similar to: Resample UltraWideBand to NarrowBand

The resampler will convert raw audio from one sample rate to another. You are starting with Speex-encoded audio frames. So, what you need to do is: 1. Run the media frames through the Speex decoder (look at Speexdec or testenc_uwb). This will give you raw audio with a 32 KHz sample rate. 2. Use the resampler to convert the audio from 32 KHz to 8 KHz sample rate. 3. Run the 8KHz audio

Hi list, I am researcher in VoIP Applications and my challenge now is convert one RTP data frame that is in 32KHz to other RTP data frame in 32KHz. Do someone help me about it? Very thanks, Thiago. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090316/fbbaf566/attachment.htm

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

Hi all, I'm new to speex and there are quite a number of things I am stuggling with. Any help much appreciated. -bob- Environment: Ubuntu 10.04 Speex 1.2 rc-1 GNU C compiler IDE Code::Blocks My application is actually 'D' but I've written a 'C' wrapper following the examples on speex.org. I have raw data which is in float format with a nominal block size of 512 samples

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Look, how about you start from the testresample.c file? Oh, and you seem to be reading 320 samples at a time and processing 640, so that can't be good (and even beyond that your code is wrong for other reasons). Jean-Marc On 11/6/19 1:56 AM, zhouyuchen at iauto.com wrote: > Hello, > I printed the log, in/out len is not truncated, which means that the > input and output are not

Be aware that inlen is an in/out parameter. It tells you how many samples the resampler read. That number can be smaller than what you passed -- in which case it means you need to buffer them and give them to the resampler again in the next call. Jean-Marc On 11/5/19 8:27 AM, zhouyuchen at iauto.com wrote: > Hello, > I have encountered some problems. I want to use speex to convert the

good afternoon everyone, I have a problem that I can not solve I have installed a samba 4.9.0 in centos 7.5 using XFS. In the DPTO share I have the departmental folders, which I gave the rights to the groups. The problem: when a user creates a file within some sub-folders the group's rights do not arrive in the file is read-only. When the user accesses a website and downloads

Hi, In Google Chrome version (win) dev-m 25.0.13655 ran a test call to play () randomly, and several times in 10 different files opus, each with 4K in size; Ex: <audio src="midia/phone-0.mp3" type="audio/opus" id="som0" preload="auto"></audio> ... <button id="btn0"

Hi, I found this list of Speex bitrates in the mail archive. http://www.xiph.org/archives/speex-dev/200306/0004.html Can somebody confirm that this list is correct? I am wondering about the following: - On the Speex website it says: "Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." while the bitrates listed here are e.g. 84400 for

Hi I am using speex for cancelling echo.but i am not succeeded. I need echo cancellation in urgent.Please help m. Thanks in advance Devi

Hello Jean-Marc, thank you for your answer! > I'll look into this. There's basically no overflow prevention for now. > I'll think about how to add that without affecting performance too much > (on CPUs that don't have hardware saturation). I'm open to suggestions :-) I'm not sure if I can really help, but I did a few more tests. Reducing the volume of the input

> I believe there is new resampling functionality in the speex svn head, > although I haven't tested it yet. You might also want to check out > 'Secret Rabbit Code' for your resampling. Yes, I've just been working on a resampler recently. Its changing a lot, but it's now usable. I'd actually be quite happy to have some feedback on it. Jean-Marc > Hope this

Actually opus-tools-0.2 has the same issue. Sincerely Forrest Zhang -------------- next part -------------- Add "#define FIXED_POINT 1" into the top of resample.c, outputs of resample are all zeros. Actually the data transform is required before/after calling speex_resampler_process_native(). diff -Naupr libopusenc-0.2.1-vanilla/src/resample.c libopusenc-0.2.1/src/resample.c ---

hello is there a windows plug in and a linux plug in for net meeting to use speaks codes for voice chatting? there is a linux version of net meeting. thanks hank --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word

Dear Speex developers I am going to port Speex on LPC2368 I tested Speex encoding and the mesurments shows ~40ms cpu time for one frame Do you know who ported speex to NXP or other 32bit platform? Best Regards Zohar fox -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20081105/8261e962/attachment.htm

if the new draft in the manual is used. I don't find how to tell the decoder which mode(NB/WB/UWB) is used in the encoder. The RTP header don't contain the mode field and I don't find the mode information in the coded frame either. Does this mean we have to use NB decoder in all cases? Lianghu On 11/22/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > >

Thank you for the quick response Thorvald, but I think that's not the problem here :[ I know how to capture the buffer and how to play it in the output buffer of the DS. The problem is (probably) with same kind of short/floats/bytes error in conversion/copying that the coder doesn't get. I can have my buffer locked during the compression, it not the problem at the moment. What I really

Good morning List On Friday I had to leave so I could no longer continue with our lab. Weekend and holy all have to rest as much as possible. = D So I'm back and I will put the smb.conf for analysis. # Global parameters [global] workgroup = CMC realm = CMC.CORP netbios name = SAMBA server role = active directory domain controller dns forwarder =

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