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• Recommend a Linux video editor. [closed]

  - by joeforker

  Possible Duplicate: Looking For Video Editing Software for Ubuntu I'd like a working video editor for 720p MPEG4 (.mov) with 16Khz ulaw audio in Linux. I've already tried pitivi (audio encoding issues) and kdenlive (crashes almost immediately after a clip is imported). What should I try?

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• Sound problem in my Desktop

  - by bobby521

  I have a Dell Dimension 5150 desktop, I am not able to hear any sound when any audio file is playing, I have checked the audio drivers and have downloaded the drivers from the dell site, Is this a Hardware problem with the sound card, in case is there any way to find out whether its the problem with the sound card or any suggestions is greatly appreciated

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• Sound device doesn't work in Windows 7

  - by Alex Farber

  Device manager shows that device is properly installed: High Definition Audio Device Device type: Sound, video and game controllers Manufacter: Microsoft Location: Location 0 (Internal High Definition Audio Bus) But every program trying to play sound reports that devicce is unavailable. In Windows XP it works properly.

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• How do you splice out a part of an xvid encoded avi file, with ffmpeg? (no problems with other files)

  - by user11955

  Im using the following command, which works for most files, except what seems to be xvid encoded ones /usr/bin/ffmpeg -sameq -i file.avi -ss 00:01:00 -t 00:00:30 -ac 2 -r 25 -copyts output.avi So this should basically splice out 30 seconds of video + audio, starting from 1 minute mark. It does START encoding at the 00:01:00 mark but it goes all the way to the end of the file for some reason, ignoring that I want just 30 seconds. The output looks like this. FFmpeg version git-ecc4bdd, Copyright (c) 2000-2010 the FFmpeg developers built on May 31 2010 04:52:24 with gcc 4.4.3 20100127 (Red Hat 4.4.3-4) configuration: --enable-libx264 --enable-libxvid --enable-libmp3lame --enable-libopenjpeg --enable-libfaac --enable-libvorbis --enable-gpl --enable-nonfree --enable-libxvid --enable-pthreads --enable-libfaad --extra-cflags=-fPIC --enable-postproc --enable-libtheora --enable-libvorbis --enable-shared libavutil 50.15. 2 / 50.15. 2 libavcodec 52.67. 0 / 52.67. 0 libavformat 52.62. 0 / 52.62. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.20. 0 / 1.20. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'file.avi': Metadata: ISFT : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:02:00.00, start: 0.000000, bitrate: 1587 kb/s Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s File 'lol6.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to 'lol6.avi': Metadata: ISFT : Lavf52.62.0 Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected [buffer @ 0x184b610]Buffering several frames is not supported. Please consume all available frames before adding a new one. frame= 1501 fps=104 q=0.0 Lsize= 15612kB time=30.02 bitrate=4259.7kbits/s ts/s video:15303kB audio:235kB global headers:0kB muxing overhead 0.482620% if I convert this file to mp4 for example, and then perform the same action, it works perfectly.

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• 500 MB Avi video from CD lagging/glitching when reproduced

  - by Caki Esther

  I created an extra cd with Nero Burning that contains both audio tracks (I can hear them correctly) and a .avi presentation (pretty big one, 540 MB on a 700 MB cd). The audio is fine but the problem is that when the video is played from the cd (with whatever media player: Windows Media Player, VLC, etc..) it lags/glitches/stutters. I'd like the video to be smooth, how should I burn the video to reduce this effect? I mean: what kind of compression/format and why?

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• Can my PC run Need for Speed Shift

  - by John

  Here are my PC's specs: Operating System MS Windows 7 32-bit CPU Intel Mobile Core 2 Duo T8100 @ 2.10GHz Penryn 45nm Technology RAM 3.0GB Dual-Channel DDR2 @ 332MHz 5-5-5-15 Motherboard Sony Corporation VAIO (N/A) Graphics Nvidia Defaul @ 1280x800 256MB GeForce 8400M GT (Sony) Hard Drives 250GB Hitachi Hitachi HTS542525K9SA00 ATA Device (IDE) Optical Drives Optiarc DVD RW AD-7560A ATA Device Audio High Definition Audio Device

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• reencode several videos with virtualdub?

  - by acidzombie24

  I have about 50 small videos (and a few large videos). I want to convert them all with the SAME settings. Its basically change audio to X with Y bitrate, change video to xvid. and do full processing on the video and audio. Then force the FPS to 15 since every program i tried (including virtualdub) thinks it 0.3 FPS. How do i apply all of these settings to all of my files?

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• Listening to the iPhone mic with SCListener and playing music at the same time: how?

  - by Eamon Ford

  Hello, I am using Stephen Celis' SCListener class (for iPhone) to "listen" from the microphone, but I also need to be playing music at the same time using the MediaPlayer framework. However, when I start listening with SCListener, the music fades out and stops. I have set the kAudioSessionCategory_PlayAndRecord property on the audio session in SCListener, which should allow me to play audio and record audio at the same time, but as far as I can tell it has no effect. I'm confused, because according to other developers' results, this works just fine, but not for me. I'm thinking maybe the kAudioSessionCategory_PlayAndRecord property allows you to play sound and record if you're using the AVAudioPlayer framework or something to play the sound, but maybe not the MediaPlayer framework? This would be a problem for me because I need to play music from the user's iPod library, which, as far as I know is only possible to do using the MediaPlayer framework. Does anyone know how I can get around this problem? Thanks in advance!

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• using AudioQueues with AudioFileReadBytes

  - by Santosh

  Hey Im trying to work with Audio queues to play a very big mp3 file (arround 23 hours long). when audio queue asks for buffers though callback, im using AudioFileReadBytes() API to read the bytes from audio file and feed the queue. startQueue fails with the error : prime failed any inputs????? Also I succeeded playing file using AudioFileReadPackets API instead of AudioFileReadBytes(). But the problem with API is that when I seek (fast forward) by a long interval, say 9 hours (for example fast forward from 32 mins playtime to 9:32 mins) then AudioFileReadPackets() takes a long time (almost 2 mins) to read from new location. any comments would be greatly appreciated.

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• Adobe Flash player Secuirty Pop-Up question

  - by kapildalwani

  I am building a Audio Recording tool using Flash and Wowza. I dont want to start the recording until the use clicks the Allow Button is the Security Pop-up question represented here http://www.macromedia.com/support/documentation/en/flashplayer/help/help05.html In Audio I dont get this until I attach the stream to it. In Video can get thsi question when I attach the camera to Video. I want to avoid making a connection until the user clicks Accept and this doesn't happen until I make the connection request in Audio. I am able to display the http://www.macromedia.com/support/documentation/en/flashplayer/help/help09.html pop-up using SecurityManager Is there a way I can call the pop-up from my code. http://www.macromedia.com/support/documentation/en/flashplayer/help/help05.html

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• Sending most correct mimetype

  - by Roland Franssen

  Hi all, I have a list of extension to mimetype in a INI file. However some extensions have multiple mimetypes, for example; midi[] = "application/x-midi" midi[] = "audio/midi" midi[] = "audio/x-mid" midi[] = "audio/x-midi" midi[] = "music/crescendo" midi[] = "x-music/x-midi" 6 (possible) mimetypes for 1 extension. Whats common practice to determine the correct mimetype? (e.g. i need to set a HTTP content-type header). I know its not ideal; determining mimetypes based on extension.. but i need consistent (cross-server) results (e.g. fileinfo extension in PHP is making terrible guesses*). * Some fileinfo results for example; js - text/plain css - text/c-h

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• Expression Encoder SDK - WMA Pro Codec Issues with Windows Server 2003

  - by PortageMonkey

  I am using the Expression Encoder SDK to encode .avi and Flash files to a .wmv format suitable for Silverlight. By default, EE encodes files with audio using the the WMA PRO codec. If you are running Windows Server 2003, this is a problem as it doesn't support the WMA PRO codec and produces and error message similar to the following. Error Message: The Audio Profile settings do not match a valid system profile. Error Source: Microsoft.Expression.Encoder Error Target Site: System.String GetProfileString() I am looking for a way to change the default audio codec to something suitable for WS 2003. I am aware that although not supported natively, there is a highly invasive way to install Windows Media Player 11 and it's codecs on WS 2003 but this involves registry tinkering and other hacks not suitable for our production environments so that solution is out.

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• AVAudioPlayer Memory Leak - Media Player Framework

  - by Krishnan

  Hi Friends, I am using AVAudioPlayer object to play an audio. I create an audioPlayer object initially. I play an animation and when ever animation starts I play the audio and pause the audio when the animation is finished. I initially found three memory Leaks using Instruments. (The responsible caller mentioned was RegisterEmbedCodecs). After suggestion from a "ahmet emrah" in this forum to add MediaPlayer framework, the number of leaks reduced to one. And is there any way to completely get rid of it? Thanks and regards, krishnan.

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• unable to download a file from rtmp server

  - by user309815

  Hi Team I want to download an audio file from red5 server using rtmp server. string strUri; strUri = "rtmp://XXX/oflaDemo/" + Session["streamName"].ToString(); string strUploadto; strUploadto = Server.MapPath("") + "\Audio\" + "myaudio.flv"; WebClient webClient = new WebClient(); //webClient.DownloadFile("rtmp://begoniaprojects.com/oflaDemo/" + Session["streamName"].ToString(), Page.MapPath("") + "\Audio\" +"myaudio.flv"); webClient.DownloadFile(strUri, strUploadto); but i am getting uri prefix is not recognized message while downloading. please suggest me.

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• Is it possible to avoid downloading song up to the supplied position using SoundManager2?

  - by dinjas

  I'm using the SoundManager2 JavaScript SDK on a site that streams synchronized audio from SoundCloud to multiple clients simultaneously. When a new user loads the page, the audio is loaded, and a position parameter is set to specify where playback should begin. The problem arises when the track is really long (say 60 minutes), and the current track position is substantially far into the track (e.g. 30 minutes). When this is the case, it takes a really long time before playback begins because the track has to download/buffer up to the current position. Is there a way to avoid downloading the 30 minutes of audio that I don't need?

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• Preload *.wav with SystemSoundID?

  - by fuzzygoat

  I am playing a wav file to give a little audio feedback when a button in my UI is pressed. My question is when you first press the button there is a delay (about 1.5secs) whilst the sound file "sound.wav" is loaded and cached. Is there a way to pre-cache this file (maybe in my viewDidLoad)? I guess I could do it by just playing it a viewDidLoad, but would really need to disable the audio so it does not "beeb" each time the app starts. many thanks for and help. gary EDIT: Looks like my question is a duplicate of this post unless anyone has any new info? Maybe a way to turn the play volume down temporarily, unless the audio is cleared each time through the run loop.

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• SOS: AudioFormat when writing to file in FreeTTS

  - by user330793

  Very annoying problem. I have developed a freeTTS application of the freetts class that write captured audio to file however I am having some very annoying problems. When setting the audio player to singlefileaudio player I try to also set the audioformat with my own default values for sampleRate, sampleSizeInBits, channels, signed and bigEndian. Now I access AudioPlayer.get methods to show these values in runtime just to ensure they are set to what I set them and they match those values. However when file writing completes and I check the properties of the resulting wave file, they are set to the audioPlayer default settings. Normally this will be fine except I have to read the files into another application which has fixed audio property settings so I always get a resulting output that sounds like am fast forwarding the sound and listening to it at the same time. Obviously because of the different sampling rates. I need help please. Thanx, Henry

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• Adding a red5 app in a multiuser website

  - by Zakaria

  hi everybody, I have an mvc php website where users can publish their public information: http://www.example.com/foobar/profile. Beside this project, based on some red5 samples, I have an application (done with Flex) that sends audio: rtmp://server/sendAudio (very basic but works). I want to create for each subscribed on my website an admin part where can send an audio stream: http://admin.example.com/foobar. And, when someone goes on their public profile, they can listen to the streamed audio: http://www.example.com/foobar/profile). How can I use my red5/flash app dynamically with my php website so that my users can broadcast their proper canal? Do you have some experience to share ? Thank you, Regards.

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• Is it bad practice to have a long initialization method?

  - by Paperflyer

  many people have argued about function size. They say that functions in general should be pretty short. Opinions vary from something like 15 lines to "about one screen", which today is probably about 40-80 lines. Also, functions should always fulfill one task only. However, there is one kind of function that frequently fails in both criteria in my code: initialization functions. For example in an audio application, the audio hardware/API has to be set up, audio data has to be converted to a suitable format and the object state has to properly initialized. These are clearly three different tasks and depending on the API this can easily span more than 50 lines. The thing with init-functions is that they are generally only called once, so there is no need to re-use any of the components. Would you still break them up into several smaller functions would you consider big initialization functions to be ok?

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• Does background iOS require specific provisioning?

  - by Moshe

  I've added the appropriate XML to my PLIST ("audio key") and my app did stream audio in the background. I installed the app on to an iPhone 3GS running iOS 4 via Ad Hoc distribution. I played audio in my app and pressed the home key. It was still playing. Then I switched computers and reset my provisioning profile. I recompiled and exported with a generic provisioning profile. I sent the app to someone else to test on their 3GS via Ad Hoc and the app does not work in the background.

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• BlackBerry Technical Specification

  - by Sam

  I'm having trouble locating BlackBerry techical specifications and their website is a mess. They also don't have a number that I can use to easily contact them. This isn't exactly a coding question, but what does the BlackBerry audio API look like, and where can I get technical specifications on audio? Specifically, I'm trying to find out more information on Audio-In, specifically, through the Mic-In on the 3.5 mm jack. Unfortunately, before I can proceed, I need to know such things like sampling rate, data width, etc. Direction to the right resource or if you know off of the top of your head is appreciated.

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• Installing Skype on Amazon EC2 instance

  - by Adrian

  For my application, I need to have Skype working on my Amazon EC2 Windows instance. I got the application installed and am able to log in, however, I can't make a phone call, since I am getting an 'Can't detect your sound card' error. Since I'm trying to inject audio from an audio file into the phone call, I don't need the sound card on the server. Thus, I need a way to bypass this error message. I have tried installing Virtual Audio Cable, which unfortunately didn't work (even though it worked on my desktop machine).

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• Easy way to combine php code (lame question)

  - by alekseygr

  Hi, I have vary easy and LAME question. I have code: <?php if (function_exists("insert_audio_player")) {insert_audio_player("[audio:|titles=]"} ?> This code outputs audio player to my page in wordpress. I need to call custom field inside this code. My custom field code is: <?php print_custom_field('tc_filename'); ?> Something like: <?php if (function_exists("insert_audio_player")) {insert_audio_player("[audio:<?php print_custom_field('tc_filename'); ?>|titles=<?php print_custom_field('tc_title'); ?>]"} ?> How can I call for second code inside first? Thx.

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• Paperclip generating wrong URLs in Heroku

  - by Tony

  Paperclip is generating wrong URLs in Heroku. I have an Audio model which has a mp3 field as follows: class Audio < ActiveRecord::Base has_attached_file :mp3, :storage => :s3, :s3_credentials => S3_CREDENTIALS, :bucket => S3_CREDENTIALS[:bucket], :path => ":rails_root/public/system/:attachment/:id/:style/:filename", :url => "/system/:attachment/:id/:style/:filename" I am calling audio.mp3.url from a controller, and it returns http://s3.amazonaws.com/MyApp/audios/mp3s//original/96a9ae89302fdf8462ee05eb829f2e17578b144e20120908-2-11f61zr.mp3?1347135050 instead of http://s3.amazonaws.com/MyApp/audios/mp3s/000/000/004/original/96a9ae89302fdf8462ee05eb829f2e17578b144e20120908-2-11f61zr.mp3?1347135050 (which works) Why is it missing the '000/000/004' part of the route? The same model is generating the right URL when used in a view. Any help? I am using paperclip 3.2.0 and Rails 3.1.8. Any help?

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• Need to sync two lists with atrribute time, but times aren't equal

  - by virgula24

  I gonna try to describe my problem the best i can. I have two lists, one with audio frames and other with color frames (not relevant). Both of them have timestamps, they were captured at the same moment but at different instants. So, i have like this: index COLOR AUDIO 0 841 846 1 873 897 2 905 948 3 940 1000 the frames start at high numbers because they were captured and then trimmed to specific parts, but im shot, frame 0 is synced with only 5ms apart(timestamp in ms). On every case i have, the audio frames count is less than the color count. I need to make them have the same count. The stating frames may be coloraudio, color

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