Search Results

Search found 21392 results on 856 pages for 'audio output'.

Page 44/856 | < Previous Page | 40 41 42 43 44 45 46 47 48 49 50 51  | Next Page >

• Using SQL Server's Output Clause

  When you are inserting, updating, or deleting records from a table, SQL Server keeps track of the records that are changed in two different pseudo tables: INSERTED, and DELETED. These tables are normally used in DML triggers. If you use the OUTPUT clause on an INSERT, UPDATE, DELETE or MERGE statement you can expose the records that go to these pseudo tables to your application and/or T-SQL code.

  Read the article

• How to join two command output

  - by UAdapter

  for example I have command that shows how much space folder takes du folder | sort -n it works great, however I would like to have human readable form du -h folder however if I do that than I cannot sort it as numeric. How to join "du folder" and "du -h folder" to see output sorted as "du folder", but with first column from "du -h folder" P.S. this is just an example. this technique might be very useful for me (if its possible)

  Read the article

• FFSERVER - streaming an ASF video as Webm output

  - by Emmanuel Brunet

  I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Environment Debian 7.5 ffmpeg 2.2 Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://192.168.1.62:8091/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://192.168.1.62:8091/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream.

  Read the article

• Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

  - by jdl2003

  I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

  Read the article

• How come the ls command prints in multiple columns on tty but only one column everywhere else?

  - by David Lou

  Even after using Unix-like OSes for a couple years, this behaviour still baffles me. When I use the ls command in a directory that has lots of files, the output is usually nicely formatted into multiple columns. Here's an example: $ ls a.txt C.txt f.txt H.txt k.txt M.txt p.txt R.txt u.txt W.txt z.txt A.txt d.txt F.txt i.txt K.txt n.txt P.txt s.txt U.txt x.txt Z.txt b.txt D.txt g.txt I.txt l.txt N.txt q.txt S.txt v.txt X.txt B.txt e.txt G.txt j.txt L.txt o.txt Q.txt t.txt V.txt y.txt c.txt E.txt h.txt J.txt m.txt O.txt r.txt T.txt w.txt Y.txt However, if I try to redirect the output to a file, or pipe it to another command, only a single column appears in the output. Using the same example directory as above, here's what I get when I pipe ls to wc: $ ls | wc 52 52 312 In other words, wc thinks there are 52 lines, even though the output to the terminal has only 5. I haven't observed this behaviour in any other command. Would you like to explain this to me?

  Read the article

• ffserver-2.2 - streaming an ASF video as Webm output with ffserver on Debian 7.5

  - by Emmanuel Brunet

  I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://ffserver_ip:port/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://ffserver_ip:port/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream. Thanks for your help again.

  Read the article

• How can I pause console output in rxvt?

  - by Javid Jamae

  I'm running rxvt in Cygwin on a Windows box. This is how I invoke it: rxvt -sr -sl 2500 -sb -geometry 90x30 -tn rxvt -fn "Lucida Console-14" -e /usr/bin/bash --login -i Anyone know how to pause the console output in rxvt? I can use Ctrl-S / Ctrl-Q to pause / un-pause, but this won't work if a script is already running and spewing output to stdout. Highlighting the terminal window with the mouse doesn't seem to work like with other consoles such as the standard Cygwin console, or the Windows command prompt console. Some sort of scroll lock would be nice, but I can't seem to find any way to do this. I know I could just pipe my output to a file, but I want a way to pause the output for something that I didn't expect to explode with console output. Basically I want to scroll back while its running without it constantly moving me to the bottom of the output buffer as it updates more data to stdout. I don't particularly care if the solution given actually pauses the script (like when you highlight the mouse in the Windows Command window), or just scroll locks and let's me scroll while its still running the underlying script, though I'd like to know how to do both if its possible.

  Read the article

• How to setup Dual Head with "radeon" driver for R770?

  - by user1709408

  I want to make dual head setup without xrandr but with Xinerama. I put "Screen 1" line into xorg.conf, but card still show identical output on DVI-2 and DVI-3 It is important to use xinerama for me (to glue three monitors), that's why i decide not to use ranrd (randr is incompatible with xinerama as i read somewhere) Here is my videocard (HD 4850 X2): lspci | grep R700 03:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI R700 [Radeon HD 4850] 04:00.0 Display controller: Advanced Micro Devices [AMD] nee ATI R700 [Radeon HD 4850] Here is how monitors are connected: grep "DVI" /var/log/Xorg.0.log [ 1210.002] (II) RADEON(0): Output DVI-0 using monitor section Monitor0 [ 1210.048] (II) RADEON(0): Output DVI-1 has no monitor section [ 1210.079] (II) RADEON(0): EDID for output DVI-0 [ 1210.080] (II) RADEON(0): Printing probed modes for output DVI-0 [ 1210.128] (II) RADEON(0): EDID for output DVI-1 [ 1210.128] (II) RADEON(0): Output DVI-0 connected [ 1210.128] (II) RADEON(0): Output DVI-1 disconnected [ 1210.128] (II) RADEON(0): Output DVI-0 using initial mode 1920x1200 [ 1210.160] (II) RADEON(1): Output DVI-2 using monitor section Monitor2 [ 1210.215] (II) RADEON(1): Output DVI-3 has no monitor section [ 1210.246] (II) RADEON(1): EDID for output DVI-2 [ 1210.247] (II) RADEON(1): Printing probed modes for output DVI-2 [ 1210.299] (II) RADEON(1): EDID for output DVI-3 [ 1210.300] (II) RADEON(1): Printing probed modes for output DVI-3 [ 1210.300] (II) RADEON(1): Output DVI-2 connected [ 1210.300] (II) RADEON(1): Output DVI-3 connected [ 1210.300] (II) RADEON(1): Output DVI-2 using initial mode 1920x1200 [ 1210.300] (II) RADEON(1): Output DVI-3 using initial mode 1920x1200 Here is my /etc/X11/xorg.conf Section "ServerFlags" Option "RandR" "0" Option "Xinerama" "1" EndSection Section "ServerLayout" Identifier "Three Head Layout" Screen "MyPrecious0" Screen "MyPrecious2" RightOf "MyPrecious0" Screen "MyPrecious3" LeftOf "MyPrecious0" EndSection Section "Screen" Identifier "MyPrecious0" Monitor "Monitor0" Device "Device300" EndSection Section "Screen" Identifier "MyPrecious2" Monitor "Monitor2" Device "Device400" EndSection Section "Screen" Identifier "MyPrecious3" Monitor "Monitor3" Device "Device401" EndSection Section "Device" Identifier "Device300" BusID "PCI:3:0:0" Screen 0 Driver "radeon" EndSection Section "Device" Identifier "Device400" BusID "PCI:4:0:0" Screen 0 Driver "radeon" EndSection Section "Device" Identifier "Device401" BusID "PCI:4:0:0" Screen 1 Driver "radeon" EndSection Section "Monitor" Identifier "Monitor0" EndSection Section "Monitor" Identifier "Monitor2" EndSection Section "Monitor" Identifier "Monitor3" EndSection I tried to switch to vesa driver (didn't work for me) I tried to add options like Option "ZaphodHeads" "DVI-2" and Option "ZaphodHeads" "DVI-3" into sections "Device 400" and "Device 401" (this didn't help because "ZaphodHeads" option is for ranrd, and randr is disabled by decision) I tried to merge sections "Device 400" and "Device 401" into one section and add Option "ZaphodHeads" "DVI-2,DVI-3" (see comment about randr above) single section setup helps to change log line RADEON(1): Output DVI-3 has no monitor section into RADEON(1): Output DVI-3 using monitor section Monitor3 but nothing was enough to switch from screen cloning to separate screens. This problem (lack of documentation on radeon driver) is similar to these: Radeon display driver clones monitors while using Xinerama (moderators decision to close that problem was wrong) Ubuntu 12.10 multi-monitor setup isn't working The problem is solvable, because this hardware worked as three headed for me earlier with gentoo/xorg-server-1.3 Xorg -configure creates setup for the first monitor on the first GPU Please don't advise to use fglrx/aticonfig/amdcccle (this goes against my religion beliefs)

  Read the article

• Android: Voice Recording and saving audio

  - by user1320912

  I am working on application that will record the voice of the user and save the file on the SD card and then allow the user to listen to the audio again. I am able to allow the user to record his voice using the RecognizerIntent, but I cant figure out how to save the audio file and allow the user to hear the audio. I would appreciate it if someone could help me out. I have displayed my code below: // Setting up the onClickListener for Audio Button attachVoice = (Button) findViewById(R.id.AttachVoice_questionandanswer); attachVoice.setOnClickListener(new OnClickListener() { public void onClick(View v) { Intent voiceIntent = new Intent(RecognizerIntent.ACTION_RECOGNIZE_SPEECH); voiceIntent.putExtra(RecognizerIntent.EXTRA_LANGUAGE_MODEL, RecognizerIntent.LANGUAGE_MODEL_FREE_FORM); voiceIntent.putExtra(RecognizerIntent.EXTRA_PROMPT, "Please Speak"); startActivityForResult(voiceIntent, VOICE_REQUEST); } }); protected void onActivityResult(int requestCode, int resultCode, Intent data) { if(requestCode == VOICE_REQUEST && resultCode == RESULT_OK){ }

  Read the article

• Testing background audio in the simulator

  - by Cactuar

  I'm experimenting with the new background audio service in iPhone OS 4.0 but I can't get it to work in the simulator. According to this page: iPhone Application Programming Guide: Executing Code in the Background it seems that all I have to do is add the a UIBackgroundModes key with an array containing audio to my Info.plist file and the audio my application plays should automatically continue when I switch to another app. I have done this but the audio still pauses as I switch to another app, when I switch back it continues where it left off. This is the code I'm using to play the sound: NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/audio.mp3", [[NSBundle mainBundle] resourcePath]]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; audioPlayer.numberOfLoops = -1; if (audioPlayer == nil) NSLog(@"%@", [error userInfo]); else [audioPlayer play]; Has anyone gotten this to work? Could it be that it would work on an actual device and it's just a problem with the simulator? I'm a bit hesitant to install 4.0 on my phone since I've heard it's still very buggy. Wish I had another device to use only for development.

  Read the article

• Custom flash mp3 player stopping in the middle of playing audio on windows nt ie6 system

  - by Charlotte Moller

  We have used a custom MP3 flash player for a lot of years on our website without any issues, but recently, a client of ours is reporting that the audio is playing for several seconds and then stopping. When they refresh the page or click play in the player again the audio plays fine. We are puzzled as to what could be causing this issue after this running successfully for our clients for so many years. The client system is Windows NT running IE6. Does anyone have any idea what could cause the audio to behave this way? Could audio drivers or the version of flash cause problems? We do not have flash programmers on our team so we are not even sure where to start looking within the flash code of the player. Any ideas?

  Read the article

• Capture Flash Audio in 4.7 Edge?

  - by emcmanus

  Is there a way to capture plugin (Flash) audio before it gets to the sound card? I'd like to record plugin audio, hopefully without actually playing the sound. Capturing audio at the device level is an absolute last resort, as the application would pick up all system audio rather than just the Webkit plugin. I'm aware of the recent switch back from QTMultimedia; is this possible with phonon? I spent the night looking for some way to access the phonon graph via QWebFrame (or any of the QtWebkit widgets) -- and didn't turn up much. I also started digging through QTWebkit, particularly NPAPI, without success. For reference, I'm using the edge version of 4.7 (6aa50af000f85cc4497749fcf0860c8ed244a60e) This seems to be a fairly challenging problem. Any hints would be greatly appreciated.

  Read the article

• Audio queue start failed

  - by mobapps99

  Hi , i'm developing a project which has both audio streaming and playing audio from file. For audio streaming i'm using AudioStreamer and for playing from file i'm using avaudioplayer. Both streaming and playing works perfectly as long as the app is not interrupted by a phone call or sms. But when a call/sms comes after dismissing the call when i try to restart streaming i'm getting the error "Audio queue start failed" . This happens only when i have used avaudioplayer at least once and after that used streaming. When the avaudioplayer obeject is not created , in this scenario the there is no problem with resuming streaming after dismissing the call. My guess is that some thing is wrong with audioqueue. Help is very much appreciated.......

  Read the article

• Making a DVD video with a still image and PCM 16bit audio with ffmpeg

  - by João

  I'm trying to make a small video with a still image and a sound file playing in the background to pass it to dvdauthor and create a DVD. The command I'm using is this: ffmpeg -loop_input -i image.jpg -qscale 2 -i song.flac -aspect 4:3 -target pal-dvd -acodec pcm_s16le -shortest output.mpg However, the resulting video file doesn't have sound at all (testing it on VLC Player). I don't know if I can't combine "-acodec pcm_s16le" with "-target pal-dvd" to override the later, or if there is something else wrong with the command. If I try without the "-acodec pcm_s16le" parameter the video and audio works, I can even create a DVD ISO with it. However, the audio stays as AC3. I wanted to include with the video the lossless audio, not a compressed one. I suppose the DVD standart allows to have PCM audio in it, am I right?

  Read the article

• How to get the default audio format of a TTS Engine

  - by Itslava

  In Microsoft TTS 5.1 or newer. The SpVoice.AudioOutputStream property says: The AudioOutputStream property gets and sets the current audio stream object used by the voice. Setting the voice's AudioOutputStream property may cause its audio output format to be automatically changed to match the text-to-speech (TTS) engine's preferred audio output format. If the voice's AllowAudioOutputFormatChangesOnNextSet property is True, the format change takes place; if False, the format remains unchanged. In order to set the AudioOutputStream property of a voice to a specific format, its AllowOutputFormatChangesOnNextSet should be False. It means a engine's always has a preferred audio output format. So, how can i get it.. i have not found any interface to get that attribute.

  Read the article

• merge CAF (core audio format,Linear PCM) file into a video(mjpeg) file using ffmpeg (programaticall

  - by Sridhar

  Hello, As the title said, I have a CAF file and I want to insert that file into a video file using ffmpeg(Programmatically ) using C. CAF file has Linear PCM audio format I am successfully able to build the video file and now want to add the audio to the video file from CAF (Core Audio Format) file. Can anyone provide a sample please. Thanks

  Read the article

• Access MP3 audio data independently of ID3 tags?

  - by kyl191

  Hi, this is a 2 part question. First off, is it possible to access the audio data in an MP3 independently of the ID3 tags, and secondly, is there any way to do so using available libraries? I recently consolidated my music collection from 3 computers and ended up with songs which had changed ID3 tags, but the audio data itself was unmodified. Running a search for duplicate files failed because the file changed with the ID3 tag change, but I think it should be possible to identify duplicate files if I just run a deduplication using the audio data for comparison. I know that it's possible to seek to a particular position past the ID3 header in the file, and directly read the data, but was wondering if there's a library that would expose the audio data so I could just extract the data, run a checksum on it, and store the computed result somewhere, then look for identical checksums. (Also, I'd probably have to use some kind of library when you take into account variable length headers.)

  Read the article

• Playing Audio in WPF with repeat on finish

  - by user316323

  Hi I want to create a audio player class but having some problems like when audio is finished it is not repeating. While playing audio some UI controls are not working.

  Read the article

• Live noise-filter on line-in

  - by Damon Gant

  I'm running the following setup: Xbox 360 is hooked up to my (PC) screen via HDMI/DVI converter. Because the Xbox has no dedicated sound output, except for optical S/PIDF, I'm also using the AV/RCA output, namely just the audio, which is connected to an old stereo, which is then connected to my PCs line-in. I'm now experiencing a some of noise. I'm using one of the standard "Realtek High Definition Audio" cards, which doesn't seem to offer this kind of functionality. Is there a software that will playback audio right off a device while running filters on it? It doesn't have to create a device on its own, I just want to listen to it. Here's a sample: http://puu.sh/1suY6

  Read the article

• DTS to AC3 conversion for LG TV using mediatomb DLNA server

  - by prion crawler

  I want to convert a MKV video file containing DTS audio to a stream with AC3 audio. I want to pass this resulting stream to mediatomb's transcoding feature. Mediatomb will transfer the stream via DLNA to a LG TV, which does not support DTS audio. I have tried the VLC command below but the TV does not recognize the stream, and playing the destination stream on PC does not produce sound. vlc -vvv -I dummy INPUT.file --sout \ '#transcode{acodec=ac3,ab=256k,channels=2,threads=4} \ :std{mux=ts,access=file,dst=DEST.file}' The following ffmpeg command give a stream that plays on the TV with sound, but the ffmpeg process gets killed (with signal 15) within 10-15 seconds, and then the TV restarts the playback from the beginning. This goes on in loops. ffmpeg -i INPUT.file -acodec ac3 -ab 384k -vcodec copy \ -vbsf h264_mp4toannexb -f mpegts -y DEST.file I want to have a working DLNA server which transcodes DTS to AC3, any help is appreciated.

  Read the article

• 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.

  Read the article

• connect 2.1 stereo speakers to LG LCD-TV (5500 series)

  - by rMaero

  I bought a pair of speakers for my dad's TV, LG 32LE5500. When I installed them, it just sounded worse than the integrated ones and that's where I realized the subwoofer didn't work at all and both speakers make lower volume than the internal ones. The audio output jack says "H/P" (standing for headphones, and a matching symbol) before buying I checked this output with my phone's headphones and it worked so I figured it would work with a set of speakers since it's a standard audio output. I guess it's literally for headphones and not any other kind of sound players. There is only one other audio output and it is the optical-digital, so I can't use that. Not at least with these speakers.. am I screwed? or is there any workaround?

  Read the article

• Share a USB sound card over a network/bluetooth (Mac & PC)

  - by AlexW

  I've been wondering how I can stream audio to an external Edirol USB sound card, wirelessly, on both Mac and PC. I'm not looking for high quality transmission, just to play mp3s from my Mac laptop to a USB sound card that is attached to two very nice balanced studio reference monitors. Is there any way I can firstly power the sound card box, and secondly, provide with an audio stream along it's USB input. I've looked at the Belkin USB hub, and I have a Time Capsule with the AirPort interface inside. These things seem to do vaguely what I want but when it comes to audio, the specifications are less clear. Any suggestions very welcome.

  Read the article

• No sound through DisplayPort

  - by Chris Koknat

  I'm trying to connect my Elitebook 8440p laptop to my Samsung HDTV. The laptop does not have a HDMI connection, but it does have DisplayPort. I bought a DisplayPort-to-HDMI adapter here http://www.amazon.com/gp/product/B002CSRFD8/ref=oss_product, and connected it with a 3' HDMI cable. The video shows up fine, but there is no audio. DisplayPort, HDMI, and the adapter all support audio. I contacted HP tech support, who told me to update my sound drivers. I installed the driver and rebooted. Supposedly, I should see a "HD Audio" tab. No luck, even after installing the driver again and rebooting. HP closed the case. I'm using XP Pro.

  Read the article

• Speakers will not work after I use USB Headset

  - by Josh K

  I am trying to configure my SteelSeries Siberia V2 Frost USB Headset to work with my 2.0 speakers using a jack. My goal is to find a easy way (no restart) to switch playback from my headset and my speakers and vice-versa. If i plug my headset in and make it the default device then restart my application/web page then the sounds works out of headset. If I switch the default to my speakers and restart apps/web pages then sound does not play. I know my speakers are on because if I configure them through windows and test, the sounds play, and sounds also play when I test it through my audio manager. Even if I unplug my headset, I still cannot get sound out of my speakers unless I restart My audio manager is RealTek HD Audio Manager, Windows 7 x64. I have tried the speaker back, usb front. speaker front, usb in front port. I have not tried speaker back, usb back.

  Read the article

< Previous Page | 40 41 42 43 44 45 46 47 48 49 50 51  | Next Page >