similar to: Vorbis files merging

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:

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

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,

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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

Hi there, First of all, sorry for my English I'm not a native english speaker. I need to develop and application witch does 3 tasks with oggs files : 1/ chain 2 ogg files 2/ extract a part from an ogg file 3/ add silence between two chained oggs files. Basically, - For the 1st task, I think that the best way to chain ogg files is to use the "cat" (on linux) or "copy

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

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

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'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

2009/6/18 Adam Rosi-Kessel <adam at rosi-kessel.org>: > Ivo Emanuel Gon?alves wrote, on 6/18/2009 6:14 AM: >> >> On 6/18/09, Conrad Parker<conrad at metadecks.org> ?wrote: >>> >>> This sounds like it needs a new tool specifically for fixing broken >>> Ogg files. >> >> I see two solutions: >> >> 1) the new vcut which I

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.

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

Greetings, How does the ogg_page_bos function work? I call this function directly after calling ogg_sync_pageout. It works. Then I place a guard around it until the start of a new stream (chain mode). If this is the start of a new stream, should it return zero after the next call. I looked in ogg_page structure to see if there is a member named b_o_s, but I did not see anything. What is being

Hi, I was wondering what oggz_get_bos and oggz_get_eos are supposed to do, specifically are they meant to be used in oggz_read mode? The documentation says "Determine if a given logical bitstream is at [bos/eos]". However if get_bos is given a specific serialno it returns that stream's b_o_s flag which, in read mode, is set on initialisation. The serialno=-1 (for all streams)

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

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

Mandag, 2 juni 2003, skrev du: >On Mon, 02 Jun 2003 21:22, Vitaly Nechyos wrote: >This is what people usually want when they say 'normalization'. AFAIK, you >can't easily tinker with the vorbis frames to add a scale factor in the same >way you can with MP3, but I could be wrong. I could be wrong too, but if I haven't overseen anything, it should be possible to

--- 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