similar to: decoder give me values over 32000!

I think there is a problem. If we are coding PCM 16 mono, then, when we decode the samples we should obtain PCM 16 mono again, but wave data over 16bits is extracted from the decoder. If the input level is reduced we could loss wave information. Thanks for your help <p>Jean-Marc Valin wrote: >> we are using speex 1.0.3 under windows. We code samples with

> I have been receiving the W32.Netsky virus from the speex email server. > Just giving you a heads up. Our mail server is running Linux, so unless Monty spent two hours to get it to run under Wine, I think you should look elsewhere for the origin ;) Jean-Marc --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

Hello I had an idea about judging the quality of ogg vorbis (or any other lossy codec) I took a wave-file and encoded it to ogg. Then decoded ogg to wav and inverted it's phase. When mixing the original wav with the phase-inverted decoded ogg-file, any identic parts of compressed and uncompressed audio should be eliminated. Of course there's always a "rest" of sound because

Hi, Does anyone have an idea about the possibility to apply psychoacoustic models as the ones in mp3 or AAC to a CELP coder? Thanks! /Pontus <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

On Tue, Dec 17, 2002 at 11:55:21PM -0800, Greg Herlein wrote: > If such a thing happens, discussion of the RTP profile draft > would be most welcome - please get responses back to the > list! Now, if this were finalised before the conference then we could do a demo and use it for broadcasting the lectures streams around the world... What is currently the best way of doing this? I'm

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the

Hi Masakazu, I have reproduced the Speex high frequency noise issue on a Blackfin 537 by sending 'silence' into the Speex fixed point encoder. By recording the Speex decoder output and plotting the spectrum using Audacity the following two frequencies. 1596Hz at -31.3dB 3200Hz at -48dB These two frequencies are present in both Narrow Band and Wide Band modes. I've tried a number

?Hi to everybody. I have data in a buffer. when i encode it and decode it i does not obtain the same data. i think encode(decode(data)) = decode(encode(data)) = data. I also try it with trivial example in opus boundle downloded on the official website. My probleme is that how can i write two function myencode and mydecode such as myencode(mydecode(data)) = mydecode(myencode(data)) = data? Thank

I was wondering if the developers were using anything to "objectively" test the quality of the speex vocoder. For instance PSQM or one of the many derivatives. Mean Opinion Scoring seems an expensive route. Is there some open source software to use for this? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

The latest revision of the draft RTP Profile is attached for review. This will be submitted to the IETF Audio-Video Transport Working Group for consideration immediately, so if you have any more comments, let us know. In addition, we will be applying for an official MIME type. Note that the AVP code and the MIME type in this latest revision have been changed from "SPX" to

Hi, I have been using speex version 1.0.5 on a text-to-speech program. Recently I upgraded to version 1.2beta1 and noticed that the waveform the I got after encoding and decoding on the new versions (beta1,beta2) is much more different than the original than on version 1.0.5. I also ran a PESQ comparison test on 700 voice samples and got better results in the older version (I used quality 9, and

I have successfully encoded a wav file in android using speex(ndk). But when i try to decode it back, the file size keep on increasing to a very large size. The recorded audio file consists of sample rate - 8000 , 16 BIT, MONO. Why the decoder gives such a large size wav file? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

It seems that speex (speex 1.0.4 on OSX 10.3.9) is writing over the input buffer (with what looks like a sine wave). Included below is a short program that demonstrates the problem. Is this a bug, or am I doing something wrong? I don't see this behavior mentioned in the docs. I don't really need the input buffer, but I'm guessing this problem is related to a high pitched sound

Hello Greg If money isn't a problem Intel has an optimized compiler for eVC and XScale processors http://www.intel.com/software/products/compilers/techtopics/PCA_Optimization_WP.pdf If you have any luck getting the eVC compiler closer to realtime I'd really like to know. I'm still far from realtime when using Speex 1.1.3 on a HP iPAQ (Intel pxa255). Best regards Bjoern D.

Hi all. I'm trying to decode speex using version 1.1.10's libspeex with fixed_point enabled. copied the sample in the manual (1.1.11) with minor revisions. While running the program, encountered this warning: "Packet is larger than allocated buffer : 38" when calling speex_bits_read_from (&bits, cbits, nBytes) then my program terminated unexpectedly with errors (Unhandled

I was recently talking to a friend of mine who produces what's widely referred to as "techno", music that's heavy on electronically generated loops and effects, samples, and has plenty of bass. He doesn't like ogg for a few reasons - he sent me a rather long email, and I've stripped out the nonsense from it (he has some technically irrelevant reasons for not liking

The only way you can track these things is by IP address, looking through the message headers. Don't try to draw any conclusions from email addresses used in the message, be they From, To, or in the body, as these are always forged and are harvested in many ways. Tom tony t. (digicastipv7@sbcglobal.net) wrote: > > I figured so. I should have been more clear. Someone through the

I was also wondering if there is a standard set of input sequences people are using to test Speex. I haven't stumbled upon it/them yet. > -----Original Message----- > From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]On > Behalf Of Jean-Marc Valin > Sent: Tuesday, February 25, 2003 7:24 PM > To: speex > Subject: Re: [speex-dev] Quality > > > > I