Search Results

Search found 5322 results on 213 pages for 'audio chat'.

Page 89/213 | < Previous Page | 85 86 87 88 89 90 91 92 93 94 95 96  | Next Page >

• Networkmanager in systray gone and sound not working after update 13.10

  - by rubo77

  After upgrading my Xubuntu 13.04 to 13.10 I have no sound. I still have sound if I start VLC with sudo mpg123 test.mp3 So it seemd there is a right problem EDIT after adding myself to the group audio with adduser myself audio I could play sounds again from the desktop with VLC But one problem remaining: The systray, usually looking like this: is not working anymore. No audio-settings and no network-manager in the taskbar in XFCE: there is just one small box with nothing in it. When I install stalonetray, There I see the status of wicd and all the other statuses, so the systray seems to be broken.

  Read the article

• 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?

  Read the article

• Cheap sound on speakers - Dell XPS L502X

  - by Rafael

  I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

  Read the article

• Can I get the Waves Maxx speaker effects to work in Ubuntu?

  - by Rafael

  I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

  Read the article

• There's no Sound Mixer menu, missing menu option in Sound Recorder

  - by AlexN

  I am using: -Ubuntu 11.10 -Skype -PS3 Eye Toy camera to input video and sound This setup has been properly working in former Ubuntu releases. To use the mic already built in on the PS3 Eye Toy camera I open de Sound Recorder app (notice: not inside Skype, from inside Skype it is not possible to do this) that is included in Gnome and then I go to FileSound Mixer, from this menu I can choose Gnome to get the input audio from the PS3 Eye Toy, instead of from the Audio-In of the computer. Now in Ubuntu 11.10 this Sound Mixer menu inside Sound Recorder is missing, Gnome says something like this: gnome-volume-control is not installed in the proper directory Note: I have tried this on Unity, Unity 2D, Gnome Classic, Gnome Classic 2D and Gnome Shell. In all of them the problem is the same. What can I do? Basically what I want to do is to be able to tell the computer to get the audio in from the PS3 Camera. Thanks in advance.

  Read the article

• Is there now any way to convert mp3 files to m4a or aac 192kbit?

  - by piedro

  Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

  Read the article

• 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.

  Read the article

• How to Install Linux on my PC

  - by Holic

  Hi i need some help to install the drivers from my pc, on Ubuntu 10.10 i just installed it, and i a newbie on Ubuntu, but i understand a bit of Windows...but i want to try ubuntu and then Maybe change to UBUNTU!!! My hardware: QuadCore Intel Core i7-870, 3266 MHz (24 x 136) Asus P7P55D-E (2 PCI, 3 PCI-E x1, 2 PCI-E x16, 4 DDR3 DIMM, Audio, Gigabit LAN, IEEE-1394) NVIDIA GeForce GTX 480 (1536 MB) nVIDIA HDMI @ nVIDIA GF100 - High Definition Audio Controller VIA VT1828S @ Intel Ibex Peak PCH - High Definition Audio Controller [B-3] DIMM1: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) DIMM3: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) my pc is not connected to the internet with a wire(RJ45) but with a wireless LAn Asus WL-167G-V3(wich i also whant to install if possible) Anything would've help me :) Cheers & Thank you!

  Read the article

• 12.04 sound keeps auto-muting when idle

  - by fali

  I just installed 12.04 on an HP8510W. Everything works fine except for one weird behavior which I have noticed. When ever there is no audio playing, the audio mute indicator on the laptop is on. As soon as I start playing a you tube video the mute indicator turns off and I get sound. Here is my pulse audio output which says that the sink is suspended because it is idle: Welcome to PulseAudio! Use "help" for usage information. list-sinks 1 sink(s) available. index: 0 name: <alsa_output.pci-0000_00_1b.0.analog-stereo> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE I tried running alsamixer, but I don't see the auto-mute option.

  Read the article

• Sound issue in Lubuntu

  - by jvsa90

  I'm recently having a problem in my Lubuntu deskptop: sound through the speakers doesn't seem to work. The funny thing is: it works when I plug in my earphones. I've tried to unmute everything with pavucontrol and alsamixer, but everything seems to be OK. $ sudo aplay -l **** Liste der Hardware-Geräte (PLAYBACK) **** Karte 0: Intel [HDA Intel], Gerät 0: HDA Generic [HDA Generic] Sub-Geräte: 0/1 Sub-Gerät #0: subdevice #0 $ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 02) Subsystem: Acer Incorporated [ALI] Device 034a Flags: bus master, fast devsel, latency 0, IRQ 44 Memory at 58200000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel Can anyone guess what's happening? It has worked until recently and it definitely works in my Windows partition.

  Read the article

• Generic Content Player?

  - by Jantire

  The general idea on the web appears to be that video/audio are to be separated with plain text. By separated, I mean you have a place that plays video/audio and a place that you read text. This is because it is widely understood that they are vastly different. However, audio and video are just another way of communication, just like text. So why do we separate the two even if they are nearly the same thing? Correct me if I'm wrong but, most tutorials are either plain text how-to's (wiki-style) or visual/auditory instructional videos (YouTube). Why aren't the two combined? Or, if it's already been done can someone reply with the link? This might be bordering off-topic and if it is off-topic then please point me to the right place so it won't be. This might also appear to be an obvious question, however I'm not sure if this subject has really been deeply thought-out by more than a few individuals.

  Read the article

• Is it possible to use .data() as a search criteria?

  - by Andrew

  I have a pretty complex chat application going on, and there are multiple chat panes, chat entries, chat submits, etc. going on in the same window. At first I was going to do something like.... <input type="text" class="chattext" id="chattext-42"> <input type="text" class="chattext" id="chattext-93"> <input type="button" class="chatsubmit" id="chatsubmit-42"> <input type="button" class="chatsubmit" id="chatsubmit-93"> ... etc. (of course this is vastly simplified, they'd be in separate divs, separate visibilities, etc) So, when they clicked on a .chatsubmit, it would then get the id of that and find the last two characters for the chat ID. This presents some problems, as it would require rewrites if IDs changed lengths, and seems just plain inelegant to me. I then remembered the .data() facility in jQuery... I thought, maybe I could do it more like this: <input type="text" class="chattext"> ... and add a .data("id", 42) to this one <input type="button" class="chatsubmit"> ... and add a .data("id", 42) So that when they click chatsubmit, it gets the ID, and then finds the chattext with that ID and processes it. But looking at the documentation, I don't see an easy way to search by this. For example, let's say the event target in this case is the chatsubmit with the data('id') of 42... var ID = $(event.target).data('id'); // Sets it to 42 var chattext = ... And here I run into the trouble. How do I find which DOM element matches a class of chattext and a data('id') of 42? Is there any easy method, or do I have to search every .chattext for the one with an id of 42? Or is there another easy way of doing this? I did consider the possibility of the container div having the ID, which would make it, I think,? slightly easier to get. But if this works, it could be dealing with things in other container divs as well, making that not a long-term solution. Edit: Literally seconds after posting this, I found this: http://james.padolsey.com/javascript/extending-jquerys-selector-capabilities/ which includes information on extending the selector to data. So I'll try that out, and in the meantime, is this a completely foolhardy way of handling this?

  Read the article

• 12.04 - sound is laggy when running games through Wine

  - by orzechowskid

  Lenovo U400 Wine 1.5.5 Ubuntu 12.04 with all updates applied I'm experiencing severe (~500ms) audio lag in all games run in Wine. Portal 2, Half-Life, World of Goo, and Fallout are all exhibiting this problem. When I run winecfg though and click the "Test Sound" button at the bottom of the Audio tab, the sound effect appears to play immediately. So I'm not sure what's going on. I don't think it's a problem with PulseAudio by itself since totem videos and Youtube clips both play in perfect sync. Anyone have any ideas on where to start fixing this? thanks! (edit: I thought this was limited to Steam games but I installed a non-Steam game and I now see that's not the case. I get audio lag in other apps too.)

  Read the article

• Don't fire onfocus when selecting text?

  - by Casey Hope

  I'm writing a JavaScript chatting application, but I'm running into a minor problem. Here is the HTML structure: <div id="chat"> <div id="messages"></div> <textarea></textarea> </div> When the user clicks/focuses on the chat box, I want the textbox to be automatically focused. I have this onfocus handler on the chat box: chat.onfocus = function () { textarea.focus(); } This works, but the problem is that in Firefox, this makes it impossible to select text in the messages div, since when you try to click on it, the focus shifts to the textarea. How can I avoid this problem? (Semi-related issues: In Chrome, textarea.focus() doesn't seem to shift the keyboard focus to the textarea; it only highlights the box. IE8 does not seem to respond to the onfocus at all when clicking, even if it tabindex is set. Any idea why?)

  Read the article

• Python: Networked IDLE/Redo IDLE front-end while using the same back-end?

  - by Rosarch

  Is there any existing web app that lets multiple users work with an interactive IDLE type session at once? Something like: IDLE 2.6.4 Morgan: >>> letters = list("abcdefg") Morgan: >>> # now, how would you iterate over letters? Jack: >>> for char in letters: print "char %s" % char char a char b char c char d char e char f char g Morgan: >>> # nice nice If not, I would like to create one. Is there some module I can use that simulates an interactive session? I'd want an interface like this: def class InteractiveSession(): ''' An interactive Python session ''' def putLine(line): ''' Evaluates line ''' pass def outputLines(): ''' A list of all lines that have been output by the session ''' pass def currentVars(): ''' A dictionary of currently defined variables and their values ''' pass (Although that last function would be more of an extra feature.) To formulate my problem another way: I'd like to create a new front end for IDLE. How can I do this? UPDATE: Or maybe I can simulate IDLE through eval()? UPDATE 2: What if I did something like this: I already have a simple GAE Python chat app set up, that allows users to sign in, make chat rooms, and chat with each other. Instead of just saving incoming messages to the datastore, I could do something like this: def putLine(line, user, chat_room): ''' Evaluates line for the session used by chat_room ''' # get the interactive session for this chat room curr_vars = InteractiveSession.objects.where("chatRoom = %s" % chat_room).get() result = eval(prepared_line, curr_vars.state, {}) curr_vars.state = curr_globals curr_vars.lines.append((user, line)) if result: curr_vars.lines.append(('SELF', result.__str__())) curr_vars.put() The InteractiveSession model: def class InteractiveSession(db.Model): # a dictionary mapping variables to values # it looks like GAE doesn't actually have a dictionary field, so what would be best to use here? state = db.DictionaryProperty() # a transcript of the session # # a list of tuples of the form (user, line_entered) # # looks something like: # # [('Morgan', '# hello'), # ('Jack', 'x = []'), # ('Morgan', 'x.append(1)'), # ('Jack', 'x'), # ('SELF', '[1]')] lines = db.ListProperty() Could this work, or am I way off/this approach is infeasible/I'm duplicating work when I should use something already built?

  Read the article

• Drupal module permissions

  - by Trevor Newhook

  When I run the code with an admin user, the module returns what it should. However, when I run it with a normal user, I get a 403 error. The module returns data from an AJAX call. I've already tried adding a 'access callback' = 'user_access'); line to the exoticlang_chat_logger_menu() function. I'd appreciate any pointers you might have. Thanks for the help The AJAX call: jQuery.ajax({ type: 'POST', url: '/chatlog', success: exoticlangAjaxCompleted, data:'messageLog=' + privateMessageLogJson, dataType: 'json' }); The module code: function exoticlang_chat_logger_init(){ drupal_add_js('misc/jquery.form.js'); drupal_add_library('system', 'drupal.ajax'); } function exoticlang_chat_logger_permission() { return array( 'Save chat data' => array( 'title' => t('Save ExoticLang Chat Data'), 'description' => t('Send private message on chat close') ), ); } /** * Implementation of hook_menu(). */ function exoticlang_chat_logger_menu() { $items = array(); $items['chatlog'] = array( 'type' => MENU_CALLBACK, 'page callback' => 'exoticlang_chat_log_ajax', 'access arguments' => 'Save chat data'); //'access callback' => 'user_access'); return $items; } function exoticlang_chat_logger_ajax(){ $messageLog=stripslashes($_POST['messageLog']); $chatLog= 'Drupal has processed this. Message log is: '.$messageLog; $chatLog=str_replace('":"{[{','":[{',$chatLog); $chatLog=str_replace(',,',',',$chatLog); $chatLog=str_replace('"}"','"}',$chatLog); $chatLog=str_replace('"}]}"','"}]',$chatLog); echo json_encode(array('messageLog' => $chatLog)); // echo $chatLog; echo print_r(privatemsg_new_thread(array(user_load(1)), 'The subject', 'The body text')); drupal_exit(); }

  Read the article

• Need help with threads in a client/server

  - by nunos

  For college, I am developing a local relay chat. I have to program a chat server and client that will only work on sending messages on different terminal windows on the same computer with threads and fifos. The fifos part I am having no trouble, the threads part is the one that is giving me some headaches. The server has one thread for receiving commands from a fifo (used by all clients) and another thread for each client that is connected. For each client that is connected I need to know a certain information. Firstly, I was using global variables, which worked as longs as there was only one client connected, which is much of a chat, to chat alone. So, ideally I would have some data like: -nickname -name -email -etc... per client that is connected. However, I don't know how to do that. I could create a client_data[MAX_NUMBER_OF_THREADS] where client_data was a struct with everything I needed to have access to, but this would require to, in every communication between server and client to ask for the id of the client in the array client_data and that does not seem very pratical I could also instantiate a client_data immediately after creating the thread but it would only be available in that block, and that is not very pratical either. As you can see I am in need of a little guidance here. Any comment, piece of code or link to any relevant information is greatly appreciated. Thanks.

  Read the article

• Can I get away with this or is it just too crude and unpractical ?

  - by The_AlienCoder

  I spent the whole of last night searching for a free AspNet web chat control that I could simply drag into my website. Well the search was in vain as I could not find a control that matched my needs i.e List of users, 1 to 1 chat, Ability to kick out users.. In the end I decided to create my own control from scractch. Although it works well on my machine Im concerned that It maybe a little crude and unpractical on a shared hosting enviroment. Basically this is what I did : Created an sql database that stores the chat messages. Wrote the stored procedures and and included a statement that clears old messages Then the 'crude' part : Dragged an update panel and timer control on my page Dragged a Repeater databound to the chat messages table inside the update panel Dragged another update panel and inside it put a textbox and a button Configured the timer control to tick every 5 seconds. ..and then I made it all work like this In the timer tick event I 'refreshed' the messages display by invoking Databind() on my repeater i.e protected void Timer1_Tick(object sender, EventArgs e) { MyRepeater.DataBind(); } Then in my send button click event protected void btnSend_Click(object sender, EventArgs e) { MyDataLayer.InsertMessage(Message, Sender, CurrTime); } Well It works well on my machine and Ive got the other functionalities(users list, kick out user..) to work by simply creating more tables. But like I said it seems a little crude to me. so I need a proffesional opinion. Should I run with this or try another Approach ?

  Read the article

• Use DivX settings to encode to mp4 with ffmpeg

  - by sjngm

  I'm used to use VirtualDub to encode a video to AVI container with DivX-codec (and MP3 for audio). Now I'm planning to use ffmpeg to encode videos to MP4 container with h264-codec. What I've figured out is that I need to use libx264 and one of those presets to make anything work. However, I'm amazed about the video bitrate ffmpeg uses for encoding. What I currently have is this little batch file: @ECHO OFF SETLOCAL SET IN=source.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-fpre "%FFMPEG_PATH%\presets\libx264-lossless_slow.ffpreset" SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %VIDEO% %PRESET% test.mp4 ENDLOCAL With this I tell ffmpeg to use 1978k as the bitrate, but ffmpeg uses 15000k+! I tried other presets, but they don't use my specified bitrate. Here are the presets I have: libx264-baseline.ffpreset libx264-ipod320.ffpreset libx264-ipod640.ffpreset libx264-lossless_fast.ffpreset libx264-lossless_max.ffpreset libx264-lossless_medium.ffpreset libx264-lossless_slow.ffpreset libx264-lossless_slower.ffpreset libx264-lossless_ultrafast.ffpreset ffmpeg version: FFmpeg git-N-29181-ga304071 libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.120. 0 / 52.120. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 79. 0 / 1. 79. 0 libswscale 0. 13. 0 / 0. 13. 0 Note that I don't use the latest version as it has problems with spaces in filenames. Here's what seems to be the full parameter list DivX 6.9.2 uses: -bvnn 1978000 -vbv 218691200,100663296,100663296 -dir "C:\Users\sjngm\AppData\Roaming\DivX\DivX Codec" -w -b 1 -use_presets=1 -preset=10 -windowed_fullsearch=2 -thread_delay=1 What command line parameters would that be for ffmpeg? EDIT: Going with slhck's suggestion I tried a new 32-bit version. I have no idea if that is 0.9 or newer, I can't find that info. ffmpeg version N-36890-g67f5650 libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 I reworked my batch file to look like this (interestingly enough I can't find parameter -vprofile in the documentation): @ECHO OFF SETLOCAL SET IN=VTS_01_1.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-vprofile high -preset veryslow SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %PRESET% %VIDEO% test.mp4 ENDLOCAL I see that it now uses the bitrate properly (thanks to LongNeckbeard for pointing out that the lossless-stuff ignores the bitrate!). Just in case you wonder how I came up with the 1978000, I'm using this formula which I found valid for DivX-files (I'm guessing the bitrate won't change that much for h264): width * height * 25 * 0.22 / 1000 I'm not sure if the 0.22 correlates with the CRF somehow. Overall I forgot to say the I will use a two-pass scenario, which is why I don't use the CRF here. I will try to read more about this. Currently I'm just trying to get something running that shows me that I'm doing something right (ffmpeg isn't the easiest tool to understand ;)). C:\Program Files (x86)\ffmpeg\ffmpeg.exe" -i VTS_01_1.avs -acodec libmp3lame -ab 128000 -vcodec libx264 -vb 1978000 -vprofile high -preset veryslow test.mp4 The output is now: ffmpeg version N-36890-g67f5650 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 16 2012 21:57:13 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, avs, from 'VTS_01_1.avs': Duration: 00:58:46.12, start: 0.000000, bitrate: 0 kb/s Stream #0:0: Video: rawvideo (YV12 / 0x32315659), yuv420p, 576x448, 77414 kb/s, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s File 'test.mp4' already exists. Overwrite ? [y/N] y w:576 h:448 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 05A2C400] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 05A2C400] profile High, level 3.1 [libx264 @ 05A2C400] 264 - core 120 r2120 0c7dab9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1978 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf53.30.100 Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 576x448, q=-1--1, 1978 kb/s, 25 tbn, 25 tbc Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 48000 Hz, 2 channels, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:1 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 3 fps= 1 q=22.0 size= 39kB time=00:00:00.04 bitrate=8063.8kbits/ frame= 8 fps= 2 q=22.0 size= 82kB time=00:00:00.24 bitrate=2801.3kbits/ frame= 13 fps= 3 q=23.0 size= 120kB time=00:00:00.44 bitrate=2229.5kbits/ frame= 16 fps= 4 q=23.0 size= 147kB time=00:00:00.56 bitrate=2156.7kbits/ frame= 20 fps= 4 q=22.0 size= 175kB time=00:00:00.72 bitrate=1987.4kbits/ : video:4387kB audio:273kB global headers:0kB muxing overhead 0.260038% [libx264 @ 05A2C400] frame I:2 Avg QP:19.53 size: 29850 [libx264 @ 05A2C400] frame P:76 Avg QP:22.24 size: 19541 [libx264 @ 05A2C400] frame B:359 Avg QP:25.93 size: 8210 [libx264 @ 05A2C400] consecutive B-frames: 0.5% 0.5% 0.0% 8.2% 17.2% 52.2% 16.0% 5.5% 0.0% [libx264 @ 05A2C400] mb I I16..4: 5.4% 75.3% 19.3% [libx264 @ 05A2C400] mb P I16..4: 1.3% 16.5% 2.2% P16..4: 36.3% 28.6% 12.7% 1.8% 0.2% skip: 0.4% [libx264 @ 05A2C400] mb B I16..4: 0.4% 3.8% 0.3% B16..8: 40.0% 18.4% 4.7% direct:18.5% skip:13.9% L0:45.4% L1:38.1% BI:16.5% [libx264 @ 05A2C400] final ratefactor: 20.35 [libx264 @ 05A2C400] 8x8 transform intra:83.1% inter:68.5% [libx264 @ 05A2C400] direct mvs spatial:99.2% temporal:0.8% [libx264 @ 05A2C400] coded y,uvDC,uvAC intra: 64.9% 83.4% 49.2% inter: 49.0% 50.4% 4.4% [libx264 @ 05A2C400] i16 v,h,dc,p: 25% 22% 27% 26% [libx264 @ 05A2C400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 10% 7% 23% 9% 10% 10% 10%10% 13% [libx264 @ 05A2C400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 11% 13% 9% 12% 11% 10% 9% 12% [libx264 @ 05A2C400] i8c dc,h,v,p: 42% 28% 16% 14% [libx264 @ 05A2C400] Weighted P-Frames: Y:18.4% UV:7.9% [libx264 @ 05A2C400] ref P L0: 29.1% 11.3% 15.7% 7.3% 6.9% 4.9% 5.1% 3.4%3.9% 2.7% 2.8% 1.8% 1.7% 1.2% 1.4% 0.9% [libx264 @ 05A2C400] ref B L0: 68.8% 11.4% 5.5% 2.9% 2.3% 1.9% 1.5% 1.1%1.1% 1.0% 0.9% 0.7% 0.5% 0.3% 0.1% [libx264 @ 05A2C400] ref B L1: 91.9% 8.1% [libx264 @ 05A2C400] kb/s:2055.88 As far as I'm concerned it doesn't look that bad to me.

  Read the article

• Sound Effects/Manipulation?

  - by Adam

  Hello, I am creating an android app that basically records an applies an "Effect" on the audio track then plays it back. I got my app to record an play back but I am stuck an not sure where do go from here. I have been Googling for days now trying to find a open source audio library or some way to change the audio after I record it. I currently have it setup to play back using SoundPool an I't lets me speed up an slow down the audio. I would like to do things like change pitch an add echo etc. I will appreciate any responses because I am totally stumped right now. Thanks Adam

  Read the article

• VU meter implementaion in iphone

  - by Sreelal

  Hi, I am developing an aplication for iphone which records audio and save that audio file .I need to create a UI similar to that in Voice Memo app with VU meter .I implemented codes to record audio,but i have no idea about VU meter implementation.Looking forward for a reply ......Thanks in advance

  Read the article

• avaudioplayer interferes with mpmovieplayer on ipad

  - by user175826

  my app plays video and audio. however, i have a problem where once i play an audio file using avaudioplayer, the video refuses to play. when i play the video first, everything is fine. but if the audio is played first, any time i try to play the video it simply pops up the video player but will not play the actual video (you can use the scroller to go to any point in the video, but no playback will happen). this issue does not come up on the iphone, nor on the ipad simulator. clearly there is some resource conflict here, probably related to the audio, and i'd welcome some input on how to address it.

  Read the article

• Sound File editing in Objective C

  - by Biranchi

  Hi All, I am able to record and create audio files using AudioFileCreateWithURL in the AudioToolbox Framework. I want to figure out if there is any way to edit the .caf sound files. I want to insert another recoreded audio inside the main audio file. Any thoughts or suggestions how to proceed ?? Thanks.

  Read the article

• How do i pipe stdout/stderr in .NET?

  - by acidzombie24

  I want to do something like this ffmpeg -i audio.mp3 -f flac - | oggenc2.exe - -o audio.ogg i know how to do ffmpeg -i audio.mp3 -f flac using the process class in .NET but how do i pipe it to oggenc2? Any example of how to do this (it doesnt need to be ffmpeg or oggenc2) would be fine.

  Read the article

• Avoiding shutdown hook

  - by meryl

  Through the following code I can play and cut and audio file. Is there any other way to avoid using a shutdown hook? The problem is that whenever I push the cut button , the file doesn't get saved until I close the application thanks ...................... void play_cut() { try { // First, we get the format of the input file final AudioFileFormat.Type fileType = AudioSystem.getAudioFileFormat(inputAudio).getType(); // Then, we get a clip for playing the audio. c = AudioSystem.getClip(); // We get a stream for playing the input file. AudioInputStream ais = AudioSystem.getAudioInputStream(inputAudio); // We use the clip to open (but not start) the input stream c.open(ais); // We get the format of the audio codec (not the file format we got above) final AudioFormat audioFormat = ais.getFormat(); // We add a shutdown hook, an anonymous inner class. Runtime.getRuntime().addShutdownHook(new Thread() { public void run() { // We're now in the hook, which means the program is shutting down. // You would need to use better exception handling in a production application. try { // Stop the audio clip. c.stop(); // Create a new input stream, with the duration set to the frame count we reached. Note that we use the previously determined audio format AudioInputStream startStream = new AudioInputStream(new FileInputStream(inputAudio), audioFormat, c.getLongFramePosition()); // Write it out to the output file, using the same file type. AudioSystem.write(startStream, fileType, outputAudio); } catch(IOException e) { e.printStackTrace(); } } }); // After setting up the hook, we start the clip. c.start(); } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play_cut ......................

  Read the article

< Previous Page | 85 86 87 88 89 90 91 92 93 94 95 96  | Next Page >