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• MPlayer does not work

  - by Soham Pal

  Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

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• audio controls in xfce 4.8

  - by Peter

  I am seeing several questions similar to mine, but none of the answers are sufficient. I am pretty green with Ubuntu, so here goes: I was just automatically upgraded to xfce 4.8 for Ubuntu studio. The volume control no longer works in my panel. When I launch 'mixer' I don't see any settings, either. When I try to run "linux audio configuration" I get an error: JACK can only be configured with a loaded and stopped studio. Please create a new studio or load and stop an existing one. I understand that I can change the volume using command line, but I can't understand why I got upgraded to something that fails on basic features. I much less likely to recommend ubuntu to others as a result. thanks!

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• DX9 Deferred Rendering, GBuffer displays as clear color only

  - by Fire31

  I'm trying to implement Catalin Zima's Deferred Renderer in a very lightweight c++ DirectX 9 app (only renders a skydome and a model), at this moment I'm trying to render the gbuffer, but I'm having a problem, the screen shows only the clear color, no matter how much I move the camera around. However, removing all the render target operations lets the app render the scene normally, even if the models are being applied the renderGBuffer effect. Any ideas of what I'm doing wrong?

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• How do display a "mucus spreading" effect in a 2D environment?

  - by nathan

  Here is an example of such a mucus spreading. The substance is spread around the source (in this example, the source would be the main alien building). The game is starcraft, the purple substance is called creep. How this kind of substance spreading would be achieved in a top down 2D environment? Recalculating the substance progression and regenerate the effect on the fly each frame or rather use a large collection of tiles or something else?

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• How to change the volume of left channel using pulseaudio

  - by user2622247

  I am recording the video using logitech camera and bluetooth microphone. Logitech can used for recording both audio and video. When I turn on bluetooth microphone in the middle of recording, it is replacing the logitech audio channel due to this we are getting the bluetooth audio from the left channel. But when I turn off the bluetooth then I get the logitech audio on left channel but the volume is very low and also getting the some noise. I am using PulseAudio and ffmpeg for recording purpose. So how can I increase / change the volume of left channel during runtime?

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• How to negate current window in gnome shell?

  - by k0pernikus

  I dislike that most websites use a black font on white background for their sites, as it gets too tiresome for me to read. Back in the days of 11.04, using Gnome2 with compiz, there actually was a Negative feature that could negate the content of any window, making the background black and the font white. Much easier on the eyes for me. Yet since 11.10, using gnome shell with mutter, I have no idea if there is something alike out there. Hence my question: How do I negate the currently active window in gnome shell? I am not interested in alternative methods, e.g. user styles. I am aware of their existence but I find it much easier to just invert the screen by the hit of a key shortcut. I also want the solution to be application-agnostic. As I also from time to time would want to invert libre-office or some other glaringly white application.

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• How to remove the boundary effects arising due to zero padding in scipy/numpy fft?

  - by Omkar

  I have made a python code to smoothen a given signal using the Weierstrass transform, which is basically the convolution of a normalised gaussian with a signal. The code is as follows: #Importing relevant libraries from __future__ import division from scipy.signal import fftconvolve import numpy as np def smooth_func(sig, x, t= 0.002): N = len(x) x1 = x[-1] x0 = x[0] # defining a new array y which is symmetric around zero, to make the gaussian symmetric. y = np.linspace(-(x1-x0)/2, (x1-x0)/2, N) #gaussian centered around zero. gaus = np.exp(-y**(2)/t) #using fftconvolve to speed up the convolution; gaus.sum() is the normalization constant. return fftconvolve(sig, gaus/gaus.sum(), mode='same') If I run this code for say a step function, it smoothens the corner, but at the boundary it interprets another corner and smoothens that too, as a result giving unnecessary behaviour at the boundary. I explain this with a figure shown in the link below. Boundary effects This problem does not arise if we directly integrate to find convolution. Hence the problem is not in Weierstrass transform, and hence the problem is in the fftconvolve function of scipy. To understand why this problem arises we first need to understand the working of fftconvolve in scipy. The fftconvolve function basically uses the convolution theorem to speed up the computation. In short it says: convolution(int1,int2)=ifft(fft(int1)*fft(int2)) If we directly apply this theorem we dont get the desired result. To get the desired result we need to take the fft on a array double the size of max(int1,int2). But this leads to the undesired boundary effects. This is because in the fft code, if size(int) is greater than the size(over which to take fft) it zero pads the input and then takes the fft. This zero padding is exactly what is responsible for the undesired boundary effects. Can you suggest a way to remove this boundary effects? I have tried to remove it by a simple trick. After smoothening the function I am compairing the value of the smoothened signal with the original signal near the boundaries and if they dont match I replace the value of the smoothened func with the input signal at that point. It is as follows: i = 0 eps=1e-3 while abs(smooth[i]-sig[i])> eps: #compairing the signals on the left boundary smooth[i] = sig[i] i = i + 1 j = -1 while abs(smooth[j]-sig[j])> eps: # compairing on the right boundary. smooth[j] = sig[j] j = j - 1 There is a problem with this method, because of using an epsilon there are small jumps in the smoothened function, as shown below: jumps in the smooth func Can there be any changes made in the above method to solve this boundary problem?

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• Alsa version numbers not consistent

  - by user69245

  I've recently moved from 10.04 to 12.04 and have problems with crackles on audio when the screen changes (especially obvious in a Skype conversation with video). !!Kernel Information !!------------------ Kernel release: 3.2.0-26-generic-pae Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: 1.0.25 Utilities version: 1.0.25 Running through the SoundTroubleShooting procedures I got the above details at Step 3. Is the mismatch between the Driver version (1.0.24) and the Library and Utilities versions (1.0.25) the cause of my problem, or should I be looking elsewhere? In answers to the question "Will ALSA 1.0.25 version have time to enter 12.04?" I read that 1.0.25 would be part of the new kernel, but my kernel version appears to be newer than the original release.

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• How do produce a "mucus spreading" effect in a 2D environment?

  - by nathan

  Here is an example of such a mucus spreading. The substance is spread around the source (in this example, the source would be the main alien building). The game is starcraft, the purple substance is called creep. How this kind of substance spreading would be achieved in a top down 2D environment? Recalculating the substance progression and regenerate the effect on the fly each frame or rather use a large collection of tiles or something else?

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• Microsoft LifeCam Audio Issue

  - by Daniel Samson

  Running Ubuntu 12.04.1 LTS 32-bit on a computer with an Intel core i5 processor, 4GB RAM and a NVIDIA 450GTS Graphics card. I am trying to set up Skype with my webcam. So far everything works almost perfectly, however my Microsoft LifeCam-VX700 v2.0 audio does not work correctly. The audio feed is just static/white noise. I have checked the hardware using windows and it worked perfectly there. The video works great in Ubuntu. It is just the audio that is the problem. What can I do to rectify this problem?

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• Audio Stutter in in ubuntu 12.04

  - by Andrew Redd

  After upgrading to precise my audio is stuttering. It is happening, in VLC, mplayer, and anything streaming from the internet. I followed the procedures in https://help.ubuntu.com/community/SoundTroubleshootingProcedure but nothing has helped so far. There is the problem that the driver version is out of date but it does not seem to want to update with the given commands. $ bash alsa-info.sh --stdout |grep version Driver version: 1.0.24 Library version: 1.0.25 Utilities version: 1.0.25 How can I upgrade the driver and fix the stuttering?

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• Is there now any way to convert mp3 files to m4a or aac 192kbit?

  - by piedro

  Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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• Unable to configure/setup 5.1 audio with 12.04

  - by Vipin Vinayan

  I am kinda new to Ubuntu as well. I have been having this issue with audio for quite sometime now. Initially, when I installed version 11.10 (I guess), I was able to use my 5.1 speakers without any issues. If my memory serves me right, it was after an update that the 5.1 audio stopped working and the video resolution would not get saved. I temporarily fixed the resolution issue by creating a start-up shell script that would update the resolution and load it. But the issue with audio has been going on for quite sometime now. Even though I have option for 5.1, only two speakers seem to be working. I thought an upgrade should fix the issue and so upgraded the OS to version 12.04. I also tried uninstalling alsa and pulse audio, reinstalling them, changing the /etc/pulse/daemon.conf channels from 2 to 6. I have also tried installing pavucontrol but nothing seems to have worked and the issue still persists. Is there anything else you could suggest? The lspci log on my computer is as follows 00:00.0 Host bridge: Intel Corporation 82G33/G31/P35/P31 Express DRAM Controller (rev 10) 00:01.0 PCI bridge: Intel Corporation 82G33/G31/P35/P31 Express PCI Express Root Port (rev 10) 00:02.0 VGA compatible controller: Intel Corporation 82G33/G31 Express Integrated Graphics Controller (rev 10) 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) 00:1c.0 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 1 (rev 01) 00:1c.1 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 2 (rev 01) 00:1d.0 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #1 (rev 01) 00:1d.1 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #2 (rev 01) 00:1d.2 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #3 (rev 01) 00:1d.3 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #4 (rev 01) 00:1d.7 USB controller: Intel Corporation N10/ICH 7 Family USB2 EHCI Controller (rev 01) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation N10/ICH7 Family SATA Controller [IDE mode] (rev 01) 00:1f.3 SMBus: Intel Corporation N10/ICH 7 Family SMBus Controller (rev 01) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) Would really appreciate a response that will assist me in resolving my issue. Thanks in advance Vipin

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• Managing constant buffers without FX interface

  - by xcrypt

  I am aware that there is a sample on working without FX in the samplebrowser, and I already checked that one. However, some questions arise: In the sample: D3DXMATRIXA16 mWorldViewProj; D3DXMATRIXA16 mWorld; D3DXMATRIXA16 mView; D3DXMATRIXA16 mProj; mWorld = g_World; mView = g_View; mProj = g_Projection; mWorldViewProj = mWorld * mView * mProj; VS_CONSTANT_BUFFER* pConstData; g_pConstantBuffer10->Map( D3D10_MAP_WRITE_DISCARD, NULL, ( void** )&pConstData ); pConstData->mWorldViewProj = mWorldViewProj; pConstData->fTime = fBoundedTime; g_pConstantBuffer10->Unmap(); They are copying their D3DXMATRIX'es to D3DXMATRIXA16. Checked on msdn, these new matrices are 16 byte aligned and optimised for intel pentium 4. So as my first question: 1) Is it necessary to copy matrices to D3DXMATRIXA16 before sending them to the constant buffer? And if no, why don't we just use D3DXMATRIXA16 all the time? I have another question about managing multiple constant buffers within one shader. Suppose that, within your shader, you have multiple constant buffers that need to be updated at different times: cbuffer cbNeverChanges { matrix View; }; cbuffer cbChangeOnResize { matrix Projection; }; cbuffer cbChangesEveryFrame { matrix World; float4 vMeshColor; }; Then how would I set these buffers all at different times? g_pd3dDevice->VSSetConstantBuffers( 0, 1, &g_pConstantBuffer10 ); gives me the possibility to set multiple buffers, but that is within one call. 2) Is that okay even if my constant buffers are updated at different times? And do I suppose I have to make sure the constantbuffers are in the same position in the array as the order they appear in the shader?

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• How to use Epos?

  - by Jinx

  When I installed epos: sudo apt-get install epos and try to use say: say hello the terminal says: hello#######################################################################################################################################################T Could not open localsound device, error 2 Client side error: Could not set up a stream How can I configure this software to make it run correctly?

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• Pokemon Yellow wrap transitions

  - by Alex Koukoulas

  So I've been trying to make a pretty accurate clone of the good old Pokemon Yellow for quite some time now and one puzzling but nonetheless subtle mechanic has puzzled me. As you can see in the uploaded image there is a certain colour manipulation done in two stages after entering a wrap to another game location (such as stairs or entering a building). One easy (and sloppy) way of achieving this and the one I have been using so far is to make three copies of each image rendered on the screen all of them with their colours adjusted accordingly to match each stage of the transition. Of course after a while this becomes tremendously time consuming. So my question is does anyone know any better way of achieving this colour manipulation effect using java? Thanks in advance, Alex

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• Looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs <= $5

  - by CyanPrime

  I'm looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs less then, or is $5 usd. I want to have a nice BGM for my engine demo I'm going to release for a game I'm planing on having go commercial. I don't want to spend too much money on it, so my limit is $5 usd. I want it to be at least a 1:00 in length. Where should I look? Or even better, do you have a link to a song that meets the criteria?

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• Phonon - Audio Hardware Setup

  - by Michael

  Does anyone know what configuration file is modified when I change settings within this area of the system settings? I keep choosing my Logitech USB Headset as the soundard, and, when I hit apply, it accepts that change and all is good. However, as soon as I close the system settings window, it loses the setting, so I have to leave it open all the time. How can I make this setting permanent? Thanks for any assistance. I'm actually running Kubuntu 11.10.

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• Audio playback: part of song is skipped

  - by Homulvas

  I am experiencing some problems with music playback after upgrading to Ubuntu 12.10. Basically some of the songs stop playing after some time as if the song has ended. It's always the same songs and the same time. The weird thing that it happens with Clementine and Totem but VLC doesn't have this problem and it also plays as it should on Windows. I'm guessing there might be a problem with some library that's shared with by the first two applications. I don't know if it's relevant but the file format of the audio files is flac(don't know if the problem affects mp3, because I don't have many of them).

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• Music player with a few specific requirements

  - by Jordan Uggla

  I am looking for a music player with a few specific requirements: Must have a search function that whittles down results as you type, searching the entire library. Must start playing a song when double clicked, and not continue to another song when that song finishes. Must be approachable and immediately usable by people completely unfamiliar with the program. I think this is mostly covered by the first two requirements being met. I've tried many players but unfortunately every one has failed to meet at least one of the requirements. Rhythmbox meets 1 and 3, but continues to the next search result after the song which was double clicked ends. Banshee is basically the same as Rhythmbox. While it has an option to "Stop when finished" this cannot (as far as I can tell) be made the default when double clicking a song. Audacious (as far as I can tell) fails at 1. Muine meets requirements 1 and 2, but unfortunately I couldn't make the search dialog always shown like it is with Rhythmbox / Banshee which, despite its very simple interface, made Muine incomprehensible to people trying to use it for the first time. Amarok I could not configure to meet requirement 1, but I think it's likely I was just missing something, and with its configurability I'm confident that I can set it up to meet requirements 2 and 3.

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• ubuntu lite volume control

  - by idio

  i've recently gave up on the main ubuntu, especially since new updates ruined Quantal and Raring is just a mess and went on to install a more friendly version: Ubuntu Lite. while i am quite happy with most functionalities of this lite version of 12.04, one of the modifications volume control is totally ridiculos, there is an applet that is completely counter-intutitive. So my question is: is there a way to install back the ubuntu classic volume control so i can let go of this applet alltogether? thanx

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• Pulseaudio not detecting bluetooth headset automatically

  - by adityap174

  The Pavucontrol on my Xubuntu 13.10 detects my bluetooth headset, but only after I type the command sudo alsa force-reload After a few seconds the headset appears in the configuration tab. I have also tried inserting the line load-module module-switch-on-connect in the /etc/pulse/default.pa but it doesn't help. I have tried purging and autoremoving and reinstalling everything right from alsa to pulseaudio and pulseaudio-module-bluetooth. Nothing works. Is there a workaround to load the headset without using the command force-reload ? To make the pulseaudio daemon recognize the device automatically ?

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• Batch normalize audio volume on .ogg-files

  - by Embridioum

  Are there any simple tools to normalize an .ogg-library of 10'000+ songs so that the volume is the same throughout all songs? Terminal or GUI doesn't matter, it only need to be simple. One caveat though, I don't want soft interludes/intermissions and ballads blown out of proportion. Preferably the process should find the overall gain of the album (I have all my CD's ripped into separate folders) and normalize the level thereafter.

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• Record and display Kinect audio and video via avconv

  - by ThomasH

  I'm trying to record audio and video from a Kinect using avconv but I seem to have trouble specifying the right options. avconv -f video4linux2 -video_size 640x480 -c:v h264 -c:a ac3 -i /dev/video1 test.mp4 results in: avconv: /build/buildd/libav-extra-0.8.3ubuntu0.12.04.1/libav/libavutil/mathematics.c:79: av_rescale_rnd: Assertion `c > 0' failed. Aborted (core dumped) Cheese is perfectly happy recording the audio and video from the Kinect, so it's not a setup issue. For added fun, I also need to display the video at the same time.

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• How do i use latest Pulseaudio in 11.10?

  - by YumYumYum

  Ubuntu 11.04 i had pulseaudio from source compiled and i used it to learn, it always worked (git versions). But since i have Ubuntu 11.10, i can install it but i can not use it anymore like i do in 11.04 before. Everytime i play something its throwing this: $ speaker-test speaker-test 1.0.24.2 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 192 to 2097152 Period size range from 64 to 699051 Using max buffer size 2097152 Periods = 4 ALSA lib pcm_pulse.c:746:(pulse_prepare) PulseAudio: Unable to create stream: Invalid argument Unable to set hw params for playback: Input/output error Setting of hwparams failed: Input/output error How to make pulseaudio work in 11.10 from source?

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