similar to: Ogg Vorbis and streaming

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 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

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

> All you say look normal to me. Okay, clarification requires less code to be written than a bug. :) > > Does that makes sense? I would have thought that --dtx alone would have = > doen it, > > but maybe it needs perfect digital silence to work? Or only uses the voi= > ce activation > > to know when there's silence? > > DTX requires the VAD to know when

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, 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

I'm well aware how compression works. But images and document files do not depend on the relative timing of the data to reproduce themselves. They are in essence only two-dimensional in space, whereas the data in a sound file is time-dependent. The question really has more to do with the decoded FLAC stream output, which I presume is a linear PCM file, e.g. WAV. If FLAC is lossless and

There is a big, big difference between cbr and vbr, btw. The machine in question is a Pentium 3 700 oc'ed to 800, 512MB ram, win nt 5.1 (2k and xp being nt) etc In an earlier version the oddcast dsp ate up so much cpu time when set at say 64kbps (for examples sake) 44.1kHz, stereo that the machine became totally unusable and the winamp task had to be (slowly) ended with task manager. Now,

Hello folks I am encoding music to ogg vorbis format for the first time. From what i have researched, ogg vorbis is natively a vbr format. Also encoding using the -q switch is encouraged instead of specifying a bitrate through -b. Now, while encoding using -b i get this: Encoding "file.wav" to "file.ogg" at approximate bitrate 224 kbps (VBR encoding enabled) Notice

> From: Jack Moffitt [mailto:jack@icecast.org] > > So in the very near future, I don't think that there will be any > non-vorbis capable players. And for the most part, I think Vorbis will > be included in most players, though I have little hope that Microsoft > will allow Vorbis to stand side-by-side with WMA. There is a big difference between having a codec available for

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

Brian... You've been both polite and helpful. Thanks. I do understand the dimensional nature of images and sound, though I admittedly glossed over the details while trying to draw attention to time rather than spatial artifacts. What I was looking for was confirmation that a properly designed application would decode FLAC without temporal issues. I believe you've made that perfectly

Mark Casey wrote: > I think when the real-time cbr encoding issues are sorted > Vorbis (this is my experience with oddcast dsp, winamp and > icecast 2 win32, big cpu hog) will start gaining more ground, I can't use the managed bitrate option in OddCast on my K6II-500 CPU! (Overclocked to 525, WinXP Pro,192MB RAM). I would like to. Normal VBR uses about 90% CPU. This seems too

Dear All, I like to simulate a physiologically based pharmacokinetics model using R but am having a problem with the daspk routine. The same problem has been implemented in Berkeley madonna and Winbugs so that I know that it is working. However, with daspk it is not, and the numbers are everywhere! Please see the following and let me know if I am missing something... Thanks a lot in advance,

Hi, I'm quite familiar w/ mp3 cbr/abr/vbr encoding, as well as mpeg4 (cbr/vbr,etc). And I can always calc the bit rate for a given file size with: file size * 8000 / length in seconds = kbits/sec Works great w/ mpeg4 + mp3. BUT FOR THE LIFE OF ME: I cannot get oggenc (1.0x version) to give me the file size I want. I calc. it with the above formula, and nothing comes out right. Then I do

Hi All, On Tue, Sep 18, 2007 at 06:04:59PM -0700, Stephen F. Booth wrote: >> I think that what I actually want is for the operating systems to use >> .flac >> instead. Similar to what Mac OS X does with .aiff. This should probably >> have been posted in the advocacy emailing list instead. > > This is an interesting idea. There already is a FUSE filesystem called

Hi, I'm not anywhere near an expert, but I had successfully used Darkice, TwoLame, and Icast231 to netcast a mp2 stream. If it will help, here is a snip from the related area of my darkice.cfg: [icecast2-1] format = mp2 bitrateMode = cbr bitrate = 384 quality = 1.0 server = 127.0.0.1 port = 32710 password = (duh!) sampleRate =

I'm running a: for band in n w u do for quality in 0 1 2 3 4 5 6 7 8 9 10 do for complexity in 3 do for vad in "" "--vad" do for dtx in "" "--dtx" do echo speexenc -${band} --quality ${quality} --comp ${complexity} ${vad} ${dtx} input.${band}.wav

Hi, Is there are document where the interaction between all the configuration options of speex is explained? Speex seems to have CBR, VBR and ABR. You can also use SPEEX_SET_QUALITY (SPEEX_SET_VBR_QUALITY) and SPEEX_SET_BITRATE which I suppose can't be set at the same time. Is there a list of possible combinations somewhere? I also have another question related to jitter. To minimize jitter,

Quoting "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>: > Speex it's too cpu expensive for general S60 usage, it would require a > lot of ASM optimization. Did a quick search and saw ARM CPUs with speeds above 100 MHz. That should actually be enough for Speex, at least for narrowband. > If you are using CSD, like for a secure telephony solution >