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  • Seeking through a streamed MP3 file with HTML5 <audio> tag

    - by Kyle Slattery
    Hopefully someone can help me out with this. I'm playing around with a node.js server that streams audio to a client, and I want to create an HTML5 player. Right now, I'm streaming the code from node using chunked encoding, and if you go directly to the URL, it works great. What I'd like to do is embed this using the HTML5 <audio> tag, like so: <audio src="http://server/stream?file=123"> where /stream is the endpoint for the node server to stream the MP3. The HTML5 player loads fine in Safari and Chrome, but it doesn't allow me to seek, and Safari even says it's a "Live Broadcast". In the headers of /stream, I include the file size and file type, and the response gets ended properly. Any thoughts on how I could get around this? I certainly could just send the whole file at once, but then the player would wait until the whole thing is downloaded--I'd rather stream it.

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  • How to get musicbrainz track information from audio file

    - by Baki
    Can anyone tell me how to get track information from the MusicBrainz database from an audio file (mp3, wav, wma, ogg, etc...) using audio fingerprinting. I'm using MusicBrainz Sharp library, but any other library is ok. I've seen that you must use the libofa library, that you can't use MusicBrainz Sharp to get puid from the audio file, but I can't figure out how to use libofa with C#. Please show some examples and code snippets to help me, because I can't find them anywhere. Thanks in advance!

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  • PHP Video Editing and Streaming

    - by Gaurav Padia
    Hi, I am developing online video streaming website on PHP. I need two functionalities: Need to add title/text at bottom of the video dynamically. Need to add background music to video dynamically. Is it possible with PHP or any available open source library? Can anyone guide me or provide links to this type of library ? Thanks.

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  • How to Use Steam In-Home Streaming

    - by Chris Hoffman
    Steam’s In-Home Streaming is now available to everyone, allowing you to stream PC games from one PC to another PC on the same local network. Use your gaming PC to power your laptops and home theater system. This feature doesn’t allow you to stream games over the Internet, only the same local network. Even if you tricked Steam, you probably wouldn’t get good streaming performance over the Internet. Why Stream? When you use Steam In-Home streaming, one PC sends its video and audio to another PC. The other PC views the video and audio like it’s watching a movie, sending back mouse, keyboard, and controller input to the other PC. This allows you to have a fast gaming PC power your gaming experience on slower PCs. For example, you could play graphically demanding games on a laptop in another room of your house, even if that laptop has slower integrated graphics. You could connect a slower PC to your television and use your gaming PC without hauling it into a different room in your house. Streaming also enables cross-platform compatibility. You could have a Windows gaming PC and stream games to a Mac or Linux system. This will be Valve’s official solution for compatibility with old Windows-only games on the Linux (Steam OS) Steam Machines arriving later this year. NVIDIA offers their own game streaming solution, but it requires certain NVIDIA graphics hardware and can only stream to an NVIDIA Shield device. How to Get Started In-Home Streaming is simple to use and doesn’t require any complex configuration — or any configuration, really. First, log into the Steam program on a Windows PC. This should ideally be a powerful gaming PC with a powerful CPU and fast graphics hardware. Install the games you want to stream if you haven’t already — you’ll be streaming from your PC, not from Valve’s servers. (Valve will eventually allow you to stream games from Mac OS X, Linux, and Steam OS systems, but that feature isn’t yet available. You can still stream games to these other operating systems.) Next, log into Steam on another computer on the same network with the same Steam username. Both computers have to be on the same subnet of the same local network. You’ll see the games installed on your other PC in the Steam client’s library. Click the Stream button to start streaming a game from your other PC. The game will launch on your host PC, and it will send its audio and video to the PC in front of you. Your input on the client will be sent back to the server. Be sure to update Steam on both computers if you don’t see this feature. Use the Steam > Check for Updates option within Steam and install the latest update. Updating to the latest graphics drivers for your computer’s hardware is always a good idea, too. Improving Performance Here’s what Valve recommends for good streaming performance: Host PC: A quad-core CPU for the computer running the game, minimum. The computer needs enough processor power to run the game, compress the video and audio, and send it over the network with low latency. Streaming Client: A GPU that supports hardware-accelerated H.264 decoding on the client PC. This hardware is included on all recent laptops and PCs. Ifyou have an older PC or netbook, it may not be able to decode the video stream quickly enough. Network Hardware: A wired network connection is ideal. You may have success with wireless N or AC networks with good signals, but this isn’t guaranteed. Game Settings: While streaming a game, visit the game’s setting screen and lower the resolution or turn off VSync to speed things up. In-Home Steaming Settings: On the host PC, click Steam > Settings and select In-Home Streaming to view the In-Home Streaming settings. You can modify your streaming settings to improve performance and reduce latency. Feel free to experiment with the options here and see how they affect performance — they should be self-explanatory. Check Valve’s In-Home Streaming documentation for troubleshooting information. You can also try streaming non-Steam games. Click Games > Add a Non-Steam Game to My Library on your host PC and add a PC game you have installed elsewhere on your system. You can then try streaming it from your client PC. Valve says this “may work but is not officially supported.” Image Credit: Robert Couse-Baker on Flickr, Milestoned on Flickr

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  • What are my audio devices?

    - by hellocatfood
    I'm attempting to use easycap to record from my camcorder but I'm having a slight problem. Using their test script I'm able to get audio and video. I've noticed that in the script on line 159 it makes a call to "DEV_ADUIO", which is reported as being "plughw:2,0". Exactly what is this device? Is it located in /dev/ somewhere? I've done "ls /dev/" and I can't find anything that would suggest an audio device

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  • Streaming audio not working in Android

    - by user320293
    Hi, I'm sure that this question has been asked before but I've been unable to find a solid answer. I'm trying to load a streaming audio from a server. Its a audio/aac file http://3363.live.streamtheworld.com:80/CHUMFMAACCMP3 The code that I'm using is private void playAudio(String str) { try { final String path = str; if (path == null || path.length() == 0) { Toast.makeText(RadioPlayer.this, "File URL/path is empty", Toast.LENGTH_LONG).show(); } else { // If the path has not changed, just start the media player MediaPlayer mp = new MediaPlayer(); mp.setAudioStreamType(AudioManager.STREAM_MUSIC); try{ mp.setDataSource(getDataSource(path)); mp.prepareAsync(); mp.start(); }catch(IOException e){ Log.i("ONCREATE IOEXCEPTION", e.getMessage()); }catch(Exception e){ Log.i("ONCREATE EXCEPTION", e.getMessage()); } } } catch (Exception e) { Log.e("RPLAYER EXCEPTION", "error: " + e.getMessage(), e); } } private String getDataSource(String path) throws IOException { if (!URLUtil.isNetworkUrl(path)) { return path; } else { URL url = new URL(path); URLConnection cn = url.openConnection(); cn.connect(); InputStream stream = cn.getInputStream(); if (stream == null) throw new RuntimeException("stream is null"); File temp = File.createTempFile("mediaplayertmp", ".dat"); temp.deleteOnExit(); String tempPath = temp.getAbsolutePath(); FileOutputStream out = new FileOutputStream(temp); byte buf[] = new byte[128]; do { int numread = stream.read(buf); if (numread <= 0) break; out.write(buf, 0, numread); } while (true); try { stream.close(); } catch (IOException ex) { Log.e("RPLAYER IOEXCEPTION", "error: " + ex.getMessage(), ex); } return tempPath; } } Is this the correct implementation? I'm not sure where I'm going wrong. Can someone please please help me on this.

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  • How to find an audio file's length (in seconds)

    - by mIL3S
    Hi all! (Objective C) Just using simple AudioServicesPlaySystemSoundID and its counterparts, but I can't find in the documentation if there is already a way to find the length of an audio file. I know there is AudioServicesGetPropertyInfo, but that seems to return a byte-buffer - do audio files embed their length in themselves and I can just extract it with this? Or is there perhaps a formula based on bit-rate * fileSize to convert to length-of-time? mIL3S www.milkdrinkingcow.com

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  • Audio not working

    - by user3215
    Anybody could help me in troubleshooting audio problem on ubutnu 9.04 desktop edition?. For some reason I've to keep this os not upgraded and I'm trying to fix the audio problem on this for months. It works well on upgraded version(9.10,10.04) but not on jaunty. aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod | grep snd: snd_hda_intel 436148 7 snd_pcm_oss 46336 0 snd_mixer_oss 22656 1 snd_pcm_oss snd_pcm 83076 4 snd_hda_intel,snd_pcm_oss snd_seq_dummy 10756 0 snd_seq_oss 37760 0 snd_seq_midi 14336 0 snd_rawmidi 29696 1 snd_seq_midi snd_seq_midi_event 15104 2 snd_seq_oss,snd_seq_midi snd_seq 56880 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 29704 2 snd_pcm,snd_seq snd_seq_device 14988 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 62756 21 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15200 1 snd snd_page_alloc 16904 2 snd_hda_intel,snd_pcm cat /proc/asound/cards: 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xe1280000 irq 16 cat /proc/asound/version: Advanced Linux Sound Architecture Driver Version 1.0.18rc3. vim /etc/modules: # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with "#" are ignored. lp Audio Settings:

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  • SEHException throw using Microsoft XACT Audio Framework (XACT3)

    - by Sweta Dwivedi
    I have been developing a game using Kinect + XNA and using Microsoft Audio Creation tool (XACT3) for managing my sound files and music, however in the code an SEHException is thrown whenever it tries to get the wave file from the wave Bank . . Sometimes the code works magically and all of a sudden it will start throwing this exception randomly ..I need a help on solving this exception /*Declaring Audio Engine for music*/ AudioEngine engine; SoundBank soundBank; WaveBank waveBank; Cue cue; /*Declaring Audio engine for sound effects*/ AudioEngine engine1; SoundBank soundbank; WaveBank wavebank; Cue effect; engine = new AudioEngine(@"Content\therapy.xgs"); soundBank = new SoundBank(engine, @"Content\Sound Bank.xsb"); **waveBank = new WaveBank(engine, @"Content\Wave Bank.xwb");** cue = null; engine1 = new AudioEngine(@"Content\Music_Manager\Sound_effects.xgs"); soundbank = new SoundBank(engine1, @"Content\Music_Manager\Sound1.xsb"); **wavebank = new WaveBank(engine1, @"Content\Music_Manager\Wave1.xwb");** effect = null; cue = soundBank.GetCue("hypnotizing"); cue.Play();

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  • Android 3.2 HTTP streaming

    - by user1281053
    I'm trying to create an app to stream live TV. Currently the problem I'm facing is that after say 10 minutes of playing, the video will freeze but the audio will carry on. This is on a 1.3mbps stream. I also have lower streams, such as a 384kbps stream, that might last an hour or so, but will still do the same. I've tested this with a local video, that is high quality (file size is 2.3gb) and that has no lag and doesn't freeze at all, so it must be something to do with the way HLS is streamed to android. Does anyone have any idea on how to solve this problem? Thanks

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  • Ubuntu one, music streaming

    - by iknowshall
    I'm running Ubuntu 11.10, and just signed up for Ubuntu One and Ubuntu One Music. Generally file sync is working fine, but music has been a complete failure so far. Here's what I'm looking at: After 24 hours, not a single mp3 or ogg file from my laptop has sync'd to the Cloud Music folder. I have 4 gigs of data used, which is definitely not enough to include music files. That's about the right amount for my docs and photos. With music, it should be more like 13 gigs. No music shows up on the Ubuntu One Music app on my phone, nor on the web view of my Ubuntu One Cloud folder I followed all the online instructions I could find. Basically, on my laptop, right click on the Music folder and select "Ubuntu One Synchronize this folder" The Music folder now has a green check mark on it indicating its sync'd The Ubuntu One app on my laptop says "File Sync is up to date" I did provide credit card information, and on the Ubuntu One website "Music Streaming" is listed as one of my services. So what am I doing wrong? Just read this post as I'm experiencing the same problem, the answer for this post says that due to the large number of files trying to be uploaded, that it could take up to a week. However I am, as above also being told that "File sync is up to date" Can somebody confirm that this is a bug and that I just need to wait for File sync to actually be up to date? Cheers

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  • Flash video streaming choppy for Chrome, alright for Firefox

    - by Ben
    I'm using Ubuntu 12.04.1, Chrome 21 and Firefox 15. Flash player 11.2 has been installed, and I've just started using Ubuntu... yesterday. And I'm using a Lenovo T61. The problem is that it doesn't matter if it's youtube or vimeo or some other flash player, it streams fine on chrome but every 30 seconds or so, there is a pause in video playback (with audio continuing) before it catches up, skipping quite a few seconds of video. It works perfectly fine in Firefox, and I've tried disabling PepperFlash/libpepflashplayer.so in Chrome but it doesn't seem to affect the performance. Anyone know how to work around this? It's more a problem of convenience because I don't like the idea of having to switch between Chrome and Firefox just to watch videos.

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  • Apache DVB http video Streaming bandwidth or priority problem

    - by igino manfre'
    I'm streaming few precompressed DVB videos from cloud. The streams are generated from VLC on "impossible" ports (such as 64085, 64086 etc) reverse proxed by Apache on port 80 and 8080. All the generated streams are listed in "http://95.110.164.61/indexv.html". From an ADSL connection with enough downlink bandwidth, recalling the stream generated by VLC (such as "http://95.110.164.61:64087/mpg2_6.4") it flows fluently. Recalling the same stream proxed by Apache ("http://95.110.164.61/mpg2_6.4") the stream stops and goes. The only situation in which the Apache proxed streams flow regularly is from a site connected through 64 Mbps warranted bandwith with RTT to the server less than 10 mseconds. Please note that streams below 2 Mbps are fluently proxed. The system is a single core xeon with windows 2008 R2 on 4 GB of RAM with 1 Gbps of network bandwidth. The drain of computational and bandwidth resources is negligeable, the RAM usage always lower than 50%. On the system I run many VLC streamers. Any of them drains a variable amount of RAM (from about 25 to 70 MB). On the contrary the couple of httpd.exe processes drain no more than 7 MB. Using Wireshark (on the server) I see that VLC directy send to the client much more packets than Apache, and the stream is framgmented on many frames. I'm not a programmer, a newby of Apache. Can anyone please address me to a specific portion of the Apache's huge documentation? Thank you. igino

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  • Synced audio ouput on multiple machines? VLC? hardware solutions?

    - by zimmer62
    I'm wondering if there is any software or hardware solutions to synced audio or audio and video across multiple computers or devices on a network. I've seen Sonos, and it might be a good solution, but it's also a very expensive solution. I'd like to be able to play something with realtime audio output on one PC, but hear it on speakers throughout the house, being it the home theater receiver, or another computer in another room. I saw a solution using the apple iport express, but the latency was unacceptable for anything other than just music. I'd like to avoid running audio wires with baluns to a bunch of amplifiers scattered all over the place when I have cat5 run everywhere. Is anyone familiar with using this kind of process for whole home audio? The latency is a big deal for me, if I've got video attached to the sound (e.g. watching a hockey game)

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  • C#: Streaming an Audio file from a Server to a Client

    - by Andreas Grech
    I am currently writing an application that will allow a user to install some form of an application (maybe a Windows Service) that will open a port on it's PC and given a particular destination on the hard disk, will then be able to stream mp3 files. I will then have another application that will connect to the server (being the user's pc) and be able to browse the hosted data by connecting to that PC (remotely ofcourse) given the port, and stream mp3 files from the server to the application I have found some tutorials online but most of them are about File Servers in C# and they download allow you to download a whole file. What I want is to stream an mp3 file so that it starts playing when a certain number of bytes are download (ie, whilst it is being buffered) How do I go about in accomplishing such a task? What I need to know specifically is how to write this application (that I will turn into a Windows Service later on) that will listen on a specified port a stream files, so that I can then access the files by something of the sort: http://<serverip>:65000/acdc/wholelottarosie.mp3 and hopefully be able to stream that file in a WPF MediaPlayer. [Update] I was following this tutorial about building a file server and sending the file from the server to the client. Is what I have to do something of the sort? [Update] Currently reading this post: Play Audio from a Stream using C# and I think it looks very promising as to how I can play streamed files; but I still don't know how I can actually stream the files from the server.

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  • How can I stream audio signals from various devices/computers to my home server?

    - by Breakthrough
    I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

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  • Physics-based dynamic audio generation in games

    - by alexc
    I wonder if it is possible to generate audio dynamically without any (!) audio assets, using pure mathematics/physics and some input values like material properties and spatial distribution of content in scene space. What I have in mind is something like a scene, with concrete floor, wooden table and glass on it. Now let's assume force pushes the glass towards the edge of table and then the glass falls onto the floor and shatters. The near-realistic glass destruction itself would be possible using voxels and good physics engine, but what about the sound the glass makes while shattering? I believe there is a way to generate that sound, because physics of sound is fairly known these days, but how computationaly costy that would be? Consumer hardware or supercomputers? Do any of you know some good resources/videos of such an experiment?

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  • Playing part of a sfx audio file in HTML5 using WebAudio

    - by Matthew James Davis
    I have compiled all of my sound effects into one sequenced .ogg file. I have the start and stop times for each sound effect. How do I play the individual effects? That is, how do I play part of an audio file. More specificially, I've created a dictionary { 'sword_hit': { src: 'sfx.ogg', start: 265, // ms length: 212 // ms } } that my play_sound() function can use to look up 'sword_hit' and play the correct audio file at the correct start time for the correct duration. I simply need to know how to tell the WebAudio API to start playing at start ms and only play for length ms.

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  • Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

    - by Ole Jak
    I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

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  • mplayer (mplayerhq.hu) repeats ending audio frames

    - by kamikatze
    mplayer (from mplayerhq.hu) on windows repeats the last few audio frames upon exit. When the video ends, before you can see Exiting... (End of file) in the command prompt, you will hear the last 1/2 second or so of the audio track again. This behavior is the same for multiple containers/codecs/soundcards Vista or Windows 7. Is there a workaround for this? My playback specs: MPlayer Sherpya-MT-SVN-r31027-4.2.5 (C) 2000-2010 MPlayer Team 150 audio & 343 video codecs Playing splash_final.wmv. ASF file format detected. [asfheader] Audio stream found, -aid 1 [asfheader] Video stream found, -vid 2 VIDEO: [WMV3] 1280x720 24bpp 1000.000 fps 6291.5 kbps (768.0 kbyte/s) ========================================================================== Opening video decoder: [dmo] DMO video codecs DMO dll supports VO Optimizations 0 1 DMO dll might use previous sample when requested Decoder supports the following formats: YV12 YUY2 UYVY YVYU RGB8 [..] Decoder is capable of YUV output (flags 0x1b) Movie-Aspect is undefined - no prescaling applied. VO: [directx] 1280x720 = 1280x720 Planar YV12 Selected video codec: [wmv9dmo] vfm: dmo (Windows Media Video 9 DMO) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, s16le, 329.8 kbit/23.37% (ratio: 41221-176400) Selected audio codec: [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg)) ========================================================================== AO: [dsound] 44100Hz 2ch s16le (2 bytes per sample) Starting playback...

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  • No audio input deviced are installed

    - by Meowbits
    If I go to Sound Recording Devices and it says "No audio devices are installed" If I click to set up a microphone I get an error "Wizard could not launch, No audio input device found, make sure your audio hardware is working properly and check your audio configuration in the Audio Devices and Sound Themes control panel. Where can I get an audio input device? I just want something so I can actually use the microphone on my headset. This is ridiculous. I have tried to look for any file but I simply cannot find a way to add an audio input device... I really do not want to format my computer just for this problem but I am starting to feel like that is the only option I have. I have the latest chipsets

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  • mix audio with h264 mp4 video with ffmpeg

    - by user2362912
    I have 2 files : Input #0, wav, from '105426_1.wav': Duration: 00:00:09.98, bitrate: 1312 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 41000 Hz, stereo, s16, 1312 kb/s and: Duration: 00:00:41.29, start: 0.000000, bitrate: 1313 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1211 kb/s, 24.42 fps, 25 tbr, 90k tbn, 48 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 99 kb/s Metadata: handler_name : SoundHandler I want to insert first audio file into video in special place (for example in 10 secunde of video) and mix it with audio stream of video file. I try to /usr/local/bin/ffmpeg -i 105426_1.wav -i 105426.mp4 -map 0:0 -map 1:1 -map 1:0 video_finale.mp4 but result is : Duration: 00:00:41.31, start: 0.046440, bitrate: 755 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:2(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 588 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc Metadata: handler_name : VideoHandler I need only one audio stream and first stream play not from beginig but from 10 sec

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  • How can I split a stereo audio track of a movie into two separate audio tracks?

    - by pesche
    I often record TV shows with a hard disk recorder/DVD writer, burn them as VRO file and convert to MP4 with Handbrake. The shows are bilingual broadcasts with two mono audio channels instead of a stereo one: dubbed voice on the left, original voice on the right. The TV set and VLC are both perfectly capable to play only the left or the right channel, but other video players may just offer to select between different stereo audio tracks (like they are present on many DVDs). I'd like to have an easy process to create MP4 or MKV files of these shows where the two audio channels are split into two separate audio tracks. The only way that I know of is to extract the audio track (e.g. using MPEG Streamclip), split it into two tracks using an audio tool like Audacity and then merge the audio tracks back (using a DVD authoring software, don't remember all details). Clearly not a thing to repeat regularly. Preferably a solution should run on Mac OS X, but Linux or Windows solutions are very welcome, too.

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