similar to: Is SPEEX_GET_QUALITY implemented ?

I am embedding speex in boost python when i execute int TSpeexEncoder::getQuality() { int Quality; speex_encoder_ctl(enc_state,SPEEX_GET_QUALITY,&Quality); return Quality; } i get the error warning: Unknown nb_ctl request: 5 looking in the header SPEEX_GET_QUALITY is equal to 5 I am able to read all the other settings. the codec appears to be set with VBR=0 I am

Hello everyone, I'm finally beginning to use Speex for a project. As expected, the implementation so far has been very easy and I am thrilled with the capabilities and flexibility of Speex, especially the new features in 1.1. My efforts so far have resulted in a couple of questions: 1) I was interested in fetching the default quality setting for narrowband:

To close debian bug #271052, http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=271052 I went ahead and did the following: $ diff -u ../original/speex-1.1.6/libspeex/nb_celp.c libspeex/nb_celp.c --- ../original/speex-1.1.6/libspeex/nb_celp.c 2004-07-15 01:16:52.000000000 -0400 +++ libspeex/nb_celp.c 2005-02-27 08:24:49.000000000 -0500 @@ -1746,6 +1746,21 @@ st->submodeSelect =

> 1) I was interested in fetching the default quality setting > fornarrowband: > speex_encoder_ctl(_encodeState, SPEEX_GET_QUALITY, &value); > > Unfortunately, the above fails with libspeex 1.1.4. The console > shows: > > warning: Unknown nb_ctl request: 5 Actually, I never implemented SPEEX_GET_QUALITY. It's not that usefule anyway, but I might do it in the

Hi Maitland, I think your solution is probably the best implementation you could do for SPEEX_GET_QUALITY. However, unless you have something against that, I think removing it would be better, as in many cases (e.g. VBR), the call just doesn't make sense). This would of course break the API (not the ABI), but given the fact that it was never implemented, any app that uses it is already broken

I am using Speex 1.0.4. I was very confused about why speex_encoder_ctl with SPEEX_GET_QUALITY *always* failed, until I read <http://lists.xiph.org/pipermail/speex-dev/2003-September/001896.html> from a whole year ago! Can SPEEX_GET_QUALITY be removed from speex.h and from the documentation (or actually implemented, whichever is easier :) Best! Reed

On Sun, Feb 27, 2005 at 06:09:48PM -0500, Jean-Marc Valin wrote: > Hi Maitland, > > I think your solution is probably the best implementation you could do > for SPEEX_GET_QUALITY. However, unless you have something against that, > I think removing it would be better, as in many cases (e.g. VBR), the > call just doesn't make sense). This would of course break the API (not

On Mon, 17 Jan 2005, Jean-Marc Valin wrote: > > 0) Is it possible to manipulate voice data representing more than > > the frame size of encoder (this seems possible in Speex Java code > > used until now). > > Speex itself manipulates audio one frame at the time, but what you do > after that if up to you. > What about the section 4.5 of the

On Mar 29, 2004, at 5:05 PM, Jean-Marc Valin wrote: > Can you try manually swapping the bytes on the raw output and see if > you > get something useful (I think by default wav is always little endian > while raw is native endian)? Well, my apologies. Apparently I got confused when trying all the different formats and somehow skipped little endian. The decoded raw file is in

Dear Sirs, I was trying to compile openssh for Solaris 2.8 in 64bits mode. For that, I have defined the variable CFLAGS with the following value "-fast -I/isdc/include -L/isdc/lib -xtarget=ultra -xarch=v9" before running the configure command. The configure command used was the following: ./configure --prefix=/isdc --with-ssl-dir=/isdc --with-prngd-socket=/var/run/entropy

Hi. I have just finished Googling the list archive for information on doing CTI with Asterisk. I never found a definitive "here's how it's done" answer, but I did see a number of postings indicating that it was at least a possibility. (For those of you who already know about CTI, please pardon the pedantic introductions to each section.) Here are my questions: [First Party

A few weeks ago I reinstall my laptop with Debian etch (upgraded to lenny), kernel 2.6.27.8, wine 1.1.10, fglrx v8.11 (ati x2300) and Steam api version v007 There is no problem to launch Steam.exe (`wine /path/to/Steam.exe &`). However, after Steam.exe startup, I can not launch Day of Defeat. No specific error displayed on the screen, the `Preparing to launch Day of Defeat' dialog popped

Hi I apologise if the following message is inappropriate for this list. However, I feel this is the only place where I can get some help with my problem. BOOL CALLBACK RecordProc(HRECORD hRecord, void *buffer, DWORD length, DWORD user) { try { long sent = 0; long inputOffset = 0; long outputOffset = 0; long nbBytes = 0; char *outputBuffer = NULL;

Hi, I'm using the latest version of libspeex (checked out today) and have some problems with checking the target bitrate in the encoder. Calling: peex_encoder_ctl(globals->enc_state,SPEEX_GET_BITRATE, &bitrate); Seems to set bitrate to 24600 no matter what the qualitysetting is.. The functions used previous to this call is: peex_bits_init(&globals->bits);

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

Hi All, i need your help in determining the problem in the following sample code (taken from speex manual) for fixed point encoding. i tested encoding a 160 sample frame and then decoding it back But The problem is that the decoded frame is totally different from the original frame(see output below).What is the possible reason for this?.thanks for any help #include<stdio.h>

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

I've targeted Speex to run under eCos with Thumb interworking using arm-elf-gcc 3.3.3 I'd written a small test case to verifiy the operation: #include <speex/speex.h> void speex_test(void) { SpeexBits bits; unsigned int frame_size; void *enc_state; float *speex_frame; cyg_uint64 t1, t2; int i; speex_frame = (float *)0x00030000; /* Address of Speex data

When I just copy the microphone input buffer to the output buffer the sound plays OK. But if I encode and decode the buffer through Speex I get a high pitched constant tone in the background. I actually do hear my voice speaking when I talk, but it's faint and much quieter than the tone. Here's what my data looks like: Input is the first 5 floats of each input buffer. Output is

Hello, I'm trying to use the speexlib but I just can't do it. I've made just like the example (on the site) says. Some times I get SIGFAULT. Also, I'm reading the input_frames from /dev/dsp (directly). What is missing here? speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frame_size);