similar to: streaming and VBR

Hi there, for a streaming application developed for a project at the Society for Arts and Technology Montreal, Canada, we would need to know the exact delay of our ogg/vorbis streams. It seems there are several factors that make up the total delay I get. Is there a way to find out how large the delay introduced by the encoder and decoder is? Or in other words, how many unprocessed samples

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

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

--- 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

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.

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

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

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

Hi, Maybe this is documented somewhere, if so, please send me a link to the documentation. My question is: how to set up different (VBR, CBR, ABR) modes when calling the Ogg Vorbis API? Currently I do: CBR: ret = vorbis_encode_setup_managed( &vorbisInfo, getInChannel(), getOutSampleRate(),

Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr => true)? 2) If there relation between setting the vbr => true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value?

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

Hi, Is there any difference between decoding of CBR or VBR streams in realtime in terms of CPU usage/cost? Are sample rates other than 44100Hz supported by Vorbis officially ? If it is supported, is it efficient to use them or is Vorbis specifically (or at the moment) optimized for 44100Hz ? Thanks in advance. --- Mete BALCI Senior Game Programmer Momentum DMT Istanbul, TURKEY

I'm writing a voice communication application, and I've got a few issues that I'd like to get ironed out, but I don't know enough about the speex implementation. First of all, this application is mainly used for conferencing - many people are in a room and only 1-2 are ever talking at a time. So, always encoding and transmitting everyone's audio stream would be rather

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.