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  • Burn 24/96 flac files to play on standalone player

    - by takeshin
    I have vinyl record rip in 24/96 flac format. Each track is almost 200 MB big, so the album won't fit on CD. How to burn these files on a DVD to play with the same quality on standalone DVD player? My player supports SACD, DVD Audio and DVD video as well. My OS is Ubuntu Lucid (preferred), but I have also WinXp with Nero installed. BTW, is there any difference between DVD+ and DVD- for audio?

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  • Replace sound in another YouTube video

    - by Tom
    I have received permission from someone to translate the audio in their movies. The problem I am facing is that the video quality is quite poor and the author does not have the original videos any more. How can I replace the audio in the YouTube videos without further degrading the quality of the videos? Thanks, Tom

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  • Any plugins for Skype that support "Soundboard" usage?

    - by Axxmasterr
    I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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  • Debian sound on hdmi instead of jack

    - by Hans de Jong
    I installed debian (gnome) and i can't get my sound working. When i use inxi -A i get the following result: Audio: Card-1: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series] driver: snd_hda_intel Card-2: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) driver: snd_hda_intel Sound: Advanced Linux Sound Architecture ver: 1.0.24 My feeling tells me my sound output is on the HDMI instead of my jackplug on my motherboard. How can i change this to my motherboard sound output?

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  • Which connector do I need for a "line level" subwoofer?

    - by Ben Brocka
    I've got a separate pair of speakers and I'm looking at adding a subwoofer (this, specifically). I noticed on the detail page it's inputs are listed as such: Inputs: Speaker level, line level If I'm not mistaken "line level" are the standard 3.5 audio jacks on your motherboard/sound card, right? My motherboard has the standard 6 ports for sound, if I get a subwoofer like this can I simply plug the input into the orange 3.5 jack? My audio software supports up to 7.1 so software-wise, 2.1 wouldn't be a problem.

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  • Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

    - by Roy Rico
    I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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  • Changing default playback device on Windows 8

    - by emartel
    Previously, on Vista and Windows 7, changing the Default Playback device would occur instantly. For example, audio is coming out of my speakers, I right click the Volume Control, click Playback Devices then I select another device and click Set Default. Audio would be transferred immediately. Unfortunately, now, with Windows 8, I need to kill whatever process what outputting sound, and restart it for the change to take effect. Is there something that can be done about it so that changes are taken into account immediately?

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  • Distributed and/or Parallel SSIS processing

    - by Jeff
    Background: Our company hosts SaaS DSS applications, where clients provide us data Daily and/or Weekly, which we process & merge into their existing database. During business hours, load in the servers are pretty minimal as it's mostly users running simple pre-defined queries via the website, or running drill-through reports that mostly hit the SSAS OLAP cube. I manage the IT Operations Team, and so far this has presented an interesting "scaling" issue for us. For our daily-refreshed clients, the server is only "busy" for about 4-6 hrs at night. For our weekly-refresh clients, the server is only "busy" for maybe 8-10 hrs per week! We've done our best to use some simple methods of distributing the load by spreading the daily clients evenly among the servers such that we're not trying to process daily clients back-to-back over night. But long-term this scaling strategy creates two notable issues. First, it's going to consume a pretty immense amount of hardware that sits idle for large periods of time. Second, it takes significant Production Support over-head to basically "schedule" the ETL such that they don't over-lap, and move clients/schedules around if they out-grow the resources on a particular server or allocated time-slot. As the title would imply, one option we've tried is running multiple SSIS packages in parallel, but in most cases this has yielded VERY inconsistent results. The most common failures are DTExec, SQL, and SSAS fighting for physical memory and throwing out-of-memory errors, and ETLs running 3,4,5x longer than expected. So from my practical experience thus far, it seems like running multiple ETL packages on the same hardware isn't a good idea, but I can't be the first person that doesn't want to scale multiple ETLs around manual scheduling, and sequential processing. One option we've considered is virtualizing the servers, which obviously doesn't give you any additional resources, but moves the resource contention onto the hypervisor, which (from my experience) seems to manage simultaneous CPU/RAM/Disk I/O a little more gracefully than letting DTExec, SQL, and SSAS battle it out within Windows. Question to the forum: So my question to the forum is, are we missing something obvious here? Are there tools out there that can help manage running multiple SSIS packages on the same hardware? Would it be more "efficient" in terms of parallel execution if instead of running DTExec, SQL, and SSAS same machine (with every machine running that configuration), we run in pairs of three machines with SSIS running on one machine, SQL on another, and SSAS on a third? Obviously that would only make sense if we could process more than the three ETL we were able to process on the machine independently. Another option we've considered is completely re-architecting our SSIS package to have one "master" package for all clients that attempts to intelligently chose a server based off how "busy" it already is in terms of CPU/Memory/Disk utilization, but that would be a herculean effort, and seems like we're trying to reinvent something that you would think someone would sell (although I haven't had any luck finding it). So in summary, are we missing an obvious solution for this, and does anyone know if any tools (for free or for purchase, doesn't matter) that facilitate running multiple SSIS ETL packages in parallel and on multiple servers? (What I would call a "queue & node based" system, but that's not an official term). Ultimately VMWare's Distributed Resource Scheduler addresses this as you simply run a consistent number of clients per VM that you know will never conflict scheduleing-wise, then leave it up to VMWare to move the VMs around to balance out hardware usage. I'm definitely not against using VMWare to do this, but since we're a 100% Microsoft app stack, it seems like -someone- out there would have solved this problem at the application layer instead of the hypervisor layer by checking on resource utilization at the OS, SQL, SSAS levels. I'm open to ANY discussion on this, and remember no suggestion is too crazy or radical! :-) Right now, VMWare is the only option we've found to get away from "manually" balancing our resources, so any suggestions that leave us on a pure Microsoft stack would be great. Thanks guys, Jeff

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  • How to fix "Sub-process /usr/bin/dpkg returned an error code (1)" when installing and upgrading packages?

    - by soum
    I am getting this error whenever tring to install or update anything: "Sub-process /usr/bin/dpkg returned an error code (1)" I need help, as I cannot install or upgrade any packages on my Ubuntu 11.10 system. Here is the rest of the error: unknown argument `triggered' dpkg: error processing mtools (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager-pptp-gnome ... No apport report written because MaxReports is reached already postinst called with unknown argument `triggered' dpkg: error processing network-manager-pptp-gnome (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-pptp ... postinst called with unknown argument `triggered' dpkg: error processing network-manager-pptp (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-gnome ... /var/lib/dpkg/info/network-manager-gnome.postinst called with unknown argument `triggered' dpkg: error processing network-manager-gnome (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager ... No apport report written because MaxReports is reached already /var/lib/dpkg/info/network-manager.postinst called with unknown argument `triggered' dpkg: error processing network-manager (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for mscompress ... postinst called with unknown argument `triggered' dpkg: error processing mscompress (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Errors were encountered while processing: netbase mtr-tiny module-init-tools mountmanager mono-4.0-gac mousetweaks mozilla-plugin-vlc mtools network-manager-pptp-gnome network-manager-pptp network-manager-gnome network-manager mscompress E: Sub-process /usr/bin/dpkg returned an error code (1)

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  • can't install anything ,getting error "Sub-process /usr/bin/dpkg returned an error code (1)"

    - by soum
    i am getting error whenever tring to install or update anything. "Sub-process /usr/bin/dpkg returned an error code (1)" please help me i am just stopped with my ubuntu 11.10. no installation or update. th unknown argument `triggered' dpkg: error processing mtools (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager-pptp-gnome ... No apport report written because MaxReports is reached already postinst called with unknown argument `triggered' dpkg: error processing network-manager-pptp-gnome (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-pptp ... postinst called with unknown argument triggered' dpkg: error processing network-manager-pptp (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-gnome ... /var/lib/dpkg/info/network-manager-gnome.postinst called with unknown argumenttriggered' dpkg: error processing network-manager-gnome (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager ... No apport report written because MaxReports is reached already /var/lib/dpkg/info/network-manager.postinst called with unknown argument triggered' dpkg: error processing network-manager (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for mscompress ... postinst called with unknown argumenttriggered' dpkg: error processing mscompress (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Errors were encountered while processing: netbase mtr-tiny module-init-tools mountmanager mono-4.0-gac mousetweaks mozilla-plugin-vlc mtools network-manager-pptp-gnome network-manager-pptp network-manager-gnome network-manager mscompress E: Sub-process /usr/bin/dpkg returned an error code (1)

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  • Uneditable file and Unreadable(for further processing) file( WHY? ) after processing it through C++

    - by mgj
    Hi...:) This might look to be a very long question to you I understand, but trust me on this its not long. I am not able to identify why after processing this text is not being able to be read and edited. I tried using the ord() function in python to check if the text contains any Unicode characters( non ascii characters) apart from the ascii ones.. I found quite a number of them. I have a strong feeling that this could be due to the original text itself( The INPUT ). Input-File: Just copy paste it into a file "acle5v1.txt" The objective of this code below is to check for upper case characters and to convert it to lower case and also to remove all punctuations so that these words are taken for further processing for word alignment #include<iostrea> #include<fstream> #include<ctype.h> #include<cstring> using namespace std; ifstream fin2("acle5v1.txt"); ofstream fin3("acle5v1_op.txt"); ofstream fin4("chkcharadded.txt"); ofstream fin5("chkcharntadded.txt"); ofstream fin6("chkprintchar.txt"); ofstream fin7("chknonasci.txt"); ofstream fin8("nonprinchar.txt"); int main() { char ch,ch1; fin2.seekg(0); fin3.seekp(0); int flag = 0; while(!fin2.eof()) { ch1=ch; fin2.get(ch); if (isprint(ch))// if the character is printable flag = 1; if(flag) { fin6<<"Printable character:\t"<<ch<<"\t"<<(int)ch<<endl; flag = 0; } else { fin8<<"Non printable character caught:\t"<<ch<<"\t"<<int(ch)<<endl; } if( isalnum(ch) || ch == '@' || ch == ' ' )// checks for alpha numeric characters { fin4<<"char added: "<<ch<<"\tits ascii value: "<<int(ch)<<endl; if(isupper(ch)) { //tolower(ch); fin3<<(char)tolower(ch); } else { fin3<<ch; } } else if( ( ch=='\t' || ch=='.' || ch==',' || ch=='#' || ch=='?' || ch=='!' || ch=='"' || ch != ';' || ch != ':') && ch1 != ' ' ) { fin3<<' '; } else if( (ch=='\t' || ch=='.' || ch==',' || ch=='#' || ch=='?' || ch=='!' || ch=='"' || ch != ';' || ch != ':') && ch1 == ' ' ) { //fin3<<" '; } else if( !(int(ch)>=0 && int(ch)<=127) ) { fin5<<"Char of ascii within range not added: "<<ch<<"\tits ascii value: "<<int(ch)<<endl; } else { fin7<<"Non ascii character caught(could be a -ve value also)\t"<<ch<<int(ch)<<endl; } } return 0; } I have a similar code as the above written in python which gives me an otput which is again not readable and not editable The code in python looks like this: #!/usr/bin/python # -*- coding: UTF-8 -*- import sys input_file=sys.argv[1] output_file=sys.argv[2] list1=[] f=open(input_file) for line in f: line=line.strip() #line=line.rstrip('.') line=line.replace('.','') line=line.replace(',','') line=line.replace('#','') line=line.replace('?','') line=line.replace('!','') line=line.replace('"','') line=line.replace('?','') line=line.replace('|','') line = line.lower() list1.append(line) f.close() f1=open(output_file,'w') f1.write(' '.join(list1)) f1.close() the file takes ip and op at runtime.. as: python punc_remover.py acle5v1.txt acle5v1_op.txt The output of this file is in "acle5v1_op.txt" now after processing this particular output file is needed for further processing. This particular file "aclee5v1_op.txt" is the UNREADABLE Aand UNEDITABLE File that I am not being able to use for further processing. I need this for Word alignment in NLP. I tried readin this output with the following program #include<iostream> #include<fstream> using namespace std; ifstream fin1("acle5v1_op.txt"); ofstream fout1("chckread_acle5v1_op.txt"); ofstream fout2("chcknotread_acle5v1_op.txt"); int main() { char ch; int flag = 0; long int r = 0; long int nr = 0; while(!(fin1)) { fin1.get(ch); if(ch) { flag = 1; } if(flag) { fout1<<ch; flag = 0; r++; } else { fout2<<"Char not been able to be read from source file\n"; nr++; } } cout<<"Number of characters able to be read: "<<r; cout<<endl<<"Number of characters not been able to be read: "<<nr; return 0; } which prints the character if its readable and if not it doesn't print them but I observed the output of both the file is blank thus I could draw a conclusion that this file "acle5v1_op.txt" is UNREADABLE AND UNEDITABLE. Could you please help me on how to deal with this problem.. To tell you a bit about the statistics wrt the original input file "acle5v1.txt" file it has around 3441 lines in it and around 3 million characters in it. Keeping in mind the number of characters in the file you editor might/might not be able to manage to open the file.. I was able to open the file in gedit of Fedora 10 which I am currently using .. This is just to notify you that opening with a particular editor was not actually an issue at least in my case... Can I use scripting languages like Python and Perl to deal with this problem if Yes how? could please be specific on that regard as I am a novice to Perl and Python. Or could you please tell me how do I solve this problem using C++ itself.. Thank you...:) I am really looking forward to some help or guidance on how to go about this problem....

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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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  • 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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  • We are hiring (take a minute to read this, is not another BS talk ;) )

    - by gsusx
    I really wanted to wait until our new website was out to blog about this but I hope you can put up with the ugly website for a few more days J. Tellago keeps growing and, after a quick break at the beginning of the year, we are back in hiring mode J. We are currently expanding our teams in the United States and Argentina and have various positions open in the following categories. .NET developers: If you are an exceptional .NET programmer with a passion for creating great software solutions working...(read more)

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  • Tellago is still hiring….

    - by gsusx
    Tellago 's SOA practice is rapidly growing and we are still hiring. In that sense, we are looking to for Connected Systems (WCF, BizTalk, WF) experts who are passionate about building game changing solutions with the latest Microsoft technologies. You will be working alongside technology gurus like DonXml , Pablo Cibraro or Dwight Goins . If you are interested and not afraid of working with a bunch of crazy people ;)please drop me a line at jesus dot rodriguez at tellago dot com. Hope to hear from...(read more)

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  • Back from Teched US

    - by gsusx
    It's been a few weeks since I last blogged and, trust me, I am not happy about it :( I have been crazily busy with some of our projects at Tellago which you are going to hear more about in the upcoming weeks :) I was so busy that I didn't even have time to blog about my sessions at Teched US last week. This year I ended up presenting three sessions on three different tracks: BIE403 | Real-Time Business Intelligence with Microsoft SQL Server 2008 R2 Session Type: Breakout Session Real-time business...(read more)

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  • Tellago && Tellago Studios 2010

    - by gsusx
    With 2011 around the corner we, at Tellago and Tellago Studios , we have been spending a lot of times evaluating our successes and failures (yes those too ;)) of 2010 and delineating some of our goals and strategies for 2011. When I look at 2010 here are some of the things that quickly jump off the page: Growing Tellago by 300% Launching a brand new company: Tellago Studios Expanding our customer base Establishing our business intelligence practice http://tellago.com/what-we-say/events/business-intelligence...(read more)

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  • Limiting DOPs &ndash; Who rules over whom?

    - by jean-pierre.dijcks
    I've gotten a couple of questions from Dan Morgan and figured I start to answer them in this way. While Dan is running on a big system he is running with Database Resource Manager and he is trying to make sure the system doesn't go crazy (remember end user are never, ever crazy!) on very high DOPs. Q: How do I control statements with very high DOPs driven from user hints in queries? A: The best way to do this is to work with DBRM and impose limits on consumer groups. The Max DOP setting you can set in DBRM allows you to overwrite the hint. Now let's go into some more detail here. Assume my object (and for simplicity we assume there is a single object - and do remember that we always pick the highest DOP when in doubt and when conflicting DOPs are available in a query) has PARALLEL 64 as its setting. Assume that the query that selects something cool from that table lives in a consumer group with a max DOP of 32. Assume no goofy things (like running out of parallel_max_servers) are happening. A query selecting from this table will run at DOP 32 because DBRM caps the DOP. As of 11.2.0.1 we also use the DBRM cap to create the original plan (at compile time) and not just enforce the cap at runtime. Now, my user is smart and writes a query with a parallel hint requesting DOP 128. This query is still capped by DBRM and DBRM overrules the hint in the statement. The statement, despite the hint, runs at DOP 32. Note that in the hinted scenario we do compile the statement with DOP 128 (the optimizer obeys the hint). This is another reason to use table decoration rather than hints. Q: What happens if I set parallel_max_servers higher than processes (e.g. the max number of processes allowed to run on my machine)? A: Processes rules. It is important to understand that processes are fixed at startup time. If you increase parallel_max_servers above the number of processes in the processes parameter you should get a warning in the alert log stating it can not take effect. As a follow up, a hinted query requesting more parallel processes than either parallel_max_servers or processes will not be able to acquire the requested number. Parallel_max_processes will prevent this. And since parallel_max_servers should be lower than max processes you can never go over either...

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  • Tellago speaks about Business Intellligence with SQL Server 2008 R2

    - by gsusx
    At Tellago , we always try to stay in the frontlines of technology that can enhance our solution development practices. This year we are putting a lot of emphasis on business intelligence and in particular the new set of BI technologies such as Microsoft's PowerPivot, Master Data Services and StreamInsight that are scheduled to be release with SQL Server 2008 R2. In the last few weeks we have been working closely with different Microsoft field offices to coordinate a series of customers events that...(read more)

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