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  • xvidcap: Error accessing sound input from /dev/dsp

    - by stivlo
    I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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  • Will proprietary software-based sound enhancements work with Ubuntu? (BeatsAudio, Dolby)

    - by LiveWireBT
    This question is targeted at mainstream or gamer-grade software-based audio/sound enhancements, found in highly integrated computing and entertainment systems like laptops, tablets and smartphones. These are mostly marketed with fancy badges of known audio-releated brands on the product or packaging, while being mostly uncertain about the actual implementation or components used and poorly differentiated from the general audio capabilities of the system or device. This question is not about actual hardware like speakers. If your headphones are not properly detected, your speakers are assigned wrong, work partially or not at all then your soundcard or chip is not properly detected and you should take a look at troubleshooting audio issues. This question is also not about enthusiast or recording-grade hardware like recording interfaces, amplifiers and DACs in a variety of formfactors. And this question is also not about audio encoding and playback of different audio formats like Dolby Digital, Dolby TrueHD and DTS. Most of these may be subject to patents and licensing, see restricted formats. If you are just searching for an equalizer, please take a look at this question: Is there any Sound enhancers/equalizer? Simply speaking: Every feature where you would flip a switch or check a box in a fancy looking interface in Windows that makes the sound change from neutral to fancy.

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  • I can't hear any sounds on ubuntu 11.10 on Dell inspiron N5010

    - by Ahmed
    I have a problem that I can't hear any sounds and I don't know where to start. I did the following : lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at fbf00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel -- 01:00.1 Audio device: ATI Technologies Inc Manhattan HDMI Audio [Mobility Radeon HD 5000 Series] Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at fbe40000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel And It seems that I have 2 soundcards. Is that normal ?? I also did this: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 Also on the sound setting GUI. I have 2 hardware profiles for sound cards but none of them works when I test the speakers. Where should I start searching ?

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  • How to call Twiter's Streaming/Filter Feed with urllib2/httplib?

    - by Simon
    Update: I switched this back from answered as I tried the solution posed in cogent Nick's answer and switched to Google's urlfetch: logging.debug("starting urlfetch for http://%s%s" % (self.host, self.url)) result = urlfetch.fetch("http://%s%s" % (self.host, self.url), payload=self.body, method="POST", headers=self.headers, allow_truncated=True, deadline=5) logging.debug("finished urlfetch") but unfortunately finished urlfetch is never printed - I see the timeout happen in the logs (it returns 200 after 5 seconds), but execution doesn't seem tor return. Hi All- I'm attempting to play around with Twitter's Streaming (aka firehose) API with Google App Engine (I'm aware this probably isn't a great long term play as you can't keep the connection perpetually open with GAE), but so far I haven't had any luck getting my program to actually parse the results returned by Twitter. Some code: logging.debug("firing up urllib2") req = urllib2.Request(url="http://%s%s" % (self.host, self.url), data=self.body, headers=self.headers) logging.debug("called urlopen for %s %s, about to call urlopen" % (self.host, self.url)) fobj = urllib2.urlopen(req) logging.debug("called urlopen") When this executes, unfortunately, my debug output never shows the called urlopen line printed. I suspect what's happening is that Twitter keeps the connection open and urllib2 doesn't return because the server doesn't terminate the connection. Wireshark shows the request being sent properly and a response returned with results. I tried adding Connection: close to my request header, but that didn't yield a successful result. Any ideas on how to get this to work? thanks -Simon

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  • Writing a Makefile.am to invoke googletest unit tests

    - by jmglov
    I am trying to add my first unit test to an existing Open Source project. Specifically, I added a new class, called audio_manager: src/audio/audio_manager.h src/audio/audio_manager.cc I created a src/test directory structure that mirrors the structure of the implementation files, and wrote my googletest unit tests: src/test/audio/audio_manager.cc Now, I am trying to set up my Makefile.am to compile and run the unit test: src/test/audio/Makefile.am I copied Makefile.am from: src/audio/Makefile.am Does anyone have a simple recipe for me, or is it to the cryptic automake documentation for me? :)

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  • How to remove music/videos DRM protection and convert to Mobile Devices such as iPod, iPhone, PSP, Z

    - by tonywesley
    The music/video files you purchased from online music stores like iTunes, Yahoo Music or Wal-Mart are under DRM protection. So you can't convert them to the formats supported by your own mobile devices such as Nokia phone, Creative Zen palyer, iPod, PSP, Walkman, Zune… You also can't share your purchased music/videos with your friends. The following step by step tutorial is dedicated to instructing music lovers to how to convert your DRM protected music/videos to mobile devices. Method 1: If you only want to remove DRM protection from your protected music, this method will not spend your money. Step 1: Burn your protected music files to CD-R/RW disc to make an audio CD Step 2: Find a free CD Ripper software to convert the audio CD track back to MP3, WAV, WMA, M4A, AAC, RA… Method 2: This guide will show you how to crack drm from protected wmv, wma, m4p, m4v, m4a, aac files and convert to unprotected WMV, MP4, MP3, WMA or any video and audio formats you like, such as AVI, MP4, Flv, MPEG, MOV, 3GP, m4a, aac, wmv, ogg, wav... I have been using Media Converter software, it is the quickest and easiest solution to remove drm from WMV, M4V, M4P, WMA, M4A, AAC, M4B, AA files by quick recording. It gets audio and video stream at the bottom of operating system, so the output quality is lossless and the conversion speed is fast . The process is as follows. Step 1: Download and install the software Step 2: Run the software and click "Add…" button to load WMA or M4A, M4B, AAC, WMV, M4P, M4V, ASF files Step 3: Choose output formats. If you want to convert protected audio files, please select "Convert audio to" list; If you want to convert protected video files, please select "Convert video to" list. Step 4: You can click "Settings" button to custom preference for output files. Click "Settings" button bellow "Convert audio to" list for protected audio files Click "Settings" button bellow "Convert video to" list for protected video files Step 5: Start remove DRM and convert your DRM protected music and videos by click on "Start" button. What is DRM? DRM, which is most commonly found in movies and music files, doesn't mean just basic copy-protection of video, audio and ebooks, but it basically means full protection for digital content, ranging from delivery to end user's ways to use the content. We can remove the Drm from video and audio files legally by quick recording.

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  • ALSA samples capture: cannot open device

    - by Randagio
    I'm quite new to Linux (Lubuntu 12.04 for sake of precision) and ALSA programming at all. I'm trying to write a C program to capture audio from internal PC microphone for processing it. So as first step I google a bit and I found this article for capturing audio samples A tutorial on using the ALSA Audio API but when I compile it and execute it with: ./capture "default" or ./capture "hw:0,0" and all the possible variants on theme it always raises the error: cannot open device hw:0,0 (no such file or directory). So the issue is: what is the name of the mic audio device to pass as parameter to record the audio from mic ? The mic is working ok because the Sound Recorder program records sounds perfectly and I can playback them. The output of the aplay -l is the following : **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 and this is the amixer output (cut) Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 4 [13%] [-40.50dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [-1.50dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] so for aplay it seems I have no recording device, but for amixer I've got the mic, a mic boost and mic stereo as well with all those gorgeous stuffs on their place !!. If so, how could my Sound Recorder record the audio without any problem at all ?!?! For sure I'm giving the wrong device name to the command line for capturing audio but I'm loosing the hope for finding the correct one ! Please help....before I tear my hair out !!!

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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • Streaming binary data to WCF rest service gives Bad Request (400) when content length is greater than 64k

    - by Mikey Cee
    I have a WCF service that takes a stream: [ServiceContract] public class UploadService : BaseService { [OperationContract] [WebInvoke(BodyStyle=WebMessageBodyStyle.Bare, Method=WebRequestMethods.Http.Post)] public void Upload(Stream data) { // etc. } } This method is to allow my Silverlight application to upload large binary files, the easiest way being to craft the HTTP request by hand from the client. Here is the code in the Silverlight client that does this: const int contentLength = 64 * 1024; // 64 Kb var request = (HttpWebRequest)WebRequest.Create("http://localhost:8732/UploadService/"); request.AllowWriteStreamBuffering = false; request.Method = WebRequestMethods.Http.Post; request.ContentType = "application/octet-stream"; request.ContentLength = contentLength; using (var outputStream = request.GetRequestStream()) { outputStream.Write(new byte[contentLength], 0, contentLength); outputStream.Flush(); using (var response = request.GetResponse()); } Now, in the case above, where I am streaming 64 kB of data (or less), this works OK and if I set a breakpoint in my WCF method, and I can examine the stream and see 64 kB worth of zeros - yay! The problem arises if I send anything more than 64 kB of data, for instance by changing the first line of my client code to the following: const int contentLength = 64 * 1024 + 1; // 64 kB + 1 B This now throws an exception when I call request.GetResponse(): The remote server returned an error: (400) Bad Request. In my WCF configuration I have set maxReceivedMessageSize, maxBufferSize and maxBufferPoolSize to 2147483647, but to no avail. Here are the relevant sections from my service's app.config: <service name="UploadService"> <endpoint address="" binding="webHttpBinding" bindingName="StreamedRequestWebBinding" contract="UploadService" behaviorConfiguration="webBehavior"> <identity> <dns value="localhost" /> </identity> </endpoint> <host> <baseAddresses> <add baseAddress="http://localhost:8732/UploadService/" /> </baseAddresses> </host> </service> <bindings> <webHttpBinding> <binding name="StreamedRequestWebBinding" bypassProxyOnLocal="true" useDefaultWebProxy="false" hostNameComparisonMode="WeakWildcard" sendTimeout="00:05:00" openTimeout="00:05:00" receiveTimeout="00:05:00" maxReceivedMessageSize="2147483647" maxBufferSize="2147483647" maxBufferPoolSize="2147483647" transferMode="StreamedRequest"> <readerQuotas maxArrayLength="2147483647" maxStringContentLength="2147483647" /> </binding> </webHttpBinding> </bindings> <behaviors> <endpointBehaviors> <behavior name="webBehavior"> <webHttp /> </behavior> <endpointBehaviors> </behaviors> How do I make my service accept more than 64 kB of streamed post data?

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  • Using the Onboard VGA output with a PCIe video card. Both nVidia

    - by sebikul
    I have 2 video cards, one On board, a nVidia 6150SE nForce 430 and a PCIe nVidia GeForce GT 220 1GB DDR2 RAM I have already configured the PCIe card to use the dual monitor feature, using the VGA and HDMI ports, but now I want to add a third monitor, using the On board VGA port I have managed to enable the On board graphics processor, which is taking 400MB of ram, but I cant manage to use it, nvidia-settings does not detect it, like it's not usable (but is there) My questions are the following: How can I manage to get the On board VGA display to work together with the PCIe graphics card? If possible, how can I recover those 400 MB the on board card is taking (even without being used) or how can I get it to use the PCIe card available memory? System Details: Linux 2.6.35-28-generic i686 Ubuntu 10.10 (All updates installed) NVIDIA Driver Version: 260.19.06 (Official) If more info is needed please let me know. Here is the lspci output when the On board card is disabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) And this is when both are enabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) Output of lshw -class display: *-display description: VGA compatible controller product: GT216 [GeForce GT 220] vendor: nVidia Corporation physical id: 0 bus info: pci@0000:02:00.0 version: a2 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:18 memory:df000000-dfffffff memory:c0000000-cfffffff memory:da000000-dbffffff ioport:ef80(size=128) memory:def80000-deffffff *-display description: VGA compatible controller product: C61 [GeForce 6150SE nForce 430] vendor: nVidia Corporation physical id: d bus info: pci@0000:00:0d.0 version: a2 width: 64 bits clock: 66MHz capabilities: pm msi vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:22 memory:dd000000-ddffffff memory:b0000000-bfffffff memory:dc000000-dcffffff memory:deb40000-deb5ffff If what I'm looking for is not possible, please tell me, so I can disable the On board card and recover those 400MB of wasted RAM Thanks for your help!

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  • How can I synchronize a text with audio/sound in XNA/XACT?

    - by Omkar
    Hello Geeks, I wanted to display the text while sound is playing at background. In short if there is sound/audio for "What is this", I want to display the text "What is this" in text box synchronously. Is this possible with XNA/XACT? and can I use this in standard C# based WPF or Silverlight applications? Appreciating your help.

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  • How to stream audio from ASP.NET MVC controller when it's still encoding?

    - by kyrisu
    Background I have wave files on my server that I want to stream. Because of the size I want to encode them to mp3. I've tried to use FileStreamResult - but it doesn't work because as soon as program leaves the controller stream is closed and I get - "Cannot access a closed stream" FileContentResult - but it's not a stream and the user would need to wait for encoding to finish Question Is there a way to stream audio from the controller while it's still encoding?

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  • convert decrypted .vobs to .avi with ffmpeg on ubuntu

    - by Arcath
    I have a .vob file that has bee ripped from a dvd, when I watch the .vob its very good quality video and 5.1 english audio but when I use ffmpeg it has rubbish video and mono french audio. That was using this command: ffmpeg -i /samba/ripping/vobs/12161840#2.vob -f avi /samba/ripping/avis/test.avi I've tried a few different variations on that but it never comes back with anything good just bigger files with bad video and incorrect sound. I know the videos good and the correct audio streams exist so how do I select a 5.1 track and get good video? ffmpeg gives the .vob details as: Input #0, mpeg, from '/samba/ripping/vobs/12161840#2.vob': Duration: 00:42:05.56, start: 0.287267, bitrate: 5738 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 8436 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, mono, s16, 192 kb/s Output #0, avi, to '/samba/ripping/avis/test.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.3 -> #0.1

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  • How to write a Media Center plugin like the Netflix plugin? Source code/reference samples?

    - by Vin
    I am looking to write a Windows Media Center plugin just like the Netflix WMC plugin. Once logged in, I know the streaming urls that I need to hook in to. Any source code, reference samples would be great. Found one on codeplex for swedish TV channels, but right now it's not working for some reason... Previously asked the following question, with no answers, so updated with a question asked in a easy to relate fashion Host a streaming video in my client, from a streaming url that is behind a login session? I am building a Silverlight 4 desktop client to show streaming video from a site that is login based. So that website has a Silverlight player that does streaming video, the player is behind a login sesion, so just by getting the url from fiddler and trying to play it in my Silverlight 4 desktop client won't work. Actually after that, I want to build a Windows Media Center plugin to build a Netflix-like client, that allows login through WMC and then allows you to watch streaming video. Any pointers on how to go about doing any of this?

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  • Is it possible to broadcast audio to shoutcast / icecast / other server? from flash player?

    - by Jeffrey
    I am trying to create a flash client that can stream audio to an online radio server. Theoretically a user could enter the server info / login, and then connect and start sending data to the server which could then be broadcasted and listened to by other clients. I don't think this would be very hard, but am unsure about what data formats to use and what is the best server for the job. I'd like to be able to use one of the most popular radio servers like shoutCast. Any ideas? Thanks in advance.

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  • VST plugin : using FFT on audio input buffer with arbitrary size, how ?

    - by Led
    I'm getting interested in programming a VST plugin, and I have a basic knowledge of audio dsp's and FFT's. I'd like to use VST.Net, and I'm wondering how to implement an FFT-based effect. The process-code looks like public override void Process(VstAudioBuffer[] inChannels, VstAudioBuffer[] outChannels) If I'm correct, normally the FFT would be applied on the input, some processing would be done on the FFT'd data, and then an inverse-FFT would create the processed soundbuffer. But since the FFT works on a specified buffersize that will most probably be different then the (arbitrary) amount of input/output-samples, how would you handle this ?

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  • Successfully concatenating multiple videos

    - by wiseguydigital
    My mission is to create videos out of old web slideshows. To start with I have jpegs and audio files that worked as Flash slideshows in an old system, structured such as this: Audio structure my_audio_1.mp3 (this file is a 3 second mp3 of silence) my_audio_2.mp3 my_audio_3.mp3 my_audio_4 etc... roughly 30 mp3s per slideshow Image structure my_image_1.jpg (this acts as the opening slide) my_image_2.jpg my_image_3.jpg my_image_4. etc... roughly 30 images per slideshow. As there are almost 100 slideshows that must be converted to video, I have created a web-based interface using PHP to automate the process, that sits on a local system and attempts to combine the files using shell_exec(). The process uses the following workflow: Loop through each slide and make an avi or mpeg. So for instance my_mini_video_2.avi would be a video that consists of my_image_2.jpg and has a soundtrack of my_audio_2.mp3. This slide would last the length of my_audio_2.mp3. Join / stitch / concat all of the mini videos to create the final video (Using a combination of cat and either mencoder or ffmpeg (I have also tried avimerge but to no avail). Transcode the new 'master' video to various formats such as flv etc. I thought this would be simple and have been close on many occasions but it still won't work. I can't get past stage 2 as I can't get a perfect 'master' video. I have now experimented with Mencoder, FFMpeg and seem to have been through every combination I can think of. The problem is that the audio and visuals never sync, no matter what I try. Also, I have even tried created audio-less mini videos, joining the MP3s into one long MP3 using both cat and mp3wrap and then assigning the new long MP3 as the audio track, but this always produces either a very short file or a badly slowed down file and makes the female voiceover sound like a male boxer!!! There appears to be no problems at all with the original files. Does anybody have any experience in producing a video successfully from the same kind of starting point? Or any ideas on what I may be doing wrong? As an example: If I create silent mini-videos, and stitch them together into 'temp-master.mpg' and then join the MP3s together into single MP3 called 'temp-master-audio.mp3', the audio file's duration is 09:10 and the video file's duration is 08:35. They should be the same and the audio will seem sloooow. I haven't posted code as I have written lots and lots of combinations.

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  • Reproduce PIPE functionality in IronPython

    - by Muppet Geoff
    Hi, I am hoping some genious out there can help me out with this... I am using sox to merge and resample a group of WAV files, and pipe the output directly to the input of NeroAACEnc for encoding to AAC format. I originally ran the process in a script, which included: sox.exe d:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav - | neroAacEnc.exe -q 0.5 -if - -of test.m4a This worked as expected. The '-' in the comand line translates as 'Pipe/redirect input/output (stdin/stdout)' - So Sox pipes to stdout, and NeroAACEnc reads from stdin, the | joins them together. I then migrated the whole solution to Python, and the equivalent command became: from subprocess import call, Popen, PIPE runwav = Popen(['sox.exe', 'd:\audio\1.wav', 'd:\audio\2.wav', 'd:\audio\3.wav', '-c', '1', '-r', '22050', '-t', 'wav', '-'], shell=False, stdout=PIPE) runm4b = call(['neroAacEnc.exe', '-q', '0.5', '-if', '-', '-of', 'test.m4a'], shell=False, stdin=runwav.stdout) This also worked like a charm, exactly as expected. Slightly more convoluted, but hey :) Well now I have to move it to IronPython, and the Subprocess module isn't available (the partial implementation that is, doesn't have Popen/PIPE support - plus it seems silly to add a custom library when there is probably a native alternative). I should mention here, that I opted for IronPython over C#, because I am comfortable with Python now - however, there is a chance of moving it again later to C# native, and I am using IronPython to ease myself into it :) I have no C# or .net experience. So far I have the following equivalent, that sets up the 2 processes: from System.Diagnostics import Process wav = Process() wav.StartInfo.UseShellExecute = False wav.StartInfo.RedirectStandardOutput = True wav.StartInfo.FileName = 'sox.exe' wav.StartInfo.Arguments = 'd:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav -' wav.Start() m4b = Process() m4b.StartInfo.UseShellExecute = False m4b.StartInfo.RedirectStandardInput = True m4b.StartInfo.FileName = 'neroAacEnc.exe' m4b.StartInfo.Arguments = '-q 0.5 -if - -of test.m4a' m4b.Start() I know that these 2 processes start (I can see Nero and Sox in the task manager) but what I can't figure out (for the life of me) is how to string the two output/input streams together, as with the previous two solutions. I have searched and searched, so I thought I'd ask! If anyone knows either: How to join the two streams with the same net result as the Python and Commandline versions; or A better way to acheive what I am trying to do. I would be extremely grateful! Many thanks in advance, Geoff P.S. A code sample based off the above would be awesome :) or a specific code example of a similar process that I can easily translate... this has broked my brayne.

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  • How to use infinit live streams with JAVE library? (Java, ffmpeg)

    - by Ole Jak
    So I want to use JAVE to save mp3 radio stream to my File system. I have this code for file saving but what shall I do to save a stream (stop on timer for ex) File source = new File("source.wav"); File target = new File("target.mp3"); AudioAttributes audio = new AudioAttributes(); audio.setCodec("libmp3lame"); audio.setBitRate(new Integer(128000)); audio.setChannels(new Integer(2)); audio.setSamplingRate(new Integer(44100)); EncodingAttributes attrs = new EncodingAttributes(); attrs.setFormat("mp3"); attrs.setAudioAttributes(audio); Encoder encoder = new Encoder(); encoder.encode(source, target, attrs);

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  • Trouble installing ubuntu server on virtualbox (osx)

    - by audio.zoom
    Hello all, I'm trying to install lucid lynx 10.04.2 server on a virtualbox on snow leopard. I have 2 server iso files freshly downloaded one i386 and one 64bit. When I try to start the virtual machine with either one set to be the cd drive I'm getting the same error: Failed to open a session for the virtual machine Ub. Failed to load VMMR0.r0 (VERR_SUPLIB_OWNER_NOT_ROOT). Unknown error creating VM (VERR_SUPLIB_OWNER_NOT_ROOT). Couldn't find anything on it on google so I'm trying to see if anyone else has dealt with this issue. Thanks much in advance! edit: just downloaded the 32bit desktop edition to same avail edit2: ran Disk Utility' replair permissions then restarted. New error VERR_SUPLIB_WORLD_WRITABLE (instead of VERR_SUPLIB_OWNER_NOT_ROOT)

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  • Does any economically-feasible publicly available software compare audio files to determine if they are dupes?

    - by drachenstern
    In the vein of this question http://unix.stackexchange.com/questions/3037/is-there-an-easy-way-to-replace-duplicate-files-with-hardlinks is there any software that will automatically parse a library of my songs and find the ones that really are duplicates that one can be eliminated? Here's an example: My brother used to be a huge fan of remixing CDs. He would take all of his favorite tracks and put them on one. Then he would use my computer to read them in. So now I have like 6 copies of Californication on my HDD, and they're all a few bytes difference overall. I have hundreds of songs in my library like this. I want to trim them down to having uniques. They don't all have correct ID3 tags, so figuring out that Untitled(74).mp3 is the same as californication.mp3 is the same as whowrotethis.mp3 is tricky. I do NOT want to consider a concert album and a studio album rip to be the same (if I just did artist/title matching I would end up with this scenario, which doesn't work for me). I use Windows (pick your platform) and will be getting an OSX box later in the year. I'll run Linux if that's what it takes to get it organized. I have unprotected AAC and mp3 files. Bonus points for messing with WAV or MIDI and bonus points for converting from those into MP3 (I can always use Audacity and LAME to convert later if I know they match or to convert ahead of time if that will make things easier). Are there any suggestions, or do I need to goto Programmers or SO and build a list of requirements for comparing these things and write the software myself?

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  • How does Amarok rip Audio CDs (in Ubuntu Lucid)?

    - by Hanno Fietz
    I'm in the process of moving my CD collection into my Amarok library. Mostly, it works great. Sometimes however, the process just hangs forever. The problem seems to occur at random (i. e. often, but not always at the same disk/track) and the consequences range from none (successful after cancel/retry) to Amarok's internal db becoming completely messed up. I would like to investigate and file a proper bug report or find a fix / workaround, but I don't understand how Amarok does the ripping. When all is working, there's a lame process encoding to a temporary file, which appears in my collection once it's finished. When the process hangs, that lame command is still there, but waiting forever for data on stdin, which seems to come from a third process. That seems to be kio_audiocd, but I don't know whether that's correct and what it's supposed to do. What's going on?

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