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  • Swf files not loaed on IE

    - by Rajeev
    The swf files are not loaded on IE.IS there any settings that needs to be changed on IE <div> <table style="table-layout:fixed;width:100%;"> <tr> <td width="20%"> <object width="100" height="100" id="microphone"> <embed src="/media/players/game.swf" width="250" height="250" type='application/x-shockwave-flash'> </embed> </object> </td> </tr> </table> </div>

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  • What algorithm would you use to code a parrot?

    - by Phil H
    A parrot learns the most commonly uttered words and phrases in its vicinity so it can repeat them at inappropriate moments. So how would you create a software version? Assuming it has access to a microphone and can record sound at will, how would you code it without requiring infinite resources? The best I can imagine is to divide the stream using silences in the sound, and then use some pattern recognition to encode each one as a list of tokens, storing new ones as you meet them. Hashing the token sequences and counting occurrences in a database, you could build up a picture of the most frequently uttered phrases. But given the huge variety in phrases, how do you prevent this just becoming a huge list? And the sheer number of pairs to match would surely generate lot of false positives from the combinatorial nature of matching. Would you use a neural net, since that's how a real parrot manages it? Or is there another, cleverer way of matching large-scale patterns in analogue data?

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  • Best practice for C++ audio capture API under Linux?

    - by braddock
    I need to create a C++ application with a simple audio recording from microphone functionality. I can't say that there aren't enough audio APIs to do this! Pulse, ALSA, /dev/dsp, OpenAL, etc. My question is what is the current "Best practice" API? Pulse seems supported by most modern distros, but seems almost devoid of documentation. Will OpenAL be supported across different distros, or is it too obscure? Have I missed any? Is there not a simple answer? thanks!

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  • absolute audio synchronization

    - by user1780526
    I would like to synchronize my computer with an external camcorder recording so that I can know exactly (to the millisecond) when certain recored events happen with respect to other sensors logged by the computer. One idea is to playback short sound pulses or chirps every second from the computer that get picked up by the microphone on the camcorder. But the accuracy of a simple cron job playing a sound clip is not precise enough. I was thinking of using something like gstreamer, but how does one get it to playback a clip at precisely a certain time according to the system clock?

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  • Read data from an Android USB attachment

    - by Mark
    Is there anyway to read data from an attachment through the USB port on an Android device? In particular, an EKG. Most the work can be done by the hardware of the device to simplify the output to a single number, a voltage reading. If its not possible, what about modifying an accessory that can already communicate with an android device? Thinking of devices that attach to android phones, what about sending the data as an audio signal to be read as the microphone from a headset and then analyzing the audio signal to convert it to a number that can be used to display a value. Any ideas on how to make this work?

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  • Capturing Mac OS X System Audio output with Python

    - by richbs
    Hello, I've been trying to "hijack" the Mac OS X system audio using PyAudio and save to a wav in python. That is, I do not want to record from an input device such as a microphone. I want to grab the sound output from any or all applications. I have followed the tutorials on the PyAudio site but these do not appear to cover my use case and when I try to read from the output stream I unsurprisingly get the paCanNotReadFromAnOutputOnlyStream exception. Fair enough! Is there a way to do what I am proposing with the PyAudio or other FOSS Python Library?

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  • Going for Gold

    - by Simple-Talk Editorial Team
    There was a spring in the step of some members of our development teams here at Red Gate, on hearing that on five gold awards at 2012′s SQL Mag Community and Editors Choice Awards. And why not? It’s a nice recognition that their efforts were appreciated by many in the SQL Server community. The team at Simple-Talk don’t tend to spring, but even we felt a twinge of pride in the fact that SQL Scripts Manager received Gold for Editor’s Choice in the Best Free Tools category. The tool began life as a “Down Tools” project and is one that we’ve supported and championed in various articles on Simple-talk.com. Over a Cambridge Bitter in the Waggon and Horses, we’ve often reflected on how nice it would be to nominate our own awards. Of course, we’d have to avoid nominating Red Gate tools in each category, even the free ones, for fear of seeming biased,  but we could still award other people’s free tools, couldn’t we? So allow us to set the stage for the annual Simple-Talk Community Tool awards… Onto the platform we shuffle, to applause from the audience; Chris in immaculate tuxedo, Alice in stunning evening gown, Dave and Tony looking vaguely uncomfortable, Andrew somehow distracted, as if his mind is elsewhere. Tony strides up to the lectern, and coughs lightly…”In the free-tool category we have the three nominations, and they are…” (rustle of the envelope opening) Ola Hallengren’s SQL Server Maintenance Solution (applause) Adam Machanic’s WhoIsActive (cheers, more applause) Brent Ozar’s sp_Blitz (much clapping) “Before we declare the winner, I’d like to say a few words in recognition of a grand tradition in a SQL Server community that continues to offer its members a steady supply of excellent, free tools. It hammers home the fundamental principle that a tool should solve a single, pressing and frustrating problem, but you should only ever build your own solution to that problem if you are certain that you cannot buy it, or that someone has not already provided it free. We have only three finalists tonight, but I feel compelled to mention a few other tools that we also use and appreciate, such as Microsoft’s Logparser, Open source Curl, Microsoft’s TableDiff.exe, Performance Analysis of Logs (PAL) Tool, SQL Server Cache Manager and SQLPSX.” “And now I’ll hand over to Alice to announce the winner.” Alice strides over to the microphone, tearing open the envelope. “The winner,” she pauses for dramatic effect “… is …Ola Hallengren’s SQL Server Maintenance Solution!” Queue much applause and consumption of champagne. Did we get it wrong? What free tool would you nominate? Let us know! Cheers, Simple-Talk Editorial Team (Andrew, Alice, Chris, Dave, Tony)

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  • ALSA samples capture: cannot open device

    - by Randagio
    I'm quite new to Linux (Lubuntu 12.04 for sake of precision) and ALSA programming at all. I'm trying to write a C program to capture audio from internal PC microphone for processing it. So as first step I google a bit and I found this article for capturing audio samples A tutorial on using the ALSA Audio API but when I compile it and execute it with: ./capture "default" or ./capture "hw:0,0" and all the possible variants on theme it always raises the error: cannot open device hw:0,0 (no such file or directory). So the issue is: what is the name of the mic audio device to pass as parameter to record the audio from mic ? The mic is working ok because the Sound Recorder program records sounds perfectly and I can playback them. The output of the aplay -l is the following : **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 and this is the amixer output (cut) Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 4 [13%] [-40.50dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [-1.50dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] so for aplay it seems I have no recording device, but for amixer I've got the mic, a mic boost and mic stereo as well with all those gorgeous stuffs on their place !!. If so, how could my Sound Recorder record the audio without any problem at all ?!?! For sure I'm giving the wrong device name to the command line for capturing audio but I'm loosing the hope for finding the correct one ! Please help....before I tear my hair out !!!

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  • mongoDB Management Studio

    - by Liam McLennan
    This weekend I have been in Sydney at the MS Web Camp, learning about web application development. At the end of the first day we came up with application ideas and pitched them. My idea was to build a web management application for mongoDB. mongoDB I pitched my idea, put down the microphone, and then someone asked, “what’s mongo?”. Good question. MongoDB is a document database that stores JSON style documents. This is a JSON document for a tweet from twitter: db.tweets.find()[0] { "_id" : ObjectId("4bfe4946cfbfb01420000001"), "created_at" : "Thu, 27 May 2010 10:25:46 +0000", "profile_image_url" : "http://a3.twimg.com/profile_images/600304197/Snapshot_2009-07-26_13-12-43_normal.jpg", "from_user" : "drearyclocks", "text" : "Does anyone know who has better coverage, Optus or Vodafone? Telstra is still too expensive.", "to_user_id" : null, "metadata" : { "result_type" : "recent" }, "id" : { "floatApprox" : 14825648892 }, "geo" : null, "from_user_id" : 6825770, "search_term" : "telstra", "iso_language_code" : "en", "source" : "&lt;a href=&quot;http://www.tweetdeck.com&quot; rel=&quot;nofollow&quot;&gt;TweetDeck&lt;/a&gt;" } A mongodb server can have many databases, each database has many collections (instead of tables) and a collection has many documents (instead of rows). Development Day 2 of the Sydney MS Web Camp was allocated to building our applications. First thing in the morning I identified the stories that I wanted to implement: Scenario: View databases Scenario: View Collections in a database Scenario: View Documents in a Collection Scenario: Delete a Collection Scenario: Delete a Database Scenario: Delete Documents Over the course of the day the team (3.5 developers) implemented all of the planned stories (except ‘delete a database’) and also implemented the following: Scenario: Create Database Scenario: Create Collection Lessons Learned I’m new to MongoDB and in the past I have only accessed it from Ruby (for my hare-brained scheme). When it came to implementing our MongoDB management studio we discovered that their is no official MongoDB driver for .NET. We chose to use NoRM, honestly just because it was the only one I had heard of. NoRM was a challenge. I think it is a fine library but it is focused on mapping strongly typed objects to MongoDB. For our application we had no prior knowledge of the types that would be in the MongoDB database so NoRM was probably a poor choice. Here are some screens (click to enlarge):

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  • KVM recommendations

    - by alex
    I recently tried out a cheaper KVM solution, a dual monitor one with a USB hub and audio/mic support. It seemed the perfect KVM. However, these are my experiences. When using my keyboard via the PS/2 connection, none of my multimedia keys work (useful, because it has a volume control and my speakers do not, only on a wireless remote control.) I plugged the keyboard in via the USB port - and it seemed to work. However, I believe to switch the hub from PC to Mac, you need to use a keyboard combo, which is only supported when the keyboard is plugged in via PS/2 Sometimes the middle mouse button doesn't work when connected via PS/2. The multi monitor is VGA - I just found out by the looks of things my Mac Mini outputs DVI Digital only (though this is my fault!). Mac works with 2 screens, but switching and switching back can cause it not to display on the 2nd screen unless I go detect displays again. My question is - is there a KVM out there that supports these features? USB keyboard and mouse inputs will full multimedia keys support Dual monitor DVI connections Hotkey to change PCs and physical button USB hub Emulate screens attached when switching to a different computer Works with PC and Mac Audio / Microphone support Does one exist, that won't cost the world? So far the only one that seems to support all this (that I can tell) is this one. UPDATE I ended up buying the Aten CS1642. It's expensive, but it seems to work great!

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  • No audio with streaming video

    - by Chris Barnhill
    I am having trouble with audio when playing streaming videos. My sound card is fine. I know this because if I play sounds from my local machine, there's no problem. It's only when I try to play sounds from the internet that I lose audio. This only started happening recently when I did 2 things: I connected a USB headphone/microphone set to record screencasts I began recording/publishing screencasts from screenr.com. I have tried playing video both with the headset connected and without it connected: it makes no difference. If I record a screencast on screenr.com and preview it, I hear the audio. But once I publish is and play it, there is no audio. I also hear no audio with YouTube videos. I really hope someone can help. Thanks. The latest is that the problem went away after I powered my system off and on. A reboot didn't do it, I had to actually shut down the power.

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  • This Week in Geek History: Gmail Goes Public, Deep Blue Wins at Chess, and the Birth of Thomas Edison

    - by Jason Fitzpatrick
    Every week we bring you a snapshot of the week in Geek History. This week we’re taking a peek at the public release of Gmail, the first time a computer won against a chess champion, and the birth of prolific inventor Thomas Edison. Gmail Goes Public It’s hard to believe that Gmail has only been around for seven years and that for the first three years of its life it was invite only. In 2007 Gmail dropped the invite only requirement (although they would hold onto the “beta” tag for another two years) and opened its doors for anyone to grab a username @gmail. For what seemed like an entire epoch in internet history Gmail had the slickest web-based email around with constant innovations and features rolling out from Gmail Labs. Only in the last year or so have major overhauls at competitors like Hotmail and Yahoo! Mail brought other services up to speed. Can’t stand reading a Week in Geek History entry without a random fact? Here you go: gmail.com was originally owned by the Garfield franchise and ran a service that delivered Garfield comics to your email inbox. No, we’re not kidding. Deep Blue Proves Itself a Chess Master Deep Blue was a super computer constructed by IBM with the sole purpose of winning chess matches. In 2011 with the all seeing eye of Google and the amazing computational abilities of engines like Wolfram Alpha we simply take powerful computers immersed in our daily lives for granted. The 1996 match against reigning world chest champion Garry Kasparov where in Deep Blue held its own, but ultimately lost, in a  4-2 match shook a lot of people up. What did it mean if something that was considered such an elegant and quintessentially human endeavor such as chess was so easy for a machine? A series of upgrades helped Deep Blue outright win a match against Kasparov in 1997 (seen in the photo above). After the win Deep Blue was retired and disassembled. Parts of Deep Blue are housed in the National Museum of History and the Computer History Museum. Birth of Thomas Edison Thomas Alva Edison was one of the most prolific inventors in history and holds an astounding 1,093 US Patents. He is responsible for outright inventing or greatly refining major innovations in the history of world culture including the phonograph, the movie camera, the carbon microphone used in nearly every telephone well into the 1980s, batteries for electric cars (a notion we’d take over a century to take seriously), voting machines, and of course his enormous contribution to electric distribution systems. Despite the role of scientist and inventor being largely unglamorous, Thomas Edison and his tumultuous relationship with fellow inventor Nikola Tesla have been fodder for everything from books, to comics, to movies, and video games. Other Notable Moments from This Week in Geek History Although we only shine the spotlight on three interesting facts a week in our Geek History column, that doesn’t mean we don’t have space to highlight a few more in passing. This week in Geek History: 1971 – Apollo 14 returns to Earth after third Lunar mission. 1974 – Birth of Robot Chicken creator Seth Green. 1986 – Death of Dune creator Frank Herbert. Goodnight Dune. 1997 – Simpsons becomes longest running animated show on television. Have an interesting bit of geek trivia to share? Shoot us an email to [email protected] with “history” in the subject line and we’ll be sure to add it to our list of trivia. Latest Features How-To Geek ETC Here’s a Super Simple Trick to Defeating Fake Anti-Virus Malware How to Change the Default Application for Android Tasks Stop Believing TV’s Lies: The Real Truth About "Enhancing" Images The How-To Geek Valentine’s Day Gift Guide Inspire Geek Love with These Hilarious Geek Valentines RGB? CMYK? Alpha? What Are Image Channels and What Do They Mean? Clean Up Google Calendar’s Interface in Chrome and Iron The Rise and Fall of Kramerica? [Seinfeld Video] GNOME Shell 3 Live CDs for OpenSUSE and Fedora Available for Testing Picplz Offers Special FX, Sharing, and Backup of Your Smartphone Pics BUILD! An Epic LEGO Stop Motion Film [VIDEO] The Lingering Glow of Sunset over a Winter Landscape Wallpaper

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  • 5 Ways to Determine Mobile Location

    - by David Dorf
    In my previous post, I mentioned the importance of determining the location of a consumer using their mobile phone.  Retailers can track anonymous mobile phones to determine traffic patterns both inside and outside their stores.  And with consumers' permission, retailers can send location-aware offers to mobile phones; for example, a coupon for cereal as you walk down that aisle.  When paying with Square, your location is matched with the transaction.  So there are lots of reasons for retailers to want to know the location of their customers.  But how is it done? I thought I'd dive a little deeper on that topic and consider the approaches to determining location. 1. Tower Triangulation By comparing the relative signal strength from multiple antenna towers, a general location of a phone can be roughly determined to an accuracy of 200-1000 meters.  The more towers involved, the more accurate the location. 2. GPS Using Global Positioning Satellites is more accurate than using cell towers, but it takes longer to find the satellites, it uses more battery, and it won't well indoors.  For geo-fencing applications, like those provided by Placecast and Digby, cell towers are often used to determine if the consumer is nearing a "fence" then switches to GPS to determine the actual crossing of the fence. 3. WiFi Triangulation WiFi triangulation is usually more accurate than using towers just because there are so many more WiFi access points (i.e. radios in routers) around. The position of each WiFi AP needs to be recorded in a database and used in the calculations, which is what Skyhook has been doing since 2008.  Another advantage to this method is that works well indoors, although it usually requires additional WiFi beacons to get the accuracy down to 5-10 meters.  Companies like ZuluTime, Aisle411, and PointInside have been perfecting this approach for retailers like Meijer, Walgreens, and HomeDepot. Keep in mind that a mobile phone doesn't have to connect to the WiFi network in order for it to be located.  The WiFi radio in the phone only needs to be on.  Even when not connected, WiFi radios talk to each other to prepare for a possible connection. 4. Hybrid Approaches Naturally the most accurate approach is to combine the approaches described above.  The more available data points, the greater the accuracy.  Companies like ShopKick like to add in acoustic triangulation using the phone's microphone, and NearBuy can use video analytics to increase accuracy. 5. Magnetic Fields The latest approach, and this one is really new, takes a page from the animal kingdom.  As you've probably learned from guys like Marlin Perkins, some animals use the Earth's magnetic fields to navigate.  By recording magnetic variations within a store, then matching those readings with ones from a consumer's phone, location can be accurately determined.  At least that's the approach IndoorAtlas is taking, and the science seems to bear out.  It works well indoors, and doesn't require retailers to purchase any additional hardware.  Keep an eye on this one.

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  • Integrated webcam in lenovo t410 not working with 12.04

    - by kristianp
    I have a Lenovo T410 with an inbuilt webcam and I haven't been able to get the webcam working. I tried skype, cheese, both just give me a black window. The microphone works fine with skype, by the way. Can anyone provide any clues please? The webcam is enabled in the bios, but there is no light indicating the webcam is on (not sure if there should be, though). I tried this on Kubuntu 11.10 and have upgraded to 12.04 with the same results. The Fn-F6 keyboard combination doens't seem to do anything either. EDIT: I got the webcam replaced, it looks like it was a hardware problem, because it works fine now. Thanks guys. $ ls /dev/v4l/* /dev/v4l/by-id: usb-Chicony_Electronics_Co.__Ltd._Integrated_Camera-video-index0 /dev/v4l/by-path: pci-0000:00:1a.0-usb-0:1.6:1.0-video-index0 And lsusb: $ lsusb Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 147e:2016 Upek Biometric Touchchip/Touchstrip Fingerprint Sensor Bus 001 Device 004: ID 0a5c:217f Broadcom Corp. Bluetooth Controller Bus 001 Device 005: ID 17ef:480f Lenovo Integrated Webcam [R5U877] Bus 002 Device 003: ID 05c6:9204 Qualcomm, Inc. Bus 002 Device 004: ID 17ef:1003 Lenovo Integrated Smart Card Reader Here is the output from guvcview, minus lots of lines describing the available capture formats. It says "unable to start with minimum setup. Please reconnect your camera.". guvcview 1.5.3 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started video device: /dev/video0 Init. Integrated Camera (location: usb-0000:00:1a.0-1.6) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, .... { discrete: width = 1600, height = 1200 } Time interval between frame: 1/15, vid:17ef pid:480f driver:uvcvideo checking format: 1196444237 libv4l2: error setting pixformat: Device or resource busy VIDIOC_S_FORMAT - Unable to set format: Device or resource busy Init v4L2 failed !! Init video returned -2 trying minimum setup ... video device: /dev/video0 Init. Integrated Camera (location: usb-0000:00:1a.0-1.6) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } .... vid:17ef pid:480f driver:uvcvideo checking format: 1448695129 libv4l2: error setting pixformat: Device or resource busy VIDIOC_S_FORMAT - Unable to set format: Device or resource busy Init v4L2 failed !! ERROR: Minimum Setup Failed. Exiting... VIDIOC_REQBUFS - Failed to delete buffers: Invalid argument (errno 22) cleaned allocations - 100% Closing portaudio ...OK Terminated.

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  • Problems with MediaRecorder class setting audio source - setAudioSource() - unsupported parameter

    - by arakn0
    Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaRecorder class to record just audio from the microphone. I'm following the steps indicated in the official site: http://developer.android.com/reference/android/media/MediaRecorder.html So I have a method that initializes and configure the MediaRecorder object in order to start recording. Here you have the code: this.mr = new MediaRecorder(); this.mr.setAudioSource(MediaRecorder.AudioSource.MIC); this.mr.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); this.mr.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); this.mr.setOutputFile(this.path + this.fileName); try { this.mr.prepare(); } catch (IllegalStateException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } catch (IOException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } When I execute this code in the simulator, thanks to logcat, I can see that the method setAudioSource(MediaRecorder.AudioSource.MIC) gives the next error (with the tag audio_ipunt) when it is called: ERROR/audio_input(34): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value ERROR/audio_input(34): VerifyAndSetParameter failed And then when the method prepare() is called, I get the another error again: ERROR/PVOMXEncNode(34): PVMFOMXEncNode-Audio_AMRNB::DoPrepare(): Got Component OMX.PV.amrencnb handle If I start to record bycalling the method start()... I get lots of messages saying: AudioFlinger(34):RecordThread: buffer overflow Then...after stop and release,.... I can see that a file has been created, but it doesn't seem that it been well recorderd. Anway, if i try this in a real device I can record with no problems, but I CAN'T play what I just recorded. I gues that the key is in these errors that I've mentioned before. How can I fix them? Any suggestion or help?? Thanks in advanced!!

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  • Objective C - displaying data in NSTextView

    - by Leo
    Hi, I'm having difficulties displaying data in a TextView in iPhone programming. I'm analyzing incoming audio data (from the microphone). In order to do that, I create an object "analyzer" from my SignalAnalyzer class which performs analysis of the incoming data. What I would like to do is to display each new incoming data in a TextView in realtime. So when I push a button, I create the object "analyzer" whiwh analyze the incoming data. Each time there is new data, I need to display it on the screen in a TextView. My problem is that I'm getting an error because (I think) I'm trying to send a message to the parent class (the one taking care of displaying stuff in my TextView : it has a TexView instance variable linked in Interface Builder). What should I do in order to tell my parent class what it needs to display ? Or how sohould I design my classes to display automaticlally something ? Thank you for your help. PS : Here is my error : 2010-04-19 14:59:39.360 MyApp[1421:5003] void WebThreadLockFromAnyThread(), 0x14a890: Obtaining the web lock from a thread other than the main thread or the web thread. UIKit should not be called from a secondary thread. 2010-04-19 14:59:39.369 MyApp[1421:5003] bool _WebTryThreadLock(bool), 0x14a890: Tried to obtain the web lock from a thread other than the main thread or the web thread. This may be a result of calling to UIKit from a secondary thread. Crashing now... Program received signal: “EXC_BAD_ACCESS”.

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  • iOS 5.0 AVAudioPlayer Error loading audio clip: The operation couldn’t be completed. (OSStatus error -50.)

    - by Jason Catudal
    So I'm trying to test out the audio player on the iPhone, and I went off Troy Brant's iOS book. I have the Core Audio, Core Foundation, AudioToolbox, and AVFoundation frameworks added to my project. The error message I get is in the subject field. I read like 20 pages of Google search results before resorting to asking here! /sigh. Thanks if you can help. Here's my code, pretty much verbatim out of his book: NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:@"Yonah" ofType:@"caf"]; NSLog(@"%@", soundFilePath); NSURL *fileURL = [NSURL URLWithString:soundFilePath]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:fileURL error:&error]; if (!error) { audioPlayer.delegate = self; //audioPlayer.numberOfLoops = -1; [audioPlayer play]; } else { NSLog(@"Error loading audio clip: %@", [error localizedDescription]); } EDIT: Holy Shinto. I figured out what it was. I changed NSURL *fileURL = [NSURL URLWithString:soundFilePath]; to NSURL *fileURL = [NSURL fileURLWithPath:soundFilePath]; to the latter and I was getting a weird error, weirder than the one in the subject BUT I googled that and I changed my OS input device from my webcam to my internal microphone and guess what, it worked under the fileURLWithPath method. I'll be. Damned.

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  • General question about DirectShow.NET, DirectShow and Windows Media Format

    - by Paul Andrews
    I searched and googled for an answer but couldn't find one. Basically I'm developing a webcam/audio streaming application which should capture audio and video from a pc (usb webcam/microphone) and send them to a receiving server. What the server will do with that it's another story and phase two (which I'm skipping for now) I wrote some code using DirectShow and Windows Media Format and it worked great for capture audio/video and sending them to another client, but there's a major problem: latency. Everywhere in the internet everyone gave me the same answer: "sorry dude but media format isn't for video conferencing, their codecs have too high latency". I thought I could skip the .wmv problems but seems like it's not possible to do... this road ends here then. So I saw a few examples with DirectShow.NET which were faster for both audio and video.. my question is: how come that DirectShow.NET is faster and better for video/audio conferencing? Shouldn't it be just a .NET porting of C++'s DirectShow? Am I missing something? I'm a bit confused at this point

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  • video calling (center)

    - by rrejc
    We are starting to develop a new application and I'm searching for information/tips/guides on application architecture. Application should: read the data from an external (USB) device send the data to the remote server (through internet) receive the data from the remote server perform a video call with to the calling (support) center receive a video call call from the calling (support) center support touch screens In addition: some of the data should also be visible through the web page. So I was thinking about: On the server side: use the database (probably MS SQL) use ORM (nHibernate) to map the data from the DB to the domain objects create a layer with business logic in C# create a web (WCF) services (for client application) create an asp.net mvc application (for item 7.) to enable data view through the browser On the client side I would use WPF 4 application which will communicate with external device and the wcf services on the server. So far so good. Now the problem begins. I have no idea how to create a video call (outgoing or incoming) part of the application. I believe that there is no problem to communicate with microphone, speaker, camera with WPF/C#. But how to communicate with the call center? What protocol and encoding should be used? I think that I will need to create some kind of server which will: have a list of operators in the calling center and track which operator is occupied and which operator is free have a list of connected end users receive incoming calls from end users and delegate call to free operator delegate calls from calling center to the end user Any info, link, anything on where to start would be much appreciated. Many thanks!

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  • Trimming bit of the beginning off a recorder waveform

    - by Lowgain
    I've got a flash 10.1 app that lets me record microphone input to a wav without a media server, which I am saving to an Amazon S3 bucket. I have another process running on a server which gets wavs from this bucket, converts to mp3 using LAME and puts them into another bucket. This all works fine, but in converting wav mp3, about 0.1sec or so of silence is added to my sound. In the application this are being used in, perfect sync is critical, so I need to trim off that little bit. If I have to trim it off the original waveform that is okay, I don't expect anything important to happen in that first fraction of a second. What is the best way to go about this? I am using Adobe's WavWriter to convert by ByteArray into a proper waveform. Is there a way I can easily trim off the first few samples from my ByteArray without invalidating the structure? Alternatively, is there a good server-side tool I can use to trim the wav before running it through LAME, or an argument I can give LAME? Or, could I even trim that sound off the mp3 after it has been converted? Thanks!

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  • How can I make an even more random number in ActionScript 2.0

    - by Theo
    I write a piece of software that runs inside banner ads which generates millions of session IDs every day. For a long time I've known that the random number generator in Flash is't random enough to generate sufficiently unique IDs, so I've employed a number of tricks to get even more random numbers. However, in ActionScript 2.0 it's not easy, and I'm seeing more and more collisions, so I wonder if there is something I've overlooked. As far as I can tell the problem with Math.random() is that it's seeded by the system time, and when you have sufficient numbers of simultaneous attempts you're bound to see collisions. In ActionScript 3.0 I use the System.totalMemory, but there's no equivalent in ActionScript 2.0. AS3 also has Font.enumerateFonts, and a few other things that are different from system to system. On the server side I also add the IP address to the session ID, but even that isn't enough (for example, many large companies use a single proxy server and that means that thousands of people all have the same IP -- and since they tend to look at the same sites, with the same ads, roughly at the same time, there are many session ID collisions). What I need isn't something perfectly random, just something that is random enough to dilute the randomness I get from Math.random(). Think of it this way: there is a certain chance that two people will generate the same random number sequence using only Math.random(), but the chance of two people generating the same sequence and having, say, the exact same list of fonts is significantly lower. I cannot rely on having sufficient script access to use ExternalInterface to get hold of things like the user agent, or the URL of the page. I don't need suggestions of how to do it in AS3, or any other system, only AS2 -- using only what's available in the standard APIs. The best I've come up with so far is to use the list of microphones (Microphone.names), but I've also tried to make some fingerprinting using some of the properties in System.capabilities, I'm not sure how much randomness I can get out of that though so I'm not using that at the moment. I hope I've overlooked something.

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  • Android Signal analysis + some filters.

    - by Profete162
    Hello, as the world cup is the main sport event and the Vuvuzelas are the most annoying sound in the world, I had an idea to remove them definitively by reading this new ( http://www.popsci.com/diy/article/2010-06/simple-software-can-filter-out-vuvuzela-whine) that told us that the sound has some frequencies at 233Hz + 466,932,1864Hz. I have already made a lot of Android application by myself but never touching the signal analysis and filtering part, so here are a few questions, I do not ask for precise answer but maybe links and tutorial to find something to work on. I guess that a new Android phone has the CPU and power to make real-time filtering. 1) How can I intercept the sound coming from the Jack microphone - Line-IN plug- ( I plan to link my TV to my phone with Jack to Jack plug). My question is totally software and coding, I have all the wires and adapters to plug a jack into my android phone Line IN. 2) Are there some Fourier analysis librairies, may I have a look to Java libraries on the web and import them to my Android project? I really apologize because my question seem not precise, but I think that would be something great. Thank you for your answers.

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  • iPhone rotation woes

    - by skooter500
    I have been spending many frustrating hours trying to get rotations working on the iPhone version of Tunepal. Firstly, I have a tab bar controller, with a navigation controller controlling each of the views. I actually only want one of my views to be able to rotate and that is the TuneDisplay. I have a subclassed the UITabBarController and overridden theshouldAutorotateToInterfaceOrientation: (BOOL)shouldAutorotateToInterfaceOrientation:(UIInterfaceOrientation) interfaceOrientation { if (self.selectedViewController != nil) { return [self.selectedViewController shouldAutorotateToInterfaceOrientation:interfaceOrientation]; } else { return (interfaceOrientation == UIInterfaceOrientationPortrait); } } In each of the view controllers for each of the tabs I have overridden the method and returned YES for each orientation I want to support. All well and good and everything works as it should. If I try and do a rotation on a tab that doesn’t support the rotation, nothing happens. The problem occurs if I move from a tab thats rotated to a tab that isnt supposed to support that rotation. The new tab is displayed rotated too! Screenshots for all this are included here: http://tunepal.wordpress.com/2010/04/20/rotation-woes/ Is there any way I can make it rotate back to portrait on tapping the tab? I have tried the unsupported setOrientation trick, but firstly it doesnt work correctly and secondly I received a warning from Apple for including it in my last build. If (as I suspect) there is no way to limit this behavior: How do I make the microphone image scale when I rotate the device? How do I make the buttons and the progress bar expand to fit the witdh of the toolbar? Also, one of the tabs that rotates ok has a table, with a search bar. The first time I rotate to the right or to the left, I get a black bar to the right of the search bar. If I subsequently rotate back and rotate again, the bar disappears! I have enabled the struts and springs things on the search bar in the interface builder and it looks like it should behave correctly. Any ideas about how to fix this? Ideas, feedback much appreciated Bryan

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  • CPU overheating after cleaning it

    - by Roberts
    I wanted to clean my computer CPU heatsink and fan itself, because the temperature is not what I wanted. About (50C ~ 70C). I have Intel Core 2 Duo E4300 @1.8 GHz (LGA775). The heatsink wasn't so scary filled with dust but I wanted to clean it anyway. I didn't know how to get heatsink with fan from the socket. So after 25 minutes I've figured it out. But I didn't know how to get it back on so I spent a lot time getting out the motherboard from the case. The fan and heatsink... The case and all components are clear of dust. (I'm tired now). Then I put all back just the way it was, well did few things on cable management. But the problem was that I didn't know how to connect front audio connectors. I had Windows XP hibernated. So I started the PC and everything was normal, except CMOS memory was clear. I configured the BIOS just the way it was and while I was doing that I saw about 58C CPU temperature and fan at 1789 RPM. Restarted the computer with new settings applied. But Windows halted with Blue Screen (I forgot what error it was but something with KERNEL). Restarted the PC and deleted hibernation session and everything was back normal. But couldn't record any sound from front panel microphone. The problem was that I messed ground wire with mic. Again after fixing it I turned computer on. No problems. The fan currently is noisy and temperature was 78C. The temperature before was 55C - 60C at idle. Now it's about 60C. If I do something then temperature raises to 79C. While speaking in skype the temperature was 82C. Could this problem occur because of the thermal grease (it's old and never replaced)? Edit The problem wasn't in thermal paste (because I didn't touch it). The problem was that I installed heatsink wrong. Now instead of regular 60C CPU temperature the CPU is at 48C (cool).

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  • Ubuntu stopped recognizing my iPod

    - by flashnode
    Rythmbox on Ubuntu 10.10 used to recognize my 3rd gen Nano and transfer mp3s. Now I plug it in and Ubuntu doesn't pop-up that box that asks what you want to do anymore. It is only recognized if I reboot and the thing is plugged in. Here is the output to 'lsusb -v -s bus:device' Bus 001 Device 008: ID 05ac:1262 Apple, Inc. iPod Nano 3.Gen Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x05ac Apple, Inc. idProduct 0x1262 iPod Nano 3.Gen bcdDevice 0.01 iManufacturer 1 Apple Inc. iProduct 2 iPod iSerial 3 000A27001A670128 bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 32 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 8 Mass Storage bInterfaceSubClass 6 SCSI bInterfaceProtocol 80 Bulk (Zip) iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 149 bNumInterfaces 3 bConfigurationValue 2 iConfiguration 4 iPod USB Interface bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 30 bInCollection 1 baInterfaceNr( 0) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 2 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 1 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 35 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 9 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 11025 tSamFreq[ 2] 12000 tSamFreq[ 3] 16000 tSamFreq[ 4] 22050 tSamFreq[ 5] 24000 tSamFreq[ 6] 32000 tSamFreq[ 7] 44100 tSamFreq[ 8] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 208 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 2 Device Status: 0x0000 (Bus Powered) This ubuntu forum told me to check the automount settings under /apps/nautilus/preferences/media_automount_open in gconf-editor. And I did that. Any clues?

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