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...re performing as I expect: SP AutoCorrelation: (nx/2) * nr + (nr/2) * 5 + 10 - (nr * nr)/4 + nr SP FIR Filter: 4*floor((nh-1)/2)+14)*(ceil(nr/4)) + 8 SP Inner product: nx/2 + 25 But unless the whole algorithm gets down near 10MIPS, I'm going to have to go to the 64xx fixed point. Jerry J. Trantow Applied Signal Processing, Inc. jtrantow@ieee.org -----Original Message----- From: Jim Crichton [mailto:jim.crichton@comcast.net] Sent: Thursday, January 19, 2006 10:33 AM To: Jerry Trantow Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] TI 6xxx platform performance Jerry, I think that just re...

...cific optimizations that could go into the next release. What's the procedure for submitting code? I've been working with this code for about a week now. I'm still trying to understand it all, but I'm particularly impressed by the float vs fixed flexibility of the code. Jerry J. Trantow Applied Signal Processing, Inc. jtrantow@ieee.org -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Thursday, January 19, 2006 1:00 AM To: Jerry Trantow Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] TI 6xxx platform performance > To get a feel f...

...ugh I do not need as many channels). But if you are experienced enough in this area to be doing 6416 optimizations, I am probably not telling you anything that you don't already know. That is not a task for the faint of heart. Jim Crichton ----- Original Message ----- From: "Jerry Trantow" <jtrantow@ieee.org> To: "'Jean-Marc Valin'" <jean-marc.valin@usherbrooke.ca> Cc: <speex-dev@xiph.org> Sent: Thursday, January 19, 2006 10:40 AM Subject: RE: [Speex-dev] TI 6xxx platform performance I started my project using the CodeComposerStudio speex_C...

...the case of a constant -16384 value because it can't really happen in practice (especially after filtering). BTW, on platforms that have a 40-bit accumulator, it's possible to even remove the shift from the loop and apply it only at the end. Le vendredi 03 f?vrier 2006 ? 11:27 -0600, Jerry Trantow a ?crit : > I am overriding the inner product routine in ltp.c. To test my replacement, > I threw some test vectors at it. I understand the loss of resolution caused > by the shift. I also see a FIXED_POINT danger with the summation of four > mults overflowing the 32 bit before the s...

> That's possible. In any case, u-law conversion can be done with far less > than 1 MHz... About Speex, you would likely need to enable ARM > optimizations and set the complexity to 1 (default it 2). done with arm optimizations and i was still getting high load ... guess it's from gstreamer somewhere. I'll check that next week. thanks, - Christophe

...I can share the st->window, lagWindow, bw_lpc1, bw_lpc2, variables which would save 562 bytes per encoder. Are there more variables that can be shared or eliminated if I stick to narrow band? The biggest variables are the inBuf, excBuf, swBuf, innov and I think these have to stay. Jerry J. Trantow Applied Signal Processing, Inc. jtrantow@ieee.org

...speex-dev-owner@xiph.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Speex-dev digest..." > > > Today's Topics: > > 1. Speex in C# Please help (Mohammed Ibrahim) > 2. AEC and resample question (Jerry Trantow) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 8 Feb 2007 06:21:04 -0800 (PST) > From: Mohammed Ibrahim <snouto1984@yahoo.com> > Subject: [Speex-dev] Speex in C# Please help > To: speex-dev@xiph.org > Mes...

I'm trying to make a design decision between a TI 6416 or DM642 (fixed point) and 6713 (floating point) platform. The application is a 32 channel speech encoder. (CBR only, 8khz, 8kbps) To get a feel for the computational load, I am running 1 second (50 frames) of voice through the encoder. My profile of the 6416 indicates I'm at 27.4M cycles/channel. I need to get below 720Mhz/32

I understand that the capture/playback signals need to be sync'd for an AEC to adapt. I'm a little bit confused on the requirements of synchronous sampling between the near end (mic/speaker) and the far end (phone line). I have an embedded DSP system with mic and speaker getting 1msec packets containing 8 samples. We can watch the DSP and ISDN clock frames drift and every few minutes we

...ry; on the inbound flow (from ARM), the DSP gets encoded Speex data via the shared memory, decodes it and then plays it to the speaker. I know there's a Speex port for the TMS320VC5510 but will that port work for the C64xx series? Thanks. Regards Andy ----- Original Message ---- From: Jerry Trantow <jtrantow@ieee.org> To: ndno72-speex@yahoo.com; speex-dev@xiph.org Sent: Tuesday, December 5, 2006 1:55:13 PM Subject: RE: [Speex-dev] Speex support for TI DaVinci processor Do you need to encode and/or decode? I did some work optimizing the DM642 speex 1.1.11.1 encoder earlier this year....

> >I fixed it in svn. Could you check that? > > Now all platforms match again. Note that the measured SNR for this test > sample is lower than with the broken code (10.87 vs 11.10), but of course > this is no way to judge the real quality. SNR, especially on a single sample, can be very misleading. Yet, could you just check that the DSP results match what you get on a PC?

...ed Ibrahim) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 8 Feb 2007 16:27:28 -0500 > From: "Jim Crichton" <jim.crichton@comcast.net> > Subject: Re: [Speex-dev] AEC and resample question > To: <jtrantow@ieee.org>, <speex-dev@xiph.org> > Message-ID: <00b601c74bc7$ef43c6c0$7100a8c0@Jimlaptop2> > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > reply-type=original > > Jerry, > > Looking at the paper that you referenced (which I hav...

...an-Marc. Is there a Speex trunk for the C64xx I can use? Or do I have to figure what options to turn on (in configure) when building the library? Thanks. Regards, Andy Ngo ----- Original Message ---- From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> To: ndno72-speex@yahoo.com Cc: jtrantow@ieee.org; speex-dev@xiph.org Sent: Tuesday, December 5, 2006 2:52:02 PM Subject: Re: [Speex-dev] Speex support for TI DaVinci processor Hi Andy, Speex itself (the library) should work fine on the C64xx because from the libspeex point of view, just just another 32-bit architecture with no FPU. Th...

...ode scales the x[],y[] vectors to avoid this problem I could use parallel 16x16 multiply/adds. You can see this problem with the following test case. for (i=0;i<40;i++) { x[i]=-16384; y[i]=-32768; } sum0=inner_prod(x, y, 40); fprintf(stderr,"inner_prod0(%8d).\n",sum0); Jerry J. Trantow Applied Signal Processing, Inc. jtrantow@ieee.org

...ft_ti); } // norm_shift(). PS. Here are the C64x MIPS vs Complexity results for the original code. I have been able to reduce the complexity 1 encoder to 15.7 MIPS. Encoder Complexity Original 32 Original 16 1 31.2 29.6 2 41.7 39.8 3 51.4 49.0 4 61.6 5 6 7 93.1 8 9 120.8 Jerry J. Trantow Applied Signal Processing, Inc. jtrantow@ieee.org

Hi, all. I have two questions about wideband encoding. >From "testenc_wb.c"... 1) tmp=8; speex_encoder_ctl(st, SPEEX_SET_QUALITY, &tmp); tmp=3; speex_encoder_ctl(st, SPEEX_SET_HIGH_MODE, &tmp); tmp=6; speex_encoder_ctl(st, SPEEX_SET_LOW_MODE, &tmp); How to set high_mode and low_mode, if quality is set to '9'? When I set quality '9'

...int kissfft. For example, the forward fft is normalised, while the backward one isn't. Also, in some cases, I do some renormalisation before the fft to increase accuracy. Please send me the patch when you have something working. I'm sure other people would be interested. Jean-Marc Jerry Trantow a ?crit : > I am in the process of dropping the TI DSP lib fft routines into the AEC. > > Currently, FFTWrap.c uses the USE_SMALLFT and USE_KISS_FFT to select between > two packages of FFT routines. ] > > #ifdef USE_SMALLFT > ... > #elif defined(USE_KISS_FFT) > ......

I am in the process of dropping the TI DSP lib fft routines into the AEC. Currently, FFTWrap.c uses the USE_SMALLFT and USE_KISS_FFT to select between two packages of FFT routines. ] #ifdef USE_SMALLFT ... #elif defined(USE_KISS_FFT) ... #else #error No other FFT implemented #endif My guess is that every DSP and many other processors will have their own specific FFT and iFFT. What method

Hi, My company just recently bought a TI DaVinci development board and I'm trying to port Speex to run on it. The TI DaVinci chip has an ARM926 processor and a TMS320C64+ DSP in it. I was able to get Speex somewhat working on a TMS320VC5510 about a year ago but I was wondering is there anything special I need to do to get Speex working on the TMS320C64+ DSP in the DaVinci chip. I

Hello, I am from a small Czech software company (circletech.net) working on software for mobile phones. We are currently starting a development of an Voice IP software, and consider using speex for voice compression and decompression. The software will be Symbian OS C++ program. Is there anyone who has actually used speex Symbian OS port for realtime voice compression and decompression on mobile