similar to: link problem with VC 2008

Greetings. I just upgraded to the latest 1.2beta3 and now I get the following link error: unresolved external symbol speex_wb_mode. Using VS2005. If I put back the old libspeex.lib it compiles just fine, but I would like to upgrade to use the new AEC stuff if I can. My code looks like: enc_state = speex_encoder_init(&speex_wb_mode) ; Worked before the upgrade so did the API change?

Hi, No, it only happens with speex_wb_mode. But like I said, it is because in wideband mode, the function sb_encoder_init is called and this function initializes the sampling rate using speex_*de*coder_ctl instead of speex_*en*coder_ctl In narrowband mode, the sampling rate is just initialized to 8000 st->sampling_rate=8000; BTW, I saw that my email was added to an existing thread instead of

Hi, I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE as in speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, &m_SampleRate ); I get 2 The encoder is initialized with m_pEncoderState = speex_encoder_init ( &speex_wb_mode ); On the decoder side, the correct sampling rate is returned: m_pDecoderState = speex_decoder_init ( &speex_wb_mode );

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

On 12/13/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > I am a newbie here and just started using speex in our lab. > > During compiling my voice application with windows-compiled version provided > > today, I am getting "unresolved external symbol _speex_wb_mode". > > I am probably missing some setting with VS6.0 linker... (I am a bit new

Hi, Has anybody some information on on the "Redundant audio data" header in the speex rtp payload? Is this header always present? Is its value always the same? Can it be modified through some speex_*_ctl function? Thanks, Emmanuel -- ------------------------------------------------------------- Emmanuel Wauters Tel : (+32) 11 30 13 30

I built the CVS version of the Speex library - v1.2 it calls itself. Asterisk seg faults trying to use codec_speex.so. I'll have a look to try to fix it, but thought I'd just ask if anyone else knows what needs to be done? Steve

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, Do you get that problem with speex_nb_mode too (or only speex_wb_mode)? Jean-Marc Le lun 29/03/2004 à 05:35, Chris Flerackers a écrit : > Hi, > > I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE > as in > speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, > &m_SampleRate ); > I get > 2 > The encoder is initialized with

Hi, Right now, I'm thinking of adding a speex_lib_ctl() call that would support SPEEX_GET_VERSION (and return a string) or SPEEX_GET_MAJOR_VERSION and SPEEX_GET_MINOR_VERSION (and return ints). I'm open to other suggestions though. If there's anything you'd like to see in the API for 1.2, say it now. ...and no, I won't add a speex_do_all_the_work_for_me() call :)

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

On Sat, Mar 20, 2010 at 12:04 AM, Jozsef Vass <jovass at adobe.com> wrote: > Nicer way: > > void* speexState = speex_encoder_init(&speex_wb_mode); > int speexFrameSize, speexRate; > speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); > speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); > SpeexPreprocessState*

What are you trying to accomplish? My code sample was about how Flash Player microphone input in speex. If you want to do decode, please see speexdec.c Jozsef -----Original Message----- From: Max Lapshin [mailto:max.lapshin at gmail.com] Sent: Tuesday, March 30, 2010 8:53 AM To: Jozsef Vass Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] Speex in flash player: how to work with? On Sat, Mar

I successfully encoded and decoded a wav file of PCM 16 , MONO and with sample rate 8000 in Android OS. The size of the original wav file and decoded file is near. But i am not able to play the decoded audio file, the mediaplayer says that the file has been corrupted. ENCODING: #include <jni.h> #include <stdio.h> #include "speex/speex.h" #define FRAME_SIZE 320 void

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:

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:

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

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

hello, we start to develop an open source project. This project is a VoIp application with the same features like "Teamspeak" (http://www.goteamspeak.com). Our test application run in peer to peer mode under windows 2000 and above. At the moment we are using the gsm codec (13 kbps bandwidth). Later there will be a port for *unix systems. As aforementioned the project will be released

I am using speex for encoding and decoding speech files in a speech synthesis program. I am using concatenative approach for speech synthesis using diphones (small speech chunks) as basic unit. Now in my application, the order of diphones to be joined after decoding is determined at runtime. Therefore I have to encode each diphone independent of others. I find out that when I encode diphones