Search Results

Search found 4461 results on 179 pages for 'pic audio'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 28/179 | < Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >

  • Problem Installing Xubuntu 12.04 Audio-Video codecs

    - by Seib
    I used Crouton to install Xubuntu 12.04.4 LTS with the XFCE environment on my HP Chromebook 11. I've gotten it all fully installed and everything; the only thing I'm doing now is the basic setup things, like adding codecs and other things like LibreOffice, VLC, Firefox, Ubuntu Software Center, etc. This information I got from 2 sources: http://www.efytimes.com/e1/fullnews.asp?edid=137269 http://www.binarytides.com/better-xubuntu-14-04/ . I'm currently on the same step at both URLs, which is #6 on Link 1 and #8 on Link 2. Per the articles, which both said the same thing, I typed in sudo apt-get install xubuntu-restricted-extras libavcodec-extra and it didn't do anything. It just kept on saying the same thing: Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package libavcodec-extra I've spend the last hour or so scouring the internet for a solution, for a hint even at what is going on. I don't want to do a clean reinstall, for two reasons: 1) it's takes like 1.5h just to get the croot fully installed, and 2) everything but this out of what I've done so far (up to #6 at Link 1 & up to #8 at Link 2) works except the audio. I've already installed flash, so YouTube works fine. It's just I can't hear any audio. Please help? Thanks in advance. I appreciate all the great help I've been getting from AskUbuntu lately. You all are great.

    Read the article

  • M-Audio Delta starts up at wrong sample rate

    - by steevc
    When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

    Read the article

  • ffmpeg add two audio streams to video

    - by Tossin Hausen
    I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

    Read the article

  • 5.1 sound in Unity3d 3.5

    - by N0xus
    I'm trying to implement 5.1 surround sound in my game. I've set Unity's AudioManager to a default of 5.1 surround and loaded in a 6 channel audio clip that should play a sound in each of the different audio spots. However, when I go to run my game, all I get is flat sound coming out of my front two speakers. Even then, these don't play the sound they should (front speaker should play "front speaker" right should play "right speaker" and so). Both speakers just end up playing the entire sound file. I've tried looking to see if there is a parameter that I have missed, but information on how to set up 5.1 sound in Unity is lacking (or my google skills aren't that good) and I can't get it to work as intended. Could someone please either tell me what I'm missing, or point me in the right direction? My audio source is situated at point (0, 0, 0) with my camera also being in the same point. I've moved about the scene but the same thing happens as I've already described.

    Read the article

  • Alsa devices under Wine

    - by Roberto Aloi
    Hi all, I'm running OpenSuse 11.2 and Wine 1.1.28. Even if audio is perfectly working fine for me (Skype, Banshee, etc), when I try to configure audio for Wine (to use Spotify) I cannot hear anything from the audio test. In the winecfg audio tab, ALSA is checked, but no devices are available. I tried to run alsaconf (it needs root permissions) but it returns: No supported PnP or PCI card found No legacy drivers available, either. Any idea?

    Read the article

  • Audio stream mangement in Linux

    - by User1
    I have a very complicated audio setup for a project. Here's what we have: 3 applications playing sound 2 applications recording sound 2 sound cards I really don't really have the code to any of these applications. All I want to do is monitor and control the audio streams. Here are a few examples of operations I'd like to do while the applications are running: Mute one of the incoming audio streams. Have one of the incoming audio streams do a "solo" (be the only stream that can "talk"). Get a graph (about 30 seconds worth) of the audio that each stream produced. Send one of the audio streams to soundcard #1, but all three audio streams to soundcard #2. I would likely switch audio streams every 2 minutes or so with one of the operations listed above. A GUI would be preferred. I started looking at the sound systems in Linux and it gets extremely complex and I feel like there have been many new advances in the past few years. I see jack, pulseaudio, artsd, and several other packages. They all have some promise but where should I start? Is there something someone already built that can help?

    Read the article

  • Is there an easy way to copy an audio CD in Mac OS X?

    - by Bob D
    (not a commercial CD). I did some recordings of a band years ago and ran into one of the band members who asked me if I could make copies. I assumed that this would be easy. I know that I can rip the CD into iTunes and then burn a new CD, but I have two optical drives available, is there a way to simply copy the CD from one drive to the other in one step?

    Read the article

  • Ubuntu: how to get audio to work in both Spotify (under Wine) and Flash (in Firefox)?

    - by Jonik
    I'm running Spotify on Linux using Wine. Sound worked great (even though the sound test in winecfg failed!), until I installed alsa-oss package yesterday to get Flash sound working in Firefox. Now Spotify says: "There is a problem with your sound card. Spotify can't play music." So the question is, how to get the sound in Spotify working again, so that it also keeps working in Flash & Firefox? Tweak some ALSA settings? Spotify settings? Add/remove some packages? By the way, curiously, now that sound doesn't work in Spotify, winecfg's "Test Sound" does work! This is Ubuntu 8.04 (Hardy). Sound card / driver is probably an integrated AC'97. Please mention if any additional information about the system is needed! Update: I have Flash 10 installed (outside the packaging system, using $MOZ_PLUGIN_PATH env variable), but also had Flash 9 from flashplugin-nonfree package - and the earlier version was being used by Firefox! Based on what Mike Arthur said about Flash and alsa-oss, I removed the older Flash (flashplugin-nonfree package) and alsa-oss - and Flash sound still works, which is nice. But for some reason Spotify still doesn't play sound, even though things should now be like they were originally... Update 2: Got it working, all smoothly, finally.

    Read the article

  • Audio recording, a tool for human-aided drum quantizing.

    - by basilio.mp
    I have this situation: the drummer records the track (8 tracks in a multitrack session). Now, how do I check how distant are the recorded beats from their theoretical position i.e.: there is always some error in human recorded tracks, but is there any software that can show me the ideal (theoretical, quantized) beat and the recorded one and could alert me if the error is too big. P.S.: I'm searching for a standalone tool, or for a plugin that can work with Adobe Audition 3 or Nuendo 3.

    Read the article

  • Is the Zune HD's audio card better or worse than the iPod touch's?

    - by MatthewThepc
    Firstly, if this is the wrong site to ask this question I apologize, but I didn't see one for "music players" on the stack exchange website :) After reading a few people online say that music playing from a Zune HD sounds better to them than that on an iPod touch, I was wondering whether there's any truth to that? From what I can tell, the Zune HD uses a Wolfson Microelectronics WM8352, while the first-generation iPod Touch (which the Zune HD was competing with) used a Wolfson Microelectronics WM8758BG, and newer models use the Cirrus Logic CS4398 and CS42L61. Which ones are better (to make the question less subjective, let's say in terms of quality, range, & accuracy of output)? Admittedly, I have almost no idea how everything compares and works together, but it would seem to me that, just by looking at the version numbers, the iPod has been better since it's launch. Is there anything else that effects sound quality? Thanks!

    Read the article

  • How to fix audio/game stuttering in Google Chrome's Flash plug-in?

    - by Simon Belmont
    I'm having an issue. Windows XP, running the latest Chrome 23 build. I'm using Flash 11.5 built into Chrome (Pepper Flash). It runs horribly. Chrome 22 did not have this issue as far as I recall. What a shame. YouTube videos stutter badly and after a while, they begin to lag and lose sync with the video. I disabled Pepper Flash and tested HTML5 video in YouTube and it was smooth as glass. Additionally, certain Flash based games are almost unusable now. The plug-in is using 100% CPU and it lags horribly in these games. Google/Adobe, please fix this. I shouldn't have to disable the built-in Flash plug-in (with added sandboxing security) and use regular Flash to resolve this. Short of waiting for an update to Chrome, does anyone have a better solution to fixing this? I am all ears.

    Read the article

  • Using kAudioSessionProperty_OtherMixableAudioShouldDuck on iPhone

    - by Cliff
    Hello, I'm trying to get consistant behavior out of the kAudioSessionProperty_OtherMixableAudioShouldDuck property on the iPhone to allow iPod music blending and I'm having trouble. At the start of my app I set an Ambient category like so: -(void) initialize { [[AVAudioSession sharedInstance] setCategory: AVAudioSessionCategoryAmbient error: nil]; } Later on when I attempt to play audio I set the duck property using the following method: //this will crossfade the audio with the ipod music - (void) toggleCrossfadeOn:(UInt32)onOff { //crossfade the ipod music AudioSessionSetProperty(kAudioSessionProperty_OtherMixableAudioShouldDuck,sizeof(onOff),&onOff); AudioSessionSetActive(onOff); } I call this passing a numeric "1" just before playing audio like so: [self toggleCrossfadeOn:1]; [player play]; I then call the crossfade method again passing a zero when my app's audio completes using a playback is stopping callback like so: -(void) playbackIsStoppingForPlayer:(MQAudioPlayer*)audioPlayer { NSLog(@"Releasing player"); [audioPlayer release]; [self toggleCrossfadeOn:0]; } In my production app this exact code works as expected, causing the ipod to fade out just before playing my app's audio then fade back in when the audio finishes playing. In a new project I recently started, I get different behavior. The iPod audio fades down and never fades back in. In my production app I use the AVAudioPlayer where in my new app I've written a custom audio player that uses audio queues. Could somebody help me understand the differences and how to properly use this API?

    Read the article

  • Audio on beagleboard xM

    - by Francesco
    I am running Ubuntu 12.04 Precise for omap on a beagleboard xM. I am trying to setup audio. No soundcards are listed in /proc/asound/cards. Alsamixer fails with cannot open mixer: No such file or directory Under /dev/snd/ I have only: seq timer Driver's name should be omap3beagle - twl4030. I am using alsa 1.0.24 that is installed by default with Ubuntu 12.04. I've googled a lot but I have not found anything yet. Thanks

    Read the article

  • Video Conferencing Vs. Audio Web Conferencing

    Organizations are generally confused whether to use audio web conferencing or video conferencing to communicate with their clients, stakeholders, members and all other relevant individuals. Both type... [Author: Zaibatt Zaki - Computers and Internet - August 24, 2009]

    Read the article

  • Empathy Audio Call Option Disabled 13.04

    - by PJ Singh
    Recently, I noticed that the Audio Call option is now disabled (greyed out) in Empathy when I right click on an available GoogleTalk contact. This option used to be available a few days ago. I do have libtelepathy-farstream3, libfarstream-0.1-0, and libfarstream-0.2-2 installed on Ubuntu 13.04 x64 with kernel 3.8.0-27. (Note, I just upgraded the kernel to 3.8.0-29 and still have this issue). Has anyone else experienced this recently, and if so, is there a resolution?

    Read the article

  • I can't change audio/volume preferences?

    - by genesis
    When I click to sound icon on the panel, I have 3 options: "Mute all" is gray and could not be clicked Slider - I can slide but it DOESNT change anything Preferences - Shows this (waiting for the response from audio device), but it doesn't show anything for more than a hour This is from aplay -l : root@fb:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC272 Analog [ALC272 Analog] Subdevices: 1/1 Subdevices #0: subdevice #0 karta 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevices #0: subdevice #0 root@fb:~# What's wrong?

    Read the article

  • Audio Problems with my Mac running Ubuntu 12.04

    - by Tomtom
    I've got a mac (desktop around 2008). I installed Ubuntu 11.10, then remembered I had 12.04 on a disc. I installed 12.04 and now the audio will not work. I've tried all the killallpulse and looked briefly at the Alsa thing. I put in speakers and they only work with the headphone setting. When i put in cat /proc/asound/cards I get: 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x50600000 irq 47 I do have the Mac startup noise but no sound in Ubuntu

    Read the article

  • .NET Rocks! Internet Audio

    - by Editor
    NET Rocks! is a weekly talk show for anyone interested in programming on the Microsoft .NET platform. The shows range from introductory information to hardcore geekiness. Many of their listeners download the MP3 files and burn CDs for the commute to and from work, or simply listen on a portable media player.  Download .NET Rocks! audio.

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

< Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >