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  • Lenovo R61i - audio not working with windows 7

    - by ann o connor
    Has any one got a solution to this problem. I have installed windows 7 pro and installed audio (coexant) latest driver from the lenovo web site but it still does not work. I have seen many similar posts but no solution - just the usual - do you have the most recent driver.... etc. Is there any one that has solved this problem out there? (Driver details: Audio Driver 2.57 MB Normal Version 4.98.5.50)

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  • usb audio recording: clicking noice

    - by snot
    I'm having problems recording audio on windows using usb audio devices. The recorded material will have some kind of jitter/clicking noice. I tested three different usb microphones and three different recording applications. Every combination will have the same problem. Does some one know a solution for this problem?

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  • Streaming audio from microphone to network

    - by Janusz
    I have the following problem: I want to stream the audio I record with one machine to another machine in the same network. It seems that vlc is the best shot at the moment. I was able to stream a music file via vlc but streaming the audio from the microphone the same doesn't work. EDIT If I enable play locally the captured sound is played. Even streaming to another instance of VLC on the same machine doesn't work.

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  • Sending computer audio output to multiple devices

    - by barfoon
    I'm looking to take an audio signal from a computer (1/8"), or multiple computers if possible, and send its output to some sort of switch/audio box that would allow me to choose between different receiving devices - say between headphones and a receiver. I don't want to split the signal, and I don't want to send the signal to both at once, I want to be able to choose where the signal goes. Is there a box that will do this?

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  • Front panel audio replacement

    - by develroot
    I have experienced some problems with my headphones and it turned out the front 3.5mm audio jack is defective..and it doesn't make a good contact. I can't replace the jack because it's built-in. I am wondering if there are such things as "modular" Front panels to be inserted into the available external 3.5" bay and, of course, to support (natively, internal connector) Realtek HD Audio. (my case is Asus TA-K5, but I guess it doesn't make any difference)

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  • windows-7 hdmi audio

    - by YDG
    Hello, I am using windows 7 and here is my question : Is there a way for windows to swtich my audio output to HDMI automatically when a HDMI cable is plugged in ? Right now, to switch the audio output I need to go specify my default output to HDMI everytime I plug it in and set it back to speakers once I plug it out. Cheers !

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  • merging video and audio with custom panning

    - by cherouvim
    I have: a video which has mono audio inside a audio (mono) I'd like to merge those two to a single video file containing: video from #1 audio from #1 full left pan + audio #2 full right pan Is this possible in ffmpeg using 1 command? I've tried the following which almost does this but the video/audio gets out of sync: ffmpeg -i video.mp4 -filter_complex "amovie=audio.wav [r] ; [r] amerge" output.mp4 -y I've managed to do it with multiple commands: #1 create right panned audio ffmpeg -i audio.wav -ac 2 -vbr 5 audio-stereo.mp3 -y ffmpeg -i audio-stereo.mp3 -af pan=stereo:c1=c1 audio-right.mp3 -y #2 create left panned video ffmpeg -i video.mp4 -af pan=stereo:c0=c0 video-left.mp4 -y #3 merge the two ffmpeg -i video-left.mp4 -i audio-right.mp3 -filter_complex "amix=inputs=2" video-mixed.mp4 -y It does the job, but is it possible with 1 command?

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  • Timing issues with playback of the HTML5 Audio API

    - by pat
    I'm using the following code to try to play a sound clip with the HTML5 Audio API: HTMLAudioElement.prototype.playClip = function(startTime, stopTime) { this.stopTime = stopTime; this.currentTime = startTime; this.play(); $(this).bind('timeupdate', function(){ if (this.ended || this.currentTime >= stopTime) { this.pause(); $(this).unbind('timeupdate'); } }); } I utilize this new playClip method as follows. First I have a link with some data attributes: <a href=# data-stop=1.051 data-start=0.000>And then I was thinking,</a> And finally this bit of jQuery which runs on $(document).ready to hook up a click on the link with the playback: $('a').click(function(ev){ $('a').click(function(ev){ var start = $(this).data('start'), stop = $(this).data('stop'), audio = $('audio').get(0), $audio = $(audio); ev.preventDefault(); audio.playClip(start,stop); }) This approach seems to work, but there's a frustrating bug: sometimes, the playback of a given clip plays beyond the correct data-stop time. I suspect it could have something to do with the timing of the timeupdate event, but I'm no JS guru and I don't know how to begin debugging the problem. Here are a few clues I've gathered: The same behavior appears to come up in both FF and Chrome. The playback of a given clip actually seems to vary a bit -- if I play the same clip a couple times in a row, it may over-play a different amount of time on each playing. Is the problem here the inherent accuracy of the Audio API? My app needs milliseconds. Is there a problem with the way I'm using jQuery to bind and unbind the timeupdate event? I tried using the jQuery-less approach with addEventListener but I couldn't get it to work. Thanks in advance, I would really love to know what's going wrong.

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  • Win 7 Media Center - TV - No audio on HD channels

    - by Chuzein Part II
    I wonder if you can help. I have recently purchased a PCTV Nanostick DVD-T2 290e I am running Windows 7 Home Premium 64bit and I use Windows media center for almost everything. I run a media center machine in my lounge through my Sony AV Amp onto my HD TV. I bought the Nanostick to watch ‘Freeview HD’ programmes when watching the TV through the Media Center – I also have a Sony Freeview Plus system that the Mrs uses. I installed the Nanostick with relatively no issues and as described by many user reviews Media Center picked up the new USB tuner and scanned all the new channels and I was presented with all the HD channels. The issue I have is that there is no audio sound from the HD channels but perfect video. SD works fine. The software that comes with the Nanostick works fine – both sound and vision are perfect for the HD channels. As mentioned, I run my Media Centre lots. I have an extensive DVD and BD collection stored on a server and DVD and BD all play perfectly through Arcsoft Total Media Theatre 5 – with no issues on either SD or HD. My sound card is built into my motherboard and that sends the audio signal to my gfx card and that in turn passes it through to my Sony AV Amp that decodes the audio to be heard on my 5.1 set up. Does anyone have any ideas. I have searched lots on the net and I cant find anyone else with the same issue. I am aware of codecs etc but I don’t really understand it. It also puzzles me that when I read the user/buyer reviews for this product so many people tell the story of faultless installations on the same kind of set up as me.

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  • iPhone PlayAndRecord silences all system audio??

    - by Eamon Ford
    Hi, In my iPhone app I am trying to record audio and play iPod music at the same time, so I set the audio session category to kAudioSessionCategory_PlayAndRecord. But when I set this, all system audio (including vibrate) doesn't work anymore, although the iPod audio still does work. Does anyone know if this is a bug in the SDK or something, or how to get around it? Please help! Thanks in advance!

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  • iPhone SDK: Audio Queue control

    - by codemercenary
    Hi all, I am new to the audio queue services so I have taken an example from a book called iPhone Cool Projects where it describes how to stream audio. I want to extend this to being able to play a continuous playlist of links to mp3 files like an internet radio. The problem with the example code it that it does not detect when a stream ends and does not call AudioQueueStop at any point, so I added a counter to number of buffers added to the queue, and then decrement this counter each time audioQueueOutputCallback is called by the queue. This works fine except if when the buffer count goes to 0, and then I add a call AudioQueueFlush(audioQueue) and then AudioQueueStop(audioQueue, false) I get an error. If I only call AudioQueueReset, it continues to load the buffers again, but plays them out faster then it loads them... getting stuck in a loop and then crashing. 2010-04-14 13:56:29.745 AudioPlayer[2269:207] init player with URL 2010-04-14 13:56:29.941 AudioPlayer[2269:207] did recieve data 2010-04-14 13:56:29.942 AudioPlayer[2269:207] audio request didReceiveData 2010-04-14 13:56:29.944 AudioPlayer[2269:207] >>> start audio queue 2010-04-14 13:56:29.960 AudioPlayer[2269:207] packetCallback count 2 2010-04-14 13:56:29.961 AudioPlayer[2269:207] add buffer: 1 2010-04-14 13:56:29.962 AudioPlayer[2269:207] did recieve data 2010-04-14 13:56:29.963 AudioPlayer[2269:207] audio request didReceiveData 2010-04-14 13:56:29.963 AudioPlayer[2269:207] packetCallback count 1 2010-04-14 13:56:29.964 AudioPlayer[2269:207] add buffer: 2 2010-04-14 13:56:29.965 AudioPlayer[2269:207] packetCallback count 13 2010-04-14 13:56:29.967 AudioPlayer[2269:207] add buffer: 3 2010-04-14 13:56:29.968 AudioPlayer[2269:207] done with buffer: 3 2010-04-14 13:56:29.969 AudioPlayer[2269:207] done with buffer: 2 2010-04-14 13:56:29.974 AudioPlayer[2269:207] done with buffer: 1 So this loop continues some 20 - 30 times and then it crashes. The first time it plays an audio file it queues up the buffers and then plays sound, but doesn't callback to delete them until some 100 or more have been played. Can anyone explain this behavior? I read that there was a limit of 1 audio queue for MP3 playback for the iPhone. Is that still true? If not then I suppose I should use another audio queue for the next mp3 stream. I've had a look through the apple docs but it doesn't explain this in any particular detail. A better insight into this would be great. TIA.

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  • using QTkit for recording audio

    - by RW
    It looks like using core audio to record audio is overly complicated. While QTkit is basic and down to earth However. All of the examples I have see integrate video and audio together. Does some one have or know an example of using QTkit for recording audio? rw

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  • How to process audio in real time?

    - by user1756648
    I am giving some audio input through microphone. I recorded it in Audacity, it looks something like as shown below. I want to process this audio in real time. I mainly want to do this. 1) see real time audio amplitude vs time graph 2) perform some actions based on some thing (like if a specific type of hike is seen in audio, then do something, else do something else) Is there any python module or C library that can allow me to do this ?

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  • Windows Vista/7: Managing multiple audio playback devices

    - by BrianLy
    I've got speakers (audio in) and headphones (USB headset with it's own soundcard) connected to my desktop computer. Under Windows 7, I can right-click the Audio Mixer and select Playback Devices and toggle between my these devices. Is there an easier way, perhaps a keyboard shortcut, that would make it easier to toggle? I'm working in an shared space were sometimes I want headphones to avoid annoying other people, but at other times speakers are OK. I want to be able to toggle quickly. In an ideal world, the solution to my question would work in Vista too.

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  • Change audio output depending on which one is on

    - by pkrish
    I have a PC that is hooked up to my HDTV (via a long hdmi cable) and to a monitor, which is in another room. I have speakers directly plugged to the PC audio out. The PC is next to the monitor and speakers. I am not sure but I think Windows can play sound on only 1 audio device at a time. And I can only set 1 device as the default output. I can get sound on the TV or speaker depending on which device I set to default. But I would have to do this every time I switch between using my TV or my monitor! Is there some way to configure such that sound plays through the TV if the TV is on, else it plays in the speakers? If this is not possible, then the next best alternative would be to get the sound to play on both the devices at the same time. Thanks!

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  • Audio input problem in Ubuntu 9.10

    - by Andrea Ambu
    My audio input is a mix of my mic output and my sound card output. I'd like it to be just my mic output. I was able to do so in Ubuntu 9.04 but the interface is 9.10 is totally changed and I tried every my creativity was able to think. It's really annoying when talking to other people over the internet because they keep hearing their voice back. I'm not sure I explained it in clear way so I'll give you an example: What I do: I put an mp3 on play or a video on youtube then open a recorder and start to talk on my mic. What happens: both my voice and audio from mp3/youtube get reordered, even if I put headphones volume to 0 (via hardware). What I'd like to happen: Only my voice should be recorded. I'm sure I'm missing some technical term, but that's the problem and I'd like to solve it in Ubuntu 9.10, any idea?

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  • 802.11n Audio Interference

    - by colithium
    Symptoms My audio is riddled with pops/crackles/glitches. Sometimes I swear the glitches sound exactly like the facebook messenger sound (best comparison I can give). Cause Using DPC Latency Checker, it reports the latency to be an abysmal 17,500µs (0.0175s). The first thing I did was disable my 802.11n wireless adapter. This immediately dropped the latency to a nice 250µs. When I re-enabled the adapter, it jumped right back up. I'm 99% certain that this is the cause of my audio glitches. Solution What can I do about it besides using wired Ethernet or buying a whole new adapter? My adapter is a Dell Wireless 1505 Draft 802.11n WLAN Mini-Card. To be honest, I've had nothing but trouble with the 802.11n standard and am contemplating just going back to g.

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  • Change Audio title from English to Sinhalese using ffmpeg

    - by user330461
    I insert an extra Sound track in my video file and it works well. ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 1 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 -an -y /dev/null && ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 2 -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 news.mp4 The default audio track come with the label "English" and I would like to give it a label "Sinhalese" The Second Audio track come up without a label as "track#1" and I would like to give that a label of "Tamil". How do I do that ?

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  • How to generate a 8 bit per sample wav audio file in VLC

    - by Ahmed safan
    I'm using the following vlc command line to extract first 5 minutes of audio from video file "-I dummy -vvv --no-sout-video --sout-audio --no-sout-rtp-sap --no-sout-standard-sap --ttl=1 --sout-transcode-threads=5 --sout-transcode-high-priority --sout-keep --sout #transcode{acodec=s16l,channels=1,samplerate=8000,ab=64}:std{mux=wav,access=file,dst="c:\dest.wav"} "c:\originalvideo.mpg" --start-time=0 --stop-time=300 vlc://quit"; if ab=64 =64 k bits per second and samples per second=8 k samples then bits per sample=64/8=8 bits per sample but the problem is that the output file always has samples of 16 bits per sample. I know that sample can contain bits from 8 , 16, 24 to 32 bits per sample. i want to get 8 bits per sample file how can this be done ?

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