similar to: timestamping / latency

Hi list, I have a very anoying problem when streaming ogg vorbis: due to the variable bitrate encoding data rates sometimes drop to a very low value when there is no audio input signal. After a while the server (Icecast2) just drops my stream because no new ogg pages arrive. For streaming I use a software that I have written but I can't find a way to tell it to never drop data rate below a

Hi, While building an ogg-vorbis stream encoder, I encountered some problems with silence in the audiostream. The bitrate drops to almost zero, and pages going out less then ones a minute what makes the stream to stop / buffer. In earlier postings I read that I shout use ogg_stream_flush as an alternative to ogg_stream_pageout, in case of silence. My question is this; Is there any

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Erik de Castro Lopo wrote: > > > Josh et al, > > > > I've been tracking down a problem with generating OggFlac files. > > <snip> > > > I have looked into this and suspect that the > FLAC__stream_encoder_finish() > > function needs to finish off the stream using a call to

Josh et al, I've been tracking down a problem with generating OggFlac files. While investigatint this issue I hacked one of the test files from the test_libFLAC directory of the flac-1.1.3-beta2. The result of this hacking is this standalone C file: http://www.mega-nerd.com/tmp/erikd_test.c Directions for compiling this is in the comments at the top of the file. When the program is

1.) after initializing a ogg_stream_state may I just keep calling ogg_stream_packetin with valid ogg_packet's until no more ogg_packets needs to be added? Or do I need to call ogg_stream_pageout after every ogg_stream_packetin? Iaw may I delay calling ogg_stream_pageout until there are no more packets to be added? 2.) I assume I have call ogg_stream_pageout until no more pages can be

Hi, after banging my head against the wall for quite a while I finally found what crashes my code: Because I'm working with Codewarrior 8 on OS X I had to compile the ogg / vorbis libs with CW. To do so I used the CW projects for OS 9 and modified them. Howerver, they don't work as expected. Everything seems to be fine, encoding & decoding works, only vorbis_comment_add_tag() crashes

I was wondering. When doing ogg only encoding, how do you mark end of stream? I am setting ogg_packet.e_o_s and then submitting it via ogg_stream_packetin() and then, because its the last potential page ogg_stream_flush()ing it. But when I walk through the file, EOS is not set on that logical bitstream. I have been using oggenc.c/encoder.c and oggdec.c/decoder.c for examples, but because the

I need it to be able to take a stream of PCM data and encode it. Right now it works if I am recording from a source (i.e. microphone or with "what you hear") but when I try to use another API (Team Speak) to capture the PCM data it doesn't seem to be encoding correctly. (I know the PCM data is correct. I am able to use it to create a wav file.) I'm also having a problem with

dear list, i am trying to write a small program which reads an ogg file and writes it to another ogg file (and changes serial number, granulepos etc on the fly). reading the ogg file is ok (ogg_sync_pageout, ogg_stream_pagein, ogg_stream_packetout). but writing the file doesn't work - the granulepos and page structures don't match with the original file. here's what i am doing.

hi, in my quest to grok ogg i tried to write a small program which extracts ogg_packets, and stuffs them back into ogg pages (which are then written to a file). extracting the packets works fine, but i have problems stuffing them into a new ogg stream. pseudo code: ogg_stream_init(os, serialno) foreach p in packets: ogg_stream_packetin(os, p) // try to extract a page

I'm trying to learn how to use libtheora but I'm having some problems: I believe to have encoded a video, but I can't decode it. I've tried 2 things: to record a .ogg file and to write a libtheora decoder, using the encoded ogg_packets present in memory. The source code is here: http://opensvn.csie.org/ribamar/projects/streaming/cvaenc.c (it has a makefile in

Hi, I have just integrated libshout2 and libvorbisenc into mserv, my music server (www.mserv.org). It all works well. Thanks for a great product. The problem I have is that when there is a pause between songs of around 10 seconds, Winamp (2.91) decides (due to the 64kbps dropping to a 2kbps stream I guess), that it needs to re-buffer. It fails miserably and generally dumps the connection, or

Hi everyone, I'm trying to build theora encoding into my game engine. Essentially, I'd like to do something similar to png2theora.c, where I take individual screen shots of my game and encode them into a video. Because png2theora.c uses the legacy C API, I used encoder_example.c as a template instead. encoder_example.c seems to be the only code out there that actually uses the th_* API.

Hello Does anyone have experience with _real_ real time streaming - i.e. with very little delay/latency? I need to stream from point A to point B in near-CD quality via a 100 Mbit network. That is easily done using icecast. But here is the tricky bit: I want as little delay in the signal as possibble - preferrably below 50ms! I have made a test setup encoding on and serving from an 800 MHz

On Thursday 05 February 2004 12:55, Jørgen Elgaard Larsen wrote: > Hello > > Does anyone have experience with _real_ real time streaming - i.e. with > very little delay/latency? > > I need to stream from point A to point B in near-CD quality via a 100 > Mbit network. That is easily done using icecast. But here is the tricky > bit: > > I want as little delay in the

Okay - so I stepped through the code slowly this morning making notes as I went. And I can see what’s wrong. The last Ogg header of the troublesome file contains -1 as the granule position. The spec says this has a special meaning "A special value of '-1' (in two's complement) indicates that no packets finish on this page." 4F 67 67 53 00 04 FF FF FF FF FF FF FF FF 16 A2

Hi everybody I'm trying to encode the soundcard signal to serve it with IceCast, but I'm having problems with making darkice-0.11. Anybody could help? I'm compiling in a Solaris 6 OS with gcc 2.8.1, and I've installed lame and Vorbis libraries. This is my config phrase: ./configure --prefix=/aplicacions/icecast-1.3.12/darkice --with-lame --with-vorbis

Dear R-help, We are using the Platform AMD 64 Bit Opteron. Our operating system is SUSE LINUX ENTERPRISESERVER SLES9.0 We are using R version 2.2.0 . This is our problem: None of the device drivers postscript() or pdf() is opening, e.g. the following message is coming: postscript("splines-Ex.ps") Fehler in postscript("splines-Ex.ps") : unable to start device PostScript

Hi Developers, I have a 16-bit PCM data buffer, I want to write that to ogg file. Could you help me to understand how to write pcm data to the ogg? Thanks in advance. Arun balaji -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20150518/6c8c9a63/attachment.htm

I want to compile ogg and vorbis(version 1.01) under mingw. In the console i typed ./configure make but there is a lot of error while making. here is the errors $ ./configure checking for a BSD-compatible install... /bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether to enable maintainer-specific