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  • iPad Video Playback only delivers audio, not visuals.

    - by Dwaine Bailey
    Hi guys, Recently we've developed an iPhone app for an external company, and everything works fine in the app. There is a section where the app pulls video from the client's server, and streams it into the iPhone's MPMoviePlayerController. This works fine on the iPhone and iPodTouch - both the video and the audio show up just great. The problem, however, is that when the app is run on an iPad (using the iPad's iPhone simulator thingo that it does) only the audio plays, and no video can be seen. Does anybody have any suggestions about what may be causing this? I thought perhaps it was the encoding, but then why would this prevent the video from playing on the iPad, and not the iPhone?

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  • GUI Control For Audio Presentation

    - by Boris
    I need GUI control for audio file presentation. The language is not very important but it should run on windows platform. I should be able to :- load the file play the sound put and move markers across the audio bar. it would be nice if it can load itself from RTP wireshark captures (and not wav files). An example may be seen in audacity (may be someone even had an experience extracting it from there). Writing nyquist scripts in audacity is not a good option because I have to operate on RTP captures and not on raw sound samples. Another example of such control is wireshark RTP analyzer. Any advise?

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  • Gapless (looping) audio playback with DirectX in C#

    - by horsedrowner
    I'm currently using the following code (C#): private static void PlayLoop(string filename) { Audio player = new Audio(filename); player.Play(); while (player.Playing) { if (player.CurrentPosition >= player.Duration) { player.SeekCurrentPosition(0, SeekPositionFlags.AbsolutePositioning); } System.Threading.Thread.Sleep(100); } } This code works, and the file I'm playing is looping. But, obviously, there is a small gap between each playback. I tried reducing the Thread.Sleep it to 10 or 5, but the gap remains. I also tried removing it completely, but then the CPU usage raises to 100% and there's still a small gap. Is there any (simple) way to make playback in DirectX gapless? It's not a big deal since it's only a personal project, but if I'm doing something foolish or otherwise completely wrong, I'd love to know. Thanks in advance.

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  • Audio Player with Custom Buttons

    - by Bryan Wong
    I am developing a website but require help regarding a simple javascript audio player. Currently, I have four divs set up as the "buttons" : previous song; pause; play; and next song. Pretty much self explanatory, each button serves its obvious function, previous song, pause the song, play the song, and next song. With this in mind, I am also hoping to have the music start playing right after the page completes loading. I understand there are numerous javascript solutions that involve the use of third-party "applications" such as jplayer, however, I am not well learned in javascript and would like to request the aid of the general body in this matter. LOL. that was awkwardly formal. Um, but yes. I am looking for a way to use my four divs as the controllers of a multi-track audio player. Thanks,

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  • Recording audio from many source/microphones

    - by user657429
    I'm curious if it's possible to record audio from many sources and if not, what's the limitation. Many current devices have two internal microphones (basically for noise reduction). On top of that it's possible to plug additional external one using audiojack. You can as well have another audio stream via bluetooth headset. You are allowed to specify AudioSource in android but is it possible to do recording from many sources at the same time? I'm also interested how the situation look like on the iOS devices.

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  • Learning to work with audio in C++

    - by Skilldrick
    My degree was in audio engineering, but I'm fairly new to programming. I'd like to learn how to work with audio in a programming environment, partly so I can learn C++ better through interesting projects. First off, is C++ the right language for this? Is there any reason I shouldn't be using it? I've heard of Soundfile and some other libraries - what would you recommend? Finally, does anyone know of any good tutorials in this subject? I've learnt the basics of DSP - I just want to program it! EDIT: I use Windows. I'd like to play about with real-time stuff, a bit like Max/MSP but with more control.

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  • Getting level values from PCM raw data using Core Audio

    - by John
    I am trying to extract level data from a PCM audio file using core audio. I have gotten as far as (I believe) getting the raw data into a byte array (UInt8) but it is 16 bit PCM data and I am having trouble reading the data out. The input is from the iPhone microphone, which I have set as: [recordSetting setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; which is obviously 16 bits. I am then trying to just print out a few values to see if they look reasonable for debug purposes below, and they do not look reasonable (many 0's). ExtAudioFileRef inputFile = NULL; ExtAudioFileOpenURL(track.location, &inputFile); AudioStreamBasicDescription inputFileFormat; UInt32 dataSize = (UInt32)sizeof(inputFileFormat); ExtAudioFileGetProperty(inputFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &inputFileFormat); UInt8 *buffer = malloc(BUFFER_SIZE); AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mData = buffer; //pointer to buffer of audio data bufferList.mBuffers[0].mDataByteSize = BUFFER_SIZE; //number of bytes in the buffer while(true) { UInt32 frameCount = (bufferList.mBuffers[0].mDataByteSize / inputFileFormat.mBytesPerFrame); // Read a chunk of input OSStatus status = ExtAudioFileRead(inputFile, &frameCount, &bufferList); // If no frames were returned, conversion is finished if(0 == frameCount) break; NSLog(@"---"); int16_t *bufferl = &buffer; for(int i=0;i<100;i++){ //const int16_t *bufferl = bufferl[i]; NSLog(@"%d",bufferl[i]); } } Not sure what I am doing wrong, I think it has to do with reading the byte array. Sorry for the long code post...

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  • SVG text - total length changes depending on zoom

    - by skco
    In SVG (for web-browsers), if i add a <text>-element and add some text to it the total rendered width of the text string will change depending on the scale of the text. Lets say i add "mmmmmmmmmmmmmmmmmmmmmmmmmmA" as text, then i want to draw a vertical line(or other exactly positioned element) intersecting the very last character. Works fine but if i zoom out the text will become shorter or longer and the line will not intersect the text in the right place anymore. The error can be as much as +/- 5 characters width which is unacceptable. The error is also unpredictable, 150% and 160% zoom can add 3 characters length while 155% is 2 charlengths shorter. My zoom is implemented as a scale-transform on the root element of my canvas which is a <g>. I have tried to multiply the font-size with 1000x and scale down equally on the zoom-transform and vice versa in case it was a floating point error but the result is the same. I found the textLength-attribute[1] which is supposed to adjust the total length so the text always end where i choose but it only works in Webkit. Firefox and Opera seems to not care at all about this value (haven't tried in IE9 yet). Is there any way to render text exactly positioned without resorting to homemade filling of font-outlines? [1] http://www.w3.org/TR/SVG11/text.html#TextElementTextLengthAttribute Update Snippet of the structure i'm using <svg> <g transform="scale(1)"> <!--This is the root, i'm changing the scale of this element to zoom --> <g transform="scale(0.014)"> <!--This is a wrapper for multi-line text, scaling, other grouping etc --> <text font-size="1000" textLength="40000">ABDCDEFGHIJKLMNOPQRSTUVXYZÅÄÖabcdefghijklmnopqrstxyzåäö1234567890</text> </g> </g>

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  • Convert Video and Remove Commercials in Windows 7 Media Center with MCEBuddy 1.1

    - by DigitalGeekery
    Today look at MCEBuddy for Windows 7 Media Center. This handy app automatically takes your recorded TV files and converts them to MP4, AVI, WMV, or MPEG format. It even has the option to cut out those annoying commercials during the conversion process. Installation and Configuration Download and extract MCE Buddy. (Download link below) Run the setup.exe file and take all the default settings.   Open MCEBuddy Configuration by going to Start > All Programs > MCEBuddy > MCEBuddy Configuration.   Video Paths The MCEBuddy application is comprised of a single window. The first step you’ll want to take is to define your Source and Destination paths. The “Source” will most likely be your Recorded TV directory. The Destination should NOT be the same as the Source folder. Note: The Recorded TV directory in Windows 7 Media Center will only display and play WTV & DVR-MS files. To watch the converted MP4, AVI, WMV, or MPEG files in Windows Media Center you’ll need to add them to your Video Library or Movie Library. Video Conversion Next, choose your preferred format for conversion from the “Convert to” drop down list. The default is MP4 with the H.264 codec. You’ll find a wide variety of formats. The first set of conversion options in the drop down list will resize the video to 720 pixels wide. The next two sections maintain the original size, and the final section is for a variety of portable devices.   Next, you’ll see a group of check boxes below the “Convert to” drop down list. The Commercial Skipping option will cut the commercials while converting the file. Sort By Series will create a sub-folder in your Destination folder for each TV show. Delete Original will delete the WTV file after conversion is complete. (This option is not recommended unless you are sure your files are converting properly and you no longer need the WTV file.) Start Minimized is ideal if you want to run MCEBuddy on Windows startup. Note: MCEBuddy installs and uses Comskip for commercial cutting by default. However, if you have ShowAnalyzer installed, it will use that application instead. Advanced Options To choose a specific time of day to perform the conversions, click the checkbox under the “Advanced Options,” and select the starting and ending times for conversion. For example, convert between 2 hours and 5 hours would be between 2 am and 5am. If you want MCEBuddy to constantly look for and immediately convert new recordings, leave the box unchecked.   The “Video age” option lets you choose a specific number of days to wait before performing the conversion. This can be useful if you want to watch the recordings first and delete those you don’t wish to convert. You can also choose the “Sub Directories” if you’d like MCEBuddy to convert files that are in a sub-folder in your “Source” directory. Second Conversion As you might expect, this option allows MCEBuddy to perform a second conversion of your file. This can be useful if you want to use your first conversion to create a higher quality MP4 or AVI file for playback on a larger screen, and a second one for a portable device such as Zune or iPhone. The same options from the first conversion are also available for the second. You’ll want to choose a separate Destination folder for the second conversion.   Start and Monitor Progress To start converting your video files, simply press the “Start” button at the bottom. You’ll be able to follow the progress in the “Current Activity” section. When all the video files have finished converting, or there are no current files to convert, MCEBuddy will display a “Started – Idle” status. Click “Stop” if you don’t want MCEBuddy to continue scanning for new files.   Conclusion MCEBuddy 1.1 will convert all WTV files in it’s source folder. If you want to pick and choose which recordings to convert, you may want to define a source folder different than the Recorded TV folder and then just copy or move the files you wish to convert into the new source folder. The conversion process does take a good bit of time. If you choose the commercial skipping and second conversion options it can take several hours to fully convert one TV recording. Overall, MCEBuddy makes a nice Media Center addition for those that want to save some space with smaller size files, convert Recorded TV files for their portable device, or automatically remove commercials. If you’re looking for a different method to skip commercials check out our post on how to skip commercials in Windows 7 Media Center. Download MCEBuddy 1.1 Similar Articles Productive Geek Tips Using Netflix Watchnow in Windows Vista Media Center (Gmedia)How To Skip Commercials in Windows 7 Media CenterHow To Convert Video Files to MP3 with VLCStartup Customizations for Media Center in Windows 7Add Folders to the Movie Library in Windows 7 Media Center TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 PCmover Professional The Ultimate Excel Cheatsheet Convert the Quick Launch Bar into a Super Application Launcher Automate Tasks in Linux with Crontab Discover New Bundled Feeds in Google Reader Play Music in Chrome by Simply Dragging a File 15 Great Illustrations by Chow Hon Lam

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • Easy line break removal from text fields and/or selections

    - by AmV
    I'm looking for a tool that allows me to easily re-wrap text (i.e. remove line breaks, but not paragraph breaks from a text selection or the current text field that is being edited), and that works at least with text fields in my browser (Chrome) and on Windows. Bonus points for anything that works outside the browser, and that works in-place (i.e. that doesn't require copy-pasting the text through a separate window or using something like http://www.textfixer.com/tools/remove-line-breaks.php) Browser extensions, GreaseMonkey scripts or applications that also work on Linux and/or Mac (or even better, that are multi-platform) are all welcomed. Here is an example of how the tool should behave. If I have the following in a text field: This is a test for SuperUser.com. This is a test for SuperUser.com. This is a test for SuperUser.com. This is a test for SuperUser.com This is a test for SuperUser.com. This is a test for SuperUser.com. This is a test for SuperUser.com. This is a test for SuperUser.com I'd like to have an easy tool that allows me to, for example, select the text, and with a keyboard shortcut convert it to: This is a test for SuperUser.com. This is a test for SuperUser.com. This a test for SuperUser.com. This is a test for SuperUser.com This is a test for SuperUser.com. This is a test for SuperUser.com. This a test for SuperUser.com. This is a test for SuperUser.com Thanks in advance!

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  • Software Suggestion for Managing Voice Recordings (Windows)

    - by Cbeppe
    I'm looking for Windows software that allows me to effectlively manage already made voice recordings. I have a series of recordings taken from an iPhone and I have extracted the files. The problem is that these are very long recordings and therefore I'm looking for software that allows me to: Bookmark a time in the recording Effectively manage multiple files (like Adobe Bridge does with images) Freeware or Payware Possibly other features, I haven't done this before and I'm sorry I'm unable to give a more professional description. Thanks in advance to everyone who can help! If you have any other questions, please don't hesitate to ask - I will try my best to provide useful answers.

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  • Silverlight 2.0 - Can't get the text wrapping behaviour that I want

    - by Anthony
    I am having trouble getting Silverlight 2.0 to lay out text exactly how I want. I want text with line breaks and embedded links, with wrapping, like HTML text in a web page. Here's the closest that I have come: <UserControl x:Class="FlowPanelTest.Page" xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation" xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml" xmlns:Controls="clr-namespace:Microsoft.Windows.Controls;assembly=Microsoft.Windows.Controls" Width="250" Height="300"> <Border BorderBrush="Black" BorderThickness="2" > <Controls:WrapPanel> <TextBlock x:Name="tb1" TextWrapping="Wrap">Short text. </TextBlock> <TextBlock x:Name="tb2" TextWrapping="Wrap">A bit of text. </TextBlock> <TextBlock x:Name="tb3" TextWrapping="Wrap">About half of a line of text.</TextBlock> <TextBlock x:Name="tb4" TextWrapping="Wrap">More than half a line of longer text.</TextBlock> <TextBlock x:Name="tb5" TextWrapping="Wrap">More than one line of text, so it will wrap onto the following line.</TextBlock> </Controls:WrapPanel> </Border> </UserControl> But the issue is that although the text blocks tb1 and tb2 will go onto the same line because there is room enough for them completely, tb3 onwards will not start on the same line as the previous block, even though it will wrap onto following lines. I want each text block to start where the previous one ends, on the same line. I want to put click event handlers on some of the text. I also want paragraph breaks. Essentially I'm trying to work around the lack of FlowDocument and Hyperlink controls in Silverlight 2.0's subset of XAML. To answer the questions posed in the answers: Why not use runs for the non-clickable text? If I just use individual TextBlocks only on the clickable text, then those bits of text will still suffer from the wrapping problem illustrated above. And the TextBlock just before the link, and the TextBlock just after. Essentially all of it. It doesn't look like I have many opportunities for putting multiple runs in the same TextBlock. Dividing the links from the other text with RegExs and loops is not the issue at all, the issue is display layout. Why not put each word in an individual TextBlock in a WrapPanel Aside from being an ugly hack, this does not play at all well with linebreaks - the layout is incorrect. It would also make the underline style of linked text into a broken line. Here's an example with each word in its own TextBlock. Try running it, note that the linebreak isn't shown in the right place at all. <UserControl x:Class="SilverlightApplication2.Page" xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation" xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml" xmlns:Controls="clr-namespace:Microsoft.Windows.Controls;assembly=Microsoft.Windows.Controls" Width="300" Height="300"> <Controls:WrapPanel> <TextBlock TextWrapping="Wrap">Short1 </TextBlock> <TextBlock TextWrapping="Wrap">Longer1 </TextBlock> <TextBlock TextWrapping="Wrap">Longerest1 </TextBlock> <TextBlock TextWrapping="Wrap"> <Run>Break</Run> <LineBreak></LineBreak> </TextBlock> <TextBlock TextWrapping="Wrap">Short2</TextBlock> <TextBlock TextWrapping="Wrap">Longer2</TextBlock> <TextBlock TextWrapping="Wrap">Longerest2</TextBlock> <TextBlock TextWrapping="Wrap">Short3</TextBlock> <TextBlock TextWrapping="Wrap">Longer3</TextBlock> <TextBlock TextWrapping="Wrap">Longerest3</TextBlock> </Controls:WrapPanel> </UserControl> What about The LinkLabelControl as here and here. It has the same problems as the approach above, since it's much the same. Try running the sample, and make the link text longer and longer until it wraps. Note that the link starts on a new line, which it shouldn't. Make the link text even longer, so that the link text is longer than a line. Note that it doesn't wrap at all, it cuts off. This control doesn't handle line breaks and paragraph breaks either. Why not put the text all in runs, detect clicks on the containing TextBlock and work out which run was clicked Runs do not have mouse events, but the containing TextBlock does. I can't find a way to check if the run is under the mouse (IsMouseOver is not present in SilverLight) or to find the bounding geometry of the run (no clip property). There is VisualTreeHelper.FindElementsInHostCoordinates() The code below uses VisualTreeHelper.FindElementsInHostCoordinates to get the controls under the click. The output lists the TextBlock but not the Run, since a Run is not a UiElement. private void theText_MouseLeftButtonDown(object sender, System.Windows.Input.MouseButtonEventArgs e) { // get the elements under the click UIElement uiElementSender = sender as UIElement; Point clickPos = e.GetPosition(uiElementSender); var UiElementsUnderClick = VisualTreeHelper.FindElementsInHostCoordinates(clickPos, uiElementSender); // show the controls string outputText = ""; foreach (var uiElement in UiElementsUnderClick) { outputText += uiElement.GetType().ToString() + "\n"; } this.outText.Text = outputText; } Use an empty text block with a margin to space following content onto a following line I'm still thinking about this one. How do you calculate the right width for a line-breaking block to force following content onto the following line? Too short and the following content will still be on the same line, at the right. Too long and the "linebreak" will be on the following line, with content after it. You would have to resize the breaks when the control is resized. Some of the code for this is: TextBlock lineBreak = new TextBlock(); lineBreak.TextWrapping = TextWrapping.Wrap; lineBreak.Text = " "; // need adaptive width lineBreak.Margin = new Thickness(0, 0, 200, 0);

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  • Restore audio settings - cannot open mixer: No such file or directory

    - by Alfred M.
    The internal speaker of my laptop never functionned under Ubuntu. I tried to follow indication on the web and now the jack audio does not work either. The graphic interface for audio management now displays a 'dummy output' instead of the three possible outputs I used to have (one of them was working for the jack output). In a terminal alsamixer raises an error: cannot open mixer: No such file or directory I did try to remove and reinstall alsa-utils but it did not change anything. This happened after a failed atempt to install alsa-driver-linuxant_1.0.23.1_all.deb from here. My sound card seems to be not recognised anymore. After reboot I have no more the sound icon in menu bar the upper right corner. I think I have removed my sound card driver. Indeed, the command sudo lshw -class multimedia indicated audi device as unclaimed. Any idea how I could revert to a better situation (that is jack support and alsa working)? EDIT: The command lspci -nnk | grep -iEA3 audio gives lspci -nnk | grep -iEA3 audio 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) Subsystem: ASUSTeK Computer Inc. Device [1043:1893] 00:1c.0 PCI bridge [0604]: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 [8086:2940] (rev 03)

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  • How can i compare Audio, what programming language should i use

    - by Pimmetje
    I have 2 audio files that are from almost the same source. But at some points there shifted a bit. Also the codecs does not match. I would like to make a program that takes a sample 2 - 4 seconds. And looks for it in the other file. (Most of the time it's not shifted more than 30 seconds). Than take the time and store it, Go ahead for a few seconds take a sample and find it again. This way i want to create a file where i can see on what points the file is shifted. For people who are more interested in what i want. I have a audio/video file speech and subtitles. But i have same speech from different sources with differs a bit in time. And i like to make a program that can correct the subtitle time for me. Enough about the problem I looked on the Internet for ways to compare audio files. Based on what i read comparing 2 audio files isn't that easy as i had hoped. Some talk about algorithms http://www.perlmonks.org/?node_id=169641 Some audio-library's portaudio.com aubio.org sourceforge.net/projects/ccaudio/ ambiera.com/irrklang/ The biggest problem i have is that i can't find something i can generate from the audio that i can use to compare with. I hope someone here can point me in the right direction.

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  • Text-Editing program : muti-search-replace/multi-regex?

    - by rlb.usa
    I have a long and arduous text file, and I need to do lots and lots of the same search-replaces on it inside of selections. Is there a text editing program where I can do multiple find/replace (or regex) at one time? That is, I want to : (select text) - (do-find-replace-set-A) - (do other stuff) - (repeat) Instead of : (select text) - (f&r #1, f&r #2, f&r #3 ... ) - (do other stuff) - (repeat) I have textpad, but it's macro's won't handle find/replace.

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  • Encode audio to aac with libavcodec

    - by ryan
    I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

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  • In LaTeX prefer figures on text-heavy pages.

    - by bjarkef
    Hi LaTeX seems to have a preference for placing figures together on a page, and placing surrounding text on a separate page. Can I somehow change that balance a bit, as I prefer figures to break up the text to avoid too black text-heavy pages. Example: \section{Some section} [Half a page of text] \begin{figure} [...] \caption{Figure text 1} \end{figure} [Half a page of text] \begin{figure} [...] \caption{Figure text 2} \end{figure} [More text] So what LaTeX usually does is to stack the two half pages of text on a single page, and the figures on the following page. I believe this really gives a bad balance, and bores the reader. So can I change that somehow? I know about postfixing the \begin{figure} with [ht!], but often it does not really matter. I would like to configure the balancing algorithms in LaTeX to naturally prefer pages with combined figures and text.

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  • VERY large text files and Snow Leopard

    - by cbmeeks
    Sometimes I need to work on EXTREMELY large text files. 200-300 megs or more. My favorite text editor on my MacBook Pro is TextMate. However, TM chokes on very large text files. Even ones around the 100MB mark. Is there a text editor that can handle such files for Snow Leopard? Thanks!

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