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  • Parse metadata from http live stream

    - by supo
    Hi, I'd like to extract the info string from an internet radio streamed over HTTP. By info string I mean the short note about the currently played song, band name etc. Preferably I'd like to do it in python. So far I've tried opening a socket but from there I got a bunch of binary data that I could not parse... thanks for any hints

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  • Preinitialize BackgroundAudioPlayer in WP7?

    - by kgrevehagen
    When I am using the BackgroundAudioPlayer in my Windows Phone 7 application, it takes a lot of time to load the first time I want to play a song. Is there any way of preinitializing the BackgroundAudioPlayer before playing the first track, so that when I start playing, it starts right along? I have googled it, but no luck. I am just using BackgroundAudioPlayer.Instance when I e.g. want to play, pause, stop etc an audiotrack. Is there something other i could do to fix this? Thanks in advance!

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  • How to stream authenticated content with MediaPlayer on Android

    - by 102790073222983779908
    I've seen quite a few posts askign this question on SO but there doesn't seem to be a definitive answer (or at least an answer I like!) I've got content protected behind basic auth (username/password) -- I can download it fine using the various HTTP download clases but for the life of me I can't sort out how to tell media player to stream it (and provide the authentication). I saw one post that suggested it wasn't possible since the MediaPlayer is all native code and doesn't things like the Authenticator. There are plenty of examples of how to first download to a cached copy and then play that back but....That sort of sucks (and the files maybe 100's of MB's). I saw at least one proposal to download it in smalish chunks and then start & stop the playback (redirecting to the new file) but that sort of sucks also since there would (I presume) be a stutter (I haven't tried it though) The best idea I have at this point is to start downloading to a cache file and then when it's 'full enough' start up playback while I continue to fill the file.... I hope that this works (but again, haven't tried it). Am I missing something obvious? It's so painful to have all the various pieces almost working and I sort of convinced myself that there had to be a way to natively stream protected content (or have it take a already established & qualified InputStream) but it appears no joy. BTW I'm a Mac/iPhone guy and a newb at Android so I'm still fighting a bit of Java learning.... Excuse me if I'm missing somthing obvious. -john

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  • Drawing a waveform in C#

    - by user488792
    Hi! I want to be able to display a WaveForm in C#, along with some simple features such as zooming and selection. I already have the data as a short[] of amplitude values. However, I am an amateur when it comes to hardcoding GUI. I have already found a possible helper class WaveFormClass that may help me achieve this but as a backup, I want to learn how to manually do it. So may I ask for some methods and possibly some links that will help? Thanks!

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  • Getting Frequency Components with FFT

    - by ruhig brauner
    so I was able to solv my last problem but i stubmled upon the next already. So I want to make a simple spectrogram but in oder to do so I want to understand how FFT-libaries work and what they actually calculate and return. (FFT and Signal Processing is the number 1 topic I will get into as soon as I have time but right now, I only have time for some programming exercises in the evening. ;) ) Here I just summarized the most important parts: int framesPerSecond; int samplesPerSecond; int samplesPerCycle; // right now i want to refresh the spectogram every DoubleFFT_1D fft; WAVReader audioIn; double audioL[], audioR[]; double fftL[], fftR[]; ..... framesPerSecond = 30; audioIn= new WAVReader("Strobe.wav"); int samplesPerSecond = (int)audioIn.GetSampleRate(); samplesPerCycle = (int)(audioIn.GetSampleRate()/framesPerSecond); audioL = new double[samplesPerCycle*2]; audioR = new double[samplesPerCycle*2]; fftL = new double[samplesPerCycle]; fftR = new double[samplesPerCycle]; for(int i = 0; i < samplesPerCycle; i++) { // don't even know why,... fftL[i] = 0; fftR[i] = 0; } fft = new DoubleFFT_1D(samplesPerCycle); ..... for(int i = 0; i < samplesPerCycle; i++) { audioIn.GetStereoSamples(temp); audioL[i]=temp[0]; audioR[i]=temp[1]; } fft.realForwardFull(audioL); //still stereo fft.realForwardFull(audioR); System.out.println("Check"); for(int i = 0; i < samplesPerCycle; i++) { //storing the magnitude in the fftL/R arrays fftL[i] = Math.sqrt(audioL[2*i]*audioL[2*i] + audioL[2*i+1]*audioL[2*i+1]); fftR[i] = Math.sqrt(audioR[2*i]*audioR[2*i] + audioR[2*i+1]*audioR[2*i+1]); } So the question is, if I want to know, what frequencys are in the sampled signal, how do I calculate them? (When I want to print the fftL / fftR arrays, I get some exponential formes at both ends of the array.) Thx :)

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  • sound not playing when i press the button and how to fix overlapping sounds

    - by alfredjunco
    the code is giving me an error"Unused variable'path'" and when i press a button there is no sound playing how do i fix this the aSound is in the h file - (void)playOnce:(NSString *)aSound; - (IBAction) beatButton50 { [self playOnce:@"racecars"]; } - (void)playOnce:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if([theAudio isPlaying]) { [theAudio stop]; } } - (void)playLooped:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if (!theAudio) { theAudio = [[AVAudioPlayer alloc] initWithContentsOfURL: [NSURL fileURLWithPath: path] error: NULL]; } [theAudio setDelegate: self]; // loop indefinitely [theAudio setNumberOfLoops: -1]; [theAudio setVolume: 1.0]; [theAudio play]; } - (void)stopAudio { [theAudio stop]; [theAudio setCurrentTime:0]; }

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  • Uninterrupted mp3 play on a website?

    - by Kevin
    Client is requesting a single track to be heard across the website. Generally I advise against it, but they insist. So, what is the most straightforward way of having a flash player embedded in a site, and when a user goes to another page there isn't a gap/interruption? I am thinking an iframe is required.. I am using a flash player that has autoresume, but that only solves picking up where you last left off on the song before going to another page. I tried searching SO for an answer..

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  • CMake: Mac OS X: ld: unknown option: -soname

    - by Alex Ivasyuv
    I try to build my app with CMake on Mac OS X, I get the following error: Linking CXX shared library libsml.so ld: unknown option: -soname collect2: ld returned 1 exit status make[2]: *** [libsml.so] Error 1 make[1]: *** [CMakeFiles/sml.dir/all] Error 2 make: *** [all] Error 2 This is strange, as Mac has .dylib extension instead of .so. There's my CMakeLists.txt: cmake_minimum_required(VERSION 2.6) PROJECT (SilentMedia) SET(SourcePath src/libsml) IF (DEFINED OSS) SET(OSS_src ${SourcePath}/Media/Audio/SoundSystem/OSS/DSP/DSP.cpp ${SourcePath}/Media/Audio/SoundSystem/OSS/Mixer/Mixer.cpp ) ENDIF(DEFINED OSS) IF (DEFINED ALSA) SET(ALSA_src ${SourcePath}/Media/Audio/SoundSystem/ALSA/DSP/DSP.cpp ${SourcePath}/Media/Audio/SoundSystem/ALSA/Mixer/Mixer.cpp ) ENDIF(DEFINED ALSA) SET(SilentMedia_src ${SourcePath}/Utils/Base64/Base64.cpp ${SourcePath}/Utils/String/String.cpp ${SourcePath}/Utils/Random/Random.cpp ${SourcePath}/Media/Container/FileLoader.cpp ${SourcePath}/Media/Container/OGG/OGG.cpp ${SourcePath}/Media/PlayList/XSPF/XSPF.cpp ${SourcePath}/Media/PlayList/XSPF/libXSPF.cpp ${SourcePath}/Media/PlayList/PlayList.cpp ${OSS_src} ${ALSA_src} ${SourcePath}/Media/Audio/Audio.cpp ${SourcePath}/Media/Audio/AudioInfo.cpp ${SourcePath}/Media/Audio/AudioProxy.cpp ${SourcePath}/Media/Audio/SoundSystem/SoundSystem.cpp ${SourcePath}/Media/Audio/SoundSystem/libao/AO.cpp ${SourcePath}/Media/Audio/Codec/WAV/WAV.cpp ${SourcePath}/Media/Audio/Codec/Vorbis/Vorbis.cpp ${SourcePath}/Media/Audio/Codec/WavPack/WavPack.cpp ${SourcePath}/Media/Audio/Codec/FLAC/FLAC.cpp ) SET(SilentMedia_LINKED_LIBRARY sml vorbisfile FLAC++ wavpack ao #asound boost_thread-mt boost_filesystem-mt xspf gtest ) INCLUDE_DIRECTORIES( /usr/include /usr/local/include /usr/include/c++/4.4 /Users/alex/Downloads/boost_1_45_0 ${SilentMedia_SOURCE_DIR}/src ${SilentMedia_SOURCE_DIR}/${SourcePath} ) #link_directories( # /usr/lib # /usr/local/lib # /Users/alex/Downloads/boost_1_45_0/stage/lib #) IF(LibraryType STREQUAL "static") ADD_LIBRARY(sml-static STATIC ${SilentMedia_src}) # rename library from libsml-static.a => libsml.a SET_TARGET_PROPERTIES(sml-static PROPERTIES OUTPUT_NAME "sml") SET_TARGET_PROPERTIES(sml-static PROPERTIES CLEAN_DIRECT_OUTPUT 1) ELSEIF(LibraryType STREQUAL "shared") ADD_LIBRARY(sml SHARED ${SilentMedia_src}) # change compile optimization/debug flags # -Werror -pedantic IF(BuildType STREQUAL "Debug") SET_TARGET_PROPERTIES(sml PROPERTIES COMPILE_FLAGS "-pipe -Wall -W -ggdb") ELSEIF(BuildType STREQUAL "Release") SET_TARGET_PROPERTIES(sml PROPERTIES COMPILE_FLAGS "-pipe -Wall -W -O3 -fomit-frame-pointer") ENDIF() SET_TARGET_PROPERTIES(sml PROPERTIES CLEAN_DIRECT_OUTPUT 1) ENDIF() ### TEST ### IF(Test STREQUAL "true") ADD_EXECUTABLE (bin/TestXSPF ${SourcePath}/Test/Media/PlayLists/XSPF/TestXSPF.cpp) TARGET_LINK_LIBRARIES (bin/TestXSPF ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/test1 ${SourcePath}/Test/test.cpp) TARGET_LINK_LIBRARIES (bin/test1 ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/TestFileLoader ${SourcePath}/Test/Media/Container/FileLoader/TestFileLoader.cpp) TARGET_LINK_LIBRARIES (bin/TestFileLoader ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/testMixer ${SourcePath}/Test/testMixer.cpp) TARGET_LINK_LIBRARIES (bin/testMixer ${SilentMedia_LINKED_LIBRARY}) ENDIF (Test STREQUAL "true") ### TEST ### ADD_CUSTOM_TARGET(doc COMMAND doxygen ${SilentMedia_SOURCE_DIR}/doc/Doxyfile) There was no error on Linux. Build process: cmake -D BuildType=Debug -D LibraryType=shared . make I found, that incorrect command generate in CMakeFiles/sml.dir/link.txt. But why, as the goal of CMake is cross-platforming.. How to fix it?

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  • How to sync audio files with Logitech media server in MAC OS?

    - by Abhishek
    I want to customize the Logitech Media Server (web interface on localhost) so that N number of DIFFERENT audio files will start to play at the same time on N number of wifi receivers, each file on a different receiver. Currently, the server will sync only 1 track to N number(amount) of receivers. Is it possible with Logitech media server is open source. How can I able to do this? can you explain me sample code?

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  • Ask HTG: Dealing with Windows 8 CP Expiry, Nintendo DS Save Backups, Jumbled Audio Tracks in Windows Media Player

    - by Jason Fitzpatrick
    Once a week we round up some great reader questions and share the answers with everyone. This week we’re looking at what to do when Windows 8 Consumer Preview expires, backing up your Nintendo DS saves, and how to sort out jumbled audio tracks in Windows Media Player movies. How To Be Your Own Personal Clone Army (With a Little Photoshop) How To Properly Scan a Photograph (And Get An Even Better Image) The HTG Guide to Hiding Your Data in a TrueCrypt Hidden Volume

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  • How to record both audio, Where i have one music running and my microphone is in use?

    - by YumYumYum
    I have one music playing, and i have microphone open, already the microphone is used by other application. In such case, how can i record that music and the microphone audio to a file? (if possible with command line). Follow up: $ rec new-file.wav Input File : 'default' (alsa) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:25.94 [00:00:00.00] Out:1.24M [ | ] Clip:0 ^C $ sox -d new-file.wav

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  • Join mp4 files in linux

    - by Jose Armando
    I want to join two mp4 files to create a single one. The video streams are encoded in h264 and the audio in aac. I can not re-encode the videos to another format due to computational reasons. Also, I cannot use any gui programs, all processing must be performed with linux command line utilities. FFmpeg cannot do this for mpeg4 files so instead I used MP4Box e.g. MP4Box -add video1.mp4 -cat video2.mp4 newvideo.mp4 unfortunately the audio gets all mixed up. I thought that the problem was that the audio was in aac so I transcoded it in mp3 and used again MP4Box. In this case the audio is fine for the first half of newvideo.mp4 (corresponding to video1.mp4) but then their is no audio and I cannot navigate in the video also. My next thought was that the audio and video streams had some small discrepancies in their lengths that I should fix. So for each input video I splitted the video and audio streams and then joined them with the -shortest option in ffmpeg. thus for the first video I ran avconv -y -i video1.mp4 -c copy -map 0:0 videostream1.mp4 avconv -y -i video1.mp4 -c copy -map 0:1 audiostream1.m4a avconv -y -i videostream1.mp4 -i audiostream1.m4a -c copy -shortest video1_aligned.mp4 similarly for the second video and then used MP4Box as previously. Unfortunately this didn't work either. The only success I had was when I joined the video streams separetely (i.e. videostream1.mp4 and videostream2.mp4) and the audio streams (i.e. audiostream1.m4a and audiostream2.m4a) and then joined the video and audio in a final file. However, the synchronization is lost for the second half of the video. Concretelly, there is a 1 sec delay of audio and video. Any suggestions are really welcome.

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  • How to upload binary (audio) data from a Flash AS3 client to .NET server (WCF/REST/HTTP/?)?

    - by Bobby
    Simply stated: I'm trying to record audio in a browser, and get that data back up to the server. I originally tried to capture, encode and upload the audio using Silverlight, but because of the lack of suitable client-side encoding options, I'm now giving Flash a shot (Flash has baked-in support for encoding to Speex). I think I've figured out how to capture and encode the audio... But now what was easy in Silverlight, is the challenge in Flash. My server-side is .NET: MVC2- I'm open to receiving the audio in whatever manner is best- REST, WCF.. So that's my question: How could one upload binary data from Flash, to a .NET server-side endpoint. If the answer is WCF: then how would one setup the client-side proxies to communicate with the service? If the answer is REST or HTTP Post, then how would one construct this HTTP request and pass along the data? I've been reading up on AS3, but am new to Flash dev... Thanks for any help!

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  • MPlayer does not work

    - by Soham Pal
    Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

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  • alsa - sound issues on ubuntu 12.04

    - by tam_ubuuser
    i am having an sony E series laptop.i have an HDMI port .at this stage ,i have tested my sound card , which provides audio out on my laptop i.e i could hear songs .my laptop has two sound cards amd 5450 and an intel-hda(alsamixer shows that as s/pdif) . i decided to connect HDMI output to my new HD-TV.but, i could get only visuals on my TV,NO AUDIO OUTPUT ( HDMI cable works fine with win 7).my laptop has two sound cards.but i couldn't switch output to other card.( i don't know ,how to do that) i decided to update alsa. complied the following code in terminal. sudo apt-add-repository ppa:ubuntu-audio-dev/alsa-daily sudo apt-get update sudo apt-get install alsa-hda-dkms then,strangely no login sound, and no audio output on my laptop at all .then, started complied code from step1 sound troubleshooting procedure from offical ubuntu site.then, my speaker icon taskbar disappeared .obivously $aplay -l ,provided output as no soundcards detected . so , i implemented step 4 from that guide, it provides a output of all hardware devices in my laptop. *-multimedia UNCLAIMED description: Audio device product: Cedar HDMI Audio [Radeon HD 5400/6300 Series] vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 0.1 bus info: pci@0000:01:00.1 version: 00 width: 64 bits clock: 33MHz capabilities: pm pciexpress msi bus_master cap_list configuration: latency=0 resources: memory:f0040000-f0043fff *-multimedia UNCLAIMED description: Audio device product: 5 Series/3400 Series Chipset High Definition Audio vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 05 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: latency=0 resources: memory:f5e00000-f5e03fff that command displayed output name of the two cards . but , still i have no positive output on $aplay -l. so therfore, i think alsa couldn't detect my sound cards . is there solution to this problem? it could be better,if alsa would channel output from multiple sound cards ? how should install and configure alsa such that detects HDMI cable as soon i connect to my HD tv? is it possible to alsa and pluseaudio 2.0 to co-exist, if so how?

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  • why some mp3s on mime_content_type return application/octet-stream

    - by robertdd
    why on some mp3s file when i call mime_content_type($mp3_file_path) it's return application/octet-stream? i have this: if (!empty($_FILES)) { $tempFile = $_FILES['Filedata']['tmp_name']; $image = getimagesize($tempFile); $mp3_mimes = array('audio/mpeg', 'audio/x-mpeg', 'audio/mp3', 'audio/x-mp3', 'audio/mpeg3', 'audio/x-mpeg3', 'audio/mpg', 'audio/x-mpg', 'audio/x-mpegaudio'); if (in_array(mime_content_type($tempFile), $mp3_mimes)) { echo json_encode("mp3"); } elseif ($image['mime']=='image/jpeg') { echo json_encode("jpg"); } else{ echo json_encode("error"); } }

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  • Where can I buy a Stereo audio to 3.5mm adapter?

    - by iftrue
    I need a stereo (6.33mm) to PC audio (3.5mm) adapter, and I'd like it to have an inch or two of cable so that yanking the connector doesn't break the audio port the 3.5mm is plugged into. I used to own one of these, but I lost the adapter. Where can I buy something like this online? I can only find solid adapters or 25' cables.

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  • How can I split 200Mbps of streaming traffic into routers?

    - by Jared
    As the title says, I have 200Mbps of streaming video traffic coming into my command center. How do I split the load between routers? Setup is like this: fiber --- router --- switch --- workstations I'm sorry I haven't dealt with this much traffic before. so please be gentle if you're going to kick me out :) EDITED FOR DETAILS: Okay, this specific project is for our company's IP CCTV system. We have deployed over 100++ cameras all over a building/campus and we have estimated each camera to take about 2Mbps of bandwidth each. Now, they're all connected to a switch and that's entirely fine. But coming into our command center, they have to be on a router since it'll get more than 200++ cameras next year (and I don't want to have too many hosts on one subnet). My plan was to have the 1st hundred on a 172.16.9.x block and the 2nd hundred on a 172.16.10.x block (all /24). The servers I have are currently sized to match (about 5 dual 6-core xeons) and I'd have about 19 workstations all streaming video from the 5 servers. (servers pull video from the cameras). But 200Mbps of constant traffic? How the hell do I even break this up? I need to have 1 gateway, to manage the routes... I honestly think I'm way in over my head.

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  • JNI Stream binary data from C++ to Java

    - by Cliff
    I need help passing binary data into Java. I'm trying to use jbytearray but when the data gets into Java it appears corrupt. Can somebody give me a hand? Here's a snip of some example code. First the native C++ side: printf("Building audio array copy\n"); jbyteArray rawAudioCopy = env-NewByteArray(10); jbyte toCopy[10]; printf("Filling audio array copy\n"); char theBytes[10] = {0,1,2,3,4,5,6,7,8,9}; for (int i = 0; i < sizeof(theBytes); i++) { toCopy[i] = theBytes[i]; } env->SetByteArrayRegion(rawAudioCopy,0,10,toCopy); printf("Finding object callback\n"); jmethodID aMethodId = env->GetMethodID(env->GetObjectClass(obj),"handleAudio","([B)V"); if(0==aMethodId) throw MyRuntimeException("Method not found error",99); printf("Invoking the callback\n"); env->CallVoidMethod(obj,aMethodId, &rawAudioCopy); and then the Java callback method: public void handleAudio(byte[] audio){ System.out.println("Audio supplied to Java [" + audio.length + "] bytes"); byte[] expectedAudio = {0,1,2,3,4,5,6,7,8,9}; for (int i = 0; i < audio.length; i++) { if(audio[i]!= expectedAudio[i]) System.err.println("Expected byte " + expectedAudio[i] + " at byte " + i + " but got byte " + audio[i]); else System.out.print('.'); } System.out.println("Audio passed back accordingly!"); } I get the following output when the callback is invoked: library loaded! Audio supplied to Java [-2019659176] bytes Audio passed back accordingly!

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  • Stream Media and Live TV Across the Internet with Orb

    - by DigitalGeekery
    Looking for a way to stream your media collection across the Internet? Or perhaps watch and record TV remotely? Today we are going to look at how to do all that and more with Orb. Requirements Windows XP / Vista / 7 or Intel based Mac w/ OS X 10.5 or later. 1 GB RAM or more Pentium 4 2.4 GHz or higher / AMD Athlon 3200+ Broadband connections TV Tuner for streaming and recording live TV (optional) Note: Slower internet connections may result in stuttering during playback. Installation and Setup Download and install Orb on your home computer. (Download link below) You’ll want to take the defaults for the initial portion of the install. When we get to the Orb Account setup portion of the install is when we will have to enter information and make some decisions. Choose your language and click Next. We’ll need to create and user account and password. A valid email address is required as we’ll need to confirm the account later. Click Next.   Now you’ll want to choose your media sources. Orb will automatically look for folders that may contain media files. You can add or remove folders click on the (+) or (-) buttons. To remove a folder, click on it once to select it from the list and then click the minus (-) button. To add a folder, click the plus (+) button and browse for the folder. You can add local folders as well as shared folders from networked computers and USB attached storage. Note: Both the host computer running Orb and the networked computer will need to be running to access shared network folders remotely. When you’ve selected all your media files, click Next. Orb will proceed to index your media files… When the indexing is complete, click Next. Orb TV Setup Note: Streaming Live TV to Macs is not currently supported. If you have a TV tuner card connected to your PC, you can opt to configure Orb to stream live or recorded TV. Click Next  to configure TV. Or, choose Skip if you don’t wish to configure Orb for TV.   If you have a Digital tuner card, type in your Zip Code and click Get List to pull your channel listings. Select a TV provider from the list and click Next. If not, click Skip.   You can select or deselect any channels by checking or un-checking the box to each channel. Select Auto Scan to let Orb find more channels or disable the ones with no reception. Click Next when finished.   Next choose an analog provider, if necessary, and click Next.   Select “Yes” or “No” for a set top box and click Next. Just as we did with the Digital tuner, select or deselect any channels by checking or un-checking the box to each channel. Select Auto Scan to let Orb find more channels or disable the ones with no reception. Click Next when finished.   Now we’re finished with the setup. Click Close. Accessing your Media Remotely Media files are accessed through a web-based interface. Before we go any further, however, we’ll need to confirm our username and password. Check your inbox for an email from Orb Networks. Click the enclosed confirmation link. You’ll be prompted to enter the username and password you selected in your browser then click Next.   Your account will be confirmed. Now, we’re ready to enjoy our media remotely. To get started, point your browser to the MyCast website from your remote computer. (See link below) Enter your credentials and click Log In. Once logged in, you’ll be presented with the MyCast Home screen. By default you’ll see a handful of “channels” such as a TV program guide, random audio and photos, video favorites, and weather. You can add, remove, or customize channels. To add additional channels, click on Add Channels at the top right…   …and select from the dropdown list. To access your full media libraries, click Open Application at the top left and select from one of the options. Live and Recorded TV If you have a TV tuner card you configured for Orb, you’ll see your program guide on the TV / Webcams screen. To watch or record a show, click on the program listing to bring up a detail box. Then click the red button to record, or the green button to play. When recording a show, you’ll see a pulsating red icon at the top right of the listing in the program guide. If you want to watch Live TV, you may be prompted to choose your media player, depending on your browser and settings. Playback should begin shortly.   Note for Windows Media Center Users If you try to stream live TV in Orb while Windows Media Center is running on your PC, you’ll get an error message. Click the Stop MediaCenter button and then try again.   Audio On the Audio screen, you’ll find your music files indexed by genre, artist, and album. You can play a selection by clicking once and then clicking the green play button, or by simply double-clicking.   Playback will begin in the default media player for the streaming format.   Video Video works essentially the same as audio. Click on a selection and press the green play button, or double-click on the video title. Video playback will begin in the default media player for the streaming format.   Streaming Formats You can change the default streaming format in the control panel settings. To access the Control Panel, click on Open Applications  and select Control Panel. You can also click Settings at the top right.   Select General from the drop down list and then click on the Streaming Formats tab. You are provided four options. Flash, Windows Media, .SDP, and .PLS.   Creating Playlists To create playlists, drag and drop your media title to the playlist work area on the right, or click Add to playlist on the top menu. Click Save when finished.    Sharing your Media Orb allows you to share media playlists across the Internet with friends and family. There are a few ways to accomplish this. We’ll start by click the Share button at the bottom of the playlist work area after you’ve compiled your playlist. You’ll be prompted to choose a method by which to share your playlist. You’ll have the option to share your playlist publicly or privately. You can share publically through links, blogs, or on your Orb public profile.  By choosing the Public Profile option, Orb will automatically create a profile page for you with a URL like http://public.orb.com/username that anyone can easily access on the Internet. The private sharing option allows you to invite friends by email and requires recipients to register with Orb. You can also give your playlist a custom name, or accept the auto-generated title. Click OK when finished. Users who visit your public profile will be able to view and stream any of your shared playlists to their computer or supported device.   Portable Media Devices and Smartphones Orb can stream media to many portable devices and 3G phones. Streaming audio is supported on the iPhone and iPod Touch through the Safari browser. However, video and live TV streaming requires the Orb Live iPhone App.  Orb Live is available in the App store for $9.99. To stream media to your portable device, go to the MyCast website in your mobile browser and login. Browse for your media or playlist. Make a selection and play the media. Playback will begin. We found streaming music to both the Droid and the iPhone to work quite nicely. Video playback on the Droid, however, left a bit to be desired. The video looked good, but the audio tended to be out of sync. System Tray Control Panel By default Orb runs in the system tray on start up. To access the System Tray Control Panel, right-click on the Orb icon in the system tray and select Control Panel. Login with your Orb username and  password and click OK.   From here you can add or remove media sources, add manage accounts, change your password, and more. If you’d rather not run Orb on Startup, click the General icon.   Unselect the checkbox next to Start Orb when the system starts. Conclusion It may seem like a lot of steps, but getting Orb up and running isn’t terribly difficult. Orb is available for both Windows and Intel based Macs. It also supports streaming to many Game Consoles such as the Wii, PS3, and XBox 360. If you are running Windows 7 on multiple computers, you may want to check out our write-up on how to stream music and video over the Internet with Windows Media Player 12. 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