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  • Letting search engines know that different links to identical pages stress different parts of the page

    - by balpha
    When you follow a permalink to a chat message in the Stack Exchange chat, you get a view of the transcript page for the day that contains the particular message. This message is highlighted in yellow, and the page is scrolled to its position. Sometimes – admittedly rarely, but it happens – a web search will result in such a transcript link. Here's a (constructed, obviously) example: A Google search for strange behavior of the \bibliography command site:chat.stackexchange.com gives me a link to this chat message. This message is obiously unrelated to my query, but the transcript page does indeed contain my search terms – just in a totally different spot. Both the above links lead to the same content, and Google knows this, since both pages have <link rel="canonical" href="/transcript/41/2012/4/9/0-24" /> in their <head>. The only difference between the two links is Which message has the highlight css class?. Is there a way to let Google know that while all three links have the same content, they put an emphasis on a different part of the content? Note that the permalinks on the transcript page already have a #12345 hash to "point" to the relavant chat message, but Google appears to drop it.

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  • Flash audio problem; redirecting to Pulse?

    - by Makaze
    I am having a problem where the sound on flash objects other than YouTube videos are giving a strange distortion. It has a high pitched pulse sound every second or less that resembles skips on a record. I have no idea what the problem is or how to fix it. Does anyone have an idea what I could do to fix this? I am running Xubuntu 11.10. I think I might have redirected everything to pulse using a config file but I cannot seem to find it anywhere. It used the lines 'type pulse' in it. If anyone knows what I am talking about and how to find that file or whether or not it has any relevance I would greatly appreciate it.

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  • Live Chats .NET : le transcript de la deuxième session disponible, prochain rendez-vous avec les experts de Microsoft le 25 juin

    Live Chats .NET : le transcript de la deuxième session disponible prochain rendez-vous avec les experts de Microsoft le 25 juinLe 14 mai dernier, s'est tenue la deuxième session du live chat .NET sur le chat de Developpez.com. Pour rappel, les live chats .NET sont organisés en collaboration avec Microsoft. Ce sont des séances de chat en ligne avec des experts de la société sur le développement d'applications Desktop avec la plate-forme .NET et le futur des outils de développement Microsoft.Les...

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  • Facebook username too long in Pidgin

    - by user41676
    Currently when chatting in pidgin my name that is displayed whenever I send a chat is too long and makes reading the chat difficult and sometimes confusing. Is there a way to make the display name for all of the different protocols be something shorter like a nickname or something? An example my facebook reads like this (01:14:16 PM) [email protected]/df747fe6_4BBB0493F66AE: and I want it to look like this (01:14:16 PM) username:

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  • YSlow Grade F on Add Expires headers - help please

    - by gwmbox
    I am using Joomla for my site and I have included Expires Headers in my htaccess file, however when checking the site via YSlow the grade is still F, the code in the htaccess file for this is <IfModule mod_expires.c> # Enable expiration control ExpiresActive On # Default expiration: Immediate after request ExpiresDefault "now" # CSS and JS expiration: 1 week after request ExpiresByType text/css "now plus 1 week" ExpiresByType application/javascript "now plus 1 week" ExpiresByType application/x-javascript "now plus 1 week" # Image files expiration: 1 month after request ExpiresByType image/bmp "now plus 1 month" ExpiresByType image/gif "now plus 1 month" ExpiresByType image/jpeg "now plus 1 month" ExpiresByType image/jp2 "now plus 1 month" ExpiresByType image/pipeg "now plus 1 month" ExpiresByType image/png "now plus 1 month" ExpiresByType image/svg+xml "now plus 1 month" ExpiresByType image/tiff "now plus 1 month" ExpiresByType image/vnd.microsoft.icon "now plus 1 month" ExpiresByType image/x-icon "now plus 1 month" ExpiresByType image/ico "now plus 1 month" ExpiresByType image/icon "now plus 1 month" ExpiresByType text/ico "now plus 1 month" ExpiresByType application/ico "now plus 1 month" ExpiresByType image/vnd.wap.wbmp "now plus 1 month" ExpiresByType application/vnd.wap.wbxml "now plus 1 month" ExpiresByType application/smil "now plus 1 month" # Audio files expiration: 1 month after request ExpiresByType audio/basic "now plus 1 month" ExpiresByType audio/mid "now plus 1 month" ExpiresByType audio/midi "now plus 1 month" ExpiresByType audio/mpeg "now plus 1 month" ExpiresByType audio/x-aiff "now plus 1 month" ExpiresByType audio/x-mpegurl "now plus 1 month" ExpiresByType audio/x-pn-realaudio "now plus 1 month" ExpiresByType audio/x-wav "now plus 1 month" # Movie files expiration: 1 month after request ExpiresByType application/x-shockwave-flash "now plus 1 month" ExpiresByType x-world/x-vrml "now plus 1 month" ExpiresByType video/x-msvideo "now plus 1 month" ExpiresByType video/mpeg "now plus 1 month" ExpiresByType video/mp4 "now plus 1 month" ExpiresByType video/quicktime "now plus 1 month" ExpiresByType video/x-la-asf "now plus 1 month" ExpiresByType video/x-ms-asf "now plus 1 month" # webfonts ExpiresByType font/truetype "access plus 1 month" ExpiresByType font/opentype "access plus 1 month" ExpiresByType application/x-font-woff "access plus 1 month" ExpiresByType image/svg+xml "access plus 1 month" ExpiresByType application/vnd.ms-fontobject "access plus 1 month" </IfModule> Can someone please tell me why it is not being graded by Yslow? Thanks GW

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  • Setting Ringtone notification from SD card file

    - by sgarman
    My goal is to set the users notification sound from a file that is stored onto the SD card from with in the application. I am using this code: if(path != null){ File k = new File(path, "moment.mp3"); ContentValues values = new ContentValues(); values.put(MediaStore.MediaColumns.DATA, k.getAbsolutePath()); values.put(MediaStore.MediaColumns.TITLE, "My Song title"); values.put(MediaStore.MediaColumns.SIZE, 215454); values.put(MediaStore.MediaColumns.MIME_TYPE, "audio/mp3"); values.put(MediaStore.Audio.Media.ARTIST, "Some Artist"); values.put(MediaStore.Audio.Media.DURATION, 230); values.put(MediaStore.Audio.Media.IS_RINGTONE, false); values.put(MediaStore.Audio.Media.IS_NOTIFICATION, true); values.put(MediaStore.Audio.Media.IS_ALARM, false); values.put(MediaStore.Audio.Media.IS_MUSIC, false); values.put(MediaStore.MediaColumns.DISPLAY_NAME, "Some Name"); //Insert it into the database Uri uri = MediaStore.Audio.Media.getContentUriForPath(k.getAbsolutePath()); Uri newUri = MainActivity.this.getContentResolver().insert(uri, values); RingtoneManager.setActualDefaultRingtoneUri( MainActivity.this, RingtoneManager.TYPE_NOTIFICATION, newUri ); //RingtoneManager.setActualDefaultRingtoneUri(this, RingtoneManager.TYPE_NOTIFICATION, newUri); Toast.makeText(this, "Notification Ringtone Set", Toast.LENGTH_SHORT).show(); } When I run this on the device I keep getting the error: 06-12 15:19:36.741: ERROR/Database(2847): Error inserting is_alarm=false is_ringtone=false artist_id=35 is_music=false album_id=-1 title=My Song title duration=230 is_notification=true title_key=%D%\%%P%H%F%8%%R%<%R%B%4% mime_type=audio/mp3 date_added=1276370376 _display_name=moment.mp3 _size=215454 _data=/mnt/sdcard/Android/data/_MY APP PATH_/files/moment.mp3 06-12 15:19:36.741: ERROR/Database(2847): android.database.sqlite.SQLiteConstraintException: error code 19: constraint failed I have seen others using this technique and I can't find any documentation on which values actually need to be passed in to successfully add the file into the Android system so that it can be set as a notification.

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  • Git: changes not reflecting on other checkouts - huh?

    - by Chad Johnson
    Okay, so, I have my branches (git branch -a): * chat master remotes/origin/HEAD -> origin/master remotes/origin/chat I make changes (still with the 'chat' branch checkout out), commit, and push. I go to my server, on which I have a clone of the repository, and I do a fetch: git getch then I switch to the chat branch: git checkout --track -b chat origin/chat and I event do a pull, just to make sure everything is up to date: git pull and my changes from my other computer are NOT. THERE. What the heck am I doing wrong? If I had hair, I would have pulled it out. Thankfully I am bald. When I try a 'git commit' again, I get this # On branch chat # Changed but not updated: # (use "git add/rm <file>..." to update what will be committed) # (use "git checkout -- <file>..." to discard changes in working directory) # # modified: app/controllers/chat_controller.rb # modified: app/views/dashboard/index.html.erb # modified: app/views/dashboard/layout.js.erb # modified: app/views/layouts/dashboard.html.erb # deleted: app/views/project/.tmp_edit.html.erb.55742~ # deleted: app/views/project/.tmp_edit.html.erb.83482~ # modified: public/stylesheets/dashboard/layout.css # # Untracked files: # (use "git add <file>..." to include in what will be committed) # # .loadpath # .project # config/database.yml # config/environments/development.yml # config/environments/production.yml # config/environments/test.yml # log/ no changes added to commit (use "git add" and/or "git commit -a")

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  • Ubuntu 12.04, Can hear the sound but Sound option in settings shows no sound card

    - by Vivek Srivastava
    I have weired issue. I did a fresh installation of Ubuntu 12.04. Then I installed Nvidia drives for my graphics card. I executed the command "modprobe nvidia" after installing the Nvidia drivers and rebooted. After reboot, sound indicator in top panel is disabled and I can't control the volume from there. I opened Settings Sound and it does not show any sound card installed. However, I can hear the sound. Please help. Output of lspci | grep Audio 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) 01:00.1 Audio device: NVIDIA Corporation GF110 High Definition Audio Controller (rev a1) Output of lsmod | grep snd snd_hda_codec_hdmi 32191 4 snd_hda_codec_realtek 73851 1 snd_hda_intel 33367 0 snd_hda_codec 134156 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 13668 1 snd_hda_codec snd_pcm 97188 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_timer 29990 1 snd_pcm snd 78855 7 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm

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  • Streaming desktop with avconv - severe sound issues

    - by Tommy Brunn
    I'm trying to do some live streaming in Ubuntu 12.10, but I'm having some problems with audio. More specifically, the quality is complete garbage and it's at least 10 seconds out of sync with the video. I'm using an excellent guide found here to set up my loopback devices so that I can combine the desktop audio with the microphone input. It seems to work, as I'm able to stream both audio and video to Twitch.tv. But, as I said, the audio quality is terrible. The microphone audio is very, very low, but if I increase it, I get a horrible garbled sound that is absolutely unbearable. Nothing like that is present during VoIP calls or when recording sound alone with the sound recorder, so it's not an issue with the microphone itself. The entire audio stream is also delayed about 10-15 seconds compared to the video stream. I put together an imgur album of my settings. Here is some example output from when I'm streaming: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [x11grab @ 0x162fd80] device: :0.0+570,262 -> display: :0.0 x: 570 y: 262 width: 1280 height: 720 [x11grab @ 0x162fd80] shared memory extension found [x11grab @ 0x162fd80] Estimating duration from bitrate, this may be inaccurate Input #0, x11grab, from ':0.0+570,262': Duration: N/A, start: 1353181686.735113, bitrate: 884736 kb/s Stream #0.0: Video: rawvideo, bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc [alsa @ 0x163fce0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x163fce0] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'pulse': Duration: N/A, start: 1353181686.773841, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1641ec0] w:1280 h:720 pixfmt:bgra [scale @ 0x1642480] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x165ae80] VBV maxrate unspecified, assuming CBR [libx264 @ 0x165ae80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x165ae80] profile Main, level 3.1 [libx264 @ 0x165ae80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=cbr mbtree=1 bitrate=712 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=712 vbv_bufsize=512 nal_hrd=none ip_ratio=1.25 aq=1:1.00 Output #0, flv, to 'rtmp://live.justin.tv/app/live_23011330_Pt1plSRM0z5WVNJ0QmCHvTPmpUnfC4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuv420p, 852x480, q=-1--1, 712 kb/s, 1k tbn, 30 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 712 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #1:0 -> #0:1 (pcm_s16le -> libmp3lame) Press ctrl-c to stop encoding frame= 17 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 32 fps= 31 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 40 fps= 23 q=29.0 size= 44kB time=0.03 bitrate=13786.2kbits/s dup=frame= 47 fps= 21 q=31.0 size= 93kB time=2.73 bitrate= 277.7kbits/s dup=0frame= 62 fps= 23 q=29.0 size= 160kB time=3.23 bitrate= 406.2kbits/s dup=0frame= 77 fps= 24 q=23.0 size= 209kB time=3.71 bitrate= 462.5kbits/s dup=0frame= 92 fps= 25 q=20.0 size= 267kB time=4.91 bitrate= 445.2kbits/s dup=0frame= 107 fps= 25 q=20.0 size= 318kB time=5.41 bitrate= 482.1kbits/s dup=0frame= 123 fps= 26 q=18.0 size= 368kB time=5.96 bitrate= 505.7kbits/s dup=0frame= 139 fps= 26 q=16.0 size= 419kB time=6.48 bitrate= 529.7kbits/s dup=0frame= 155 fps= 27 q=15.0 size= 473kB time=7.00 bitrate= 553.6kbits/s dup=0frame= 170 fps= 27 q=14.0 size= 525kB time=7.52 bitrate= 571.7kbits/s dup=0 frame= 180 fps= 25 q=-1.0 Lsize= 652kB time=7.97 bitrate= 670.0kbits/s dup=0 drop=32 //Here I stop the streaming video:531kB audio:112kB global headers:0kB muxing overhead 1.345945% [libx264 @ 0x165ae80] frame I:1 Avg QP:30.43 size: 39748 [libx264 @ 0x165ae80] frame P:45 Avg QP:11.37 size: 11110 [libx264 @ 0x165ae80] frame B:134 Avg QP:15.93 size: 27 [libx264 @ 0x165ae80] consecutive B-frames: 0.6% 0.0% 1.7% 97.8% [libx264 @ 0x165ae80] mb I I16..4: 7.3% 0.0% 92.7% [libx264 @ 0x165ae80] mb P I16..4: 0.1% 0.0% 0.1% P16..4: 49.1% 1.2% 2.1% 0.0% 0.0% skip:47.4% [libx264 @ 0x165ae80] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.1% 0.0% 0.0% direct: 0.0% skip:99.9% L0:42.5% L1:56.9% BI: 0.6% [libx264 @ 0x165ae80] coded y,uvDC,uvAC intra: 82.3% 87.4% 71.9% inter: 7.1% 8.4% 7.0% [libx264 @ 0x165ae80] i16 v,h,dc,p: 27% 29% 16% 28% [libx264 @ 0x165ae80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 14% 8% 8% 8% 7% 5% 7% [libx264 @ 0x165ae80] i8c dc,h,v,p: 47% 22% 20% 11% [libx264 @ 0x165ae80] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x165ae80] ref P L0: 96.4% 3.6% [libx264 @ 0x165ae80] kb/s:474.19 Received signal 2: terminating. Any ideas on how I can resolve this? The video delay is perfectly acceptable, so I wouldn't think that it's a network issue that's causing the delay in the audio. Any help would be appreciated.

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  • Need to convert a video file from mp4 to xvid

    - by Shawn
    I checked out the questions with similar titles and didn't find anything that I thought would help. I am attempting to convert a video into an avi, preferably xvid. The video file's Video and Audio Properties are as follows: Video Dimensions: 1280x544 Codec H.264/AVC Framerate: 24 frames per second Bitrate: 774 kpbs Audio Codec: MPEG-4 AAC audio Channels: Stereo Sample Rate: 48000 Hz Bitrate: 32 kpbs I have tried numerous times to convert this into an Xvid codec AVI but I have had no luck successfully getting the audio to sync properly. I am using Openshot to attempt conversion, using the libxvid codec and AVI format, but I am unsure of the proper audio settings I should use. What settings should I use to convert this video with Openshot? If it is not possible with Openshot, or if there is a better application to use, I would be grateful to know that as well.

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  • XUbuntu 12.04 sound becomes distorted on ASUS-computers

    - by Slava Fomin II
    On my XUbuntu 12.04 Desktop from time to time audio becomes distorted, not the audio from some specific applications but every possible sounds are very noisy and barely recognizable. Then i go to: Applications Menu Multimedia PulseAudio Volume Control "Configuration"-tab and change Built-in Audio's Profile from my current profile to something else. After that audio becomes normal, until it breaks again and i have to repeat these steps. It's happening on two different computers: one is an ASUS-based Desktop and other is ASUS notebook. Maybe it's related to some common motherboard audio components. Motherboard is: ASUS P8P67 EVO REV 3.0 Netbook is: ASUS EEPC VX6 Any help will be much appreciated = )

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  • Where to get streaming (live) video and audio from camera example app for Nokia?

    - by Ole Jak
    Where to get streaming (live) video and audio from camera example for Nokia (5800 for ex)? Suppose I want to create some live video streaming service app so I'll have some cool server at the back end. And I know how to do that part. Suppose I have some stand alone app for PCs now I want to go on to mobile devices. So I decided to start from Nokia because I have it and can do with it what I want (Nokia 5800 XpressMusic). So I want to see some sample app grabing audio and video streams from Phone, Synchronizing them, and sending LIVE stream to server. I need any OpenSource sample (JAVA or C or C++) that ll do this or something like this. Where can I get one?

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  • I want to record a screencast of a processing sketch

    - by nathanvda
    I have a created a music visualisation using Processing. I now want to convert that to a video, and the least obtrusive way I could think of is to record a screencast. I figured exporting Processing to video including audio, from within Processing itself, on ubuntu seemed an unsolved issue. Very hard and also could cause timing sync issues (since the music keeps running while images are captured). So move on to the screencast method. Dead-easy, I figured. But I was wrong. First hurdle was to find a way to record the sound from the audio (and not the mic). I did find a tutorial for that here. In short: use gtk-recordmydesktop and pulse audio. But, apparently, what happens: Processing does not use ALSA. When the sound is playing, it does not appear in the Pulse Audio mixer. How can I record the audio now?

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  • Where to get streaming (live) video and audio from camera example app for Android?

    - by Ole Jak
    Where to get streaming (live) video and audio from camera example for Android? Suppose I want to create some live video streaming service app so I'll have some cool server at the back end. And I know how to do that part. Suppose I have some stand alone app for PCs now I want to go on to mobile devices. So I want to see some sample app grabing audio and video streams from Phone, Synchronizing them, encoding somehow, and sending LIVE stream to server. I need any Open-Source sample that will do this or something like this. Where can I get one?

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  • How do I write the audio stream to a memory buffer instead of a file using DirectShow?

    - by yngvedh
    Hi, I have made a sample application which constructs a filter graph to capture audio from the microphone and stream it to a file. Is there any filter which allows me to stream to a memory buffer instead? I'm following the approach outlined in an article on msdn and are currently using the CLSID_FileWriter object to write the audio to file. This works nicely, but I cannot figure out how to write to a memory buffer. Is there such a memory sink filter or do I have to create it myself? (I would prefer one which is bundled with windows XP)

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  • What open source C/C++ audio compression options are there besides LAME MP3?

    - by Ole Jak
    Are there any C/C++ open source audio encoder besides LAME MP3? It doesn't need to be exactly mp3 format, I need a "compressed digital audio file". I do not want to use Lame because it is too big while no programmer can answer a simple question on it (share simple but easily downloadable and readable project containing only needed 2 simple functions... So I'm tired of searching for help with it.. I need something fresh powerful but more readable than this lib I found (mp3stego) ) "I don't want LAME because I am a fighter with its monopoly" Haha..

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