similar to: AW: PlusV algorithm -> CBR

Fredag, 31 januar 2003, skrev du: <p>>Next thing is, that I would like to clarify the difference between a codec's >latency (which I would die to know in exact figures for the vorbis codec one >day - as well as block sizes, but that's another story) and the (bandwidth) >"smoothing window" latency - call it buffering or whatever. One thing is the codec

Important (imho)! I found this site in the internet: http://www.plusv.org/ PlusV is an audio enhancement algorithm similar to SRB of Mp3Pro, but 1) better than SBR 2) Fully open-source It seems to me wise to include PlusV into new versions of Ogg Vorbis. P.S. what about some long-awaited features in Ogg Vorbis? 1) ability to turn off the frequency filter in the encoder (especially for high

Just in case my quotation causeed misunderstnaing... Forget about what I quoted before. Probably, by CBR, Monty meant ABR since CBR is stupidly patented and this is the reason why Vorbis has no CBR option. If you know this, it cannot be a problem but I see there is a question asking what is ABR in the mailing list. >On Thu 28 Jun 2001, 23:27:41 PDT, Monty (xiphmont@xiph.org)Wrote: >then

Hi, Maybe this is documented somewhere, if so, please send me a link to the documentation. My question is: how to set up different (VBR, CBR, ABR) modes when calling the Ogg Vorbis API? Currently I do: CBR: ret = vorbis_encode_setup_managed( &vorbisInfo, getInChannel(), getOutSampleRate(),

Hi, I try to use the video tag to post videos on my blog I'm using ffmeg2theora to transcode videos in theora Resolution of 320x240 pixels and constant bitrate (CBR) of ~400 kbps seems to be ok for publishing on my blog. I've read that CBR is a better choice than VBR for web publishing regarding the way TCP/IP works. ("In TCP it takes a while for a connection to increase its

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

Greetings, I was wondering some things: For a internet radio-station (e.g. using mp3 and hopefully using ogg soon), appart from the basic 'streaming' functionallity, there are two things that are also important - mixing music - 'downgrading' a signal to a lower datarate. Concerning 'mixing' music, it would be interesting to be able to mix two ogg-files without the need to

Greetings Fellow Vorbis Gurus! How can I encode, using CDex, at a CBR (constant bit rate) rather than the VBR? Also, can one take a 128kbs MP3 file and re-encode it to ogg? If so, please explain the procedure. By the way, I'm using a Windows based system. Thank you! /jp/ dojpaul@coffey.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Hi, Is there any difference between decoding of CBR or VBR streams in realtime in terms of CPU usage/cost? Are sample rates other than 44100Hz supported by Vorbis officially ? If it is supported, is it efficient to use them or is Vorbis specifically (or at the moment) optimized for 44100Hz ? Thanks in advance. --- Mete BALCI Senior Game Programmer Momentum DMT Istanbul, TURKEY

I found out about vp3 from the new winamp and coincidentally i also happen to be trying to find a suitable solution to broadband streaming. I have used flaskmpeg to create a 17mb .avi file from an 82mb Mpeg 1 file and it looks great it took about twenty minutes to encode. It runs for eight minutes and the audio is also fantastic using the lame encoder. However when I try to do the same in

Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr => true)? 2) If there relation between setting the vbr => true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value?

The following commit from about a year ago removed nouveau_dp_dpms() which did steps required to suspend and resume a monitor connected via DisplayPort. commit 0a0afd282fd715dd63d64b243299a64da14f8e8d Author: Ben Skeggs <bskeggs at redhat.com> Date: Mon Feb 18 23:17:53 2013 -0500 drm/nv50-/disp: move DP link training to core and train from supervisor My computer with

Hi, I haven't seen this mentioned here before. http://www.plusv.org/ "With traditional MP3, a typical Near CD Quality audio file has been encoded with a data rate of 128 kbits/s. While this is ok for people with big hard disks and fast Internet connections, this data speed has clearly been a bottleneck for people using modems or storing their music into 32 or 64 MB portable player

Craig Dickson (crdic@yahoo.com) wrote : > Jack Moffitt wrote: > > > I agree that this might be the case. But what happens with the > > violation is by the author? :) > > > > Ie, in the case of PlusV releasing a patented codec under the GPL or in > > the case of LAME or FreeType's authors doing similar things. What > > state is that? > >

On 5 May 2014 01:43, Ben Skeggs <skeggsb at gmail.com> wrote: > On Mon, May 5, 2014 at 4:48 AM, Sergei Antonov <saproj at gmail.com> wrote: >> The following commit from about a year ago removed nouveau_dp_dpms() which >> did steps required to suspend and resume a monitor connected via DisplayPort. >> >> commit 0a0afd282fd715dd63d64b243299a64da14f8e8d

Question: is it permissible to put more than one frame packet into a page? At 30 fps, Ogg page overhead is on the order of 8 - 10 kbits/sec. At low datarates, this is going to be unacceptable. If I shove a few frames into a page, how does this affect the granulepos stuff? ___ Dan Miller (++,) Founder, On2 Technologies <p>--- >8 ---- List archives: http://www.xiph.org/archives/

Hi, Regarding to my previous question, I think I have found out from oggenc source code that the correct way to initialize encoding is: vorbis_encode_init_vbr (or vorbis_encode_setup_managed for cbr etc.) vorbis_encode_ctl (to turn off or use avg bitrate management) vorbis_encode_setup_init By this way, there is no problem with encoded stream. However, there is no documentation of the three

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,

Hi, I am trying to encode PCM to OGG based on example code in OggVorbis SDK Win 1.0. There is no problem with "vorbis_encode_init_vbr(&vi, 1, 44100, 0.4f);" setting. However, using "vorbis_encode_init(&vi, 1, 44100, 96000, 96000, 96000);" setting gives such a warning by ogginfo: "Warning: EOS not set on stream 1" I have modified ogginfo to display exact

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 | Milestone: Component: Opus |