similar to: libfishsound/liboggz: How to properly set e_o_s marker when encoding?

On 21 May 2013 00:17, Bob Ingraham <bobi at ingrahams.us> wrote: > I apologize if this is the wrong list for this, but after searching the > available lists at xiph.org, this is the closest I could find. > > I ran into a bug in the example fishsound-encode.c program where it gets > OGGZ_ERR_BAD_SERIALNO errors from oggz: > > fish_sound_flush (fsound); > oggz_run

I apologize if this is the wrong list for this, but after searching the available lists at xiph.org, this is the closest I could find. I ran into a bug in the example fishsound-encode.c program where it gets OGGZ_ERR_BAD_SERIALNO errors from oggz: fish_sound_flush (fsound); oggz_run (oggz); oggz_close (oggz); fish_sound_delete (fsound); After debugging, it appears that

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)

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

FishSound 0.8.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (Vorbis and Speex). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.8.0.tar.gz New in this release ------------------- This release includes compatibility with the

FishSound 0.8.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (Vorbis and Speex). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.8.0.tar.gz New in this release ------------------- This release includes compatibility with the

Jan, Here's a hacky patch to add a few commandline options for setting comment header fields in ffmpeg2theora. It's a bit big because I virtualized the global info struct in theorautils.c. In retrospect that probably wasn't necessary, but I think it's cleaner anyway. I didn't test it because I couldn't compile ffmpeg2theora, but modulo bugs it should support

Hi, ffmpeg2theora currently assumes that the first frame in the input file also corresponds to the first audio sample in the input file, which might not be true for many file formats. I have especially bad problems with MPEG-TS streams that I recorded via DVB-T (sync off by 0.5 seconds etc.). Encoding chapter-ranges from DVD might yield similar problems. The attached patch fixes that, using

Hi, the attached patch (against current SVN) adds three new options to ffmpeg2theora: --keyint (set keyframe interval) --smoothness (set the theora_info::sharpness encoding parameter) and --noautosync (disable the new frame dropping/duplicating code). It also fixes a bug with the processing of --cropright that sometimes segfaulted. The sync adjustment code is now tuned to be more reliable. BTW

On Fri, Apr 3, 2009 at 1:03 AM, <j at v2v.cc> wrote: > How is write access to the official repository handled right now? > in the past it was possible for people with svn access to commit minor > fixes or apply patches. would be sad if now write access to the official > repository is limited to one person. might be a good idea to set git up > in a way that makes it possible

For last few days I was trying to learn ffmpeg and libtheora API. In the process, I have modified ffmpeg2theora code to include support for start time and end time. ffmpeg2theora -s 60 -e 130 file.avi will produce file.ogg which will be from 60th to 130th second of input file (something like -ss and -endpos in mencoder). This is a useful feature for someone who wants to cut a part of video

Hi, A week or two ago Viktor Gal migrated the liboggplay source repo to git.xiph.org, and this week I moved the liboggz and libfishsound repositories. You can see the git repositories available via gitweb at http://git.xiph.org/ I wrote some brief news about the move, and instructions for getting the code: http://blog.kfish.org/2009/04/liboggplay-liboggz-libfishsound.html Thanks to Ralph Giles

Hi, A week or two ago Viktor Gal migrated the liboggplay source repo to git.xiph.org, and this week I moved the liboggz and libfishsound repositories. You can see the git repositories available via gitweb at http://git.xiph.org/ I wrote some brief news about the move, and instructions for getting the code: http://blog.kfish.org/2009/04/liboggplay-liboggz-libfishsound.html Thanks to Ralph Giles

Hi, A week or two ago Viktor Gal migrated the liboggplay source repo to git.xiph.org, and this week I moved the liboggz and libfishsound repositories. You can see the git repositories available via gitweb at http://git.xiph.org/ I wrote some brief news about the move, and instructions for getting the code: http://blog.kfish.org/2009/04/liboggplay-liboggz-libfishsound.html Thanks to Ralph Giles

Hi, A week or two ago Viktor Gal migrated the liboggplay source repo to git.xiph.org, and this week I moved the liboggz and libfishsound repositories. You can see the git repositories available via gitweb at http://git.xiph.org/ I wrote some brief news about the move, and instructions for getting the code: http://blog.kfish.org/2009/04/liboggplay-liboggz-libfishsound.html Thanks to Ralph Giles

Josh Coalson wrote: > thanks Erik, can you confirm that this is happening only because > the total_samples_estimate is 0? (so the e_o_s setting in the > existing code is never triggered). In the Ogg code, I believe that there are two things that must be satisfied to correctly write an end of stream: - packet.e_o_s must be true - Need to call ogg_stream_flush() instead of

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

--- 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

On Tue, Nov 07, 2006 at 06:31:04AM +1100, Erik de Castro Lopo wrote: > I haven't tested it, but I don't think just setting packet.e_o_s > is sufficient to close the stream. The forced ogg_stream_flush() > is required. libogg checks the e_o_s flag (and has always done so, according to svn) so you shouldn't have to call ogg_stream_flush() except when you want to force a page

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: