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  • With the advent of HTML 5, is there a point in using COMET anymore?

    - by h2g2java
    I am very tempted to use long wait http or periodic polling by the client to set up pseudo-sockets on the browser side, for an application that would be used publicly. But then on the 2nd thought, I am thinking HTML 5 is here. But on the 3rd thought, what is the percentage of browsers out there that remain non-HTML5 within 12 months, 24 months, 36 months? If there are at least 20% of browsers still incapable of HTML5, then I cannot depend on HTML5 because 20% of users not being able to access an application is a significant amount. What do you think, how would your advice be (to me and to developers in general)? Q1. Is there any point in rigging in COMET into an application anymore? I am thinking of gwt comet - http://code.google.com/p/gwt-comet/. Q2. Should we release a new public application within the next 2 months that is dependent on HTML5 sockets and tell non-HTML5 browser users "sorry, your browser version cannot access this application"? Or should we architect the apps to use communication like GWT RPC? Q3. I am also very distrustful of long wait http request. I have never used it before but I have a horrible feeling about it. I have been using 10 to 20 second client-side polling. Is long wait http request risky (risk of hanging a browser session)? Does long wait request present any additional security risk?

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  • Get binary data from audio impulses

    - by Timo
    I have IR sensor which have TRS plug and I can record my remotes signals into audio. Now I want to control my computer with TV remote, but I don't have any clue how to compare audio input with pre-recorded audio. But after I realized that these audio waves contains only some kind data (binary) I can turn these into binary or hex, so it is much easier to compare. Waves look just like this: http://i.imgur.com/lCIyl.png And this: ttp://i.imgur.com/goJ6d.png These are records of "OK" button, sometimes there are some impulses on right channel too and I don't know why, it seems like connections in sensor are damaged maybe. Ok thats not matter, anyway I need help with python program which read these impulses and turn these into binary, in realtime from audio input(mic). I know it's sounds like "Do it for me, while I enjoy my life", but I don't have experiences with sound transforming/reading... I've looking for python examples for recording and reading audio, but unsuccessfully.

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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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  • Custom Dreamweaver DocTypes

    - by Hugh Guiney
    Dreamweaver CS5 with Dreamweaver HTML5 Pack 1.2.7 Windows 7 x64 When I go to create a new document and select the HTML5 DocType, Dreamweaver gives me the legacy encoding/character set declaration: <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> I want to replace it with the new, abbreviated style: <meta charset="utf-8"> The relevant file seems to be %ProgramFiles(x86)%\Adobe\Adobe Dreamweaver CS5\configuration\DocumentTypes\NewDocuments\Default.html, which has a blank charset, that is then apparently replaced with the appropriate character set dynamically: <meta http-equiv="Content-Type" content="text/html; charset="> I changed it, but then new documents show up like this: <meta charset=""> <title>Untitled Document</title> <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> It seems Dreamweaver added the legacy declaration back in after my modification—and as far as I can tell, there's no way to specify that the charset definition should go in-between the quotes, either. Additionally, any modifications to Default.html apply to every DocType, whereas I only want this change to apply to the HTML5 DocType. Is there anything in the configuration files that would allow me to make any of these customizations? If not, is there an extension that does it?

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  • ffmpeg fully html 5 converion supported

    - by user58542
    I need to know about library that required for ffmpeg for convert any format of audio and video for supported following format to have best configuration to convert audio and video files for html5 formats. I need for support mp3 and ogg for audio files. Also need for support FLV, H.264, Ogg Theora and VP8 (WebM) for video files. I'm using debian also using deb-multimedia repository. I need list of packages required for this formats(any format to this formats). also any configuration for install from package management or compile via latest ffmpeg repository. Thanks a lot.

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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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  • Audio Panning using RtAudio

    - by user1801724
    I use the RtAudio library. I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use RtAudio in duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. I have searched on the web, but I did not find anything useful. Should I apply a filter on the output buffer? What kind of filter?

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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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  • 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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  • 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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  • How to get audio spectrum analysis?

    - by Mrwolfy
    I need to find or create a tool that analyzes the audio spectrum of a sound file (like a .wav or .mp3). I need to output the "volume" or power of x number of frequency bands and output the data as text. This will be used to produce a visualization, a graphic equalizer like you'd see on a stereo. I am currently looking at python to do it. My question is are there some tools out there that would do this (signal processing), like math works or others? I don't have any experience with them so any advice would be appreciated.

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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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