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• Type casting Collections using Conversion Operators

  - by Vyas Bharghava

  The below code gives me User-defined conversion must convert to or from enclosing type, while snippet #2 doesn't... It seems that a user-defined conversion routine must convert to or from the class that contains the routine. What are my alternatives? Explicit operator as extension method? Anything else? public static explicit operator ObservableCollection<ViewModel>(ObservableCollection<Model> modelCollection) { var viewModelCollection = new ObservableCollection<ViewModel>(); foreach (var model in modelCollection) { viewModelCollection.Add(new ViewModel() { Model = model }); } return viewModelCollection; } Snippet #2 public static explicit operator ViewModel(Model model) { return new ViewModel() {Model = model}; } Thanks in advance!

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• C++ conversion operator between types in other libraries

  - by Dave

  For convenience, I'd like to be able to cast between two types defined in other libraries. (Specifically, QString from the Qt library and UnicodeString from the ICU library.) Right now, I have created utility functions in a project namespace: namespace MyProject { const icu_44::UnicodeString ToUnicodeString(const QString& value); const QString ToQString(const icu_44::UnicodeString& value); } That's all well and good, but I'm wondering if there's a more elegant way. Ideally, I'd like to be able to convert between them using a cast operator. I do, however, want to retain the explicit nature of the conversion. An implicit conversion should not be possible. Is there a more elegant way to achieve this without modifying the source code of the libraries? Some operator overload syntax, perhaps?

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• Mysterious different conversion to string[] of seemingly same input data

  - by Roman Kuzmin

  During investigation of some problem I found that the reason was unexpected different conversion to string[] of seemingly same input data. Namely, in the code below two commands both return the same two items File1.txt and File2.txt. But conversion to string[] gives different results, see the comments. Any ideas why is it? This might be a bug. If anybody also thinks so, I’ll submit it. But it would nice to understand what’s going on and avoid traps like that. # *** WARNING # *** Make sure you do not have anything in C:\TEMP\Test # *** The code creates C:\TEMP\Test with File1.txt, File2.txt # Make C:\TEMP\Test and two test files $null = mkdir C:\TEMP\Test -Force 1 | Set-Content C:\TEMP\Test\File1.txt 1 | Set-Content C:\TEMP\Test\File2.txt # This gets just file names [string[]](Get-ChildItem C:\TEMP\Test) # This gets full file paths [string[]](Get-ChildItem C:\TEMP\Test -Include *) # Output: # File1.txt # File2.txt # C:\TEMP\Test\File1.txt # C:\TEMP\Test\File2.txt

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• Parametrised and Conversion Constructors

  - by viswanathan

  Is there any difference between parametrised constructor and conversion constructor. If so what is it?

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• Sending microphone input over Remote Desktop 7.0

  - by Taylor Price

  I am using Remote Desktop 7 (the new version that came out with Windows 7) to control a Windows XP Pro machine. I have selected "Record from this computer" in the Remote Audio settings. When I connect to the machine, go to the control panel, open the sound panel, and go to the audio tab, I find that the default sound playback device is "Microsoft RDP Audio Driver". However, there is no default sound recording device. As expected, my IP phone thinks there is no recording device. If I am sitting in front of the computer with a mic plugged in, it works just fine. Has anybody else been able to get this work appropriately? Is there anything that I have to setup on the XP machine to get this working? Thanks in advance. Edit: As John T pointed out below, you have to be connecting to a Windows 7 Enterprise or Ultimate machine for this to work. I've also found out that Multi-monitor support has the same requirement.

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• QuickTime Player sounds much better than iTunes

  - by Gene Goykhman

  I am playing a 320 kpbs encoded music MP3 in iTunes and the sound is substantially worse than the exact same file played back in QuickTime Player (Max OS X 10.8.5). I have maxed out system volume and iTunes playback volume. I have disabled all the audio processing features in iTunes (equalization, sound enhancer, etc.) The audio coming from iTunes still sounds resampled and/or processed, whereas QuickTime Player appears to be playing it "as is". Even when I Get Info on the MP3 file in Finder and play it back directly from the Get Info window it sounds good. It's just iTunes that seems to be mangling the song. I can notice a difference on virtually all my music, so it's not just one particular MP3. I suspect the issue is that iTunes is doing some kind of audio processing but I can't find a way to turn it off. This is the newest iTunes (11.1), but the problem has probably been going on for a while... I just switched to decent earbuds and started noticing the difference. What's the best way to force iTunes to play back the file as-is, or as close as possible to how QuickTime Player/Finder would play it?

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• Speakers silent, headphones work in Ubuntu 9.04

  - by CarlF

  I'm running Ubuntu 9.04. Worked fine for months, then I rebooted yesterday after weeks of continuous operation. Now audio won't play through the speakers. The USB headset works fine, but the Conexant audio (CX20549) does not. Weirdly, it thinks it's playing. pavumeter shows appropriate levels, volume looks OK in alsamixer, but no sound. I did find this page: http://www.eugeneteplitsky.com/fixing-silent-pulseaudio-in-ubuntu-9-04/ Unfortunately the advice there doesn't help me. For one thing, the syntax for the alsa-base.conf file is apparently not actually documented anywhere. For another, my chipset isn't listed in the kernel.org docs he links to! EDIT: would upgrading to 9.10 help? Is there a major change in the audio subsystem between 9.04 and 9.10? Any suggestions? EDIT 2: This is stranger than I thought. Sound works normally in Xine, but is silent in Audacity, VLC and mplayer. What the?

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• Problems with 5.1 digital out on Ubuntu 12.04

  - by user895319

  I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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• 5.1 Surround Channels are Jumbled

  - by stickynips

  I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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• How to get the mic on the Creative X-Mod soundcard working correctly?

  - by Nyamiou The Galeanthrope

  Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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• Renoise and dssi and jack

  - by Bojan

  This may be a little complicated, but, is jack a necessity? I mean, i use renoise, and, since i dont have the need for low latencies, do i really need to use it? My basic setup ( or workflow ) is that i use csound to render stuff to wav, then import it as a sample in renoise. That goes with field recordings, my own samples, etc. So, i dont need ultra low latencies, and i dont need to patch "cords", but i want to use dssi plugins, and dssi-vst. What would be something of a minimum requirements of apps that should work. Can renoise load dssi-vst plugins by itself or do i need to use jack to patch thru or something third, i tried to read lot of articles but i got lost in the different setups...

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• Reformating xml document

  - by Joseph Reeves

  I have an xml document in the format below: <key>value</key> <key>value</key> <key>value</key> But need to convert it to the following: <tag k='key' v='value' /> <tag k='key' v='value' /> <tag k='key' v='value' /> The original xml file is roughly 20,000 lines long, so I'm keen to automate as much as possible! I've looked at xmlstarlet, but drew a blank with it. Presumably it would be a good place to start though? Help gratefully received, thanks.

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• How do I reduce the screen and file size of a recorded video, plus convert to FLV?

  - by Volomike

  I have used gtk-recordMyDesktop to make a video as an OGV file using the default settings. I need to do 3 things: How can I reduce the screen resolution (height and width) so that it can fit into a smaller video size on my website? How can I pull out like every third frame so that the file size is not so large, yet not mess up the sound? Not all Windows IE users can view OGV files. How can I convert to FLV (or, as a fallback, MP4) so that I can share on my blog?

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• How do you convert many files from .xlsx to .xls ?

  - by David Oneill

  What is a way to convert a batch of .xlsx files to .xls format? I would prefer it to be a command-line solution, but anything is better than opening each manually, and manually saving in the new format. ~~Edit~~ So is there a way to get around that error? errored: Leaking python objects bridged to UNO for reason pyuno runtime is not initialized, (the pyuno.bootstrap needs to be called before using any uno classes) python: tpp.c:63: __pthread_tpp_change_priority: Assertion `new_prio == -1 || (new_prio >= __sched_fifo_min_prio && new_prio <= __sched_fifo_max_prio)' failed. Aborted

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• Beat detection, weird detection

  - by Quincy

  I made this soundanalyzer class to detect beats in songs : // put it on pastebin for the big size, will put it here if people rather want that. pastebin.com/8PdgZPP3 but for some reason its only detecting beats from 637 sec to around 641(sec) and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates and it seems as its assigning a beat to each instant energy value in between those values. Its modeled after this : http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats properly register ?

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• Computer SOMETIMES recognizes when headphones are plugged in.

  - by rcrobot

  Whenever I plug my headphones into my computer's front headphone jack, I get a weird situation. Sometimes, the computer will recognize the headphones and work properly. But other times, the computer will play sound through both the headphones and my monitor's speaker. When this happens, the sound section of the system settings does not list the headphones. I can fix the issue temporarily by wiggling the headphone port, but if it gets wiggled the wrong way again, then the issue returns. My PC's case is a Rosewill Challenger. I have tried multiple headphones and the same issue is there. I suspect that this might be a hardware related issue, but if there is any way to fix it with software, that would be helpful. This is what it looks like when everything is working properly: This happens when I wiggle the headphone port. I can quickly switch between these two by doing so:

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• Converting a video file in arbitrary file format into MPEG4/H.264?

  - by knorv

  I want to convert a large number of video files in various formats into .mp4 files (container MPEG-4, codec H.264). I want to do this on an Ubuntu machine, using only command-line tools and I'm willing to install packages from main, restricted, universe and multiverse. Ideally I'd like to be able to do ... for VIDEO_FILE in *; do some_conversion_program $VIDEO_FILE $VIDEO_FILE.mp4 done ... and have all my video files in .mp4 format with container MPEG-4 and codec H.264. How would you tackle this problem on an Ubuntu machine? What packages do I need to install?

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• How to correctly Dispose a SourceVoice once its finished

  - by clamp

  i am starting to play a sound with XAudio2 and SourceVoice and once its finished, it should be correctly disposed to not have any leaks. i was expecting it to be something like this: sourceVoice.Start(); sourceVoice.StreamEnd += delegate { if (!sourceVoice.IsDisposed) { sourceVoice.DestroyVoice(); sourceVoice.Dispose(); } }; but that crashes with a read access violation in native code deep in XAudio2.dll which i cant debug.

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• How to convert rmvb to mp4? (or, why ffmpeg doesn't work?)

  - by Tom Brito

  Hi, I'm trying to use this script to convert an rmvb video to mp4, but I'm having problems with the ffmpeg. In apt-get there's only ffmpeg0 and ffmpeg-dev, I installed both, but the script doesn't work, it's saying that ffmpeg was not found. Any hint on this? --update The script I'm talking about: #!/bin/bash tipo=$1 arqv=$2 resolucao=$3 tipoarq=$4 help() { clear echo "Convertor de Vídeos para MP4" echo "Parametro 1 = Tipo: (A - Arquivo/D - Diretório)" echo "Parametro 2 = Arquivo/Caminho" echo "Parametro 3 = Resolução" echo "Parametro 4 = Tipo de Arquivos de Entrada (rmvb, avi, mpeg)" } if [ "$tipo" = "" -o "$arqv" = "" -o "$resolucao" = "" -o "$tipoarq" = "" ]; then help; exit fi if [ "$tipo" = "D" ]; then count=`ls "$arqv"/*.$tipoarq | wc -l` else count=1 fi echo "$count arquivos encontrados para converter." x=0 while [ ! $x -ge $count ]; do x=`echo $x + 1 | bc` if [ "$tipo" = "D" ]; then nome=`ls "$arqv"/*.$tipoarq | head -n $x | tail -n 1` else nome=$arqv fi echo "Convertendo $nome ..." ffmpeg -i "$nome" -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -cmp 2 -subcmp 2 -s $resolucao "`echo $nome | sed "s/\.$tipoarq//g"`".mp4 done exit --update Using WinFF it gives: Unknown encoder 'libx264' I've installed both the existing packages libx264-67 and libx264-dev, but none solved. Looking for more alternatives...

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• Playing a Song causing WP7 to crash on phone, but not on emulator

  - by Michael Zehnich

  Hi there, I am trying to implement a song into a game that begins playing and continually loops on Windows Phone 7 via XNA 4.0. On the emulator, this works fine, however when deployed to a phone, it simply gives a black screen before going back to the home screen. Here is the rogue code in question, and commenting this code out makes the app run fine on the phone: // in the constructor fields private Song song; // in the LoadContent() method song = Content.Load<Song>("song"); // in the Update() method if (MediaPlayer.GameHasControl && MediaPlayer.State != MediaState.Playing) { MediaPlayer.Play(song); } The song file itself is a 2:53 long, 2.28mb .wma file at 106kbps bitrate. Again this works perfectly on emulator but does not run at all on phone. Thanks for any help you can provide!

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• MEncoder Install on Ubuntu

  - by Tauqeer Ahmad

  I am writing this after checking almost all the posts but none of those solved my problem. I am trying to install mencoder to process some videos but there are strange errors coming. For examples when I try sudo apt-get install mencoder the following errors comes out: Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: mencoder : Depends: mplayer Depends: libasound2 (> 1.0.24.1) Depends: libavcodec53 (>= 4:0.8~beta2-2) but it is not installable or libavcodec-extra-53 (>= 4:0.8~beta2-2) but it is not going to be installed Depends: libavformat53 (>= 4:0.8~beta2-2) but it is not installable or libavformat-extra-53 (>= 4:0.8~beta2-2) but it is not going to be installed Depends: libcdparanoia0 (>= 3.10.2+debian) but it is not installable Depends: libenca0 (>= 1.9) but it is not installable Depends: libfontconfig1 (>= 2.8.0) but it is not installable Depends: libgif4 (>= 4.1.4) but it is not installable Depends: libjpeg8 (>= 8c) but it is not installable Depends: liblzo2-2 but it is not installable Depends: libsmbclient (>= 3.0.24) but it is not installable Depends: libspeex1 (>= 1.2~beta3-1) but it is not installable Depends: libtheora0 (>= 1.0) but it is not installable E: Unable to correct problems, you have held broken packages. Can anyone help to solve this issue. I tried to find static builds of MEncoder but could not.

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• How to compile FFmpeg with x265 support?

  - by Levan

  Today I found out that x265 is already present in ffmpeg so I compiled ffmpeg with this guide Sadly libx265 did not work on ubuntu, however on windows I tried the same thing with zeranoe ffmpeg build and it worked without a problem. So do you think i did something wrong or it is not yet implemented in linux build (using that guide)? The results of the command ffmpeg -codecs | grep -i hevc show: ffmpeg version 2.1.git Copyright (c) 2000-2014 the FFmpeg developers built on Feb 19 2014 19:00:17 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9) configuration: --prefix=/home/levan/ffmpeg_build --extra-cflags=-I/home/levan/ffmpeg_build/include --extra-ldflags=-L/home/levan/ffmpeg_build/lib --bindir=/home/levan/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 64.100 / 52. 64.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 1.102 / 4. 1.102 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 D.V.L. hevc H.265 / HEVC (High Efficiency Video Coding) Thank you for your time

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• How do I get my ART USB Dual Pre preamp to work?

  - by Zach

  I am using Audacity. I have an ART USB Dual Pre preamp. Ubuntu is not recognizing it whatsoever. I am able to record in Audacity, but it is using the mic that is built into my computer (which is a compaq Presario CQ50) instead of the one plugged into the preamp. How do I get Ubuntu to recognize the preamp that is plugged into my computer? Something tells me it has to do with the installation of the preamp software. It came with a installation CD, but when I go to "install", the nothing happens. I can view what is on the CD, but there is no installing of anything. Please help!

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