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  • reCAPTCHA Ajax API + custom theme not working

    - by Felix
    I can't see where I'm going wrong. I've tried everything I could think of, reCAPTCHA is just not working with the Ajax API. Here's what my code looks like: <!-- this is in <head> --> <script type="text/javascript" src="http://code.jquery.com/jquery-1.4.2.min.js"></script> <script type="text/javascript" src="http://api.recaptcha.net/js/recaptcha_ajax.js"></script> <script type="text/javascript"> $(document).ready(function() { Recaptcha.create("my key here", "recaptcha_widget", { "theme": "custom", "lang": "en", "callback": function() { console.log("callback"); } // this doesn't get called }); }); </script> <!-- ... this is in <body> --> <div id="recaptcha_widget" style="display: none"> <div id="recaptcha_image"></div> <div id="recaptcha_links"> <a href="javascript:Recaptcha.reload()">get another</a> &bull; <a class="recaptcha_only_if_image" href="javascript:Recaptcha.switch_type('audio')">switch to audio</a> <a class="recaptcha_only_if_audio" href="javascript:Recaptcha.switch_type('image')">switch to image</a> &bull; <a href="javascript:Recaptcha.showhelp()">help</a> </div> <dt>Type the words</dt> <dd><input type="text" id="recaptcha_response_field" name="recaptcha_response_field"></dd> </div>

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  • AVAudioPlayer making noise when playing multiple sounds at the same time

    - by Rob
    I am having an issue where AVAudioPlayer is introducing noise into playback ONLY when I play multiple sound files at the same time. If I play them each individually, they all sound perfect. But, if I play sound clip B while sound clip A is still playing, the speakers start crackling like there is noise. I have tried both m4a files AND caf files and both make the same noise, so it has to be something with how I am implementing this method or a quirk with AVAudioPlayer. Any insights? code I am using: UITouch* touch = [[event allTouches] anyObject]; NSString* filename = [soundArray objectAtIndex:[touch view].tag]; NSString *path = [[NSBundle mainBundle] pathForResource:filename ofType:@"m4a"]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely theAudio.delegate = self; [theAudio prepareToPlay]; [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio play];

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  • Differences between iPhone/iPod Simulator and Devices

    - by Allisone
    Hi, since I started iPhone/iPod Development I have come across some differences between how the simulator and how real device react. Maybe I will come across some other differences I will have to figure out as well, maybe other people haven't met these problems here (YET) and can profit from the knowledge, and maybe you know some problems/differences that you would have been happy to know about earlier before you spent several hours or days figuring out what the heck is going on. So here is what I came across. Simulator is not case sensitive, Devices are case sensitive. This means a default.png or Icon.png will work in simulator, but not on a device where they must be named Default.png and icon.png (if it's still not working read this answer) Simulator has different codecs to play audio and video If you use f.e. MPMoviePlayerController you might play certain video on the simulator while on the device it won't work (use Handbrake-presets-iPhone & iPod Touch to create playable videos for Simulator and Device). If you play audio with AudioServicesPlaySystemSound(&soundID) you might here the sound on simulator but not an a device. (use Audacity to open your soundfile, export as wav and run afconvert -f caff -d LEI16@44100 -c 1 audacity.wav output.caf in terminal) Also there is this flickering on second run problem which can be resolved with an playerViewCtrl.initialPlaybackTime = -1.0; either on the end of playing or before each beginning. Simulator is mostly much faster cause it doesn't simulate the hardware but uses Mac resources, therefore f.e. sio2 Apps (OpenGL,OpenAL,etc. framework) run much better on simulator, well everything that uses more resources will run visibly better in simulator than on device. I hope we can add some more to this.

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  • What Qt container class to use for a sorted list?

    - by Dave
    Part of my application involves rendering audio waveforms. The user will be able to zoom in/out of the waveform. Starting at fully zoomed-out, I only want to sample the audio at the necessary internals to draw the waveform at the given resolution. Then, when they zoom in, asynchronously resample the "missing points" and provide a clearer waveform. (Think Google Maps.) I'm not sure the best data structure to use in Qt world. Ideally, I would like to store data samples sorted by time, but with the ability to fill-in points as needed. So, for example, the data points might initially look like: data[0 ms] = 10 data[10 ms] = 32 data[20 ms] = 21 ... But when they zoom in, I would get more points as necessary, perhaps: data[0 ms] = 10 data[2 ms] = 11 data[4 ms] = 18 data[6 ms] = 30 data[10 ms] = 32 data[20 ms] = 21 ... Note that the values in brackets are lookup values (milliseconds), not array indices. In .Net I might have used a SortedList<int, int>. What would be the best class to use in Qt? Or should I use a STL container?

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  • video streaming infrastructure

    - by alchemical
    We would like to set-up a live video-chat web site and are looking for basic architectural advice and/or a recomendation for a particular framework to use. Here are the basic features of the site: Most streams will be broadcast live from a single person with a web cam, etc., and viewed by typically 1-10 people, although there could be up to 100+ viewers on the high side. Audio and video do not have to be super-high quality, but do need to be "good enough". The main point is to convey the basic info in the video (and audio). If occasionally the frame-rate drops low and then goes back to normal fairly soon, we could live with that. Budget is an issue, so we are in general looking for a lower cost solution that will give us most of what we need in temers of performance and quality. We are looking at Peer1 for co-lo. The rest of our web site will be .Net / Windows platform. We are open to looking at any platform for the best streaming solution, although our technical expertise is currently more on the Windows side.

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  • AVAudioPlayer working in Simulator, but not on device

    - by cannyboy
    My mp3 playing code is: NSError *error; soundObject = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:audioPathString] error:&error]; if (soundObject == nil) NSLog(@"%@", [error description]); soundObject.delegate = self; soundObject.numberOfLoops = 0; soundObject.volume = 1.0; NSLog(@"about to play"); [soundObject prepareToPlay]; [soundObject play]; NSLog(@"[soundObject play];"); The mp3 used to play fine, and it still does on the simulator. But not on the device. I've recently added some sound recording code (not mine) to the software. It uses AudioQueue stuff which is slightly beyond me. Does that conflict with AVAudioPlayer? Or what could be the problem? I've noticed that as soon as the audiorecording code starts working, I can't adjust the volume on the device anymore, so maybe it blocks the audio playback?. EDIT The solution seems to be to put this in my code. I put it in applicationDidFinishLaunching: [[AVAudioSession sharedInstance] setCategory: AVAudioSessionCategoryPlayAndRecord error: nil]; UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker; AudioSessionSetProperty (kAudioSessionProperty_OverrideAudioRoute,sizeof (audioRouteOverride),&audioRouteOverride); The first line allows both play and record, whilst the other lines apparently reroute things to make the volume louder. All audio code is voodoo to me.

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  • html5 vs flash - full comparison chart anywhere?

    - by iddqd
    So since Steve Jobs said Flash sucks and implied that HTML5 can do everything Flash can without the need for a Plugin, I keep hearing those exact words from a lot of People. I would really like to have a Chart somewhere (similar to http://en.wikipedia.org/wiki/Comparison_of_layout_engines_%28HTML5%29#Form_elements_and_attributes ) that I can just show to those people. Showing all the little things that Flash can do right now, that HTML5/Ajax/CSS is not yet even thinking about. But of course also the things that HTML5 does better. I would like to see details compared like audio playback, realtime audio processing, byte level access, bitmap data manipulation, webcam access, binary sockets, stuff in the works such as P2P technology (adobe stratus) and all the stuff I don't know about myself. Ideally with examples of what can be accomplished with, lets say Binary Sockets (such as a POP3 client) because otherwise it won't mean a lot to non-programmers since they will just say "well we can do without Binary Sockets". And ideally with some current benchmarks and some examples of websites that use this technology. I've searched the web and am surprised not to find anything. So is there such a comparison somewhere? Or does anybody want to create this and post it to Wikipedia? ;-)

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  • Progressive MP4 video issues in Flash- Video stops rendering

    - by Conor
    I'm currently working on a flash project that has an intro video that plays before heading into the main app. This video is an H.264 .mp4, 1550x540, and around 10MB. The problem thats currently driving me insane is that when I test it, occasionally the video will begin playing, and then suddenly stop rendering the video frames, leaving the audio playing in the background with nothing on screen. Once the file is played through fully (based on listening to the audio), my playback complete event fires like it should, but I can't find any info of people having similar issues. Attached is a trace of the .mp4 metadata in case that helps. videoframerate : 24 audiochannels : 2 audiocodecid : mp4a audiosamplerate : 48000 trackinfo: 0: length : 608000 timescale : 24000 language : eng sampledescription: 0: sampletype : avc1 1: length : 1218560 timescale : 48000 language : eng sampledescription: 0: sampletype : mp4a duration : 25.386666666666667 width : 1540 videocodecid : avc1 seekpoints: 0: time : 0 offset : 13964 1: time : 0.333 offset : 16893 2: time : 0.667 offset : 34212 ... 73: time : 24.333 offset : 9770329 74: time : 24.667 offset : 9845709 75: time : 25 offset : 9895215 moovposition : 32 height : 540 avcprofile : 77 avclevel : 51 aacaot : 2 This has been driving me absolutely insane... any help would be much appreciated!

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  • iphone-AVAudio Player Crashes

    - by user2779450
    I have an app that uses an avaudio player for two things. One of them is to play an explosion sound when a uiimageview collision is detected, and the other is to play a lazer sound when a button is pressed. I declared the audioplayer in the .h class, and I call it each time the button is clicked by doing this: NSURL *url = [NSURL fileURLWithPath:[[NSBundle mainBundle] pathForResource:@"/lazer" ofType:@"mp3"]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; if (error) { NSLog(@"Error in audioPlayer: %@", [error localizedDescription]); } else { [audioPlayer prepareToPlay]; } [audioPlayer play]; This works fine, but after many uses of the game, the audio will stop play when i hit the button, and when a collision is detected, the game crashes. Here is my crash log: 2013-09-18 18:09:19.618 BattleShip[506:907] 18:09:19.617 shm_open failed: "AppleAudioQueue.41.2619" (23) flags=0x2 errno=24 (lldb) Suggestions? Could there be something to do with repeatedly creating an audio player? Alternatives maybe?

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  • Tomcat Application Generating too many logs

    - by rohitgu
    Hi, I have an application which runs on tomcat 6.0.20 server on linux ubuntu server. It generates a huge amount of logs in the catalina.out folder, most of these are generated while using the application, but are not generated by the application. Some of the logs it generates are given below, Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester startElement FINE: startElement(,,mime-type) Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester startElement FINE: Pushing body text ' ' Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester startElement FINE: New match='web-app/mime-mapping/mime-type' Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester startElement FINE: Fire begin() for CallParamRule[paramIndex=1, attributeName=null, from stack=false] Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester characters FINE: characters(audio/x-mpeg) Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester endElement FINE: endElement(,,mime-type) Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester endElement FINE: match='web-app/mime-mapping/mime-type' Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester endElement FINE: bodyText='audio/x-mpeg' Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester endElement FINE: Fire body() for CallParamRule[paramIndex=1, attributeName=null, from stack=false] Apr 16, 2010 2:55:24 PM org.apache.tomcat.util.digester.Digester endElement FINE: Popping body text ' How can I turn them off? This is very important, since this a production application. Regards, Rohit

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  • How to build Android for Samsung Galaxy Note

    - by Tr?n Ð?i
    I'd like to modify and build my own Android for my Samsung Galaxy Note I've downloaded Android 4.1.2 from http://source.android.com and Samsung open source for my Samsung Galaxy Note. After extract Samsung open source, I get 2 folders: Kernel and Platform, and 2 README text file README_Kernel.txt 1. How to Build - get Toolchain From android git server , codesourcery and etc .. - arm-eabi-4.6 - edit build_kernel.sh edit "CROSS_COMPILE" to right toolchain path(You downloaded). EX) CROSS_COMPILE= $(android platform directory you download)/android/prebuilts/gcc/linux-x86/arm/arm-eabi-4.6/bin/arm-eabi- Ex) CROSS_COMPILE=/usr/local/toolchain/arm-eabi-4.6/bin/arm-eabi- // check the location of toolchain - execute Kernel script $ ./build_kernel.sh 2. Output files - Kernel : arch/arm/boot/zImage - module : drivers/*/*.ko 3. How to Clean $ make clean README_Platform.txt [Step to build] 1. Get android open source. : version info - Android 4.1 ( Download site : http://source.android.com ) 2. Copy module that you want to build - to original android open source If same module exist in android open source, you should replace it. (no overwrite) # It is possible to build all modules at once. 3. You should add module name to 'PRODUCT_PACKAGES' in 'build\target\product\core.mk' as following case. case 1) bluetooth : should add 'audio.a2dp.default' to PRODUCT_PACKAGES case 2) e2fsprog : should add 'e2fsck' to PRODUCT_PACKAGES case 3) libexifa : should add 'libexifa' to PRODUCT_PACKAGES case 4) libjpega : should add 'libjpega' to PRODUCT_PACKAGES case 5) KeyUtils : should add 'libkeyutils' to PRODUCT_PACKAGES case 6) bluetoothtest\bcm_dut : should add 'bcm_dut' to PRODUCT_PACKAGES ex.) [build\target\product\core.mk] - add all module name for case 1 ~ 6 at once PRODUCT_PACKAGES += \ e2fsck \ libexifa \ libjpega \ libkeyutils \ bcm_dut \ audio.a2dp.default 4. In case of 'bluetooth', you should add following text in 'build\target\board\generic\BoardConfig.mk' BOARD_HAVE_BLUETOOTH := true BOARD_HAVE_BLUETOOTH_BCM := true 5. excute build command ./build.sh user What I need to do after followed 2 above files

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  • iPhone - Memory Management - Using Leaks tool and getting some bizarre readings.

    - by Robert
    Hey all, putting the finishing touches on a project of mine so I figured I would run through it and see if and where I had any memory leaks. Found and fixed most of them but there are a couple of things regarding the memory leaks and object alloc that I am confused about. 1) There are 2 memory leaks that do not show me as responsible. There are 8 leaks attributed to AudioToolbox with the function being RegisterEmbeddedAudioCodecs(). This accounts for about 1.5 kb of leaks. The other one is detected immediately when the app begins. Core Graphics is responsible with the extra info being open_handle_to_dylib_path. For the audio leak I have looked over my audio code and to me it seems ok. self.musicPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:songFilePath] error:NULL]; [musicPlayer prepareToPlay]; [musicPlayer play] is called later on in a function. 2) Is it normal for there to be a spike in Object Allocation whenever a new view or controller is presented? My total memory usage is very, very low except for whenever I present a view controller. It spikes then immediately goes back down. I am guessing that this is just the phone handling all the information for switching or something. Blegh. Wall of text. Thanks in advance to anyone who helps! =)

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  • AVAudioPlayer not unloading cached memory after each new allocation

    - by Rob
    I am seeing in Instruments that when I play a sound via the standard "AddMusic" example method that Apple provides, it allocates 32kb of memory via the prepareToPlay call (which references the AudioToolBox framework's Cache_DataSource::ReadBytes function) each time a new player is allocated (i.e. each time a different sound is played). However, that cached data never gets released. This obviously poses a huge problem if it doesn't get released and you have a lot of sound files to play, since it tends to keep allocating memory and eventually crashes if you have enough unique sound files (which I unfortunately do). Have any of you run across this or what am I doing wrong in my code? I've had this issue for a while now and it's really bugging me since my code is verbatim of what Apple's is (I think). How I call the function: - (void)playOnce:(NSString *)aSound { // Gets the file system path to the sound to play. NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; // Converts the sound's file path to an NSURL object NSURL *soundURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; self.soundFileURL = soundURL; [soundURL release]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL: soundFileURL error:nil]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely // this is where the prior cached data never gets released [theAudio prepareToPlay]; // set it up and play [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio setDelegate: self]; [theAudio play]; } and then theAudio gets released in the dealloc method of course.

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  • NSTimer to smooth out playback position

    - by Michael
    I have an audio player and I want to show the current time of the the playback. I'm using a custom play class. The app downloads the mp3 to a file then plays from the file when 5% has been downloaded. I have a progress view update as the file plays and update a label on each call to the progress view. However, this is jerky... sometimes even going backward a digit or two. I was considering using an NSTimer to smooth things out. I would be fired every second to a method and pass the percentage played figure to the method then update the label. First, does this seem reasonable? Second, how do I pass the percentage (a float) over to the target of the timer. Right now I am putting the percent played into a dictionary but this seems less than optimal. This is what is called update the progress bar: -(void)updateAudioProgress:(Percentage)percent { audio = percent; if (!seekChanging) slider.value = percent; NSMutableDictionary *myDictionary = [[NSMutableDictionary alloc] init]; [myDictionary setValue:[NSNumber numberWithFloat:percent] forKey:@"myPercent"]; [NSTimer scheduledTimerWithTimeInterval:5 target:self selector:@selector(myTimerMethod:) userInfo:myDictionary repeats:YES]; [myDictionary release]; } This is called first after 5 seconds but then updates each time the method is called. As always, comments and pointers appreciated.

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  • Pulling specific entries from RSS feed [PHP]

    - by n0s
    So, I have an RSS feed with variations of each item. What I want to do is just get entries that contain a specific section of text. For example: <item> <title>RADIO SHOW - CF64K - 05-20-10 + WRAPUP </title> <link>http://linktoradioshow.com</link> <comments>Radio show from 05-20-10</comments> <pubDate>Thu, 20 May 2010 19:12:12 +0200</pubDate> <category domain="http://linktoradioshow.com/browse/199">Audio / Other</category> <dc:creator>n0s</dc:creator> <guid>http://otherlinktoradioshow.com/</guid> <enclosure url="http://linktoradioshow.com/" length="13005" /> </item> <item> <title>RADIO SHOW - CF128K - 05-20-10 + WRAPUP </title> <link>http://linktoradioshow.com</link> <comments>Radio show from 05-20-10</comments> <pubDate>Thu, 20 May 2010 19:12:12 +0200</pubDate> <category domain="http://linktoradioshow.com/browse/199">Audio / Other</category> <dc:creator>n0s</dc:creator> <guid>http://otherlinktoradioshow.com/</guid> <enclosure url="http://linktoradioshow.com/" length="13005" /> </item> I only want to display the results that contain the string CF64K. While it's probably really simple regex, I can't seem to wrap my head around getting it right. I always get seem to only be able to display the string 'CF64K', and not the stuff that surrounds it. Thanks in advance.

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  • Data Modeling of Entity with Attributes

    - by StackOverflowNewbie
    I'm storing some very basic information "data sources" coming into my application. These data sources can be in the form of a document (e.g. PDF, etc.), audio (e.g. MP3, etc.) or video (e.g. AVI, etc.). Say, for example, I am only interested in the filename of the data source. Thus, I have the following table: DataSource Id (PK) Filename For each data source, I also need to store some of its attributes. Example for a PDF would be "numbe of pages." Example for audio would be "bit rate." Example for video would be "duration." Each DataSource will have different requirements for the attributes that need to be stored. So, I have modeled "data source attribute" this way: DataSourceAttribute Id (PK) DataSourceId (FK) Name Value Thus, I would have records like these: DataSource->Id = 1 DataSource->Filename = 'mydoc.pdf' DataSource->Id = 2 DataSource->Filename = 'mysong.mp3' DataSource->Id = 3 DataSource->Filename = 'myvideo.avi' DataSourceAttribute->Id = 1 DataSourceAttribute->DataSourceId = 1 DataSourceAttribute->Name = 'TotalPages' DataSourceAttribute->Value = '10' DataSourceAttribute->Id = 2 DataSourceAttribute->DataSourceId = 2 DataSourceAttribute->Name = 'BitRate' DataSourceAttribute->Value '16' DataSourceAttribute->Id = 3 DataSourceAttribute->DataSourceId = 3 DataSourceAttribute->Name = 'Duration' DataSourceAttribute->Value = '1:32' My problem is that this doesn't seem to scale. For example, say I need to query for all the PDF documents along with thier total number of pages: Filename, TotalPages 'mydoc.pdf', '10' 'myotherdoc.pdf', '23' ... The JOINs needed to produce the above result is just too costly. How should I address this problem?

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  • MySQL FULLTEXT not working

    - by Ross
    I'm attempting to add searching support for my PHP web app using MySQL's FULLTEXT indexes. I created a test table (using the MyISAM type, with a single text field a) and entered some sample data. Now if I'm right the following query should return both those rows: SELECT * FROM test WHERE MATCH(a) AGAINST('databases') However it returns none. I've done a bit of research and I'm doing everything right as far as I can tell - the table is a MyISAM table, the FULLTEXT indexes are set. I've tried running the query from the prompt and from phpMyAdmin, with no luck. Am I missing something crucial? UPDATE: Ok, while Cody's solution worked in my test case it doesn't seem to work on my actual table: CREATE TABLE IF NOT EXISTS `uploads` ( `id` int(11) NOT NULL AUTO_INCREMENT, `name` text NOT NULL, `size` int(11) NOT NULL, `type` text NOT NULL, `alias` text NOT NULL, `md5sum` text NOT NULL, `uploaded` datetime NOT NULL, PRIMARY KEY (`id`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=6 ; And the data I'm using: INSERT INTO `uploads` (`id`, `name`, `size`, `type`, `alias`, `md5sum`, `uploaded`) VALUES (1, '04 Sickman.mp3', 5261182, 'audio/mp3', '1', 'df2eb6a360fbfa8e0c9893aadc2289de', '2009-07-14 16:08:02'), (2, '07 Dirt.mp3', 5056435, 'audio/mp3', '2', 'edcb873a75c94b5d0368681e4bd9ca41', '2009-07-14 16:08:08'), (3, 'header_bg2.png', 16765, 'image/png', '3', '5bc5cb5c45c7fa329dc881a8476a2af6', '2009-07-14 16:08:30'), (4, 'page_top_right2.png', 5299, 'image/png', '4', '53ea39f826b7c7aeba11060c0d8f4e81', '2009-07-14 16:08:37'), (5, 'todo.txt', 392, 'text/plain', '5', '7ee46db77d1b98b145c9a95444d8dc67', '2009-07-14 16:08:46'); The query I'm now running is: SELECT * FROM `uploads` WHERE MATCH(name) AGAINST ('header' IN BOOLEAN MODE) Which should return row 3, header_bg2.png. Instead I get another empty result set. My options for boolean searching are below: mysql> show variables like 'ft_%'; +--------------------------+----------------+ | Variable_name | Value | +--------------------------+----------------+ | ft_boolean_syntax | + -><()~*:""&| | | ft_max_word_len | 84 | | ft_min_word_len | 4 | | ft_query_expansion_limit | 20 | | ft_stopword_file | (built-in) | +--------------------------+----------------+ 5 rows in set (0.02 sec) "header" is within the word length restrictions and I doubt it's a stop word (I'm not sure how to get the list). Any ideas?

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  • iPhone AVAudioPlayer failed to find codec

    - by Anthony
    Hello, I am writing an app that downloads a wav file from a server and needs to play that file. The files use the mulaw codec with 2:1 compression. These wav files are dynamically created by a seperate process so there is no way for me to preconvert the files to a different format or codec, I need to be able to play them as is. I am using an AVAudioPlayer instance initialized as follows: NSURL *audioURL = [[NSURL alloc] initWithString:@"http://xxx.../file.wav"]; NSData *audioData = [[NSData alloc] initWithContentsOfURL:audioURL]; AVAudioPlayer *audio = [[AVAudioPlayer alloc] initWithData:audioData error:nil]; [audio play]; However, when the play method executes, I get the following Console Output when executing on the Simulator: AudioQueue codec policy 1: failed to find a codec of the requested type I also tried saving the downloaded data to a local file and using a file URL, however that yeilds the same results. The downloaded file does play fine on both Mac and Windows based desktop media players. The SDK docs state that the mulaw codec is supported on the iPhone, so I am unsure why it is failing to find it. Any assistance would be greatly appreciated. Thanks.

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  • SIP UAS asks for OPTIONS

    - by TacB0sS
    Hey, I have UAC that registers to a UAS, after registration the UAS sends me an OPTIONS request, what should I answer it? only the audio media streams? Update I: Allow me to explain myself better... if I want to invite someone to a session I USE the INVITE method and negotiate the media then, for that specific session. But once I register to the server, and it asks me for OPTIONS, then what should I supply, everything my client supports? once I answer it would it deduce that every INVITE I would request from now on would use these medias? or would I need to supply new media with every request? Update II: Hi Wiz, I was in the process of building a negotiation system, so i tried it out and replied the UAS here is the sort dialog we had: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Freeswitch 1.2.3 Max-Forwards: 70 Date: Sat, 05 Jun 2010 12:06:43 GMT Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 OPTIONS In Response To 102: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> CSeq: 102 OPTIONS Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Content-Length: 248 v=0 o=310 4515233118481497946 4515233118481497946 IN IP4 10.0.0.1 s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 10.0.0.1 m=audio 40000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 This response caused the server to stop sending me the options request, does this means I can only use these parameters with the server now? or as you said, it does not matter? Thanks, Adam.

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  • Performance: float to int cast and clipping result to range

    - by durandai
    I'm doing some audio processing with float. The result needs to be converted back to PCM samples, and I noticed that the cast from float to int is surprisingly expensive. Whats furthermore frustrating that I need to clip the result to the range of a short (-32768 to 32767). While I would normally instictively assume that this could be assured by simply casting float to short, this fails miserably in Java, since on the bytecode level it results in F2I followed by I2S. So instead of a simple: int sample = (short) flotVal; I needed to resort to this ugly sequence: int sample = (int) floatVal; if (sample > 32767) { sample = 32767; } else if (sample < -32768) { sample = -32768; } Is there a faster way to do this? (about ~6% of the total runtime seems to be spent on casting, while 6% seem to be not that much at first glance, its astounding when I consider that the processing part involves a good chunk of matrix multiplications and IDCT) EDIT The cast/clipping code above is (not surprisingly) in the body of a loop that reads float values from a float[] and puts them into a byte[]. I have a test suite that measures total runtime on several test cases (processing about 200MB of raw audio data). The 6% were concluded from the runtime difference when the cast assignment "int sample = (int) floatVal" was replaced by assigning the loop index to sample. EDIT @leopoldkot: I'm aware of the truncation in Java, as stated in the original question (F2I, I2S bytecode sequence). I only tried the cast to short because I assumed that Java had an F2S bytecode, which it unfortunately does not (comming originally from an 68K assembly background, where a simple "fmove.w FP0, D0" would have done exactly what I wanted).

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  • Web Hosting: Any web host that supports files more than 50,000 in number?

    - by Devner
    Hi all, For my PHP & mySQL based application, I am trying to buy website hosting from a host who does not have a limit on the number of files I carry in my hosting account. Almost all the websites have a common limit of 50,000 files (some websites call it 50,000 nodes). The rest(to the extent of my search) are not even close. I have gone through the various websites, Googled lot of information, have spoken with the customer service of the hosting companies and they said that they have a limit of 50,000 files and that's why they call it the LIMIT. Now I have my application, which is a kind of social networking website, where people can upload various files of varying file size. So say if 50,000 users were to join the website and upload 1 file each, the limit of 50,000 will be reached very easily and my 50,001 customer will start facing file upload problems (& so will my account). So I would like to know if there's any website hosting services that do NOT levy such restrictions. In summary, I need the following options: No maximum file limit (more than 50,000 files in account). No maximum file upload limit in server setting (10MB, 12MB, 15MB, 20MB, etc.). Ability to upload files of various types (zip, flv, jg, png, etc.). Ability to stream Audio and Video (live audio & video not necessary). Access to .htaccess Access to php.ini, my.cnf or my.ini (this would be a plus) Supports SSL. Provides dedicated hosting(& IP) as well. Monthly payments without contracts are a plus. If you know of any such website hosting services, please post a reply ( a link to the same will be appreciated ). Thank you.

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  • Play Multiple iPod Library Songs On iPhone At The Same Time With Pitch Bending & Other Effects

    - by Dino
    Hi, I have been going at this for the past two weeks and it is driving me crazy. I asked this question a couple of days ago (Extract iPod Library raw PCM samples and play with sound effects) and whilst the answer got me half way there I am still stuck. Basically what I am trying to achieve is load up multiple songs from the iPod library for playback with effects such as pitch bend, eq effects etc... I have gone down the route of AVPlayer and AVAudioPlayer which are too simple. The only framework I've seen that can play audio with these effects is OpenAL. I have tried a few objective c wrappers (Finch and ObjectAL) Finch does not play compressed audio whilst ObjectAL will only convert it for me if I pass in a URL for the file (which is something I cannot do because I only have an incompatible iPod library URL). An example of an app that does what I want beautifilly is Tap DJ. It can load up songs from the iPod library fast (unlike TouchDJ and it plays them with all sorts of effects. Any help would be much appreciated.

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  • Play Video From Raw Folder

    - by SterAllures
    Evening, I've just started programming with android and made a few programs and everything so I'm still kind of a novice but im trying to understand it all. So here's my problem, I'm trying to play a video, the thing is, I got it working when I Stream it from an URL with VideoView over the internet or when i place in on my sdcard. What I want to do now is play a video I've got in my res/raw folder, but it only plays audio and I don't understand why, it doesn't give any error in my logcat as far as I can see, also couldn't really find a solution with google since most of the answers are about VideoView and just put the video on your SDCard. Now someone told me I had to use setDisplay (SurfaceHolder) and I've also tried that but I still only get the audio. I hope somebody can help me to find a solution to this problem. VideoDemo.java package nl.melvin.videodemo; import android.app.Activity; import android.os.Bundle; import android.media.MediaPlayer; import android.view.SurfaceHolder; import android.view.SurfaceView; public class videodemo extends Activity { public SurfaceHolder holder; public SurfaceView surfaceView; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); MediaPlayer mp = MediaPlayer.create(this, R.raw.mac); mp.setDisplay(holder); mp.start(); } } XML <?xml version="1.0" encoding="utf-8"?> <LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/LinearLayout01" android:layout_width="fill_parent" android:layout_height="fill_parent" > <SurfaceView android:id="@+id/surfaceview" android:layout_width="fill_parent" android:layout_height="fill_parent"> </SurfaceView>> </LinearLayout> I've also tried Uri.parse but it says it can't play the video (.mp4 format).

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  • Any guidelines for handling the Headset and Bluetooth AVRC transport controls in Android 2.2

    - by StefanK
    I am trying to figure out what is the correct (new) approach for handling the Intent.ACTION_MEDIA_BUTTON in Froyo. In pre 2.2 days we had to register a BroadcastReceiver (either permanently or at run-time) and the Media Button events would arrive, as long as no other application intercepts them and aborts the broadcast. Froyo seems to still somewhat support that model (at least for the wired headset), but it also introduces the registerMediaButtonEventReceiver, and unregisterMediaButtonEventReceiver methods that seem to control the "transport focus" between applications. During my experiments, using registerMediaButtonEventReceiver does cause both the bluetooth and the wired headset button presses to be routed to the application's broadcast receiver (the app gets the "transport focus"), but it looks like any change in the audio routing (for example unplugging the headset) shits the focus back to the default media player. What is the logic behind the implementation in Android 2.2? What is correct way to handle transport controls? Do we have to detect the change in the audio routing and try to re-gain the focus? This is an issue that any 3rd party media player on the Android platform has to deal with, so I hope that somebody (probably a Google Engineer) can provide some guidelines that we can all follow. Having a standard approach may make headset button controls a bit more predictable for the end users. Stefan

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  • Which is the best way to encode batch videos on server side?

    - by albanx
    Hello I am making a general question since I am a developer and I have no advance experience on video elaboration. I have to preparare a web application with the purpose to allow video files upload on our company server and then video elaboration by server, on user command. The purpose of the web application is to allow to the user to make some elaboration on video depending on user action launch from the web app: (server has to ) convert video in different format(mp4, flv...) extact keyframes from video and saves them in jpeg format possibility to extract audio from video automatic control of quality audio & video (black frames,silences detection) change scene detection and keyframe extraction ..... This what's my bosses wanted from the web based application (with the server support obviously), and I understand only the first 3 points of this list, the rest for me was arabic.... My question is: Which is the best and fastest server side application for this works, that can support multiple batch video conversions, from command line (comand line for php-soap-socket interaction or something else..)? Is suitable Adobe Media Server for batch video conversion? Which are adobe products that can be used for this purpose? Note: I have experience with Indesign Server scripting programing (sending xml with php and soap call...), and I am looking to something similiar for video elaboration. I will appreciate any answers. THANKS ALL

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