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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 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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  • 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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  • 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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  • Can't find generic USB audio driver for a Samson COU1 USB microphone

    - by marcipollo
    I am unable to use a Samson USB CO1U microphone on a PC running XP, SP3. When I plug it into the USB port, Windows generates the sound indicating that it has found new hardware, and the green LED on the mic lights. But, it does not work, and the device manager reports that it cannot find a driver after searching. The same mic works on a Vista machine. Samson has no driver on their Web site, and insists that the generic audio driver in Windows should work. (http://www.samsontech.com/PRODUCTS/productpage.cfm?prodID=1810). I cannot find a generic USB audio driver at Microsoft.com. Can anyone help? Larry

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  • No HDMI audio - Windows 8 - ASUS H81M-PLUS

    - by Paul Wright
    I have an issue with HDMI audio on Windows 8 using an ASUS H81M-PLUS motherboard (without an external GFX card). There are many forum posts advising you to go into playback devices and setting HDMI to be default - I have done this. To eliminate what works and what doesn't work: I have not been able to get sound from my HDTV using HDMI. I have used this HDMI cable with my PS3, so this cable should be fine. I am able to use the HDMI cable in extended mode, so that I have two monitors (including the TV), just no audio. This HDMI cable goes straight from the motherboard to the TV. Below I have included 'Device manager', and 'Playback Devices' (Sound). Device Manager Playback Devices, showing disabled and disconnected devices I am at a loss. I have uninstalled all drivers, and then rebooted and made windows look for the correct ones, made sure the HDMI device was default. Thanks, Paul

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  • Can't find generic USB audio driver for a Samson COU1 USB microphone

    - by user10321
    I am unable to use a Samson USB CO1U microphone on a PC running XP, SP3. When I plug it into the USB port, Windows generates the sound indicating that it has found new hardware, and the green LED on the mic lights. But, it does not work, and the device manager reports that it cannot find a driver after searching. The same mic works on a Vista machine. Samson has no driver on their Web site, and insists that the generic audio driver in Windows should work. (http://www.samsontech.com/PRODUCTS/productpage.cfm?prodID=1810). I cannot find a generic USB audio driver at Microsoft.com. Can anyone help? Larry

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  • Tool to bulk speed up/convert an audio file

    - by User1
    I want to listen to certain podcasts on my phone but I have two common problems: The audio is in some weird format (some don't play on my phone). The audio is slow. I want to use something like sox or avconv to bulk convert the files. Since this is just voice and going on a cell phone, small low-quality files would be best for me. I had some good success using avconv: avconv -i weird.wma normal.ogg Unforunately, this command creates an enormous ogg file and I can't get it play faster. Ideally, this particular file would play at 170% of the original speed.

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  • Solutions for exporting a remote desktop app (display and audio)

    - by Richard
    I'm looking for a solution that will allow me to export a desktop app running on a server to a client machine. The server is ideally Linux, the desktop is Windows (+Mac for icing on the cake). The export should be encrypted and I need to support multiple clients from one server. I only want to export an individual app, not a whole desktop, and ideally am looking for open source solutions. The obvious, cheapest, simplest choice is to use X tunnelled over ssh (e.g using Xming on the desktop) but X doesn't support audio. What are the alternatives? Or is there a way to support audio using X or in parallel to X? Thanks

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  • FreePBX: Asterisk in the Cloud (EC2) Audio Problems

    - by neezer
    Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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