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  • Terrible noises from subwoofer of ACER Aspire 6930 with Realtek sound chip

    - by OneWorld
    After approximately 5-15 min of listening to music my subwoofer begins to make terrible noises. He's just "coughing". That began after 6 months I had this computer. Now I found out, that I can temporarily fix this problem by "restarting" the audio stream of the application that plays music. For example reloading last.fm page (reloads the flash file). Another way to reset the audio playback is switching the speaker configuration shown below in the screenshot. According to many posts on the internet like http://www.tomshardware.co.uk/forum/52918-20-acer-aspire-6935g-speaker-problem ACER support isn't any help Exchanging hardware doesn't fix the problem Even the later models have this problem Turning off the volume of the subwoofer is not an option to me. I still have warranty (I bought an extension of one year). I already tried about 15 versions of the Realtek driver with no success. I am not sure but MAYBE the problem did not occur on the original windows vista that was shipped with this computer. However, I removed the original windows for good reasons (english). What do you suggest me? Did anyone fix this problem? Maybe by writing a script which resets the audio streams every 5 minutes? Shall I take the effort to deal with the acer support until they give me another model? (I won't have a computer than for a longer time, will spend money on telephone hotlines (1,30 EUR / min)......) Here are additional infos, if they are any help: Windows 7 64 Bit (Original was Windows Vista Home Premium 32 Bit) All specs Audio driver version:

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  • How do I activate the F_LINE input in a transplanted HP chassis?

    - by admin
    I have an HP Pavilion Media Center PC chassis, vintage 2003 or so and I replaced the motherboard in it with a newer (vintage 2009) HP motherboard, M2N68-LA (Narra 5). I have scoured the internet trying to find pinouts for the motherboard to no avail. My question concerns the front panel audio, specifically Line In. The old chassis was built for AC97 but the new mobo is build for the newer HD audio standard. I figured out by comparison & experimentally how to connect the Mic & Headphone jacks to the HD audio header of the mobo by adding a manual switch to set the SENSE lines. Now all works fine for Mic & headphone. The old chassis also has a front panel Line In jack that the newer HP chassis does not have. However, the new mobo has a 4 pin white connector labeled F_LINE that I believe is a line input. Under Windows 7 I see the two Line Inputs in the mixer but I can't get one of them to become active. The 4 pin F_LINE connector uses the two middle pins for ground, and presumably the other two for left and right audio inputs. There are no pins for sensing on that connector. Can anyone tell me how to use that F_LINE input for the front panel, or how to activate it?

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  • Sound doesn't work anymore after replacing RAM

    - by thejh
    Hello, today, I replaced one old RAM module with two newer, bigger ones, but now, the sound doesn't seem to work anymore. Already ran alsaconf and it didn't help. Output of lspci for the audio device: 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f5100000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel The audio device is onboard and has six configurable outputs, two or so are also capable of being an input (if I remember it correctly), but I don't know how to control it under linux. Does somebody know how/whether replacing the RAM could be related to my problem and/or how to fix it?

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  • How to boost playback volume in real time on media recorded with a very low volume.

    - by L Marksman
    I have never heard a satisfactory answer to this often misunderstood question, let me explain. Lets say I have a sound card and earphones/speakers that can play back audio loud enough in most cases. This is great but the problem is that you always find people who do not know how to record audio, from Youtube video's to music. So now you end up with a audio playback that only uses 10% or less of the capacity of your sound hardware, in vista/win 7 you will see this frequently in the mixer with the volume pushed up to max but the green sound level only goes up a millimeter or two. I am looking for (preferably free) software or a method to boost the sound level of any audio from any source in real time to use more of my hardware capacity similar to what VLC media player can do. Oh and please, do not tell me it is impossible. I am not trying to boost the volume past what my hardware is capable of, I am just trying to use my hardware's full capacity. Also please do not tell met to buy new hardware, I know I can use hardware amplification, I don't want to (like many others) spend money on a simple little problem like this. Thanks!

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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  • Oracle User Productivity Kit Translation

    - by ultan o'broin
    Oracle's customers just love the User Productivity Kit (UPK). I hear only great things about it from our international customers at the Oracle Usability Advisory Board meetings too. The UPK is the perfect solution for enterprise applications training needs (I previously reviewed a fine book about UPK btw). One question I am often asked is how source content created using the UPK can be translated into another language. I spoke with Peter Maravelias, Principal Product Strategy Manager for UPK about this recently. UPK is already optimized for easy source-target translation already. There is even a solution for re-recording demos. Here's what you can do to get your source content into another language: Use UPK's ability to automatically translate events and actions. UPK comes with XML templates that allow you to accomplish this in 21 languages with a simple publishing action switch. These templates even deal with the tricky business of using gender-based translations. Spanish localization template sample Japanese localization template sample Use the Import and Export localization features to export additional custom content in a format like XLIFF, easily handled by translation tools. You could also export and import in Word format. Re-record the sound (audio) files that go with the recordings, one per screen. UPK's granular approach to the sound files means that timing isn't an option. Retiming demos isn't required. A tip here with sound files and XLFF-exported custom content is to facilitate translation context by avoiding explicit references to actions going on in the screen recordings. A text based storyboard with screenshots accompanying the sound files should also be provided to the translators. Provide a glossary of terms too. Use the re-record option in UPK to record any demo from a translated application. This will allow all the translated UI labels to be automatically captured. You may be required to resize any action events here due to text expansion issues. Of course, you will need translated data in the translated application too, so plan for this in advance. However, source-target language skills aren't required for the re-recording. The UPK Player itself, of course, is also available from Oracle along with content and doc in 21 languages. The Developer and Setup is also translated in a smaller number of languages. Check the Oracle UPK website for latest details. UPK is a super solution for global enterprise applications training deployments allowing source content to be translated into multiple languages easily. See this post on the UPK blog for more insight too!

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  • Beat detection and FFT

    - by Quincy
    So I am working on a platformer game which includes music with beat detection. I am currently using a simple if the energy that is stored in the history buffer is smaller then the current energy there is a beat. The problem with this is that ofcourse if you use songs like rock songs where you have a pretty steady amplitude this isn't going to work. So I looked further and found algorithms splitting the sound into multiple bands using FFT. I then found this : http://en.literateprograms.org/Cooley-Tukey_FFT_algorithm_(C) The only problem I'm having is that I am quite new to audio and I have no idea how to use that to split the signal up into multiple signals. So my question is : How do you use a FFT to split a signal into multiple bands ? Also for the guys interested, this is my algorithm in c# : // C = threshold, N = size of history buffer / 1024 public void PlaceBeatMarkers(float C, int N) { List<float> instantEnergyList = new List<float>(); short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; // Calculate instant energy for every 1024 samples. while (sampleIndex + nextSamples < samples.Length) { float instantEnergy = 0; for (int i = 0; i < nextSamples; i++) { instantEnergy += Math.Abs((float)samples[sampleIndex + i]); } instantEnergy /= nextSamples; instantEnergyList.Add(instantEnergy); if(sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; } int index = N; int numInBuffer = index; float historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } }

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  • Android: Voice Recording and saving audio

    - by user1320912
    I am working on application that will record the voice of the user and save the file on the SD card and then allow the user to listen to the audio again. I am able to allow the user to record his voice using the RecognizerIntent, but I cant figure out how to save the audio file and allow the user to hear the audio. I would appreciate it if someone could help me out. I have displayed my code below: // Setting up the onClickListener for Audio Button attachVoice = (Button) findViewById(R.id.AttachVoice_questionandanswer); attachVoice.setOnClickListener(new OnClickListener() { public void onClick(View v) { Intent voiceIntent = new Intent(RecognizerIntent.ACTION_RECOGNIZE_SPEECH); voiceIntent.putExtra(RecognizerIntent.EXTRA_LANGUAGE_MODEL, RecognizerIntent.LANGUAGE_MODEL_FREE_FORM); voiceIntent.putExtra(RecognizerIntent.EXTRA_PROMPT, "Please Speak"); startActivityForResult(voiceIntent, VOICE_REQUEST); } }); protected void onActivityResult(int requestCode, int resultCode, Intent data) { if(requestCode == VOICE_REQUEST && resultCode == RESULT_OK){ }

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  • Ubuntu 12.04 - No sound - HELP!

    - by Bruno Tacca
    I'm panicking... my sound stopped working after I tried to set-up my notebook speakers, plus two headphone jacks... My idea was to multichannel the sound to 3 channels, built-in speakers, and sound-card 2 headphone jacks. After a couple efforts I did it with 2 channels, speakers and 1 headphone jack, but the other wasn't working. After more tries and tries, sound stop working. I just want my sound back... crying like a baby on the floor. And, if possible, but not necessary, a simple guide to active the 3 channels. xD I will post the diagnosis according to https://help.ubuntu.com/community/SoundTroubleshootingProcedure STEP 1 Did it, still no sound. STEP 2 Did it, still no sound. STEP 3 and #STEP 4 (I removed the log cause there is a limit of characters to be posted.) The log can be found here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/238653 STEP 5 Rebooted, still no sound. STEP 6 Did it. In the Output Devices tab, nothing is muted. I play a music with the Rhythmbox Music Player, I don't hear anything but in the pavucontrol I can see in the Built-in Audio Analog Stereo a sound bar shaking... but, no sound. STEP 7 In alsamixer, AlsaMixer v1.0.25 Card: HDA Intel PCH Chip: Creative CA0132 information View: F3:[Playback] F4: Capture F5: All Item: Headphone [dB gain: 25.00, 25.00] Then, I have 5 columns Headphone, Speaker, PCM, S/PDIF, S/PDIF Default PCM A little weird when I try to mute the Headphone and the Speaker, here what happens: Starting both unmutted, mutting headphone cause speaker being mutted automaticaly. Starting both unmutted, mutting speaker cause headphone being mutted automaticaly. Starting both mutted, possible to unmute both separately. STEP 8 I cannot hear sound on both (headphone and/or speaker). STEP 9 Dual boot... Restarted, windows was with sound at max volume. Restarted again, still no sound at ubuntu. I heard something when ubuntu started, a little noise, then silence again. The sound icon always start mutted, after unmutting, I have no sound. STEP 10 I dont have this command in my ubuntu. STEP 11 Tried at STEP 8, no sound. There are no problem with jumpers or hardware, cause I have sound working on windows. STEP 12 No way to open my alienware and loss the warranty x.X" STEP 13 I think it's loaded, judging my the logs STEP 14 Alienware M17xR4, the hardware is listed in the logs above, at STEP 4. There are two headphone hacks, one with just an headphone printed above, and the other with an headset (with mic) printed, there is a mic jack too, and a spdif (optical) too. STEP 15 I dont want to enable S/PDIF STEP 16 I never used the HDMI output, yet... Thanks in advance. I hope I listed all the information you need.

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  • Testing background audio in the simulator

    - by Cactuar
    I'm experimenting with the new background audio service in iPhone OS 4.0 but I can't get it to work in the simulator. According to this page: iPhone Application Programming Guide: Executing Code in the Background it seems that all I have to do is add the a UIBackgroundModes key with an array containing audio to my Info.plist file and the audio my application plays should automatically continue when I switch to another app. I have done this but the audio still pauses as I switch to another app, when I switch back it continues where it left off. This is the code I'm using to play the sound: NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/audio.mp3", [[NSBundle mainBundle] resourcePath]]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; audioPlayer.numberOfLoops = -1; if (audioPlayer == nil) NSLog(@"%@", [error userInfo]); else [audioPlayer play]; Has anyone gotten this to work? Could it be that it would work on an actual device and it's just a problem with the simulator? I'm a bit hesitant to install 4.0 on my phone since I've heard it's still very buggy. Wish I had another device to use only for development.

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  • Custom flash mp3 player stopping in the middle of playing audio on windows nt ie6 system

    - by Charlotte Moller
    We have used a custom MP3 flash player for a lot of years on our website without any issues, but recently, a client of ours is reporting that the audio is playing for several seconds and then stopping. When they refresh the page or click play in the player again the audio plays fine. We are puzzled as to what could be causing this issue after this running successfully for our clients for so many years. The client system is Windows NT running IE6. Does anyone have any idea what could cause the audio to behave this way? Could audio drivers or the version of flash cause problems? We do not have flash programmers on our team so we are not even sure where to start looking within the flash code of the player. Any ideas?

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  • Capture Flash Audio in 4.7 Edge?

    - by emcmanus
    Is there a way to capture plugin (Flash) audio before it gets to the sound card? I'd like to record plugin audio, hopefully without actually playing the sound. Capturing audio at the device level is an absolute last resort, as the application would pick up all system audio rather than just the Webkit plugin. I'm aware of the recent switch back from QTMultimedia; is this possible with phonon? I spent the night looking for some way to access the phonon graph via QWebFrame (or any of the QtWebkit widgets) -- and didn't turn up much. I also started digging through QTWebkit, particularly NPAPI, without success. For reference, I'm using the edge version of 4.7 (6aa50af000f85cc4497749fcf0860c8ed244a60e) This seems to be a fairly challenging problem. Any hints would be greatly appreciated.

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  • Making a DVD video with a still image and PCM 16bit audio with ffmpeg

    - by João
    I'm trying to make a small video with a still image and a sound file playing in the background to pass it to dvdauthor and create a DVD. The command I'm using is this: ffmpeg -loop_input -i image.jpg -qscale 2 -i song.flac -aspect 4:3 -target pal-dvd -acodec pcm_s16le -shortest output.mpg However, the resulting video file doesn't have sound at all (testing it on VLC Player). I don't know if I can't combine "-acodec pcm_s16le" with "-target pal-dvd" to override the later, or if there is something else wrong with the command. If I try without the "-acodec pcm_s16le" parameter the video and audio works, I can even create a DVD ISO with it. However, the audio stays as AC3. I wanted to include with the video the lossless audio, not a compressed one. I suppose the DVD standart allows to have PCM audio in it, am I right?

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  • How to get the default audio format of a TTS Engine

    - by Itslava
    In Microsoft TTS 5.1 or newer. The SpVoice.AudioOutputStream property says: The AudioOutputStream property gets and sets the current audio stream object used by the voice. Setting the voice's AudioOutputStream property may cause its audio output format to be automatically changed to match the text-to-speech (TTS) engine's preferred audio output format. If the voice's AllowAudioOutputFormatChangesOnNextSet property is True, the format change takes place; if False, the format remains unchanged. In order to set the AudioOutputStream property of a voice to a specific format, its AllowOutputFormatChangesOnNextSet should be False. It means a engine's always has a preferred audio output format. So, how can i get it.. i have not found any interface to get that attribute.

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  • Access MP3 audio data independently of ID3 tags?

    - by kyl191
    Hi, this is a 2 part question. First off, is it possible to access the audio data in an MP3 independently of the ID3 tags, and secondly, is there any way to do so using available libraries? I recently consolidated my music collection from 3 computers and ended up with songs which had changed ID3 tags, but the audio data itself was unmodified. Running a search for duplicate files failed because the file changed with the ID3 tag change, but I think it should be possible to identify duplicate files if I just run a deduplication using the audio data for comparison. I know that it's possible to seek to a particular position past the ID3 header in the file, and directly read the data, but was wondering if there's a library that would expose the audio data so I could just extract the data, run a checksum on it, and store the computed result somewhere, then look for identical checksums. (Also, I'd probably have to use some kind of library when you take into account variable length headers.)

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  • h264 inside FLV container vs. MP4 container?

    - by Gotys
    I am developing a tube site, and currently having issues with h264 format . By looking at youtube, I noticed they are putting their hi-def videos into mp4 container, so logically I did the same. Next, I installed mod_h264_streaming for lighttpd to make streaming and timeline-scrubbing work. Problem is, that large files (500mb+ at somewhat high resolution) take for EVER to even start buffering ( I read the flowplayer or other flash players need to download metadata first) . I moved the xmov atom to the front of the file with MP4Box (i tried qt-quickstart too) , and the problem didn't go away. Next I read online I need to interleave audio tracks, so I did that too. No change in slowness. So I tried putting the same exact h264 movie into an FLV container, and the playback buffering starts almost instantly - no slowness. So what am I missing here? Why would I choose MP4 container with mod_264_streaming module , which seems super-slow over a regular FLV container with lighttpd's built-in mod_flv_streaming ? Obviously many websites pick mp4 container , but I fail to understand why ? And as a side question - I tried using HTML5's VIDEO tag to try the same h264 MP4 movie, and the scrubbing is LIGHTING FAST! I looked into lighttpd's log file, and i noticed taht Flash Players append video.mp4?start=234 each time timeline is scrubbed, wheres HTML5's video tag does no such thing . Is this some sort of limitations of Flash ? Why Can't flash streaming be same fast as HTML5 streaming? Thanks to ALL who can help. I very much appreciate this community.

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  • h264 inside FLV container vs. MP4 container?

    - by Gotys
    I am developing a tube site, and currently having issues with h264 format . By looking at youtube, I noticed they are putting their hi-def videos into mp4 container, so logically I did the same. Next, I installed mod_h264_streaming for lighttpd to make streaming and timeline-scrubbing work. Problem is, that large files (500mb+ at somewhat high resolution) take for EVER to even start buffering ( I read the flowplayer or other flash players need to download metadata first) . I moved the xmov atom to the front of the file with MP4Box (i tried qt-quickstart too) , and the problem didn't go away. Next I read online I need to interleave audio tracks, so I did that too. No change in slowness. So I tried putting the same exact h264 movie into an FLV container, and the playback buffering starts almost instantly - no slowness. So what am I missing here? Why would I choose MP4 container with mod_264_streaming module , which seems super-slow over a regular FLV container with lighttpd's built-in mod_flv_streaming ? Obviously many websites pick mp4 container , but I fail to understand why ? And as a side question - I tried using HTML5's VIDEO tag to try the same h264 MP4 movie, and the scrubbing is LIGHTING FAST! I looked into lighttpd's log file, and i noticed taht Flash Players append video.mp4?start=234 each time timeline is scrubbed, wheres HTML5's video tag does no such thing . Is this some sort of limitations of Flash ? Why Can't flash streaming be same fast as HTML5 streaming? Thanks to ALL who can help. I very much appreciate this community.

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  • Live noise-filter on line-in

    - by Damon Gant
    I'm running the following setup: Xbox 360 is hooked up to my (PC) screen via HDMI/DVI converter. Because the Xbox has no dedicated sound output, except for optical S/PIDF, I'm also using the AV/RCA output, namely just the audio, which is connected to an old stereo, which is then connected to my PCs line-in. I'm now experiencing a some of noise. I'm using one of the standard "Realtek High Definition Audio" cards, which doesn't seem to offer this kind of functionality. Is there a software that will playback audio right off a device while running filters on it? It doesn't have to create a device on its own, I just want to listen to it. Here's a sample: http://puu.sh/1suY6

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  • connect 2.1 stereo speakers to LG LCD-TV (5500 series)

    - by rMaero
    I bought a pair of speakers for my dad's TV, LG 32LE5500. When I installed them, it just sounded worse than the integrated ones and that's where I realized the subwoofer didn't work at all and both speakers make lower volume than the internal ones. The audio output jack says "H/P" (standing for headphones, and a matching symbol) before buying I checked this output with my phone's headphones and it worked so I figured it would work with a set of speakers since it's a standard audio output. I guess it's literally for headphones and not any other kind of sound players. There is only one other audio output and it is the optical-digital, so I can't use that. Not at least with these speakers.. am I screwed? or is there any workaround?

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  • Squid Authentication & streaming

    - by Steve Butler
    I've got squid setup using Kerberos authentication. I'm also using squidguard as an URL redirector to block out the usual nastiness of the web. There are some sites though that we allow certain users to, and others not. This all works well, assuming I'm not using any streaming. From what i can determine from the squid logs and the wireshark traces I've done, when the initial request to stream is sent, everything is good, the authenticated username is sent with the request to squidguard. The problem is that on subsequent traffic the username is not sent to squidguard, causing it to be blocked based on default policy. I've tried using the squid built-in allow/deny stuff, but its relatively clunky, and so far squidguard has been pretty easy and fast. Here comes the question(s): How do i get Squid to pass username on all requests? (something tells me this isn't the best way) How do i get squidguard to see traffic is authenticated to a specific user even when a username isn't passed? Is there any other way of accomplishing this? A few details that may be of importance: I'm using a list of users stored in a text file for squidguard to compare against. I'm using full kerberos auth with Squid. CentOS 6.0 Squid 3.1.4 Squidguard 1.3

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  • VIDEO Streaming - How to output the video timestamp?

    - by Emmanuel Brunet
    I would like to backup an ASF video stream with the time stamp (I mean the original recording date/time) on the output stream ? Usage I convert / store my video using the mkv format (Matroska) with libx264 (video) and aac (audio) codecs. Assume the IP camera webcam user account is account, the password password $ ffmpeg -i http://admin:alpha1237@webcam/videostream.asf -c:v libx264 -s 768X432 -crf 13 -b:v 2500K -pix_fmt yuv420p -c:a libfdk_aac output.mkv This works fine on a tenvis JPT3815W camera How to I need to get the video timestamp available for display as a subtitle or other meta data field managed by standard video players, and ideally to be able to hide it or not during video reading. Does anybody knows how to achieve that ? Thanks in advance for your help.

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  • Share a USB sound card over a network/bluetooth (Mac & PC)

    - by AlexW
    I've been wondering how I can stream audio to an external Edirol USB sound card, wirelessly, on both Mac and PC. I'm not looking for high quality transmission, just to play mp3s from my Mac laptop to a USB sound card that is attached to two very nice balanced studio reference monitors. Is there any way I can firstly power the sound card box, and secondly, provide with an audio stream along it's USB input. I've looked at the Belkin USB hub, and I have a Time Capsule with the AirPort interface inside. These things seem to do vaguely what I want but when it comes to audio, the specifications are less clear. Any suggestions very welcome.

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