similar to: Enabling Speex VBR for VoIP

Hi, I'm trying to cover all Speex configuration options to set quality/mode/bitrate in my application and I was wondering if the following all makes sense: - abr_bitrate - use average bitrate and set it using bitrate parameter - SPEEX_SET_ABR - cbr_bitrate - use constant bitrate and set it using bitrate parameter - SPEEX_SET_BITRATE - cbr_mode - use constant bitrate and

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

I have two questions: 1. ?I would like to enable VBR with the Speex codec for VoIP communications. Is this setting made within VoIP clients themselves or is it a setting that must be changed on the specific version of Speex (for example, via command line)? 2. If not, is there any way to create RTP packets via the Speex command line? (The goal is to view the lengths of the packets created by

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

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 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 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

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(),

I'm writing a voice communication application, and I've got a few issues that I'd like to get ironed out, but I don't know enough about the speex implementation. First of all, this application is mainly used for conferencing - many people are in a room and only 1-2 are ever talking at a time. So, always encoding and transmitting everyone's audio stream would be rather

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,

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?

On 2010-04-11 10:09, Randy Yates wrote: > If I specify VBR mode via > > speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_VBR,&isTrue); Try: speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_ABR, &desiredRate);

The audio frame speex generates sounds pretty terrible most of the time, and I don't use it for jitter correction instead I just use it for dropped packets - so I usually drop the late packet. It sounds acceptable as long as I drop less than 5% of traffic (dropping 2 in a row makes a bad robot noise, so I reset the stream in that case). The good news is that on an unsaturated DSL line jitter

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

hi, im a student from a local university in our place. i'm not sure if this is the place to post this question so please pardon me if im wrong. my questions is that when i transfer speex encoded samples through a rtp session wire, upon receiving the samples i immediately decode and play them, but i get a crackling sound when i decode and play each sample received. 1) i used 160*5 as my wave

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any

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

________________________________ From: "speex-dev-request at xiph.org" <speex-dev-request at xiph.org> To: speex-dev at xiph.org Sent: Wednesday, August 7, 2013 11:43 PM Subject: Speex-dev Digest, Vol 110, Issue 1 Send Speex-dev mailing list submissions to ??? speex-dev at xiph.org To subscribe or unsubscribe via the World Wide Web, visit ???

Thanks for the advice. With complexity=1, bit rates 5950,8000 and 15000 (CBR only), I'm still getting 30-50 seconds encoding time for a 10-second file. To anyone who has made this work in ARMv4, or knows how to, can I get some advice on the settings? FIXED_POINT is already defined in all relevant files. My source code is patterned after sampleenc and I haven't tried using Ogg.. my source