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• Hard crash when using bluetooth headset on MacBook Pro and Lion 10.7.2

  - by jtalarico

  I recently picked up a bluetooth headset (Motorola S10-HD) and started using it with my MacBook Pro (17" purchased new in 2010) running Lion 10.7.2. Here's what works - stereo audio: iTunes Spotify Pandora (via browser) games (e.g. Minecraft, which is a Java app) audio from YouTube Plex, VLC, other video players Here's what doesn't work well - stereo audio fails and the headset seems to go into mono (i.e. tinny-as-hell) mode: Google Hangout Skype GoTo Meeting Here's what's just downright catastrophic. If I'm listening to stereo audio and then decide to jump into a Skype call (Google Hangout, or GoTo Meeting), bluetooth often crashes and I can only get things working again by shutting down the device, disabling bluetooth, and getting things back up and running again. But the audio is still horrible, and MUCH better using just a simple set of iPhone earbuds and mic. About 80% of the time during such a call, bluetooth crashes. And about 90% of the time, after the call ends, Skype is shut down, or I try to switch back to playing stereo audio, I get a hard crash!! The gray screen of death descends and I'm told I need to restart my machine. In one such instance, even after a reboot, I could not enable bluetooth again ("Turn Bluetooth On" was grayed out in taskbar). Is this just a weak implementation of bluetooth by Apple, or is this a hardware issue? I've seen others posting similar issues even on the Apple support site indicating that bluetooth headsets are failing left and right, but I haven't seen anyone mention hard crashes.

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• How to send Bluetooth audio to non-Bluetooth speakers?

  - by wonsungi

  In short, I am looking for a Bluetooth-3.5 mini stereo converter. What is this type of device called, and what are some of the best models (is there a difference in audio quality/lag)? I wish to connect some speakers (Altec Lansing inMotion IM7), which does not support Bluetooth, to my laptop (Lenovo X301) wirelessly. Currently, I can connect my laptop's headphone jack to the AUX jack on my speakers via a mini stereo cable. How do I replace this cable with some type of Bluetooth setup? I am not sure what this Bluetooth device is called. I thought I found something, but it actually does the opposite of what I need (3.5 mini stereo-Bluetooth). (My OS is Vista Enterprise, if that matters)

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• MPlayer refuses to generate mono wav file

  - by JCCyC

  I want to downsample an existing audio file to 8KHz mono. This command line downsamples it to stereo: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1 -ao pcm:waveheader:file="/tmp/blah1.wav" ~/from_my_cellphone.3ga It generates a file that the file utility identifies as stereo: $ file /tmp/blah1.wav /tmp/blah1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Now, if I read the documentation correctly, I should add pan=1:0.5:0.5 so I get a file that's half the size: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1:pan=1:0.5:0.5 -ao pcm:waveheader:file="/tmp/blah2.wav" ~/from_my_cellphone.3ga But it doesn't! blah2.wav is identical to blah1.wav! What am I doing wrong?

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• xvidcap: Error accessing sound input from /dev/dsp

  - by stivlo

  I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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• Unable To Get Sound Working to External Speaker on HP TouchSmart 320 on 11.04 or 11.10

  - by Schof

  This is an HP TouchSmart 320, model number 320-1200m. I'm using Ubuntu 11.04. Hardware information: root@hp320:/home/mpower# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 root@hp320:~$ cat /proc/asound/card0/codec#0 | grep Codec Codec: IDT 92HD91BXX Sound to headphone jack works properly, but sound to built-in speakers does not work. I have installed Windows, and with Windows 7 installed, all audio hardware works properly, so this isn't a hardware fault. I've looked at https://help.ubuntu.com/community/HdaIntelSoundHowto and have been unable to find my card in http://www.kernel.org/doc/Documentation/sound/alsa/HD-Audio-Models.txt . I have tried adding almost every conceivable model in the line "options snd-hda-intel model=MODEL" line I added to /etc/modprobe.d/alsa-base.conf. Update 2011-11-09 2:31 PM PST: I've gone to Control Center - Sound Preferences to attempt settings that make sound work. The "Hardware" tab shows one device: "Internal Audio 1 Output / 1 Input Analog Stereo Duplex." There are two output profiles listed in the selection box at the bottom of the tag: Analog Stereo Duplex and Analog Stereo Output. Neither cause sound to emit from the speakers. I've also run alsamixer on the command-line and ensured that everything is set to maximum and nothing is muted. Update 2011-11-09 5:15 PM PST: I've replicated the exact same symptoms in 11.10. Update 2011-11-10 11:31 AM PST: I've filed a bug in launchpad: https://launchpad.net/ubuntu/+source/alsa-driver/+bug/888703

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• speakers dont work in macbook pro

  - by Ali_IT

  I have a macbook pro but my Built-in stereo speakers don't work and it comes a red light from Headphone out/optical digital audio out port. my Built-in stereo speakers aren't dead because at first that OS runs it comes a sound from them but as soon as the macbook pro is ready when i play music they don't work and In the SOUND in system preferences the name of device for sound output is Digital Out. Is the problem from hardware or software. Is there any solution?

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• Repackage m4v file

  - by Daniel Huckstep

  I have some m4v files I made with Handbrake where the AC3 audio channel was the first one, and the stereo was the second. This causes problems with some things (like Quicktime) so I want to repackage the file such that the stereo track is the first audio track. I don't want to re-encode things. Can I do this? Can I do it for free?

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• Samsung Galaxy S II bluetooth headset

  - by tumchaaditya

  I want to buy a stereo bluetooth headset for Galaxy SII and Galaxy SIII. I am not asking for any product recommendation. But, I just want to be sure that I will be able to receive phone calls using the button on headset as well as listen music in stereo(which will be taken care of by A2DP). So, any guidelines on choosing the headsets? What to look for to ensure that the buttons on the headset work?

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• windows live playback left and right audio channel

  - by user1254761

  I have a multichannel (4x stereo) audiocard (m-audio delta1010lt) and want to playback /playthru some of the channels live. But I am only able to playback/playthru the left channel on each stereo-input (CH1, CH3, CH5, CH7). For CH2,CH4,CH6,CH8 I see the Windows Volume-Indicator going up and down in the Windows Record-Audiosettings but I don't hear any playback sound. Is there a way to playback/playthru all input channels?

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• How hard is it to create a not-so-random number generator?

  - by Duracell

  Backstory: So I was driving to band practice this evening. My car has a USB port where you can plug in a USB stick with MP3 files on it and the stereo will play them. I have about 100 MP3s on my stick so I pushed the 'Random' button. So from here to band practice, it played: Track 22 Track 45 Track 4 Track 11 Track 87 Track 66 Track 98 Then on the way home, it played Track 16 Track 27 Track 33 And then I stopped at the petrol station. I filled up, got back in the car and the stereo fired up again. It played Track 22 Track 45 Track 4 Track 11 Track 87 I thought, WTF? What's with this 'random' generator? What are they using as a seed, if not time? Is a car stereo so memory-tight that it can't even use the C stdlib? Does anyone know how this kind of thing happens?

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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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• Composite Moon Map Offers Stunning Views of the Lunar Surface [Astronomy]

  - by Jason Fitzpatrick

  Researchers at Arizona State University have stitched together a massive high-resolution map of the moon; seen the moon in astounding detail. Using images fro the Lunar Reconnaissance Orbiter (LRO) they carefully stitch a massive map of the moon with a higher resolution than the public has ever seen before: The WAC has a pixel scale of about 75 meters, and with an average altitude of 50 km, a WAC image swath is 70 km wide across the ground-track. Because the equatorial distance between orbits is about 30 km, there is nearly complete orbit-to-orbit stereo overlap all the way around the Moon, every month. Using digital photogrammetric techniques, a terrain model was computed from this stereo overlap. Hit up the link below to check out the images and the process they used. Lunar Topography as Never Seen Before [via NASA] How to Make the Kindle Fire Silk Browser *Actually* Fast! Amazon’s New Kindle Fire Tablet: the How-To Geek Review HTG Explains: How Hackers Take Over Web Sites with SQL Injection / DDoS

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• Acer Revo (ION platform) + Maverick + 5.1 surround over HDMI

  - by Oli

  I've had a turbulent relationship with my media centre box. Every upgrade I perform on it seems to bring a brand new set of audio issues (the opposite of my desktop where things seem to get better and better). It's a Acer Revo 3600. That's basically an low-end Intel Atom chip with a Nvidia 9400M onboard. On paper that's perfect for something like a media centre. But having just upgraded to Maverick, the sound properties box only wants to offer me stereo sound over HDMI. The exact setup goes: Revo - Onkyo AV receiver - LG TV. The Onkyo box strips off the audio (supporting 7.1 -- though we're only using 6 speakers) and feeds the video onto the TV. I'd like to get to a point where Ubuntu thinks it's doing 5.1 over HDMI, upmixing stereo to 6ch and supporting DTS/AC3 (through Boxee). I've had this working before but it's frankly been a bit of a hacktastrophe. The audio chip is recognised as Nvidia MCP79/7A HDMI in alsamixer if that helps.

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• Why won't cron run my sh script?

  - by Dmitry Narkevich

  Used gnome-schedule to create a script to set my headset as the fallback audio device because it keeps unsetting it when the headset gets disconnected or pc goes into sleep mode. Anyway, crontab is this: SHELL=/bin/sh PATH=/bin:/sbin:/usr/bin:/usr/sbin:/home/dmitry/bin * * * * * headsetfix /home/dmitry/bin/headsetfix is #!/bin/sh pacmd set-default-sink alsa_output.usb-Logitech_Inc_Logitech_USB_Headset_H540_00000000-00-H540.analog-stereo pacmd set-default-source alsa_input.usb-Logitech_Inc_Logitech_USB_Headset_H540_00000000-00-H540.analog-stereo It runs fine from the terminal. I've made sure it's chmodded to be executable, and "which headsetfix", run from cron, outputs "/home/dmitry/bin/headsetfix" so not sure what the problem is.

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• Which of VLC's dependencies causes sound device detection?

  - by Raphael

  I am setting up a headless music server based on the minimal Ubuntu image. After having installed the packages openssh-server,pulseaudio, libmad0,flac,liboff0,libid3tag0,libvorbis0a,ffmpeg, mpd,mpc,mpdscribble, paman,paprefs,pavumeter neither my internal soundcard nor the external DAC where detected by pulseaudio, that is pactl list did only list the dummy devices. Several reboots did not change that. The hardware devices are detected properly: ~$ lsusb | grep Texas Bus 002 Device 002: ID 08bb:2706 Texas Instruments Japan ~$ lspci | grep Audio 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Following a hunch, I installed vlc with all dependencies. After a reboot, both devices are detected! ~$ pactl list | grep "Sink: alsa_output" Monitor of Sink: alsa_output.pci-0000_00_1b.0.analog-stereo Monitor of Sink: alsa_output.usb-Burr-Brown_from_TI_USB_Audio_DAC-00-DAC.analog-stereo Now I would like to remove VLC again but keep the devices. The question is: which of the many dependencies of VLC enables proper device detection? And why on earth is it not a dependency of pulseaudio?

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• Problem with sound in Kubuntu 12.10

  - by Mihkel

  I'm really enjoying Kubuntu 12.10 experience, but the problem starts with sound. It wasn't here before, but today sound sounds garbled and echoed and wrong. It happens in Audacity and VLC. It doesn't happen when I test the sound devices nor when I use Amarok to play the music files (but come on, who uses Amarok to listen to a random music file, it's much more natural to use VLC for that ;-) ) Kubuntu/Phonon recognizes 2 sound devices: 1) RV770 HDMI Audio [Radeon HD 4850/4870] Digital Stereo [HDMI] 2) Built-in Audio Analog Stereo I know it has to use the second option, and it probably does, but that's not the case. What I did find out was that I had to rescan for audio devices in Audacity (and probably select "sysdefault") for it to sound normal. Why does it happen? I've tried following some other questions, but well.

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• Speakers don't work in 12.10 but they work fine on windows7

  - by giri

  I have recently upgraded my Ubuntu 12.04 to 12.10 version and find issues with my speakers as well as microphone. When I boot the system they don't work, but(don't know why) when I restart once or twice they work fine. There is no problem with my laptop(dell xps) as they work well on windows7. I have my sound settings as follows Hardware --- Built-in Audio 1 Outpu/1 Input Analog Stereo Duplex Input(Internal Microphone) & Output(Speakers) -----Built-in audio Analog Stereo Any suggestions to fix the problem??

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• 5.1 surround sound for 12.04

  - by iwhorl

  I am very new to Ubuntu and linux as I have used windows only until yesterday. I am attempting to send my audio signal to my pioneer reciever using a single digital optical cable which is plugged into my M2N Sli Deluxe motherboard. I am sending video with a Geforce 8600gt video card through HDMI. That card does not support audio through HDMI. The only thing I have got to work so far is my left channel speaker acts as a stereo left and my sub woofer is trying to act as a stereo right. I was able to achieve this through altering the alsamixer settings. Can anyone point me in the right direction on this?

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• Changing in pavucontrols tab "Recording" via command line

  - by Mojo

  I'm using pavucontrol to make changes in the "Recording". I'm changing the source (??) of a Loopback to Null-Output from "Internes Audio Analog Stereo" to "Monitor of Internes Audio Analog Stereo" see the screenshot http://picpaste.de/Bildschirmfoto_vom_2013-10-26_11_32_03-z0KwnFDE.png I'm now looking for a possibility to do this via command line. So far I've done the following: pactl load-module module-null-sink ? creates a new sink pactl load-module module-loopback ? creates a new sink input pactl load-module module-loopback ? creates another sink input pacmd move-sink-input 0 1 ? changes the sink of the sink-input (to Null-Output); this is like changing manually in the pacucontrol tab "Playback". It's just the last part (making the change like shown in the screenshot) via command line that I'm not able to do. I'd be very happy for any advice or suggestions. Thanks already!

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• Microphone not working at all on a Toshiba sattelite p505 s8980

  - by flamingburner

  My built-in microphone doesn't work in "record audio" program and in Skype. I'm using ubuntu 11.10 in Toshiba sattelite p505 s8980. No mute button checked I have PulseAudio Volume Control installed currently when aim trying to use record audio program and hear what I record it plays noisy sound not my voice I changed my computer from 'Analog Stereo Output' to 'Analog Stereo Duplex', in both sound setting ( ubuntu ) and in PulseAudio Volume Control no problem in bios and in my windows seven sound worked perfectly and in ubuntu no problem in sound at all i can play music and movies and even in skype i can hear the sounf of who is taking to me ( it's about the microphone only "

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• Connect bluetooth headphones both to PC and phone at the same time

  - by Sergiy Byelozyorov

  I have recently bought Philips SHB6110. Extract from the 13th page of manual: Therefore you can connect your Bluetooth stereo headset. with a Bluetooth stereo enabled phone to both listen to music and lead calls, or with a Bluetooth phone that does not support Bluetooth stereo (A2DP) to lead calls and at the same time to a Bluetooth audio device (Bluetooth enabled MP3 player, Bluetooth audio adapter etc.) to listen to music. Make sure to pair the phone first with your Bluetooth headset, then turn both the phone and headset off to then pair the Bluetooth audio device. With the SwitchStream feature you can listen to music and monitor your calls at the same time. Even while listening to music, you will hear a ring tone when receiving a call and can switch to the call simply by tapping the button. The manual however doesn't specify how do I connect to both device at the same time. I use Toshiba Satellite Pro P300-1CG laptop with Belkin Mini Bluetooth Adapter and Nokia N95 phone. Operating system is Windows 7 64-bit and I have Skype installed. Both phone and compute can be used for listening to music and talking on the phone (on PC via Skype). Best solution would be if I could connect to PC and phone as the same time and monitor calls both mobile and Skype calls while listening music from Winamp. If that is not possible, then I would like at least to be able to listen music from PC, while monitoring calls from mobile. So, please tell me how do I connect both PC and phone to headphones?

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• This .mpg video clip doesn't play well

  - by Roey

  I've installed K-lite mega codec pack v6.9.0 with playback essentials without player. My default and only media player is windows media player. here are the clip's media info: General Complete name : D:\Users\Roey\Downloads\B384MV.mpg Format : MPEG-PS File size : 273 MiB Duration : 4mn 59s Overall bit rate : 7 643 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@High Format settings, BVOP : No Format settings, Matrix : Default Format settings, GOP : M=1, N=15 Duration : 4mn 57s Bit rate mode : Variable Bit rate : 7 363 Kbps Nominal bit rate : 9 000 Kbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.142 Stream size : 261 MiB (96%) Audio ID : 192 (0xC0) Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Duration : 4mn 59s Bit rate mode : Constant Bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 44.1 KHz Compression mode : Lossy Stream size : 4.56 MiB (2%) Menu When I play it there is no sound (just a little "kahhhh" noise every 10-20 seconds) and the frames are moving very slow - it "jumps" frames. A blue tray icon [FFa] "ffdshow audio decoder" pops with the following details: Input:MP3, stereo, 44100 Hz (libavocodec) Output:PCM, stereo, 44100 Hz, 16-bit integer Any help will be much appreciated. Thanks

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• AVConv increases song duration when converting MP3

  - by chauffch

  I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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• Converting from mp4 to Xvid avi using avconv?

  - by Ricardo Gladwell

  I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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• Are there any 5.1 surround audio switches on the market?

  - by thepurplepixel

  (Somewhat related to this question) I have a set of Logitech 5.1 surround speakers, which use 3 stereo 3.5mm TRS connectors (minijacks) to transfer the audio (the typical green/black/orange audio outputs). I have a Griffin Firewave hooked up to my MacBook Pro, and my desktop has a Realtek ALC889 audio chipset. I have looked for a way to, essentially, switch the speaker inputs between my Firewave and my desktop without having to disconnect the cables from one, route them around my desk, and plug them into the other. I'd love to have something like an old Belkin DB-25/LPT switch, but for these audio cables. Of course, purchasing one and soldering my own cables on the connection terminals is always an option, but, is there a reasonably priced 5.1 audio switch (or 3x stereo) on the market that will accomplish the simple task of switching audio outputs between two computers into a set of 5.1 speakers? Thanks in advance!

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