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  • How to show or direct a business analyst to a data modelling subject?

    - by AaronLS
    Our business analysts pushed hard to collect data through a spreadsheet. I am the programmer responsible for importing that data. Usually when they push hard for something like this, I never know how well it will work out until a few weeks later when I have time assigned to work on the task of programming the import of the data. I have tried to do as much as possible along the way, named ranges, data validations, etc. But I usually don't have time to take a detailed look at all the data and compare to the destination in the database to determine how well it matches up. A lot of times there will be maybe a little table of items that somehow I have to relate to something else in the database, but there are not natural or business keys present that would allow me to do so. Make the best of this, trying to write something that can compare strings and make a best guess at it and then go through the effort of creating interfaces for a user to match the imported data to the destination. I feel like if the business analyst was actually creating a data model, they would be forced to think about these relationships, and have an appreciation for the need of natural or business keys to be part of the spreadsheet for the purposes of smoothly importing the data. The closest they come to business analysis is a big flat list of fields, and that would be fine if it were like any other data dictionary and include data types+relationships, but it isn't. They are just a bunch of names. No indication of what type of data they might hold, and it is up to me to guess. When I have pushed for more detail, they say that it is just busy work. How can I explain the importance of data modelling? How can I tell them what it is and how to do it? It feels impossible, because they don't have an appreciation for its importance. They do however, usually have an interest in helping out in whatever way they can, it's just this in particular has never gotten a motivated response.

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  • Tender vs. Requirements vs. Solution Design

    - by Tom Tom
    Conventionally, which of the above documents is deemed to hold the most weight when it comes to system acceptance? I recently had a conversation along these lines: It was argued that the initial requirements / tender documentation should be used to determine system acceptance. It was said that the solution design only serves to describe the way in which the system will solve the problem, not the problem it will solve. Furthermore, it was argued that if requirements are missed during solution design, the requirements should be referenced during system acceptance and that if any requirements were missed then the original tender should be referenced. Conversely, I suggested that - while requirements may be based on the original tender - they supersede it once agreed with the stakeholders. Furthermore, during solution design, analysis is performed to address and refine these initial requirements, translating them into a system capable of meeting the actual requirements. Once signed off by the relevant users, this solution design should absolutely represent the requirements (by virtue of the fact that it's designed upon them) but actually supersedes them as the basis for system acceptance. Is one of the above arguments more valid than the other?

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  • How to show or direct a business analyst to do data modelling?

    - by AaronLS
    Our business analysts pushed hard to collect data through a spreadsheet. I am the programmer responsible for importing that data. Usually when they push hard for something like this, I never know how well it will work out until a few weeks later when I have time assigned to work on the task of programming the import of the data. I have tried to do as much as possible along the way, named ranges, data validations, etc. But I usually don't have time to take a detailed look at all the data and compare to the destination in the database to determine how well it matches up. A lot of times there will be maybe a little table of items that somehow I have to relate to something else in the database, but there are not natural or business keys present that would allow me to do so. Make the best of this, trying to write something that can compare strings and make a best guess at it and then go through the effort of creating interfaces for a user to match the imported data to the destination. I feel like if the business analyst was actually creating a data model, they would be forced to think about these relationships, and have an appreciation for the need of natural or business keys to be part of the spreadsheet for the purposes of smoothly importing the data. The closest they come to business analysis is a big flat list of fields, and that would be fine if it were like any other data dictionary and include data types+relationships, but it isn't. They are just a bunch of names. No indication of what type of data they might hold, and it is up to me to guess. When I have pushed for more detail, they say that it is just busy work. How can I explain the importance of data modelling? How can I tell them what it is and how to do it? It feels impossible, because they don't have an appreciation for its importance. They do however, usually have an interest in helping out in whatever way they can, it's just this in particular has never gotten a motivated response.

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  • How many copies are needed to enlarge an array?

    - by user10326
    I am reading an analysis on dynamic arrays (from the Skiena's algorithm manual). I.e. when we have an array structure and each time we are out of space we allocate a new array of double the size of the original. It describes the waste that occurs when the array has to be resized. It says that (n/2)+1 through n will be moved at most once or not at all. This is clear. Then by describing that half the elements move once, a quarter of the elements twice, and so on, the total number of movements M is given by: This seems to me that it adds more copies than actually happen. E.g. if we have the following: array of 1 element +--+ |a | +--+ double the array (2 elements) +--++--+ |a ||b | +--++--+ double the array (4 elements) +--++--++--++--+ |a ||b ||c ||c | +--++--++--++--+ double the array (8 elements) +--++--++--++--++--++--++--++--+ |a ||b ||c ||c ||x ||x ||x ||x | +--++--++--++--++--++--++--++--+ double the array (16 elements) +--++--++--++--++--++--++--++--++--++--++--++--++--++--++--++--+ |a ||b ||c ||c ||x ||x ||x ||x || || || || || || || || | +--++--++--++--++--++--++--++--++--++--++--++--++--++--++--++--+ We have the x element copied 4 times, c element copied 4 times, b element copied 4 times and a element copied 5 times so total is 4+4+4+5 = 17 copies/movements. But according to formula we should have 1*(16/2)+2*(16/4)+3*(16/8)+4*(16/16)= 8+8+6+4=26 copies of elements for the enlargement of the array to 16 elements. Is this some mistake or the aim of the formula is to provide a rough upper limit approximation? Or am I missunderstanding something here?

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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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  • 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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  • 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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  • 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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  • 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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  • Turn off all sounds from websites

    - by David Oneill
    Often, I am listening to music of my choosing. Is there a way to preemptively turn off all sounds originating from websites? I don't want to click the 'mute' button once the page loads. And sometimes, it won't even have a mute. :-/ I use Chromium and FireFox. ~~EDIT~~ I use XFCE, so my menu options are different. Is this a gnome-specific utility? Or, what is the command for this utility?

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  • The Best Text to Speech (TTS) Software Programs and Online Tools

    - by Lori Kaufman
    Text to Speech (TTS) software allows you to have text read aloud to you. This is useful for struggling readers and for writers, when editing and revising their work. You can also convert eBooks to audiobooks so you can listen to them on long drives. We’ve posted some websites here where you can find some good TTS software programs and online tools that are free or at least have free versions available. 8 Deadly Commands You Should Never Run on Linux 14 Special Google Searches That Show Instant Answers How To Create a Customized Windows 7 Installation Disc With Integrated Updates

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  • IrrKlang with Ogre

    - by Vinnie
    I'm trying to set up sound in my Ogre3D project. I have installed irrKlang 1.4.0 and added it's include and lib directories to my projects VC++ Include and Library directories, but I'm still getting a Linker error when I attempt to build. Any suggestions? (Error 4007 error LNK2019: unresolved external symbol "__declspec(dllimport) class irrklang::ISoundEngine * __cdecl irrklang::createIrrKlangDevice(enum irrklang::E_SOUND_OUTPUT_DRIVER,int,char const *,char const *)" (_imp?createIrrKlangDevice@irrklang@@YAPAVISoundEngine@1@W4E_SOUND_OUTPUT_DRIVER@1@HPBD1@Z) referenced in function "public: __thiscall SoundManager::SoundManager(void)" (??0SoundManager@@QAE@XZ)

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  • music for an arcade game?

    - by user717572
    I'm thinking about music for my brick breaker game, but I don't know how to choose any. If I'd make a loop from a few seconds, I think it would get annoying very quickly. I also found some longer length tracks (about 2 minutes), but when this is over, it's going to be repeated anyway, just like when you'd select a new level, you'd have to listen to the same beginning of the song again. I can't put an hour of music in my application, so what would you recommend I'd do for the music?

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  • How to disable Alert volume from the command line?

    - by Bryce
    There is an option in the Sound Preferences dialog, Sound Effects tab, to toggle Alert volume 'mute'. It works and suffices for my needs to disable the irritating system beep/bell. However, I reinstall systems a LOT for testing purposes and would like to set this setting in a shell script so it's off without having to fiddle with a GUI. But for the life of me I can't seem to find where this can be toggled via a command line tool. I've scanned through gconf-editor, pulseaudio's pacmd, grepped through /etc, even dug through the gnome-volume-control source code, but I am not seeing how this can be set. I gather that gnome-volume-control has changed since a few releases ago. Ideas?

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  • How to disable Alert volume from the command line in Natty?

    - by Bryce
    There is an option in the Sound Preferences dialog, Sound Effects tab, to toggle Alert volume 'mute'. It works and suffices for my needs to disable the irritating system beep/bell. However, I reinstall systems a LOT for testing purposes and would like to set this setting in a shell script so it's off without having to fiddle with a GUI. But for the life of me I can't seem to find where this can be toggled via a command line tool. I've scanned through gconf-editor, pulseaudio's pacmd, grepped through /etc, even dug through the gnome-volume-control source code, but I am not seeing how this can be set. I gather that gnome-volume-control has changed since a few releases ago. I'm using Natty fwiw. Ideas?

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  • How can I automatically mute the volume at every boot?

    - by ændrük
    Sometimes I forget to enable mute before shutting down my laptop. Can I set it up to be muted by default every time Ubuntu boots, before the login screen is displayed? When I try DoR's suggestion of sudo alsactl store, the settings stored in /var/lib/alsa/asound.state are lost on the next reboot. Something is using this file to automatically save the current volume settings every time I reboot.

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