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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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  • Looking for Mini itx board suitable for hooking up to a television.

    - by Jaxsun
    I am building a small computer to hook up to my tv for playing some games and playing media. I would like to find a board with both HDMI and standard RCA hookups that would be powerful enough to play full HD video (if it has an embedded cpu). I came across the VIA VB8003 which seemed to fit my needs but it doesn't seem to be widely available. Any help would be greatly appreciated, thank you.

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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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  • 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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  • 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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  • How to prevent Vista from entering Sleep mode when certain conditions exist?

    - by idyllhands
    Mainly, I would like to prevent my laptop from entering Sleep mode when I am playing music or streaming video. This is on my laptop, so another option that would work would be to prevent sleep mode whenever the laptop is plugged into my tv via HDMI (ie when HDMI port is in use). Or prevent sleep mode whenever there is audio playing.. I am soon upgrading to Windows 7, so solutions using 7 would be great also/instead. Thanks!

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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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  • Flash FLVPlayback states

    - by rob5408
    I'm writing my own class to manage a skin for an FLVPlayback component. It works 90% of the time, but sometimes the state get really messed up. Specifically, the video is playing, meaning I can see it play and the VideoEvent.PLAYHEAD_UPDATE event is firing, but when I poll the FLVPlayback component about its playing property, it returns false. I assume this may be because 'buffering' is kind of a subset of 'playing', but I cannot confirm this in the documentation. I guess another way to ask this question is, "Does the FLVPlayback component ever buffer while it is in a stopped state?"

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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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  • Querying current number of touches on screen without using events on iPhone

    - by nikhil
    I have an application that starts playing a sound when user touches the uiview and changing to different tones as the user slides the finger on the screen. The sound stops when the user lifts the finger. I am using the touchesBegan, Moved and Ended Events for this. My problem is touches Ended (and/or cancelled) is sometimes not fired properly and the sound keeps playing even after the finger is lifted from screen. So as a workaround I would like to implement a timer that would check for the number of touches on the screen and if it is zero it will check and stop the audioplayer if playing. I have been searching for some code that could get me the number of touches like UITouch *touches=[self getAllTouchesonScreen]; or something :)

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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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  • 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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  • Polling versus socket servers for online Flash games

    - by justin
    Hi, I want to make an online flash game, it will have social features but the gameplay will be primarily single-player. For example, no two players will appear on the screen at once, the social interaction will be through asynchronous messages, there won't be real-time chat or anything. Much of the logic would happen in the client, the server would validate the client logic, but it wouldn't need to be totally synchronous, which is why I'm thinking polling might be satisfactory. I have read in many places that socket servers can be more efficient than using polling for online games, but is that mainly a consideration for games that are more multi-player with more mult-player interactions than the game I have descriebed? If many users are playing online at the same time, but each playing a relatively isolated game, and not interacting to in real-time with each players, could polling be okay, or would using sockets be advisable no matter what if you have an online game that you envision many people playing at the same time? Thanks!

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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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  • how to change orientation of the iphone /ipod ?

    - by Balraj
    HI All i am using the device orientation for some functionality like as in stating the as we run the app it should be in the landscape mode and in this mode if we orientate the phone in 90 angle than it should play some video than problem comes video play by default in the landscape mode but we need it in the portrait mode than for playing video in the portrait mode we put the code [ mPlayer setOrientation:UIDeviceOrientationPortrait animated:NO]; like this even this method show a warning but it do work for playing the video in the portrait mode. now what is the problem :-as we start the app the screen we were showing in the starting of the app just comes and disaapper and it just starts playing the video and not come to that screen we even use the default rotation function to make it usable but it doesn't work any suggestion thanks Balraj Verma

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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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  • Using system Sound to play sounds

    - by Shoaibi
    Here is the code: -(void)stop { NSLog(@"Disposing Sounds"); AudioServicesDisposeSystemSoundID (soundID); //AudioServicesRemoveSystemSoundCompletion (soundID); } static void completionCallback (SystemSoundID mySSID, void* myself) { NSLog(@"completion Callback"); } - (void) playall: (id) sender { [self stop]; AudioServicesAddSystemSoundCompletion (soundID,NULL,NULL, completionCallback, (void*) self); OSStatus err = kAudioServicesNoError; NSString *aiffPath = [[NSBundle mainBundle] pathForResource:@"slide1" ofType:@"m4a"]; NSURL *aiffURL = [NSURL fileURLWithPath:aiffPath]; err = AudioServicesCreateSystemSoundID((CFURLRef) aiffURL, &soundID); AudioServicesPlaySystemSound (soundID); NSLog(@"Done Playing"); } Output: Disposing Sounds Done Playing In actual no sound gets play at all and completion call back isn't called as well. Any idea what could be wrong here? I want to stop any previous sound before playing current.

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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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