similar to: Oggz use

Hi, I'm the main author of Speex (http://speex.sourceforge.net), an open-source speech codec that uses Ogg for file storage. I'd like to have feedback on the way we store Speex data in Ogg files to make sure we do the right thing. The Speex documentation can be found at http://speex.sourceforge.net/manual/ but here's the part about Ogg: <p><p>Speex bit-streams can be

Hi. I assume that I am deeply stupid and have missed something obvious, but why does this bare-bones code not generate a valid (If trivial) opus file? Running opusinfo I ge the following which I interpret this to mean that not even the first packet gets a pass: New logical stream (#1, serial: 1f0cce42): type opus WARNING: Could not decode Opus header packet 0 - invalid Opus stream (1) WARNING:

Okay, how do I drop a changeset/patchset/tag for you folks from SVN? At this point, I have written three examples of how to use the basics of the ogg streaming and decoding in Tremor. I heartily welcome any suggestions, improvements and corrections that you can point out in the code. The examples required me to make some small modifications to the main tremor library. However, the changes

Hey there I had implemented Speex encoding nicely in an application I was working on when I realized that I needed to change the API. I changed it, and the change somehow broke the speex encoding section of the code. The program writes SPX headers that other programs can read, but does not generate the rest of the file correctly. I have been through the guide "programming with

Good day. I have some questions dealing with merging Ogg Vorbis files. Suppose I want merge two Ogg Vorbis files encoded with the same parameters into one file that composed of one logical bitstream. In this connection some questions: A) Are codebooks in file headers depend only on Vorbis version and encoding parameters and don't depend on the particular source file? B) Are the following

Hi, I am creating a new ogg stream for theatrical/stage/disco lighting and am having trouble encoding my comment header with the following code in _tp_writelsbint function, it does not write the second byte to the ogg buffer. I am using windows and have created a new win32 library project with visual studio and added my code, what do i have to do to get the function working? Is there a project

Hello, I recently added discontinuous stream support to libogg1. The patch is attached. I also wrote Writ codec for libogg1 (based on original code by Arc), and sample Writ encoder (SubRip to Writ converter) and decoder. Is anybody interested? WBR, Roman. -------------- next part -------------- Index: include/ogg/ogg.h =================================================================== ---

I have a 1.1G Ogg file with vorbis and theora. oggz_seek_units() takes 14 seconds to find a position in the file towards the end. Now, the function guess() in oggz_seek() guesses a position at about 1.5G and then slowly searches back until it finds the end of the file (continously seeking beyond the end of the file and then calling read which returns 0). Then it does a linear scan from the

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.

On 23/02/15 10:30 AM, Tony wrote: > I __think__ opus-tools' opusenc has code which can be used as a > template/sample for how to create an .Ogg file with Opus content. As > expected, it makes use of libopus and libogg. Sadly that's the best option currently. As you say it's a bit complicated, although oggenc has a lot of extra support for controlling encoder options you

Hi. Le duodi 2 frimaire, an CCXIV, Wesley J. Landaker a ?crit?: > speexenc always crashes on amd64; here is an example, and lots of info: I tracked this bug a bit further than you. Here is the origin of the crash: op.packet = (unsigned char *)speex_header_to_packet(&header, (int*)&(op.bytes)); (from speexenc.c) op.bytes is a long, not an int, it is wrong to pass it as an int,

I'm currently working on a speex + theora webchat. I have both working alone, but to make them work together and to know where the package goes, I'm trying to use ogg packages. With theora there has been no problem, but with speex I can't make it work. The problem is that for a second it works great, but then it begings to gain lag, and in a little time it has 5-10 lag seconds (into

If by webchat you mean anything interactive, then I can already tell you what the problem is. TCP+Ogg. For anything real-time, you want UDP/RTP without Ogg. Also, you need to take into account the fact that computers do *not* have accurate clock and their soundcards drift. None of these problems are fatal, but you need to take that into account in the design. Jean-Marc On Mon, 2006-08-07 at

2008/10/29 Romain Beauxis <toots at rastageeks.org>: > > I am currently implementing theora for our application. > In our model for generating ogg streams, we may want to stop > a stream while not providing a new YUV data buffer for encoding. > > Current API doesn't allow such thing, since the eos flag is set by the > packetout function only when the last_p parameter

Hello all, I'm currently in the process of implementing a mixing matrix into the header of OPUS files for the proposed channel mapping 3 seen here: https://tools.ietf.org/html/draft-ietf-codec-ambisonics-03#section-3.2 My implementation simply replaces writing the stream mapping char array with the Q15 col-wise matrix vals described in the above document. I'm able to write/read matrices

Perhaps I am really dense, but after writing an app to encode multi-stream audio files using libfishsound and liboggz, I cannot figure out a way to reliably determine the which packet is the FINAL packet output from the encoder, so I can set the e_o_s flag on it. What is the magic for determining WHEN to set the e_o_s marker on a packet in the FishSoundEncoded callback function? Even though I

sorry, no webchat, is a videoconference program. And I'm using UDP as you can read down on the mail. The problem is that speex packages alone work great, but to split speex/theora packages on the arrive I need to use some kind of container, and I'm trying ogg. On 8/7/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > If by webchat you mean anything interactive,

I should also say the the ogg_packet.packet[] char array is zero-filled for packet mentioned. On Thu, May 18, 2017 at 12:19 PM Drew Allen <bitllama at google.com> wrote: > Hello all, > > I'm currently in the process of implementing a mixing matrix into the > header of OPUS files for the proposed channel mapping 3 seen here: >

Hi Drew, In what code are you implementing this? Note that I've been working on a library called libopusenc to make it easier to create Ogg Opus files. Maybe you want to have a look at that. The code is currently at: https://github.com/jmvalin/libopusenc Cheers, Jean-Marc On 18/05/17 03:19 PM, Drew Allen wrote: > Hello all, > > I'm currently in the process of implementing a

Greetings, I''ve had no success with mounting a vfat file system created by a Windows XP guest on a lvm volume. # mount -t vfat /dev/vg1/win1 /mnt/ mount: wrong fs type, bad option, bad superblock on /dev/vg1/win1, missing codepage or other error In some cases useful info is found in syslog - try dmesg | tail or so # dmesg FAT: invalid media value (0xb9) VFS: