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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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• Is RAID 0 or JBOD better for home media server?

  - by Donald Hughes

  I have an external two-bay drive enclosure (the OWC Mercury Elite-AL Pro) connected to a Mac Mini (my home media server) over FireWire 800. I'm streaming media to other computers in the house over wired gigabit. I have two 1.5 TB drives that I'm using independently right now. The media is on one, and I'm mirroring the files to the other drive at night as a backup. But as I approach filling up the drive I'm wanting to span those two drives together to give me a total of about 3 TB, and then buy another drive for backups. The external enclosure supports both RAID 0 and JBOD, but I'm not clear on which would be better in this situation. Would RAID 0 provide any performance improvements over JBOD for streaming video (possibly several streams at once? How does each affect the MTBF of the drives? In general, should I choose RAID 0, JBOD, or keep them independent?

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• One codec to rule them all

  - by AngryHacker

  I am streaming videos in my house via Windows Media Player Streaming, which is basically DLNA. So theoretically any DLNA compliant device can pick up the stream. However, I've quickly found that this is only one part of the solution. Over the years I've accumulated a ton of video-capable devices. While all these devices can see the Windows Media Player stream, they all speak in different codecs. And frankly, I am confused by codecs. In the beginning, I thought that the codecs were defined by the filename extension they carried (e.g. avi, mp4, wmv, etc...), but after further research, it looks like the extensions are simply containers. Inside an .avi file could reside several different codecs. So my question is this: is there a format/codec that plays equally well on any device.

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• Setup Windows Media Player 11 to stream from TVersity

  - by snorfys

  I've got TVersity installed on a Windows 2003 server box (work had an extra license that they donated to let me install at home to get some practice setting up/administering a domain etc.) I found out that Windows Media Player 11 won't install on Windows 2003, but installed TVersity instead and streaming to my 360 is working great. Problem is that I don't know how to setup streaming to any other PC on the network. All of the PCs have access to the shared network folder, but playing from there doesn't stream and the stutter is pretty bad. Is there a way to setup Windows Media Player 11 or another player to stream from TVersity?

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• Is an Adnroid-based Phone a Suitable MP3 Player for Music Streamed over the Internet?

  - by James McFarland

  I am considering getting an HTC phone running Android from Verizon Wireless when I next upgrade my phone. I also have an online account with a music vendor, where I have rights to listen to my collection, but not download the MP3s. Further, I have an unlimited data plan and Wi-Fi, so I have full access to bandwidth volume without any concerns. I am especially interested in mounting my phone in a car kit, and streaming my online music to my car's sound system while driving. If you are experienced in this scenario, or have tried this scenario - Is is reasonable to expect my HTC Android phone to provide me with streaming music via my cell data plan anywhere I get cell service?

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• Wireless dropouts that only affect subset of devices

  - by jwaddell

  When watching videos streamed over WiFi from a NAS box (D-Link DNS-323) I am getting wireless dropouts. However they only appear to occur when I have left my laptop (Dell Inspiron 9300 running Windows XP SP3) running; the laptop is usually suspended if I'm not using it. The dropouts have occurred when streaming to a Netgear EVA8000 streaming device, and also to a PS3. I'm using a Netgear DG834G as the wireless modem/router. When a dropout occurs I go to the laptop and see that its wireless connection has also dropped out. The odd thing is that my wife's MacBook and my iPhone still maintain their connections. What could be causing this behaviour, and how do I go about fixing it?

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• How to stream multiple files on demand in VLC?

  - by romkyns

  Is there any way at all that I can set up VLC on a server PC in such a way that I can access a list of all my videos from another PC, and pick one to be streamed on demand? I've been pointed at this streaming guide (pdf), but it's pretty useless. For a start, most of the menus in those screenshots don't match the actual current version VLC, and then it sort of assumes you already know what you're doing. So far I managed to figure out how to stream a single file, which I must choose before watching on the server PC - pretty useless if you ask me! The impenetrable "UI" doesn't help either... (P.S. The reason I'm going for streaming rather than the very simple to set up network drive is described in this question)

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• Improving sound quality with remote ESD server

  - by cuu508

  Hi, I'm investigating low-budget ways to get audio from my PC (Ubuntu) to HiFi without wires. I'm currently testing a setup where Asus WL-500gP wireless router runs ESD daemon and has attached USB soundcard which is then plugged into HiFi. I'm testing playback on PC with mpg123-esd and Spotify under Wine. The sound is there, latency is unexpectedly low, but I also hear occassional clicks and some distortion from time to time. I suppose that's because of the low latency and wireless streaming of uncompressed audio--any packet drops, CPU temporarily being busy etc. will cause clicks in sound output. Is there a way around this problem, increasing latency / buffer size somehow perhaps? Streaming using shoutcast protocol seems to be a way out but I have feeling that would be a complex and brittle setup.

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• Les États-Unis veulent durcir la législation anti-streaming, diffuser une vidéo protégée pourrait devenir passible de prison

  Les États-Unis cherchent à renforcer leur législation anti-streaming Diffuser une vidéo protégée sur une plate-forme de streaming pourrait devenir passible de prison Pendant que la loi HADOPI est implémentée en France avec ses notions de riposte graduée et de coupure de connexion Internet en cas de partage sur le réseau P2P, le congrès américain se penche sur un texte de loi proposant une approche bien plus radicale pour renforcer le droit d'auteur. La proposition de loi répondant au nom de "Bill S.978" propose en effet de sanctionner lourdement presque toute diffusion de contenu protégé par le droit d'auteur via une plate-forme de streaming telle que YouTube. Ain...

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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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• Ask How-To Geek: iPad Battery Life, Batch Resizing Photos, and Syncing Massive Music Collections

  - by Jason Fitzpatrick

  Christmas was good to many of you and now you’ve got all sorts of tech questions related to your holiday spoils. Come on in and we’ll clear up how to squeeze more life out of your iPad, resize all those photos, and sync massive music collections to mobile devices. Once a week we dip into our reader mailbag and help readers solve their problems, sharing the useful solutions with you in the process. Read on to see our fixes for this week’s reader dilemmas. Latest Features How-To Geek ETC How to Use the Avira Rescue CD to Clean Your Infected PC The Complete List of iPad Tips, Tricks, and Tutorials Is Your Desktop Printer More Expensive Than Printing Services? 20 OS X Keyboard Shortcuts You Might Not Know HTG Explains: Which Linux File System Should You Choose? HTG Explains: Why Does Photo Paper Improve Print Quality? Orbiting at the Edge of the Atmosphere Wallpaper Simon’s Cat Explores the Christmas Tree! [Video] The Outdoor Lights Scene from National Lampoon’s Christmas Vacation [Video] The Famous Home Alone Pizza Delivery Scene [Classic Video] Chronicles of Narnia: The Voyage of the Dawn Treader Theme for Windows 7 Cardinal and Rabbit Sharing a Tree on a Cold Winter Morning Wallpaper

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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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