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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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  • No apparent reason for high load average

    - by Oz.
    We have several web servers running on Amazon (ec2) c1.xlarge, over Amazon AMI. The servers are duplicates of each other, running the exact same hardware and software. Each server spec is: 7 GB of memory 20 EC2 Compute Units (8 virtual cores with 2.5 EC2 Compute Units each) 1690 GB of instance storage 64-bit platform I/O Performance: High API name: c1.xlarge A couple of weeks ago we have run a yum upgrade on one of the servers. Starting on this upgrade the upgraded server started showing a high load average. Needless to say, we did not update the other servers and we can not do so until we understand the reason for this behavior. The strange thing is that when we compare the servers using top or iostat, we can not find the reason for the high load. Note that we have moved traffic from the "problematic" server to the others, which have made the "problematic" server less crowded in terms of requests, and still his load is higher. Do you have any idea what could it be, or where else can we check? Many thanks for the help! Oz. # # proper server # w command # 00:42:26 up 2 days, 19:54, 2 users, load average: 0.41, 0.48, 0.49 USER TTY FROM LOGIN@ IDLE JCPU PCPU WHAT pts/1 82.80.137.29 00:28 14:05 0.01s 0.01s -bash pts/2 82.80.137.29 00:38 0.00s 0.02s 0.00s w # # proper server # iostat command # Linux 3.2.12-3.2.4.amzn1.x86_64 _x86_64_ (8 CPU) avg-cpu: %user %nice %system %iowait %steal %idle 9.03 0.02 4.26 0.17 0.13 86.39 Device: tps Blk_read/s Blk_wrtn/s Blk_read Blk_wrtn xvdap1 1.63 1.50 55.00 367236 13444008 xvdfp1 4.41 45.93 70.48 11227226 17228552 xvdfp2 2.61 2.01 59.81 491890 14620104 xvdfp3 8.16 14.47 94.23 3536522 23034376 xvdfp4 0.98 0.79 45.86 192818 11209784 # # problematic server # w command # 00:43:26 up 2 days, 21:52, 2 users, load average: 1.35, 1.10, 1.17 USER TTY FROM LOGIN@ IDLE JCPU PCPU WHAT pts/0 82.80.137.29 00:28 15:04 0.02s 0.02s -bash pts/1 82.80.137.29 00:38 0.00s 0.05s 0.00s w # # problematic server # iostat command # Linux 3.2.20-1.29.6.amzn1.x86_64 _x86_64_ (8 CPU) avg-cpu: %user %nice %system %iowait %steal %idle 7.97 0.04 3.43 0.19 0.07 88.30 Device: tps Blk_read/s Blk_wrtn/s Blk_read Blk_wrtn xvdap1 2.10 1.49 76.54 374660 19253592 xvdfp1 5.64 40.98 85.92 10308946 21612112 xvdfp2 3.97 4.32 93.18 1087090 23439488 xvdfp3 10.87 30.30 115.14 7622474 28961720 xvdfp4 1.12 0.28 65.54 71034 16487112

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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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  • High quality / high speed dvd reader for Mac Pro

    - by deadprogrammer
    I have a high end Mac pro, but one thing in is that I'm unhappy with is a DVD drive. It's a Hitachi GH41N. Apple calls it a "superdrive", but it's anything but. The damn thing makes an amazing amount of noise, and isn't too fast either. What I want is a state of the art, fast, quiet DVD reader, preferably not even a burner. What should I get?

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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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  • 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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  • 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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  • Audio problems with asus notebook with Bluetooth and usb devices in win 7

    - by QuickSilver
    My notebook is Asus P53E - core i5 Windows 7 installed Audio from PC speakers and headphone is distorted when i turn on bluetooth or some usb device plugged in. I belive this is a software issue. I tried updating my audio drivers but nothing help. Any help will be appreciated. Update: After a few days digging I found that this problem is causing by the asus sound enhancement application SonicFocus. The distortion stops while turning off sonic focus. Can anyone help me with a solution other than turning off SonicFocus

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  • Screenflick Audio option in MacBook Pro

    - by John
    When after I shut down my MacBook Pro by holding the power button for a few sec, (which I found is bad for the computer, so I will not do anymore) I found that my speaker doesn't play until I plug in and out earphone into the machine. When my speaker is not working like this, and when I am on a random webcam chatting site like chatroulette.com, they can hear the music playing on my iTunes when I choose Screenflick Audio option in the Mic setting. But when the Speaker is working back again, they don't hear the music playing even when I do Screenflick Audio mode. How can I make it work? Also, how do you make the chatting partner hear my music playing on my computer while I talk to them (not via my speaker, since it's bad quality).

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  • Windows audio service fails to automatically start after VirtualBox install

    - by humble_coder
    I'm having a completely nonsensical issue in Windows XP SP3. Basically my "Windows Audio" service no longer starts automatically. Despite being set to "Automatic" I have to manually go in and start it. This issue didn't start until the most recent update of VirtualBox, but I can't find anything on the forums related to this specific issue. I've tried reinstalling the RealTek drivers as well, in the event that that had something to do with it. Any assistance is most appreciated! EDIT 1: It is the host's audio that won't start. The update of Virtualbox was merely the "marker" of when these events started occurring. Given it's the only variable/change I'm assuming a correlation.

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