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  • WCF on Windows Phone 7 (Silverlight 4)

    - by Igor Zevaka
    Has anyone been able to communicate using WCF on Windows Phone Series 7 emulator? I've been trying for the past two days and it's just happening for me. I can get a normal Silverlight control to work in both Silverlight 3 and Silverlight 4, but not the phone version. Here are two versions that I've tried: Version 1 - Using Async Pattern BasicHttpBinding basicHttpBinding = new BasicHttpBinding(); EndpointAddress endpointAddress = new EndpointAddress("http://localhost/wcf/Authentication.svc"); Wcf.IAuthentication auth1 = new ChannelFactory<Wcf.IAuthentication>(basicHttpBinding, endpointAddress).CreateChannel(endpointAddress); AsyncCallback callback = (result) => { Action<string> write = (str) => { this.Dispatcher.BeginInvoke(delegate { //Display something }); }; try { Wcf.IAuthentication auth = result.AsyncState as Wcf.IAuthentication; Wcf.AuthenticationResponse response = auth.EndLogin(result); write(response.Success.ToString()); } catch (Exception ex) { write(ex.Message); System.Diagnostics.Debug.WriteLine(ex.Message); } }; auth1.BeginLogin("user0", "test0", callback, auth1); This version breaks on this line: Wcf.IAuthentication auth1 = new ChannelFactory<Wcf.IAuthentication>(basicHttpBinding, endpointAddress).CreateChannel(endpointAddress); Throwing System.NotSupportedException. The exception is not very descriptive and the callstack is equally not very helpful: at System.ServiceModel.DiagnosticUtility.ExceptionUtility.BuildMessage(Exception x) at System.ServiceModel.DiagnosticUtility.ExceptionUtility.LogException(Exception x) at System.ServiceModel.DiagnosticUtility.ExceptionUtility.ThrowHelperError(Exception e) at System.ServiceModel.ChannelFactory`1.CreateChannel(EndpointAddress address) at WindowsPhoneApplication2.MainPage.DoLogin() .... Version 2 - Blocking WCF call Here is the version that doesn't use the async pattern. [System.ServiceModel.ServiceContract] public interface IAuthentication { [System.ServiceModel.OperationContract] AuthenticationResponse Login(string user, string password); } public class WcfClientBase<TChannel> : System.ServiceModel.ClientBase<TChannel> where TChannel : class { public WcfClientBase(string name, bool streaming) : base(GetBinding(streaming), GetEndpoint(name)) { ClientCredentials.UserName.UserName = WcfConfig.UserName; ClientCredentials.UserName.Password = WcfConfig.Password; } public WcfClientBase(string name) : this(name, false) {} private static System.ServiceModel.Channels.Binding GetBinding(bool streaming) { System.ServiceModel.BasicHttpBinding binding = new System.ServiceModel.BasicHttpBinding(); binding.MaxReceivedMessageSize = 1073741824; if(streaming) { //binding.TransferMode = System.ServiceModel.TransferMode.Streamed; } /*if(XXXURLXXX.StartsWith("https")) { binding.Security.Mode = BasicHttpSecurityMode.Transport; binding.Security.Transport.ClientCredentialType = HttpClientCredentialType.None; }*/ return binding; } private static System.ServiceModel.EndpointAddress GetEndpoint(string name) { return new System.ServiceModel.EndpointAddress(WcfConfig.Endpoint + name + ".svc"); } protected override TChannel CreateChannel() { throw new System.NotImplementedException(); } } auth.Login("test0", "password0"); This version crashes in System.ServiceModel.ClientBase<TChannel> constructor. The call stack is a bit different: at System.Reflection.MethodInfo.get_ReturnParameter() at System.ServiceModel.Description.ServiceReflector.HasNoDisposableParameters(MethodInfo methodInfo) at System.ServiceModel.Description.TypeLoader.CreateOperationDescription(ContractDescription contractDescription, MethodInfo methodInfo, MessageDirection direction, ContractReflectionInfo reflectionInfo, ContractDescription declaringContract) at System.ServiceModel.Description.TypeLoader.CreateOperationDescriptions(ContractDescription contractDescription, ContractReflectionInfo reflectionInfo, Type contractToGetMethodsFrom, ContractDescription declaringContract, MessageDirection direction) at System.ServiceModel.Description.TypeLoader.CreateContractDescription(ServiceContractAttribute contractAttr, Type contractType, Type serviceType, ContractReflectionInfo& reflectionInfo, Object serviceImplementation) at System.ServiceModel.Description.TypeLoader.LoadContractDescriptionHelper(Type contractType, Type serviceType, Object serviceImplementation) at System.ServiceModel.Description.TypeLoader.LoadContractDescription(Type contractType) at System.ServiceModel.ChannelFactory1.CreateDescription() at System.ServiceModel.ChannelFactory.InitializeEndpoint(Binding binding, EndpointAddress address) at System.ServiceModel.ChannelFactory1..ctor(Binding binding, EndpointAddress remoteAddress) at System.ServiceModel.ClientBase1..ctor(Binding binding, EndpointAddress remoteAddress) at Wcf.WcfClientBase1..ctor(String name, Boolean streaming) at Wcf.WcfClientBase`1..ctor(String name) at Wcf.AuthenticationClient..ctor() at WindowsPhoneApplication2.MainPage.DoLogin() ... Any ideas?

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  • System.UnsupportedException using WCF on Windows Phone 7

    - by Igor Zevaka
    Has anyone been able to communicate using WCF on Windows Phone Series 7 emulator? I've been trying for the past two days and it's just happening for me. I can get a normal Silverlight control to work in both Silverlight 3 and Silverlight 4, but not the phone version. Here are two versions that I've tried: Version 1 - Using Async Pattern BasicHttpBinding basicHttpBinding = new BasicHttpBinding(); EndpointAddress endpointAddress = new EndpointAddress("http://localhost/wcf/Authentication.svc"); Wcf.IAuthentication auth1 = new ChannelFactory<Wcf.IAuthentication>(basicHttpBinding, endpointAddress).CreateChannel(endpointAddress); AsyncCallback callback = (result) => { Action<string> write = (str) => { this.Dispatcher.BeginInvoke(delegate { //Display something }); }; try { Wcf.IAuthentication auth = result.AsyncState as Wcf.IAuthentication; Wcf.AuthenticationResponse response = auth.EndLogin(result); write(response.Success.ToString()); } catch (Exception ex) { write(ex.Message); System.Diagnostics.Debug.WriteLine(ex.Message); } }; auth1.BeginLogin("user0", "test0", callback, auth1); This version breaks on this line: Wcf.IAuthentication auth1 = new ChannelFactory<Wcf.IAuthentication>(basicHttpBinding, endpointAddress).CreateChannel(endpointAddress); Throwing System.NotSupportedException. The exception is not very descriptive and the callstack is equally not very helpful: at System.ServiceModel.DiagnosticUtility.ExceptionUtility.BuildMessage(Exception x) at System.ServiceModel.DiagnosticUtility.ExceptionUtility.LogException(Exception x) at System.ServiceModel.DiagnosticUtility.ExceptionUtility.ThrowHelperError(Exception e) at System.ServiceModel.ChannelFactory`1.CreateChannel(EndpointAddress address) at WindowsPhoneApplication2.MainPage.DoLogin() .... Version 2 - Blocking WCF call Here is the version that doesn't use the async pattern. [System.ServiceModel.ServiceContract] public interface IAuthentication { [System.ServiceModel.OperationContract] AuthenticationResponse Login(string user, string password); } public class WcfClientBase<TChannel> : System.ServiceModel.ClientBase<TChannel> where TChannel : class { public WcfClientBase(string name, bool streaming) : base(GetBinding(streaming), GetEndpoint(name)) { ClientCredentials.UserName.UserName = WcfConfig.UserName; ClientCredentials.UserName.Password = WcfConfig.Password; } public WcfClientBase(string name) : this(name, false) {} private static System.ServiceModel.Channels.Binding GetBinding(bool streaming) { System.ServiceModel.BasicHttpBinding binding = new System.ServiceModel.BasicHttpBinding(); binding.MaxReceivedMessageSize = 1073741824; if(streaming) { //binding.TransferMode = System.ServiceModel.TransferMode.Streamed; } /*if(XXXURLXXX.StartsWith("https")) { binding.Security.Mode = BasicHttpSecurityMode.Transport; binding.Security.Transport.ClientCredentialType = HttpClientCredentialType.None; }*/ return binding; } private static System.ServiceModel.EndpointAddress GetEndpoint(string name) { return new System.ServiceModel.EndpointAddress(WcfConfig.Endpoint + name + ".svc"); } protected override TChannel CreateChannel() { throw new System.NotImplementedException(); } } auth.Login("test0", "password0"); This version crashes in System.ServiceModel.ClientBase<TChannel> constructor. The call stack is a bit different: at System.Reflection.MethodInfo.get_ReturnParameter() at System.ServiceModel.Description.ServiceReflector.HasNoDisposableParameters(MethodInfo methodInfo) at System.ServiceModel.Description.TypeLoader.CreateOperationDescription(ContractDescription contractDescription, MethodInfo methodInfo, MessageDirection direction, ContractReflectionInfo reflectionInfo, ContractDescription declaringContract) at System.ServiceModel.Description.TypeLoader.CreateOperationDescriptions(ContractDescription contractDescription, ContractReflectionInfo reflectionInfo, Type contractToGetMethodsFrom, ContractDescription declaringContract, MessageDirection direction) at System.ServiceModel.Description.TypeLoader.CreateContractDescription(ServiceContractAttribute contractAttr, Type contractType, Type serviceType, ContractReflectionInfo& reflectionInfo, Object serviceImplementation) at System.ServiceModel.Description.TypeLoader.LoadContractDescriptionHelper(Type contractType, Type serviceType, Object serviceImplementation) at System.ServiceModel.Description.TypeLoader.LoadContractDescription(Type contractType) at System.ServiceModel.ChannelFactory1.CreateDescription() at System.ServiceModel.ChannelFactory.InitializeEndpoint(Binding binding, EndpointAddress address) at System.ServiceModel.ChannelFactory1..ctor(Binding binding, EndpointAddress remoteAddress) at System.ServiceModel.ClientBase1..ctor(Binding binding, EndpointAddress remoteAddress) at Wcf.WcfClientBase1..ctor(String name, Boolean streaming) at Wcf.WcfClientBase`1..ctor(String name) at Wcf.AuthenticationClient..ctor() at WindowsPhoneApplication2.MainPage.DoLogin() ... Any ideas?

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  • Best way to 'harden' embedded ext4 file server against unexpected loss of power?

    - by Jeremy Friesner
    Hi all, First, a little background: my company makes an audio streaming device that is a headless, rack-mounted Linux box with a couple of SSDs attached. Each SSD is formatted with ext4. The users can connect to the system using Samba/CIFS to upload new audio files or access existing ones. There is also custom software for streaming out audio over the network. This is all fine. The only problem is that the users are audio people, not computer people, and see the system as a 'black box', not as a computer. Which means that at the end of the day, they aren't going to ssh in to the box and enter "/sbin/shutdown -h"; they are just going to cut power to the rack and leave, and expect things to still work properly the next day. Since ext4 has journalling, journal checksumming, etc, this mostly works. The only time it doesn't work is when someone uploads a new file via Samba and then cuts power to the system before the uploaded data has been fully flushed to the disk. In that case, they come in the next day and find that their new file has been truncated or is missing entirely, and are unhappy. My question is, what is the best way to avoid this problem? Is there a way to get smbd to call "sync" at the end of every upload? (Performance on uploads isn't so important, since they only happen occasionally). Or is there a way to tell ext4 to automatically flush within a few seconds of any change to a file? (Again, performance can be sacrificed for safety here) Should I set a particular write-ordering mode, activate barriers, etc?

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • AVConv increases song duration when converting MP3

    - by chauffch
    I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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  • How to embed/hardcode SRT subtitles into mp4 videos with VLC?

    - by Jens Bannmann
    I'm looking for a way to "burn in" or render/rembed/hardcode subtitles (from an SRT file) into an MP4 video with VLC. But no matter what options I use, it never works properly. I get a file that plays video way too fast (audio is normal), or one that plays normally, but actually does not have embedded subtitles. Also, with some options (like the one below) it does not play in QuickTime, only in VLC. So the main question is: how can I make this work in VLC? Secondary questions are: How do I decide which options I should set? Which settings are best if I want to leave the file bitrate etc. the same as much as possible, only embed subtitles? It seems I cannot leave the field empty or Video/Audio unchecked, so I guess I would first need to figure out the original audio and video bitrate. What do the "Scale" and "Channels" options mean? ... none of which are answered within the VLC documentation. For example, this is one set of options I used in the "Advanced Open File…" dialog: Advanced Open File… myFileName.mp4 [ ] Treat as a pipe rather than as a file [x] Load subtitles file: mySubtitleFileName.srt [ ] Play another media synchronously [x] Streaming/Saving Streaming and Transcoding Options [ ] Display the stream locally (o) File [outputFileName.mp4 ] [ ] Dump raw input Encapsulation Method: (MPEG 4 ) Transcoding options [x] Video (mp4v ) Bitrate (kb/s) [256 ] Scale [1 ] [x] Audio (mp3 ) Bitrate (kb/s) [128 ] Channels [1 ]

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • Running computer from separate room

    - by Dan
    I want my computer to be in the basement, but to use it on the first floor. Which cables should I run through the floor? Can some be wireless or other methods? Here are some of the options I've thought of: Basic: Run DVI, usb (mouse), usb (keyboard), and audio cable (4 cables) USB Hub option: Run DVI, and 1 usb, then using a usb hub split it into mouse, keyboard, maybe even audio (2-3 cables) HDMI Option: If I get a new video card and monitor that supports HDMI, would I be able to run both audio and video through it? Would the monitor have to have an audio out? Also there is a lot of extra bandwidth in the HDMI cables, could I send two monitors on 1 cable or would I have to use 2 cables? How about sending mouse/keyboard through the HDMI cable? I see a lot of monitors with USB hubs built in, but I assume I'd still have to wire HDMI + 1 USB cable to use the USB hubs? X Terminal Machine/Thin Client: I don't really know much about this option. Not sure if it would allow me to run graphics acceleration and watch movies, does anyone know more details about what this would allow me to do? Other options: Any other ways to do this? Can any of this be wireless?

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  • Lync Server 2010

    - by ManojDhobale
    Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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  • Sound issues after trying everything

    - by Lerp
    I cannot get my sound working properly, no matter what I do, there's always some problem. It's very annoying as it's the only thing preventing me from making Ubuntu my main OS. At the moment my sound always plays through both my speakers and my headphones regardless except the sound through the headphones is crackly. It is also a bit quiet even though everything is maxed. I've managed to improve the situation to a point where the sound out of my speakers is perfect but I have none at all from my headphones. I do have two connectors listed in the sound settings but regardless of which one is selected it always plays through the speakers. I think this might have something to do with the fact that my speakers are plugging into the front of my computer, typically the headphone jack, and my headphones are plugging into the back but when I try disconnecting the speakers from the front there is still no sound from the headphones. I fixed the speaker sound by going through the sound settings and making sure they were all set to 100% then rebooting. Things I have tried: Maxing everything and unmuting everything in alsamixer Uninstalling pulseaudio Making gstreamer use only alsa via gstreamer-properties. This worked with the sound test button including independent sound between headphones and speakers but when I reset the computer it no longer worked. So I tried setting it manually in gconf-editor which didn't work either. Reinstalling alsa and pulseaudio Setting the model in /etc/modprobe.d/alsa-base.conf to 6stack and 6stack-dig neither worked. Upgrading to 12.10 Here's some command output to help you diagnose my problem. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 sudo lshw -C sound *-multimedia description: Audio device product: 82801JI (ICH10 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:70 memory:f7ff8000-f7ffbfff cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 options snd-hda-intel model=6stack

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  • How to setup equivalent USVIDEO.ORG DNS-Proxy on Linux

    - by Gary
    I have a VPS in the USA running Ubuntu. I want to setup something similar to http://www.usvideo.org Basically, USVIDEO is a DNS service that allows Canadians to access American content like Hulu, Netflix, NBC, and etc (restricted by geographical IP). Here is how I think USVideo does it: Clients (PS3, XBOX, PC) specifies the DNS server(s) as specified on USVIDEO.org's website. If the DNS request is a video/audio site such as Netflix or Pandora, forward the request to a proxy. Otherwise, for all other requests, forward it to a different DNS server. If the specific video/audio URL is requested, return the address of the proxy server, which in turn relays traffic to the destination video/audio domain via the U.S. gateway so that it appears that the access is coming from a U.S. IP address. Once the DNS request has passed the U.S. IP address check, their proxy server steps out of the loop and lets the video streaming site contact you directly to start the video stream. This trick relies on the way that the video streaming sites check the country of your IP address once up front, but don't actually check the country of the destination IP address while the video is streaming. What is elegant about this solution is that a VPN Tunnel is not required to bypass geographical IP checks from certain websites. All that is required on the client side is to specify the DNS server (the VPS). If a certain site is geographically locked, just forward the traffic to a proxy, and that's it. These sites can be specified in the DNS entries, or perhaps in the proxy service to redirect the DNS request to its own proxy. I believe what I need to setup something similar is Squid Proxy, IPTables, and DNS. What I need help is how to exactly approach this? Would Squid Proxy be setup as a transparent proxy?

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  • Nginx, HAproxy, Unicorn, Rails and Node settings

    - by Julien Genestoux
    Our application is currently only a "regular" web app, with no fancy things like streaming HTTP or websockets. It's mostly a Rails app, served by a few (20 on 2 machines) Unicorn workers, proxied by a venerable nginx server which deals with load balancing. This has been working quite well for the past year and the app now serves between 400 and 800 requests per second at any point during the day. We're soon releasing 2 new APIs, which are both served by a Node application : a websocket one, as well as a long polling HTTP one. (the fancy thing like the Twitter streaming API where HTTP connections never end). They both use the same port on node and since the node app is stateless, we can certainly deploy a few of them to handle the traffic. The app (node) is now deployed in 5 instances and are now listening on 5 different 'private' ports on the same host. We need to put something in front of them to load balance, but also something that is able to deal with sockets (either websocket or HTTP streaming) which are intended to stay 'up' for days. The question is then : what? I read somewhere that HAProxy does a better job than Nginx at this. What do you recommend?

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  • How to disable Bluetooth auto-answer on Windows 7?

    - by MarkR
    When my BlackBerry 9630 is connected via Bluetooth to my Dell desktop running Windows 7 x64 there are a bunch of Bluetooth services enabled on Windows Advanced Audio BB Bypass service BB Desktop Service Dial-up Networking Headset Audio Gateway Remote Control Remotely Controlled Device When a call arrives on the 9630 Windows immediately answer the call. This is annoying when windows answers the phone before I even hear an audible ring and the hapless caller is saying, "hello. hello" on top of music I have playing. Does anyone know how to tell Windows not to answer the call? I want to be able to don my headset, switch audio to my headset, THEN manually answer the call.

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  • problem in AudioStreaming in Iphone Sdk?

    - by senthilmuthu
    I am using sample code of AudioStream.zip,but when i use this to play Mp3 file , it gives wrong total amount of playing time(after played completely through streaming).... i checked through downloading that Mp3 file into Document Directory and played in Itune it exactly is played for 2.10 seconds.but in streaming through that code(- (double)progress method) gives total playing time only 2.3 sec, is there any sample code for AudioStreaming except that one to give right Total playing Time?

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  • v4l - capture and watch at the same time

    - by John Barrett
    Capturing v4l and line-in audio using mencoder works very well, but I would like to record real-time gameplay video from consoles plugged into the video card. I've used xawtv for this (Works quite well, can preview and record in real time), but when I enable any deinterlacing or aspect ration options the video fails to record. I have to record raw and re-encode the video with the appropriate filters later to get something workable. Other things I have tried: tvtime with xvidcap and jack audio capture - xvidcap drops frames and muxing the audio is impossible as it will go out of sync (I have not found muxer options that work to force a correct frame rate) mencoder capture to file, attempt to pipe tail of file to mplayer... mencoder works great, piping the file is far too heavy to attempt gameplay. Soooo, v4l capture and preview simultaneously, recommendations?

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  • graphics cards with no HDMI output

    - by Noam Gal
    I am currently looking at Gainward GTX260 896MB GS GLH, but I've seen it also on the x275 version - the card doesn't have an HDMI output, only two dvi and one tv out (s-video?). They claim they support HDMI using a dvi-HDMI converter. Will I get a true high definition quality on my TV (assuming it supports it) like that? Or is it not as good, and I should stick to cards that have an HDMI output (ATI), or pay way too much for x295? What about connecting the audio? The x260 comes with an internal spdif cable - does that mean I can connect my soundcard to my graphics card, and have the audio come out through the dvi, and into the HDMI cable? Or am I mixing it all up here, and I have to somehow connect the sound to the TV using a seperate cable (Hoping it has a seperate audio-in for the HDMI channel)?

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  • Enabling Surround sound on a Realtek ALC892 via SPDIF, on Windows 7

    - by Alex
    I have a problem with my ALC892, on an ASRock mainboard (ASRock 890FX Deluxe4). I get only stereo sounds if I use SPDIF connection, in general. My amp shows that is getting surround sound only when I use the Test feature of Windows 7. This test feature allows to know which formats are supported by the audio chip. The tests render correctly both Dolby Digital and DTS. You can find this test under Sounds, Playback Devices, Select Digital Audio, then "Properties". I am using Windows 7 x64, with the latest drivers from the official Realtek website. I also tested other driver versions, both from the Realtek website and from the ASRock one, but had no luck. Thanks for the help. Some specs: CPU: AMD Phenom II X4 965 MOBO: ASRock 890FX Deluxe4 (with onboard Realtek ALC892) Audio amp: Onkyo R-380 (works fine with other sources like PS3 and Xbox 360)

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  • Use external speakers with laptop hooked to separate monitor?

    - by lhan16
    I have a laptop with a set of external speakers hooked up to it on my computer desk. The speakers use the standard 3.5mm audio (headphones) jack. The speakers work fine, but I've recently added a separate monitor to my laptop via HDMI. With the monitor hooked up to my laptop and the speakers still hooked up to the laptop, sound will only come out of the built-in monitor speakers. When I look at my audio settings, there are three different "audio playback devices" showing up, but only the built-in monitor speakers make noise when I click "test" (and I hear nothing when I set any of the other devices as the default. Does anyone know how I can still use my external speakers when using a separate monitor with my laptop? I'm hoping there is a solution that doesn't require the laptop to be open or closed, because I use both scenarios. I came across this post, but it doesn't look like they had much luck.

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  • Android: Use XML Layout for List Cell rather than Java Code Layout (Widgets)

    - by Stephen Finucane
    Hi, I'm in the process of making a music app and I'm currently working on the library functionality. I'm having some problems, however, in working with a list view (In particular, the cells). I'm trying to move from a simple textview layout in each cell that's created within java to one that uses an XML file for layout (Hence keeping the Java file mostly semantic) This is my original code for the cell layout: public View getView(int position, View convertView, ViewGroup parent) { String id = null; TextView tv = new TextView(mContext.getApplicationContext()); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); id = musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); tv.setText(id); } else tv = (TextView) convertView; return tv; } And my new version: public View getView(int position, View convertView, ViewGroup parent) { View cellLayout = findViewById(R.id.albums_list_cell); ImageView album_art = (ImageView) findViewById(R.id.album_cover); TextView album_title = (TextView) findViewById(R.id.album_title); TextView artist_title = (TextView) findViewById(R.id.artist_title); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); album_title.setText(musiccursor.getString(music_column_index)); //music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); //musiccursor.moveToPosition(position); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); artist_title.setText(musiccursor.getString(music_column_index)); } else{ cellLayout = (TextView) convertView; } return cellLayout; } The initialisation (done in the on create file): musiclist = (ListView) findViewById(R.id.PhoneMusicList); musiclist.setAdapter(new MusicAdapter(this)); musiclist.setOnItemClickListener(musicgridlistener); And the respective XML files: (main) <?xml version="1.0" encoding="utf-8"?> <LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" android:orientation="vertical" android:layout_width="fill_parent" android:layout_height="fill_parent"> <ListView android:id="@+id/PhoneMusicList" android:layout_width="fill_parent" android:layout_height="fill_parent" /> <TextView android:id="@android:id/empty" android:layout_width="wrap_content" android:layout_height="0dip" android:layout_weight="1.0" android:text="@string/no_list_data" /> </LinearLayout> (albums_list_cell) <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/albums_list_cell" android:layout_width="wrap_content" android:layout_height="wrap_content"> <ImageView android:id="@+id/album_cover" android:layout_alignParentLeft="true" android:layout_alignParentTop="true" android:layout_width="50dip" android:layout_height="50dip" /> <TextView android:id="@+id/album_title" android:layout_toRightOf="@+id/album_cover" android:layout_alignParentTop="true" android:layout_width="wrap_content" android:layout_height="wrap_content" /> <TextView android:id="@+id/artist_title" android:layout_toRightOf="@+id/album_cover" android:layout_below="@+id/album_title" android:layout_width="wrap_content" android:layout_height="15dip" /> </RelativeLayout> In theory (based on the tiny bit of Android I've done so far) this should work..it doesn't though. Logcat gives me a null pointer exception at line 96 of the faulty code, which is the album_title.setText line. It could be a problem with my casting but Google tells me this is ok :D Thanks for any help and let me know if you need more info!

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  • XPP-32 over W7-64 on music production laptop

    - by quarlo
    I need to upgrade my laptop and need high performance for music production (recording and mixing). My audio interface manufacturer seems to be unable to successfully convert their drivers to 64-bit. I do not trust a virtual machine to handle real-time audio recording at low enough latency so ... I would like to install XP Pro 32-bit on a separate partition and dual boot since most of the machines that can handle this application now ship with Windows 7 64-bit flavors. I'd like to transit to 64-bit over time assuming M-Audio does eventually get a handle on 64-bit drivers, but really need to ensure that I can stay at 32-bit for now. Does anyone have any experience with this or something similar?

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  • How do I concatenate a lot of files into one inside Hadoop, with no mapping or reduction

    - by Leonard
    I'm trying to combine multiple files in multiple input directories into a single file, for various odd reasons I won't go into. My initial try was to write a 'nul' mapper and reducer that just copied input to output, but that failed. My latest try is: vcm_hadoop lester jar /vcm/home/apps/hadoop/contrib/streaming/hadoop-*-streaming.jar -input /cruncher/201004/08/17/00 -output /lcuffcat9 -mapper /bin/cat -reducer NONE but I end up with multiple output files anyway. Anybody know how I can coax everything into a single output file?

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  • Video Recording on iPhone

    - by gn-mithun
    I have this iPhone app, which plays streaming video from a source(Just video, no audio). Is there any mechanism by which i can capture this streaming video or record them, for later playback?

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  • How can I access the external microphone with Ubuntu?

    - by Charles Merriam
    Sound in Ubuntu, it has its own special joy. I would like my external microphone to work. Symptoms: I can play sound through the speakers I can play sound through the headsets. Plugging and and plugging headphone output correctly switches. I can record from the built-in microphone, using "Sound Recorder" and others. but: I cannot record from the external microphone. My Sound Preferences/Input panel has no option for an external microphone. If the answer is upgrade the ALSA drivers, please say exactly what to type. Thank you. ======== I'm using Ubuntu 9.10 Karmic Koala on a laptop (Gateway W3501), Sigmatel. That is: ~$ head -1 /proc/asound/card0/code* ==> /proc/asound/card0/codec#0 <== Codec: SigmaTel STAC9205 ==> /proc/asound/card0/codec#1 <== Codec: Conexant ID 2c06 ~$ lspci | grep -i audio 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)

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  • Batch Convert .mkv to .mp4

    - by IamHere
    I want to batch convert all .mkv files in a folder into .mp4 with VLC. It should use the original video-/audio stream and if possible the .ass subtitle of the .mkv. It's not really a conversion, it's more like changing the container – my player can't read the MKV videos. If I do this conversion by hand (manually) it works, but I have a lot of MKV files to convert, so it would take a lot of time. I have searched the internet for a batch file to do this and I found a few. I tried to modify them to my wish, but all attempts I tried just created a .mp4 file that doesn't contain the audio stream and the video stream also cannot be rendered by all my media players on the PC. So could someone tell me how the batch has to look like, so it works with the original video and audio stream (and maybe .ass subtitles)?

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  • What is the best free service to host images and mp3 files?

    - by Edward Tanguay
    I am making an educational social software silverlight application. I would like users to be able to point the application to a URL with text, images, and audio files which they have created. Many users will not have their own website to do this, so we are looking for a free service they can use to upload, and manage their own text/image/audio content. What is the best free service for non-technical users to upload and make available text, images and audio? For instance, sites.google.com allows you to upload pictures and access them via http so that would work, but that is more about making a website. For this purpose we just need the ability to upload files, without the website creation tools.

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