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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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  • How to handle URLs with diacritic characters

    - by user359650
    I am wondering how to handle URLs which correspond to strings containing diacritic (á, u, ´...). I believe what we're seeing mostly are URLs where diacritic characters where converted to their closest ASCII equivalent, for instance Rånades på Skyttis i Ö-vik converted to ranades-pa-skyttis-i-o-vik. However depending on the corresponding language, such conversion might be incorrect. For instance in German, ü should be converted to ue and not just u, as seen with the below URL representing the Bayern München string as bayern-muenchen: http://www.bundesliga.de/en/liga/clubs/fc-bayern-muenchen/index.php However what I've also noticed, is that browsers can render non-ASCII characters when they are percent-encoded in the URL, which is the approach Wikipedia has chosen, for instance http://de.wikipedia.org/wiki/FC_Bayern_M%C3%BCnchen which is rendered as: Therefore I'm considering the following approach for creating URL slugs: -(1) convert strings while replacing non-ASCII characters to their recommended ASCII representation: Bayern München - bayern-muenchen -(2) also convert strings to percent encoding: Bayern München - bayern_m%C3%BCnchen -create a 301 redirect from version (1) to version (2) Version (1) URLs could be used for marketing purposes (e.g. mywebsite.com/bayern-muenchen) but the URLs that would end being displayed in the browser bar would be version (2) URLs (e.g. mywebsite.com/bayern-münchen). Can you foresee particular problems with this approach? (Wikipedia is not doing it and I wonder why, apart from the fact that they don't need to market their URLs)

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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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  • My sound stopped working today, how can I fix it?

    - by Oli
    This seems to be a problem with pulseaudio. I was logged in over VNC on my phone and started playing a video this caused X to crash (as sometimes happens). I restarted and suddenly the sound doesn't work. I have a Intel HDA/Realtek ALC889 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller alsamixer is detecting this just fine. PulseAudio doesn't detect this alsa device so is using auto_null as the default sink (logs below). When I properly kill PulseAudio (tell it not to auto-start) direct ALSA communication with the sound card works just fine. speaker-test, for example, works. So the hardware and ALSA layers are fine IMO. In the logs, it seems that the card might be "busy" but I really don't know how or why it would be now (and never before). Is there an ALSA lock file somewhere that it still there because of my crash? I just ran sudo fuser /dev/snd/* and saw this: oli@bert:~$ sudo fuser /dev/snd/* /dev/snd/controlC0: 1884 /dev/snd/pcmC0D0c: 1884m /dev/snd/timer: 1884 A look at the process list (ps aux | grep 1884) tells me process 1884 is arecord -c 1 -f S16_LE -r 8000 -t raw. No idea what this is or why it's running. When I try and kill arecord (as root), it just respawns and rebinds on the hardware. I'm in a very annoying situation where I don't know what is going on and don't know how to find out. I'm open to all suggestions to get this working again. Fire away. And here's what I get when I stop PA auto-loading, kill it and then start it with -vvvv. oli@bert:~$ pulseaudio -vvvvv I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted D: core-rtclock.c: Timer slack is set to 50 us. D: core-util.c: RealtimeKit worked. I: core-util.c: Successfully gained nice level -11. I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: main.c: Running on host: Linux x86_64 2.6.38-rc3 #1 SMP Tue Feb 1 10:53:04 GMT 2011 D: main.c: Found 8 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimised build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 8310740c4729ef474fe5ecec4bbf5a6b. I: main.c: Session ID is 8310740c4729ef474fe5ecec4bbf5a6b-1297338553.571075-1050119523. I: main.c: Using runtime directory /home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-runtime. I: main.c: Using state directory /home/oli/.pulse. I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Enjoy ol' chap! I: cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes.tdb' I: module-device-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes'. I: module.c: Loaded "module-device-restore" (index: #0; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes.tdb' I: module-stream-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes'. I: module.c: Loaded "module-stream-restore" (index: #1; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database.tdb' I: module-card-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database'. I: module.c: Loaded "module-card-restore" (index: #2; argument: ""). I: module.c: Loaded "module-augment-properties" (index: #3; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards. I: module.c: Loaded "module-udev-detect" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': success D: dbus-util.c: Successfully connected to D-Bus system bus ba7c9a1f90b3d49d930bca2100000015 as :1.62 D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: bluetooth-util.c: Bluetooth daemon is apparently not available. I: module.c: Loaded "module-bluetooth-discover" (index: #5; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-gconf.so': success I: module.c: Loaded "module-gconf" (index: #8; argument: ""). I: module-default-device-restore.c: Saved default sink 'auto_null' not existant, not restoring default sink setting. I: module-default-device-restore.c: Saved default source 'auto_null.monitor' not existant, not restoring default source setting. I: module.c: Loaded "module-default-device-restore" (index: #9; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #10; argument: ""). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. I: sink.c: Created sink 0 "auto_null" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: sink.c: device.description = "Dummy Output" I: sink.c: device.class = "abstract" I: sink.c: device.icon_name = "audio-card" D: core-subscribe.c: Dropped redundant event due to change event. I: source.c: Created source 0 "auto_null.monitor" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: source.c: device.description = "Monitor of Dummy Output" I: source.c: device.class = "monitor" I: source.c: device.icon_name = "audio-input-microphone" D: module-null-sink.c: Thread starting up I: module.c: Loaded "module-null-sink" (index: #11; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: module.c: Loaded "module-intended-roles" (index: #13; argument: ""). D: module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1 I: module.c: Loaded "module-console-kit" (index: #15; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). D: dbus-util.c: Successfully connected to D-Bus session bus efbffc6788fad56cfd64d40c00000018 as :1.182 D: main.c: Got org.pulseaudio.Server! I: main.c: Daemon startup complete. I: client.c: Created 1 "Native client (UNIX socket client)" I: client.c: Created 2 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon D: core-subscribe.c: Dropped redundant event due to change event. I: module-suspend-on-idle.c: Sink auto_null idle for too long, suspending ... D: sink.c: Suspend cause of sink auto_null is 0x0004, suspending Note the one section that seems to find the hardware but says it's busy (no idea if this is relevant). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards.

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  • Dealing with numerous, simultaneous sounds in unity

    - by luxchar
    I've written a custom class that creates a fixed number of audio sources. When a new sound is played, it goes through the class, which creates a queue of sounds that will be played during that frame. The sounds that are closer to the camera are given preference. If new sounds arrive in the next frame, I have a complex set of rules that determines how to replace the old ones. Ideally, "big" or "important" sounds should not be replaced by small ones. Sound replacement is necessary since the game can be fast-paced at times, and should try to play new sounds by replacing old ones. Otherwise, there can be "silent" moments when an old sound is about to stop playing and isn't replaced right away by a new sound. The drawback of replacing old sounds right away is that there is a harsh transition from the old sound clip to the new one. But I wonder if I could just remove that management logic altogether, and create audio sources on the fly for new sounds. I could give "important" sounds more priority (closer to 0 in the corresponding property) as opposed to less important ones, and let Unity take care of culling out sound effects that exceed the channel limit. The only drawback is that it requires many heap allocations. I wonder what strategy people use here?

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  • How to resolve a NULL cString crash

    - by hanumanDev
    I'm getting a crash with the following encoding fix I'm trying to implement: // encoding fix NSString *correctStringTitle = [NSString stringWithCString:[[item objectForKey:@"main_tag"] cStringUsingEncoding:NSISOLatin1StringEncoding] encoding:NSUTF8StringEncoding]; cell.titleLabel.text = [correctStringTitle capitalizedString]; my crash log output states: *** Terminating app due to uncaught exception 'NSInvalidArgumentException', reason: '*** +[NSString stringWithCString:encoding:]: NULL cString' thanks for any help

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  • Convert .net String object into base64 encoded string

    - by chester89
    I have a question, which Unicode encoding to use while encoding .NET string into base64? I know strings are UTF-16 encoded on Windows, so is my way of encoding is the right one? public static String ToBase64String(this String source) { return Convert.ToBase64String(Encoding.Unicode.GetBytes(source)); }

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  • How to detect generation loss of a transcoded audio.

    - by The Rook
    Lets say you have a 96 kbit mp3 and you Transcode the file into a 320 kbit mp3. How could you programmatically detect the original bit rate or quality? Generation loss is created because each time a lossy algorithm is applied new information will be deemed "unnecessary" and is discarded. How could an algorithm use this property to detect the transcoding of audio. 128 kbps LAME mp3 transcoded to 320 kbps LAME mp3 (I Feel You, Depeche Mode) 10.8 MB. This image was taken from the bottom of this site. The 2 tracks above look nearly identical, but the difference is enough to support this argument.

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  • What language/API to use for a standalone live-input audio visualizer app?

    - by knuckfubuck
    I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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  • Streaming audio to mobile phones, what technology to use ?

    - by Alx
    I'm planning on building an application where audio media is going to be streamed to the mobile phone for the user to listen. The targets are smartphones: iPhone/Blackberry/Android/(J2ME ?). I see that streaming on iPhone has to be done with HTTP Live streaming, but I don't see it supported by other platforms. Should I broadcast the streams via rstp ? http ? Is there any way to use a unified solution for all the different mobile platform ? If anyone already had to go through this, help would be gratly appreciated.

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  • Android: Using MediaRecorder to crop an existing audio file?

    - by user141146
    Hi, I'd like to take an existing mp3 file located on an SD card and arbitrarily crop it (e.g. crop from 0:12 to 1:14 in a 3 minute song). The only class that I've seen that seems remotely relevant to do this is the MediaRecorder class. My 'hope' would be to "record" an existing file like this: MediaRecorder recorder = new MediaRecorder(); recorder.setAudioSource(###some magical way of specifying an existing file??###); But this obviously doesn't work (setAudioSource() takes an int and seems to default to the phone's microphone). Is there a class or an approach that can be used to crop audio on the phone itself? TKS!!

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  • How does one record audio from a Javascript based webapp?

    - by username
    I'm trying to write a web-app that records WAV files (eg: from the user's microphone). I know Javascript alone can not do this, but I'm interested in the least proprietary method to augment my Javascript with. My targeted browsers are Firefox for PC and Mac (so no ActiveX). Please share your experiences with this. I gather it can be done with Flash (but not as a WAV formated file). I gather it can be done with Java (but not without code-signing). Are these the only options? @dominic-mazzoni I'd like to record the file as a WAV because because the purpose of the webapp will be to assemble a library of good quality short soundbites. I estimate upload will be 50 MB, which is well worth it for the quality. The app will only be used on our intranet. UPDATE: There's now an alternate solution thanks to JetPack's upcoming Audio API: See https://wiki.mozilla.org/Labs/Jetpack/JEP/18

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  • is there any Simple opensource live audio streaming Server using WCF? (see specification below)

    - by Ole Jak
    is there any Simple opensource live audio streaming Server using WCF? I need it to have simple structure: it should listen to some url format like http://example.com/service/stream?write&id=ANY_STRING and if any data comes to such address format it'll start making it avaliable by something like this http://example.com/service/stream?read&id=ANY_STRING Main thing here to be able to stream live data thru WCF service not buffering it just sharing stream. So can please any one help me with such idea? I think not only I have seen such problem with WCF alot on different sites so answer will help the WCF comunyty alot. I hope. BTW: I know some people say WCF is not prepared for live streaming over bacikHTTPbinding but hey! We all need it to, and we ask MS alot so some day they'll make it beter and we all want to be prepared for it.

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  • How to detect UTF-8-based encoded strings [closed]

    - by Diego Sendra
    A customer of asked us to build him a multi-language based support VB6 scraper, for which we had the need to detect UTF-8 based encoded strings to decode it later for proper displaying in application UI. It's necessary to point out that this need arises based on VB6 limitations to natively support UTF-8 in its controls, contrary to what it happens in .NET where you can tell a control that it should expect UTF-8 encoding. VB6 natively supports ISO 8859-1 and/or Windows-1252 encodings only, for which textboxes, dropdowns, listview controls, others can't be defined to natively support/expect UTF-8 as you can do in .NET considering what we just explained; so we would see weird symbols such as é, è among others, making it a whole mess at the time of displaying. So, next function contains whole UTF-8 encoded punctuation marks and symbols from languages like Spanish, Italian, German, Portuguese, French and others, based on an excellent UTF-8 based list we got from this link - Ref. http://home.telfort.nl/~t876506/utf8tbl.html Basically, the function compares if each and one of the listed UTF-8 encoded sentences, separated by | (pipe) are found in our passed string making a substring search first. Whether it's not found, it makes an alternative ASCII value based search to get a match. Say, a string like "Societé" (Society in english) would return FALSE through calling isUTF8("Societé") while it would return TRUE when calling isUTF8("SocietÈ") since È is the UTF-8 encoded representation of é. Once you got it TRUE or FALSE, you can decode the string through DecodeUTF8() function for properly displaying it, a function we found somewhere else time ago and also included in this post. Function isUTF8(ByVal ptstr As String) Dim tUTFencoded As String Dim tUTFencodedaux Dim tUTFencodedASCII As String Dim ptstrASCII As String Dim iaux, iaux2 As Integer Dim ffound As Boolean ffound = False ptstrASCII = "" For iaux = 1 To Len(ptstr) ptstrASCII = ptstrASCII & Asc(Mid(ptstr, iaux, 1)) & "|" Next tUTFencoded = "Ä|Ã…|Ç|É|Ñ|Ö|ÃŒ|á|Ã|â|ä|ã|Ã¥|ç|é|è|ê|ë|í|ì|î|ï|ñ|ó|ò|ô|ö|õ|ú|ù|û|ü|â€|°|¢|£|§|•|¶|ß|®|©|â„¢|´|¨|â‰|Æ|Ø|∞|±|≤|≥|Â¥|µ|∂|∑|âˆ|Ï€|∫|ª|º|Ω|æ|ø|¿|¡|¬|√|Æ’|≈|∆|«|»|…|Â|À|Ã|Õ|Å’|Å“|–|—|“|â€|‘|’|÷|â—Š|ÿ|Ÿ|â„|€|‹|›|ï¬|fl|‡|·|‚|„|‰|Â|Ú|Ã|Ë|È|Ã|ÃŽ|Ã|ÃŒ|Ó|Ô||Ã’|Ú|Û|Ù|ı|ˆ|Ëœ|¯|˘|Ë™|Ëš|¸|Ë|Ë›|ˇ" & _ "Å|Å¡|¦|²|³|¹|¼|½|¾|Ã|×|Ã|Þ|ð|ý|þ" & _ "â‰|∞|≤|≥|∂|∑|âˆ|Ï€|∫|Ω|√|≈|∆|â—Š|â„|ï¬|fl||ı|˘|Ë™|Ëš|Ë|Ë›|ˇ" tUTFencodedaux = Split(tUTFencoded, "|") If UBound(tUTFencodedaux) > 0 Then iaux = 0 Do While Not ffound And Not iaux > UBound(tUTFencodedaux) If InStr(1, ptstr, tUTFencodedaux(iaux), vbTextCompare) > 0 Then ffound = True End If If Not ffound Then 'ASCII numeric search tUTFencodedASCII = "" For iaux2 = 1 To Len(tUTFencodedaux(iaux)) 'gets ASCII numeric sequence tUTFencodedASCII = tUTFencodedASCII & Asc(Mid(tUTFencodedaux(iaux), iaux2, 1)) & "|" Next 'tUTFencodedASCII = Left(tUTFencodedASCII, Len(tUTFencodedASCII) - 1) 'compares numeric sequences If InStr(1, ptstrASCII, tUTFencodedASCII) > 0 Then ffound = True End If End If iaux = iaux + 1 Loop End If isUTF8 = ffound End Function Function DecodeUTF8(s) Dim i Dim c Dim n s = s & " " i = 1 Do While i <= Len(s) c = Asc(Mid(s, i, 1)) If c And &H80 Then n = 1 Do While i + n < Len(s) If (Asc(Mid(s, i + n, 1)) And &HC0) <> &H80 Then Exit Do End If n = n + 1 Loop If n = 2 And ((c And &HE0) = &HC0) Then c = Asc(Mid(s, i + 1, 1)) + &H40 * (c And &H1) Else c = 191 End If s = Left(s, i - 1) + Chr(c) + Mid(s, i + n) End If i = i + 1 Loop DecodeUTF8 = s End Function

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  • UITableView's NSString memory leak on iphone when encoding with NSUTF8StringEncoding

    - by vince
    my UITableView have serious memory leak problem only when the NSString is NOT encoding with NSASCIIStringEncoding. - (UITableViewCell *)tableView:(UITableView *)tableView cellForRowAtIndexPath:(NSIndexPath *)indexPath { static NSString *CellIdentifier = @"cell"; UILabel *textLabel1; UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:CellIdentifier]; if (cell == nil) { cell = [[[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault reuseIdentifier:CellIdentifier] autorelease]; textLabel1 = [[UILabel alloc] initWithFrame:CGRectMake(105, 6, 192, 22)]; textLabel1.tag = 1; textLabel1.textColor = [UIColor whiteColor]; textLabel1.backgroundColor = [UIColor blackColor]; textLabel1.numberOfLines = 1; textLabel1.adjustsFontSizeToFitWidth = NO; [textLabel1 setFont:[UIFont boldSystemFontOfSize:19]]; [cell.contentView addSubview:textLabel1]; [textLabel1 release]; } else { textLabel1 = (UILabel *)[cell.contentView viewWithTag:1]; } NSDictionary *tmpDict = [listOfInfo objectForKey:[NSString stringWithFormat:@"%@",indexPath.row]]; textLabel1.text = [tmpDict objectForKey:@"name"]; return cell; } -(void) readDatabase { NSArray *documentPaths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString *documentsDir = [documentPaths objectAtIndex:0]; databasePath = [documentsDir stringByAppendingPathComponent:[NSString stringWithFormat:@"%@",myDB]]; sqlite3 *database; if(sqlite3_open([databasePath UTF8String], &database) == SQLITE_OK) { const char sqlStatement = [[NSString stringWithFormat:@"select id,name from %@ order by orderid",myTable] UTF8String]; sqlite3_stmt *compiledStatement; if(sqlite3_prepare_v2(database, sqlStatement, -1, &compiledStatement, NULL) == SQLITE_OK) { while(sqlite3_step(compiledStatement) == SQLITE_ROW) { NSString *tmpid = [NSString stringWithUTF8String:(char *)sqlite3_column_text(compiledStatement, 0)]; NSString *tmpname = [NSString stringWithCString:(const char *)sqlite3_column_text(compiledStatement, 1) encoding:NSUTF8StringEncoding]; [listOfInfo setObject:[[NSMutableDictionary alloc] init] forKey:tmpid]; [[listOfInfo objectForKey:tmpid] setObject:[NSString stringWithFormat:@"%@", tmpname] forKey:@"name"]; } } sqlite3_finalize(compiledStatement); debugNSLog(@"sqlite closing"); } sqlite3_close(database); } when i change the line NSString *tmpname = [NSString stringWithCString:(const char *)sqlite3_column_text(compiledStatement, 1) encoding:NSUTF8StringEncoding]; to NSString *tmpname = [NSString stringWithCString:(const char *)sqlite3_column_text(compiledStatement, 1) encoding:NSASCIIStringEncoding]; the memory leak is gone i tried NSString stringWithUTF8String and it still leak. i've also tried: NSData *dtmpname = [NSData dataWithBytes:sqlite3_column_blob(compiledStatement, 1) length:sqlite3_column_bytes(compiledStatement, 1)]; NSString *tmpname = [[[NSString alloc] initWithData:dtmpname encoding:NSUTF8StringEncoding] autorelease]; and the problem remains, the leak occur when u start scrolling the tableview. i've actually tried other encoding and it seems that only NSASCIIStringEncoding works(no memory leak) any idea how to get rid of this problem?

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  • Code for decoding/encoding a modified base64 URL

    - by Kirk Liemohn
    I want to base64 encode data to put it in a URL and then decode it within my HttpHandler. I have found that Base64 Encoding allows for a '/' character which will mess up my UriTemplate matching. Then I found that there is a concept of a "modified Base64 for URL" from wikipedia: A modified Base64 for URL variant exists, where no padding '=' will be used, and the '+' and '/' characters of standard Base64 are respectively replaced by '-' and '_', so that using URL encoders/decoders is no longer necessary and has no impact on the length of the encoded value, leaving the same encoded form intact for use in relational databases, web forms, and object identifiers in general. Using .NET I want to modify my current code from doing basic base64 encoding and decoding to using the "modified base64 for URL" method. Has anyone done this? To decode, I know it starts out with something like: string base64EncodedText = base64UrlEncodedText.Replace('-', '+').Replace('_', '/'); // Append '=' char(s) if necessary - how best to do this? // My normal base64 decoding now uses encodedText But, I need to potentially add one or two '=' chars to the end which looks a little more complex. My encoding logic should be a little simpler: // Perform normal base64 encoding byte[] encodedBytes = Encoding.UTF8.GetBytes(unencodedText); string base64EncodedText = Convert.ToBase64String(encodedBytes); // Apply URL variant string base64UrlEncodedText = base64EncodedText.Replace("=", String.Empty).Replace('+', '-').Replace('/', '_'); I have seen the Guid to Base64 for URL StackOverflow entry, but that has a known length and therefore they can hardcode the number of equal signs needed at the end.

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  • Fast or asynchronous AS3 JPEG encoding

    - by Bart van Heukelom
    I'm currently using the JPGEncoder from the AS3 core lib to encode a bitmap to JPEG var enc:JPGEncoder = new JPGEncoder(90); var jpg:ByteArray = enc.encode(bitmap); Because the bitmap is rather large (3000 x 2000) the encoding takes a long while (about 20 seconds), causing the application to seemingly freeze while encoding. To solve this, I need either: An asynchronous encoder so I can keep updating the screen (with a progress bar or something) while encoding An alternative encoder which is simply faster Is either possible?

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  • Perl's use encoding pragma breaking UTF strings

    - by Karel Bílek
    I have a problem with Perl and Encoding pragma. (I use utf-8 everywhere, in input, output, the perl scripts themselves. I don't want to use other encoding, never ever.) However. When I write binmode(STDOUT, ':utf8'); use utf8; $r = "\x{ed}"; print $r; I see the string "í" (which is what I want - and what is 00+ED unicode char). But when I add the "use encoding" pragma like this binmode(STDOUT, ':utf8'); use utf8; use encoding 'utf8'; $r = "\x{ed}"; print $r; all I see is a box character. Why? Moreover, when I add Data::Dumper and let the Dumper print the new string like this binmode(STDOUT, ':utf8'); use utf8; use encoding 'utf8'; $r = "\x{ed}"; use Data::Dumper; print Dumper($r); I see that perl changed the string to "\x{fffd}". Why?

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  • python UTF16LE file to UTF8 encoding

    - by Qiao
    I have big file with utf16le (BOM) encoding. Is it possible to convert it to usual UTF8 by python? Something like file_old = open('old.txt', mode='r', encoding='utf_16_le') file_new = open('new.txt', mode='w', encoding='utf-8') text = file_old.read() file_new.write(text.encode('utf-8')) http://docs.python.org/release/2.3/lib/node126.html (-- utf_16_le UTF-16LE) Not working. Can't understand "TypeError: must be str, not bytes" error. python 3

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