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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• Problems with 5.1 digital out on Ubuntu 12.04

  - by user895319

  I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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• 5.1 Surround Channels are Jumbled

  - by stickynips

  I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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• How to get the mic on the Creative X-Mod soundcard working correctly?

  - by Nyamiou The Galeanthrope

  Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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• Renoise and dssi and jack

  - by Bojan

  This may be a little complicated, but, is jack a necessity? I mean, i use renoise, and, since i dont have the need for low latencies, do i really need to use it? My basic setup ( or workflow ) is that i use csound to render stuff to wav, then import it as a sample in renoise. That goes with field recordings, my own samples, etc. So, i dont need ultra low latencies, and i dont need to patch "cords", but i want to use dssi plugins, and dssi-vst. What would be something of a minimum requirements of apps that should work. Can renoise load dssi-vst plugins by itself or do i need to use jack to patch thru or something third, i tried to read lot of articles but i got lost in the different setups...

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• Beat detection, weird detection

  - by Quincy

  I made this soundanalyzer class to detect beats in songs : // put it on pastebin for the big size, will put it here if people rather want that. pastebin.com/8PdgZPP3 but for some reason its only detecting beats from 637 sec to around 641(sec) and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates and it seems as its assigning a beat to each instant energy value in between those values. Its modeled after this : http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats properly register ?

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• Computer SOMETIMES recognizes when headphones are plugged in.

  - by rcrobot

  Whenever I plug my headphones into my computer's front headphone jack, I get a weird situation. Sometimes, the computer will recognize the headphones and work properly. But other times, the computer will play sound through both the headphones and my monitor's speaker. When this happens, the sound section of the system settings does not list the headphones. I can fix the issue temporarily by wiggling the headphone port, but if it gets wiggled the wrong way again, then the issue returns. My PC's case is a Rosewill Challenger. I have tried multiple headphones and the same issue is there. I suspect that this might be a hardware related issue, but if there is any way to fix it with software, that would be helpful. This is what it looks like when everything is working properly: This happens when I wiggle the headphone port. I can quickly switch between these two by doing so:

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• How to correctly Dispose a SourceVoice once its finished

  - by clamp

  i am starting to play a sound with XAudio2 and SourceVoice and once its finished, it should be correctly disposed to not have any leaks. i was expecting it to be something like this: sourceVoice.Start(); sourceVoice.StreamEnd += delegate { if (!sourceVoice.IsDisposed) { sourceVoice.DestroyVoice(); sourceVoice.Dispose(); } }; but that crashes with a read access violation in native code deep in XAudio2.dll which i cant debug.

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• Playing a Song causing WP7 to crash on phone, but not on emulator

  - by Michael Zehnich

  Hi there, I am trying to implement a song into a game that begins playing and continually loops on Windows Phone 7 via XNA 4.0. On the emulator, this works fine, however when deployed to a phone, it simply gives a black screen before going back to the home screen. Here is the rogue code in question, and commenting this code out makes the app run fine on the phone: // in the constructor fields private Song song; // in the LoadContent() method song = Content.Load<Song>("song"); // in the Update() method if (MediaPlayer.GameHasControl && MediaPlayer.State != MediaState.Playing) { MediaPlayer.Play(song); } The song file itself is a 2:53 long, 2.28mb .wma file at 106kbps bitrate. Again this works perfectly on emulator but does not run at all on phone. Thanks for any help you can provide!

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• How do I get my ART USB Dual Pre preamp to work?

  - by Zach

  I am using Audacity. I have an ART USB Dual Pre preamp. Ubuntu is not recognizing it whatsoever. I am able to record in Audacity, but it is using the mic that is built into my computer (which is a compaq Presario CQ50) instead of the one plugged into the preamp. How do I get Ubuntu to recognize the preamp that is plugged into my computer? Something tells me it has to do with the installation of the preamp software. It came with a installation CD, but when I go to "install", the nothing happens. I can view what is on the CD, but there is no installing of anything. Please help!

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• Turn off all sounds from websites

  - by David Oneill

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

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

  - by Lori Kaufman

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

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• IrrKlang with Ogre

  - by Vinnie

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

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• music for an arcade game?

  - by user717572

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

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• How to disable Alert volume from the command line?

  - by Bryce

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

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

  - by Bryce

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

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

  - by ændrük

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

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• Setting up port forwarding for 7000 appliance VM in VirtualBox

  - by uejio

  I've been using the 7000 appliance VM for a lot of testing lately and relied on others to set up the networking for the VM for me, but finally, I decided to take the dive and do it myself.  After some experimenting, I came up with a very brief number of steps to do this all using the VirtualBox CLI instead of the GUI. First download the VM image and unpack it somewhere.  I put it in /var/tmp. Then, set your VBOX_USER_HOME to some place with lots of disk space and import the VM: export VBOX_USER_HOME=/var/tmp/MyVirtualBoxVBoxManage import /var/tmp/simulator/vbox-2011.1.0.0.1.1.8/Sun\ ZFS\ Storage\ 7000.ovf (go get a cup of tea...) Then, set up port forwarding of the VM appliance BUI and shell:First set up port as NAT:VBoxManage modifyvm Sun_ZFS_Storage_7000 --nic1 nat Then set up rules for port forwarding (pick some unused port numbers):VBoxManage modifyvm Sun_ZFS_Storage_7000 --natpf1 "guestssh,tcp,,4622,,22"VBoxManage modifyvm Sun_ZFS_Storage_7000 --natpf1 "guestbui,tcp,,46215,,215" Verify the settings using:VBoxManage showvminfo Sun_ZFS_Storage_7000 | grep -i nic Start the appliance:$ VBoxHeadless --startvm Sun_ZFS_Storage_7000 & Connect to it using your favorite RDP client.  I use a Sun Ray, so I use the Sun Ray Windows Connector client: $ /opt/SUNWuttsc/bin/uttsc -g 800x600 -P <portnumber> <your-hostname> & The portnumber is displayed in the output of the --startvm command.(This did not work after I updated to VirtualBox 4.1.12, so maybe at this point, you need to use the VirtualBox GUI.) It takes a while to first bring up the VM, so please be patient. The longest time is in loading the smf service descriptions, but fortunately, that only needs to be done the first time the VM boots.  There is also a delay in just booting the appliance, so give it some time. Be sure to set the NIC rule on only one port and not all ports otherwise there will be a conflict in ports and it won't work. After going through the initial configuration screen, you can connect to it using ssh or your browser: ssh -p 45022 root@<your-host-name> https://<your-host-name>:45215 BTW, for the initial configuration, I only had to set the hostname and password.  The rest of the defaults were set by VirtualBox and seemed to work fine.

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• Lots of static/crackling noises after ALSA HDA DKMS installation

  - by MartinB

  I am using a Samsung Chronos 7 laptop with the following sound setup: $ head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Realtek ALC269VC ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI With the stock ALSA that comes with Ubuntu 12.04, I do not get any sound out of the headphones when I plug them into the headphone jack. After plugging the headphones in, I have to manually use Alsamixer to increase the volume, so that the headphones become usable. I have been told that this issue is due to my sound chip not being supported in the ALSA version that ships with Precise. A similar question at AskUbuntu and the Ubuntu Community Documentation point me to the ALSA DKMS installation. After installing the dkms module of yesterday's ALSA snapshot and rebooting, the headphone issue is indeed solved. I can now plug my headphones into the jack and instantly have sound on them. However, now I have tons of static noises and crackling when playing sound in VLC player or Skype (Firefox HTML5 playback seems to be fine, unless a Skype sound interferes with it). Is there a fix for this? I tried adding the Alsa PPA and installing the latest ALSA package proper, but that didn't have any effect, only the Alsa DKMS package seems to solve the headphone issue.

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• How to make Bluetooth headset appear as a sound device?

  - by torbengb

  I have an onboard VIA sound card but no speakers attached. Instead, I want to use my bluetooth headset as my only sound device, both input and output (mono). I plugged in the bt dongle and was impressed that it was ready to use within seconds. I then paired it to my Jabra BT500 headset. No problems so far. I then went to Sound Preferences but no bluetooth is listed, only the onboard VIA sound card. Question: How can I enable my headset, and make it the default all the time?

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• Looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs <= $5

  - by CyanPrime

  I'm looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs less then, or is $5 usd. I want to have a nice BGM for my engine demo I'm going to release for a game I'm planing on having go commercial. I don't want to spend too much money on it, so my limit is $5 usd. I want it to be at least a 1:00 in length. Where should I look? Or even better, do you have a link to a song that meets the criteria?

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• Where to get sounds for game development for kids [closed]

  - by at.

  I'm teaching kids to program using Ruby and the gaming framework Gosu/Chingu. For the sounds for their games I've been showing them http://www.bfxr.net/. It's decent, but the samples are limited and some of them are pretty cheap (check the explosion, it's like an explosion on a commodore 64 game). Is there an easy resource kids can get the sounds they want? I'm happy to pay some kind of educational license for it.

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• Music player with a few specific requirements

  - by Jordan Uggla

  I am looking for a music player with a few specific requirements: Must have a search function that whittles down results as you type, searching the entire library. Must start playing a song when double clicked, and not continue to another song when that song finishes. Must be approachable and immediately usable by people completely unfamiliar with the program. I think this is mostly covered by the first two requirements being met. I've tried many players but unfortunately every one has failed to meet at least one of the requirements. Rhythmbox meets 1 and 3, but continues to the next search result after the song which was double clicked ends. Banshee is basically the same as Rhythmbox. While it has an option to "Stop when finished" this cannot (as far as I can tell) be made the default when double clicking a song. Audacious (as far as I can tell) fails at 1. Muine meets requirements 1 and 2, but unfortunately I couldn't make the search dialog always shown like it is with Rhythmbox / Banshee which, despite its very simple interface, made Muine incomprehensible to people trying to use it for the first time. Amarok I could not configure to meet requirement 1, but I think it's likely I was just missing something, and with its configurability I'm confident that I can set it up to meet requirements 2 and 3.

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• Difference between Sound and Music

  - by Southpaw Hare

  What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

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