similar to: lpcSize

Okay, Not exactly the answer I was looking for. This sounds like a big change. I don't mind re-writing the LSP quantizer but re-training code books and breaking compatibility is not what I want to do. I am working on an optimization for an ARM cortex-A8. It is desirable to process things in 4 element blocks. Is there a simpler approach you could recommend? Vasant Shridhar

On 11-04-11 06:48 PM, Shridhar, Vasant wrote: > Okay, > > Not exactly the answer I was looking for. This sounds like a big > change. I don't mind re-writing the LSP quantizer but re-training > code books and breaking compatibility is not what I want to do. I am > working on an optimization for an ARM cortex-A8. It is desirable to > process things in 4 element blocks.

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

Hi, can someone help me understand and experiment with Speex in a better way than my knowledge. I want to change the LPC filter order of the Speex algorithm. I think I need to change the LPC order( or lpcSize) in modes.c file. but to make it work I need to modify the codebook also, can someone tell me how to go about changing the codebook entries to support the changed order of prediction.

On mer, 2011-08-10 at 09:41 -0400, Rony Nandy wrote: > Hi All, > I have downloaded libogg-1.3.0 along with speex.But,during > build speex is exiting with ogg.h missing.Any suggestions will be highly > appreciated. IIRC, speexenc encodes your data into a speex stream which is encapsulated into an OGG container, so you need to libogg to compile it. Though, it has been ages

Hi, I am using speex 1.2beta2 on a 24bit DSP that has a severe program and data space limitations. I am only interested in the speex decoder for 11kbps narrowband implementation. I am using the following parameters and structures 160, /*frameSize*/ 40, /*subframeSize*/ 10, /*lpcSize*/ 17, /*pitchStart*/ 144, /*pitchEnd*/ /* 11 kbps medium bit-rate

I have been trying to understand how to get the VAD algorithm working. I sent an input stream of all zeros into the preprocessor but still got a return value of 1 indicating that speech was detected. Is this feature not available with the latest release? I thought at the very least it would detect this as silence and return 0 but that does not seem to be the case. Does anyone have any

On Fri, Jul 8, 2011 at 6:39 AM, Ashhar Farhan <farhan at phonestack.com> wrote: > what you need to do is this: take the wireshark raw dump, read each > udp packet and write it back to another file. While writing back to > the new file skip the sizeof udp header + rtp header. I can't recall > how many bytes you need to skip, however, I suppose it would be in the > range of

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Hi, So I got a crash on the following code: k1=SUBMODE(lpc_enh_k1); k2=SUBMODE(lpc_enh_k2); which in the newer codebase is: bw_lpc(SUBMODE(lpc_enh_k1), st->interp_qlpc, awk1, st->lpcSize); bw_lpc(SUBMODE(lpc_enh_k2), st->interp_qlpc, awk2, st->lpcSize); I am not sure if the newer code will have the same issue but the following check is

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a

Speexdec plays ogg files. You can simply modify it to play your raw file or even rtp capture. Jozsef -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of speex-dev-request at xiph.org Sent: Saturday, July 09, 2011 12:00 PM To: speex-dev at xiph.org Subject: Speex-dev Digest, Vol 86, Issue 3 Send Speex-dev mailing list submissions to

Hi All, I am trying to use VAD opption of speex encode on STM32F4 chip. But when I use *speex_preprocess_run *function, It allways return 1. Anyone can show me the problem? And please, give me an example about using VAD. Thanks in advance, Regards, Nhan. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

Hi I am currently trying to port speex v1.1.6 to a microcontroller with very limited memory (<64Kbyte RAM). what I found when initialising the encoder, a chunk of 32Kb was attempted to be alloced, which failed: src/nb_celp.c: void *nb_encoder_init(const SpeexMode *m) { /* snip */ st = (EncState*)speex_alloc(sizeof(EncState)+8000*sizeof(spx_sig_t)); /* snip */ } same goes for the

Firstly, running for more channels will not break my hack. All that's needed is to call RECOPLAY_MARK with different identifiers (say nb, wb or uwb) before doing the appropriate initialization. Secondly, my attempts to do the Gaussian in fixed point went like this : Define a new constant lag_factor_gauss that is manually set equal to exp(sqr(2*M_PI*lag_factor)/-2) by whoever changes the