer setting, all on. I installed gnome-alsamixer, and I noticed headphone was ticked, if I untick the main audio mutes, and the headphone no longer works. Headset only works with internal speaker. Audio works fine on my other desktop and laptop running this release 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) salvatore@salvatore-Aspire-7730:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. salvatore@salvatore-Aspire-7730:~$ head -n 1 /proc/asound/card*/codec#* ==> /proc/asound/card0/codec#0 <== Codec: Realtek ALC888 ==> /proc/asound/card0/codec#1 <== Codec: LSI ID 1040 ==> /proc/asound/card0/codec#2 <== Codec: Intel Cantiga HDMI salvatore@salvatore-Aspire-7730:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 salvatore@salvatore-Aspire-7730:~$ uname -a Linux salvatore-Aspire-7730 3.2.0-23-generic #36-Ubuntu SMP Tue Apr 10 20:39:51 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux salvatore@salvatore-Aspire-7730:~$ The alsa-base.conf does not exist Tried this: sudo apt-get remove --purge alsa-base sudo apt-get remove --purge pulseaudio sudo apt-get install alsa-base sudo apt-get install pulseaudio sudo alsa force-reload Then: sudo apt-get purge pulseaudio gstreamer0.10-pulseaudio sudo apt-get install pulseaudio gstreamer0.10-pulseaudio indicator-sound Tred this. sudo gedit Then open terminal: sudo /etc/modprobe.d/alsa-base.conf At the end of the file add a new line: options snd-hda-intel model=generic Save and then reboot But alsa-base.conf does not exist

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  • Design pattern for an automated mechanical test bench

    - by JJS
    Background I have a test fixture with a number of communication/data acquisition devices on it that is used as an end of line test for a product. Because of all the various sensors used in the bench and the need to run the test procedure in near real-time, I'm having a hard time structuring the program to be more friendly to modify later on. For example, a National Instruments USB data acquisition device is used to control an analog output (load) and monitor an analog input (current), a digital scale with a serial data interface measures position, an air pressure gauge with a different serial data interface, and the product is interfaced through a proprietary DLL that handles its own serial communication. The hard part The "real-time" aspect of the program is my biggest tripping point. For example, I need to time how long the product needs to go from position 0 to position 10,000 to the tenth of a second. While it's traveling, I need to ramp up an output of the NI DAQ when it reaches position 6,000 and ramp it down when it reaches position 8,000. This sort of control looks easy from browsing NI's LabVIEW docs but I'm stuck with C# for now. All external communication is done by polling which makes for lots of annoying loops. I've slapped together a loose Producer Consumer model where the Producer thread loops through reading the sensors and sets the outputs. The Consumer thread executes functions containing timed loops that poll the Producer for current data and execute movement commands as required. The UI thread polls both threads for updating some gauges indicating current test progress. Unsure where to start Is there a more appropriate pattern for this type of application? Are there any good resources for writing control loops in software (non-LabVIEW) that interface with external sensors and whatnot?

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  • 12.10 no audio via hdmi and video speeds up

    - by jackson
    I have a laptop with an ati radeon 4200, on 12.04 everything worked fine, since upgrading to 12.10 I cannot get sound over the hdmi. When I switch to hdmi audio the video speeds up to about 2x. I can use the speakers in my laptop and watch video via hdmi with no problems. Things I have tried: Various tutorials to install the AMD/ATI drivers, all of which resulted in low graphics mode. Checked that everything is properly set in alsamixer, the sound utility and - installed pavucontrol and checked everything in there. Verified the output from cat /proc/asound/cards looks normal When I initially upgraded there was a plethora of problems which I believe were due to the old proprietary driver still being used but not compatible, after a few hours trying to fix that I decided just to back up and do a fresh install which works great except for the above stated problem. Any help would be greatly appreciated!! Finally hopefully this hasn't already been answered, I have tried a few different searches on the boards and haven't come up with anything. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0

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  • Sound card not detected in 13.04

    - by Ganessh Kumar R P
    I have a problem with my sound card. I don't have volume up or down option anywhere. In the setting -> Sound I don't have any card detected. But when I run the command sudo aplay -l, I get the following output **** List of PLAYBACK Hardware Devices **** Failed to create secure directory (/home/ganessh/.config/pulse): Permission denied card 0: MID [HDA Intel MID], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 0/1 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 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 And the command lspci -v | grep -A7 -i "audio" outputs 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 02a2 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at f0f20000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 5 Series/3400 Series Chipset PCI Express Root Port 1 (rev 06) (prog-if 00 [Normal decode]) -- 02:00.1 Audio device: NVIDIA Corporation GF106 High Definition Audio Controller (rev a1) Subsystem: Dell Device 02a2 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at d3efc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 07:00.0 Network controller: Intel Corporation Ultimate N WiFi Link 5300 So, I assume that the drivers are properly installed but still I don't get any option in the settings or volume control. The same card used to work well back in 2010 versions(04 and 10) Any help is appreciated. Thanks

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  • I can't hear any sounds on ubuntu 11.10 on Dell inspiron N5010

    - by Ahmed
    I have a problem that I can't hear any sounds and I don't know where to start. I did the following : lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at fbf00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel -- 01:00.1 Audio device: ATI Technologies Inc Manhattan HDMI Audio [Mobility Radeon HD 5000 Series] Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at fbe40000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel And It seems that I have 2 soundcards. Is that normal ?? I also did this: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 Also on the sound setting GUI. I have 2 hardware profiles for sound cards but none of them works when I test the speakers. Where should I start searching ?

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  • Sound issues after trying everything

    - by Lerp
    I cannot get my sound working properly, no matter what I do, there's always some problem. It's very annoying as it's the only thing preventing me from making Ubuntu my main OS. At the moment my sound always plays through both my speakers and my headphones regardless except the sound through the headphones is crackly. It is also a bit quiet even though everything is maxed. I've managed to improve the situation to a point where the sound out of my speakers is perfect but I have none at all from my headphones. I do have two connectors listed in the sound settings but regardless of which one is selected it always plays through the speakers. I think this might have something to do with the fact that my speakers are plugging into the front of my computer, typically the headphone jack, and my headphones are plugging into the back but when I try disconnecting the speakers from the front there is still no sound from the headphones. I fixed the speaker sound by going through the sound settings and making sure they were all set to 100% then rebooting. Things I have tried: Maxing everything and unmuting everything in alsamixer Uninstalling pulseaudio Making gstreamer use only alsa via gstreamer-properties. This worked with the sound test button including independent sound between headphones and speakers but when I reset the computer it no longer worked. So I tried setting it manually in gconf-editor which didn't work either. Reinstalling alsa and pulseaudio Setting the model in /etc/modprobe.d/alsa-base.conf to 6stack and 6stack-dig neither worked. Upgrading to 12.10 Here's some command output to help you diagnose my problem. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 sudo lshw -C sound *-multimedia description: Audio device product: 82801JI (ICH10 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:70 memory:f7ff8000-f7ffbfff cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 options snd-hda-intel model=6stack

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  • no volume in kubuntu 10.04

    - by neha
    hello,I am having both gnome and kde on my system.as my gnome is working perfectly but in KDE is there is no sound being generated. output of apley -l and lspci commands is as follows.. neha@neha-laptop:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 and output of lspci command is: neha@neha-laptop:~$ lspci 00:00.0 Host bridge: Intel Corporation Mobile PM965/GM965/GL960 Memory Controller Hub (rev 0c) 00:02.0 VGA compatible controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (rev 0c) 00:02.1 Display controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (rev 0c) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.4 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 5 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HEM (ICH8M) LPC Interface Controller (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801HBM/HEM (ICH8M/ICH8M-E) IDE Controller (rev 02) 00:1f.2 SATA controller: Intel Corporation 82801HBM/HEM (ICH8M/ICH8M-E) SATA AHCI Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 02:09.0 FireWire (IEEE 1394): Ricoh Co Ltd R5C832 IEEE 1394 Controller (rev 05) 02:09.1 SD Host controller: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 22) 02:09.2 System peripheral: Ricoh Co Ltd R5C843 MMC Host Controller (rev 12) 02:09.3 System peripheral: Ricoh Co Ltd R5C592 Memory Stick Bus Host Adapter (rev 12) 02:09.4 System peripheral: Ricoh Co Ltd xD-Picture Card Controller (rev ff) 09:00.0 Ethernet controller: Marvell Technology Group Ltd. 88E8040 PCI-E Fast Ethernet Controller (rev 12) 0b:00.0 Network controller: Broadcom Corporation BCM4312 802.11a/b/g (rev 01) can anyone help me??

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