similar to: Encoding 2-Pass VBR?

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

Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > > I've been using Speex in my voice-over-IP program on Win32, in > > wideband (16kHz) mode. I just starting using VBR recently and > > have run into something that might be a problem within Speex: > > Are you turning on DTX in addition to VBR? Also, what version are you > using. As of 1.0, DTX is

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 have just written a small c programm which reads in speex files and outputs them as asci text numbers. This is necessary as for writing a hw testbench for decoding tests, since thy only read ascii files :-(.I can't read the bitrate properly from the header file. It's allways -1. Please see example below and attached c programm. I wan't to check for a fixed bitrate (8000).

I want to display info about Ogg files to Windows users. The docs tell me how to distinguish CBR (nom=min=max bitrate), but how do I distinguish between ABR and VBR streams? When I created my Ogg files (with CDex) I set the 'quality'. Is there some way to read this from the Ogg file? Should I avoid calling the ov_bitrate function because it has to read through the whole stream?

I know speex is not supposed to do a great job compressing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down

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

> 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

Hi, I've been using Speex in my voice-over-IP program on Win32, in wideband (16kHz) mode. I just starting using VBR recently and have run into something that might be a problem within Speex: If someone hasn't spoken for a little while, and the bitrate drops to very low, sometimes the high half of the spectrum becomes frozen with a looping sound. The bottom half of the spectrum is

Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3),

I did some further investigating into this issue and here's what I found: It appears somewhere between ices0.1.0 and ices 0.2.0 is where the "bug" (for lack of a better term) was introduced that i'm noticing. When I compile ices0.2.0 w/lame 3.89beta, the VBR is *NOT* reencoded, but instead streamed out at full VBR bitrate(s). However, when using ices0.1.0 w/lame 3.89beta, it

Dear all, I have been going through below mentioned update. http://web.mit.edu/xiphmont/Public/theora/demo.html I found this line- Timothy Terriberry estimates a simple self-training range-coding backend could be reasonably expected to decrease bit usage overhead by 15-20%. Is it really true? I found one implementation.... http://svn.xiph.org/trunk/ghost/libentcode/ Did anyone verified this

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,

Josh, Yes, you can change VBR quality or ABR average bitrate at any time. You don't need to do anything special. I use this feature (changing VBR quality midstream) from time to time and it's quite convenient. Tom Joshua Gargus <schwa at fastmail.us> wrote: > > Hi there (from speex-dev newbie), > > Is it possible to change the target VBR on the fly? The

Hello, I purchased the "CD Ripper 2" plugin for Winamp 3 and need help choosing an ogg encoding format. It seems I have the choice between VBR and ABR formatted oggs. I am familiar with VBR (variable bit rate) but not sure what ABR stands for? Is it possible/desirable to specify a bit rate when encoding oggs, or does the nature of the ogg format make use of varying bit rates?

Dear FLAC team, I have some fundamental doubts regarding FLAC encoder: 1) In FLAC encoder there is no option of specifying the bitrate. There is something called compression-level. I believe that based on the nature of input WAV file and compression-level chosen bitrate is coming into picture. As such there is no user control on the bitrate. Is that correct? 2) In FLAC encoder I am also not

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

> 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

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

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