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• VBA olMailItem .display, recording when/if sent manually

  - by ExcelCyclist

  My code to displays a message with basic subject, body, attachment. Next the user manually updates and customizes the message and should send it. I want to record when (if) the email is sent. Is this possible or any tips? My environment is Office 2007 with an excel based macro going to Outlook. [Excerpt] Dim OutApp As Outlook.Application Dim OutMail As Outlook.MailItem Set OutApp = CreateObject("Outlook.Application") OutApp.Session.Logon Set OutMail = OutApp.CreateItem(olMailItem) With OutMail .To = Email '.CC = .Subject = Subj .BodyFormat = olFormatHTML .Body = Msg '.HTMLBody = Msg If Not FileAttach = vbNullString Then .Attachments.Add (PathFile) .Display End With

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• Recording SELECT statements in PostgreSQL 8.4

  - by David Anniwell

  Hi All I've got a table which contains sensitive data and according to data protection policy we have to keep a record of every read/write of the data including a row identifier and the user who accessed the table. The writing is no issue using triggers but obviously triggers aren't supported for SELECT statements. What's the best method of doing this? I've looked at rules but I can't get them to INSERT into a table, and I've tried logging every query but this doesn't seem to log SELECT statements. Ideally for security I'd like to keep the log within a table on the database but logging to a file is fine too. Thanks David

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• Smooth screen recording software

  - by user85658

  Hello everyone, I am preparing for my senior thesis presentation. I'd like to back myself up in case there is no internet connection available. Therefore I want to record a video showing the functionality of my software. I have tried Camtasia but all the smooth and slick animations I've created, do not look that appealing. Is there an alternative. Something that will capture my screen 1 to 1, or near that. Camtasia is great, but it does not serve the purpose. Any help would be greatly appreciated. Best Regards, Kiril

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• Recording a screencast of an Android app using the emulator

  - by hgpc

  Unlike the iPhone simulator, the Android emulator doesn't look like an Android device. If you have to create a screencast or promotional video of your Android application, the default skin of the emulator is no good. Is there any way to configure the emulator to look like an Android device? If you have dealt with this already, what other things do you recommend taking into account to record an emulator screencast? Not a programming question per se, but an useful question for Android developers.

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• How to change default audio input device programatically

  - by f34r

  I am looking for a way to set/change default input device inside my application. I have several different recording devices and it is very anoying to go into the control panel and change default recording device. I was looking around and I did not find anything that could help me with the problem. Application is written in c# and it is targeted for Windows Vista / Windows 7.

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• Live javascript debugging by recording function calls and parameters

  - by Jenko

  Is there a debugging system that allows me to record javascript function calls and their parameters as they occur? this would allow me to trace and debug applications in live/client situations without the performance hit due to manual logging. Edit: I'm not talking about manually calling functions using a 'console' window and viewing the results, or manually adding 'trace' or 'log' commands into my javascript. I need it to work with any running javascript.

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• Asterisk auto Call recording

  - by Manjoor

  We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100). Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside). Here is the diagram |------|--------------------------------- | |--------------24 Lines ---------- Other clasic Phones PRI------ | PBX |--------------------------------- | | | | | |-----------|---------| | |--8 lines--| |--------- | |-----------|Asterisk |---------- 50 SIP phone |------| | |---------- |---------|---------- Is there a simple way to do this?

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• C# Flash ActiveX - Recording

  - by Dremation

  I wrote an application that monitors live streams from various sites. The application has done well but theres a feature that is highly requested. And that's the ability to record the stream. I'm current using the Adobe Flash ActiveX control to stream the videos from the various sites. IS there a way to record the stream? Whether it be with Flash ActiveX or another framework/control that you may know of.

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• Trouble recording unique regex output to array in perl

  - by Structure

  The goal of the following code sample is to read the contents of $target and assign all unique regex search results to an array. I have confirmed my regex statement works so I am simplifying that so as not to focus on it. When I execute the script I get a list of all the regex results, however, the results are not unique which leads me to believe that my manipulation of the array or my if (grep{$_ eq $1} @array) { check is causing a problem(s). #!/usr/bin/env perl $target = "string to search"; $inc = 0; $once = 1; while ($target =~ m/(regex)/g) { #While a regex result is returned if ($once) { #If $once is not equal to zero @array[$inc] = $1; #Set the first regex result equal to @array[0] $once = 0; #Set $once equal to zero so this is not executed more than once } else { if (grep{$_ eq $1 } @array ) { #From the second regex result, check to see if the result is already in the array #If so, do nothing } else { @array[$inc] = $1; #If it is not, then assign the regex search result to the next unused position in the array in any position. $inc++; #Increment to next unused array position. } } } print @array; exit 0;

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• Save response from certain WEB resources while recording scenario

  - by jdevelop

  I need to create scenario for user interaction with single-page WEB application. The application does lots of AJAX calls in order to authenticate user and get user data. So I created simple scenario with HTTP Test Script Recorder and tried to record my script. Everything went well, however I noticed that whilst request data is recorder properly, the response data is not recorder at all. I tried to enable Add assertions and Regex matching - but that didn't work as well. Can you please advice how do I record response texts as well?

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• Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

  - by jdl2003

  I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

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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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