similar to: Please 30 second to look a my code

Well, you seem to be using FRAME_SIZE but only defining frame_size. Otherwise, the code looks OK, but it's always hard to tell. I suggest you start from speexenc/speexdec or from the example I wrote in the manual at: http://www.speex.org/manual/node12.html Jean-Marc Le ven 19/12/2003 à 05:22, Fabio a écrit : > Hi > i'm developing a sort of VoIP application > for my

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

Hi, i'm not too familiar with how the ogg vorbis community works, so i hope this is within the topic of the list, but i didnt really want to leave this thing i've did out in the dark either. :) i've just ported libvorbis, libogg, and vorbisfile stuff to the pocket pc a while ago, it was pretty straight forward (just add a few more #ifdefs around the ftoi function and some others) and

Hi All, Are multi-channel (more than 2) formats fully supported in the OggVorbis specification ? I couldn't find any information about multi-channel support on xiph.org. I've used 'oggdropXPd' to encode a 5.1 wavefile and the Xiph OggVorbis libraries (vorbisfile.dll) to decode the file successfully, however the order of the channel interleaving is different to the original wave

The majority of a Speex encoder app does fit in a 6713. The 6713 has 8K of L1 and another 256K of memory 64K of which can be configured as L2 cache. (16,32,48, or 64K). One level of TI's website seems to incorrectly indicate only 64K of L2. I turned off MANUAL_ALLOC and have it allocating internal memory using calloc(). I did change the L2 cache to 2 way (32K) and adjusted the heap size to

I'm shopping for a small/tiny audio recorder, the kind for recording in a class, interviews, etc... not really music, just voice. Per usual, a lot of these write their audio files in some Windows format, e.g., WMA. As a confirmed Linux guy, I'd want to offload the audio files in some format that Linux can read/play natively. I've read a sketchy suggestion that there's a Linux

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

I started my project using the CodeComposerStudio speex_C64_test.pjt in speex 1.1.11.1. To build using floating point, I created a new project with the same files and modified ti\config.h to #undef FIXED_POINT. Is there a better way to configure a floating point processor? I have a few TI specific optimizations that could go into the next release. What's the procedure for submitting code?

This script was developed by Mark Johnson. All I did (Dean Collins) was type up the instructions and make it easy to understand. This Script will allow you to dial and extension number on your Asterisk@home V0.6 or later pabx and have it read you the weather for your city How it does this is; 1/ When you dial your extension it causes the weather.agi script to run 2/ This makes the program

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Erik de Castro Lopo wrote: > David Siebert wrote: > >> Anyone know of some example code that shows storing speex audio into a >> WAV? For a lot of reasons a WAV is a better choice than ogg for my >> application. > > If you have decoded the speex data to PCM, the easiest way to get > it to WAV is to use libsndfile: > >

Hey All, I now have Festival compiled, installed and running using the instructions on the Wiki page. When I try to change the voice that is being used however, I am running into a problem. I get the following in the festival server log: Cannot open file /tmp/est_10877_00000/utt.wav as tokenstream Wave load: can't open file "/tmp/est_10877_00000/utt.wav" Cannot load wavefile:

Hi, I frequently encode multi-hour radio recordings in 44.1 kHz 24 bit stereo and am currently using the -6 setting with verify (via the "FLAC frontend" application). Would there be any parameter(s) that may lead to noticeably better compression without increasing encoding time a lot? (My machine isn't the most powerful in the world, even with two CPUs (see sig).) Currently

This is release 0.9.51 of Wine, a free implementation of Windows on Unix. What's new in this release: - A bunch of WinHelp improvements. - Better Japanese font support. - A ton of rpcrt4 fixes. - Several Alsa capture fixes. - Improved support for screen resolution changes. - Lots of bug fixes. Because of lags created by using mirrors, this message may reach you before the release

Hi, Tank you for your reply. I have tried the sample programs given the manual and done encoding and decoding a wave file. That is working fine. I have already read the manual and API reference. But this problem did not occured with the wave file encoding/decoding. While capturing the data from MIC and encoding/decoding is giving this problem. Thanks and Regards, Vijender Reddy, -----

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>

I've cross-compiled flac for the armv4l processor (rio receiver), and i'm trying to start up a decode thread : #include <FLAC/stream_decoder.h> .... FLAC__StreamDecoder *flac = NULL; flac = FLAC__stream_decoder_new(); if (flac == NULL) { printf("[DECODE] Unable to initalize flac object\n");

[cc'd to vorbis and gstreamer-devel because I thought both lists would like to see this ] > I'm curious if anyone else is at all fed up with the current > state-of-affairs of media support under *IX. As things stand it's rather [...] Hi, Caught your post on vorbis-dev, but I'm not on vorbis, where Monty suggested this go, so email me directly with any replies.

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your