similar to: live streaming

Thanks a lot. Well, the problem is that it's hard to find any informaton about converting *.mjr files. I'll keep trying. Regards, Mateusz 2015-07-09 0:57 GMT+02:00 Mark Harris <mark.hsj at gmail.com>: > Hi Mateusz, > > It looks like the Janus code that writes Ogg Opus will not produce > correct Ogg Opus output when there are packets missing. You can quote > me or

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, In our project we use janus-gateway (http://janus.conf.meetecho.com/) as a webRTC gateway and also as a stream recorder. We are on the tests stage of our project, and after very long development time we have ecountered a bug that is a blocker for whole project. After real tests of recording streams using janus we realized that audio and video are out of sync in recordings, despite of fact,

Hello All, This is my first appearance to the opus mailing list. My name is Michael Mehari and i am a researcher at Ghent University mainly on cognitive radio networking. Recently i am working on a demo scenario on self adjustable wireless audio conferencing and i selected opus for the audio encoding and decoding part. specifically the applications (i.e opusenc and opusdec) from opus-tools

Dear Opus developers. This is the first time I write here, so hello to everybody! Sorry to disturb you but I would like to ask you something I could not answer by googling and by looking at this mailing list archive. I have just started investigating this new and promising codec for real-time audio transport over the internet for industrial applications. I was previously experimenting with RTP

Hello! I need to extract audio data at a certain position in respect to the original audio data. Is the "-cbr" switch meant to ensure that data can be found on a specific position? I want to be able to predict where certain audio can be found in the encoded data. An example: If you put 10 apples into a box and then compress the box, you will not be able to predict where apple C

Which one of the various libraries on xiph.org allow me to create an Opus file? The 3 or so libraries on the Opus download page all seem to be for reading files, converting and/or encode/decode streams. I have device that outputs a stream of Opus 'frames' and I need to save them into an Opus file. -- Tony -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I am interested in being able to encode a single Opus stream using several CPU cores. I get a raw audio input and "opusenc" can transcode it at 1200% speed (Raspberry PI 3B+). It saturates a single CPU core, but the other three are idle. Is out there any project to add multithreading options to "opusenc", or something in that line? Looking around, I have found this:

Hello, I'm currently using Liquidsoap to stream a group of files to Icecast. The stream is sent to Icecast in a few different formats (MP3,HE-AAC v2,OggFlac,OggOpus). The "Current Song" metadata is correctly displayed for the MP3 and the AAC stream, but not the OggFLAC or OggOpus stream. I see a few entries in the Icecast forums, but it does look like FLAC metadata support never

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Martin Leese wrote: > Subject: [opus] MIME Types and File Extensions > To: opus at xiph.org > Message-ID: > <CAAzqGd_uzR646Nsdt=O2HDxLOYE2=K=5n9UOHLr3Y4BGzdVasw at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi All, > > On the Xiph Wiki page at: > https://wiki.xiph.org/MIME_Types_and_File_Extensions ... > Could somebody more

Hi Mark, Well, so do you think that any of this libraries probably won't provide correct audio conversion? Do you know any other possible solutions for described problems, ommiting trying to fix on my own any of this libraries or waiting for fix? Will you mind if I quote you in disscussion I mentioned? Sorry for asking you so many questions at once. Regards, Mateusz 2015-07-08 17:42

This patch to opus-tools adds optional support to WavPack lossless format as input to opusenc. Like support to FLAC, it depends on an external library, libwavpack, and may be disabled on configure. Lucas Clemente Vella (1): Reading input from WavPack files. Makefile.am | 7 +- configure.ac | 37 ++++++++ src/audio-in.c | 71 ++++++++------- src/opusenc.c | 19 +++- src/opusenc.h

On Wed, 2014-02-12 at 14:29 +0000, Ian Malone wrote: > On 12 February 2014 13:02, Alice Wonder <alicewonder at shastaherps.org> wrote: > > Hello, > > > > I am new to using opus but so far I really am loving it. At 16kbps > > spoken content sounds fabulous. > > > > I am using opusenc on 64-bit linux - mostly with flac input. > > > > The one

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hello!, i've just compared the 1.0.3 release with the master branch on a BeagleBone Black (AM335x 1GHz ARM Cortex-A8 with NEON floating-point accelerator) and Arch Linux ARM. At the moment I dont no why, but I see that 1.1 is much slower in encoding. Are there any default changes, that I missed and could explain this? Normaly I suggested a better performance with 1.1 and the ARM

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

Hello, I am new to using opus but so far I really am loving it. At 16kbps spoken content sounds fabulous. I am using opusenc on 64-bit linux - mostly with flac input. The one thing I can't figure out how to do is add album art. I tried passing the vorbis way of adding a metadata_block_picture comment field to opusenc but it complained about the length. I can't seem to find any

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and