similar to: vorbis multistream

This patch adds top level functions to create an ambisonic multistream encoder. The implementation currently just calls the analogous surround sound functions with channel mapping 255 to create an encoder that bundles uncoupled streams. Forthcoming patches will actually set channel bitrate and other configuration. My main concern is that adding additional

Hello All, I have created a Vorbis Comment Editor for the win32 environment. It is completely untested and undoubtedly has bugs. That said, it is fairly stable and functional. It is a simple single dialog based app. I have released it under the GPL and have made code available. It is compiled statically with the latest CVS version of libogg & libvorbis and it uses the vcedit.c routines

Hello All, I have created a Vorbis Comment Editor for the win32 environment. It is completely untested and undoubtedly has bugs. That said, it is fairly stable and functional. It is a simple single dialog based app. I have released it under the GPL and have made code available. It is compiled statically with the latest CVS version of libogg & libvorbis and it uses the vcedit.c routines

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

Heya guys, How long until we get a WORKING Vorbis Comment editor? For Windows? -- Robert J. Lynn, Jr. Brainbench Certified Computer Technician, Linux Administrator, and Master Windows 98 User PGP Key ID: 0xCDE22CFB (RSA) rjlynn@suscom.net, rjlynn@crosswinds.net EFNet: Vegeta99 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

I use the Opus multistream encoder for both mono and stereo encodings and after updating from 1.1.3 to 1.3 I noticed the size of the produced Opus files had doubled for 1-channel encodings whereas switching to the standand Opus encoder gave me roughly the same sizes as before. In these tests, I'm encoding an 8kHz mono stream containing only speech with the following options set: frame length

Does win32 have pathconf() and _PC_NAME_MAX? I ask for ogg123 -- it uses "NAME_MAX" right now (already been mentioned on this list a few times), which does not exist everywhere. I've proposed changing it to use the pathconf() function, but I don't know if win32 has this or not (I don't have, or want, access to win32 compilers :-). If win32 doesn't have pathconf(), we

When encoding using the Opus multistream API, is it possible for opus_multistream_encode() to return 0? For example, what happens if multiple streams are being encoded, each with DTX enabled and all streams emit DTX packets at the same time? What about the case of a single stream with DTX enabled? Thanks Kevin O'Connor -------------- next part -------------- An HTML attachment was

Are multistream packets limited to one frame per stream in a given packet? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20140326/acfa13ab/attachment.htm

Hi Michael, Is there any reason you can't just use the generic multi-stream API, i.e. opus_multistream_encoder_init() and give it the mapping you need? This is how surround was originally done (in 1.0) and only got changed when surround needed a more complex mapping and more data in the encoder. If it turns out you need this kind of thing too, then yes we would probably just want to extend

Jean-Marc, > Is there any reason you can't just use the generic multi-stream API, > i.e. opus_multistream_encoder_init() and give it the mapping you need? I would like the encoder to make decisions based on the fact that the audio is Ambisonics. For example, the bitrate allocation should not be the same for every channel. The original multistream API does not provide a way to pass in

Hi Ralph, could you please set up a area at xiph.org, where you mirror some of the test sequences, just to be shure that everybody talks about the same sequence ? The license issues are not too important, I think if we should be allowed to use 'free for non-commercial use' sequences for our tests. Perhaps it's a good idea to seperate things in 'free', 'free for test

Hi Jean-Marc, I looked at that but importantly these streams need to remain absolutely independent, Further they may have been encoded at some previous time. So my question stands. Thanks, Marc On Jul 26, 2013, at 9:10 PMEDT, Jean-Marc Valin wrote: > Hi Marc, > > I recommend you have a look at the multistream API and how we use it for > surround in the Ogg Opus draft. Sounds

Hi all, The attached patch adds/fixes a few null argument checks in the multistream API. Do these changes make sense? Thanks, Felicia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170324/1afee1e1/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name:

Hello all, Attached is the initial proposed implementation for ch.mapping 253/3, based on the IETF proposal: https://tools.ietf.org/html/draft-ietf-codec-ambisonics-03 A brief overview of the patch, as it is slightly lengthy: After discussion with Jean-Marc, we determined that ch.253/3 will need the demixing matrix as part of the encoder/decoder struct stack and thus will require a new

This patch adds a new option "channel-mapping" to opusenc which sets the channel mapping family used by the multistream encoder. Please let me know whether adding this option is worthwhile and whether the help string is okay. I tried to keep it short but accurate. The error message for an unimplemented channel mapping is "Error cannot create encoder: request not implemented".

Hi, For some time now I have been using pxelinux to add a helpful 'administrator' boot mode to our workstations with the following 'pxelinux.cfg/default' config: ---- DEFAULT local PROMPT 0 TIMEOUT 0 TOTALTIMEOUT 1 ONTIMEOUT menu ALLOWOPTIONS 0 NOESCAPE 0 LABEL local LOCALBOOT 0 LABEL menu CONFIG /pxelinux.cfg/default-menu ---- This is great as by

Hi all, We (WebRTC/Google) would like to extend Opus to natively support 120 ms encoding instead of relying on repacketization as a post processing step. This is to ensure that a valid 120 ms packet is always available. I've attached a couple of patches to add this to opus_encoder(), based on the internal repacketization process carried out by 60 ms CELT. We intend to extend this later for

Thanks for looking over this, Mark! I'm travelling this week but when I'm back I'll address all of your concerns and send,out another patch. :) On Jun 7, 2017 9:56 AM, "Mark Harris" <mark.hsj at gmail.com> wrote: > On Tue, May 30, 2017 at 3:13 PM, Drew Allen <bitllama at google.com> wrote: > > Hello all, > > > > Attached is the initial

Hi guys. I hope this isn't too off-topic for this list... I work for a company that is currently using ogg/vorbis in their audio software product as one of several possible encodings. We want to add some new functionality to the software that will require a little specialist coding from someone who knows the ogg codebase. Specifically, we need a library to provide vorbis file