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• Transcoding media server streaming to the iPhone

  - by pilif

  I have a huge collection of videos in different formats, but with one thing in common: They are not playable on an iPhone (or iPod Touch). Instead of complaining about Apple's IMHO broken world view ("there are no video formats but quicktime and mp4"), I wonder if there's a solution out there that allows streaming these different videos to the iPhone. This means that the source media needs to be transcoded on the fly. I already tried a few solutions out there, but with varying success: PS3 Media Server kind of worked, but only once and only for one single file. TVersity is said to work, but it requires UAC to be disabled and I don't see any need for this. The solution I'm looking for should run on Windows 2008 Server or Linux. I just can't believe that there's nothing out there that would allow me to stream my huge video collection on my iPhone (we're talking Wifi here, not 3G). After looking at the answers provided and after retrying TVersity without much success, I gave Orb another try and while the web interface failed to work for me, the iPhone Application (I tried the free one at first) actually worked flawlessly. And not only that, it also manages to convert the streams on-the-fly, so you don't have to wait for the transcoding process to finish before playback starts. On my 2.26 Ghz MacMini Server, this worked even with 1080p material. For Windows 2008 Server users out there: Remember to install the Desktop Experience Feature in the Server Manager if you intend this to work. Of all the stuff I had a look at, this really provided instant-success - even though I'm now probably sending the contents of my harddrive to orb's central server (sigh)

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• Why can't I use my Bluetooth Headset with my laptop?

  - by Michael Haren

  I just received a new Plantronics M50 bluetooth headset. It works great with my phone, but I can't get it working with my laptop. What are the things I should be checking? Here's what I've done so far: It pairs successfully: It's not in multipoint mode--it's only paired to my laptop I've installed all available drivers from Plantronics and Dell I have no (!) in Device Manager (though I don't see the headset there either--would I?) I can "configure" the headset by double clicking on it: "Allow the computer to turn off this device to save power" is unchecked in the Bluetooth radio settings Apps that let me choose the playback/mic device only list my laptop, not the headset [UPDATE] I went into the Bluetooth device's properties and Checked "headset" under the services tab. This was successful but hasn't delivered any functionality as far as I can tell I'd like to use this headset for VOIP conferencing (Goto meeting, Gmail voice chat, G+ hangouts, Skype, etc.) and listening to music (iTunes). Where else should I be digging? Is it possible that this new headset is simply not compatible with computers (i.e. it's only compatible with phones)?

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• h264 inside FLV container vs. MP4 container?

  - by Gotys

  I am developing a tube site, and currently having issues with h264 format . By looking at youtube, I noticed they are putting their hi-def videos into mp4 container, so logically I did the same. Next, I installed mod_h264_streaming for lighttpd to make streaming and timeline-scrubbing work. Problem is, that large files (500mb+ at somewhat high resolution) take for EVER to even start buffering ( I read the flowplayer or other flash players need to download metadata first) . I moved the xmov atom to the front of the file with MP4Box (i tried qt-quickstart too) , and the problem didn't go away. Next I read online I need to interleave audio tracks, so I did that too. No change in slowness. So I tried putting the same exact h264 movie into an FLV container, and the playback buffering starts almost instantly - no slowness. So what am I missing here? Why would I choose MP4 container with mod_264_streaming module , which seems super-slow over a regular FLV container with lighttpd's built-in mod_flv_streaming ? Obviously many websites pick mp4 container , but I fail to understand why ? And as a side question - I tried using HTML5's VIDEO tag to try the same h264 MP4 movie, and the scrubbing is LIGHTING FAST! I looked into lighttpd's log file, and i noticed taht Flash Players append video.mp4?start=234 each time timeline is scrubbed, wheres HTML5's video tag does no such thing . Is this some sort of limitations of Flash ? Why Can't flash streaming be same fast as HTML5 streaming? Thanks to ALL who can help. I very much appreciate this community.

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• Getting old bluetooth headset out of standby on Windows 7

  - by luagh45

  I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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• Problems setting up VLC Sever/client streaming

  - by Ayos

  I'm trying to set up a Linux machine as the server and a Windows XP machine as the client. Both machines are connected to the same local network via a Wi-Fi router. I setup the stream with the following properties : http stream port 8080 play locally And not much else. No firewall on the windows client(Windows firewall is disabled) When I try to open network stream via the client machine(Using VLC or Windows Media Player) I get the following errors: Media Player error code : 0xC00D11B3: Encountered a network Problem. VLC Console: main warning: connection timed out access_mms error: cannot connect to 192.168.1.3:8080 main debug: no access module matching "http" could be loaded main debug: TIMER module_need() : 12625.810 ms - Total 12625.810 ms / 1 intvls (Avg 12625.809 ms) main error: open of `http://192.168.1.3:8080' failed main debug: dead input main debug: repeating item main debug: starting playback of the new playlist item main debug: resyncing on http://192.168.1.3:8080 main debug: http://192.168.1.3:8080 is at 0 main debug: creating new input thread main debug: Creating an input for 'http://192.168.1.3:8080' main debug: using timeshift granularity of 50 MiB, in path 'C:\DOCUME~1\Accer\LOCALS~1\Temp' main debug: `http://192.168.1.3:8080' gives access `http' demux `' path `192.168.1.3:8080' main debug: creating demux: access='http' demux='' location='192.168.1.3:8080' file='\\192.168.1.3:8080' main debug: looking for access_demux module: 0 candidates main debug: no access_demux module matched "http" main debug: TIMER module_need() : 0.461 ms - Total 0.461 ms / 1 intvls (Avg 0.461 ms) main debug: creating access 'http' location='192.168.1.3:8080', path='\\192.168.1.3:8080' main debug: looking for access module: 2 candidates access_http debug: http: server='192.168.1.3' port=8080 file='' main debug: net: connecting to 192.168.1.3 port 8080 qt4 debug: IM: Deleting the input main debug: TIMER input launching for 'http://192.168.1.3:8080' : 13397.979 ms - Total 13397.979 ms / 1 intvls (Avg 13397.978 ms) qt4 debug: IM: Setting an input Need Help. Thanks in advance.

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• Is there a 'global media cache' in Windows 7 that may be used by third party media players?

  - by Pulse

  Here's the background. I don't use Windows Media Player or Media Centre, in fact both components have been 'turned off' via the 'Programs and Features' option. My media player of choice is a nightly build of MPC-HC, which plays virtually everything. I do, however, have VLC portable available for those rare instances when MPC-HC can't or won't play something correctly. This is the situation. I tend to download various media files via torrent, typically, game trailers or freely availably films, such as the recently released, torrent only, Pioneer One. Quite often these files are quite large, being 1GB+ so I quite often like to preview the file after it has downloaded a significant portion of the file. For the most part, this works quite well, and gives me an idea about the worth of continuing the download. Sometimes, however, the file doesn't play as expected and instead plays a completely unrelated file that has been previously played. Here's the strange thing. if I try to preview the file in MPC-HC or VLC both players play the same, previously played file, regardless of whichever player was originally responsible for playback. Most times, it's not even a file that's been played recently. I have searched the registry for some sort of MRU cache, but have found nothing. I have made sure each player has had it's respective history/cache deleted and can fine nothing on disk that seems to be storing this, apparently shared data. So, the question is, where are these unrelated players getting the file information from? Thnaks.

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• Windows 7 "freezes" (chills?), and then "unfreezes" for about 1 minute.

  - by gbc001

  Hi, I have an Acer Timeline 1810T netbook (4GB RAM) with Windows 7 x64. About once or twice a day, it "freezes" - the reason i put this in quotation marks is that it does not really freeze, as in you cant move mouse, etc. I can move my mouse and jump between different applications, but I cant use the applications for anything. So I can jump between notepad and Firefox, but I cant browse to a new web page. I have been trying to determine the source of this misery for a while now, and I suspect it has something to do with the hard drive - indirectly if not directly. Here are some screen shots of the resource monitor during a "freeze" and during normal operation: Freeze: http://imgur.com/Gcgq1.jpg Normal operation: imgur.com/mlHaI.jpg As you can see, CPU is fine during freeze, but the disk is going bananas.. Does anyone have an idea of what these reading mean, or about the problem in general? There seems to be no specific activity that sets this off - it can be during browsing, or during media playback with nothing else open. Very appreciative of any help!

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• Does VLC Player work well on Windows 7 64-bit?

  - by ????

  I tried VLC Player on Windows 7 64-bit version and the playback was fine, but when the video is maximized the image is very pixelated. I tried this on a Radeon HD 2600 XT graphics card as well as a computer with an Intel graphics chipset and both have the same result. If I use VLC Player on Windows 7 32-bit instead of 64-bit, there is no pixelation. Is this a known problem or is there any method to fix it on Windows 7 64-bit? Update: there isn't a 32-bit vs 64-bit version of VLC player, is there? (unlike 7-Zip) I also tried GOM Player and it doesn't have the problem on Windows 7 64-bit. Update: Nov 4, 2009 VLC displays an update notice: VLC 1.0.3 is a minor release fixing many bugs, especially for Windows Vista and 7, but it also introduces 2 new modes for deinterlacing, and a new udev module. Major fixes are about WMA Pro support, Dolby tracks in 4.0, v4l/v4l2 and atsc and a crash in mjpeg demuxer. Update of translations are also part of this release.

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• PSTN Trunk TDM400P Install on Asterisk / Trixbox

  - by Jona

  Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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• Flash AS3: (VideoEvent.COMPLETE, completePlay) - listener is triggered before video is completed

  - by Tevi

  Hello, I have a flash video using the standard FLV Playback component that comes with Flash. I'm using ActionScript 3 to modify the appearance and set up an event listener. I've set it up to go to a new URL using "externalInterface" when the video completes play. The URL is set in a variable using SWFObject. On only a few instances (3 people out of 50 - tested using Amazon Turk), people reported being taken directly to the new url, before the video even started playing. It's difficult to repeat the issue, but it did happen to me once. It doesn't have anything to do with cache, since it has been reported on people going to the url for the first time. Here's the url to the video: http://www.partstown.com/is-bin/INTERSHOP.enfinity/WFS/Reedy-PartsTown-Site/en_US/-/USD/ViewStaticPage-UnFramed?page=tourthetown Here's the code: import flash.external.*; import fl.video.*; var myVideo:FLVPlayback = new FLVPlayback(); var theUrl:String = this.loaderInfo.parameters.urlName; var theScript:String = this.loaderInfo.parameters.scriptName; myVideo.source = this.loaderInfo.parameters.videoPath;//"partstown.flv"; myVideo.skin = this.loaderInfo.parameters.skinPath;//"SkinUnderPlayStopSeekMuteVol.swf" myVideo.skinBackgroundColor = 0xAEBEFB; myVideo.skinBackgroundAlpha = 0.5; myVideo.width = 939; myVideo.height = 660; myVideo.addEventListener(VideoEvent.COMPLETE, completePlay); function completePlay(e:VideoEvent):void { myVideo.alpha=0.2; ExternalInterface.call(theScript); } addChild(myVideo); Why would the listener be triggered before the event complete? How can I fix it? Thanks!

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• Testing movie with Flash IDE fails to load file from localhost

  - by davgothic

  Hi, I'm just wondering if anybody can help me with my simple but frustrating problem. I have created an SWF that loads an XML file from http://localhost/flash/Projects/MEL/Quiz/Quiz/bin/xml/quiz.xml, but I get this error when running the movie using Test Movie in the Flash IDE. Error #2044: Unhandled ioError:. text=Error #2032: Stream Error. URL: http://localhost/flash/Projects/MEL/Quiz/Quiz/bin/xml/quiz.xml at Main/loadConfig()[D:\www\webroot\flash\Projects\MEL\Quiz\Quiz\src\Main.as:126] at Main/configLoadError()[D:\www\webroot\flash\Projects\MEL\Quiz\Quiz\src\Main.as:143] at flash.events::EventDispatcher/dispatchEventFunction() at flash.events::EventDispatcher/dispatchEvent() at flash.net::URLLoader/onComplete() The error I get if I handle the exception is: [IOErrorEvent type="ioError" bubbles=false cancelable=false eventPhase=2 text="Error #2032: Stream Error. URL: http://localhost/flash/Projects/MEL/Quiz/Quiz/bin/xml/quiz.xml"] Trouble is running the SWF in a browser locally does work, it only throws these errors in the Flash IDE. I have tried a adding wildcard crossdomain.xml file in my root web directory and setting the SWF publish properties for local playback security to Allow network only, but neither of these have solved my problem. I know Windows 7 handles localhost name resolution differently compared to previous versions of Windows but I have even added 127.0.0.1 localhost to my hosts file to no avail. Can anyone shed any light on this issue?

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• MP3 Decoding on Android

  - by Rob Szumlakowski

  Hi. We're implementing a program for Android phones that plays audio streamed from the internet. Here's approximately what we do: Download a custom encrypted format. Decrypt to get chunks of regular MP3 data. Decode MP3 data to raw PCM data in a memory buffer. Pipe the raw PCM data to an AudioTrack Our target devices so far are Droid and Nexus One. Everything works great on Nexus One, but the MP3 decode is too slow on Droid. The audio playback starts to skip if we put the Droid under load. We are not permitted to decode the MP3 data to SD card, but I know that's not our problem anyways. We didn't write our own MP3 decoder, but used MPADEC (http://sourceforge.net/projects/mpadec/). It's free and was easy to integrate with our program. We compile it with the NDK. After exhaustive analysis with various profiling tools, we're convinced that it's this decoder that is falling behind. Here's the options we're thinking about: Find another MP3 decoder that we can compile with the Android NDK. This MP3 decoder would have to be either optimized to run on mobile ARM devices or maybe use integer-only math or some other optimizations to increase performance. Since the built-in Android MediaPlayer service will take URLs, we might be able to implement a tiny HTTP server in our program and serve the MediaPlayer with the decrypted MP3s. That way we can take advantage of the built-in MP3 decoder. Get access to the built-in MP3 decoder through the NDK. I don't know if this is possible. Does anyone have any suggestions on what we can do to speed up our MP3 decoding? -- Rob Sz

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• How to download a wav file from the web to a location on iPhone using NSFileHandle and NSURLConnecti

  - by Jinru

  Hi all, I want to download a wav file from a web service, cache it on the iphone and playback it using AVAudioPlayer. Using NSFileHandle and NSURLConnection seems a viable solution when dealing with relatively large files. However, after running the app in the simulator I don't see any saved file under the defined directory (NSHomeDirectory/tmp). Below is my basic code. Where am I doing wrong? Any thoughts are appreciated! #define TEMP_FOLDER [NSHomeDirectory() stringByAppendingPathComponent:@"tmp"] - (void)downloadToFile:(NSString*)name { NSString* filePath = [[NSString stringWithFormat:@"%@/%@.wav", TEMP_FOLDER, name] retain]; self.localFilePath = filePath; // set up FileHandle self.audioFile = [[NSFileHandle fileHandleForWritingAtPath:localFilePath] retain]; [filePath release]; // Open the connection NSURLRequest* request = [NSURLRequest requestWithURL:self.webURL cachePolicy:NSURLRequestUseProtocolCachePolicy timeoutInterval:60.0]; NSURLConnection* connection = [[NSURLConnection alloc] initWithRequest:request delegate:self]; } #pragma mark - #pragma mark NSURLConnection methods - (void)connection:(NSURLConnection *)connection didReceiveData:(NSData*)data { [self.audioFile writeData:data]; } - (void)connection:(NSURLConnection *)connection didFailWithError:(NSError*)error { NSLog(@"Connection failed to downloading sound: %@", [error description]); } - (void)connectionDidFinishLoading:(NSURLConnection *)connection { [connection release]; [audioFile closeFile]; }

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• How to stream partial content with ASP.NET MVC FileStreamResult

  - by o_o

  We're using a FileStreamResult to provide video data to a Silverlight MediaElement based video player: public ActionResult Preview(Guid id) { return new FileStreamResult( Services.AssetStore.GetStream(id, ContentType.Preview), "application/octet-stream"); } Unfortunately, the Silverlight video player downloads the entire video file before it starts playing. This behavior is expected as our Preview Action does not support downloading partial content. (side note: if the file is hosted in an IIS virtual directory we can start playback at any location in the video while it is still downloading. however for security and auditing reasons we can't provide a direct download link. so this is not an option.) How can we improve the Controller Action to support partial HTTP content? I assume we first have to inform the client that we support it (adding an "Accept-Ranges:bytes" header to a HEAD request), then we have to evaluate the HTTP "Range" header and stream the requested file range with a response code of 206. Will that work with ASP.NET MVC hosted on IIS6? Is there already some code available? Also see: http://en.wikipedia.org/wiki/List_of_HTTP_headers http://blogs.msdn.com/anilkumargupta/archive/2009/04/29/downloadprogress-downloadprogressoffset-and-bufferprogress-of-the-mediaelement.aspx http://benramsey.com/archives/206-partial-content-and-range-requests/

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• SoundManager2 has irregular latency

  - by Stefan Monov

  I'm playing some notes at regular intervals. Each one is delayed by a random number of milliseconds, creating a jarring irregular effect. How do I fix it? Note: I'm OK with some latency, just as long as it's consistent. Answers of the type "implement your own small SoundManager2 replacement, optimized for timing-sensitive playback" are OK, if you know how to do that :) but I'm trying to avoid rewriting my whole app in flash for now. For an example of app with zero audible latency see the flash-based ToneMatrix. Testcase (see it here live or get it in an zip): <head> <title></title> <script type="text/javascript" src="http://www.schillmania.com/projects/soundmanager2/script/soundmanager2.js"> </script> <script type="text/javascript"> soundManager.url = '.' soundManager.flashVersion = 9 soundManager.useHighPerformance = true soundManager.useFastPolling = true soundManager.autoLoad = true function recur(func, delay) { window.setTimeout(function() { recur(func, delay); func(); }, delay) } soundManager.onload = function() { var sound = soundManager.createSound("test", "test.mp3") recur(function() { sound.play() }, 300) } </script> </head> <body> </body> </html>

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• Actionscript: NetStream stutters after buffering.

  - by meandmycode

  Using NetStream to stream content from http, I've noticed that esp with certain exported h264's, if the player encounters an empty buffer, it will stop and buffer to the requested length (as expected). However once the buffer is full, the playback doesn't resume, but instead jumps ahead, as such- instantly playing the buffered duration in a brief moment, and thusly triggering an empty buffer again.. this will then continue over and over. Presumably when the netstream pauses to buffer, the playhead position continues, and the player is attempting to snap to that position on resume- however given it could take 5 seconds to build a 2 second buffer- it ends up with a useless buffer again.. (this is an assumption) I've attempted to work around this by listening for an empty buffer netstatus event, pausing the stream, and at the same time setting up a loop to check the current buffer length vs the requested buffer length.. and resuming once the buffer length is greater than or equal to the requested buffer.. however this causes problems when there isn't enough of the video remaining.. for example, a 10 second buffer with only 5 seconds remaining, the loop just sits there waiting for a buffer length of 10 seconds when theres only 5 left... You would think that you could simply check which was smaller, the time left or the requested buffer length.. however the times flash gives are not accurate.. If you add the net streams current time index, plus the buffered time, the total is not the entire duration of the movie (when at the end).. it is close but not the same. This brings me back to the original problem, and if there is another way to fix this, clearly flash knows when the buffer is ready, so how can i get flash pause when it buffers, and resume once the buffer is ready? currently it doesn't.. it pauses and then once the buffer is full- it plays the entire buffered content in about .1 of a second. Thanks in advance, Stephen.

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• AVAudioPlayer making noise when playing multiple sounds at the same time

  - by Rob

  I am having an issue where AVAudioPlayer is introducing noise into playback ONLY when I play multiple sound files at the same time. If I play them each individually, they all sound perfect. But, if I play sound clip B while sound clip A is still playing, the speakers start crackling like there is noise. I have tried both m4a files AND caf files and both make the same noise, so it has to be something with how I am implementing this method or a quirk with AVAudioPlayer. Any insights? code I am using: UITouch* touch = [[event allTouches] anyObject]; NSString* filename = [soundArray objectAtIndex:[touch view].tag]; NSString *path = [[NSBundle mainBundle] pathForResource:filename ofType:@"m4a"]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely theAudio.delegate = self; [theAudio prepareToPlay]; [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio play];

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• Browser Based Streaming Video/Audio (not progressive download)

  - by Josh

  Hello, I am trying to understand conceptually the best way to deliver real streaming audio and video content. I would want it to be consumed with a web browser, utilizing the least amount of proprietary technology. I wouldn't be serving static files and using progressive download, this would be real audio streams being captured live. How does one broadcast a stream that will be reasonably in sync with the source? What kind of protocol is suitable? Edit: In research I've found that there are a few protocols: RTSP, HTTP Streaming, RTMP, and RTP. HTTP streaming is somewhat unsuitable if you are streaming a live performance/communication of some kind because it relies on TCP (as its HTTP based) and you don't lose packets. In a low bandwidth situation, the client can get significantly behind in playback. ref RTMP is a proprietary technology, requiring flash media server. Crap on that. The reason I looked at flash is because they are extremely flexible as far as user experience goes. SoundManager2 provides an excellent javascript interface for playing media with flash. This is what I would look for in a client application. RTSP/RTP is what Microsoft switched to using, deprecating their MMS protocol. RTSP is the control protocol. Its similar to HTTP with a few distinct difference -- server can also talk to the client, and there are additional commands, like PAUSE. Its also a stateful protocol, which is maintained with a session id. RTP is the protocol for delivering the payload (encoded audio or video). There are a few open sourced projects, one of them being supported by apple here. It seems like this might do what I want it to, and it looks like quite a few players support it. It sounds like it would be suitable for a "live" broadcast from this page here. Thanks, Josh

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• Using kAudioSessionProperty_OtherMixableAudioShouldDuck on iPhone

  - by Cliff

  Hello, I'm trying to get consistant behavior out of the kAudioSessionProperty_OtherMixableAudioShouldDuck property on the iPhone to allow iPod music blending and I'm having trouble. At the start of my app I set an Ambient category like so: -(void) initialize { [[AVAudioSession sharedInstance] setCategory: AVAudioSessionCategoryAmbient error: nil]; } Later on when I attempt to play audio I set the duck property using the following method: //this will crossfade the audio with the ipod music - (void) toggleCrossfadeOn:(UInt32)onOff { //crossfade the ipod music AudioSessionSetProperty(kAudioSessionProperty_OtherMixableAudioShouldDuck,sizeof(onOff),&onOff); AudioSessionSetActive(onOff); } I call this passing a numeric "1" just before playing audio like so: [self toggleCrossfadeOn:1]; [player play]; I then call the crossfade method again passing a zero when my app's audio completes using a playback is stopping callback like so: -(void) playbackIsStoppingForPlayer:(MQAudioPlayer*)audioPlayer { NSLog(@"Releasing player"); [audioPlayer release]; [self toggleCrossfadeOn:0]; } In my production app this exact code works as expected, causing the ipod to fade out just before playing my app's audio then fade back in when the audio finishes playing. In a new project I recently started, I get different behavior. The iPod audio fades down and never fades back in. In my production app I use the AVAudioPlayer where in my new app I've written a custom audio player that uses audio queues. Could somebody help me understand the differences and how to properly use this API?

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• Frame Accurate Browser Launchable Video Player ... ?

  - by cliftonc

  I have a requirement where I need to enable playback (full screen) of a h.264 MPEG4 (thanks for the correction!) video from a local network, launchable from a browser link on a Windows workstation, and be frame accurate. By frame accurate I mean that I need to be able to scrub through the video in the same way you would with a vtr, stop at a frame, and then move backwards and forwards frame by frame (it is for a very specific compliance requirement where have to be able to check every frame if there is something that is potentially against broadcasting guidelines). The application itself is used to capture notes while viewing the material, so the end model is for a dual monitor workstation, with a web form in one, the video playing full screen in the second (no issue launching the video and manually having to move it to the second screen), and then the user controls the video via keyboard shortcuts or a jog shuttle. I have looked at QT, but the java bindings seem to be dead or nearly so, flash isn't frame accurate, VLC given its streaming heritage seems to be only able to move forward by a frame and not backwards, and all I have left are commercial offerings that in my experience are difficult and expensive to change. Any ideas of where I should look or alternative options? Any advice appreciated!

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• Play multiple audio files using AVAudioPlayer

  - by inScript09

  Hi all, I am planning on releasing 10 of my song recordings for free but bundled in an iphone app. They are not available on web or itunes or anywhere as of now. I am new to iphone sdk (latest) as you can imagine, so I have been going through the developer documentation, various forums and stackoverflow to learn. Apple's avTouch sample application was a great start. But I want my app to play all the 10 tracks one by one. All the songs are added to resources folder and are named as track1, track2...track10. In the avTouch app code I can see the following 2 parts which is where I think I need to make changes to achieve what I am looking for. But I am lost. // Load the array with the sample file NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: [[NSBundle mainBundle] pathForResource:@"sample" ofType:@"m4a"]]; - (void)audioPlayerDidFinishPlaying:(AVAudioPlayer *)player successfully:(BOOL)flag { if (flag == NO) NSLog(@"Playback finished unsuccessfully"); [player setCurrentTime:0.]; [self updateViewForPlayerState]; } can anyone please help me on 1. how to load the array with all the 10 tracks which are added to resources folder 2. and when I hit play, player should start the first track. when the 1st track ends 2nd track should start and so on for the remaining tracks. Thank You

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• AVAudioPlayer working in Simulator, but not on device

  - by cannyboy

  My mp3 playing code is: NSError *error; soundObject = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:audioPathString] error:&error]; if (soundObject == nil) NSLog(@"%@", [error description]); soundObject.delegate = self; soundObject.numberOfLoops = 0; soundObject.volume = 1.0; NSLog(@"about to play"); [soundObject prepareToPlay]; [soundObject play]; NSLog(@"[soundObject play];"); The mp3 used to play fine, and it still does on the simulator. But not on the device. I've recently added some sound recording code (not mine) to the software. It uses AudioQueue stuff which is slightly beyond me. Does that conflict with AVAudioPlayer? Or what could be the problem? I've noticed that as soon as the audiorecording code starts working, I can't adjust the volume on the device anymore, so maybe it blocks the audio playback?. EDIT The solution seems to be to put this in my code. I put it in applicationDidFinishLaunching: [[AVAudioSession sharedInstance] setCategory: AVAudioSessionCategoryPlayAndRecord error: nil]; UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker; AudioSessionSetProperty (kAudioSessionProperty_OverrideAudioRoute,sizeof (audioRouteOverride),&audioRouteOverride); The first line allows both play and record, whilst the other lines apparently reroute things to make the volume louder. All audio code is voodoo to me.

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• html5 vs flash - full comparison chart anywhere?

  - by iddqd

  So since Steve Jobs said Flash sucks and implied that HTML5 can do everything Flash can without the need for a Plugin, I keep hearing those exact words from a lot of People. I would really like to have a Chart somewhere (similar to http://en.wikipedia.org/wiki/Comparison_of_layout_engines_%28HTML5%29#Form_elements_and_attributes ) that I can just show to those people. Showing all the little things that Flash can do right now, that HTML5/Ajax/CSS is not yet even thinking about. But of course also the things that HTML5 does better. I would like to see details compared like audio playback, realtime audio processing, byte level access, bitmap data manipulation, webcam access, binary sockets, stuff in the works such as P2P technology (adobe stratus) and all the stuff I don't know about myself. Ideally with examples of what can be accomplished with, lets say Binary Sockets (such as a POP3 client) because otherwise it won't mean a lot to non-programmers since they will just say "well we can do without Binary Sockets". And ideally with some current benchmarks and some examples of websites that use this technology. I've searched the web and am surprised not to find anything. So is there such a comparison somewhere? Or does anybody want to create this and post it to Wikipedia? ;-)

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• Progressive MP4 video issues in Flash- Video stops rendering

  - by Conor

  I'm currently working on a flash project that has an intro video that plays before heading into the main app. This video is an H.264 .mp4, 1550x540, and around 10MB. The problem thats currently driving me insane is that when I test it, occasionally the video will begin playing, and then suddenly stop rendering the video frames, leaving the audio playing in the background with nothing on screen. Once the file is played through fully (based on listening to the audio), my playback complete event fires like it should, but I can't find any info of people having similar issues. Attached is a trace of the .mp4 metadata in case that helps. videoframerate : 24 audiochannels : 2 audiocodecid : mp4a audiosamplerate : 48000 trackinfo: 0: length : 608000 timescale : 24000 language : eng sampledescription: 0: sampletype : avc1 1: length : 1218560 timescale : 48000 language : eng sampledescription: 0: sampletype : mp4a duration : 25.386666666666667 width : 1540 videocodecid : avc1 seekpoints: 0: time : 0 offset : 13964 1: time : 0.333 offset : 16893 2: time : 0.667 offset : 34212 ... 73: time : 24.333 offset : 9770329 74: time : 24.667 offset : 9845709 75: time : 25 offset : 9895215 moovposition : 32 height : 540 avcprofile : 77 avclevel : 51 aacaot : 2 This has been driving me absolutely insane... any help would be much appreciated!

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• blackberry implement audio player

  - by Prasad

  Hi, I am developing an application which let users to hear songs online. And I used Blackberry Player and Manager APIs. My application works fine and I can play songs. Now I wan't to add more controls to it. As an example I want pause, play songs. Mute the sound, Control the volume. Display the progress of the play back. Display the current time position of the song like that. I started research on that. And I tried to do that with PlayerListener. But unfortunately all the time I am getting IllegalStateException. So I can't go ahead with that research. As a help can someone please tell me how can I implement above kind of controls for a player. Appreciate if someone can post a sample code to do that. Further I will put my playback source code here. public void run() { try { p = Manager.createPlayer(requestedSong + SystemSettings.strNetwork); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } public void run() { try { p = Manager.createPlayer(strSongURL); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } Thank you very much. Prasad

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