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• ffmpeg hangs when creating a video

  - by FearUs

  I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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• gzip compression

  - by SyAu

  I'm using Weblogic application server and Apache web server in my J2EE environment and planning to implement gzip compression of response. Not sure, whether to implement compression on the Apache server or on the weblogic.

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• Cannot turn on gzip compression in JBoss 5

  - by Vladimir Bezugliy

  I added following file deployers\jbossweb.deployer\server.xml <Connector compression="force" compressionMinSize="512" noCompressionUserAgents="gozilla, traviata" compressableMimeType="text/html,text/xml,image/png,text/css,text/javascript"> </Connector> But fiddler shows that jboss does not compress responses. How to ensure that gzip compression in JBoss is turned on? Is it possible to check it in jmx-console?

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• Is it possible to transcode audio in C# using DirectSound?

  - by Robert Davis

  I want to transcode a lot of audio from its source format to PCM without resampling or messing with the sample size. I figure if Windows Media Player can play the file and it doesn't use a legacy ACM codecs it must be using DirectSound to do so (this is on Windows XP and Windows Server 2k3). So is it possible to access DirectSound from C# and do so? I've tried searching the web but all the examples have been about playback which I have no interest in doing.

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• How do I programmatically extract the audio from a YouTube video?

  - by Skoder

  Hey, I'm trying to create a C# application which allows me to extract just the audio from YouTube videos. I've come across sites that already do that, but I'm not sure how they actually work. What would be the best way to do this programmatically? Thanks for any advice

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• How to get video and audio streams from web cameras with Java?

  - by Ole Jak

  How can I get video and audio streams from web cameras with Java (in a cross-platform way)? For example, we have a computer with 3-4 USB web cameras; we want to get their streams and make them visible in the user interface. How can we perform such a thing?

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• What's the best way of playing media files (esp. audio) with Mono/C#?

  - by supercheetah

  I'm trying to create something that will be playing some sound and music for some things in Mono+C#, but I'm not sure what the best thing will be for that. I'm trying to make it usable with things like Ogg Vorbis, MP3s, and wave files. My primary platform will be Linux, although a cross platform solution would be nice. Anyone have any suggestions for libraries for playing audio files?

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• How to play an audio file ONLY while user is shaking iPhone?

  - by wgpubs

  I've read a number of posts regarding how to detect shakes but I'm not sure: What is currently the best way to detect shakes? and - How to play an audio file ONLY while the user is shaking the iPhone? Anyone have an advice, examples/tutorials or sample code? Thanks

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• iphone sdk: Core Audio How to continue recording to file after user stops recording by leaving the a

  - by Harkonian

  The iPhone's AVAudioRecorder class will not allow you to open an existing file to continue a recording. Instead, it overwrites it. I'd like to know an approach that would allow me to continue recording to an existing file using Core Audio APIs.

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• Where to start learning about audio or video codecs ?

  - by Vamsi

  Hi, I am very much confused to know what happens inside the codecs. I want to learn about the elements inside audio encoders and decoders. Would be very happy if you can provide me some links where i can find some good study material. Thanks precisely i would like to know how the codec parses the a media file.

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• Compiling a list of audio + video players (flash / javascript / otehr) that I can embed into a websi

  - by FiveTools

  I'm compiling a list of audio + video players (flash / javascript / other) that I can embed into a website. flowplayer: http://flowplayer.org/ jw player: http://www.longtailvideo.com/players/ premium beat: http://www.premiumbeat.com/flash_resources/free_flash_music_player/ xspf web player: http://musicplayer.sourceforge.net/ yahoo media player: http://mediaplayer.yahoo.com/ any popular ones I'm missing? (anyone know if I can skin / customize any of them to operate similar to the Windows vista volume control?)

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• Is it possible to capture audio stream in browser via flash/js/silverlight ?

  - by emeer

  Hi, Im working on a project where we need to capture a 5 sec clip of whatever is being played through the users line-out (audio). Is it possible to do this reliably through a browser (using flash/silvelight/js..) ?

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• How to export audio-media from a MOV-file with QuickTime-API?

  - by Berschi

  I want to export the audio-media of a MOV-File with the QuickTime-API and save it to an WAV-File (or something equivalent). How can I do that? I use Windows XP.

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• gzip compression good or bad?

  - by WarDoGG

  I have a server that currently does a lot of processing in my application and the target users are those who have a very good internet connection. The output that is sent from the server is always text/html and we do not use any media (audio/video) only images (static site images like logo,etc). We are experiencing severe performance issues and I wonder if turning off gzip/mod_deflate on the server so that the server would avoid compressing the output. Will this cause an improvement in performance?

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• MinGW tar compression problem

  - by Shiftbit

  I am unable to get the Mingw tar to work with compress files. It does not filter through the proper compression utility. However, tar will work if I manually uncompress the file first. I have tried in both the MSYS shell and Windows cmd. Has anyone had this problem or is it a MinGW bug? For example, this does not work: C:\Users\home\Desktop>tar -tzf wdiff-0.5.tar.gz tar: Cannot use compressed or remote archives tar: Error is not recoverable: exiting now C:\Users\home\Desktop>tar -t -Zgzip -f wdiff-0.5.tar.gz tar: Cannot use compressed or remote archives tar: Error is not recoverable: exiting now C:\Users\home\Desktop>tar -tf wdiff-0.5.tar.gz tar: Hmm, this doesn't look like a tar archive tar: Skipping to next file header tar: Only read 6732 bytes from archive wdiff-0.5.tar.gz tar: Error is not recoverable: exiting now However, this works: gzip -d wdiff-0.5.tar.gz tar -tf wdiff-0.5.tar

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• What is the best nginx compression gzip level?

  - by Chamnap

  I'm using nginx reverse proxy cache with gzip enabled. However, I got some problems from android applications http requests to my rails json web service. It seems when I turn off reverse proxy cache, it works ok because the response header comes without gzip. Therefore, I think the problem caused from gzip. What is the most appropriate level of gzip compression? gzip on; gzip_http_version 1.0; gzip_vary on; gzip_comp_level 6; gzip_proxied any; gzip_types text/plain text/css text/javascript application/javascript application/json application/x-javascript text/xml application/xml application/xml+rss;

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• Comparing zip/compression utilities

  - by Grant Palin

  I've used WinZip, WinRar, and 7zip for packaging and compression. I know the first two are payware, and the last is open source. Despite that, they all seem to serve the same overall purpose. Are there any other distinguishing characteristics that make any of the options stand out? I'm not really looking for a "best" package, but would like to know of noteworthy differences between the common tools. For what it's worth, I do seem to like WinRar. Not sure why, but there it is. If it matters, I'm using Windows 7.

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• how do i determine the image compression algorithm

  - by klijo

  i have a folder containing images. I need to determine the image compression algorithm used in them. Image format is TIFF. Is there a program that i can use to do this ? A program that runs on windows or Linux is ok. When i do a file it gives 100 (2).tif: TIFF image data, little-endian 100.tif: TIFF image data, little-endian It doesnt say which type of algorithm it uses. whether its lossy or lossless and the name of it ?

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• Where can I get a splitter to connect a device with a single 3.5 mm plug into the audio input/output jacks on my laptop?

  - by XinJeisan

  I recently bought the :Hype Retro Handset for Mobile Phone" -- its just a device that looks like a handset to use when chatting on a computer or mobile phone that plugs into the phone/computer with a single 3.5 mm plug. I was hoping to use it on my windows 7 Toshiba laptop. I can hear audio fine through the handset but what I'm saying is not being picked up on the handset. On the box it says "some phones and computers may need additional adapters," so I'm hoping it is possible to get a splitter or something for this to work properly. I did email the parent company (http://dglusa.com/) but I haven't heard from them, and, looking over their website, I doubt I will. I also went to the local radio shack, and the guy said I needed a splitter, but he didn't know where to get one. I can find the kind of splitter I think I need online, but I'm unsure whether they are just for output or can also do input/output.

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• Flash in browsers does not play sound accurately using Pulse network audio

  - by Dave M G

  I use PulseAudio to send sound over the LAN to an audio server. When playing any Flash media in Firefox or Chrome, the sound flutters, as if the volume were going up and down every second. The problem does not exhibit with any other software, and I think it's specific to how Flash interacts with my sound set up. How do I get Flash to play nice with the PulseAudio network sound server? Update I have discovered that I can stop the sound fluttering if I follow these steps: Start a Flash video Run pulseaudio --kill on the server Wait about 7 seconds After this, the PulseAudio server automatically respawns, and the sound in the Flash video is perfect. The problem now, though, is that I have to do this every time I start a Flash video. This is obviously not desireable. So, the question is, how do I make whatever it is that makes the sound work when I go through these steps stick so that I don't have to do them? Also, I've uploaded some PulseAudio log output to Pastebin, taken while attempting to play a Flash video, if that helps. I've tried to get logging details from Flash, but despite installing and enabling Flash for debugging, it has not generated any ouput at all. Details I have uploaded an example video of the problem onto Youtube. In the video you can see the opening of a Ted Talk video, and the sound flutters as it plays. The video also stutters while playing back. Here are my sound device output settings:

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• Jack Audio ubuntu 12.10

  - by Shaneo1

  I used to have Jack Server working with 10.10, 11.04, 11.10 but not 12.04 and now 12.10. I have installed jackd jackd2 qjackctl surfed many forums and even given advice of how to get jack working, but now I am stuck. Tue Nov 27 22:30:46 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:19.960 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Tue Nov 27 22:31:19 2012: Starting jack server... Tue Nov 27 22:31:19 2012: JACK server starting in realtime mode with priority 10 Tue Nov 27 22:31:19 2012: [1m[31mERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Audio device hw:0,0 cannot be acquired...[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Cannot initialize driver[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: JackServer::Open failed with -1[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to open server[0m Tue Nov 27 22:31:21 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:22.047 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Can anyone assist?

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• Double audio cd ripping weirdness

  - by jqno

  Since I installed Ubuntu 12.04, Rhythmbox, Banshee and Sound Juicer have started acting weird around double cd's, and specifically, cd #2 of said double cd. Sometimes, they will show the information of cd #1. Track names, durations, and even count are incorrect. Sometimes, they will first show the tracks for cd #1, then continue onto cd #2 if cd #2 has more tracks than #1. Sound Juicer seems to be unable to find any track durations at all, even for single cd's. Obviously, this is a pain when I'm trying to rip double cd's. And I have a fair number of them, which I want to rip. This happens on both my machines (a slightly aging iMac, and a 1-year-old Sony Vaio). However, on previous versions of Ubuntu, this never happened. All on the same machines. So I suspect 12.04 is using a different lib for extracting audio cd data. Just for kicks, I tried with Linux Mint 13, and there it works correctly, even though it claims to be based on Ubuntu 12.04 and therefore should be using (partially) the same software. So if the Mint guys can fix it, I should be able to do it too, right? So, my question: what changed in 12.04 that could cause this? And more importantly: what can I do to fix it?

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• How to make pulseaudio and ubuntu detect the same audio device as alsa driver

  - by Kiwy

  I use Ubuntu 14.04 x64 and I use gnome-shell on my laptop. I have a Bose companion 5 (which is basically a USB sound system) and a HDMI port, both does work perfectly when I just boot with the cable plugin. However, when my laptop go to sleep or get unplugged from those two outputs, if I plug back the device, I end up without any hardware detection (only the built-in speakers) from pulse and gnome-shell sound output selector while if I use alsamixer, the device look up and ready. gstreamer-properties allow me to select and test effectively any device but while alsa recognize any device on the run, pulse is not capable of handling things correctly, my question is then: How can I make pulse detect and use the same hardware as alsa, or how to remove completely and gracefully pulseaudio (meaning volume applet running in gnome shell) I don't mind if the project implies to recompile half gnome shell if it implies those audio outputs work all the time. Pulse does not list my soundcard when I use command pactl list cards while the module plug&play for sound card is loaded in pactl list modules. I really don't know what to do, the behavior seems pretty random.

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• 12.10 no audio via hdmi and video speeds up

  - by jackson

  I have a laptop with an ati radeon 4200, on 12.04 everything worked fine, since upgrading to 12.10 I cannot get sound over the hdmi. When I switch to hdmi audio the video speeds up to about 2x. I can use the speakers in my laptop and watch video via hdmi with no problems. Things I have tried: Various tutorials to install the AMD/ATI drivers, all of which resulted in low graphics mode. Checked that everything is properly set in alsamixer, the sound utility and - installed pavucontrol and checked everything in there. Verified the output from cat /proc/asound/cards looks normal When I initially upgraded there was a plethora of problems which I believe were due to the old proprietary driver still being used but not compatible, after a few hours trying to fix that I decided just to back up and do a fresh install which works great except for the above stated problem. Any help would be greatly appreciated!! Finally hopefully this hasn't already been answered, I have tried a few different searches on the boards and haven't come up with anything. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0

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• No audio in Google Chrome

  - by Z9iT

  I started with Ubuntu 12.04 Minimal. Then installed only 3 utils sudo apt-get install xorg xinit google-chrome-stable alsa-base alsa-utils alsa-oss I have added google-chrome to .xinitrc file. Used sudo alsamixer to unmute everything using M. Also I am able to hear sound when I run this independently in a terminal sudo aplay /usr/share/sounds/alsa/Front_Center.wav However Google Chrome is not giving any sound output be it on youtube or the same file (/usr/share/sounds/alsa/Front_Center.wav) opened by browsing in chrome. UPDATE : the moment i install some Desktop (display) Manager like gnome or lxde and launch chrome then, the audio is perfect success. However if i kill the xsession and the desktop manager (lxde) AND then start with loading only the chrome (without DM) then again i loose the sound. This makes me wonder that there is something which is not allowing the sound to be loaded into chrome directly, but once the session like lxde loads, then it works flawless. I am thinking that i should rather ask, how to authorize google-chrome to use sound software? Miscellaneous : I am surprised to know that I cannot start google-chrome by sudo command (it asks to be a normal user) && that i cannot start alsamixer as a normal user (i must use sudo alsamixer ) May someone please help what i need to do so that google chrome speaks????

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