similar to: acoustic echo canceller & noise reduction sample code

I am trying to compile speexclient.c. I downloaded all the files to ~/speex- 1.2beta1/include When I call ./compile.sh I get the following errors, speexclient.c: In function 'main': speexclient.c:164: error: 'SPEEX_PREPROCESS_SET_ECHO_STATE' undeclared (first use in this function) speexclient.c:164: error: (Each undeclared identifier is reported only once speexclient.c:164:

Hi All, I would like to communicate speech over a network compressed using speex. However, I do not want to communicate a whole Ogg-formatted file. I am interested in only the speech frames. I invoked: speexenc input-file-name - > raw-speech This, I am thinking gives the speech frames only. To play this back at the receiving end, do I need to format it into Ogg formatted file before I can

Hi all: How to use jitter.c, I see this file is not include in the lib. Sorry for this newbie Q.

Hello Jean-Marc, Hello speex-dev-people, Jean-Marc Valin wrote: > All I can suggest is having a look at my simple client at: > http://people.xiph.org/~jm/speexclient/ Unfortunately, this does not compile: <<<<< snip >>>>> bjoern@phobos:~/beleg/speex/speexclient-0.0-dev$ ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether

> Message: 1 > Date: Thu, 15 Jul 2010 00:19:43 -0600 > From: devi at purpletalk.com > Subject: [Speex-dev] Speex Echo Cancellation > To: speex-dev at xiph.org > Message-ID: <20100715001943.pr04hvtxsc8k04so at mail.purpletalk.com> > Content-Type: text/plain; charset=ISO-8859-1; DelSp="Yes"; > format="flowed" > > Hi > > I am using speex

For a streaming application like VOIP, you collect 20 ms of samples, feed this through the encoder, stick it in an RTP packet, and send if over the network. On the receive side you feed packets through a jitter buffer to the decoder, and then copy the output audio to your output device. Speex runs efficiently on most compilers, so the real-time requirement is not a big deal, as long as you

Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc;

Hello, is there any sample code that uses libspeex? I wrote an app that uses it according to the manual, but the encoding does not sound good enough. I haven't figured out if it's a bug or a configuration. Is there any simple code that encodes properly? yx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, As you may have guess, Speex 1.2 is slowly approaching, though there's still a lot left to do so I can't say how long it'll take. I thought this was the right time to ask if there's anything missing or that can be improved to make 1.2 better. At this point, it can't be anything major, but there are still some changes that are possible, e.g: - Improving some

Hello, In a way similar to speexclient voip example, I want to have a voip application where 3 or more users can interact. This creates the following problems "1. How to playback 2 or more audio streams such that they appear overlayed (parallel)". The audio streams are speex encoded. I have few approaches in mind. 1. Create multiple decoder states and multiple ALSA audio playback

Hi, Indeed I too have troubles implementing echo removal, I like ask kindly to Jean-Marc (or any other) if him can put a source code demo to show us how to use effectively echo removal API and parameters in real case scenario. A big thank you! Roberto -----Messaggio originale----- Da: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] Per conto di hs Inviato: marted? 11 ottobre

Daniel Schmidt a ?crit : > I found the cause of the problem. The function shift_timings can > produce overflows in the timing array if the jitter is huge or the > time units are very short. After changing the timing values' type from > spx_int16_t to spx_int32_t it seems to work. Hmm, I always assumed there wouldn't be any overflows. What parameter range are you using that

On Tue, 11 Oct 2005 10:36:51 +0200, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > Source code at: > http://people.xiph.org/~jm/speexclient/ I rewritten my program so it is more similar to yours, I grabbed your code for storing and retrieving echo, and I have better results, I think the echo is cancelled in about 50% but it still can be heared. I think there is a

Hi all, i've been having problems for the past little while with finding a app library for my voip application. then looking at speexclient i noticed there are timestamps and sequence numbers included in the implementation. my question is: should i get a third party RTP lib for my application or simply emulate the speex client in its RTP like properties. wouldn't it be easier and

Hi, I'd like to report a bug in speexclient/speex_jitter_buffer.c. In the function speex_jitter_get(), after the line: packet.data = data; add the line packet.len = 2048; If this variable is uninitialized, then in jitter_buffer_get() (in jitter.c), the predicate "jitter->packets[i].len > packet->len" is undetermined. When this happens, the wrong amount of

Thanks for your quick response. Do you know how many playback frames that would be considered "ok" to buffer up in this scenario? I've tested the speexclient application and got similar results. There isn't any option to the playback/capture-mechanism for adjusting the buffer size? It feels like doing my own buffering would simply be a copy paste of your buffering but with a

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

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

Hi All, I am working on my master project to port the Speex Acousic Canceller to FPGA. But I am not able to get a satisfied result in its performance whenI integrated it to a SIP softphone as my first step. The SIP softphone calls to a PSTN gateway and lands to my land line phone. I can hear the echo of my every word while I talks on the land line phone. I am using G.711 codec, frame_size 80

Hi Jean-Marc, Thanks for the suggesting. I tried the test wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo cancelleris not really