similar to: encoding 44.1Khz

hi,All, I want to know whether Opus has AEC features like Speex? Thanks 2012/10/17 <opus-request at xiph.org> > Send opus mailing list submissions to > opus at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/opus > or, via email, send a message with subject or body 'help' to

Hi, I am trying to compile the opus-tools with VS 2010 on windows, but I get many errrors and there is some files that there arent in the source like getoppt.c . Has anybody already compile the opus tools on windows successfully? Any Idea will be really helpfull. arctor -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, On the Xiph Wiki page at: https://wiki.xiph.org/MIME_Types_and_File_Extensions I added Opus to several of the sections, but wasn't sure whether to add "Theora + Opus" to: .ogv - video/ogg .axv - video/annodex As far as I can tell, according to the Opus FAQ at: https://wiki.xiph.org/OpusFAQ#Will_Opus_replace_Vorbis_in_video_files.3F Theora + Opus is

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

Hi Josh, On Sat, 2004-10-09 at 22:41, Josh Coalson wrote: > --- Marek Peteraj <marpet@naex.sk> wrote: > > Hi all, > > > > i'd like to ask what the best options are for safe 24bit 96kHz > > archiving. Currently i'm only using -8 but there are also some other > > options like block size etc. Can anyone suggest? > > I haven't heard of

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for

On 2015-01-08 12:17 PM, Martin Leese wrote: > I added Opus to several of the sections, but > wasn't sure whether to add "Theora + Opus" to: > .ogv - video/ogg > .axv - video/annodex .ogv is more correct there. .axv indicates a cmml text annotation track. > As far as I can tell, according to the Opus > FAQ at: >

Hi, I would like to use OPUS codec to compress and decompress an audio stream for full-duplex intercom application; I would like to use a microcontroller equipped with Cortex-M4 core without NEON running at 72MHz. Does the OPUS codec works fine on that architecture? Do you have some benchmark data? Thank you in advance. Best regards. Daniele -------------- next part -------------- An HTML

Hi all, i'd like to ask what the best options are for safe 24bit 96kHz archiving. Currently i'm only using -8 but there are also some other options like block size etc. Can anyone suggest? Also i'd like to ask whether a no-lax 192kHz mode is planned in flac, seems like 192kHz isn't directly supported although flac can compress such rate in lax mode. My last question - is there

Hi Tristan, well... if speed is really his problem (and looking at those 72 MHz it probably is), wouldn't Speek surpass Opus by far? I agree that Opus is way better, but it's sadly also using more resources... But yeah, Richard you should give Opus a try if possible ;) It's generally easier to use then Speex and also more feature rich. Yours sincerely, Ren? Sch?mann *From:*

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

On Sep 22, 2010, at 9:54 AM, David Given wrote: > My language is producing a lot of very short functions, typically two or > three instructions long. These should be ideal candidates for inlining, > but it isn't happening. > > My compiler is producing one big bitcode file, and all the functions are > marked as 'internal'. I'm then doing the optimisation and

My language is producing a lot of very short functions, typically two or three instructions long. These should be ideal candidates for inlining, but it isn't happening. My compiler is producing one big bitcode file, and all the functions are marked as 'internal'. I'm then doing the optimisation and translation manually using llc -O3 into a .s file, and then linking this against

Hello, Sorry for the offtopic subject, but I tried asking this on some Apple mailing lists and could not find an answer there, and as it's related to imap (provided by dovecot), someone might know something about this. The problem is the following: I have a local IMAP server provided by dovecot, so that I can store my mail in maildir format, and read it both using mutt and Apple

Hi!, I wondering if It's possible to change 48khz sample rate limit?, I'm Planing to encode with OGG codec a audio signal but I need that OGG Encoder works with 192khz of sample rate. It's Possible? Any Suggestions?

On Sat, Jun 7, 2014 at 10:58 AM, Andrew Lentvorski <bsder at allcaps.org> wrote: > > > On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > > Actually... no! 24-bit can indeed be useful as extra margin and Opus > > can actually represent even more dynamic range than 24-bit PCM. That's > > not the case for 192 kHz. There's no "margin" that 192 kHz buys

? Do you have any references for me to investigate, I am trying to understand how noise is reduced by introducing higher sampling rates. (I tried to search, but maybe it is so obvious that nobody even explains it) This is not very obvious. It requires you to understand basic signal processing theory. I will give some pointers below. Any physical signal (e.g. audio coming out of speaker, current

Hi there - I am currently in the middle of a project where I need to configure an Icecast Server and I am using Ices as the encoder. I want to configure Ices to take in a stream with a sample rate of 128 kHz and output it to the Icecast server. My question is, what are the minimum and maximum sample rates that Ices will continue to work with? Thank you, Heidi Young Software

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also