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  • Which audio playback technology do I need for this?

    - by mystify
    I have trouble choosing the right audio playback technology. There's a ton of technologies to use on the iPhone, it's so confusing. What I need to do is this: start playing short sounds ranging between 0.1 and 2 seconds high quality playback, no crackle (I heard some of the iPhone audio playback technologies do a crackle sound on start or end, which is bad!) ability to start playback of a sound, while there's already another one playing right now (two, three or more sounds at the same time) What would you suggest here, and why? Thanks :-)

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  • Is it possible to capture audio output and apply effects to it?

    - by Ciaran
    Using .NET and DirectSound I want to be able to take all output sound that is coming from my audio device and apply effects to it. I've had a quick look at the docs on MSDN and there doesn't seem to be any explanation as to how to do something like this. I've read elsewhere that you'd be better off writing a driver to sit in front of your real audio driver and have that do whatever you want with the sound. Any ideas anyone to push me in the right direction?

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  • Speech recognition webservice that scores the accuracy of one audio clips vs. another?

    - by wgpubs
    Does such a thing exist? Building a Rails based web application where users can upload an audio file of them speaking that then needs to be compared to another audio file for the purposes of determining how similar to voices are. Ideally I'd like to simply get a response that gives me a score of how similar they are in terms of percentage (e.g. 75% similar etc...). Anyone have any ideas? Thanks

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  • How to create live stream audio for web-sites ???

    - by Kathir
    Hi All, We are storing sound from mic to pc via sound forge. We would like to broadcast the sound which comes from the mic to the pc as live streaming audio. Basically a person speaks in a mic, we like to give it as live stream audio. The web-site is hosted on yahoo server. Can you please let me know in what are the ways we can achieve this? Thanks, Kathir

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  • No HDMI Audio with GeForce 9600GT and nForce board

    - by Bobby
    I've been trying to get HDMI with sound working for the last few days, and I'm a little bit out of ideas. (I've verified that the hardware/Setup works via Windows.) aplay does not list my HDMI device: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 I've already compiled the alsa drivers (1.0.24) from a snapshot (with --with-oss=no) and added the line options snd-hda-intel model=auto # Tried 3stack-dig and 6stack-dig too to /etc/modprobe.d/alsa-base.conf. Still, the device does not show up. If it is important, the HDMI TV is at the moment not configured to be part of the X session (I've tried that to, at least with X restart, and it didn't change anything). What did I miss? $ lspci 00:00.0 Host bridge: nVidia Corporation Device 07c3 (rev a2) 00:00.1 RAM memory: nVidia Corporation nForce 630i memory controller (rev a2) 00:01.0 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.1 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.2 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.3 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.4 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.5 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.6 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:02.0 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:03.0 ISA bridge: nVidia Corporation MCP73 LPC Bridge (rev a2) 00:03.1 SMBus: nVidia Corporation MCP73 SMBus (rev a1) 00:03.2 RAM memory: nVidia Corporation MCP73 Memory Controller (rev a1) 00:03.4 RAM memory: nVidia Corporation MCP73 Memory Controller (rev a1) 00:04.0 USB Controller: nVidia Corporation GeForce 7100/nForce 630i USB (rev a1) 00:04.1 USB Controller: nVidia Corporation MCP73 [nForce 630i] USB 2.0 Controller (EHCI) (rev a1) 00:08.0 IDE interface: nVidia Corporation MCP73 IDE (rev a1) 00:09.0 Audio device: nVidia Corporation MCP73 High Definition Audio (rev a1) 00:0a.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0b.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0c.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0d.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0e.0 IDE interface: nVidia Corporation MCP73 IDE (rev a2) 00:0f.0 Ethernet controller: nVidia Corporation MCP73 Ethernet (rev a2) 02:00.0 VGA compatible controller: nVidia Corporation G94 [GeForce 9600 GT] (rev a1)   $ aplay -L default pulse Playback/recording through the PulseAudio sound server front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Digital IEC958 (S/PDIF) Digital Audio Output dmix:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample mixing device dmix:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample mixing device dsnoop:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample snooping device dsnoop:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample snooping device hw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct hardware device without any conversions hw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct hardware device without any conversions plughw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Hardware device with all software conversions plughw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Hardware device with all software conversions

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  • Bose USB audio: crackling popping sound, eventually die

    - by Richard Barrett
    I've been trying to troubleshoot this issue for a while now. Any help would be much appreciated. I'm having trouble getting my Bose "Companion 5 multimedia speakers" working with my installation of Ubuntu 12.04 (link to Bose product here: http://www.bose.com/controller?url=/shop_online/digital_music_systems/computer_speakers/companion_5/index.jsp ). The issue seems to be low level (not just Ubuntu). What happens: When I boot into Ubuntu, I can get Rhythm box to play ok. However, if I try anything else (an .avi file, a webpage, or Clementine player with mp3 files) I get crackling, popping, or choppy sounds. If I move the mouse around, especially if it seems graphic intensive, the problem gets worse (more crackling noises). The more taxing it appears to be, the more likely it is that the sound will just die altogether until I reboot. For some reason the videos at www.bloomberg.com seem especially bad for it (my sound normally goes dead in under 45 seconds and won't work until reboot). Both my desktop running Ubuntu 12.04 and my laptop (running the same) have the same crackling problem. Troubleshooting so far: A friend of mine who knows linux well tried to solve it for me without any luck. He took pulseaudio out of the equation, but still had the problem just using AlSA. Among the many things he tried was adjusting the latency, but that didn't help either. I've also tried things like adjusting the USB device settings in the config file from -2 to -1 so that it will use my USB sound and I also commented out the lines that would stop that. These don't do anything. (That really seems like it's for someone who is getting no sound at all, so it's not surprising this won't work.) My friend's laptop running his Archlinux could play my Bose USB speakers without any problems. I also tried setting my daemon.conf file to use 6 channels (based on this http://lotphelp.com/lotp/configure-ubuntu-51-surround-sound ) but that didn't work either. I recently used a DVD to boot into Ubuntu Studio 12.04 (because it uses a live audio kernel) and this happened: I got perfect sound for a minute or two When I started moving windows around while sound was playing, the sound died again. Perhaps more interesting: There is a headphone out jack on the Bose system. When I use it, the audio is perfect for all applications (even the deadly bloomberg.com videos with .avi playing at the same time and moving around windows). Also, there is an audio-in jack on the Bose system. I can use a male-to-male mini jack to go from my soundcard's output to the Bose input and then all sound works perfectly. -However, it still requires the Bose to be plugged in to USB, otherwise I lose all sound. Any thoughts? Any suggestions for trouble shooting? (Or any suggestions for somewhere else to post to solve this?) Any logs or other files I can provide to help someone help me work this out? Your help is much appreciated! Rick BTW: I sometimes get people posting responses like "My Bose USB system works great with Ubuntu 12.04," without any more details. Is there anything I should ask such people to narrow down my problem? (It's kind of annoying to hear such a response because it doesn't help solve my problem.)

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  • linux mint VIA sound issue

    - by user2699451
    So I installed linux Mint 15 "Olivia" 64 bit on my Mecer W550EU laptop I have HD Audio with a VIA chipset charles-W55xEU charles # lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_via 51018 1 snd_hda_intel 39619 5 snd_hda_codec 136453 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13602 1 snd_hda_codec snd_pcm 97451 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 1 snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 19 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_via,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd And my sound card charles-W55xEU charles # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1802 Analog [VT1802 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1802 HP [VT1802 HP] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 and my audio device charles-W55xEU charles # lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High `Definition Audio Controller (rev 04)` Subsystem: CLEVO/KAPOK Computer Device 0550 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at f7c10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Sometimes when I boot up, soundworks, other times it doenst, it is completely random, so far, no-one on xchat linux help or linux mint forums was able to help me, I have always had issues with sound on VIA chipsets I have: sudo apt-get upgrade && apt-get install mint-meta-cinnamon it seemed to help but after 2-3 reboots, the problem came back, btw, everytime I checked, pulse audio is selected to Duplex Audio Input & Output and alsa mixer is always unmuted!

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  • How can I tweak this A* search pathfinding algorithm to handle different terrain movement values?

    - by user422318
    I'm creating a 2D map-based action game with similar interaction design as Diablo II. In other words, the player clicks around a map to move their player. I just finished player movement and am moving on to pathfinding. In the game, enemies should charge the player's character. There are also five different terrain types that give different movement bonuses. I want the AI to take advantage of these terrain bonuses as they try to reach the player. I was told to check out the A* search algorithm (http://en.wikipedia.org/wiki/A*_search_algorithm). I'm doing this game in HTML5 and JavaScript, and found a version in JavaScript: http://www.briangrinstead.com/blog/astar-search-algorithm-in-javascript I'm trying to figure out how to tweak it though. Below are my ideas about what I need to change. What else do I need to worry about? When I create a graph, I will need to initialize the 2D array I pass in passed on with a traversal of a map that corresponds to the different terrain types. in graph.js: "GraphNodeType" definition needs to be modified to handle the 5 terrain types. There will be no walls. in astar.js: The g and h scoring will need to be modified. How should I do this? in astar.js: isWall() should probably be removed. My game doesn't have walls. in astar.js: I'm not sure what this is. I think it indicates a node that isn't valid to be processed. When would this happen, though? At a high level, how do I change this algorithm from "oh, is there a wall there?" to "will this terrain get me to the player faster than the terrain around me?" Because of time, I'm also debating reusing my Bresenham algorithm for the enemies. Unfortunately, the different terrain movement bonuses won't be used by the AI, which will make the game suck. :/ I'd really like to have this in for the prototype, but I'm not a developer by trade nor am I a computer scientist. :D If you know of any code that does what I'm looking for, please share! Sanity check tips for this are also appreciated.

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  • what would be a good way to implement/render a 2d tiled map for a browser game?

    - by jj_
    I've made this little rpg ruby game I did while learning and now I'd like to make it into a browser game. I've already set up Sinatra framework to serve it, so what I am looking for, before everything else, is a way to represent the game map in browser (location attributes are stored in db). A new map is randomly generated by code for each new game at each game start. For now forget db, and let's say a map (say 100x100 "squares") is stored as a tridimensional array. (x,y, ...) Last "dimension" of array stores who & what is at that map cell: a player, a building, whatever. So all I have to do is render those "squares" or array cells to a 2d tiled map in the browser. The map does not need to refresh or be dynamically fetched as you scroll it, (at least at this stage of development) but, a technology which would allow me to do so in future would be a good reason for choosing it. Things that I thought of: html tables, html5 canvas, some js framework which is designed exactly with this purpose (which I do not know of = please advice). Yes I know about gamequery-js framework, but I've never used it, and I don't know if it's going to slow down everything down to inusability as I'm adding new features (scrolling, ajax). I really don't know of any other alternatives.. maybe there are lighter approaches? Easier or more minimalistic ways ? More targeted js framework which is the right tool for the job? Maybe just some html canvas code, or even simple image maps, or images with absolute positioning will be enough? The thing is I'd like to start simple, and then gradually make it better, so, as I said before, I'd prefer something that will give me room for improvement or is headed toward new web tendencies but which will also give me a bit of gratification in the beginning :) So.. advices are needed! And appreciated! :) Thanks p.s. Flash is excluded because I don't like it.

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  • Split UInt32 (audio frame) into two SInt16s (left and right)?

    - by morgancodes
    Total noob to anything lower-level than Java, diving into iPhone audio, and realing from all of the casting/pointers/raw memory access. I'm working with some example code wich reads a WAV file from disc and returns stereo samples as single UInt32 values. If I understand correctly, this is just a convenient way to return the 32 bits of memory required to create two 16 bit samples. Eventually this data gets written to a buffer, and an audio unit picks it up down the road. Even though the data is written in UInt32-sized chunks, it eventually is interpreted as pairs of 16-bit samples. What I need help with is splitting these UInt32 frames into left and right samples. I'm assuming I'll want to convert each UInt32 into an SInt16, since an audio sample is a signed value. It seems to me that for efficiency's sake, I ought to be able to simply point to the same blocks in memory, and avoid any copying. So, in pseudo-code, it would be something like this: UInt32 myStereoFrame = getFramefromFilePlayer; SInt16* leftChannel = getFirst16Bits(myStereoFrame); SInt16* rightChannel = getSecond16Bits(myStereoFrame); Can anyone help me turn my pseudo into real code?

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  • How to future-proof my touch-enabled web application?

    - by Rice Flour Cookies
    I recently went out and purchased a touch-screen monitor with the intention of learning how to program touch-enabled web applications. I had reviewed the MDN documentation about touch events, as well as the W3C specification. To get started, I wrote a very short test page with two event handlers: one for the mousedown event and one for the touchstart event. I fired up the web page in IE and touched the document and found that only the mousedown event fired. I saw the same behavior with Firefox, only to find out later that Firefox can be set to enable the touchstart event using about:config. When touch events are enabled, the touchstart event fires, but not mousedown. Chrome was even stranger: it fired both events when I touched the document: touchstart and mousedown, in that order. Only on my Android phone does it appear to be the case that only the touchstart event fires when I touch the document. I did a a Google search and ended up on two interesting pages. First, I found the page on CanIUse for touch events: http://caniuse.com/#feat=touch Can I Use clearly indicates that IE does not support touch events as of this writing, and Firefox only supports touch events if they are manually enabled. Furthermore, all four browsers I mentioned treat the touch in a completely different way. It boils down to this: IE: simulated mouse click Firefox with touch disabled: simulated mouse click Firefox with touch enabled: touch event Chrome: touch event and simulated mouse click Android: touch event What is more frustrating is that Google also found a Microsoft page called RethinkIE. RethinkIE brags about touch support in IE; as a matter of fact, one of their slogans is "Touch the Web". It links to a number of touch-based application. I followed some of these links, and as best I can tell, it's just like CanIUse described; no proper touch support; just simulated mouse clicks. The MDN (https://developer.mozilla.org/en-US/docs/Web/API/Touch) and W3C (http://www.w3.org/TR/touch-events/) documentation describe a far richer interface; an interface that doesn't just simulate mouse clicks, but keeps track of multiple touches at once, the contact area, rotation, and force of each touch, and unique identifiers for each touch so that they can be tracked individually. I don't see how simulated mouse clicks can ever touch the above described functionality, which, once again, is part of the W3C specification, although it is listed as "non-normative", meaning that a browser can claim to be standards-compliant without implementing it. (Why bother making it part of the standard, then?) What motivated my research is that I've written an HTML5 application that doesn't work on Android because Android doesn't fire mouse events. I'm now afraid to try to implement touch for my application because the browsers all behave so differently. I imagine that at some time in the future, the browsers might start handling touch similarly, but how can I tell how they might be handled in the future short of writing code to handle the behavior of each individual browser? Is it possible to write code today that will work with touch-enabled browsers for years to come? If so, how?

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

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

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  • Prefilling an SMS on Mobile Devices with the sms: Uri Scheme

    - by Rick Strahl
    Popping up the native SMS app from a mobile HTML Web page is a nice feature that allows you to pre-fill info into a text for sending by a user of your mobile site. The syntax is a bit tricky due to some device inconsistencies (and quite a bit of wrong/incomplete info on the Web), but it's definitely something that's fairly easy to do.In one of my Mobile HTML Web apps I need to share a current location via SMS. While browsing around a page I want to select a geo location, then share it by texting it to somebody. Basically I want to pre-fill an SMS message with some text, but no name or number, which instead will be filled in by the user.What worksThe syntax that seems to work fairly consistently except for iOS is this:sms:phonenumber?body=messageFor iOS instead of the ? use a ';' (because Apple is always right, standards be damned, right?):sms:phonenumber;body=messageand that works to pop up a new SMS message on the mobile device. I've only marginally tested this with a few devices: an iPhone running iOS 6, an iPad running iOS 7, Windows Phone 8 and a Nexus S in the Android Emulator. All four devices managed to pop up the SMS with the data prefilled.You can use this in a link:<a href="sms:1-111-1111;body=I made it!">Send location via SMS</a>or you can set it on the window.location in JavaScript:window.location ="sms:1-111-1111;body=" + encodeURIComponent("I made it!");to make the window pop up directly from code. Notice that the content should be URL encoded - HTML links automatically encode, but when you assign the URL directly in code the text value should be encoded.Body onlyI suspect in most applications you won't know who to text, so you only want to fill the text body, not the number. That works as you'd expect by just leaving out the number - here's what the URLs look like in that case:sms:?body=messageFor iOS same thing except with the ;sms:;body=messageHere's an example of the code I use to set up the SMS:var ua = navigator.userAgent.toLowerCase(); var url; if (ua.indexOf("iphone") > -1 || ua.indexOf("ipad") > -1) url = "sms:;body=" + encodeURIComponent("I'm at " + mapUrl + " @ " + pos.Address); else url = "sms:?body=" + encodeURIComponent("I'm at " + mapUrl + " @ " + pos.Address); location.href = url;and that also works for all the devices mentioned above.It's pretty cool that URL schemes exist to access device functionality and the SMS one will come in pretty handy for a number of things. Now if only all of the URI schemes were a bit more consistent (damn you Apple!) across devices...© Rick Strahl, West Wind Technologies, 2005-2013Posted in IOS  JavaScript  HTML5   Tweet !function(d,s,id){var js,fjs=d.getElementsByTagName(s)[0];if(!d.getElementById(id)){js=d.createElement(s);js.id=id;js.src="//platform.twitter.com/widgets.js";fjs.parentNode.insertBefore(js,fjs);}}(document,"script","twitter-wjs"); (function() { var po = document.createElement('script'); po.type = 'text/javascript'; po.async = true; po.src = 'https://apis.google.com/js/plusone.js'; var s = document.getElementsByTagName('script')[0]; s.parentNode.insertBefore(po, s); })();

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  • bluetooth headset can connect, but not visible in pulse audio

    - by Kim Marivoet
    I have a plantronics bluetooth headset, and until yesterday I could use it without any problem. However, today it suddenly stopped working (maybe related to the last software update I did). I can still connect/disconnect my headset, but it doesn't show up in pulse audio anymore. I read through various posts that describes kind of the same problem, but none of the suggested solutions worked. I get following error in the syslog: Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/HFPAG Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSource Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSink Oct 13 16:50:09 desktop kernel: [ 17.340943] input: 48:C1:AC:08:FE:8F as /devices/virtual/input/input14 Oct 13 16:50:09 desktop bluetoothd[1040]: /org/bluez/1040/hci0/dev_48_C1_AC_08_FE_8F/fd0: fd(36) ready Oct 13 16:50:09 desktop rtkit-daemon[1894]: Successfully made thread 2213 of process 1892 (n/a) owned by '1000' RT at priority 5. Oct 13 16:50:09 desktop rtkit-daemon[1894]: Supervising 5 threads of 1 processes of 1 users. Oct 13 16:50:10 desktop bluetoothd[1040]: Badly formated or unrecognized command: AT+XEVENT=USER-AGENT,COM.PLANTRONICS,PLT_VOYAGERPRO,0109,27.90,FFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFF Oct 13 16:50:10 desktop bluetoothd[1040]: Audio connection got disconnected Any help would be much appreciated. I'm using Ubuntu 12.04. Thanks, Kim

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  • Setting up Beats audio on HP Pavilion m6

    - by Joel Auterson
    I have an HP Pavilion m6-1054sa laptop, with a Beats subwoofer on the bottom. The normal laptop speakers work fine under Ubuntu but the Beats speaker(s?) does not. Anyone know how to get this working? Here's my lspci output, if it helps... 00:00.0 Host bridge: Intel Corporation Ivy Bridge DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Ivy Bridge PCI Express Root Port (rev 09) 00:02.0 VGA compatible controller: Intel Corporation Ivy Bridge Graphics Controller (rev 09) 00:14.0 USB controller: Intel Corporation Panther Point USB xHCI Host Controller (rev 04) 00:16.0 Communication controller: Intel Corporation Panther Point MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 1 (rev c4) 00:1c.1 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 2 (rev c4) 00:1d.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Panther Point LPC Controller (rev 04) 00:1f.2 RAID bus controller: Intel Corporation 82801 Mobile SATA Controller [RAID mode] (rev 04) 00:1f.3 SMBus: Intel Corporation Panther Point SMBus Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Thames XT/GL [Radeon HD 7600M Series] (rev ff) 07:00.0 Unassigned class [ff00]: Realtek Semiconductor Co., Ltd. Device 5289 (rev 01) 07:00.2 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 0a) 08:00.0 Network controller: Intel Corporation Centrino Wireless-N 2230 (rev c4)

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  • Problem Installing Xubuntu 12.04 Audio-Video codecs

    - by Seib
    I used Crouton to install Xubuntu 12.04.4 LTS with the XFCE environment on my HP Chromebook 11. I've gotten it all fully installed and everything; the only thing I'm doing now is the basic setup things, like adding codecs and other things like LibreOffice, VLC, Firefox, Ubuntu Software Center, etc. This information I got from 2 sources: http://www.efytimes.com/e1/fullnews.asp?edid=137269 http://www.binarytides.com/better-xubuntu-14-04/ . I'm currently on the same step at both URLs, which is #6 on Link 1 and #8 on Link 2. Per the articles, which both said the same thing, I typed in sudo apt-get install xubuntu-restricted-extras libavcodec-extra and it didn't do anything. It just kept on saying the same thing: Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package libavcodec-extra I've spend the last hour or so scouring the internet for a solution, for a hint even at what is going on. I don't want to do a clean reinstall, for two reasons: 1) it's takes like 1.5h just to get the croot fully installed, and 2) everything but this out of what I've done so far (up to #6 at Link 1 & up to #8 at Link 2) works except the audio. I've already installed flash, so YouTube works fine. It's just I can't hear any audio. Please help? Thanks in advance. I appreciate all the great help I've been getting from AskUbuntu lately. You all are great.

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  • M-Audio Delta starts up at wrong sample rate

    - by steevc
    When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

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  • ffmpeg add two audio streams to video

    - by Tossin Hausen
    I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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  • 5.1 sound in Unity3d 3.5

    - by N0xus
    I'm trying to implement 5.1 surround sound in my game. I've set Unity's AudioManager to a default of 5.1 surround and loaded in a 6 channel audio clip that should play a sound in each of the different audio spots. However, when I go to run my game, all I get is flat sound coming out of my front two speakers. Even then, these don't play the sound they should (front speaker should play "front speaker" right should play "right speaker" and so). Both speakers just end up playing the entire sound file. I've tried looking to see if there is a parameter that I have missed, but information on how to set up 5.1 sound in Unity is lacking (or my google skills aren't that good) and I can't get it to work as intended. Could someone please either tell me what I'm missing, or point me in the right direction? My audio source is situated at point (0, 0, 0) with my camera also being in the same point. I've moved about the scene but the same thing happens as I've already described.

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  • Alsa devices under Wine

    - by Roberto Aloi
    Hi all, I'm running OpenSuse 11.2 and Wine 1.1.28. Even if audio is perfectly working fine for me (Skype, Banshee, etc), when I try to configure audio for Wine (to use Spotify) I cannot hear anything from the audio test. In the winecfg audio tab, ALSA is checked, but no devices are available. I tried to run alsaconf (it needs root permissions) but it returns: No supported PnP or PCI card found No legacy drivers available, either. Any idea?

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