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  • How to stream audio from ASP.NET MVC controller when it's still encoding?

    - by kyrisu
    Background I have wave files on my server that I want to stream. Because of the size I want to encode them to mp3. I've tried to use FileStreamResult - but it doesn't work because as soon as program leaves the controller stream is closed and I get - "Cannot access a closed stream" FileContentResult - but it's not a stream and the user would need to wait for encoding to finish Question Is there a way to stream audio from the controller while it's still encoding?

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  • convert decrypted .vobs to .avi with ffmpeg on ubuntu

    - by Arcath
    I have a .vob file that has bee ripped from a dvd, when I watch the .vob its very good quality video and 5.1 english audio but when I use ffmpeg it has rubbish video and mono french audio. That was using this command: ffmpeg -i /samba/ripping/vobs/12161840#2.vob -f avi /samba/ripping/avis/test.avi I've tried a few different variations on that but it never comes back with anything good just bigger files with bad video and incorrect sound. I know the videos good and the correct audio streams exist so how do I select a 5.1 track and get good video? ffmpeg gives the .vob details as: Input #0, mpeg, from '/samba/ripping/vobs/12161840#2.vob': Duration: 00:42:05.56, start: 0.287267, bitrate: 5738 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 8436 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, mono, s16, 192 kb/s Output #0, avi, to '/samba/ripping/avis/test.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.3 -> #0.1

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  • Is it possible to broadcast audio to shoutcast / icecast / other server? from flash player?

    - by Jeffrey
    I am trying to create a flash client that can stream audio to an online radio server. Theoretically a user could enter the server info / login, and then connect and start sending data to the server which could then be broadcasted and listened to by other clients. I don't think this would be very hard, but am unsure about what data formats to use and what is the best server for the job. I'd like to be able to use one of the most popular radio servers like shoutCast. Any ideas? Thanks in advance.

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  • Linux application that bundles multiple incoming audio and video streams into one container file?

    - by StackedCrooked
    I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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  • VST plugin : using FFT on audio input buffer with arbitrary size, how ?

    - by Led
    I'm getting interested in programming a VST plugin, and I have a basic knowledge of audio dsp's and FFT's. I'd like to use VST.Net, and I'm wondering how to implement an FFT-based effect. The process-code looks like public override void Process(VstAudioBuffer[] inChannels, VstAudioBuffer[] outChannels) If I'm correct, normally the FFT would be applied on the input, some processing would be done on the FFT'd data, and then an inverse-FFT would create the processed soundbuffer. But since the FFT works on a specified buffersize that will most probably be different then the (arbitrary) amount of input/output-samples, how would you handle this ?

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  • Successfully concatenating multiple videos

    - by wiseguydigital
    My mission is to create videos out of old web slideshows. To start with I have jpegs and audio files that worked as Flash slideshows in an old system, structured such as this: Audio structure my_audio_1.mp3 (this file is a 3 second mp3 of silence) my_audio_2.mp3 my_audio_3.mp3 my_audio_4 etc... roughly 30 mp3s per slideshow Image structure my_image_1.jpg (this acts as the opening slide) my_image_2.jpg my_image_3.jpg my_image_4. etc... roughly 30 images per slideshow. As there are almost 100 slideshows that must be converted to video, I have created a web-based interface using PHP to automate the process, that sits on a local system and attempts to combine the files using shell_exec(). The process uses the following workflow: Loop through each slide and make an avi or mpeg. So for instance my_mini_video_2.avi would be a video that consists of my_image_2.jpg and has a soundtrack of my_audio_2.mp3. This slide would last the length of my_audio_2.mp3. Join / stitch / concat all of the mini videos to create the final video (Using a combination of cat and either mencoder or ffmpeg (I have also tried avimerge but to no avail). Transcode the new 'master' video to various formats such as flv etc. I thought this would be simple and have been close on many occasions but it still won't work. I can't get past stage 2 as I can't get a perfect 'master' video. I have now experimented with Mencoder, FFMpeg and seem to have been through every combination I can think of. The problem is that the audio and visuals never sync, no matter what I try. Also, I have even tried created audio-less mini videos, joining the MP3s into one long MP3 using both cat and mp3wrap and then assigning the new long MP3 as the audio track, but this always produces either a very short file or a badly slowed down file and makes the female voiceover sound like a male boxer!!! There appears to be no problems at all with the original files. Does anybody have any experience in producing a video successfully from the same kind of starting point? Or any ideas on what I may be doing wrong? As an example: If I create silent mini-videos, and stitch them together into 'temp-master.mpg' and then join the MP3s together into single MP3 called 'temp-master-audio.mp3', the audio file's duration is 09:10 and the video file's duration is 08:35. They should be the same and the audio will seem sloooow. I haven't posted code as I have written lots and lots of combinations.

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  • Reproduce PIPE functionality in IronPython

    - by Muppet Geoff
    Hi, I am hoping some genious out there can help me out with this... I am using sox to merge and resample a group of WAV files, and pipe the output directly to the input of NeroAACEnc for encoding to AAC format. I originally ran the process in a script, which included: sox.exe d:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav - | neroAacEnc.exe -q 0.5 -if - -of test.m4a This worked as expected. The '-' in the comand line translates as 'Pipe/redirect input/output (stdin/stdout)' - So Sox pipes to stdout, and NeroAACEnc reads from stdin, the | joins them together. I then migrated the whole solution to Python, and the equivalent command became: from subprocess import call, Popen, PIPE runwav = Popen(['sox.exe', 'd:\audio\1.wav', 'd:\audio\2.wav', 'd:\audio\3.wav', '-c', '1', '-r', '22050', '-t', 'wav', '-'], shell=False, stdout=PIPE) runm4b = call(['neroAacEnc.exe', '-q', '0.5', '-if', '-', '-of', 'test.m4a'], shell=False, stdin=runwav.stdout) This also worked like a charm, exactly as expected. Slightly more convoluted, but hey :) Well now I have to move it to IronPython, and the Subprocess module isn't available (the partial implementation that is, doesn't have Popen/PIPE support - plus it seems silly to add a custom library when there is probably a native alternative). I should mention here, that I opted for IronPython over C#, because I am comfortable with Python now - however, there is a chance of moving it again later to C# native, and I am using IronPython to ease myself into it :) I have no C# or .net experience. So far I have the following equivalent, that sets up the 2 processes: from System.Diagnostics import Process wav = Process() wav.StartInfo.UseShellExecute = False wav.StartInfo.RedirectStandardOutput = True wav.StartInfo.FileName = 'sox.exe' wav.StartInfo.Arguments = 'd:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav -' wav.Start() m4b = Process() m4b.StartInfo.UseShellExecute = False m4b.StartInfo.RedirectStandardInput = True m4b.StartInfo.FileName = 'neroAacEnc.exe' m4b.StartInfo.Arguments = '-q 0.5 -if - -of test.m4a' m4b.Start() I know that these 2 processes start (I can see Nero and Sox in the task manager) but what I can't figure out (for the life of me) is how to string the two output/input streams together, as with the previous two solutions. I have searched and searched, so I thought I'd ask! If anyone knows either: How to join the two streams with the same net result as the Python and Commandline versions; or A better way to acheive what I am trying to do. I would be extremely grateful! Many thanks in advance, Geoff P.S. A code sample based off the above would be awesome :) or a specific code example of a similar process that I can easily translate... this has broked my brayne.

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  • How to use infinit live streams with JAVE library? (Java, ffmpeg)

    - by Ole Jak
    So I want to use JAVE to save mp3 radio stream to my File system. I have this code for file saving but what shall I do to save a stream (stop on timer for ex) File source = new File("source.wav"); File target = new File("target.mp3"); AudioAttributes audio = new AudioAttributes(); audio.setCodec("libmp3lame"); audio.setBitRate(new Integer(128000)); audio.setChannels(new Integer(2)); audio.setSamplingRate(new Integer(44100)); EncodingAttributes attrs = new EncodingAttributes(); attrs.setFormat("mp3"); attrs.setAudioAttributes(audio); Encoder encoder = new Encoder(); encoder.encode(source, target, attrs);

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  • Trouble installing ubuntu server on virtualbox (osx)

    - by audio.zoom
    Hello all, I'm trying to install lucid lynx 10.04.2 server on a virtualbox on snow leopard. I have 2 server iso files freshly downloaded one i386 and one 64bit. When I try to start the virtual machine with either one set to be the cd drive I'm getting the same error: Failed to open a session for the virtual machine Ub. Failed to load VMMR0.r0 (VERR_SUPLIB_OWNER_NOT_ROOT). Unknown error creating VM (VERR_SUPLIB_OWNER_NOT_ROOT). Couldn't find anything on it on google so I'm trying to see if anyone else has dealt with this issue. Thanks much in advance! edit: just downloaded the 32bit desktop edition to same avail edit2: ran Disk Utility' replair permissions then restarted. New error VERR_SUPLIB_WORLD_WRITABLE (instead of VERR_SUPLIB_OWNER_NOT_ROOT)

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  • Does any economically-feasible publicly available software compare audio files to determine if they are dupes?

    - by drachenstern
    In the vein of this question http://unix.stackexchange.com/questions/3037/is-there-an-easy-way-to-replace-duplicate-files-with-hardlinks is there any software that will automatically parse a library of my songs and find the ones that really are duplicates that one can be eliminated? Here's an example: My brother used to be a huge fan of remixing CDs. He would take all of his favorite tracks and put them on one. Then he would use my computer to read them in. So now I have like 6 copies of Californication on my HDD, and they're all a few bytes difference overall. I have hundreds of songs in my library like this. I want to trim them down to having uniques. They don't all have correct ID3 tags, so figuring out that Untitled(74).mp3 is the same as californication.mp3 is the same as whowrotethis.mp3 is tricky. I do NOT want to consider a concert album and a studio album rip to be the same (if I just did artist/title matching I would end up with this scenario, which doesn't work for me). I use Windows (pick your platform) and will be getting an OSX box later in the year. I'll run Linux if that's what it takes to get it organized. I have unprotected AAC and mp3 files. Bonus points for messing with WAV or MIDI and bonus points for converting from those into MP3 (I can always use Audacity and LAME to convert later if I know they match or to convert ahead of time if that will make things easier). Are there any suggestions, or do I need to goto Programmers or SO and build a list of requirements for comparing these things and write the software myself?

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  • How does Amarok rip Audio CDs (in Ubuntu Lucid)?

    - by Hanno Fietz
    I'm in the process of moving my CD collection into my Amarok library. Mostly, it works great. Sometimes however, the process just hangs forever. The problem seems to occur at random (i. e. often, but not always at the same disk/track) and the consequences range from none (successful after cancel/retry) to Amarok's internal db becoming completely messed up. I would like to investigate and file a proper bug report or find a fix / workaround, but I don't understand how Amarok does the ripping. When all is working, there's a lame process encoding to a temporary file, which appears in my collection once it's finished. When the process hangs, that lame command is still there, but waiting forever for data on stdin, which seems to come from a third process. That seems to be kio_audiocd, but I don't know whether that's correct and what it's supposed to do. What's going on?

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  • how to stream audio and video files, but use any media player on Windows (without using Windows file

    - by RamyenHead
    I want to access and play media files on machine S (Windows XP) from machine C (Windows XP). Using Windows File Sharing ("share this folder" stuff), if it works, I would share the folder containing media files on machine S, and I would be able to play media files, sitting in front of C, using any media player I want. Windows somehow ensures that the remote files behave like local files. But Windows file sharing won't work for me, is there any alternative? If two machines were both Linux, I would install an SSH server on S and use Nautilus from C to access and play media files. The reason why I can't use Windows file sharing is, my campus use two different subnets, I have S and C on different subnets and it seems that the firewall governing the whole network in campus doesn't allow file sharing between different subnets. I tried changing Windows Firewall settings on S to allow C in, it still wouldn't work, so it must be the other firewall.

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  • How to split audio into multiple channels from optical S/PDIF or 1/8"?

    - by Josh M.
    I have a motherboard which has an optical S/PDIF output or 1/8". I'd like to "split" that signal into the appropriate channels so that I can then connect that to the wires behind my car's headunit which, in turn, run to the amp. The factory Bose amp just takes a single connector with a million wires running out of it, so that's why I would need to separate the signal into separate channels. On the other end there are four RCA connectors: front left, front right, rear left, rear right. The sub-woofer signal does not require an additional connection. Edit: Revised to include S/PDIF or 1/8".

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  • 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?

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  • Hardware Mediaplayer display

    - by Eric Audio
    I'm looking for a keyboard or just a little display to attach on my keyboard or something like that, what will show me the music tracks i'm playing in windowsmedia player, itunes, etc. I did some research and the only thing I found are gaming keyboards, but i'm not shure if these show my music tracks. So my question: Does somebody knows a keyboard who show the music tracks or just a little display? Bye, Eric

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  • How can I get Windows 7 to switch audio from a monitor (with built-in speakers) to headphones when t

    - by tnorthcutt
    I have an HP dv5t laptop running Windows 7 64 bit with an Acer H235H monitor connected to it via an HDMI cable. The monitor has built-in speakers, which are a huge improvement over the laptop's speakers. However, when I want to use headphones, right now, I have to connect them to the laptop, then right-click the sound icon in the task bar, select Playback Devices, right click the monitor, and disable it. Is there any way to get Windows 7 to automatically switch the output to the headphones when they're plugged in? That's the behavior that happens without the monitor attached (i.e. it will switch from the laptop speakers to headphones when headphones are plugged in). I have the same issue with a Sony Vaio laptop running Windows 7 64-bit and an identical monitor, for reference.

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  • Android app hanging, sometimes until Force Close / Wait dialog appears

    - by fredley
    I'm making an app that records uncompressed (wav format) audio. I'm using this class to actually record the audio. Currently, my application records fine (I can play the file), however when I click the button to stop the recording, the app hangs for 10 seconds or so, with no log output or any signs of life. Finally it comes round, dumps a load of errors into the log, updates the UI etc. I'm using AsyncTasks to try and avoid this kind of thing but it's not working. Here's my code: //Called on clicks of the record button. rar is the instance of RehearsalAudioRecorder private OnClickListener RecordListener = new OnClickListener(){ @Override public void onClick(View v) { Log.d("Record","Click"); if (recording){ new stopRecordingTask().execute(rar,null,null); startStop.setText("Record"); statusBar.setText("Recording Finished, ready to Encode"); }else{ recording = true; new startRecordingTask().execute(rar,null,null); startStop.setText("Stop"); statusBar.setText("Recording Started"); } } }; private class startRecordingTask extends AsyncTask<RehearsalAudioRecorder,Void,Void>{ @Override protected Void doInBackground(RehearsalAudioRecorder... rs) { RehearsalAudioRecorder r = rs[0]; r.setOutputFile("/sdcard/rarOut.wav"); r.prepare(); r.start(); return null; } } private class stopRecordingTask extends AsyncTask<RehearsalAudioRecorder,Void,Void>{ @Override protected Void doInBackground(RehearsalAudioRecorder... rs) { RehearsalAudioRecorder r = rs[0]; r.stop(); r.reset(); return null; } } In Logcat, I always get output like this, which has me stumped. I have no idea what's causing it (I'm logging the RehearsalAudioRecorder class, and it's being started/stopped correctly by the button clicks. This output occurs after the log output for the button click and correct stop() method call) 12-19 11:59:11.172: ERROR/AudioRecord-JNI(22662): Unable to retrieve AudioRecord object, can't record 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): Error occured in updateListener, recording is aborted 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): stop() called on illegal state: STOPPED 12-19 11:59:11.172: ERROR/AudioRecord-JNI(22662): Unable to retrieve AudioRecord object, can't record 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): Error occured in updateListener, recording is aborted 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): stop() called on illegal state: ERROR 12-19 11:59:11.172: ERROR/AudioRecord-JNI(22662): Unable to retrieve AudioRecord object, can't record 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): Error occured in updateListener, recording is aborted 12-19 11:59:11.172: ERROR/uk.ac.cam.tfmw2.steg.RehearsalAudioRecorder(22662): stop() called on illegal state: ERROR ... 10 or more times I've been fiddling with this all day and I'm not getting anywhere, any input would be greatly appreciated. Update I've replace the AsyncTasks with Threads, still doesn't work, the app completely hangs when I click record, despite the fact the Log indicates there's nothing going on in the main thread. Still completely stumped.

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  • Converting from mp4 to Xvid avi using avconv?

    - by Ricardo Gladwell
    I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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  • which default.list should i modify for default applications and what are the differences between the 2

    - by damien
    I would like to add miro to the default application GUI in system settings/default applications.I added ;miro.desktopnext to all rhythmbox.desktop entries eventually discovering if it was not added to audio/x-vorbis+ogg=rhythmbox.desktop as audio/x-vorbis+ogg=rhythmbox.desktop;miro.desktop it would not appear in the system settings/default applications drop down list for audio. I can find default.list in either /etc/gnome/defaults.list or /usr/share/applications/defaults.list modifying either gives me the same results.What is the difference and which is the correct list to modify?

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  • map based on specific type of stream

    - by Steven Penny
    Consider the following file Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'infile.mp4': Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1038 Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 386 kb/s Stream #0:2(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 124 kb/s I would like to run a command that uses the video and only the stereo audio. With this particular file I could just run ffmpeg -i infile.mp4 -c copy -map v -map :2 outfile.mp4 However this will not work if the audio streams are switched. How can I tell FFmpeg to use the 2 channel audio only?

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  • I got my z-5 Logitech speakers to work, but whenever I restart, I have to reconfigure them

    - by The Bill
    This is the content of my alsa-base.conf file (for some reason, the entries preceded by # are bolded--anyway): autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-usb-audio index=0 options snd-hda-intel index=1 Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 options snd-usb-audio index=-2 options snd-usb-audio index=0 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-hda-intel index=1 I deleted a line that said something like "#Keep usb-audio from being loaded as first soundcard" and that made the speakers work for the first time (before this, they never showed up). I also added the last four lines. Anyway, what can I add to this so that I don't have to reconfigure them each time I restart? Currently, I have to open Sound Settings, then under the hardware tab, select Analog Stereo Output, and then unplug my USB speakers and plug them back in. This makes them pop up so that I can see them. Otherwise, it will not show my Z-5 speakers as a device that can be configured.

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  • No sound after installing a new Graphics Card

    - by Dan
    I've just upgraded my graphics card to an Asus Geforce 210 and now my system has no sound. I've ran Update Manager and the Additional Drivers utility which installed the latest Nvida driver. The graphics card is connected to my TV via a DVI-to-HDMI (DVI at the PC end) cable for the visual connection, and an audio jack from my onboard soundcard for my audio connection. Any ideas on how to resolve this? I ran this command ubuntu-bug audio And it outputted this: You seem to have configured PulseAudio to use the "pci-0000_05_00.1" card, while you want output from "NFORCE - NVidia CK804". I've tired a bit of messing about with the audio settings but can't get anything to work.

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