similar to: non-audio packets

hi all, im sending my speex packets from one PC to another across the Internet via TCP. rigth now, i am working on a buffering solution to ensure the continuity of voice playback on the recieving end. to solve this, i need to determine the size (in milliseconds) of the incoming packets. how do i compute this? my speex settings are as follows: - narrow-band mode - quality level 4 - 10 speex

Hi All, We are using speex decoder only for narrow band decoding. Quite often, even though source frame is not silence, decoded audio is silence. I wonder there is a problem in the way we uses speex decoder API. Basically, we initialize speex decoder only once and just reset the decoder before we decoding another file. Do we need to intialize speex decoder and destory decoder for each audio file

Somewhere in the new packet-loss concealment code, there is a bug. I tried encoding a 44100-hz stereo file and it decodes just fine, but if I try it with packet loss, I get a segmentation fault. If you want to see the file, I can put it up somewhere publicly accessible. If someone here can't work on it, I'll be back on my home computer in a few days, so I'll work on it then.

I believe if you add the *decoder* lookahead time to the encoder lookahead time you are already referencing, you will get the numbers that you have calculated. --John On 4/11/2012 1:00 PM, speex-dev-request at xiph.org wrote: > Message: 1 > Date: 10 Apr 2012 22:07:20 +0200 > From: Thilo K?hler<koehlerthilo at gmx.de> > Subject: [Speex-dev] Speex Codec Delay Problem > To:

Whoops, left this message in my outbox. I managed to fix the problem. Apparently I was only copying 160 bytes (Frame Size) back into the audio stream when I should have been copying 320 (chars <-> shorts confused me there). Hence why I could hear myself yet it was distorted. Half the wav was missing =) To answer some of the other questions here, for any insight into what I'm doing:

Fabio, You can't mix encoded audio directly. You would have to decode, mix, and reencode. There will likely be some loss of quality. Tom Fabio <fabpet@gmail.com> wrote: > > Hi to all, > > I developed a VoIP application using Speex. Now i want to record the > conversation to a file. > As i have two indipendent thread transmitting e receiving voice i > whish to

Hi to all, I developed a VoIP application using Speex. Now i want to record the conversation to a file. As i have two indipendent thread transmitting e receiving voice i whish to mix both audio streams in one stream before saving the conversation to a file. I'm using Speex in narrow band mode, 8 bits per sample, mono. i found at this link http://www.vttoth.com/digimix.htm a simple way to

comment inline. On Wed, 16 May 2007, Jean-Marc Valin wrote: >> Page 3: >> >> To be compliant with this specification, implementations MUST support >> 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. >> The sampling rate MUST be 8, 16 or 32 kHz. >> >> There is a type above after (narrowband), there is a " extra

Hi, We'd like to announce the first draft for the Speex RTP profile. It was written essentially by Greg Herlein, with some help from Simon Morlat and I. We'd like to get some feedback on it before it is sent to the IETF. Basically this will allow all SIP based VoIP applications using Speex to inter-operate. For those interested, there's already Simon's LinPhone (www.linphone.org)

> Otherwise Greg, can you send the latest version of the RTP draft so I > can put it on the site (the current one is getting old)? Attached for all to see. Greg -------------- next part -------------- Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin draft-herlein-speex-rtp-profile-06

The latest revision of the draft profile for use of Speex in RTP is attached. We plan on submitting this - or a modified version of this, based on immediate feedback - to the IETF on Monday for consideration at the next meeting. Major differences in this revision are: - removed the discussion in the MIME section. It's a duplicate of the SDP discussion anyway, and may or may not match the

All: Attached please find yet another RTP profile revision (v5). You can also find the document at: http://www.herlein.com/downloads/speex/docs/ Changes: - added vbr, cng, ebw, sr optional parameters to MIME - added vbr, cng, ebw a=fmtp options for SDP use - added required document attributes for submission to IETF and IANA (format and author contact information). Note that we

Hi, I'm moving my "metadata" which accompanies some of the speech packets into the speex bits. However, as usual, I'm a bit confused. The manual (PDF from current SVN), section 5.10 (bottom of page 21) states that: "Finally, applications may define custom in-band messages using mode 13. The size of the message in bytes is encoded with 5 bits, so that the decoder can

Alfred E. Heggestad wrote: > <...> > > If we don't get any comments in 1 week (by 22. May 2007) we will go ahead > and submit it to the IETF. Of course you can comment on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the

hi, I have a small and quick question. When speex divide the UWB speech signal into High and low bands, JMV said to me in HA that : "For ultra-wideband (0-16 kHz, 32 kHz sampling), I first split the band into wideband (0-8 kHz) and "very high band" (8-16 kHz). Then, the wideband itself is split into low (0-4 kHz) and high (4-8 kHz) band. So there's a total of 3 bands

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hi, On http://www.speex.org/docs/manual/speex-manual/node11.html#SECTION001140000000000000000 the bitrate for Speex Wideband mode 9 and 10 is 9 34,400 10 42,400 On http://tools.ietf.org/html/draft-ietf-avt-rtp-speex-03 the bitrate for Speex Wideband mode 9 and 10 is 9 34.2 kbit/s 10 42.2 kbit/s I believe the draft contains the right values: E.g. mode 10 Narrowband mode 7 -> 492

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

Hi Please find an updated version of the Speex I-D attached. The only change is addition of the copyright conditions in Appendix A, as requested by Ivo. Many thanks for your input. I will give you a few more days before submitting to AVT working group /alfred Ivo Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check