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  • After playing a MediaElement, how can I play it again?

    - by Edward Tanguay
    I have a variable MediaElement variable named TestAudio in my Silverlight app. When I click the button, it plays the audio correctly. But when I click the button again, it does not play the audio. How can I make the MediaElement play a second time? None of the tries below to put position back to 0 worked: private void Button_Click_PlayTest(object sender, RoutedEventArgs e) { //TestAudio.Position = new TimeSpan(0, 0, 0); //TestAudio.Position = TestAudio.Position.Add(new TimeSpan(0, 0, 0)); //TestAudio.Position = new TimeSpan(0, 0, 0, 0, 0); //TestAudio.Position = TimeSpan.Zero; TestAudio.Play(); }

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  • How to View Android Native Code Profiling?

    - by David R.
    I started my emulator with ./emulator -trace profile -avd emulator_15. I then tracked down the trace files to ~/.android/avd/rodgers_emulator_15.avd/traces/profile, where there are six files: qtrace.bb, qtrace.exc, qtrace.insn, qtrace.method, qtrace.pid, qtrace.static. I can't figure out what to do with these files. I've tried both dmtracedump and traceview on all of the files, but none seem to generate any output I can do anything with. How can I view the proportion of time taken by native method calls on Android?

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  • How can I set a ringtone for an individual contact on Android ?

    - by PHP_Jedi
    How can I set a ringtone for an individual contact on Android ? I have found a way to set the default ringtone that applies to all contacts without an individual ringtone. But that is not what i'm trying to accomplish. I want the application to have a button "Apply ringtone to contact". When i click, I start an activityForResult displaying a list of all contacts on the phone. When a contact is selected, the contact activity closes and returns with a uri to the contact. Now all the app needs to do is to apply the selected ringtone to that spesific contact. The code for displaying and selecting contacts by an activity is already implemented and seems to work on with the app. Its only the last part left, and I have no clue about how to solve this. Any help would be usefull.

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  • Delphi: Error when starting MCI

    - by marco92w
    I use the TMediaPlayer component for playing music. It works fine with most of my tracks. But it doesn't work with some tracks. When I want to play them, the following error message is shown: Which is German but roughly means that: In the project pMusicPlayer.exe an exception of the class EMCIDeviceError occurred. Message: "Error when starting MCI.". Process was stopped. Continue with "Single Command/Statement" or "Start". The program quits directly after calling the procedure "Play" of TMediaPlayer. This error occurred with the following file for example: file size: 7.40 MB duration: 4:02 minutes bitrate: 256 kBit/s I've encoded this file with a bitrate of 128 kBit/s and thus a file size of 3.70 MB: It works fine! What's wrong with the first file? Windows Media Player or other programs can play it without any problems. Is it possible that Delphi's TMediaPlayer cannot handle big files (e.g. 5 MB) or files with a high bitrate (e.g. 128 kBit/s)? What can I do to solve the problem?

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  • Why is native libmpg123 taking so long on android with libgdx?

    - by cmbryan
    I'm trying to use the gdx-audio extensions, but am having trouble decoding mp3s. It works, but very slowly!! A 10-second file is taking 6.57 seconds to decode :( Here is the method: public void decode() { Mpg123Decoder decoder = new Mpg123Decoder(externalFile); short[] sampleArray = new short[1024]; // read until we reach the end of the file while (decoder.readSamples(sampleArray, 0, sampleArray.length) > 0) {} } Can anyone tell me why this is taking so long?

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  • Why is my htaccess file preventing access to my MP3 file?

    - by Andrew
    My Zend Framework application has a public directory which contains an htaccess file. If the file isn't found in the public directory, it routes the request through the application. I have an MP3 file within my public directory, but the htaccess file is routing the request through the application! Do you see anything wrong with my htaccess file? AddDefaultCharset utf-8 RewriteEngine on RewriteRule ^Resources/.* - [L] RewriteCond %{REQUEST_FILENAME} !-f RewriteCond %{REQUEST_FILENAME} !-d RewriteRule !\.(js|ico|gif|jpg|png|css|htm|html|php|pdf|doc|txt|swf|xml|mp3)$ /index.php [NC]

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  • ffmpeg add two audio streams to video

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

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  • Are there any good music mixers available, equivalent to Windows "MP3 Tunes"?

    - by RobinJ
    In Windows my dad used to have a program called MP3 Tunes. I have tried running it with Wine, and it worked. But strange things kept happening to the program, so it's not a reliable way to play music. Basically I just want to have 2 players (in a single window) with these features: Preloading tracks in a player without immediately starting them. Fading from one track to the other. A timer on each player. These features are also desired, but not required: Microphone input. Prelistening before loading a song in a player (through a seperate sound card). Pitch/Tempo control. Just being able to browse folders in the filesystem (without things like a music library). Here are some screenshots of the program to clarify what I'm looking for:

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  • Is there now any way to convert mp3 files to m4a or aac 192kbit?

    - by piedro
    Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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  • How to get my windows docs ; pics, mp3, and docs off an external usb hard drive (sata 3.5 enclosure)

    - by Alan
    Back when I had windows 7, I cloned my internal hard drive to an enclosed external usb hard drive. I then formatted my internal hard drive and installed ubuntu 11.10 as my only OS. How do I migrate my files, pics, mp3's, and etc off the external drive and back to my internal drive which now only has ubuntu on it. (and why is this process not made easier?) I have tried logging in as the root user but I cannot find the external device. I have downloaded several different apps to manage files, usb sticks and etc. PLEASE HELP!!!

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  • Almost 2013 - Any decent options for mp3 to text? (Speech Recognition)

    - by ajacian81
    I know there's some questions here on s/u regarding converting spoken word mp3 to text, however, most are pretty old (2010 and earlier). I'm just wondering if there's any new legitimate options for this task - if google has shown us anything, speech recognition has come a long way. Personally, I'd prefer a linux based solution, but I'm not picky. I've heard a lot about something called Sphinx, but I tried to set it up and get it going but I couldn't. I know there's a number of different componenents for Sphinx so maybe I was doing it wrong? Either way, are there any new applications for Speech recognition, especially from MP3 files? Thanks!

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  • How can I force Xbox Music to find music in all subfolders of my Music library?

    - by Matthew
    My music library is organized (roughly) like this: Music mp3 (original) Artist/Album/Song mp3 (from Tom) Artist/Album/Song mp3 (from Dick) Artist/Album/Song mp3 (from Harry) Artist/Album/Song ... etc. When I use the desktop Zune Software, it finds all of this music. However, the Xbox Music metro app only seems to find music in the "mp3 (original)" folder. How can I force it to find all of my music?

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  • How to programmatically generate an MP3 podcast file with chapters and text track?

    - by adib
    Hi Anybody know how to programmatically generate MP3 files with bookmarks that can be used in iTunes / iPod / iPhone / iPod touch? Specifically text bookmarks (bookmarks with titles) that the listener can skip to a specific point in time in the audio file. Also how to add the text transcription of the podcast's content. Even better if you have an example Cocoa code or library to write the MP3 file. Thanks.

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  • HTML5 Audio: Which formats? Ditch Ogg Vorbis in favor of Ogg Opus? Is MP3 still needed?

    - by phoibos
    I'm currently working on a website which has to stream audio files. Since bandwidth is always an issue, the file size should be as small as possible. I wonder what audio formats I should provide. MP3 - Most common format but low quality, I don't know if it's even required, since AAC is well supported by the browsers incapable of playing free codecs MP4 AAC - Nice quality / small filesize, supported by Safari / Mobile Devices / IE9 / Flash / Chrome A free codec - well, until recently, there only was Ogg Vorbis, but Ogg Opus is standardized now and it's really good! Questions: Is it time yet to use Opus instead if Vorbis? Firefox supports Opus since version 15, and Opera has support on its roadmap - I guess Chrome will follow in the future too. Do I still have to provide an MP3 file?

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  • How to remove HTML code from search result page content

    - by Jack Torris
    I have music website. There are 46 album pages and each page has different player and files. I just entered the one of album's URLs in a search engine. I found that Google is displaying player code in search result content. For example, enter this URL in Google and check the results. Each result displays a .mp3 file in content section. I see this: This page contains a demo of and documentation for the new jPlayer Playlist add-on, ... mp3:"http://www.jplayer.org/audio/mp3/Miaow-01-Tempered-song.mp3", ... I don't want Google to show the player code and mp3 files in search result. How can I hide audio files and player code from search engine? What would be the best solution for it?

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  • VLC doesnot play any file [ any video/.mp3] in local machine , closes when a file is opened

    - by hsemarap
    When I run vlc from terminal I get: In the VLC dialog box: Your input can't be opened: VLC is unable to open the MRL 'file:///media/Ent/movies/the%20mask.avi'. Check the log for details. In terminal: VLC media player 2.0.1 Twoflower (revision 2.0.1-0-gf432547) [0x8fe8f8] main input error: open of `file/xspf-open:///home/para/.local/share/vlc/ml.xspf' failed [0x8fe8f8] main input error: Your input can't be opened [0x8fe8f8] main input error: VLC is unable to open the MRL 'file/xspf-open:///home/para/.local/share/vlc/ml.xspf'. Check the log for details. [0x8f1aa8] main interface error: no suitable interface module [0x8f9b08] main interface error: no suitable interface module [0x8be008] main libvlc error: option http-user-agent does not exist [0x8be008] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface. [0x8f9b08] qt4 interface error: Unable to load extensions module [0x7f5280000b78] main input error: open of `file:///media/Ent/movies/the%20mask.avi' failed [0x8fc718] main playlist error: could not export playlist

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  • How to track/sniff mp3 files posted on Zippyshare.com? [closed]

    - by Stoan
    I'm not sure if this is a right place to ask this question, We starting a indie recording label, I want to minimize piracy of our music. I want to track/sniff our songs that are posted to Zippyshare.com How can I right a tool to automate this process? we would supply our song names and it would search and notify us if our songs are posted on Zippyshare.com. I'm a junior Java developer. I'm looking for direction on how to right an app that would achieve this, any help is appreciated. Thanks

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  • Anyone know of a .net library/utility that will convert a word document to an mp3 format

    - by EJB
    Anyone know of any well-supported/proven methods for converting a Microsoft word document to an MP3 or wav format such that hearing-impaired folks could "listen" to documents that I have stored in my web-based document management system? I already have the interface built such that someone can use the telephone to get the list of documents available, with the dates and titles "read" to them over the phone, but now I would like the ability to let someone actually listen to the contents of word files stored in the system. Ideally a .net library or utility that would let me convert the DOC - MP3 after each upload would be best, but one that "read" the file on demand would be OK too.

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  • execute a command in all subdirectories bash

    - by Luigi R. Viggiano
    I have a directory structure composed by: iTunes/Music/${author}/${album}/${song.mp3} I implemented a script to strip my mp3 bitrate to 128 kbps using lame (which works on a single file at time). My script looks like this 'normalize_mp3.sh': #!/bin/bash SAVEIFS=$IFS IFS=$(echo -en "\n\b") for f in *.mp3 do lame --cbr $f __out.mp3 mv __out.mp3 $f done IFS=$SAVEIFS This works fine, if I go folder by folder and execute this command. But I'd like to have a "global" command, like in 4DOS so I can run: $ cd iTunes/Music $ global normalize_mp3.sh and the global command would traverse all subdirs and execute the normalize_mp3.sh to strip all my mp3 in all subfolders. Anyone knows if there is a unix equivalent to the 4dos global command? I tried to play with find -exec but I just managed to get an headache.

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  • Why won't Media Monkey add one particular folder of mp3's?

    - by ChrisF
    I'm using the latest and greatest version on Media Monkey (free version) and it won't find the mp3's in one particular folder in my music tree. It can see all the other files in the tree and the folder shows up when I click Add/Rescan files to the library... I have full control over the folder and all the files in the folder. The files play in Windows Media Player. The files play in Media Monkey if I right click and play from the context menu. All the tracks are at least 2 minutes long and over 5MB long and Media Monkey is set to ignore files shorter than 20KB and include all files regardless of length. There was an issue in that the that the genre of the tracks was set to "Classical" and the option that allows you to browse the classical music independently of the other music isn't enabled in the free version. It's a Gold version option only. I hadn't spotted that my other classical music was also missing from the library (I have rather a large library). Once I retagged the music with a different tag and tried to add the files again it reported that it added the tracks, but they still didn't show up in the library.

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  • Is there a best practice for concatenating MP3 Files, adjusting sample rates to match, while preserving original files?

    - by Scott
    Hello overflow community! Does anyone know if there is a "best practice" to concatenate mp3 files to create new files, while preserving the original files? I am working on a CentOS Linux machine, in command line. I will eventually call the command line from a PHP script. I have been doing research and I have come up with a process that I think could work. It combines general advice from different forums, blogs, and sources like this one. So here I go: Create a temporary folder Loop through files to create a new, converted copy, of file into a "raw" format (which one, I don't know. I didn't know "raw" files existed before too long ago. I could use some suggestions on this) Store the path to the temporary files, in the temporary folder, and then loop through the files to concatenate them and then put the new merged file the final "processed directory" Delete the contents of the temporary file with the temporary raw files inside. Convert the final file from "raw" to mp3 and enjoy the finished result I'm thinking that this course of action might be best because I can't necessarily control the quality of the original "source" mp3s. The only other option I could think of would be to create a script that would perform a similar process upon files being added to the system leaving only the files with the "proper" format and removing the original "erroneous" file. Hopefully you can see that I have put some thought into this and that I'm trying to leverage the collective knowledge of this community to choose the best direction. Perhaps there is a better path that I could take? By concatenate, I mean to join together in sequence to create a new audio file from the "concatenated files."

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