Search Results

Search found 345 results on 14 pages for 'microphone'.

Page 8/14 | < Previous Page | 4 5 6 7 8 9 10 11 12 13 14  | Next Page >

  • Problems with webcame

    - by Murtaza
    My friends is using windows 8.1 and he has a webcam. He says it is working very well even without installing drivers. Webcam is A4tech - PK-331F. I asked for this webcam for use. I plugged that that webcam and windows detected it and installed drivers automatically. Then I opened Skype for checking webcam. I got massage on top of Skype that "your webcam has microphone" when I clicked over it, it directed me to sound settings where I microphone of webcam was working. I went to videos settings but webcam wasn't listed their. As you can see in image. Webcam is listed in device manager but isn't listed in Skype. I am also using windows 7 professional. Any idea why it is happening?

    Read the article

  • Silverlight Cream for April 30, 2010 -- #852

    - by Dave Campbell
    In this Issue: Michael Washington, Tim Greenfield, Jaime Rodriguez, and The WP7 Team. Shoutouts: Mike Taulty has a pretty complete set of links up for information about VS2010, Silverlight, Blend, Phone 7 Upgrade Christian Schormann announced Blend for Windows Phone: Update Available, and has other links up as well. From SilverlightCream.com: Silverlight Simplified MVVM Modal Popup Michael Washington is demonstrating a modal popup in MVVM and also shows the implementation of a value converter XPath support in Silverlight 4 + XPathPad Tim Greenfield blogged about XPath support in Silverlight 4 and his XPathPad tool... check out what all you can do with it... then go grab it, or the source too! Windows phone capabilities security model Jaime Rodriguez is discussing the WP7 capabilities exposed with the latest refresh such as location services, microphone, media library, gamer services, phone dialoer, push notification... how to code for them and other tips. Windows Phone 7 Series Developer Training Kit The WP7 Team is discussing the WP7 capabilities exposed with the latest refresh such as location services, microphone, media library, gamer services, phone dialoer, push notification... how to code for them and other tips. Stay in the 'Light! Twitter SilverlightNews | Twitter WynApse | WynApse.com | Tagged Posts | SilverlightCream Join me @ SilverlightCream | Phoenix Silverlight User Group Technorati Tags: Silverlight    Silverlight 3    Silverlight 4    Windows Phone MIX10

    Read the article

  • Streaming desktop with avconv - severe sound issues

    - by Tommy Brunn
    I'm trying to do some live streaming in Ubuntu 12.10, but I'm having some problems with audio. More specifically, the quality is complete garbage and it's at least 10 seconds out of sync with the video. I'm using an excellent guide found here to set up my loopback devices so that I can combine the desktop audio with the microphone input. It seems to work, as I'm able to stream both audio and video to Twitch.tv. But, as I said, the audio quality is terrible. The microphone audio is very, very low, but if I increase it, I get a horrible garbled sound that is absolutely unbearable. Nothing like that is present during VoIP calls or when recording sound alone with the sound recorder, so it's not an issue with the microphone itself. The entire audio stream is also delayed about 10-15 seconds compared to the video stream. I put together an imgur album of my settings. Here is some example output from when I'm streaming: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [x11grab @ 0x162fd80] device: :0.0+570,262 -> display: :0.0 x: 570 y: 262 width: 1280 height: 720 [x11grab @ 0x162fd80] shared memory extension found [x11grab @ 0x162fd80] Estimating duration from bitrate, this may be inaccurate Input #0, x11grab, from ':0.0+570,262': Duration: N/A, start: 1353181686.735113, bitrate: 884736 kb/s Stream #0.0: Video: rawvideo, bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc [alsa @ 0x163fce0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x163fce0] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'pulse': Duration: N/A, start: 1353181686.773841, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1641ec0] w:1280 h:720 pixfmt:bgra [scale @ 0x1642480] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x165ae80] VBV maxrate unspecified, assuming CBR [libx264 @ 0x165ae80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x165ae80] profile Main, level 3.1 [libx264 @ 0x165ae80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=cbr mbtree=1 bitrate=712 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=712 vbv_bufsize=512 nal_hrd=none ip_ratio=1.25 aq=1:1.00 Output #0, flv, to 'rtmp://live.justin.tv/app/live_23011330_Pt1plSRM0z5WVNJ0QmCHvTPmpUnfC4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuv420p, 852x480, q=-1--1, 712 kb/s, 1k tbn, 30 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 712 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #1:0 -> #0:1 (pcm_s16le -> libmp3lame) Press ctrl-c to stop encoding frame= 17 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 32 fps= 31 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 40 fps= 23 q=29.0 size= 44kB time=0.03 bitrate=13786.2kbits/s dup=frame= 47 fps= 21 q=31.0 size= 93kB time=2.73 bitrate= 277.7kbits/s dup=0frame= 62 fps= 23 q=29.0 size= 160kB time=3.23 bitrate= 406.2kbits/s dup=0frame= 77 fps= 24 q=23.0 size= 209kB time=3.71 bitrate= 462.5kbits/s dup=0frame= 92 fps= 25 q=20.0 size= 267kB time=4.91 bitrate= 445.2kbits/s dup=0frame= 107 fps= 25 q=20.0 size= 318kB time=5.41 bitrate= 482.1kbits/s dup=0frame= 123 fps= 26 q=18.0 size= 368kB time=5.96 bitrate= 505.7kbits/s dup=0frame= 139 fps= 26 q=16.0 size= 419kB time=6.48 bitrate= 529.7kbits/s dup=0frame= 155 fps= 27 q=15.0 size= 473kB time=7.00 bitrate= 553.6kbits/s dup=0frame= 170 fps= 27 q=14.0 size= 525kB time=7.52 bitrate= 571.7kbits/s dup=0 frame= 180 fps= 25 q=-1.0 Lsize= 652kB time=7.97 bitrate= 670.0kbits/s dup=0 drop=32 //Here I stop the streaming video:531kB audio:112kB global headers:0kB muxing overhead 1.345945% [libx264 @ 0x165ae80] frame I:1 Avg QP:30.43 size: 39748 [libx264 @ 0x165ae80] frame P:45 Avg QP:11.37 size: 11110 [libx264 @ 0x165ae80] frame B:134 Avg QP:15.93 size: 27 [libx264 @ 0x165ae80] consecutive B-frames: 0.6% 0.0% 1.7% 97.8% [libx264 @ 0x165ae80] mb I I16..4: 7.3% 0.0% 92.7% [libx264 @ 0x165ae80] mb P I16..4: 0.1% 0.0% 0.1% P16..4: 49.1% 1.2% 2.1% 0.0% 0.0% skip:47.4% [libx264 @ 0x165ae80] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.1% 0.0% 0.0% direct: 0.0% skip:99.9% L0:42.5% L1:56.9% BI: 0.6% [libx264 @ 0x165ae80] coded y,uvDC,uvAC intra: 82.3% 87.4% 71.9% inter: 7.1% 8.4% 7.0% [libx264 @ 0x165ae80] i16 v,h,dc,p: 27% 29% 16% 28% [libx264 @ 0x165ae80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 14% 8% 8% 8% 7% 5% 7% [libx264 @ 0x165ae80] i8c dc,h,v,p: 47% 22% 20% 11% [libx264 @ 0x165ae80] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x165ae80] ref P L0: 96.4% 3.6% [libx264 @ 0x165ae80] kb/s:474.19 Received signal 2: terminating. Any ideas on how I can resolve this? The video delay is perfectly acceptable, so I wouldn't think that it's a network issue that's causing the delay in the audio. Any help would be appreciated.

    Read the article

  • My headset doesn't work [closed]

    - by Kristian Flatheim Jensen
    Hello! I am not sure if this question fits on this site :S if it doesn't please let me know and i remove it :) well... here is my problem I just got a steel series headset for christmas(yay!) and i quickly plugged it into my MacBook Pro. The audio output works fine, but i am having issues with the audio input :( i had these kinds of problems before and i think my mac may be broken :( but then i saw this post: http://www.biloca.com/blog/?p=25 and they talked about some power issues to the microphone on the Mac Mini... I did not quite get what they were discussing so i have a simple question :) Do any of you have an idea why my microphone doesn't work? Please help! :) Best Regards Kristian

    Read the article

  • Voice Control iOS

    - by Marc Tanis
    I would like to build a simple reader app for the iPad 2 that would allow users to navigate/read via voice controls. The app would allow the user to enter a mode where the microphone was live and listened for predefined keywords like 'down', 'up', 'next', 'back', 'home', etc. I don't want to reinvent the wheel on this so I'm just wondering first, if someone has done this already and if not, are there any good tutorials or SDKs available to help with recording someone's voice, and then comparing future output to see if it matches, or just dealing with the microphone in general?

    Read the article

  • How to receive a datastream from a device on your computer, in C#

    - by WebDevHobo
    I plan to build a small audio-recorder app in C#. My laptop has a built in Microphone that's always active, so I want to use that as an early-stage test. I would simply start recording, save the file as a .wav or even use the LAME dll to make it into an MP3. The problem is, I don't know how to contact that microphone. Do I use a library that can detect a device, or do I just catch a stream of bytes from the port that the device is on? I don't have any experience with receiving data from connected devices. I suppose that I'll need to enter all the data into a byte array and then Serialize that into a WAV file, but I'm not sure. Can I get some pointers on this subject?

    Read the article

  • android spectrum analysis of streaming input

    - by TheBeeKeeper
    for a school project I am trying to make an android application that, once started, will perform a spectrum analysis of live audio received from the microphone or a bluetooth headset. I know I should be using FFT, and have been looking at moonblink's open source audio analyzer ( http://code.google.com/p/moonblink/wiki/Audalyzer ) but am not familiar with android development, and his code is turning out to be too difficult for me to work with. So I suppose my questions are, are there any easier java based, or open source android apps that do spectrum analysis I can reference? Or is there any helpful information that can be given, such as; steps that need be taken to get the microphone input, put it into an fft algorithm, then display a graph of frequency and pitch over time from its output? Any help would be appreciated, thanks.

    Read the article

  • Problem in installation in My Hp g4 1226se

    - by vivek Verma
    1vivek.100 Dual booting error in Hp pavilion g4 1226se Dear sir or Madam, My name is vivek verma.... I am the user of my Hp laptop which series and model name is HP PAVILION G4 1226SE........ i have purchase in the year of 2012 and month is February.....the windows 7 home basic 64 Bit is already installed in in my laptop.... Now i want to install Ubuntu 12.04 Lts or 13.10 lts..... i have try many time to install in my laptop via live CD or USB installer....and i have try many live CD and many pen drive to install Ubuntu ... but it is not done......now i am in very big problem...... when i put my CD or USB drive to boot and install the Ubuntu......my laptop screen is goes the some black (brightness of my laptop screen is very low and there is very low visibility ) and not showing any thing on my laptop screen..... and when i move the my laptop screen.....then there is graphics option in this screen to installation of the Ubuntu option......and when i press the dual boot with setting button and press to continue them my laptop is goes for shutdown after 2 or 5 minutes..... ...... and Hp service center person is saying to me our laptop hardware has no problem.....please contact to Ubuntu tech support............. show please help me if possible..... My laptop configuration is here...... Hardware Product Name g4-1226se Product Number QJ551EA Microprocessor 2.4 GHz Intel Core i5-2430M Microprocessor Cache 3 MB L3 cache Memory 4 GB DDR3 Memory Max Upgradeable to 4 GB DDR3 Video Graphics Intel HD 3000 (up to 1.65 GB) Display 35,5 cm (14,0") High-Definition LED-backlit BrightView Display (1366 x 768) Hard Drive 500 GB SATA (5400 rpm) Multimedia Drive SuperMulti DVD±R/RW with Double Layer Support Network Card Integrated 10/100 BASE-T Ethernet LAN Wireless Connectivity 802.11 b/g/n Sound Altec Lansing speakers Keyboard Full size island-style keyboard with home roll keys Pointing Device TouchPad supporting Multi-Touch gestures with On/Off button PC Card Slots Multi-Format Digital Media Card Reader for Secure Digital cards, Multimedia cards External Ports 1 VGA 1 headphone-out 1 microphone-in 3 USB 2.0 1 RJ45 Dimensions 34.1 x 23.1 x 3.56 cm Weight Starting at 2.1 kg Power 65W AC Power Adapter 6-cell Lithium-Ion (Li-Ion) What's In The Box Webcam with Integrated Digital Microphone (VGA) Software Operating System: Windows 7 Home Basic 64bit....Genuine..... ......... Sir please help me if possible....... Name =vivek verma Contact no.+919911146737 Email [email protected]

    Read the article

  • How can I automatically switch to USB headset when plugged in?

    - by d3vid
    Whenever I plugged in my old audio jack headset, sound was immediately diverted from my speakers to the headset speakers, and the microphone was immediately available. When I plug in my new USB headset, I have to open Sound Preferences and switch both input and output to the headset. Is there any way to make this happen automatically? I'm using a Fujitsu-Siemens Amilo Pi laptop, Maverick and a Logitech H330 USB headset.

    Read the article

  • Silverlight Cream for November 22, 2011 -- #1172

    - by Dave Campbell
    In this Issue: XAMLGeek, WindowsPhoneGeek, Nigel Sampson, Jesse Liberty, Sumit Dutta(-2-), Dave Bost, Jared Bienz, Joost van Schaik, and Michael Crump. Above the Fold: Silverlight: "10 Laps around Silverlight 5 (Part 7 of 10)" Michael Crump WP7: "Using MVVMLight, ItemsControl, Blend and behaviors to make a ‘heads up compass’" Joost van Schaik Metro/WinRT/W8: "“Badevand” for Windows 8" XAMLGeek Shoutouts: Michael Palermo's latest Desert Mountain Developers is up Michael Washington's latest Visual Studio #LightSwitch Daily is up From SilverlightCream.com:“Badevand” for Windows 8XAMLGeek posted a Metro app that shows water and air temperature and rain level for 5 beaches in Copenhagen, Denmark... no source, but good to see people posting appsGetting Started with Windows Phone RemindersWindowsPhoneGeek digs into Reminders in this WP7.1 post... the code you need, description, and a project to downloadHelp my app has been revoked!Nigel Sampson had a surprise when his latest app was revoked on his device, and then another... read what the solution wasA Dozen Windows Phone Videos… And CountingJesse Liberty posted his 12th WP7.1 video on Channel 9 - all about Reminders in MangoPart 23 - Windows Phone 7 - Detect Operator and Network InformationSumit Dutta has 2 more parts to his WP7 quest up... this part 23 is about getting mobile operator information and hot to get network capabilities using Microsoft.Phone.Net.DeviceNetworkInformationPart 24 - Windows Phone 7 - Microphone RepeaterIn part 24, Sumit Dutta uses the Microsoft.Xna.Framework Microphone class to record and play back voice.31 Days of Mango | Day #13: Marketplace Test KitDave Bost is at the helm of Jeff Blankenburg's Day 13 in his 31 day quest, discussing the Marketplace Test Kit and showing how to use it to determine if your app is ready for certification31 Days of Mango | Day #12; Beta DistributionJeff Blankenburg's Day 12 is written by guest author Jared Bienz, and shows how to submit an application for Beta testingUsing MVVMLight, ItemsControl, Blend and behaviors to make a ‘heads up compass’Joost van Schaik has a tutorial up showing how to make a WP7 Compass app using MVVMLight, Expression Blend, and then shows his thoughts on using the ItemsControl and Behaviors... code, descriptions and a project to download.... and I think I got your name right for the first time, Joost :)10 Laps around Silverlight 5 (Part 7 of 10)Michael Crump put out part 7 of his Silverlight 5 series at SilverlightShow... this is actually part 2 of OS Integration with Silverlight covering, among other things, 64-bit browser support and Power AwarenessStay in the 'Light!Twitter SilverlightNews | Twitter WynApse | WynApse.com | Tagged Posts | SilverlightCreamJoin me @ SilverlightCream | Phoenix Silverlight User GroupTechnorati Tags:Silverlight    Silverlight 3    Silverlight 4    Windows PhoneMIX10

    Read the article

  • signal processing libraries

    - by khinester
    Are there any open source libraries/projects which work in a similar way to http://www.tagattitude.fr/en/products/technology? I am trying to understand the process. At first I thought this could work like when you send a fax to a fax machine. It is basically using the mobile phone’s microphone as a captor and its audio channel as a transporter. Are there any libraries for generating the signal and then being able to decode it?

    Read the article

  • Removing pulseaudio, about ALSAMixer

    - by allenskd
    I removed pulseaudio today because rosetta stone had conflicts identifying the microphone which kinda makes it useless to me to learn a new language. Thing is, Alsamixer seems to have screwed up so I'm not really familiar on which configuration file I have to tweak to make the whole system use alsamixer (actually it does... but the problem is this error) $ alsamixer ALSA lib pulse.c:229:(pulse_connect) PulseAudio: Unable to connect: Connection refused cannot open mixer: Connection refused Could anyone enlighten me on which configuration file I have to change?

    Read the article

  • Windows Phone 7 Prototype 001: Speech Recognition on WP7

    At some point in the future it will be awesome when you can just tell your computer what to do and it does it - without typing to help those of us with a blistering 11 WPM hunk and peck technique. Siri, a mobile digital assistant using speech recognition was voted best tech at SXSW. I dont know about that one. Although, I'm sure it will get better when Apple rebuilds it and  bundles on iPhone 5. So how would you do that on WP7? There have been some videos floating around showing Bing with some voice control so obviously the phone has speech recognition. So what options are there: System.Speech? Not included in WP7/SL Nuance software like Siri? No WP7/SL version yet. Invoking the SAPI dlls on the phone? No automation factory in WP7 SL. Web services using System.Speech and mic on the phone? YES! The last one was my least favorite but that works for now. I built a quick sample app to show how to do text-to-speech and speech recognition on WP7.   @eklimczak will not be happy with the developer designed UI. In this sample there is web service with provides access to the system.speech APIs in .NET. Basically its just passing around byte arrays. On the phone its using the XNA audio frameworks to play the text-to-speech stream and to record using the microphone. The code is pretty simple and you can download from the link at the end of this post. The only things to note are adjusting the WCF config to handle larger byte uploads and the Microphone API is a little weird with that 1 second buffer. It would be nice if you could just to mic.start and mic.end which would return an array of bytes instead of managing your own stream inside the buffer ready callback. Couple of downsides to this approach: Recoding from the phone has some static. Could be my code or the my mic is bad / not calibrated right. Having to make web service calls instead of local access is not ideal (Microsoft, please add an API for the SAPI dlls) Although in the context of an app like Siri its not so bad since you need to do web service lookups to get data back Speech recognition quality really depends on either a) a limited grammar set like that pizza grammar in the sample or b) training the recognizer. For the latter it would be annoying to have users train the system. Using the System.Speech stuff youd have to have a profile for each user. So until Microsoft adds some speech client APIs on the phone or Nuance releases a wp7 product, this is a decent workaround. In the future Id like to build something similar to Siri. I shall call it Iris in homage. Im a big fan of mobile speech apps because frankly its just not safe to Google while driving. Since some of my designer co-workers have been posting UI sketches for WP7, Id like to start posting some code prototypes for things I try out on the phone. That will probably last 2 weeks, but for the moment I have like 10 posts in the queue. Sample Code 100% guaranteed to work on my emulatorDid you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

    Read the article

  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

    Read the article

  • Java Sound API: Capturing sound output from a Target Port

    - by Kyle Kampy
    I'm writing a simple piece of software that streams audio over LAN. I have all of the network parts implemented, but what I'm stumbling on is using the Java Sound API. I have successfully captured audio from the microphone, and line-in, but I can't seem to capture from any target ports, like the speakers. My question is, is it possible to capture from the Master target port? Here is the piece of code that works on initializing the line. private boolean startCapture(){ try{ DataLine.Info info = new DataLine.Info( TargetDataLine.class, format); line = (TargetDataLine)AudioSystem.getLine(info); audioBuffer = new byte[bufferSize]; line.open(format); line.start(); return true; }catch(Exception e){ System.out.println("Exception thrown when capturing audio:\n" + e); return false; } } Running the code like this will just use the microphone as my line. Here is info about my sound system. Most important is probably the fact that I'm running Linux. Thanks in advance for any and all help you can give me.

    Read the article

  • moving audio over a local network using GStreamer

    - by James Turner
    I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

    Read the article

  • Android, phone call audio stream via wlan

    - by moppel
    I am planning on developing my specific voip app for android. Here's the scenario: when a phone call occurs I want to hear the person who's calling on my local pc speakers and I want to speak to him via my own pc microphone / headset. So I need to send the audio stream of both me and the person I am talking to via the wlan network. Something like this: ... onCallStateChanged(int state, String phoneNumber){ while(state == PhoneListener.CALL_STATE_OFFHOOK){ //while phone call is happaning //send incoming speech via wlan to pc //receive audiostream from pc microphone and direct it to the phone call } } ... Is this possible with the current Android API? (Actually it should be since voip apps are available in the market) I did some research in the Android API and all I found was the AudioManager which has constant named public static final int STREAM_VOICE_CALL; //The audio stream for phone calls But I don't know how to use it our how it should give me access to the actual audiostreams which I can send via network. How do I manage to do this? The connection would be realised by TCP sockets.

    Read the article

  • Home entertainment karaoke system

    - by Mehper C. Palavuzlar
    Here is what I have: 40" Sony Bravia LCD TV, 5+1 speaker system, lots of original Karaoke CDs, and of course, a microphone. To set up a karaoke entertainment system, what kind of hardware do I need? Are there any standalone karaoke players out there? I hope my only option is not having to connect my laptop to TV. I already have karaoke software on my laptop but I wanna step up to a higher level without the help of a computer.

    Read the article

  • How can I prevent static when PC is plugged into an amplified speaker system?

    - by Kyle
    I've plugged a computer into an amp, using a 1/8 inch male extension cord, into a female adapter, that adapts into a male microphone 1/4 end. That being said, the amp sits at about half volume all the time because there are other things that play on it. (This issue is not flexible, nor is changing the amp) The problem is that now, even when I mute out the computer, you hear some static in the background. I was wondering some about some solutions (preferably multiple).

    Read the article

  • Output Audio via HDMI and Analog Out Simultaneously

    - by Alex Miller
    I need to output the system audio from a desktop to both the HDMI output (sending to the display that functions as the room speakers) and via the analog stereo output (to feed the reference input on a microphone array for Skype). The only solution I've found so far for this is an HDMI Audio De-Embedder, but I'd really like to avoid buying another piece of hardware, so I'm hoping there's a way to do this inside software. So: Is there any way to make Windows 7 output all audio over both HDMI and Analog outs, simultaneously?

    Read the article

  • To make use of 5.1 sound do i need a 5.1 headphone/speaker?

    - by Fellknight
    I recently bought a Toshiba X500 laptop with decent speakers & sound card in it. I want to use headphones but im unsure if they also need to have the same audio channels as the sound card (5.1 in this case). Also The laptop itself only has the two standard sound ports - headphones and microphone, does 5.1 headphones require a special kind of connection?.

    Read the article

  • 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.

    Read the article

  • Sending output to both bluetooth headset and normal speakers - Windows 7

    - by Christian Mann
    Hey, I have a bluetooth "headset" (it's more like a speaker with a microphone on it, but it registers as a headset) and I want to play music through it. I also want to play the same music through the "normal" speakers on the laptop. Is this possible? If so, is it possible to play two different streams on each speaker? Say if I wanted to DJ a party or something, I'd want to hear the upcoming song and mix it before sending it live.

    Read the article

< Previous Page | 4 5 6 7 8 9 10 11 12 13 14  | Next Page >