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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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• iPhone SDK: Change playback speed using core audio AVAudioPlayer

  - by Harkonian

  I'd like to be able to play back audio I've recorded using AVAudioRecorder @ 1.5x or 2.0x speed. I don't see anything in AVAudioPlayer that will support that. I'd appreciate some suggestions, with code if possible, on how to accomplish this with the iPhone 3.x SDK. I'm not overly concerned with lowering the pitch to compensate for increased playback speed, but being able to do so would be optimal.

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• GWT audio recording

  - by oelewapperke

  Are there any options for recording audio in a GWT application ? Options seem to include flex, perhaps a java applet or some other form of flash ?

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• cut audio file with iPhone SDK

  - by Dmitry

  Hi! Is it possible to cut audio file with iPhone SDK? (file has .caf extension) I just need to cut off the silence at the beginning. (Also, maybe it's possible to write new file from the existing one with specified start and end time.) Thanks in advance!

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• Video and Audio Drift

  - by Cenoc

  Hey everyone, I was wondering, how much does recorded audio and video drift from their actual recording time usually? I'm recording both separately (into unsigned 8 bit PCM (44100 Hz) and raw image data files) and I was wondering how much I can expect each to drift. Thanks in advance!

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• Audio Recording in C++

  - by Cenoc

  Hey, I was wondering, what was a good cross-platform utility for doing audio recording/ playback/ seeking in C++? I was thinking going the route of ALUT (OpenAL), but is there a better way? If not, do you guys know of any good tutorials/sample code for ALUT?

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• Handle a More Navigation Controller in an Interface Builder based TabBar Application

  - by Thomas Joulin

  Hi, I'm still not clear on how and when to use interface builder. I have a tabbar-based application, in which I added 6 navigations controllers. Instead of having 6 tabs, I would like 3 plus a "More" tab which allows the user to configure the tabs he wants. Is there any way to do that with IB ? And if not, how can I move from IB to a code-based tabbar (provided I already set up a class TabBarController which handles shouldAutoRotate:) Thanks in advance !

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• Extract music files from a Audio CD [closed]

  - by Jatin

  Possible Duplicate: What good, free audio CD ripping/extraction tools exist for Windows, and supporting multiple formats? I have an audio cd, which has audio files with the file format as .cda ( CD Audio Track ). Each one of these files have a size of 1 KB each, and the rest of the CD has nothing else. Is there a way that I can get the audio files from the CD and then convert it into mp3 format and then play it in any other devices as I like.

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• Multiple Audio I/P and O/P simultaneaously

  - by Raj Naveen

  hi (1) i saw in one of your posts that it is possible to get different outputs in windows 7. i am eager to know more. Is there any way i can create a 2 or more virtual cable between two softwares simultaneously. so that simultaneously, two or more audio inputs will be routed to equal no of audio analysers receivers, and then the audio analysers send back a filtered audio back to respective audio inputs... Please reply to email id: [email protected]

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• Nyquist won't play audio

  - by erjiang

  I downloaded Nyquist, and am having trouble playing sounds from it. If I run it normally, I get: Nyquist -- A Language for Sound Synthesis and Composition Copyright (c) 1991,1992,1995 by Roger B. Dannenberg Version 2.29 > (play (osc 60)) Saving sound file to ./eric-temp.wav error: snd_save -- could not open audio output > If I wrap it by running padsp ny, the sound plays fine for about half a second, and then I get garbage fed to my speakers. Any solutions?

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• Multiple Audio listeners in Scene

  - by Kevin Jensen Petersen

  THIS IS UNITY Im trying to make a FPS game over networking, it works fine. But now, when im trying to implement sound, it won't work. My guess would be, to add a Audio listener to the prefab, that gets instansiated whenever a player connects to the server, however the problem about this is that each player's audiolistener have been switched out which the other player(s), so the AudioSource won't play at the player, but at someone else in the game. Any suggestions ?

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• Audio Stutters at gdm

  - by Allan

  Ok I have a problem every 2 times out of 3 I login (I cant be specific it fairly random) I get a Stuttering GDM warning (not the login sound just the Bell sound to wake you up) the only way to stop it is to login I have a Fujitsu Siemens Amilo 1718 with a 2gig of memory (only hardware mod) using 10.10 Maverick and I have disabled KMS as my system was freezing as per the release notes. The only time this has happened before on the same machine was when I gave Kubuntu a try when 10.04 came out then it happened at the login screen and at random times while listening to music in any program. By the way audio is fine as is almost everything else once I have logged in. I would like an answer to this as I am an advocate of Ubuntu and its kind of embarrassing when the first thing that happens is *bing*. as requested Daniel alsa-info Pulse verbose log Not sure how useful the pulse log will be as I cant replicate the bug with a terminal open but I wouldnt be asking the question if I knew the answer so..... Edit 24/12/2010 ......been living on cocktail sausages and pickled onions for five days now made a make shift splint with cocktail sticks..... oops so updated the alsa drivers but I still get the same message in the dmesg No response from codec, disabling MSI: last cmd=0x10a90000 googleing it brings up a forum post from some other distro with a green logo the only common denominator seems to be graphics ie ATI Radeon XPRESS 200M which is why I have had to turn of kms as the chip is so old that small mice try to eat the "kernel" ;) funnily enough following the bug link at the end of the post, I found a comment about "Ubuntu Black Magic" so mabey I am coming at this from the wrong angle...... Bad Joo Joo any one. I will try the second part of Daniels Fix and Update with the result. The final Edit: (Plays air guitar) In the end neither of these solved the problem as such However I have given Roland a tick for reminding me of the solution and I gave Daniel the Bounty for the effort in trying to solve the problem. The answer for future readers was the enable the correct HD Audio Model I found the answer back when using Karmic Koala 9.10 in this forum post Amilo Li1718 Skype - Can't get it working... the model is options snd-hda-intel model=3stack position_fix=1 enable=yes which can be added to the end of alsa-base.conf thanks all for helping and hope anyone with a similar problem will find the answer here.

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• Synchronizing audio with scrolling text

  - by mr yoshida

  I am trying to have a website that vertically scrolls about 5 paragraphs of text with a matching audio file that reads along with it. It doesn't need to be synchronized word for word such as highlighting each spoken word but an accurate start and stop time. I've searched for quite a bit on the most efficient way of doing this but can't seem to find any answers. I tried Flash but really don't want to use it. Thanks in advance.

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• Create Audio file on iPhone/iPad from many other audio files (mixer)

  - by Brian

  I am trying to create something similar like Piano app on the iPhone. When people tap a key, it play a piano note. Basically, there will have only 7 notes (C) at the moment. Each note is a .caf file and its length is 5 seconds. I do not know if there is any way to save the song user played and export to mp3/caf format? The AVAudioRecord seems only record from the microphone input. Many thanks

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• Problems with MediaRecorder class setting audio source - setAudioSource() - unsupported parameter

  - by arakn0

  Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaRecorder class to record just audio from the microphone. I'm following the steps indicated in the official site: http://developer.android.com/reference/android/media/MediaRecorder.html So I have a method that initializes and configure the MediaRecorder object in order to start recording. Here you have the code: this.mr = new MediaRecorder(); this.mr.setAudioSource(MediaRecorder.AudioSource.MIC); this.mr.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); this.mr.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); this.mr.setOutputFile(this.path + this.fileName); try { this.mr.prepare(); } catch (IllegalStateException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } catch (IOException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } When I execute this code in the simulator, thanks to logcat, I can see that the method setAudioSource(MediaRecorder.AudioSource.MIC) gives the next error (with the tag audio_ipunt) when it is called: ERROR/audio_input(34): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value ERROR/audio_input(34): VerifyAndSetParameter failed And then when the method prepare() is called, I get the another error again: ERROR/PVOMXEncNode(34): PVMFOMXEncNode-Audio_AMRNB::DoPrepare(): Got Component OMX.PV.amrencnb handle If I start to record bycalling the method start()... I get lots of messages saying: AudioFlinger(34):RecordThread: buffer overflow Then...after stop and release,.... I can see that a file has been created, but it doesn't seem that it been well recorderd. Anway, if i try this in a real device I can record with no problems, but I CAN'T play what I just recorded. I gues that the key is in these errors that I've mentioned before. How can I fix them? Any suggestion or help?? Thanks in advanced!!

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• Oracle Solaris Cluster 4.2 Event and its SNMP Interface

  - by user12609115

  Background The cluster event SNMP interface was first introduced in Oracle Solaris Cluster 3.2 release. The details of the SNMP interface are described in the Oracle Solaris Cluster System Administration Guide and the Cluster 3.2 SNMP blog. Prior to the Oracle Solaris Cluster 4.2 release, when the event SNMP interface was enabled, it would take effect on WARNING or higher severity events. The events with WARNING or higher severity are usually for the status change of a cluster component from ONLINE to OFFLINE. The interface worked like an alert/alarm interface when some components in the cluster were out of service (changed to OFFLINE). The consumers of this interface could not get notification for all status changes and configuration changes in the cluster. Cluster Event and its SNMP Interface in Oracle Solaris Cluster 4.2 The user model of the cluster event SNMP interface is the same as what was provided in the previous releases. The cluster event SNMP interface is not enabled by default on a freshly installed cluster; you can enable it by using the cluster event SNMP administration commands on any cluster nodes. Usually, you only need to enable it on one of the cluster nodes or a subset of the cluster nodes because all cluster nodes get the same cluster events. When it is enabled, it is responsible for two basic tasks. • Logs up to 100 most recent NOTICE or higher severity events to the MIB. • Sends SNMP traps to the hosts that are configured to receive the above events. The changes in the Oracle Solaris Cluster 4.2 release are1) Introduction of the NOTICE severity for the cluster configuration and status change events.The NOTICE severity is introduced for the cluster event in the 4.2 release. It is the severity between the INFO and WARNING severity. Now all severities for the cluster events are (from low to high) • INFO (not exposed to the SNMP interface) • NOTICE (newly introduced in the 4.2 release) • WARNING • ERROR • CRITICAL • FATAL In the 4.2 release, the cluster event system is enhanced to make sure at least one event with the NOTICE or a higher severity will be generated when there is a configuration or status change from a cluster component instance. In other words, the cluster events from a cluster with the NOTICE or higher severities will cover all status and configuration changes in the cluster (include all component instances). The cluster component instance here refers to an instance of the following cluster componentsnode, quorum, resource group, resource, network interface, device group, disk, zone cluster and geo cluster heartbeat. For example, pnode1 is an instance of the cluster node component, and oracleRG is an instance of the cluster resource group. With the introduction of the NOTICE severity event, when the cluster event SNMP interface is enabled, the consumers of the SNMP interface will get notification for all status and configuration changes in the cluster. A thrid-party system management platform with the cluster SNMP interface integration can generate alarms and clear alarms programmatically, because it can get notifications for the status change from ONLINE to OFFLINE and also from OFFLINE to ONLINE. 2) Customization for the cluster event SNMP interface • The number of events logged to the MIB is 100. When the number of events stored in the MIB reaches 100 and a new qualified event arrives, the oldest event will be removed before storing the new event to the MIB (FIFO, first in, first out). The 100 is the default and minimum value for the number of events stored in the MIB. It can be changed by setting the log_number property value using the clsnmpmib command. The maximum number that can be set for the property is 500. • The cluster event SNMP interface takes effect on the NOTICE or high severity events. The NOTICE severity is also the default and lowest event severity for the SNMP interface. The SNMP interface can be configured to take effect on other higher severity events, such as WARNING or higher severity events by setting the min_severity property to the WARNING. When the min_severity property is set to the WARNING, the cluster event SNMP interface would behave the same as the previous releases (prior to the 4.2 release). Examples, • Set the number of events stored in the MIB to 200 # clsnmpmib set -p log_number=200 event • Set the interface to take effect on WARNING or higher severity events. # clsnmpmib set -p min_severity=WARNING event Administering the Cluster Event SNMP Interface Oracle Solaris Cluster provides the following three commands to administer the SNMP interface. • clsnmpmib: administer the SNMP interface, and the MIB configuration. • clsnmphost: administer hosts for the SNMP traps • clsnmpuser: administer SNMP users (specific for SNMP v3 protocol) Only clsnmpmib is changed in the 4.2 release to support the aforementioned customization of the SNMP interface. Here are some simple examples using the commands. Examples: 1. Enable the cluster event SNMP interface on the local node # clsnmpmib enable event 2. Display the status of the cluster event SNMP interface on the local node # clsnmpmib show -v 3. Configure my_host to receive the cluster event SNMP traps. # clsnmphost add my_host Cluster Event SNMP Interface uses the common agent container SNMP adaptor, which is based on the JDMK SNMP implementation as its SNMP agent infrastructure. By default, the port number for the SNMP MIB is 11161, and the port number for the SNMP traps is 11162. The port numbers can be changed by using the cacaoadm. For example, # cacaoadm list-params Print all changeable parameters. The output includes the snmp-adaptor-port and snmp-adaptor-trap-port properties. # cacaoadm set-param snmp-adaptor-port=1161 Set the SNMP MIB port number to 1161. # cacaoadm set-param snmp-adaptor-trap-port=1162 Set the SNMP trap port number to 1162. The cluster event SNMP MIB is defined in sun-cluster-event-mib.mib, which is located in the /usr/cluster/lib/mibdirectory. Its OID is 1.3.6.1.4.1.42.2.80, that can be used to walk through the MIB data. Again, for more detail information about the cluster event SNMP interface, please see the Oracle Solaris Cluster 4.2 System Administration Guide. - Leland Chen 

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• Looking for an application to record audio and video on a linux "embedded" device

  - by Luke404

  I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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• Create a kind of Interface c++ [migrated]

  - by Liuka

  I'm writing a little 2d rendering framework with managers for input and resources like textures and meshes (for 2d geometry models, like quads) and they are all contained in a class "engine" that interacts with them and with a directX class. So each class have some public methods like init or update. They are called by the engine class to render the resources, create them, but a lot of them should not be called by the user: //in pseudo c++ //the textures manager class class TManager { private: vector textures; .... public: init(); update(); renderTexture(); //called by the "engine class" loadtexture(); gettexture(); //called by the user } class Engine { private: Tmanager texManager; public: Init() { //initialize all the managers } Render(){...} Update(){...} Tmanager* GetTManager(){return &texManager;} //to get a pointer to the manager //if i want to create or get textures } In this way the user, calling Engine::GetTmanager will have access to all the public methods of Tmanager, including init update and rendertexture, that must be called only by Engine inside its init, render and update functions. So, is it a good idea to implement a user interface in the following way? //in pseudo c++ //the textures manager class class TManager { private: vector textures; .... public: init(); update(); renderTexture(); //called by the "engine class" friend class Tmanager_UserInterface; operator Tmanager_UserInterface*(){return reinterpret_cast<Tmanager_UserInterface*>(this)} } class Tmanager_UserInterface : private Tmanager { //delete constructor //in this class there will be only methods like: loadtexture(); gettexture(); } class Engine { private: Tmanager texManager; public: Init() Render() Update() Tmanager_UserInterface* GetTManager(){return texManager;} } //in main function //i need to load a texture //i always have access to Engine class engine-GetTmanger()-LoadTexture(...) //i can just access load and get texture; In this way i can implement several interface for each object, keeping visible only the functions i (and the user) will need. There are better ways to do the same?? Or is it just useless(i dont hide the "framework private functions" and the user will learn to dont call them)? Before i have used this method: class manager { public: //engine functions userfunction(); } class engine { private: manager m; public: init(){//call manager init function} manageruserfunciton() { //call manager::userfunction() } } in this way i have no access to the manager class but it's a bad way because if i add a new feature to the manager i need to add a new method in the engine class and it takes a lot of time. sorry for the bad english.

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• BluRay audio/video stuttering with PowerDVD 11, WinDVD 11 Pro, etc? Xonar/Auzen HD audio option?

  - by jrista

  I recently upgraded my Windows 7 MediaCenter HTPC due to a motherboard failure (really old motherboard and cpu, it was on its last legs.) I chose to upgrade to an i5 system with everything built into the motherboard. I did my due diligence, researched, and found some hardware that was within my budget. I ended up with: Core i5 2500K (3.3Ghz) Corsair XMS3 2x2Gb DDR3 (4Gb) ASUS P8H 61-M LE/CSM MicroCenter 64Gb SSD (Previous BluRay player, forget the brand) The system is pretty awesome, and plays everything I have perfectly. I almost went with an Atom solution, however there have been numerous notes that they do not play NetFlix Instant Watch well...and I am a heavy Netflix IW user. High definition BluRay rips work well, although they usually contain lower audio quality than the BluRay's they were ripped from. The real problem I am encountering is playing back BluRay video from discs. For some reason, I am encountering rather terrible stuttering problems with both the audio and video. The stuttering is synchronous in both, and occurs at seemingly random intervals. I've used PowerDVD 9, PowerDVD 11 trial, and WinDVD 11 Pro trial. All three have stuttering problems, although PowerDVD 11 seems to have the least. Watching system resource usage, CPU load is never above 20%, and memory usage tends to be a constant 1/3rd the total available system memory. When playback is fine, its superb...the video is crystal clear. The audio quality is ok, certainly not what I would expect from a BluRay disc. I did some research, and it seems that playing BluRay from a PC causes a downsampling of the audio? I am curious if the audio is my primary problem here, the cause of the stuttering I am encountering? When stuttering occurs, the audio gets REALLY bad, while the video just pauses momentarily every second until for whatever reason everything picks up and runs fine (usually after a few seconds to a couple minutes.) The audio chipset is a Realtek HD ALC887 8-channel, supposedly designed to support BluRay playback. Has anyone encountered any issues like this playing back bluray discs on a PC (namely with PowerDVD...WinDVD was FAR worse, and seemed to have real trouble even reading the discs, and I have no interest in fiddling with it further.) Is there any reason to suspect the video decoding as the problem?(Given how bad the audio gets during a stutter, and how clean the video remains, I am inclined to think the issue boils down to audio.) Is it even remotely possible that the motherboard, cpu, or ram are causing the stuttering (all three are pretty blazing fast...faster than the hardware that I replaced, which seemed to play BluRay fine with PowerDVD 9.) I've read a bit about the Asus Xonar HDAV 1.3 and the Auzen X-Fi HomeTheater HD home theater hi-fi audio cards. Seems they are the only way to get true full-quality, uncompressed BluRay audio bitstreaming over HDMI on a PC. None of the usual suspects seem to have these cards in stock, however. Are these cards worth getting? Are they even still available, or have they been discontinued (if so, that would indeed be sad...they sound simply fantastic.)

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• FFmpeg not recording audio during screen capture

  - by King

  I'm using the script below to run FFmpeg on Ubuntu 10.10. I followed these instructions to install FFmpeg & x264. While ffmpeg does capture the screen it does not capture the mic audio. I've checked that the mic works via "System Preferences". Anyone have any ideas on what the problem(s) could be and suggestions on how to resolve this issue? Thanks. ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 -y screen-capture.mkv

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• Audio Panning using RtAudio

  - by user1801724

  I use Rtaudio library. I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use a duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. I seek on the web, but I did not find anything useful. Should I apply a filter on the output buffer? What kind of filter? Can anyone help me? Thanks

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• Looking for Non Hosted Audio & Video Podcasting Solution for Church Websites

  - by motboys

  I am looking for a solution that will do the following: User uploads audio and/or video files with title, desc. image etc Solution embeds info into ID3 tags Solution generates RSS feed Solution embeds new content in our website Content on website is searchable This is for a couple of church websites I manage. I am looking for the ability to do the above with a sermon mp3 and also a video. At the moment we are doing it with multiple steps / people involved and I want to automate the process. I can't seem to find a solution that does all of the above. Thank you!

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• Which API for cross platform mobile audio?

  - by deft_code

  This question focuses on the API's available on phones. I'd been planning to use OpenAL in my game for maximum portability. It runs great on Linux so I can quickly develop the Game and leverage it's superior debugging tools. However I've recently heard that Android doesn't support OpenAL well. Instead they've gone with a OpenSL ES library. What I'm looking for is a free Audio library that I can use with minimal custom code on iPhone, Android, and my Linux desktop. Does such an API exists? Some extra details: The game is written in C++ with custom minimal front ends. ObjC for iPhone, Java for Android, and SFML for Desktops. I'm using OpenGL ES for portability as iPhone doesn't support the more advanced OpenGL APIs.

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