similar to: Opus in WebM

Yes, it's basically an intrinsic aspect of the mathematics. If you seek into the middle of an Opus stream, that's about how long it takes before the audio converges to be correct. On Jan 17, 2014 2:13 PM, "Brendan Bolles" <brendan at fnordware.com> wrote: > On Jan 7, 2014, at 11:11 AM, Brendan Bolles wrote: > > > The closest thing to a standards document for

It does not; Vorbis needs only lap consecutive frames, and the timing was constructed so that the 'timestamp' of a vorbis frame actually points to the middle of the frame data. When you combine a starting frame with the subsequent frame, the data you get back is from the starting timestamp forward. Because Opus needs to preroll a larger amount, that trick wouldn't work with Opus.

On Jan 7, 2014, at 11:11 AM, Brendan Bolles wrote: > The closest thing to a standards document for putting Opus into WebM is here: > > http://wiki.xiph.org/MatroskaOpus I'm curious about the part there where it says SeekPreRoll should be set to 80000000. I believe those are in nanoseconds, so that's 0.08 seconds, 3840 samples at 48kHz. Here's a page explaining that it

Hi everyone, I'm working on a WebM plug-in for Premiere. You probably already know that WebM is a Matroska container that uses Vorbis for audio compression. I'm pretty new to the world of audio compression and need some help. You can see my code here: http://github.com/fnordware/AdobeWebM First of all, I wonder if it's possible to seek to an exact audio sample in WebM. Whereas

I am a new developer on Vorbis. Now I want to create a audio-only webm file from the micrphone. Vorbis encoder is necessary for the creation. Here is the procedure of the steps: 1. Initialize Vorbis environment. 2. Generate the header and export to the output stream. 3. Get the PCM samples of certain size. 4. Divide the original PCM data into Vorbis block. 5. Encode the block into Ogg

Hey everyone, I've added Ogg Opus support to my Adobe Premiere plug-in here: http://github.com/fnordware/AdobeOgg Now I'll add Opus support to me WebM plug-in too. I've got this Opus stuff mostly figured out, but I have a few questions. Here's one: What do the numbers in mapping mean? I see that opus.h refers to the Vorbis channel mapping order, so does mapping in Opus take

Brendan Bolles wrote: > DiscardPadding seems to assume that you might top off an opus > frame with empty samples when the frame would spill onto the next > matroska timestamp. So rather than have continuous opus that > could be decoded one after another, you'd have to jump over these > little gaps? The intent was not to allow gaps in the file like this (that would confuse the

On Jan 17, 2014, at 2:19 PM, Benjamin Schwartz wrote: > Yes, it's basically an intrinsic aspect of the mathematics. If you seek into the middle of an Opus stream, that's about how long it takes before the audio converges to be correct. Do you happen to know if Vorbis has a similar need to do a SeekPreRoll? Brendan

On Jan 17, 2014, at 3:19 PM, Brendan Bolles wrote: > I don't think Vorbis has the fixed-frame issue of Opus where audio frames might not align with Matroska timestamps. Whoops, I meant for the total duration of the file you should be able to have your vorbis length match the file duration length exactly. As my next paragraph said, in general I think the Vorbis packets will never align

Hi Jean-Mark, thank you for your answer! Yes, you understood my question correctly. I was just about to compose a reply to my original question, where I described how I solved my problem. As you've already suggested, I've switched to Ogg/Opus, which is better supported, but does not work with the Media Source Extensions. I'll have a look whether disabling prediction will help with

Hey everyone, I've written a FLAC plug-ing for Adobe Premiere. It's actually part of the Ogg Vorbis plug-in I originally set out to write. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip

Hi all, I finally understand how lipopusenc is capable of producing chainable Opus files (in contrast to my program), and I managed to successfully implement the method [1]. Essentially, the last frame of a file is marked as a "keyframe" by disabling prediction for this frame in libopus. This encoded keyframe frame is then copied in verbatim to the next file, with the pre-skip set to

Hi Jean-Marc (and everyone else who replied), > Considering you're switching to Ogg, I think you should give libopusenc> a try. It does a really good job at getting rid of *all* discontinuities> -- to the point where you can chop a song into files less than one> millisecond each and it still sounds good. It's also pretty simple to> use. You just feed it audio and tell it

Hi ! 2011/1/31 Romain Beauxis <toots at rastageeks.org>: > Well the interesting question is: "Is webm streamable?". Well it seems to ma that Google made Webm especially for streaming and I've even read that it should be easier to stream webm than ogg for various reasons. > Alternatively, there seems to be a ogg mapping for vp8 floating around: >

Sorry I replied in private ... 2011/1/31 Quentin Drouet <kent1 at arscenic.info> > > > 2011/1/31 Romain Beauxis <toots at rastageeks.org> > > Le dimanche 30 janvier 2011 03:54:10, michel memeteau a ?crit : >> > Is there some evolution on Webm support, does it need a lot of changes >> > in Libshout for example ? or does the code structure of icecast

Le dimanche 30 janvier 2011 03:54:10, michel memeteau a ?crit : > Is there some evolution on Webm support, does it need a lot of changes > in Libshout for example ? or does the code structure of icecast makes > it easy to add new containers/codecs ? Well the interesting question is: "Is webm streamable?". I don't know much about it nut webm is wrapped into a matroska-based

Attached is a patch for adding webm streaming support to icecast svn. Also available is my git tree: https://github.com/krad-radio/icecast-oneman/tree/format_ebml_internal My git tree is behind the svn version a little bit, I will update it soon, the webm code is identical. Please note the following: 1) This version has no external dependencies, the EBML parsing is built into the format_ebml

Hi, After a few experiments I found how it works. - the easiest way is to declare a path statement in the config. - you can now copy webm files (webm streams are now supported) in the previous path - now, each files are considered as mount. BEWARE : you can't use chrome or safari or opera to open the url directly, the file will be downloaded instead of read. Works with Firefox by the way.

Hi, On 5/1/19 9:58 AM, Sytze Visser wrote: > I am streaming live with webm with ffmpeg to icecast 2.4.2. After the > stream ends, I am unable to determine the duration of the file using > ffprobe or mediainfo. Not sure but it seems that this has to do with > headers? It's a fundamental limitation of this type of streaming that the duration can not be determined beforehand and

On Sat, May 22, 2010 at 12:06:15PM +0300, Alexandru Ardelean wrote: > Hey everyone. > > If you haven't seen this already, last week was Google I/O the third > edition. > The event page is : http://code.google.com/events/io/2010/ > The youtube channel with the keynotes is : > http://www.youtube.com/googledevelopers > One of the most interesting things at that conference