similar to: bug found in speex_bits_read_whole_bytes

OK, I fixed it but: 1) you shouldn't be using the MAX_BYTES_PER_FRAME constant (just moved it to the .c where it belongs) 2) if you read more than MAX_BYTES_PER_FRAME the buffer will resize automatically, so there's no problem. Just use the size you want. Jean-Marc Le lun 25/08/2003 à 02:27, Thomas S a écrit : > Hello there, > I thought I would contribute to this wonderful

I'm trying to use speex on OMAP's DSP using dsp gateway, and have some questions. The code almost works (within few next days I'll post a some kind tutorial and patches), but it seems to be a little bug in bits.c. The code looks like this void speex_bits_read_from(SpeexBits *bits, char *chars, int len) { ////////////////// bla-bla-bla /////////////////// for (i=0;i<len;i++)

Jean-Marc, The following patch makes the above two read functions const correct. Cheers, Erik diff --git a/include/speex/speex_bits.h b/include/speex/speex_bits.h index a26fb4c..234ec53 100644 --- a/include/speex/speex_bits.h +++ b/include/speex/speex_bits.h @@ -77,7 +77,7 @@ void speex_bits_reset(SpeexBits *bits); void speex_bits_rewind(SpeexBits *bits); /** Initializes the bit-stream

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output

Hi, I have raw PCM data that is 180 seconds long and sampled at 22050 samples per second. The data is 16-bit speech audio, male and female voices. My question is: What is a good frame size to use if I capture the entire audio stream first and then want to compress it for later playback. I noticed that the example encoder code calls: speex_bits_init(&bits); and I notice that this has the

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output

> -----Original Message----- > From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] > Sent: Thursday, December 14, 2006 2:51 AM > To: Miles, Stewart > Cc: speex-dev@xiph.org > Subject: Re: [Speex-dev] Would be nice to conditionally > compile out coding modes and code tables... > > Miles, Stewart a ?crit : > > I'm only using the narrow-band encoder

Hi everybody, I need clarification about speex_bits_write function usage. There is an example usage of it. nbBytes = speex_bits_write(&bits, cbits, MAX_FRAME_BYTES); I examine the content of this function and which value should I set to MAX_FRAME_BYTE. In speexenc project, it is 2000 and in manual it is 200. For example, I set it to 200 but the length of bits->chars is 250

Hi, I am using speex 1.2beta2 on a narrowband 16-bit, 8khz system that has a severe program space problem and will not fit speex in its normal operation. In an attempt to shrink speex I placed a breakpoint in every function and ran a decode and encode and removed the breakpoints that I hit. in the functions that had a breakpoint that I didn't hit I commented out those functions (as well as

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

On Tue, 2005-03-01 at 04:31 -0500, Jean-Marc Valin wrote: > Hi, > > I just checked in some changes that remove some unnecessary buffers > (exc2 and buf2) and that reduce other buffers to the minimum. That > should save 7800 bytes for the encoder. > > Jean-Marc > thanks, I have merged the diff in $ svn merge-r 8996:8997 http://svn.xiph.org/trunk/speex to my codebase.

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Using the following command cvs -d :pserver:anoncvs@xiph.org:/usr/local/cvsroot login and anoncvs as the password, I get the following response cvs [login aborted]: connect to xiph.org(198.136.36.245):2401 failed: Connection refused can anyone clue me in? D Sledge __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software

I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be

Winamp3 is broken according to some posts here. Try Winamp 2.91 or the latest XMMS. Of course, XMMS requires the xmms-mp3 libraries. Try that and see if it helps. KJ <p>On Fri, 2003-09-12 at 15:56, Jeff Ousley wrote: > Thanks, all, for the pointers. I think I have > everything configured properly, but, something is > obviously still wrong. I'm using Kerry's guide with >

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07

Hi. My bad, but I have not been tracking the -stable list and I just updated -stable on a production machine. It seemed to work just wonderful, as normal, for about 10 minutes then it started to deny connections and when you run commands, it would just hang. I have done a reboot and fsck everything. It is up now but things are still strange. Here's the output. I just updated -stable at