similar to: playing vorbis file stored in a buffer array

----- Forwarded message from Telford Tendys <telford@triode.net.au> ----- Delivery-Date: Thu Jan 25 22:22:57 2001 Date: Fri, 26 Jan 2001 16:26:27 +1100 From: Telford Tendys <telford@triode.net.au> To: feedback@vorbis.com Subject: Comments on your code User-Agent: Mutt/1.2.5i Dear Vorbis, I didn't even know about ogg or vorbis until I was at a conference and someone said, ``if

Hi, I'm wondering if the plans for streaming support include on demand streams via rtsp -- not just a multicast of one stream like a radio station (icecast)?

I'm trying to adapt decoder_example to to support more of an event driven model. I'm building an RTSP client/server where the client will have a function called everytime a new packet comes in. So, I want to queue up these packets to be played. My current attack is as follows (note this isn't very robust and I realized that): For first and second packet i grab all the header stuff

Another one at block.c:176: --- Title: We The People Artist: DJ Lithium Presents Bitstream is 2 channel, 44100Hz Time: 58:29.07, Bitrate: 100.1 Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 1024 (LWP 27207)] _vds_shared_init (v=0xbffff73c, vi=0x4024efe0, encp=0) at block.c:176 176 b->modebits=ilog2(ci->modes); (gdb) bt #0 _vds_shared_init

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

I am trying to add a package (I know its older - but it should work) gst-rtsp-server-0.10.8 to CentOS 7. The gstreamer 0.10 packages are included in C7 and they compile just fine. When I extract and try to compile gst-rtsp-server 0.10.8 the ./configure goes fine. but the make results in errors: make make all-recursive make[1]: Entering directory

Hello all! I am wondering if anyone can help me locate some information concerning streaming Vorbis files. I am needing to locate the information for a multimedia networking class I am taking so I can learn all this audio and networking mumbo-jumbo. For the class, I need to locate 3 streaming media types and answer the following questions: Does it use Meta

Greetings one and all; I'm not subscribed to this list so I'm first sending this message to verify that mails from me make it through, and then later I'll send the juicy messages with patches. Also, the address I'm using is IPv6-only and doesn't often work, so drop me from any replies and I'll catch the archives, or drop only the hostname part to get an IPv4 address that

Are oggdec and ogg123 supposed to be able to extract or play audio from an Ogg Theora+Vorbis file? OggDec 1.0 and ogg123 from vorbis-tools 1.0.1 don't seem to be able to do so. If the programs are not supposed to do this job now, are any improvements planned so they can do this job in the future? Thanks for info on this.

Hi, There is a bug in the shuffle of ogg123 which was here at least until the CVS version of 3 days ago. Here's our fix: (it seems to be not perfectly equiprobable but at least fixes the old one which sometimes gives two times the same song and never the other song) -=-=-- --- vorbis-tools-1.0beta3/ogg123/ogg123.c.pix Sat Nov 4 01:23:21 2000 +++ vorbis-tools-1.0beta3/ogg123/ogg123.c Thu

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain

Hi Where do I find instruction to configure icecast as a reflect server for rtsp? Scenario: Client1 sends audio to IceCast via rtsp Client2 receives audio from IceCast via rtsp. I need to just setup this behaviour.. -Atul -- ------- To make world a better place to live in ------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

alpha. (that about sums it up) Will encode and play back via an sdp file and multicast on one computer (over the net if you transfer the sdp file over by hand). rc/rtenc3 and src/rtdec3 are in the style of encoder/decoder_example ...so this means you MUST edit it them to suit your system. For example, I specify my rtenc3 to multicast just on eth1 (to avoid pissing off my cable supplier on

I got this one this morning, but didn't notice it til now: --- (gdb) bt #0 0x4003be91 in vorbis_block_clear (vb=0xbffff71c) at block.c:156 #1 0x4003ce36 in vorbis_analysis_blockout (v=0xbffff71c, vb=0x8050b80) at block.c:591 #2 0x4002ff23 in _prefetch_all_headers (vf=0xbffff540, dataoffset=-1073749296) at vorbisfile.c:324 #3 0x4003039c in _process_packet (vf=0xbffff540, readp=1)

hi I am getting this error: -- Executing [50 at smvoice-sip:1] Answer("SIP/440-0856dd70", "") in new stack -- Executing [50 at smvoice-sip:2] rtsp("SIP/440-0856dd70", "rtsp://192.168.1.175/img/video.sav") in new stack [Apr 30 11:22:48] WARNING[8031]: app_rtsp.c:1037 rtsp_play: >rtsp play [Apr 30 11:22:48] NOTICE[8031]: rtp.c:1287 ast_rtp_read:

On Thu, 2020-09-24 at 16:57 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 16:31:33, hw wrote: > > > Hi, > > > > is it possible to "call" an IP camera? > > Only if it talks SIP (which some do, generally door entry cameras with a > push > button input and often a lock release output). > > > I'm thinking about something

Hi, I saw the documentation "RTP Payload Format for Vorbis Encoded Audio". My question: The Darwin Streaming Server (OpenSource and free for several plattforms) streams media with RTSP/RTP. For more information: http://www.publicsource.apple.com/projects/streaming/ I know that a QuickTime Component of Ogg Vorbis (compressor/decompressor) in progress. (Thanks guys) Could I stream Ogg

Hi, does anybody know if somebody tried implementing things like rtsp, mms or similar in icecast so far? I heard some rumors from people complaining transmissions over http or not "that secure" because using a wget you can easily write them to disk. They suggested using real servers or something - but I wonder if its (easy?) possible to implement another already widely used

I have a gst-rtsp-server 0.10.8.2 that I am trying to compile on CentOS 7.2 I get this error: rtsp-media.c error 'g_cond_free' is deprecated (declared at /usr/include/glib-2.0/glib/deprecated/gthread.h:283) [-Werror=deprecated-declarations] What can I do about that to compile this package (I'm compiing from source) ? Thanks Jerry

Hello, I am trying to stream an IP Camera output (h264) into a conference. The IP Camera supports RTSP. Searching around the web, I believe the RTSP support (was) available through app_rtsp (external to Asterisk distribution). This, I believe, has problems and has issues compiling in Asterisk 11 (I tried compiling it in Asterisk 11 and it failed). I may not be able to use DiaStar or i6net's