similar to: Compilation of Speex for use with MIPS processors

Well, it's an ARM7TDMI core, so basically one register operation per clock, with memory accesses taking longer. Having the memory on-chip should make memory access much less of an impact. I was afraid that you would answer the way you did: I thought about my question after I sent it, and the "LP" in CELP is what makes it a sequential process; it can't do linear prediction on a

Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml

Eecho cancellation is used to eliminate the echo which mixed with near end speech, while it was occured by far end speech. It always work like this: Find the similar speech in near end speech and eliminate it. In case 1: Because the there is no echo in near end speech, and the EC module still works on it. So a inverse model is been calculated, and a inverse signal is produced. In this way ,

svn trunk 31497 For the life of me, I can't get this :) I want to be able to catch the situation where the calling party hangs up *before* the call is connected to the called party. My dialplan is thus: macro DialExternal(exten) { Dial(Zap/G3/${exten},120,g,M(connected)); goto DialResult|r${HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing

Hi, There is a lot of speex quality estimations. One of this comparative estimation is even available on the official site <http://speex.org/comparison/>. I'd like to present yet another one. And I thought that the best place for this presentation would be Speex-dev mailing list. I want to get feedbacks and criticisms please. If Speex authors consider to make some parts of this work

Lis, I suggest you try tweaking Speex's VAD probabilities as Steve suggested. But consider a simple threshold-based approach as a backup option. Personally, I struggled with Speex's VAD algorithms (both encoder and preprocessor) for a long time, tweaked the probabilities, wrote special case code to work around the mistakes, and was still never satisfied with the results. In times

Tom Harper wrote: > Steve, > > The main problem I am having with the system is clipping off the start > of someone's speech when they first start talking- the ends of the > sentences > seem to be handled properly. I am wondering whether this is the fault > of the > audio playback system or whether this is a speex issue- I don't seem to get that; I suppose

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On = Behalf Of Tom Grandgent Sent: Friday, March 03, 2006 12:57 AM To: Steve Kann; Lis Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Lis, I suggest you try tweaking Speex's

I used Microsoft text to speech with speex. It sounded reasonably well. I didn't experiment with various quality settings (8KHz, default quality). I was capturing audio from a sound card, not the direct wave data from the TTS engine. .a -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Reed Hedges Sent: Friday, May 26, 2006 7:23 AM

Hi Alex, How about you start by actually reading the manual ( http://www.speex.org/docs/ )? If you have any remaining questions after that, you can post them on the list. Jean-Marc ALEX NG a ?crit : > HI, > I am new in speex. How can I use speex to implement the online voice > chat? > Can we stream the voice? > Thanks > Alex > ----- Original Message ----- From:

Hi All, I would like to communicate speech over a network compressed using speex. However, I do not want to communicate a whole Ogg-formatted file. I am interested in only the speech frames. I invoked: speexenc input-file-name - > raw-speech This, I am thinking gives the speech frames only. To play this back at the receiving end, do I need to format it into Ogg formatted file before I can

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

I am interested in using Speex in an embedded system built around an ARM microcontroller. I have seen other posts indicating that Speex can run in real-time on some iPAQ PDA's, but these are using a StrongARM 166MHz processor. I'm looking more at the chips from Atmel (SAM7), Philips (LPC2xxx), and TI (TMS 470), which are ARM7TDMI with on-chip SRAM and flash, running at speeds of 33 to

Hi, Definitely! Since I have a glance and seems to me that speex is used for static compression but not for streaming. Can you suggest me which part should I read carefully? There are so many parts inside the doc. Many thanks Alex ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca> To: "ALEX NG" <alex@gorex.com.hk> Cc:

Hi, Jean-Marc, and thanks for the quick reply. Let me just say I'm a huge fan of speex, and the work you've done. I actually barely understand what I'm reading so far in the source code and documentation, just enough to understand just how cool the algorithms are. LPC10 and MELP allow me to speed up speech with a simple hack on the decoder frame size. Playing fewer samples per

> /* FIXME: This VAD is a kludge */ > .. and it shows (or hears?) unfortunately. I've run a few tests with it > with my users, and they complain that it misdetects too often... In both > directions. Non-speech is detected as speech more often than before, and > more important it also doesn't detect speech as good as before. > I'd really like to see this

HI, I am new in speex. How can I use speex to implement the online voice chat? Can we stream the voice? Thanks Alex ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca> To: "sharanabasava matmari" <sharanabasavam@yahoo.com> Cc: <speex-dev@xiph.org> Sent: Monday, May 21, 2007 6:13 AM Subject: Re: [Speex-dev] Speex speech