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  • Stream audio to mobile device

    - by blackn1ght
    I'd like to stream the audio from Ubuntu 10.10 to my HTC Desire HD (Android 2.2). I've seen solutions so far for streaming from audio players, but I'd like to stream any audio output from the PC to my phone. My use case is for watching TV/Films in VLC or online (BBC iPlayer) in bed, without having to use my surround sound system which is likely to wake up my house mates. I'm not just talking about music from Banshee, but any audio that the system makes. I was thinking that PulseAudio is pretty powerful, is it possible to route audio through that to a mobile device? Can it be done through bluetooth? Cheers in advance!

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  • USB Audio Device Loopback Through Speakers

    - by matto1990
    I have a USB turntable which when plugged in to my ubuntu 10.10 machine appears in the audio settings as an input device (USB PnP Audio Device Analog Stereo) like a microphone. What I'd like to be able to do it to have the sound for that audio device played back through the audio output (speaker or whatever). I'm not too worried if there's a slight delay between the audio coming in and it being played out through the speakers. As far as I'm aware this is refereed to as software loopback. I can achieve exactly what I want if I open Audacity, enable software loopback and press record. Obvious this isn't ideal as I don't really want it recording what I'm playing all the time. I know this is possible because of the Audacity example however I'd like to know if there's a way to do it without it recording. I've search around for a while for a piece of software that does this, however I couldn't get anything even close. Any help would be greatly appreciated.

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  • Play audio in javascript with a good performance

    - by João
    I'm developing a browser game where the player can shoot. Everytime he shoots it play a sound. Currently i'm using this code to play sounds in JavaScript: var audio = document.createElement("audio"); audio.src = "my_sound.mp3"; audio.play(); I'm worried about performance here. Will 10 simultaneous sounds impact my game performance too much? Will all audio objects stay in memory even after they are played?

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  • How can I select an audio output device in directshow

    - by Vibhore Tanwer
    I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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  • Using interface classes and non-virtual interface idiom in C++

    - by andreas buykx
    Hi all, In C++ an interface can be implemented by a class with all its methods pure virtual: class IFoo { public: virtual void method() = 0; }; Now I want to implement this interface by a hierarchy of classes: class FooBase : public IFoo // implement interface IFoo { public: void method(); // calls methodImpl; private: virtual void methodImpl(); }; For the class hierarchy I would like to use the non-virtual interface (NVI) idiom, to deny derived classes the possibility of overriding the common behavior implemented in FooBase::method(), but it seems that all derived classes have the opportunity to override the FooBase::method() because it is declared in the interface class. Is my observation correct? And if so are there other options to both use interface classes and the NVI idiom?

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  • Implement a generic C++/CLI interface in a C# generic interface

    - by Florent
    I have a C++/CLI generic interface like this : using namespace System::Collections::Generic; namespace Test { generic <class T> public interface class IElementList { property List<T>^ Elements; }; } and I want to implement it in a C# generic interface like this : using Test; namespace U { public interface IElementLightList<T> : IElementList<T> where T : IElementLight { bool isFrozen(); bool isPastable(); } } This don't work, Visual Studio is not able to see C++/CLI IElementList interface ! I tested with a not generic C++/CLI interface and this work. What I missed ?

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  • Do you have any additions or alterations to this list of popular audio formats?

    - by roja
    All, I am trying to compile a list of common audio file formats used in both personal storage and peer transmission. I have compiled the following list, do you think that there are any significant formats missing? Are any of them not actually common formats? Any advice/alterations are highly useful. advanced audio coding, apple lossless audio file, atrac3 audio file, atrac audio file, audio interchange file format, core audio file, free lossless audio codec file, mpeg 1 audio layer 3, mpeg 2 audio, mpeg 4 audio book file, musical instrument digital interface, ogg vorbis compressed audio file, open media framework file, real audio, real audio media, waveform audio file format, windows media audio Kind regards, Roja

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  • Boost Audio Input on OS X?

    - by alanstorm
    I'm using my 13" Mac Book Pro's audio input functionality with an external microphone (recent vintage, bought around Thanksgiving). I've increased my input volume to the maximum in system preference, but the resulting recorded volume (using iShowU HD) is very low. Is there anyway to increase the input volume/sensitivity beyond Apple's default settings? I've found plenty on google about increasing the OUTPUT volume, but I want to increase the input volume.

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  • Set default system audio output port (for all accounts)

    - by Ludwik Trammer
    The default output audio port Ubuntu doesn't work on my system. It should be "Analog Mono Output/Amplifier", instead of "Analog Output/Amplifier". I can easily change that in sound preferences, just by choosing the right port in the "Output" tab. The problem is this would only apply to a single account, and I would like to change it system-wide, so it applies to all accounts on the system (I have more than 100 users...). I'm after 2 hours of Googling, so any help would be appreciated.

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  • JavaScript audio not playing outside of jQuery function

    - by user1814016
    I know the question title doesn't make much sense, but I can't think of a better way to put it. I am a newbie to jQuery and I'm using this code to fade in a <div> and play a sound: $(document).ready(function(){ $('#speech').fadeIn('medium', function() { play('msg_appear'); var sptx = $('<p class="stext">').text('There is nothing here.'); $('#speech').append(sptx); $('.stext').typeOut({marker: '', delay: 22}); }); }); This code runs fine however the sound plays after the fade-in is complete. I wanted it to play while it was fading in, so I tried placing the play() call outside of the fade-in function like this: $(document).ready(function(){ play('msg_appear'); $('#speech').fadeIn('medium', function() { However, now it's not playing at all. There's no errors on the JavaScript console so I'm unsure if it's a syntax error, and probably something obvious, but I don't know what. play() is a function I found to play audio, here it is if it matters at all. I placed it in the same file the above code is; right above the $(document).ready(). function play(sound) { if (window.HTMLAudioElement) { var snd = new Audio(''); if(snd.canPlayType('audio/ogg')) { snd = new Audio(sound + '.ogg'); } else if(snd.canPlayType('audio/mp3')) { snd = new Audio(sound + '.mp3'); } snd.play(); } else { alert('HTML5 Audio is not supported by your browser!'); } }

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  • Implementing an interface with interface members

    - by jstark
    What is the proper way to implement an interface that has its own interface members? (am I saying that correctly?) Here's what I mean: public Interface IFoo { string Forty { get; set; } string Two { get; set; } } public Interface IBar { // other stuff... IFoo Answer { get; set; } } public class Foo : IFoo { public string Forty { get; set; } public string Two { get; set; } } public class Bar : IBar { // other stuff public Foo Answer { get; set; } //why doesnt' this work? } I've gotten around my problem using explicit interface implementation, but I'm wondering if there is a better way?

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  • Audio server with best API?

    - by Wintermute
    I'm a web dev, working in a small studio with a couple of other devs and some crayon-munchers (or, "designers"). Like all the best and trendiest creative studios, we have tunes. Our tunes consists of a set of speakers that whoever wants to can plug into their machine, and DJ their little socks off via iTunes, Spotify, VLC or whatever their music player of choice happens to be. Obviously, this lacks finesse. What we WANT is this: a single, dedicated machine running some sort of audio player (ideally Win-based, but a Linux flavour isn't impossible), that exposes an API. We (ie: me and the other devs) want to write a web-based client onto it, that'll let us remotely do all sorts of funky stuff like generating on-the-fly genre-based playlists, and voting for tracks, and making tea. My question - and please forgive me if this isn't the place for such a question, I was going to ask on Stackoverflow but that didn't seem right either - is this: what's the best player to start with? What can do all of this? I know VLC can function as a streaming server, but know nothing of any API it may have. I'd rather chop my pinky off than use iTunes, but if it does what we want, then... Anyhow, thanks for reading. All comments and suggestions gratefully received.

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  • Monitoring an audio line.

    - by Stefan Liebenberg
    I need to monitor my audio line-in in linux, and in the event that audio is played, the sound must be recorded and saved to a file. Similiar to how motion monitors the video feed. Is it possible to do this with bash? something along the lines of: #!/bin/bash # audio device device=/dev/audio-line-in # below this threshold audio will not be recorded. noise_threshold=10 # folder where recordings are stored storage_folder=~/recordings # run indefenitly, until Ctrl-C is pressed while true; do # noise_level() represents a function to determine # the noise level from device if noise_level( $device ) > $noise_threshold; then # stream from device to file, can be encoded to mp3 later. cat $device > $storage_folder/`date`.raw fi; done;

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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  • Outbound traffic being blocked for MIP/VIPped servers (Juniper SSG5)

    - by Mark S. Rasmussen
    As we've been having some problems with sporadic packet loss, I've been preparing a replacement router (also an SSG5) for our current Juniper SSG5. I've setup the new SSG5 identically to the old one. We have a /29 IP range with a single IP setup as a MIP map to a server and two others being used for VIP maps. Each VIP/MIP is accompanied by relevant policies. Long story short - we tried connected the new SSG5 and some things were not working as they should. No problem, I just reconnected the old one. However, some things are still broken, even when I reconnected the old one. I fear I may have inadvertently changed some settings while browsing through old settings in my attempt to reconfigure the new SSG5 unit. All inbound traffic seems to work as expected. However, the 192.168.2.202 server can't initiate any outbound connections. It works perfectly on the local network, but any pings or DNS lookups to external IP's fail. The MIP & VIP map to it works perfectly - I can access it through HTTP and RDP without issues. Any tips on what to debug, or where I've messed up my config? I've attached the full config here (with anonymized IPs): set clock timezone 1 set vrouter trust-vr sharable set vrouter "untrust-vr" exit set vrouter "trust-vr" unset auto-route-export exit set service "MyVOIP_UDP4569" protocol udp src-port 0-65535 dst-port 4569-4569 set service "MyVOIP_TCP22" protocol tcp src-port 0-65535 dst-port 22-22 set service "MyRDP" protocol tcp src-port 0-65535 dst-port 3389-3389 set service "MyRsync" protocol tcp src-port 0-65535 dst-port 873-873 set service "NZ_FTP" protocol tcp src-port 0-65535 dst-port 40000-41000 set service "NZ_FTP" + tcp src-port 0-65535 dst-port 21-21 set service "PPTP-VPN" protocol 47 src-port 2048-2048 dst-port 2048-2048 set service "PPTP-VPN" + tcp src-port 1024-65535 dst-port 1723-1723 set service "NZ_FMS_1935" protocol tcp src-port 0-65535 dst-port 1935-1935 set service "NZ_FMS_1935" + udp src-port 0-65535 dst-port 1935-1935 set service "NZ_FMS_8080" protocol tcp src-port 0-65535 dst-port 8080-8080 set service "CrashPlan Server" protocol tcp src-port 0-65535 dst-port 4280-4280 set service "CrashPlan Console" protocol tcp src-port 0-65535 dst-port 4282-4282 unset alg sip enable set auth-server "Local" id 0 set auth-server "Local" server-name "Local" set auth default auth server "Local" set auth radius accounting port 1646 set admin auth timeout 10 set admin auth server "Local" set admin format dos set vip multi-port set zone "Trust" vrouter "trust-vr" set zone "Untrust" vrouter "trust-vr" set zone "DMZ" vrouter "trust-vr" set zone "VLAN" vrouter "trust-vr" set zone "Untrust-Tun" vrouter "trust-vr" set zone "Trust" tcp-rst set zone "Untrust" block unset zone "Untrust" tcp-rst set zone "DMZ" tcp-rst set zone "VLAN" block unset zone "VLAN" tcp-rst set zone "Untrust" screen tear-drop set zone "Untrust" screen syn-flood set zone "Untrust" screen ping-death set zone "Untrust" screen ip-filter-src set zone "Untrust" screen land set zone "V1-Untrust" screen tear-drop set zone "V1-Untrust" screen syn-flood set zone "V1-Untrust" screen ping-death set zone "V1-Untrust" screen ip-filter-src set zone "V1-Untrust" screen land set interface ethernet0/0 phy full 100mb set interface ethernet0/3 phy full 100mb set interface ethernet0/4 phy full 100mb set interface ethernet0/5 phy full 100mb set interface ethernet0/6 phy full 100mb set interface "ethernet0/0" zone "Untrust" set interface "ethernet0/1" zone "Null" set interface "bgroup0" zone "Trust" set interface "bgroup1" zone "Trust" set interface "bgroup2" zone "Trust" set interface bgroup2 port ethernet0/2 set interface bgroup0 port ethernet0/3 set interface bgroup0 port ethernet0/4 set interface bgroup1 port ethernet0/5 set interface bgroup1 port ethernet0/6 unset interface vlan1 ip set interface ethernet0/0 ip 212.242.193.18/29 set interface ethernet0/0 route set interface bgroup0 ip 192.168.1.1/24 set interface bgroup0 nat set interface bgroup1 ip 192.168.2.1/24 set interface bgroup1 nat set interface bgroup2 ip 192.168.3.1/24 set interface bgroup2 nat set interface ethernet0/0 gateway 212.242.193.17 unset interface vlan1 bypass-others-ipsec unset interface vlan1 bypass-non-ip set interface ethernet0/0 ip manageable set interface bgroup0 ip manageable set interface bgroup1 ip manageable set interface bgroup2 ip manageable set interface bgroup0 manage mtrace unset interface bgroup1 manage ssh unset interface bgroup1 manage telnet unset interface bgroup1 manage snmp unset interface bgroup1 manage ssl unset interface bgroup1 manage web unset interface bgroup2 manage ssh unset interface bgroup2 manage telnet unset interface bgroup2 manage snmp unset interface bgroup2 manage ssl unset interface bgroup2 manage web set interface ethernet0/0 vip 212.242.193.19 2048 "PPTP-VPN" 192.168.1.131 set interface ethernet0/0 vip 212.242.193.19 + 4280 "CrashPlan Server" 192.168.1.131 set interface ethernet0/0 vip 212.242.193.19 + 4282 "CrashPlan Console" 192.168.1.131 set interface ethernet0/0 vip 212.242.193.22 22 "MyVOIP_TCP22" 192.168.2.127 set interface ethernet0/0 vip 212.242.193.22 + 4569 "MyVOIP_UDP4569" 192.168.2.127 set interface ethernet0/0 vip 212.242.193.22 + 3389 "MyRDP" 192.168.2.202 set interface ethernet0/0 vip 212.242.193.22 + 873 "MyRsync" 192.168.2.201 set interface ethernet0/0 vip 212.242.193.22 + 80 "HTTP" 192.168.2.202 set interface ethernet0/0 vip 212.242.193.22 + 2048 "PPTP-VPN" 192.168.2.201 set interface ethernet0/0 vip 212.242.193.22 + 8080 "NZ_FMS_8080" 192.168.2.216 set interface ethernet0/0 vip 212.242.193.22 + 1935 "NZ_FMS_1935" 192.168.2.216 set interface bgroup0 dhcp server service set interface bgroup1 dhcp server service set interface bgroup2 dhcp server service set interface bgroup0 dhcp server auto set interface bgroup1 dhcp server auto set interface bgroup2 dhcp server auto set interface bgroup0 dhcp server option domainname iplan set interface bgroup0 dhcp server option dns1 192.168.1.131 set interface bgroup1 dhcp server option domainname nzlan set interface bgroup1 dhcp server option dns1 192.168.2.202 set interface bgroup2 dhcp server option dns1 8.8.8.8 set interface bgroup2 dhcp server option wins1 8.8.4.4 set interface bgroup0 dhcp server ip 192.168.1.2 to 192.168.1.116 set interface bgroup1 dhcp server ip 192.168.2.2 to 192.168.2.116 set interface bgroup2 dhcp server ip 192.168.3.2 to 192.168.3.126 unset interface bgroup0 dhcp server config next-server-ip unset interface bgroup1 dhcp server config next-server-ip unset interface bgroup2 dhcp server config next-server-ip set interface "ethernet0/0" mip 212.242.193.21 host 192.168.2.202 netmask 255.255.255.255 vr "trust-vr" set interface "serial0/0" modem settings "USR" init "AT&F" set interface "serial0/0" modem settings "USR" active set interface "serial0/0" modem speed 115200 set interface "serial0/0" modem retry 3 set interface "serial0/0" modem interval 10 set interface "serial0/0" modem idle-time 10 set pak-poll p1queue pak-threshold 96 set pak-poll p2queue pak-threshold 32 set flow tcp-mss unset flow tcp-syn-check set dns host dns1 0.0.0.0 set dns host dns2 0.0.0.0 set dns host dns3 0.0.0.0 set address "Trust" "192.168.1.0/24" 192.168.1.0 255.255.255.0 set address "Trust" "192.168.2.0/24" 192.168.2.0 255.255.255.0 set address "Trust" "192.168.3.0/24" 192.168.3.0 255.255.255.0 set ike respond-bad-spi 1 unset ike ikeid-enumeration unset ike dos-protection unset ipsec access-session enable set ipsec access-session maximum 5000 set ipsec access-session upper-threshold 0 set ipsec access-session lower-threshold 0 set ipsec access-session dead-p2-sa-timeout 0 unset ipsec access-session log-error unset ipsec access-session info-exch-connected unset ipsec access-session use-error-log set l2tp default ppp-auth chap set url protocol websense exit set policy id 1 from "Trust" to "Untrust" "Any" "Any" "ANY" permit traffic set policy id 1 exit set policy id 2 from "Untrust" to "Trust" "Any" "VIP(212.242.193.19)" "PPTP-VPN" permit traffic set policy id 2 exit set policy id 3 from "Untrust" to "Trust" "Any" "VIP(212.242.193.22)" "HTTP" permit traffic priority 0 set policy id 3 set service "MyRDP" set service "MyRsync" set service "MyVOIP_TCP22" set service "MyVOIP_UDP4569" exit set policy id 6 from "Trust" to "Trust" "192.168.1.0/24" "192.168.2.0/24" "ANY" deny set policy id 6 exit set policy id 7 from "Trust" to "Trust" "192.168.2.0/24" "192.168.1.0/24" "ANY" deny set policy id 7 exit set policy id 8 from "Trust" to "Trust" "192.168.3.0/24" "192.168.1.0/24" "ANY" deny set policy id 8 exit set policy id 9 from "Trust" to "Trust" "192.168.3.0/24" "192.168.2.0/24" "ANY" deny set policy id 9 exit set policy id 10 from "Untrust" to "Trust" "Any" "MIP(212.242.193.21)" "NZ_FTP" permit set policy id 10 exit set policy id 11 from "Untrust" to "Trust" "Any" "VIP(212.242.193.22)" "PPTP-VPN" permit set policy id 11 exit set policy id 12 from "Untrust" to "Trust" "Any" "VIP(212.242.193.22)" "NZ_FMS_1935" permit set policy id 12 set service "NZ_FMS_8080" exit set policy id 13 from "Untrust" to "Trust" "Any" "VIP(212.242.193.19)" "CrashPlan Console" permit set policy id 13 set service "CrashPlan Server" exit set nsmgmt bulkcli reboot-timeout 60 set ssh version v2 set config lock timeout 5 set snmp port listen 161 set snmp port trap 162 set vrouter "untrust-vr" exit set vrouter "trust-vr" unset add-default-route exit set vrouter "untrust-vr" exit set vrouter "trust-vr" exit

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  • Allowing connections initiated from outside

    - by Mark S. Rasmussen
    I've got an old Juniper SSG5 running ScreenOS 5.4.0r6.0. Once a day, more or less, it'll start randomly dropping packets at a rate of ~5-10%. We currently solve this issue by simply rebooting the unit, after which it resumes working in perfect condition. As this error has started appearing randomly, without any configuration or hardware changes, I'm assuming I've got an aging unit about to fail. As such, I've got a replacement SSG5 running ScreenOS 6.0. I've dumped the config on the 5.4 and imported it into a clean 6.0, and it seems to gladly accept it, and all my configuration seems to be A-OK. However, upon connecting the new unit, all outside-initiated connections seem to be blocked. If I browse our external IP from the inside, everything works perfectly, and it's not just port 80, SSH, Crashplan - all of our policies route correctly. All normal networking, initiated from the inside, work perfectly as well. If on the other hand I browse our external IP from the outside, everything is blocked. Barring differences between ScreenOS 5.4 and 6.0, the config is identical. Is there a setting somewhere that defines whether outside/inside initiated connections are allowed? unset key protection enable set clock timezone 1 set vrouter trust-vr sharable set vrouter "untrust-vr" exit set vrouter "trust-vr" unset auto-route-export exit set service "MyVOIP_UDP4569" protocol udp src-port 0-65535 dst-port 4569-4569 set service "MyVOIP_TCP22" protocol tcp src-port 0-65535 dst-port 22-22 set service "MyRDP" protocol tcp src-port 0-65535 dst-port 3389-3389 set service "MyRsync" protocol tcp src-port 0-65535 dst-port 873-873 set service "NZ_FTP" protocol tcp src-port 0-65535 dst-port 40000-41000 set service "NZ_FTP" + tcp src-port 0-65535 dst-port 21-21 set service "PPTP-VPN" protocol 47 src-port 2048-2048 dst-port 2048-2048 set service "PPTP-VPN" + tcp src-port 1024-65535 dst-port 1723-1723 set service "NZ_FMS_1935" protocol tcp src-port 0-65535 dst-port 1935-1935 set service "NZ_FMS_1935" + udp src-port 0-65535 dst-port 1935-1935 set service "NZ_FMS_8080" protocol tcp src-port 0-65535 dst-port 8080-8080 set service "CrashPlan Server" protocol tcp src-port 0-65535 dst-port 4280-4280 set service "CrashPlan Console" protocol tcp src-port 0-65535 dst-port 4282-4282 unset alg sip enable set alg appleichat enable unset alg appleichat re-assembly enable set alg sctp enable set auth-server "Local" id 0 set auth-server "Local" server-name "Local" set auth default auth server "Local" set auth radius accounting port 1646 set admin name "netscreen" set admin password "XXX" set admin auth web timeout 10 set admin auth dial-in timeout 3 set admin auth server "Local" set admin format dos set vip multi-port set zone "Trust" vrouter "trust-vr" set zone "Untrust" vrouter "trust-vr" set zone "DMZ" vrouter "trust-vr" set zone "VLAN" vrouter "trust-vr" set zone "Untrust-Tun" vrouter "trust-vr" set zone "Trust" tcp-rst set zone "Untrust" block unset zone "Untrust" tcp-rst set zone "MGT" block unset zone "V1-Trust" tcp-rst unset zone "V1-Untrust" tcp-rst set zone "DMZ" tcp-rst unset zone "V1-DMZ" tcp-rst unset zone "VLAN" tcp-rst set zone "Untrust" screen tear-drop set zone "Untrust" screen syn-flood set zone "Untrust" screen ping-death set zone "Untrust" screen ip-filter-src set zone "Untrust" screen land set zone "V1-Untrust" screen tear-drop set zone "V1-Untrust" screen syn-flood set zone "V1-Untrust" screen ping-death set zone "V1-Untrust" screen ip-filter-src set zone "V1-Untrust" screen land set interface ethernet0/0 phy full 100mb set interface ethernet0/3 phy full 100mb set interface ethernet0/4 phy full 100mb set interface ethernet0/5 phy full 100mb set interface ethernet0/6 phy full 100mb set interface "ethernet0/0" zone "Untrust" set interface "ethernet0/1" zone "Null" set interface "bgroup0" zone "Trust" set interface "bgroup1" zone "Trust" set interface "bgroup2" zone "Trust" set interface bgroup2 port ethernet0/2 set interface bgroup0 port ethernet0/3 set interface bgroup0 port ethernet0/4 set interface bgroup1 port ethernet0/5 set interface bgroup1 port ethernet0/6 unset interface vlan1 ip set interface ethernet0/0 ip 215.173.182.18/29 set interface ethernet0/0 route set interface bgroup0 ip 192.168.1.1/24 set interface bgroup0 nat set interface bgroup1 ip 192.168.2.1/24 set interface bgroup1 nat set interface bgroup2 ip 192.168.3.1/24 set interface bgroup2 nat set interface ethernet0/0 gateway 215.173.182.17 unset interface vlan1 bypass-others-ipsec unset interface vlan1 bypass-non-ip set interface ethernet0/0 ip manageable set interface bgroup0 ip manageable set interface bgroup1 ip manageable set interface bgroup2 ip manageable set interface bgroup0 manage mtrace unset interface bgroup1 manage ssh unset interface bgroup1 manage telnet unset interface bgroup1 manage snmp unset interface bgroup1 manage ssl unset interface bgroup1 manage web unset interface bgroup2 manage ssh unset interface bgroup2 manage telnet unset interface bgroup2 manage snmp unset interface bgroup2 manage ssl unset interface bgroup2 manage web set interface ethernet0/0 vip 215.173.182.19 2048 "PPTP-VPN" 192.168.1.131 set interface ethernet0/0 vip 215.173.182.19 + 4280 "CrashPlan Server" 192.168.1.131 set interface ethernet0/0 vip 215.173.182.19 + 4282 "CrashPlan Console" 192.168.1.131 set interface ethernet0/0 vip 215.173.182.22 22 "MyVOIP_TCP22" 192.168.2.127 set interface ethernet0/0 vip 215.173.182.22 + 4569 "MyVOIP_UDP4569" 192.168.2.127 set interface ethernet0/0 vip 215.173.182.22 + 3389 "MyRDP" 192.168.2.202 set interface ethernet0/0 vip 215.173.182.22 + 873 "MyRsync" 192.168.2.201 set interface ethernet0/0 vip 215.173.182.22 + 80 "HTTP" 192.168.2.202 set interface ethernet0/0 vip 215.173.182.22 + 2048 "PPTP-VPN" 192.168.2.201 set interface ethernet0/0 vip 215.173.182.22 + 8080 "NZ_FMS_8080" 192.168.2.216 set interface ethernet0/0 vip 215.173.182.22 + 1935 "NZ_FMS_1935" 192.168.2.216 set interface bgroup0 dhcp server service set interface bgroup1 dhcp server service set interface bgroup2 dhcp server service set interface bgroup0 dhcp server auto set interface bgroup1 dhcp server auto set interface bgroup2 dhcp server auto set interface bgroup0 dhcp server option domainname companyalan set interface bgroup0 dhcp server option dns1 192.168.1.131 set interface bgroup1 dhcp server option domainname companyblan set interface bgroup1 dhcp server option dns1 192.168.2.202 set interface bgroup2 dhcp server option dns1 8.8.8.8 set interface bgroup2 dhcp server option wins1 8.8.4.4 set interface bgroup0 dhcp server ip 192.168.1.2 to 192.168.1.116 set interface bgroup1 dhcp server ip 192.168.2.2 to 192.168.2.116 set interface bgroup2 dhcp server ip 192.168.3.2 to 192.168.3.126 unset interface bgroup0 dhcp server config next-server-ip unset interface bgroup1 dhcp server config next-server-ip unset interface bgroup2 dhcp server config next-server-ip set interface "ethernet0/0" mip 215.173.182.21 host 192.168.2.202 netmask 255.255.255.255 vr "trust-vr" set interface "serial0/0" modem settings "USR" init "AT&F" set interface "serial0/0" modem settings "USR" active set interface "serial0/0" modem speed 115200 set interface "serial0/0" modem retry 3 set interface "serial0/0" modem interval 10 set interface "serial0/0" modem idle-time 10 set flow tcp-mss unset flow tcp-syn-check unset flow tcp-syn-bit-check set flow reverse-route clear-text prefer set flow reverse-route tunnel always set pki authority default scep mode "auto" set pki x509 default cert-path partial set pki x509 dn name "[email protected]" set dns host dns1 0.0.0.0 set dns host dns2 0.0.0.0 set dns host dns3 0.0.0.0 set address "Trust" "192.168.1.0/24" 192.168.1.0 255.255.255.0 set address "Trust" "192.168.2.0/24" 192.168.2.0 255.255.255.0 set address "Trust" "192.168.3.0/24" 192.168.3.0 255.255.255.0 set crypto-policy exit set ike respond-bad-spi 1 set ike ikev2 ike-sa-soft-lifetime 60 unset ike ikeid-enumeration unset ike dos-protection unset ipsec access-session enable set ipsec access-session maximum 5000 set ipsec access-session upper-threshold 0 set ipsec access-session lower-threshold 0 set ipsec access-session dead-p2-sa-timeout 0 unset ipsec access-session log-error unset ipsec access-session info-exch-connected unset ipsec access-session use-error-log set vrouter "untrust-vr" exit set vrouter "trust-vr" exit set l2tp default ppp-auth chap set url protocol websense exit set policy id 1 from "Trust" to "Untrust" "Any" "Any" "ANY" permit set policy id 1 exit set policy id 2 from "Untrust" to "Trust" "Any" "VIP(215.173.182.19)" "PPTP-VPN" permit traffic set policy id 2 exit set policy id 3 from "Untrust" to "Trust" "Any" "VIP(215.173.182.22)" "HTTP" permit log set policy id 3 set service "MyRDP" set service "MyRsync" set service "MyVOIP_TCP22" set service "MyVOIP_UDP4569" exit set policy id 6 from "Trust" to "Trust" "192.168.1.0/24" "192.168.2.0/24" "ANY" deny set policy id 6 exit set policy id 7 from "Trust" to "Trust" "192.168.2.0/24" "192.168.1.0/24" "ANY" deny set policy id 7 exit set policy id 8 from "Trust" to "Trust" "192.168.3.0/24" "192.168.1.0/24" "ANY" deny set policy id 8 exit set policy id 9 from "Trust" to "Trust" "192.168.3.0/24" "192.168.2.0/24" "ANY" deny set policy id 9 exit set policy id 10 from "Untrust" to "Trust" "Any" "MIP(215.173.182.21)" "NZ_FTP" permit set policy id 10 exit set policy id 11 from "Untrust" to "Trust" "Any" "VIP(215.173.182.22)" "PPTP-VPN" permit set policy id 11 exit set policy id 12 from "Untrust" to "Trust" "Any" "VIP(215.173.182.22)" "NZ_FMS_1935" permit set policy id 12 set service "NZ_FMS_8080" exit set policy id 13 from "Untrust" to "Trust" "Any" "VIP(215.173.182.19)" "CrashPlan Console" permit set policy id 13 set service "CrashPlan Server" exit set nsmgmt bulkcli reboot-timeout 60 set ssh version v2 set config lock timeout 5 unset license-key auto-update set telnet client enable set snmp port listen 161 set snmp port trap 162 set vrouter "untrust-vr" exit set vrouter "trust-vr" unset add-default-route exit set vrouter "untrust-vr" exit set vrouter "trust-vr" exit Note that I've previously posted a similar question (pertaining to the same device & replacement, but ultimately caused by a malfunctioning switch, and thus clouding the current issue): Outbound traffic being blocked for MIP/VIPped servers (Juniper SSG5)

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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • How to fix these compiler errors?

    - by Sandra Schlichting
    I have this source code from 2001 that I would like to compile. It gives this: $ make g++ -O99 -Wall -DLINUX -pedantic -c -o audio.o audio.cpp In file included from audio.cpp:7: audio.h:14: error: use of enum ‘mad_flow’ without previous declaration audio.h:15: error: use of enum ‘mad_flow’ without previous declaration audio.h:17: error: use of enum ‘mad_flow’ without previous declaration audio.cpp: In function ‘mad_flow audio::input(void*, mad_stream*)’: audio.cpp:19: error: new declaration ‘mad_flow audio::input(void*, mad_stream*)’ audio.h:14: error: ambiguates old declaration ‘int audio::input(void*, mad_stream*)’ audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:23: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:23: error: within this context audio.h:10: error: ‘char* audio::stream::Buffer’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:27: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:27: error: within this context audio.cpp: In function ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’: audio.cpp:49: error: new declaration ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’ audio.h:15: error: ambiguates old declaration ‘int audio::output(void*, const mad_header*, mad_pcm*)’ audio.cpp: In function ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’: audio.cpp:83: error: new declaration ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’ audio.h:17: error: ambiguates old declaration ‘int audio::error(void*, mad_stream*, mad_frame*)’ audio.cpp: In constructor ‘audio::stream::stream(const char*)’: audio.cpp:119: error: ‘input’ was not declared in this scope audio.cpp:122: error: ‘output’ was not declared in this scope audio.cpp:123: error: ‘error’ was not declared in this scope make: *** [audio.o] Error 1 audio.h contains #include <stdlib.h> #include "mad.h" namespace audio { class stream { private: char* Buffer; size_t BufferSize, BufferPos; struct mad_decoder Decoder; friend enum mad_flow input(void* Data, struct mad_stream* MadStream); friend enum mad_flow output(void* Data, const struct mad_header* Header, struct mad_pcm* PCM); friend enum mad_flow error(void* Data, struct mad_stream* MadStream, struct mad_frame* Frame); public: stream(const char* FileName); ~stream(); void play(); }; } I have tried to just insert enum mad_flow {}; but that just gave a new problem. Can anyone see how to fix this?

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  • Is there an audio recording application/tool that has Tivo-like functionality?

    - by Bob
    I do a lot of live speech recording that requires me to quickly jump back and then transcribe a particular piece of the audio, then go back to recording again, while still maintaining the full audio file. So Far I've done this by splitting the audio and running one line to a recorder (for the whole audio), and one to my computer. Then I use something like Audacity to record, and then stop/go back whenever I hear something worth transcribing. This requires me to stop the recording, then start it up again and I end up missing chunks of the speech I'm listening to. Is there a tool that would let me rewind, then listen again and continue listening at a buffered distance from the audio recording, the way Tivo does with television shows?

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