similar to: Discontinuous encoding and VBR tradeoffs

I don't know why you're getting sound carrying over to the next time you encode - that doesn't sound normal to me. Have you tried saving and examining the raw audio you're feeding to the encoder? Have you tried encoding and decoding that using speexenc/speexdec? I use Speex in VBR mode for a VoIP app. I'm always recording and running the audio through the preprocessor

On Tue, Jan 17, 2006 at 02:19:17PM -0500, me@sbooth.org wrote: > Hi all, > > I have recently added Speex support to my app (http://sbooth.org/Max/) and > am in the process of creating a GUI for the user to control the codec > parametere. I am new to Speex and as such I am not really sure which > parameters are more important than others, and deserve prominent > placement,

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 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

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, 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 Conrad, Thanks a lot for the info. This is exactly what I needed. We want to use speex in our IP telephony client and I was indeed trying to put together a GUI configuration screen. Best regards, Chris > On Tue, May 20, 2003 at 06:04:39PM +0200, Chris Flerackers wrote: >> Hi, >> >> Is there are document where the interaction between all the configuration >> options

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; 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?

Dear all I'm currently using a simple encoding implementation based on the encoding example delivered with theora. Now i tried to change the quality of the output file. Its my understanding, that either a bitrate can be set for Constant Bitrate (CBR) encoding, or a quality (0-63) for Variable Bitrate (VBR). As you might know, the basic Encoder Example uses the so-called 2-Pass method for

Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > > See the SPEEX_SET_VBR and SPEEX_SET_VBR_QUALITY options. But be aware > that unless you're aggregating many VoIP calls on the same link, there's > little use for VBR in VoIP. I don't agree with this. I've used Speex's VBR mode exclusively in my P2P VoIP software for the past 4 years and never

> 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

> So far, I've noticed that --dtx alone doesn't seem to do anything, but --vad does > and the combination --dtx with --vad does more. The lone exception is for wide > and ultra-wide coding at quality 0, --vad, alone, makes the file slightly larger. > At all other sized the ordering is always: plain==--dtx, then --vad, then --vad > with --dtx. All you say look normal to me.

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

Hello, I've noticed that when I try to reencode VBR mp3s to broadcast @128k , they are not being reencoded at all. They stream as normal VBRs I am unable to reproduce this at lower bitrates (56-112k i've specifically tested and all work fine). It does reencode all non-VBR mp3s appropriately to 128k... Any ideas why VBR reencoding would suddenly stop working at 128k? I've attempted

Hi, I'm updating speex support in sweep to RC2 and I'd like some advice on encoder options, and would appreciate some testing of the user interface. a screenshot of the updated speex encoding dialog: http://www.vergenet.net/~conrad/tmp/sw_speexRC2_1.png you can grab a tarball of sweep for testing here: http://www.vergenet.net/~conrad/tmp/sweep-0.8.0-speexRC2.tar.gz general

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?

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'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

On 15/12/09 10:37, Blaise Potard wrote: > 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. Just curious, did you identify where exactly the regression occurred? > We are compressing some 22khz wave files in wb mode with maximum > quality / complexity in VBR, and the result was