similar to: encoding speex, (insanity looming)

In short try calling speex_encode_int and speex_decode_int. The calls to speex_encode and speex_decode expects the data to be a floating point values. The data is passed by pointer and the compiler does not do the conversion. This is why you are getting the segfault on decode as the short is 16-bits and a standard float is 32-bits. Tom -----Original Message----- From:

Hi Tom, Thanks for your reply, I'm having a few difficulty's following you advise. You mention that I should be calling 'speex_encoder_int' & 'speex_decoder_int', I'm having problems finding these functions in the speex header. I am currenlty calling speex_encoder_init & speex_decoder_init in my test code, I have repeated the relevent bits of the example

Hi, in my application i quite often (not allways) get those two warning: "Invalid mode encountered: corrupted stream?" "Invalid wideband mode encountered: corrupted stream?" The stream is not corrupted but i wish to know what those warning mean to correct my code. My encode and decode routines are those (libspeex 1.1.4): SpeexBits enc_bits, dec_bits; void *enc_state,

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

i don't think so. this is my current code: for(i=0; i<samples;i++){ /*only get max 320 bytes at a time so bits can handle*/ Client->raw_audio[i] = *input++; Client->temp_audio[i] = Client->raw_audio[i]; Client->session_File[ndx] = Client->temp_audio[i]; ndx+=1; } speex_bits_reset(&ebits);

Hi, I read in the documentation something about how to realize packing more than one encoded frame into a packet. Before I read this, I always encoded every frame seperate and packet them as following together: [encoded length][encoded frame][encoded length][encoded frame]... But if I understood the docu correct, it should be easier: I call speex_encode_int() for every frame until the whole

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any

I'm currently working on a speex + theora webchat. I have both working alone, but to make them work together and to know where the package goes, I'm trying to use ogg packages. With theora there has been no problem, but with speex I can't make it work. The problem is that for a second it works great, but then it begings to gain lag, and in a little time it has 5-10 lag seconds (into

Hi, i have some code that gives me a "warning: Buffer too small to pack bits" mesage. looking at the libspeex source/bits.c i see the warning in a a function named speex_bits_pack. i'm not using this function. can someone tell me where this may be coming from since i've had a similiar problem before and had that fixed. i can't seem to pinpoint this one. thanks Greg

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>

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

Thanks for your reply. I changed my code to: if (speex_preprocess_run(preprocess_state, shortPointer) == 1) { speex_encode_int(enc_state, shortPointer, &enc_bits); } In the mobile version of the software, compiled against the mobile build of Speech, I get 1 and 0 based on whether the speech is detected. In the version of the software compiled against the Win32 version of Speex,

Hello everyone I've a following problem with the speex codec. I'm developing application that is supposed to read sounds from the microphone with DirectSoundCapture. Unfortunately when I want to compress and decompress the sound buffer from DirectSound I only have jitter. Maybe you have some code samples how to compress standard DS buffers: LPDIRECTSOUND8 lpDS = NULL;

Changing from using floats to shorts did fix the high pitched tone problem. I'm having other problems but I'll look into it more first. SteveK wrote: > > On May 21, 2006, at 6:33 PM, Kevin Jenkins wrote: > >> 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 >>

int frameSize; speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frameSize); Is frameSize in bytes? If not, what unit is it in? I need to know so I know how big to make each element in my circular buffer. Also, do I need to call speex_encoder_ctl( enc_state, SPEEX_SET_SAMPLING_RATE, &sampleRate ); Depending on sample rate I record my audio in? If not, is there any benefit to

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, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer

Hi list, I'm trying to figure out how to do the most compatible implementation that will work with as many versions of Speex as possible. I am streaming multi frame Speex blocks over a TCP connection which works fine as long as the version of Speex is the same on both sides. When using a newer Speex (1.1.?) to encode and an older version to decode (1.0.5), it does not work. The encoder

On Saturday 10 April 2010 21.51.55 Jean-Marc Valin wrote: > All version after 1.0 are compatible with each other and with 1.0. Hi Jean-Marc and thanks for the quick reply. I have now looked at this further and got the idea to hard code the number of frames in the decoder temporarily as a test and voil?, it works! The problem seem to be some incompatibility in the termination handling. After

Hi John, Thanks for your reply. Yes, my output device is in 8-bit mode, same as my input mode. I'm actually working in char (8 bit) arrays. Is it necessary to work in signed short integers only? I changed all the short arrays in the example to char. Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To: carineliang@yahoo.com.sg; speex-dev@xiph.org Sent: Thursday,