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  • Is it possible to capture audio output and apply effects to it?

    - by Ciaran
    Using .NET and DirectSound I want to be able to take all output sound that is coming from my audio device and apply effects to it. I've had a quick look at the docs on MSDN and there doesn't seem to be any explanation as to how to do something like this. I've read elsewhere that you'd be better off writing a driver to sit in front of your real audio driver and have that do whatever you want with the sound. Any ideas anyone to push me in the right direction?

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  • Speech recognition webservice that scores the accuracy of one audio clips vs. another?

    - by wgpubs
    Does such a thing exist? Building a Rails based web application where users can upload an audio file of them speaking that then needs to be compared to another audio file for the purposes of determining how similar to voices are. Ideally I'd like to simply get a response that gives me a score of how similar they are in terms of percentage (e.g. 75% similar etc...). Anyone have any ideas? Thanks

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  • How to create live stream audio for web-sites ???

    - by Kathir
    Hi All, We are storing sound from mic to pc via sound forge. We would like to broadcast the sound which comes from the mic to the pc as live streaming audio. Basically a person speaks in a mic, we like to give it as live stream audio. The web-site is hosted on yahoo server. Can you please let me know in what are the ways we can achieve this? Thanks, Kathir

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  • Bose USB audio: crackling popping sound, eventually die

    - by Richard Barrett
    I've been trying to troubleshoot this issue for a while now. Any help would be much appreciated. I'm having trouble getting my Bose "Companion 5 multimedia speakers" working with my installation of Ubuntu 12.04 (link to Bose product here: http://www.bose.com/controller?url=/shop_online/digital_music_systems/computer_speakers/companion_5/index.jsp ). The issue seems to be low level (not just Ubuntu). What happens: When I boot into Ubuntu, I can get Rhythm box to play ok. However, if I try anything else (an .avi file, a webpage, or Clementine player with mp3 files) I get crackling, popping, or choppy sounds. If I move the mouse around, especially if it seems graphic intensive, the problem gets worse (more crackling noises). The more taxing it appears to be, the more likely it is that the sound will just die altogether until I reboot. For some reason the videos at www.bloomberg.com seem especially bad for it (my sound normally goes dead in under 45 seconds and won't work until reboot). Both my desktop running Ubuntu 12.04 and my laptop (running the same) have the same crackling problem. Troubleshooting so far: A friend of mine who knows linux well tried to solve it for me without any luck. He took pulseaudio out of the equation, but still had the problem just using AlSA. Among the many things he tried was adjusting the latency, but that didn't help either. I've also tried things like adjusting the USB device settings in the config file from -2 to -1 so that it will use my USB sound and I also commented out the lines that would stop that. These don't do anything. (That really seems like it's for someone who is getting no sound at all, so it's not surprising this won't work.) My friend's laptop running his Archlinux could play my Bose USB speakers without any problems. I also tried setting my daemon.conf file to use 6 channels (based on this http://lotphelp.com/lotp/configure-ubuntu-51-surround-sound ) but that didn't work either. I recently used a DVD to boot into Ubuntu Studio 12.04 (because it uses a live audio kernel) and this happened: I got perfect sound for a minute or two When I started moving windows around while sound was playing, the sound died again. Perhaps more interesting: There is a headphone out jack on the Bose system. When I use it, the audio is perfect for all applications (even the deadly bloomberg.com videos with .avi playing at the same time and moving around windows). Also, there is an audio-in jack on the Bose system. I can use a male-to-male mini jack to go from my soundcard's output to the Bose input and then all sound works perfectly. -However, it still requires the Bose to be plugged in to USB, otherwise I lose all sound. Any thoughts? Any suggestions for trouble shooting? (Or any suggestions for somewhere else to post to solve this?) Any logs or other files I can provide to help someone help me work this out? Your help is much appreciated! Rick BTW: I sometimes get people posting responses like "My Bose USB system works great with Ubuntu 12.04," without any more details. Is there anything I should ask such people to narrow down my problem? (It's kind of annoying to hear such a response because it doesn't help solve my problem.)

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  • linux mint VIA sound issue

    - by user2699451
    So I installed linux Mint 15 "Olivia" 64 bit on my Mecer W550EU laptop I have HD Audio with a VIA chipset charles-W55xEU charles # lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_via 51018 1 snd_hda_intel 39619 5 snd_hda_codec 136453 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13602 1 snd_hda_codec snd_pcm 97451 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 1 snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 19 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_via,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd And my sound card charles-W55xEU charles # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1802 Analog [VT1802 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1802 HP [VT1802 HP] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 and my audio device charles-W55xEU charles # lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High `Definition Audio Controller (rev 04)` Subsystem: CLEVO/KAPOK Computer Device 0550 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at f7c10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Sometimes when I boot up, soundworks, other times it doenst, it is completely random, so far, no-one on xchat linux help or linux mint forums was able to help me, I have always had issues with sound on VIA chipsets I have: sudo apt-get upgrade && apt-get install mint-meta-cinnamon it seemed to help but after 2-3 reboots, the problem came back, btw, everytime I checked, pulse audio is selected to Duplex Audio Input & Output and alsa mixer is always unmuted!

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  • PC BluRay - Multichannel HD Audio output

    - by sheepsimulator
    When playing a BluRay movie on a PC (any OS, Mac/Win/Linux), I have some questions about audio output: When playing a BluRay disc on the PC using a BluRay player program, can it decode the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) and output the audio directly to the soundcard's 7.1 line-level analog outputs? Is it possible to bitstream the the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) over the HDMI output to a receiver when using a BluRay player program? I'd kinda like to know. I'm contemplating building a home theater PC, and the above functionality is important. I'd prefer that #1 is possible, actually, because it would mean I wouldn't have to buy a receiver.

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  • How to fix error 1330 (invalid digital signature) when installing "The Rosetta Stone"

    - by victoriah
    I bought an older version of this software from a friend and the Rosetta Stone support hasn't been of much use. When installing, at the end of the process I get an error from the installer that says something along the lines of: Error 1330. A file that is required cannot be installed because the cabinet file [file.cab] has an invalid digital signature. This isn't the first time I've had such an error, some time ago I was unable to install a game I bought because of the same thing. I extracted the cab file itself just to see what it was and it's just an archive of icons and things like that, so it's not even like I'm missing out on much :/ Any advice/tips on how to work around this error would be appreciated e.g. if it's possible to make the installer ignore digital signatures. Running Windows 7.

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  • RAID 1 Mirror Error with Two Western Digital 500GB Drives

    - by bm678
    I have Windows 7 Ultimate 64 bit installed on a Western Digital 500GB drive (WD5000BEVT-22ZAT0) that was partitioned automatically by Windows as 100MB System Reserved and 465.66GB drive C. There is also an unallocated second Western Digital 500GB drive (WD5000BPVT-22HXZT1) that I want to use for RAID 1 to mirror the first drive but I get an error message stating “ALL DISKS HOLDING EXTENTS FOR A GIVEN VOLUME MUST HAVE THE SAME SECTOR SIZE, AND THE SECTOR SIZE MUST BE VALID.” I uninstalled Windows patch KB-982018 but I still get the same error message. Could you please let me know how to resolve this? Thanks.

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  • Split UInt32 (audio frame) into two SInt16s (left and right)?

    - by morgancodes
    Total noob to anything lower-level than Java, diving into iPhone audio, and realing from all of the casting/pointers/raw memory access. I'm working with some example code wich reads a WAV file from disc and returns stereo samples as single UInt32 values. If I understand correctly, this is just a convenient way to return the 32 bits of memory required to create two 16 bit samples. Eventually this data gets written to a buffer, and an audio unit picks it up down the road. Even though the data is written in UInt32-sized chunks, it eventually is interpreted as pairs of 16-bit samples. What I need help with is splitting these UInt32 frames into left and right samples. I'm assuming I'll want to convert each UInt32 into an SInt16, since an audio sample is a signed value. It seems to me that for efficiency's sake, I ought to be able to simply point to the same blocks in memory, and avoid any copying. So, in pseudo-code, it would be something like this: UInt32 myStereoFrame = getFramefromFilePlayer; SInt16* leftChannel = getFirst16Bits(myStereoFrame); SInt16* rightChannel = getSecond16Bits(myStereoFrame); Can anyone help me turn my pseudo into real code?

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  • Record 8 separate Line IN Channels from M-Audio Delta 1010 Card

    - by Peter Hoffmann
    I want to record the 8 separate Line IN Channels from my M-Audio Delta 1010 Card. The card is recogniced nicely and a can record a single channel via arecord -d 10 -f cd -t wav -D channel1 out2.wav. I've set up the different channels in ~/.asoundrc. Now if I want to record a second channel in parallel (arecord -d 10 -f cd -t wav -D channel2 out2.wav) I get the error arecord: main:564: audio open error: Device or resource busy As I understand the delta 1010 is a single Access Card, so only one application can access it at a time. Is this correct? The next step was to configure a dual channel input in .asoundrc # envy24 channel 1+2 only pcm.test { type plug ttable.0.0 1 ttable.0.1 1 slave.pcm ice1712 } Which works ok when I do a arecord -d 10 -f cd -t wav -D test -c 2 out.wav (BTW can anyone point me to a tool to split a multi channel wav into a file per channel?) But when I want to record the channels separately with (-I option) arecord -d 10 -f cd -t wav -D test -c 2 -I channel1.wav channel2.wav I get no recordings. Did I miss something with the configuration or what are my options to record all 8 channels via arecord. I've no experience with jackd. Is it an option to install jackd and record the line ins via jackd?

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  • bluetooth headset can connect, but not visible in pulse audio

    - by Kim Marivoet
    I have a plantronics bluetooth headset, and until yesterday I could use it without any problem. However, today it suddenly stopped working (maybe related to the last software update I did). I can still connect/disconnect my headset, but it doesn't show up in pulse audio anymore. I read through various posts that describes kind of the same problem, but none of the suggested solutions worked. I get following error in the syslog: Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/HFPAG Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSource Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSink Oct 13 16:50:09 desktop kernel: [ 17.340943] input: 48:C1:AC:08:FE:8F as /devices/virtual/input/input14 Oct 13 16:50:09 desktop bluetoothd[1040]: /org/bluez/1040/hci0/dev_48_C1_AC_08_FE_8F/fd0: fd(36) ready Oct 13 16:50:09 desktop rtkit-daemon[1894]: Successfully made thread 2213 of process 1892 (n/a) owned by '1000' RT at priority 5. Oct 13 16:50:09 desktop rtkit-daemon[1894]: Supervising 5 threads of 1 processes of 1 users. Oct 13 16:50:10 desktop bluetoothd[1040]: Badly formated or unrecognized command: AT+XEVENT=USER-AGENT,COM.PLANTRONICS,PLT_VOYAGERPRO,0109,27.90,FFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFF Oct 13 16:50:10 desktop bluetoothd[1040]: Audio connection got disconnected Any help would be much appreciated. I'm using Ubuntu 12.04. Thanks, Kim

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  • Setting up Beats audio on HP Pavilion m6

    - by Joel Auterson
    I have an HP Pavilion m6-1054sa laptop, with a Beats subwoofer on the bottom. The normal laptop speakers work fine under Ubuntu but the Beats speaker(s?) does not. Anyone know how to get this working? Here's my lspci output, if it helps... 00:00.0 Host bridge: Intel Corporation Ivy Bridge DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Ivy Bridge PCI Express Root Port (rev 09) 00:02.0 VGA compatible controller: Intel Corporation Ivy Bridge Graphics Controller (rev 09) 00:14.0 USB controller: Intel Corporation Panther Point USB xHCI Host Controller (rev 04) 00:16.0 Communication controller: Intel Corporation Panther Point MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 1 (rev c4) 00:1c.1 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 2 (rev c4) 00:1d.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Panther Point LPC Controller (rev 04) 00:1f.2 RAID bus controller: Intel Corporation 82801 Mobile SATA Controller [RAID mode] (rev 04) 00:1f.3 SMBus: Intel Corporation Panther Point SMBus Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Thames XT/GL [Radeon HD 7600M Series] (rev ff) 07:00.0 Unassigned class [ff00]: Realtek Semiconductor Co., Ltd. Device 5289 (rev 01) 07:00.2 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 0a) 08:00.0 Network controller: Intel Corporation Centrino Wireless-N 2230 (rev c4)

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  • Problem Installing Xubuntu 12.04 Audio-Video codecs

    - by Seib
    I used Crouton to install Xubuntu 12.04.4 LTS with the XFCE environment on my HP Chromebook 11. I've gotten it all fully installed and everything; the only thing I'm doing now is the basic setup things, like adding codecs and other things like LibreOffice, VLC, Firefox, Ubuntu Software Center, etc. This information I got from 2 sources: http://www.efytimes.com/e1/fullnews.asp?edid=137269 http://www.binarytides.com/better-xubuntu-14-04/ . I'm currently on the same step at both URLs, which is #6 on Link 1 and #8 on Link 2. Per the articles, which both said the same thing, I typed in sudo apt-get install xubuntu-restricted-extras libavcodec-extra and it didn't do anything. It just kept on saying the same thing: Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package libavcodec-extra I've spend the last hour or so scouring the internet for a solution, for a hint even at what is going on. I don't want to do a clean reinstall, for two reasons: 1) it's takes like 1.5h just to get the croot fully installed, and 2) everything but this out of what I've done so far (up to #6 at Link 1 & up to #8 at Link 2) works except the audio. I've already installed flash, so YouTube works fine. It's just I can't hear any audio. Please help? Thanks in advance. I appreciate all the great help I've been getting from AskUbuntu lately. You all are great.

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  • M-Audio Delta starts up at wrong sample rate

    - by steevc
    When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

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  • ffmpeg add two audio streams to video

    - by Tossin Hausen
    I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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  • 5.1 sound in Unity3d 3.5

    - by N0xus
    I'm trying to implement 5.1 surround sound in my game. I've set Unity's AudioManager to a default of 5.1 surround and loaded in a 6 channel audio clip that should play a sound in each of the different audio spots. However, when I go to run my game, all I get is flat sound coming out of my front two speakers. Even then, these don't play the sound they should (front speaker should play "front speaker" right should play "right speaker" and so). Both speakers just end up playing the entire sound file. I've tried looking to see if there is a parameter that I have missed, but information on how to set up 5.1 sound in Unity is lacking (or my google skills aren't that good) and I can't get it to work as intended. Could someone please either tell me what I'm missing, or point me in the right direction? My audio source is situated at point (0, 0, 0) with my camera also being in the same point. I've moved about the scene but the same thing happens as I've already described.

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  • Alsa devices under Wine

    - by Roberto Aloi
    Hi all, I'm running OpenSuse 11.2 and Wine 1.1.28. Even if audio is perfectly working fine for me (Skype, Banshee, etc), when I try to configure audio for Wine (to use Spotify) I cannot hear anything from the audio test. In the winecfg audio tab, ALSA is checked, but no devices are available. I tried to run alsaconf (it needs root permissions) but it returns: No supported PnP or PCI card found No legacy drivers available, either. Any idea?

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  • Audio stream mangement in Linux

    - by User1
    I have a very complicated audio setup for a project. Here's what we have: 3 applications playing sound 2 applications recording sound 2 sound cards I really don't really have the code to any of these applications. All I want to do is monitor and control the audio streams. Here are a few examples of operations I'd like to do while the applications are running: Mute one of the incoming audio streams. Have one of the incoming audio streams do a "solo" (be the only stream that can "talk"). Get a graph (about 30 seconds worth) of the audio that each stream produced. Send one of the audio streams to soundcard #1, but all three audio streams to soundcard #2. I would likely switch audio streams every 2 minutes or so with one of the operations listed above. A GUI would be preferred. I started looking at the sound systems in Linux and it gets extremely complex and I feel like there have been many new advances in the past few years. I see jack, pulseaudio, artsd, and several other packages. They all have some promise but where should I start? Is there something someone already built that can help?

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  • Is there an easy way to copy an audio CD in Mac OS X?

    - by Bob D
    (not a commercial CD). I did some recordings of a band years ago and ran into one of the band members who asked me if I could make copies. I assumed that this would be easy. I know that I can rip the CD into iTunes and then burn a new CD, but I have two optical drives available, is there a way to simply copy the CD from one drive to the other in one step?

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  • Ubuntu: how to get audio to work in both Spotify (under Wine) and Flash (in Firefox)?

    - by Jonik
    I'm running Spotify on Linux using Wine. Sound worked great (even though the sound test in winecfg failed!), until I installed alsa-oss package yesterday to get Flash sound working in Firefox. Now Spotify says: "There is a problem with your sound card. Spotify can't play music." So the question is, how to get the sound in Spotify working again, so that it also keeps working in Flash & Firefox? Tweak some ALSA settings? Spotify settings? Add/remove some packages? By the way, curiously, now that sound doesn't work in Spotify, winecfg's "Test Sound" does work! This is Ubuntu 8.04 (Hardy). Sound card / driver is probably an integrated AC'97. Please mention if any additional information about the system is needed! Update: I have Flash 10 installed (outside the packaging system, using $MOZ_PLUGIN_PATH env variable), but also had Flash 9 from flashplugin-nonfree package - and the earlier version was being used by Firefox! Based on what Mike Arthur said about Flash and alsa-oss, I removed the older Flash (flashplugin-nonfree package) and alsa-oss - and Flash sound still works, which is nice. But for some reason Spotify still doesn't play sound, even though things should now be like they were originally... Update 2: Got it working, all smoothly, finally.

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  • Audio recording, a tool for human-aided drum quantizing.

    - by basilio.mp
    I have this situation: the drummer records the track (8 tracks in a multitrack session). Now, how do I check how distant are the recorded beats from their theoretical position i.e.: there is always some error in human recorded tracks, but is there any software that can show me the ideal (theoretical, quantized) beat and the recorded one and could alert me if the error is too big. P.S.: I'm searching for a standalone tool, or for a plugin that can work with Adobe Audition 3 or Nuendo 3.

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  • Is the Zune HD's audio card better or worse than the iPod touch's?

    - by MatthewThepc
    Firstly, if this is the wrong site to ask this question I apologize, but I didn't see one for "music players" on the stack exchange website :) After reading a few people online say that music playing from a Zune HD sounds better to them than that on an iPod touch, I was wondering whether there's any truth to that? From what I can tell, the Zune HD uses a Wolfson Microelectronics WM8352, while the first-generation iPod Touch (which the Zune HD was competing with) used a Wolfson Microelectronics WM8758BG, and newer models use the Cirrus Logic CS4398 and CS42L61. Which ones are better (to make the question less subjective, let's say in terms of quality, range, & accuracy of output)? Admittedly, I have almost no idea how everything compares and works together, but it would seem to me that, just by looking at the version numbers, the iPod has been better since it's launch. Is there anything else that effects sound quality? Thanks!

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