Search Results

Search found 277 results on 12 pages for 'stereo'.

Page 8/12 | < Previous Page | 4 5 6 7 8 9 10 11 12  | Next Page >

• Convert audio file to FLAC with ffmpeg?

  - by elpsk

  can I convert one of this format to compatible 16000.0 Sample Rate FLAC file? kAudioFormatLinearPCM = 'lpcm', kAudioFormatAppleIMA4 = 'ima4', kAudioFormatMPEG4AAC = 'aac ', kAudioFormatMACE3 = 'MAC3', kAudioFormatMACE6 = 'MAC6', kAudioFormatULaw = 'ulaw', kAudioFormatALaw = 'alaw', kAudioFormatMPEGLayer1 = '.mp1', kAudioFormatMPEGLayer2 = '.mp2', kAudioFormatMPEGLayer3 = '.mp3', kAudioFormatAppleLossless = 'alac' I tried using ffmpeg ffmpeg -i audio.xxx -acodec flac audio.flac but result is FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard Mac OSX universal build for ffmpegX configuration: --enable-memalign-hack --enable-mp3lame --enable-gpl --disable-vhook --disable-ffplay --disable-ffserver --enable-a52 --enable-xvid --enable-faac --enable-faad --enable-amr_nb --enable-amr_wb --enable-pthreads --enable-x264 libavutil version: 49.0.0 libavcodec version: 51.9.0 libavformat version: 50.4.0 built on Apr 15 2006 04:58:19, gcc: 4.0.1 (Apple Computer, Inc. build 5250) Input #0, wsaud, from 'audio.alac': Duration: 00:00:03.8, start: 0.000000, bitrate: 199 kb/s Stream #0.0: Audio: adpcm_ima_ws, 24931 Hz, stereo, 199 kb/s Unable for find a suitable output format for 'audio.flac' I also installed flac codec for mac, but nothing... I tried also use convtoflac.sh (from http://legroom.net/software/convtoflac) but result is similar. Any idea to convert in flac?

  Read the article

• FFMPEG Segfault Solutions

  - by Brentley_11

  I'm trying to convert a bunch of movies into h.264 mp4's using FFMPEG. These movies are sourced from various portable camcorders such as the Flip Mino HD and the Kodak ZI8. One issue I'm having with video from the ZI8 is it seems to be causing FFMPEG to segfault. Here is my command: ffmpeg -i 'XmasSailor720p60fps.MOV' -threads 2 -acodec libfaac -ab 96kb -vcodec libx264 -vpre hq -b 500kb -s 484x272 XmasSailor.mp4 Here is the output: FFmpeg version SVN-r20668, Copyright (c) 2000-2009 Fabrice Bellard, et al. built on Dec 2 2009 18:37:34 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared libavutil 50. 5. 1 / 50. 5. 1 libavcodec 52.42. 0 / 52.42. 0 libavformat 52.39. 2 / 52.39. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0. 7. 2 / 0. 7. 2 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 59.94 (60000/1001) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'XmasSailor720p60fps.MOV': Duration: 00:00:05.37, start: 0.000000, bitrate: 12021 kb/s Stream #0.0(eng): Video: h264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 11994 kb/s, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Metadata major_brand : qt minor_version : 0 compatible_brands: qt comment : KODAK Zi8 Pocket Video Camera comment-eng : KODAK Zi8 Pocket Video Camera [libx264 @ 0x99e1020]using SAR=1/1 [libx264 @ 0x99e1020]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x99e1020]profile High, level 2.1 Output #0, mp4, to 'XmasSailor.mp4': Stream #0.0(eng): Video: libx264, yuv420p, 484x272 [PAR 1:1 DAR 121:68], q=10-51, 500 kb/s, 30k tbn, 29.97 tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 96 kb/s Metadata comment : Encoded with the Statusfirm Video Transcoder Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [h264 @ 0x99de950]B picture before any references, skipping [h264 @ 0x99de950]decode_slice_header error [h264 @ 0x99de950]no frame! Error while decoding stream #0.0 [h264 @ 0x99de950]B picture before any references, skipping [h264 @ 0x99de950]decode_slice_header error [h264 @ 0x99de950]no frame! Error while decoding stream #0.0 frame= 20 fps= 0 q=13797729.0 size= 0kB time=0.66 bitrate= 0.6kbits/s frame= 39 fps= 37 q=13797729.0 size= 0kB time=1.30 bitrate= 0.3kbits/s frame= 48 fps= 30 q=33.0 size= 11kB time=0.10 bitrate= 903.0kbits/s frame= 58 fps= 27 q=31.0 size= 22kB time=0.43 bitrate= 421.0kbits/s frame= 67 fps= 25 q=29.0 size= 41kB time=0.73 bitrate= 462.6kbits/s frame= 75 fps= 23 q=29.0 size= 59kB time=1.00 bitrate= 486.7kbits/s frame= 83 fps= 22 q=29.0 size= 81kB time=1.27 bitrate= 521.9kbits/s frame= 90 fps= 21 q=29.0 size= 97kB time=1.50 bitrate= 530.1kbits/s frame= 98 fps= 20 q=29.0 size= 114kB time=1.77 bitrate= 526.9kbits/s frame= 106 fps= 20 q=29.0 size= 134kB time=2.04 bitrate= 537.7kbits/s frame= 114 fps= 19 q=29.0 size= 150kB time=2.30 bitrate= 533.7kbits/s frame= 122 fps= 19 q=29.0 size= 172kB time=2.57 bitrate= 547.8kbits/s frame= 130 fps= 19 q=29.0 size= 193kB time=2.84 bitrate= 557.5kbits/s frame= 136 fps= 18 q=29.0 size= 211kB time=3.04 bitrate= 570.0kbits/s frame= 144 fps= 18 q=29.0 size= 242kB time=3.30 bitrate= 599.5kbits/s frame= 152 fps= 17 q=30.0 size= 261kB time=3.57 bitrate= 598.6kbits/s frame= 157 fps= 15 q=-1.0 Lsize= 368kB time=5.21 bitrate= 579.3kbits/s video:302kB audio:61kB global headers:0kB muxing overhead 1.416371% [libx264 @ 0x99e1020]frame I:1 Avg QP:27.22 size: 8720 [libx264 @ 0x99e1020]frame P:48 Avg QP:25.15 size: 3759 [libx264 @ 0x99e1020]frame B:108 Avg QP:30.10 size: 1105 [libx264 @ 0x99e1020]consecutive B-frames: 0.6% 11.5% 28.8% 59.0% [libx264 @ 0x99e1020]mb I I16..4: 28.5% 47.6% 23.9% [libx264 @ 0x99e1020]mb P I16..4: 0.8% 1.3% 0.5% P16..4: 50.6% 17.7% 13.1% 0.0% 0.0% skip:15.9% [libx264 @ 0x99e1020]mb B I16..4: 0.2% 0.3% 0.1% B16..8: 44.0% 1.2% 2.6% direct: 5.1% skip:46.5% L0:45.5% L1:51.0% BI: 3.5% [libx264 @ 0x99e1020]final ratefactor: 23.51 [libx264 @ 0x99e1020]8x8 transform intra:49.9% inter:67.9% [libx264 @ 0x99e1020]direct mvs spatial:98.1% temporal:1.9% [libx264 @ 0x99e1020]coded y,uvDC,uvAC intra: 54.7% 76.1% 41.4% inter: 17.1% 24.4% 7.8% [libx264 @ 0x99e1020]i16 v,h,dc,p: 18% 52% 5% 25% [libx264 @ 0x99e1020]i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 22% 9% 7% 10% 10% 9% 8% 13% [libx264 @ 0x99e1020]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 18% 8% 8% 10% 13% 10% 9% 12% [libx264 @ 0x99e1020]Weighted P-Frames: Y:10.4% [libx264 @ 0x99e1020]ref P L0: 60.2% 15.3% 11.0% 7.6% 5.2% 0.7% [libx264 @ 0x99e1020]ref B L0: 72.6% 15.6% 11.8% [libx264 @ 0x99e1020]kb/s:471.17 Segmentation fault I'm wondering if anyone else has ran into similar issues. I wasn't able to find anything helpful via Google. Another question I have is if anyone knows of a company that offers paid support for FFMPEG. Thank you for your time.

  Read the article

• FFMPEG Segfault Solutions

  - by Brentley_11

  I'm trying to convert a bunch of movies into h.264 mp4's using FFMPEG. These movies are sourced from various portable camcorders such as the Flip Mino HD and the Kodak ZI8. One issue I'm having with video from the ZI8 is it seems to be causing FFMPEG to segfault. Here is my command: ffmpeg -i 'XmasSailor720p60fps.MOV' -threads 2 -acodec libfaac -ab 96kb -vcodec libx264 -vpre hq -b 500kb -s 484x272 XmasSailor.mp4 Here is the output: FFmpeg version SVN-r20668, Copyright (c) 2000-2009 Fabrice Bellard, et al. built on Dec 2 2009 18:37:34 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared libavutil 50. 5. 1 / 50. 5. 1 libavcodec 52.42. 0 / 52.42. 0 libavformat 52.39. 2 / 52.39. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0. 7. 2 / 0. 7. 2 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 59.94 (60000/1001) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'XmasSailor720p60fps.MOV': Duration: 00:00:05.37, start: 0.000000, bitrate: 12021 kb/s Stream #0.0(eng): Video: h264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 11994 kb/s, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Metadata major_brand : qt minor_version : 0 compatible_brands: qt comment : KODAK Zi8 Pocket Video Camera comment-eng : KODAK Zi8 Pocket Video Camera [libx264 @ 0x99e1020]using SAR=1/1 [libx264 @ 0x99e1020]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x99e1020]profile High, level 2.1 Output #0, mp4, to 'XmasSailor.mp4': Stream #0.0(eng): Video: libx264, yuv420p, 484x272 [PAR 1:1 DAR 121:68], q=10-51, 500 kb/s, 30k tbn, 29.97 tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 96 kb/s Metadata comment : Encoded with the Statusfirm Video Transcoder Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [h264 @ 0x99de950]B picture before any references, skipping [h264 @ 0x99de950]decode_slice_header error [h264 @ 0x99de950]no frame! Error while decoding stream #0.0 [h264 @ 0x99de950]B picture before any references, skipping [h264 @ 0x99de950]decode_slice_header error [h264 @ 0x99de950]no frame! Error while decoding stream #0.0 frame= 20 fps= 0 q=13797729.0 size= 0kB time=0.66 bitrate= 0.6kbits/s frame= 39 fps= 37 q=13797729.0 size= 0kB time=1.30 bitrate= 0.3kbits/s frame= 48 fps= 30 q=33.0 size= 11kB time=0.10 bitrate= 903.0kbits/s frame= 58 fps= 27 q=31.0 size= 22kB time=0.43 bitrate= 421.0kbits/s frame= 67 fps= 25 q=29.0 size= 41kB time=0.73 bitrate= 462.6kbits/s frame= 75 fps= 23 q=29.0 size= 59kB time=1.00 bitrate= 486.7kbits/s frame= 83 fps= 22 q=29.0 size= 81kB time=1.27 bitrate= 521.9kbits/s frame= 90 fps= 21 q=29.0 size= 97kB time=1.50 bitrate= 530.1kbits/s frame= 98 fps= 20 q=29.0 size= 114kB time=1.77 bitrate= 526.9kbits/s frame= 106 fps= 20 q=29.0 size= 134kB time=2.04 bitrate= 537.7kbits/s frame= 114 fps= 19 q=29.0 size= 150kB time=2.30 bitrate= 533.7kbits/s frame= 122 fps= 19 q=29.0 size= 172kB time=2.57 bitrate= 547.8kbits/s frame= 130 fps= 19 q=29.0 size= 193kB time=2.84 bitrate= 557.5kbits/s frame= 136 fps= 18 q=29.0 size= 211kB time=3.04 bitrate= 570.0kbits/s frame= 144 fps= 18 q=29.0 size= 242kB time=3.30 bitrate= 599.5kbits/s frame= 152 fps= 17 q=30.0 size= 261kB time=3.57 bitrate= 598.6kbits/s frame= 157 fps= 15 q=-1.0 Lsize= 368kB time=5.21 bitrate= 579.3kbits/s video:302kB audio:61kB global headers:0kB muxing overhead 1.416371% [libx264 @ 0x99e1020]frame I:1 Avg QP:27.22 size: 8720 [libx264 @ 0x99e1020]frame P:48 Avg QP:25.15 size: 3759 [libx264 @ 0x99e1020]frame B:108 Avg QP:30.10 size: 1105 [libx264 @ 0x99e1020]consecutive B-frames: 0.6% 11.5% 28.8% 59.0% [libx264 @ 0x99e1020]mb I I16..4: 28.5% 47.6% 23.9% [libx264 @ 0x99e1020]mb P I16..4: 0.8% 1.3% 0.5% P16..4: 50.6% 17.7% 13.1% 0.0% 0.0% skip:15.9% [libx264 @ 0x99e1020]mb B I16..4: 0.2% 0.3% 0.1% B16..8: 44.0% 1.2% 2.6% direct: 5.1% skip:46.5% L0:45.5% L1:51.0% BI: 3.5% [libx264 @ 0x99e1020]final ratefactor: 23.51 [libx264 @ 0x99e1020]8x8 transform intra:49.9% inter:67.9% [libx264 @ 0x99e1020]direct mvs spatial:98.1% temporal:1.9% [libx264 @ 0x99e1020]coded y,uvDC,uvAC intra: 54.7% 76.1% 41.4% inter: 17.1% 24.4% 7.8% [libx264 @ 0x99e1020]i16 v,h,dc,p: 18% 52% 5% 25% [libx264 @ 0x99e1020]i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 22% 9% 7% 10% 10% 9% 8% 13% [libx264 @ 0x99e1020]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 18% 8% 8% 10% 13% 10% 9% 12% [libx264 @ 0x99e1020]Weighted P-Frames: Y:10.4% [libx264 @ 0x99e1020]ref P L0: 60.2% 15.3% 11.0% 7.6% 5.2% 0.7% [libx264 @ 0x99e1020]ref B L0: 72.6% 15.6% 11.8% [libx264 @ 0x99e1020]kb/s:471.17 Segmentation fault I'm wondering if anyone else has ran into similar issues. I wasn't able to find anything helpful via Google. Another question I have is if anyone knows of a company that offers paid support for FFMPEG. Thank you for your time.

  Read the article

• Webcam microphone input in Gnome/pulseaudio

  - by sdaau

  Just got a "Trust" webcam, which gets recognized on my Ubuntu Lucid. It has a built in microphone - which also gets recognized - however, I cannot really get it to act as the system microphone input? Here are some screenshots of what is shown by gnome-volume-control: The default window shows Trust webcam - which has two profiles: "Analog Mono Input" and "Off" - of course, I have it on "Analog Mono Input": However, on the "Input" tab - there is no matching "device for sound input" - neither a matching connector: Then I installed pavucontrol - but that doesn't show that much more; it tells first that gnome-volume-control reads from "Internal Audio Analog Stereo": Then in "Input devices" tab, there is again nothing resembling the mic input from webcam: Finally, under "Configuration" tab, the "Trust" webcam shows, but even if its profile is on "Analog Mono Input", nothing much happens:   So, does anyone know how I could get this webcam microphone to be recognized as the system input? Many thanks in advance for any answers, Cheers!

  Read the article

• Enable multiple audio output on Windows 7

  - by patrick

  For Windows 7, 64 bit: I have a digital SPDIF output to my stereo, which controls speakers in other rooms. I also have a set of speakers connected to the regular audio jack at the computer. This allows me to send music to the kitchen while my child plays games on the computer. Works great. Except when I'm playing games and still want to listen to music. ;-D I know I can manually switch WMP to play through the speakers instead of SPDIF, but I was wondering if there's any way to enable simultaneous audio out in Windows 7? Virtual Audio Card is a non-starter because I'm running 64 bits and the VAC driver isn't signed.

  Read the article

• squeaky sound when in 5.1 mode Audigy2 ls

  - by ageis23

  Hi I'm using alsa in 5.1 mode. pcm.ch51dup { slave.pcm surround51 slave.channels 6 type route ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } The 5.1 mode works but I get sqeaky sound in one of the speakers but it's not there once I turn it back into stereo mode and turn on the matrix mode. The speakers are the Logitech X-540. I do have onboard sound but I've disabled via the sound preferences applet in gnome I'm using a Audigy2 ls with ubuntu lucid. Why is this? The asoundrc is the only file I've played with. I enable 5.1 using the sound preferences applet provided by gnome.

  Read the article

• Volume display for laptop missing after driver update

  - by Cyclone

  I have a Dell Inspiron 1520, and I just updated my audio drivers to be able to use Stereo Mix. It's a SigmaTel audio driver. Prior to the update, whenever I used the hardware buttons to update volume or mute/unmute speakers, it would always have a nice display like this one: But, for volume instead of screen brightness. The hardware buttons still work fortunately, but the volume display is gone. Is there a place I can configure this setting and re-enable it? The speaker icon in task tray was also disabled with the update, but I re-enabled that from the Control Panel. Is this simply gone with the driver update? Thanks for the help!

  Read the article

• VMWare Workstation Dev Machine Disks: one fast or four echofriendly raid?

  - by Avi

  I'm building a new dev computer. It will be running a few VMWare Worksation virtual machines - A dev machine running VS-2010, a build machine, a version-control machine, a web server for testing, a "personal" machine running office etc. I'll be connecting the computer to my stereo, so I'll also be running iTunes (possible on a dedicated VM) and I want the computer to be a silent one. I'll probably use an Antec P183 case. I was advised on Serverfault to use Raid10 for performance. Raid 10 uses 4 disks. So, my question is as follows: In terms of heat, noise, reliability, warranty, price, capacity and performance, what would you suggest: A Raid10 4 disk array using eco-friendly disks such as the $94 1TB Western Digital Caviar Green, or one high performance disk such as the 2TB Western Digital Caviar Black at $280?

  Read the article

• How do I swap audio output of the left and right speakers?

  - by Manga Lee

  I have two speakers stereo speakers but when I use the sound control panel applet to test my audio configuration I get sound in the right speaker when the user interface indicates the right speaker and vice versa. Is there a way to swap the audio output from left to right and right to left? UPDATE: The reason for this question is that I've recently rearranged my workspace and because of physical constraints the left speaker has to go on the right side and vice versa. I could of course solve this problem with a hardware solution but I'd rather use a software solution if one is available.

  Read the article

• How to play audio through a bluetooth headset in Windows 7.

  - by palehorse

  I recently did an in-place upgrade from Vista to Windows 7 RTM. For the most part, things have went brilliantly. The only issue I've been facing is regarding my bluetooth headset, a Dell BH200. The laptop is an Inspiron 1720. I can get Windows 7 to pair with the headset, and even get the bluetooth settings to show that there is stereo audio available. The problem is that the bluetooth headset never shows up in my sound output devices list. I've uninstalled and reinstalled the bluetooth drivers from Dell, tried turning on/off all of the features of the headset, removing the bluetooth driver and letting windows redetect and install but nothing has worked so far. I guess I should mention that the headset worked fine in Vista. Anyone have any ideas?

  Read the article

• Ubuntu 11.04: No Audio Output

  - by Jason George

  I installed a fresh copy of Ubuntu 11.04 earlier this week and I'm having trouble getting my audio online. It was working fine in 10.04 and all the resources I can find on troubleshooting seem to be fairly dated so I'm not sure if they apply. CMI8788 [Oxygen HD Audio] Analog Stereo Duplex Playing a WMA file shows 0.00db output when I mouse over the sound controller in the status bar. Obviously, no output from my speakers. I tried adjusting the profile, thinking I might have the wrong one. That seems to have made things worse. Where mouse over originally said something along the lines of "Oxygen HD Audio," it now reads "Dummy Output." Selecting "Test Speakers" in sound preferences crashes the dialog. Any pointers would be great.

  Read the article

• Quality wise, is Windows Media Audio 10 Professional equivalent to WMA?

  - by Louis

  I noticed that for encoding CD rips, Zune is still using WMA 9.2 instead of WMA 10 Pro. On a given file using the highest quality VBR settings looks like this: VBR Quality 98, 44 kHz, stereo 1-pass VBR On the same file if I use WMA 10 Pro, with the same settings, the resulting file is about 20% smaller. Using my ears, I'm unable to tell the difference, but I'm wondering if this was the goal of WMA 10 Pro (to be as good as WMA at a lower bitrate). Is the quality of a WMA 10 Pro file equal to that of a WMA 9.2 file encoded with the same settings?

  Read the article

• No surround sound over HDMI

  - by Chris A

  I have my Pioneer receiver hooked up to my AMD HD4600m which supports audio output over HDMI. I used to work fine, however, after a recent driver update, it will only output stereo sound. It used to output 5.1 surround sound perfectly. Windows does not seem to recognize the device as capable of 5.1. In the past, 5.1 was given as an option in the configuration window, like this: I can not get it to properly recognize the capabilities of the device. I have tried uninstalling the relevant audio device in device manager, and I have also completely removed and reinstalled the display driver. Any suggestions would be greatly appreciated. Thanks

  Read the article

• Data CD for audiobooks?

  - by Marco7757

  I'm trying to burn my .mb4-audiobook files to a CD. I was impressed by the compression-rate (10 hours of audiobook within 150MB?!). The problem now is, that I cannot burn it as an audio CD as these allow only about 80 minutes of audio (audiobook is about +10 hours). I burned them as a data CD now. It works, but, of course, the downside of a data CD is, that not every player (e.g. car, stereo) can play data CDs. What can I do? I don't want to waste 100 CDs on such a simple problem ... is there any way to burn an audio CD? I mean, just regarding the filesize this shouldn't be a problem, shouldn't it? Why is an audio CD only able to play up to 80 minutes?

  Read the article

• Where are the Record Volume Controls in Windows 7?

  - by DJbigJack

  Windows XP (and previous versions) had a Record Volume Control panel that could be used to select between music inputs (Stereo Mix, Wav, etc) and a microphone. There doesn't seem to be an equivalent capability in Windows 7 . Is there a third party application that provides this functionality? Note: the Windows XP Record Volume Control was accessed by doubleclicking the Speaker icon in the system try which displayed the (Listen) Volume Control. In the menu there was a "properties" which gave you the option of displaying the RECORD Volume Control instead. I used this capabiliy in Win XP to select the required inputs for an Internet Radio Station and now with Win 7, I can't do it any longer

  Read the article

• Help with dynamic range compression function (audio)

  - by MusiGenesis

  I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think): public static void Compress(ref short[] input, double thresholdDb, double ratio) { double maxDb = thresholdDb - (thresholdDb / ratio); double maxGain = Math.Pow(10, -maxDb / 20.0); for (int i = 0; i < input.Length; i += 2) { // convert sample values to ABS gain and store original signs int signL = input[i] < 0 ? -1 : 1; double valL = (double)input[i] / 32768.0; if (valL < 0.0) { valL = -valL; } int signR = input[i + 1] < 0 ? -1 : 1; double valR = (double)input[i + 1] / 32768.0; if (valR < 0.0) { valR = -valR; } // calculate mono value and compress double val = (valL + valR) * 0.5; double posDb = -Math.Log10(val) * 20.0; if (posDb < thresholdDb) { posDb = thresholdDb - ((thresholdDb - posDb) / ratio); } // measure L and R sample values relative to mono value double multL = valL / val; double multR = valR / val; // convert compressed db value to gain and amplify val = Math.Pow(10, -posDb / 20.0); val = val / maxGain; // re-calculate L and R gain values relative to compressed/amplified // mono value valL = val * multL; valR = val * multR; double lim = 1.5; // determined by experimentation, with the goal // being that the lines below should never (or rarely) be hit if (valL > lim) { valL = lim; } if (valR > lim) { valR = lim; } double maxval = 32000.0 / lim; // convert gain values back to sample values input[i] = (short)(valL * maxval); input[i] *= (short)signL; input[i + 1] = (short)(valR * maxval); input[i + 1] *= (short)signR; } } and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios. Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however. I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this: Can anybody suggest a modification to my function to produce the soft knee curve?

  Read the article

• Ubuntu 11.10 - How can i stop self-feedback-loop, coming directly from my microphone to speaker?

  - by YumYumYum

  I have microphone audio, which comes instantly to my speaker. I am using default pulseaudio and alsa from the package. I have tried to setup: 1) PA /etc/pulse/default.pa /etc/asound.conf $ ls analog-input-aux.conf analog-input-fm.conf analog-input-mic.conf analog-input-tvtuner.conf analog-output-desktop-speaker.conf analog-output-mono.conf analog-input.conf analog-input-front-mic.conf analog-input-mic.conf.common analog-input-video.conf analog-output-headphones-2.conf analog-output-speaker.conf analog-input.conf.common analog-input-internal-mic.conf analog-input-mic-line.conf analog-output.conf analog-output-headphones.conf iec958-stereo-output.conf analog-input-dock-mic.conf analog-input-linein.conf analog-input-rear-mic.conf analog-output.conf.common analog-output-lfe-on-mono.conf 2) ALSA in lsmod to make sure no loopback modules are loaded etc but none is resolving it. And there are many less information available on this. Has anyone similar problem solution in Ubuntu 11.10? (this problem i have resolved in Ubuntu 11.04 by replacing the default pulseaudio version to latest source from git, but while trying the same in Ubuntu 11.10 does not worked). Any tips please?

  Read the article

• What packages are neccessary to have sound output from java applets?

  - by MvG

  I've got a very minimalistic setup of ubuntu precise, created using debootstrap. So please don't assume that any packages are installed just because they usually are. On that system, I'd like to play some sounds from a java applet. However, this always fails with the following error message: javax.sound.midi.MidiUnavailableException: Can not open line at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1132) at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1036) ... Caused by: java.lang.IllegalArgumentException: No line matching interface SourceDataLine supporting format PCM_SIGNED 44100.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian is supported. at javax.sound.sampled.AudioSystem.getLine(AudioSystem.java:476) at javax.sound.sampled.AudioSystem.getSourceDataLine(AudioSystem.java:604) at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1066) ... 35 more As the messages mention a soft synthesizer, and pcm lines, I expect that the lack of some midi daemon is not the issue here. As far as I can tell, the alsa kernel modules are loaded, including snd_hda_intel, snd_pcm, snd_seq_midi among others. I've also included the alsa-base and alsa-utils packages in my installation. alsa-mixer looks good, using “HDA Intel PCH” as its default device. What other packages, configuration settings or daemon startups does java require to make its sound output work?

  Read the article

• what is the best program to capture my workings as an AVI or MPEG

  - by raihanchy

  I have already used recordmydesktop, xvidcap and kazam. My sound working fine with other audio or videos. xvidcap doesn't record sound at all. I have tried many ways. If I try as: 'padsp xvidcap', it also gives error, like: /dev/dsp cannot found or missing. I have changed it to /dev/snd. Still no effect. Even I can record sound through gnome-sound-recorder - after pressing record button, I open pavucontrol. Then from Recording tab, I choose 'Monitor of Analog Stereo'. But if I run xvidcap, I don't get that option in pavucontrol. kazam works a bit slow. It records at the beginning of the captured video. But for unknown reason, it eventually the sounds just go off. Also the video is not smooth as xvidcap. Though Kazam output as H64/MP4. Record my Desktop also doesn't give sound. Can you guys please help me, either - how to get sound with xvidcap or how kazam could be record nicely. I am looking something Camtasia, as used for Windows. Thanks in advance. Raihan

  Read the article

• Unable to record sound from web browser (firefox / chromium ) using recordmydesktop

  - by thamurath

  I have to do some screencast tutorials and i am using recordmydesktop with gtk frontend to do it. I need to record also the sound and here is where i have found the problem. It took me some time, but now I can record the sound from almost every application in my desktop ... almost. I need to capture some sound from a web application using java, but when i load the page nothing appears in the playback tab of pavucontrol. I think this is the problem, because if there is no sound stream i think the recordmydesktop program thinks there is no sound to record ... the funny thing is that I can ear the sound in my speakers! I have tried with Firefox and Chromium with no success. Although I have been able to record youtube videos without problem, so it seems that java is the key here. Any suggestion or idea? P.S.: I am using Ubuntu 11.10 with this configuration. ( if more information is needed please let me know) sight i cannot post images ... so I have an audigy2 sound card using Analog Stereo Output profile. I have also an "Internal Audio" device, but i have it with the "Off" profile. In recordmydesktop-Advanced-Sound: Device = default

  Read the article

• Recording slow web stream

  - by Budric

  I'm trying to record an mpeg2 video stream from a website that doesn't have the greatest bandwidth. The video often buffers. I want to download the stream and watch it offline. The extract stream format received is: Stream #0.0[0x44]: Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s Stream #0.1[0x45]: Video: mpeg2video (Main), yuv420p, 704x576 [PAR 16:11 DAR 16:9], 15000 kb/s, 27.19 fps, 25 tbr, 90k tbn, 50 tbc I use the following tool to transocde the stream: ffmpeg -i "http://url" -y -vcodec libx264 -b 3000k -acodec copy /tmp/stream.mp4 Unfortunately after a few seconds ffmpeg stops recording with an error [mpegts @ 0x1f0b9c0] PES packet size mismatch [mp2 @ 0x1f14640] incomplete frame Error while decoding stream #0.0 [mpeg2video @ 0x1f16860] ac-tex damaged at 0 26 [mpeg2video @ 0x1f16860] Warning MVs not available I've tried encoding with vlc as well with similar issues. Although vlc doesn't stop encoding, the output video has regions where it hangs. vlc -I dummy "http://url" --network-caching="1000" --sout="#transcode{vcodec=h264,vb=3000,acodec=mp3,ab=192}:std{access=file,mux=mp4,dst=/tmp/stream.mp4}" [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 9 33 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 132 DC, 132 AC, 132 MV errors [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 16 17 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 836 DC, 836 AC, 836 MV errors libdvbpsi error (PSI decoder): TS discontinuity (received 4, expected 3) for PID 0 I also tried flv transcoding and it shows up with its own set of issues, like output flv file hangs in certain parts. Anyone know what's wrong or how to fix this?

  Read the article

• How do I stop sound coming from my speakers with a faulty headphone socket?

  - by Andy

  On my laptop, I have a faulty headphone socket, so when I insert headphones into it, the speakers do not mute. I can confirm that this problem is caused by faulty hardware and not software as when I twist the headphone jack, the speakers come on and off according to the movements. On previous versions of Ubuntu, I worked around this problem by going into alsamixer and disabling "Auto-Mute Mode", and then going into the sound settings and choosing "Analog Headphones". However, on 12.04, no such option exists, rendering my headphones unusable with no way to work around the problem. I momentarily thought I had this problem fixed when I installed PulseAudio Volume Control from the Software Centre. I selected the Output Devices tab, and under "Built-in Audio Analogue Stereo" I selected "Headphones" for the port. However, this almost randomly seems to change back to "Speakers", despite me setting "Auto-Mute Mode" as disabled. Basically, I would like a way to permanently mute the speakers so only the headphones will play sound, without it losing my settings. It is ridiculous that such a simple setting has been taken away to "simplify" the user interface.

  Read the article

• Cannot play windows WMA lossless files on Rhythmbox

  - by sr71

  I have installed Ubuntu 10.10 with all its updates (without Windows) on it’s own drive and everything is working fine. I want to play WMA audio files, also mp3 files. The mp3 files play fine. The WMA files do not play. Used "Rhythmbox Music Player" with and without "Ubuntu-restricted" -extras. Still does not play the lossless windows audio files. I am frustrated with searching to play a WMA file ("download this converter"), but one cannot use this until one "deletes this". I have done everything but it still does not play my windows lossless files that I made from all my CD’s. I am looking for a music player that I can use to play mp3’s and WMA lossless music files and automatically put the album cover on and update the info if one exists. Installation should be as simple as possible. Right now I am back to the original virgin Ubuntu 10.10 with all the recent updates. This computer will do nothing but play music (mp3 and WMA) through a stereo system. I also use Internet to update album info for the music. I do not care what bells and whistles the music player program has, as long as it is an easy install and just plays my mp3 and wma lossless music files. Any help would be appreciated.

  Read the article

• xubutnu Nvidia Settings not remembered

  - by hozza

  I have an Nvidia card and I'm using NVIDIA Driver Version: 304.51 and the NVIDIA X Server Settings GUI. Everything works fine but when I reboot and login again both my two screens are set to +0 +0 so they mirror each other... I change the settings to screen 2 (NEC LED) to be left of screen 1 (DELL) click Apply and Save to X Configuration File... It all works but when I login again the settings are not remembered... This is my xorg config file, can anyone help out? # nvidia-settings: X configuration file generated by nvidia-settings # nvidia-settings: version 304.51 (buildd@komainu) Fri Oct 12 12:53:49 UTC 2012 Section "ServerLayout" Identifier "Layout0" Screen 0 "Screen0" 0 0 InputDevice "Keyboard0" "CoreKeyboard" InputDevice "Mouse0" "CorePointer" Option "Xinerama" "0" EndSection Section "Files" EndSection Section "InputDevice" # generated from default Identifier "Mouse0" Driver "mouse" Option "Protocol" "auto" Option "Device" "/dev/psaux" Option "Emulate3Buttons" "no" Option "ZAxisMapping" "4 5" EndSection Section "InputDevice" # generated from default Identifier "Keyboard0" Driver "kbd" EndSection Section "Monitor" # HorizSync source: edid, VertRefresh source: edid Identifier "Monitor0" VendorName "Unknown" ModelName "DELL 2005FPW" HorizSync 30.0 - 83.0 VertRefresh 56.0 - 75.0 Option "DPMS" EndSection Section "Device" Identifier "Device0" Driver "nvidia" VendorName "NVIDIA Corporation" BoardName "GeForce GT 640" EndSection Section "Screen" Identifier "Screen0" Device "Device0" Monitor "Monitor0" DefaultDepth 24 Option "Stereo" "0" Option "nvidiaXineramaInfoOrder" "DFP-0" Option "metamodes" "DFP-0: nvidia-auto-select +1280+0, DFP-2: nvidia-auto-select +0+0" SubSection "Display" Depth 24 EndSubSection EndSection

  Read the article

• IPhone track title

  - by woodbase

  If you have an IPhone, you probably know that the name in the playlist comes from the “Title”-attribute instead of the filename. Usually that is not a problem. But when I plug my IPhone to the car stereo the tracks are sorted alphabetically by the “title”-attribute. That becomes a problem when You have an e-book where each chapter starts with “Track 01”. You can manually update this in the file properties (from the context menu in Windows Explorer), but doing so for +200 tracks – no thank you :) The FileInfo-class does not contain a property for this special audio file attribute. However the problem is easily solved using TagLib. The method below, not optimized in any way - just solving the problem at hand, will set the “title”-attribute to the file name. private static void UpdateTitleAttr(string dirPath, string fileFilter)         {             var files = System.IO.Directory.GetFiles(dirPath, fileFilter);                         foreach (var file in files)             {                 var f = TagLib.File.Create(file);                 var newTitle = f.Name.Substring(f.Name.LastIndexOf(@"\") + 1);                 f.Tag.Title = newTitle;                 f.Save();                }         } So now I can hear e-books while driving :P

  Read the article

< Previous Page | 4 5 6 7 8 9 10 11 12  | Next Page >