similar to: Optimised ARM Ogg/Theora/Vorbis decoder

I've made a start writing a Theora player for Series 60 phones (mostly Nokia phones, but some Samsung and Sony Ericsson ones too). Download here: http://symbianoggplay.sourceforge.net/OgvPlay/OgvPlay_010.zip I've been using Big Buck Bunny, Elephants Dream and a few other ogv files for testing. I've uploaded 320x180 versions of those here:

---------- Forwarded message ---------- From: David Gerard <dgerard at gmail.com> Date: Fri, Feb 5, 2010 at 21:47 Subject: [whatwg] Suddenly, ~40% of IE users get HTML5 Theora with no effort To: WHATWG <whatwg at whatwg.org> http://www.atoker.com/blog/2010/02/04/html5-theora-video-codec-for-silverlight/

[Robin, sorry if you are receiving this for the third time, have subscribed to the list because previous attempt haven't hit the list] Hello, i took the liberty to port the theorarm-merge-branch to latest svn (used the git mirror actually). The changes already contains the fix Dave Milici posted to the list, plus it adds more fixes: - some warnings about unused variables in

Hi all, I've taken preglows ARM versions of qmf_synth and iir_mem16 from rockboxes speex codec, and tweaked them a bit further for some more speed. I attach them here so you can review and take on any changes you want. Please let me know if you have questions etc. Thanks, Robin -- Robin Watts, Email: <mailto:Robin.Watts@wss.co.uk> Warm Silence Software, WWW:

Hi all-- I tried building the ARM-optimized theora codec from the theorarm- merge-branch, and encountered the following compile and runtime problems before getting something to run. If there is another way to build it, it would be nice to know, but I got the sense that its current state in svn is incomplete. I'm using a gcc cross-compiler for ARM on an x86 Linux PC. After running

Hi all. Can any one please help me with my problem. I wrote an application for Android, using Teorarm. It crushes when I try to play the video file. Native application, compiled with the help of Android NDK 9r in Android Studio. Program stack: #0 theorapackB_readARM () at jni/theorarm/src/dec/bitwise2ARM.s:150 #1 0x673fae80 in oc_dec_headerin (_opb=0x6dca5278, _info=0x6dca52b0,

Hello all. I have got problem with compiling theorarm. I followed instructions posted earlier on this list (http://permalink.gmane.org/gmane.comp.multimedia.ogg.theora.devel/3066) but still got messages about undefined symbol: oc_frag_recon_inter2_arm. I have fixed Makefile.am by adding arm/ARMbitwise2.s \ arm/ARMdecode.s \ arm/ARMfilter.s \ arm/ARMfrag.s \ arm/ARMidct.s \ arm/ARMint.h \

hi all, i'm trying to code a tool to get some images from a ogg/theora file .... so basically i need to get a frame and convert to rgb ... and here is my question (sorry, i'm newbe here) ... how to convert a yuv_buffer to a rgb matrix ? i tryied diferent ways but my mistakes are bassically cause i don't understant how yuv_buffer structure works ... anybody can help? thanks a lot.

Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username:

On Sun, 17 Sep 2006, [UTF-8] Rafael Esp?ndola wrote: > The ARM has a multiply instruction of the form Rd=Rm*Rs where Rd != Rm. > How can I add this requirement to the instruction definition? Unfortunately there is currently no great way to do this. Some options are: 1. Pin one operand to a specific register, one that isn't allocated (e.g. R12). Use copyto/fromreg to get to it.

The ARM has a multiply instruction of the form Rd=Rm*Rs where Rd != Rm. How can I add this requirement to the instruction definition? Thanks, Rafael

Hello All, I'm having some trouble figuring out what the clearest way to plot my k-means clustering result on an my existing MDS. First I performed MDS on my distance matrix (note: I performed k-means on the MDS coordinates because applying a euclidean distance measure to my raw data would have been inappropriate) canto.MDS<-cmdscale(canto) I then figured out what would be my optimum

Hi, I've eventually released the first version of my rendering library for Kate streams, and I've recently been hacking away at adding it to liboggplay. That was the missing "overlay" mode to the browser_plugin patch I sent a while ago. Now, I'm not too sure how to expose those overlays to liboggplay users: the obvious solution is to add a RGBA video type, and expose the

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =

Hi I want to du a sensitivity analysis using Latin Hypercubes. But my parameters have to fulfill two conditions: 1) ranging from 0 to 1 2) have to sum up to 1 So far I am using the lhs package and am doing the following: library(lhs) ws <- improvedLHS(1000, 7) wsSums <- rowSums(ws) wss <- ws / wsSums but I think I can't do that, as after the normalization > min(wss) [1]

Hi Alexey, This is what works for me: [http.conf]: tlsenable=yes ; enable tls - default no. tlsbindaddr=144.x.y.z:8089 ; address and port to bind to - default is bindaddr and port 8089. tlscertfile=/etc/asterisk/keys/mycert.pem ; path to the certificate file (*.pem) only. tlsprivatekey=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from the wiki [1]. > So I'm using [2] live demo from

2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]: