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  • ffmpeg: video file played OK on Ubuntu, but no sound on XP

    - by Andy Le
    I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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  • Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

    - by davr
    First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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  • Pixel Shader Giving Black output

    - by Yashwinder
    I am coding in C# using Windows Forms and the SlimDX API to show the effect of a pixel shader. When I am setting the pixel shader, I am getting a black output screen but if I am not using the pixel shader then I am getting my image rendered on the screen. I have the following C# code using System; using System.Collections.Generic; using System.Linq; using System.Windows.Forms; using System.Runtime.InteropServices; using SlimDX.Direct3D9; using SlimDX; using SlimDX.Windows; using System.Drawing; using System.Threading; namespace WindowsFormsApplication1 { // Vertex structure. [StructLayout(LayoutKind.Sequential)] struct Vertex { public Vector3 Position; public float Tu; public float Tv; public static int SizeBytes { get { return Marshal.SizeOf(typeof(Vertex)); } } public static VertexFormat Format { get { return VertexFormat.Position | VertexFormat.Texture1; } } } static class Program { public static Device D3DDevice; // Direct3D device. public static VertexBuffer Vertices; // Vertex buffer object used to hold vertices. public static Texture Image; // Texture object to hold the image loaded from a file. public static int time; // Used for rotation caculations. public static float angle; // Angle of rottaion. public static Form1 Window =new Form1(); public static string filepath; static VertexShader vertexShader = null; static ConstantTable constantTable = null; static ImageInformation info; [STAThread] static void Main() { filepath = "C:\\Users\\Public\\Pictures\\Sample Pictures\\Garden.jpg"; info = new ImageInformation(); info = ImageInformation.FromFile(filepath); PresentParameters presentParams = new PresentParameters(); // Below are the required bare mininum, needed to initialize the D3D device. presentParams.BackBufferHeight = info.Height; // BackBufferHeight, set to the Window's height. presentParams.BackBufferWidth = info.Width+200; // BackBufferWidth, set to the Window's width. presentParams.Windowed =true; presentParams.DeviceWindowHandle = Window.panel2 .Handle; // DeviceWindowHandle, set to the Window's handle. // Create the device. D3DDevice = new Device(new Direct3D (), 0, DeviceType.Hardware, Window.Handle, CreateFlags.HardwareVertexProcessing, presentParams); // Create the vertex buffer and fill with the triangle vertices. (Non-indexed) // Remember 3 vetices for a triangle, 2 tris per quad = 6. Vertices = new VertexBuffer(D3DDevice, 6 * Vertex.SizeBytes, Usage.WriteOnly, VertexFormat.None, Pool.Managed); DataStream stream = Vertices.Lock(0, 0, LockFlags.None); stream.WriteRange(BuildVertexData()); Vertices.Unlock(); // Create the texture. Image = Texture.FromFile(D3DDevice,filepath ); // Turn off culling, so we see the front and back of the triangle D3DDevice.SetRenderState(RenderState.CullMode, Cull.None); // Turn off lighting D3DDevice.SetRenderState(RenderState.Lighting, false); ShaderBytecode sbcv = ShaderBytecode.CompileFromFile("C:\\Users\\yashwinder singh\\Desktop\\vertexShader.vs", "vs_main", "vs_1_1", ShaderFlags.None); constantTable = sbcv.ConstantTable; vertexShader = new VertexShader(D3DDevice, sbcv); ShaderBytecode sbc = ShaderBytecode.CompileFromFile("C:\\Users\\yashwinder singh\\Desktop\\pixelShader.txt", "ps_main", "ps_3_0", ShaderFlags.None); PixelShader ps = new PixelShader(D3DDevice, sbc); VertexDeclaration vertexDecl = new VertexDeclaration(D3DDevice, new[] { new VertexElement(0, 0, DeclarationType.Float3, DeclarationMethod.Default, DeclarationUsage.PositionTransformed, 0), new VertexElement(0, 12, DeclarationType.Float2 , DeclarationMethod.Default, DeclarationUsage.TextureCoordinate , 0), VertexElement.VertexDeclarationEnd }); Application.EnableVisualStyles(); MessagePump.Run(Window, () => { // Clear the backbuffer to a black color. D3DDevice.Clear(ClearFlags.Target | ClearFlags.ZBuffer, Color.Black, 1.0f, 0); // Begin the scene. D3DDevice.BeginScene(); // Setup the world, view and projection matrices. //D3DDevice.VertexShader = vertexShader; //D3DDevice.PixelShader = ps; // Render the vertex buffer. D3DDevice.SetStreamSource(0, Vertices, 0, Vertex.SizeBytes); D3DDevice.VertexFormat = Vertex.Format; // Setup our texture. Using Textures introduces the texture stage states, // which govern how Textures get blended together (in the case of multiple // Textures) and lighting information. D3DDevice.SetTexture(0, Image); // Now drawing 2 triangles, for a quad. D3DDevice.DrawPrimitives(PrimitiveType.TriangleList , 0, 2); // End the scene. D3DDevice.EndScene(); // Present the backbuffer contents to the screen. D3DDevice.Present(); }); if (Image != null) Image.Dispose(); if (Vertices != null) Vertices.Dispose(); if (D3DDevice != null) D3DDevice.Dispose(); } private static Vertex[] BuildVertexData() { Vertex[] vertexData = new Vertex[6]; vertexData[0].Position = new Vector3(-1.0f, 1.0f, 0.0f); vertexData[0].Tu = 0.0f; vertexData[0].Tv = 0.0f; vertexData[1].Position = new Vector3(-1.0f, -1.0f, 0.0f); vertexData[1].Tu = 0.0f; vertexData[1].Tv = 1.0f; vertexData[2].Position = new Vector3(1.0f, 1.0f, 0.0f); vertexData[2].Tu = 1.0f; vertexData[2].Tv = 0.0f; vertexData[3].Position = new Vector3(-1.0f, -1.0f, 0.0f); vertexData[3].Tu = 0.0f; vertexData[3].Tv = 1.0f; vertexData[4].Position = new Vector3(1.0f, -1.0f, 0.0f); vertexData[4].Tu = 1.0f; vertexData[4].Tv = 1.0f; vertexData[5].Position = new Vector3(1.0f, 1.0f, 0.0f); vertexData[5].Tu = 1.0f; vertexData[5].Tv = 0.0f; return vertexData; } } } And my pixel shader and vertex shader code are as following // Pixel shader input structure struct PS_INPUT { float4 Position : POSITION; float2 Texture : TEXCOORD0; }; // Pixel shader output structure struct PS_OUTPUT { float4 Color : COLOR0; }; // Global variables sampler2D Tex0; // Name: Simple Pixel Shader // Type: Pixel shader // Desc: Fetch texture and blend with constant color // PS_OUTPUT ps_main( in PS_INPUT In ) { PS_OUTPUT Out; //create an output pixel Out.Color = tex2D(Tex0, In.Texture); //do a texture lookup Out.Color *= float4(0.9f, 0.8f, 0.0f, 1); //do a simple effect return Out; //return output pixel } // Vertex shader input structure struct VS_INPUT { float4 Position : POSITION; float2 Texture : TEXCOORD0; }; // Vertex shader output structure struct VS_OUTPUT { float4 Position : POSITION; float2 Texture : TEXCOORD0; }; // Global variables float4x4 WorldViewProj; // Name: Simple Vertex Shader // Type: Vertex shader // Desc: Vertex transformation and texture coord pass-through // VS_OUTPUT vs_main( in VS_INPUT In ) { VS_OUTPUT Out; //create an output vertex Out.Position = mul(In.Position, WorldViewProj); //apply vertex transformation Out.Texture = In.Texture; //copy original texcoords return Out; //return output vertex }

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  • [iptables] Why do 'iptables -A OUTPUT -j REJECT' at the end of the chain OUTPUT override the previous rules??

    - by Serge
    Those are my IPTABLES rules: iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A OUTPUT -p tcp --dport 22 -j ACCEPT iptables -A OUTPUT -p udp --dport 22 -j ACCEPT iptables -A OUTPUT -p tcp --dport 80 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT iptables -A OUTPUT -p tcp --dport 53 -j ACCEPT iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp -m tcp --dport 22 -m state --state NEW -m recent --set --name DEFAULT --rsource iptables -A INPUT -p tcp -m tcp --dport 22 -m state --state NEW -m recent --update --seconds 180 --hitcount 4 --name DEFAULT --rsource -j DROP iptables -A INPUT -p tcp -m state --state NEW --dport 22 -j ACCEPT iptables -A OUTPUT -j REJECT iptables -A INPUT -j REJECT iptables -A FORWARD -j REJECT Im using a remote ssh conetion to set them up, but after i set: iptables -A OUTPUT -j REJECT My connection get lost. I have read all the documentation for Iptables and i can figure out anything, the global Rejects for INPUT work well because i can access to the web page but i get a timeout for ssh. Any idea? Thanks

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  • What is the best way to get an audio file duration in Android?

    - by Gilead
    Hi! I'm using a SoundPool [ 1 ] to play audio clips in my app. All is fine but I need to know when the clip playback has finished. At the moment I track it in my app by obtaining the duration of each clip using a MediaPlayer [ 2 ] instance. That works fine but it looks wasteful to load each file twice, just to get the duration. I could roughly calculate the duration myself knowing the length of the file (available from the AssetFileDescriptor [ 3 ]) but I'd still need to know the sample rate and the number of channels. I see two potential solutions to that problem: Figuring out when a clip has finished playing (doesn't seem to be possible with SoundClip). Having a class which could load just the header of an audio file and give me the sample rate/number of channels (and, ideally, the sample count to get the exact duration). Any suggestions? Thanks, Max The code I'm using at the moment (works fine but is rather heavy for the purpose): String[] fileNames = ... MediaPlayer mp = new MediaPlayer(); for (String fileName : fileNames) { AssetFileDescriptor d = context.getAssets().openFd(fileName); mp.reset(); mp.setDataSource(d.getFileDescriptor(), d.getStartOffset(), d.getLength()); mp.prepare(); int duration = mp.getDuration(); // ... } On a side note, this question has already been asked [ 4 ] but got no answers.

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  • Issues regarding playing audio files in a JME midlet.

    - by Northernen
    I am making a midlet which is to be used to play out local audio files. It is obviously not working. I am getting a null reference on the "is" variable, in the code snippet shown below. 1. try{ 2. System.out.println("path: " + this.getClass()); 3. InputStream is = this.getClass().getResourceAsStream("res/01Track.wav"); 4. p1=Manager.createPlayer(is, "audio"); 5. p1.realize(); 6. p1.prefetch(); 7. p1.start(); 8. } 9. catch(Exception e){ 10. System.out.println(e.getMessage()); 11. } I assume there is something wrong with the "this.getClass().getResourceAsStream("res/01Track.wav")" bit, but I can not for the life of me figure out why, and I have tried referring to the file in 20 different ways. If I printline "this.getClass()" it gives me "path: class Mp3spiller". The absolute path to "01Track.wav" is "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller\res\01Track.wav". Am I completely wrong in thinking that I should refer relatively to "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller"? If anyone could point out what I am doing wrong, I would be grateful. I have basically stolen the code from a tutorial I found online, so I would have thought it would be working.

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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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  • Skipping video and audio with PS3MediaServer

    - by MaxMackie
    I'm using the latest PS3MediaServer build right from the repos suggested in the Ubuntu Wiki. I'm streaming multiple movies from my server (Ubuntu 10.04 LTS) to my PS3 over wireless. Sometimes, during some movies, the audio and the video will begin skipping. This can last anywhere between 5 and 30 seconds before it goes back to normal. I have a four core i5 processor and 8GB of DDR3 RAM so I don't think my computer is having a hard time keeping up with the transcoding. So this leads me to believe it's either sub-optimal transcoding options from within PS3MS or my network can't handle the heat. Other than the out-of-box configuration, is there any way I can tweak the settings for the application to use my resources more efficiently?

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  • Internet stops working after heavy downloading, video/audio streaming etc

    - by Kuba Szwed
    As mentioned in title, Internet stops working on my PC after heavy downloading, video/audio streaming etc. There are no errors, no disconnections etc. Simply after some time (certain amount of data downloaded) I can't get any more. If I try using ping afterwards nothing happens. If ping is running simultaneously with streaming/downloading I get some correct responses and then it keeps showing an error. What helps is re-plugging my Pentagram USB wifi card, but I hope there is a better solution. Edit: One more thing: my friend who works in IT suggested that it might have something to do with cache (DNS cache? I don't remember him specifying) getting filled while it should be emptied automatically.

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  • How to cut audio file with avconv?

    - by x-yuri
    I have a hard time trying to figure out how to cut a file with avconv. Here's the command I use: avconv -ss 52:13:49 -t 01:13:52 -i RR119Accessibility.wav RR119Accessibility-2.wav But it doesn't work. I get the whole file as a result. Well, almost the whole file. Somehow the resulting file has duration 1:16:31 instead of 1:17:23. Also I believe I executed this command in every possible way: with -ss and -t after -i, with -t specifying ending point, with mp3 files, with specifying audio codec, with ffmpeg. Am I doing it wrong?

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  • Installing old Loki games on 12.04 64-bit results in no audio

    - by FlabbergastedPickle
    All, Here's an interesting problem. I followed instructions provided online for installing Loki Games' Heroes of Might and Magic 3 (see http://www.swanson.ukfsn.org/loki/ and http://wtanaka.com/node/7641) and got it installed and patched to the latest version. However, every time I start it regardless whether the pulseaudio is running, I get the following error: LD_LIBRARY_PATH=/usr/local/lib/Loki_Compat/ /usr/local/lib/Loki_Compat/ld-linux.so.2 /usr/local/games/Heroes3/heroes3.dynamic ALSA lib conf.c:3314:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM default Couldn't open audio: My first soundcard is HDMI output and my second one is the actual soundcard (HP DM1 running 12.04 64-bit with latest updates). I did set up /etc/asound.conf as follows: asound.conf pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } So, the default soundcard should work ok. Between Shadowgrounds that also stopped working and this it appears a there may be some unfinished business/regressions in 32-bit support on 64-bit systems in 12.04. Any thoughts?

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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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  • MERGE gives better OUTPUT options

    - by Rob Farley
    MERGE is very cool. There are a ton of useful things about it – mostly around the fact that you can implement a ton of change against a table all at once. This is great for data warehousing, handling changes made to relational databases by applications, all kinds of things. One of the more subtle things about MERGE is the power of the OUTPUT clause. Useful for logging.   If you’re not familiar with the OUTPUT clause, you really should be – it basically makes your DML (INSERT/DELETE/UPDATE/MERGE) statement return data back to you. This is a great way of returning identity values from INSERT commands (so much better than SCOPE_IDENTITY() or the older (and worse) @@IDENTITY, because you can get lots of rows back). You can even use it to grab default values that are set using non-deterministic functions like NEWID() – things you couldn’t normally get back without running another query (or with a trigger, I guess, but that’s not pretty). That inserted table I referenced – that’s part of the ‘behind-the-scenes’ work that goes on with all DML changes. When you insert data, this internal table called inserted gets populated with rows, and then used to inflict the appropriate inserts on the various structures that store data (HoBTs – the Heaps or B-Trees used to store data as tables and indexes). When deleting, the deleted table gets populated. Updates get a matching row in both tables (although this doesn’t mean that an update is a delete followed by an inserted, it’s just the way it’s handled with these tables). These tables can be referenced by the OUTPUT clause, which can show you the before and after for any DML statement. Useful stuff. MERGE is slightly different though. With MERGE, you get a mix of entries. Your MERGE statement might be doing some INSERTs, some UPDATEs and some DELETEs. One of the most common examples of MERGE is to perform an UPSERT command, where data is updated if it already exists, or inserted if it’s new. And in a single operation too. Here, you can see the usefulness of the deleted and inserted tables, which clearly reflect the type of operation (but then again, MERGE lets you use an extra column called $action to show this). (Don’t worry about the fact that I turned on IDENTITY_INSERT, that’s just so that I could insert the values) One of the things I love about MERGE is that it feels almost cursor-like – the UPDATE bit feels like “WHERE CURRENT OF …”, and the INSERT bit feels like a single-row insert. And it is – but into the inserted and deleted tables. The operations to maintain the HoBTs are still done using the whole set of changes, which is very cool. And $action – very convenient. But as cool as $action is, that’s not the point of my post. If it were, I hope you’d all be disappointed, as you can’t really go near the MERGE statement without learning about it. The subtle thing that I love about MERGE with OUTPUT is that you can hook into more than just inserted and deleted. Did you notice in my earlier query that my source table had a ‘src’ field, that wasn’t used in the insert? Normally, this would be somewhat pointless to include in my source query. But with MERGE, I can put that in the OUTPUT clause. This is useful stuff, particularly when you’re needing to audit the changes. Suppose your query involved consolidating data from a number of sources, but you didn’t need to insert that into the actual table, just into a table for audit. This is now very doable, either using the INTO clause of OUTPUT, or surrounding the whole MERGE statement in brackets (parentheses if you’re American) and using a regular INSERT statement. This is also doable if you’re using MERGE to just do INSERTs. In case you hadn’t realised, you can use MERGE in place of an INSERT statement. It’s just like the UPSERT-style statement we’ve just seen, except that we want nothing to match. That’s easy to do, we just use ON 1=2. This is obviously more convoluted than a straight INSERT. And it’s slightly more effort for the database engine too. But, if you want the extra audit capabilities, the ability to hook into the other source columns is definitely useful. Oh, and before people ask if you can also hook into the target table’s columns... Yes, of course. That’s what deleted and inserted give you.

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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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  • 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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  • How to fix a gstreamer error

    - by BJsgoodlife
    I upgraded to 14.04 and now gstreamer is not working. What I am trying to accomplish is to hear the audio input on my computer coming from a ham radio. This is the command that I am using: gst-launch pulserc ! pulseink. This is the error message that I am receiving: ERROR: pipeline could not be constructed: no element "pulserc". I am wondering if I should purge gstreamer, or if there is another command that I should put it. Thank you again everyone for all of your help.

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  • Noisy audio recording in Ubuntu Linux

    - by Haresh K Miriyala
    I'm unable to record audio without noise on my Ubuntu laptop . It is not a grounding problem . My windows os is able to record voice without the static noise so, my hardware is fine . I read in the Windows forums that the os has software noise filtering mechanism that's automatically enabled. How do I do the same in Ubuntu? If not, please let me know how to filter the static noise. Any help is appreciated. Thanks!

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  • video/audio output via HDMI Ubuntu 12.04

    - by lostNfound
    I've been out of the Ubuntu loop for quite a while now and have a completely new laptop now. Just installed Ubuntu 12.04 64-bit and would like to output my video and my audio via HDMI to my television. the following is the lspci | grep VGA for my computer. please tell me if there is any additional information needed and preferably how to obtain it and i will be more than happy to oblige. thank you in advance for your time and assistance in this matter. 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 01:00.0 VGA compatible controller: NVIDIA Corporation GF108 [GeForce GT 540M] (rev a1) Edit: every time i restart my computer, after a short moment, i get an error message stating something along the lines "sorry, jockey needed to close unexpectedly." after researching, i discovered jockey is the name of the "additional drivers," which after initial installation, ubuntu informed me of proprietary drivers available. those are no longer available, and this error continues to occur.

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  • play sound from different applications on different output devices (speakers, headphones)

    - by Mike
    I want music (played e.g. via Audacious) to be played on speaker system, and all other sounds produced by other applications (including ubuntu sound effects) to be played on headphones. My computer has sound connectors at its back, and also connectors on its front panel. When I connect both headphones and speakers, only headphones work (I take it, the front connectors take precedence?). Should I purchase another sound card (in addition to the motherboard-integrated sound I have)? When I go to Audacious output settings, I see only the ouput plugin selection list, with PulseAudio selected and options like ALSA, OSS4 etc. But there's no facility to select particular output device (and I guess it wouldn't magically appear even if I had the second soundcard). Is this at all possible to bind specific application to particular output device?

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  • Radeon HD5570 HDMI Video Card 5.1 Audio doesn't work

    - by ryandlf
    I am using Ubuntu and XMBC on my HTPC and have chosen the Radeon HD5570 Video card which has an HDMI output. In the sound preferences there is no surround sound option for the video card just stereo and although I can get sound through it in XBMC, my receiver does not state Dolby Digital on movies that are in fact Dolby so its definitely not giving me the true sound it should. Does this card not support surround sound through HDMI and I somehow missed it? If that is the case does anyone have suggestion that has been tested and works? Id like to know its going to work before investing in yet another video card. UPDATE I purchased a Nvidia GeForce GTS 450, plugged it in, downloaded the proprietary driver from the system control panel, disabled the onboard audio from the BIOS (not sure if this was necessary but I did it anyways), and changed the sound settings to use the new video card. Everything works flawlessly. It was a seemless setup.

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  • Podcast site - Serve audio files with CDN

    - by Bobe
    I am managing a small podcast website hosted on a shared server. Currently there are only eight or nine episodes, each of which are about 50 MB, so bandwidth is not really an issue at the moment. However, looking forward, would it be feasible to use a "free" CDN like Cloudflare to serve the audio files? If so, how would I set this up? I took a quick look at it before, and it seems you have to have your whole site routed (is that the right term?) through the CDN rather than just specific files or filetypes. I'd like some clarification on this.

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  • sometimes, strange audio distortion(peeping, scratching), ubuntu 12.04

    - by richi902
    i have a problem with my sound in ubuntu 12.04. the problem is, that sometimes out of nowhere, when switching songs, or playing youtube videos, changing volume with my keyboard buttons, that the sound gets distorted(peeping, scratching). i dont know if it is related, but when i skip through music in rythmbox, there is also a little scratching noise. i can sometimes temporarly fix it: for youtube videos, i refresh the page, and sometimes it works agian normal, mostly not. for audio playback with rythmbox, i have to pause the song for sometime, and resume it, and hope that it works. before all that,i have changed my soundcard to "Analog Surround 5.1" in the sound settings from ubuntu, but i also used alsamixer to change it from 2 channels to 6 channels, since changing in ubuntu sound settings alone wasnt enough to make the other speaks work. i use a ASUS P8-H61-M LE B3 Revision Motherboard. which has a built in surround soundcard.

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  • ffmpeg hangs when creating a video

    - by FearUs
    I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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  • Getting input and output from a jar file run from java class?

    - by Jack L.
    Hi, I have a jar file that runs this code: public class InputOutput { /** * @param args * @throws IOException */ public static void main(String[] args) throws IOException { boolean cont = true; BufferedReader in = new BufferedReader(new InputStreamReader(System.in)); while (cont) { System.out.print("Input something: "); String temp = in.readLine(); if (temp.equals("end")) { cont = false; System.out.println("Terminated."); } else System.out.println(temp); } } } I want to program another java class that executes this jar file and can get the input and send output to it. Is it possible? The current code I have is this but it is not working: public class JarTest { /** * Test input and output of jar files * @author Jack */ public static void main(String[] args) { try { Process io = Runtime.getRuntime().exec("java -jar InputOutput.jar"); BufferedReader in = new BufferedReader(new InputStreamReader(io.getInputStream())); OutputStreamWriter out = new OutputStreamWriter(io.getOutputStream()); boolean cont = true; BufferedReader consolein = new BufferedReader(new InputStreamReader(System.in)); while (cont) { String temp = consolein.readLine(); out.write(temp); System.out.println(in.readLine()); } } catch (IOException e) { e.printStackTrace(); } } } Thanks for your help

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  • Accessing both stored procedure output parameters AND the result set in Entity Framework?

    - by MS.
    Is there any way of accessing both a result set and output parameters from a stored procedure added in as a function import in an Entity Framework model? I am finding that if I set the return type to "None" such that the designer generated code ends up calling base.ExecuteFunction(...) that I can access the output parameters fine after calling the function (but of course not the result set). Conversely if I set the return type in the designer to a collection of complex types then the designer generated code calls base.ExecuteFunction<T>(...) and the result set is returned as ObjectResult<T> but then the value property for the ObjectParameter instances is NULL rather than containing the proper value that I can see being passed back in Profiler. I speculate the second method is perhaps calling a DataReader and not closing it. Is this a known issue? Any work arounds or alternative approaches? Edit My code currently looks like public IEnumerable<FooBar> GetFooBars( int? param1, string param2, DateTime from, DateTime to, out DateTime? createdDate, out DateTime? deletedDate) { var createdDateParam = new ObjectParameter("CreatedDate", typeof(DateTime)); var deletedDateParam = new ObjectParameter("DeletedDate", typeof(DateTime)); var fooBars = MyContext.GetFooBars(param1, param2, from, to, createdDateParam, deletedDateParam); createdDate = (DateTime?)(createdDateParam.Value == DBNull.Value ? null : createdDateParam.Value); deletedDate = (DateTime?)(deletedDateParam.Value == DBNull.Value ? null : deletedDateParam.Value); return fooBars; }

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