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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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• fmod getWaveData() export to WAVE file help (C++)

  - by eddietree

  I am trying to export the current sound that is being played by the FMOD::System into a WAVE file by calling getWaveData(). I have the header of the wave file correct, and currently trying to write to the wave file each frame like so: const unsigned int samplesPerSec = 48000; const unsigned int fps = 60; const int numSamples = samplesPerSec / fps; float data[2][numSamples]; short conversion[numSamples*2]; m_fmodsys->getWaveData( &data[0][0], numSamples, 0 ); // left channel m_fmodsys->getWaveData( &data[1][0], numSamples, 1 ); // right channel int littleEndian = IsLittleEndian(); for ( int i = 0; i < numSamples; ++i ) { // left channel float coeff_left = data[0][i]; short val_left = (short)(coeff_left * 0x7FFF); // right channel float coeff_right = data[1][i]; short val_right = (short)(coeff_right * 0x7FFF); // handle endianness if ( !littleEndian ) { val_left = ((val_left & 0xff) << 8) | (val_left >> 8); val_right = ((val_right & 0xff) << 8) | (val_right >> 8); } conversion[i*2+0] = val_left; conversion[i*2+1] = val_right; } fwrite((void*)&conversion[0], sizeof(conversion[0]), numSamples*2, m_fh); m_dataLength += sizeof(conversion); Currently, the timing of the sound is correct, but the sample seems clipped way harshly. More specifically, I am outputting four beats in time. When I playback the wave-file, the beats timing is correct but it just sounds way fuzzy and clipped. Am I doing something wrong with my calculation? I am exporting in 16-bits, two channels. Thanks in advance! :) Reference (WAVE file format): http://www.sonicspot.com/guide/wavefiles.html

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• AAC 256kbit to MP3 320kbit conversion. I know it's lossy, but how?

  - by Fabian Zeindl

  Has anyone ever transcoded music from a high-quality aac to an mp3 (or vice-versa). The internet is full of people who say this should never be done, but apart from the theoretical standpoint that you can only lose information, does it matter in practise? is the difference perceivable, except on studio-equipment? does the re-encoding actually lose much information? If, p.e., high frequences are chopped away by the initial compression, those frequencies aren't there anymore, so this part of the compression-algorithm won't touch the data during the second compression. Am i wrong?

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• Convert MySQL to an MS SQL Server 2008 Database

  Converting a MySQL database to an MS SQL Server 2 8 database is a bit tricky. It is however an important database migration conversion. Is there some way to do it without resorting to costly database conversion software or facing issues with ODBC connectivity This article will teach you a new method to help you accomplish this conversion.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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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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• Android multimedia frameworks- How does audioflinger manage the connected audio devices?

  - by Tina

  what is the actuall detailed control flow while playing or recording an audio? and what is the role of audioflinger component here..

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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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• Problem with playing audio from SMB-shares using jQuery or Flash plugin.

  - by Kamerer

  When I open page with player directly and play music from my computer or netbios-share, it's OK. But when I open exactly same page through local server, I can't play audio from SMB-shares.

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• Advice: Python Framework Server/Worker Queue management (not Website)

  - by Muppet Geoff

  I am looking for some advice/opinions of which Python Framework to use in an implementation of multiple 'Worker' PCs co-ordinated from a central Queue Manager. For completeness, the 'Worker' PCs will be running Audio Conversion routines (which I do not need advice on, and have standalone code that works). The Audio conversion takes a long time, and I need to co-ordinate an arbitrary number of the 'Workers' from a central location, handing them conversion tasks (such as where to get the source files, or where to ask for the job configuration) with them reporting back some additional info, such as the runtime of the converted audio etc. At present, I have a script that makes a webservice call to get the 'configuration' for a conversion task, based on source files located on the worker already (we manually copy the source files to the worker, and that triggers a conversion routine). I want to change this, so that we can distribute conversion tasks ("Oy you, process this: xxx") based on availability, and in an ideal world, based on pending tasks too. There is a chance that Workers can go offline mid-conversion (but this is not likely). All the workers are Windows based, the co-ordinator can be WIndows or Linux. I have (in my initial searches) come across the following - and I know that some are cross-dependent: Celery (with RabbitMQ) Twisted Django Using a framework, rather than home-brewing, seems to make more sense to me right now. I have a limited timeframe in which to develop this functional extension. An additional consideration would be using a Framework that is compatible with PyQT/PySide so that I can write a simple UI to display Queue status etc. I appreciate that the specifics above are a little vague, and I hope that someone can offer me a pointer or two. Again: I am looking for general advice on which Python framework to investigate further, for developing a Server/Worker 'Queue management' solution, for non-web activities (this is why DJango didn't seem the right fit).

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• Stuck in a loop

  - by Luke

  while (true) { //read in the file StreamReader convert = new StreamReader("../../convert.txt"); //define variables string line = convert.ReadLine(); double conversion; int numberIn; double conversionFactor; //ask for the conversion information Console.WriteLine("Enter the conversion in the form (Amount, Convert from, Convert to)"); String inputMeasurement = Console.ReadLine(); string[] inputMeasurementArray = inputMeasurement.Split(','); //loop through the lines looking for a match while (line != null) { string[] fileMeasurementArray = line.Split(','); if (fileMeasurementArray[0] == inputMeasurementArray[1]) { if (fileMeasurementArray[1] == inputMeasurementArray[2]) { Console.WriteLine("The conversion factor for {0} to {1} is {2}", inputMeasurementArray[1], inputMeasurementArray[2], fileMeasurementArray[2]); //convert to int numberIn = Convert.ToInt32(inputMeasurementArray[0]); conversionFactor = Convert.ToDouble(fileMeasurementArray[2]); conversion = (numberIn * conversionFactor); Console.WriteLine("{0} {1} is {2} {3} \n", inputMeasurementArray[0], inputMeasurementArray[1], conversion, inputMeasurementArray[2]); break; } } else { Console.WriteLine("Please enter two valid conversion types \n"); break; } line = convert.ReadLine(); } } The file consists of the following: ounce,gram,28.0 pound,ounce,16.0 pound,kilogram,0.454 pint,litre,0.568 inch,centimetre,2.5 mile,inch,63360.0 The user will input something like 6,ounce,gram The idea is that it finds the correct line by checking if the first and second words in the file are the same as the second and third the user enters. The problem is that if it checks the first line and it fails the if statement, if goes through to the else statement and stops. I am trying to find a way where it will stop after the it finds the correct line but not until. If someone types in a value that isn't in the file, then it should show an error.

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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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• 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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• Linux application that bundles multiple incoming audio and video streams into one container file?

  - by StackedCrooked

  I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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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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• 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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• how to use usb speaker and built in audio speakers simultaneously on MAC OS X 10.6 using cocoa or an

  - by Saurabh

  Hi As title says i want to use built in speakers and usb speakers simultaneously on MAC OS X 10.6. I want to play different files on these (I do not want to play same file on both speakers). Is is possible using cocoa or anything else?? Any help would be appreciated! Thanks

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• how to stream audio and video files, but use any media player on Windows (without using Windows file

  - by RamyenHead

  I want to access and play media files on machine S (Windows XP) from machine C (Windows XP). Using Windows File Sharing ("share this folder" stuff), if it works, I would share the folder containing media files on machine S, and I would be able to play media files, sitting in front of C, using any media player I want. Windows somehow ensures that the remote files behave like local files. But Windows file sharing won't work for me, is there any alternative? If two machines were both Linux, I would install an SSH server on S and use Nautilus from C to access and play media files. The reason why I can't use Windows file sharing is, my campus use two different subnets, I have S and C on different subnets and it seems that the firewall governing the whole network in campus doesn't allow file sharing between different subnets. I tried changing Windows Firewall settings on S to allow C in, it still wouldn't work, so it must be the other firewall.

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• How to split audio into multiple channels from optical S/PDIF or 1/8"?

  - by Josh M.

  I have a motherboard which has an optical S/PDIF output or 1/8". I'd like to "split" that signal into the appropriate channels so that I can then connect that to the wires behind my car's headunit which, in turn, run to the amp. The factory Bose amp just takes a single connector with a million wires running out of it, so that's why I would need to separate the signal into separate channels. On the other end there are four RCA connectors: front left, front right, rear left, rear right. The sub-woofer signal does not require an additional connection. Edit: Revised to include S/PDIF or 1/8".

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• Quality wise, is Windows Media Audio 10 Professional equivalent to WMA?

  - by Louis

  I noticed that for encoding CD rips, Zune is still using WMA 9.2 instead of WMA 10 Pro. On a given file using the highest quality VBR settings looks like this: VBR Quality 98, 44 kHz, stereo 1-pass VBR On the same file if I use WMA 10 Pro, with the same settings, the resulting file is about 20% smaller. Using my ears, I'm unable to tell the difference, but I'm wondering if this was the goal of WMA 10 Pro (to be as good as WMA at a lower bitrate). Is the quality of a WMA 10 Pro file equal to that of a WMA 9.2 file encoded with the same settings?

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• Hardware Mediaplayer display

  - by Eric Audio

  I'm looking for a keyboard or just a little display to attach on my keyboard or something like that, what will show me the music tracks i'm playing in windowsmedia player, itunes, etc. I did some research and the only thing I found are gaming keyboards, but i'm not shure if these show my music tracks. So my question: Does somebody knows a keyboard who show the music tracks or just a little display? Bye, Eric

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• How can I get Windows 7 to switch audio from a monitor (with built-in speakers) to headphones when t

  - by tnorthcutt

  I have an HP dv5t laptop running Windows 7 64 bit with an Acer H235H monitor connected to it via an HDMI cable. The monitor has built-in speakers, which are a huge improvement over the laptop's speakers. However, when I want to use headphones, right now, I have to connect them to the laptop, then right-click the sound icon in the task bar, select Playback Devices, right click the monitor, and disable it. Is there any way to get Windows 7 to automatically switch the output to the headphones when they're plugged in? That's the behavior that happens without the monitor attached (i.e. it will switch from the laptop speakers to headphones when headphones are plugged in). I have the same issue with a Sony Vaio laptop running Windows 7 64-bit and an identical monitor, for reference.

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