similar to: ogg encoding

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:

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

So, I abandoned the hope of using the ogg python bindings to do pure ogg container encoding. I started looking at the libogg in the hopes of retooling the bindings to follow a better object model and it actually looks like the problem is down in libogg, not the bindings. Am I crazy or does libogg rely on libvorbis to return ogg_packets, and that there are no functions that will build an ogg_packet

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--- 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

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.

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. I'm trying to implement Ogg/Vorbis support for the MPEG4IP project. The goal is to support Ogg/Vorbis audio for MPEG-4 streaming. So far I have managed to make the encoder save Ogg packets as an Audio object in an .mp4 file. As a side effect, it can also save Ogg pages in an .ogg file playable by xmms, but that's no big deal. So what I'm doing is this. First the init part. a)

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, 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

On 25-Feb-08, at 9:34 AM, xiphmont@xiph.org wrote: > I'd say both; libogg should ignore the incomplete packet or flag a > structural error, and the spec should be clarified. I guess libogg should return -1 if you try to ogg_stream_packetout() after the eos packet. The eos flag pretty clearly applies to the *page* not the packet, so I think how that gets translated into the

Hi, I have recently implemented encoding/decoding to/from OggVorbis in my app, with the code being based upon the Libvorbisenc sample code supplied with the SDK. It all works very well indeed (encoding from a PCM file, then playback), except that no matter the length of the source PCM file, the last (approx) 4 seconds of audio is always missing from the Ogg file. I am trying to find some

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

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

I'm trying to encode a picture from my webcam using the theora codec. The final "product" must encode a frame, send it over the network and decode it on "the other side". For now, it must only work locally so we don't care about the transmission. This is what I've understood till now: I have a char * buffer from the camera RG24,I convert it to YCbCr 4:4:4, I

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

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

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, 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

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