similar to: speex and rtp

I discovered that starting from svn commit 13186, liibspeex does not work anymore. Version 13178 and earlier works ok. Revision: 13186 Author: jm Date: 0:16:56, 25 June 2007 Message: Tuned the vocoder gains and prevented the ol_gain from being reduced by the encoder for voiced segments. ---- Modified : /trunk/speex/libspeex/nb_celp.c I tested it on Windows Mobile 5.0 PocketPC platform with

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:

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

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

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, 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

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

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 everyone, I?m a graduate student in a Brazilian Intitute of Technology, and I?m doing some academic research regarding secure voice transmission over phone lines. One of our reserach goals is to implement a hardware vocoder, with low bit rates, and a preferably free algorithm, to be used in this secure voice system. Actually, there is a functional system using a proprietary AMBE

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

Hi, I'm trying to use the jitter buffer feature that comes with Speex but I'm getting unexpected latency. I wrote a client application that does VOIP-like functions and without using jitter buffer, the end-to-end latency is around 250 ms (I'm using lowband 5.97 kpbs). However, when I tried to incorporate the jitter buffer feature, the latency would grow as time elapsed (up to a few

Hi, My name is Jean Quirion and I am a DSP engineer. Currently I am working on a project where it is desired to implement a VoIP solution over a GSM GPRS link. I would like to use Speex as the vocoder for this application. This application would require the Speex encoder/decoder and possibly the pre-processor to run on a low power fixed-point DSP such as a TI C55x. Thus, I am interested in

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, 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:

Hello, I'm interested in a low rate vocoder to run on SA-1110 or XScale processor (for PDAs voice application). Is there a suitable speex vocoder for that? Thanks in advance, Eyal. ________________________________________________ Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag --- >8 ---- List archives:

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hello? My name is Hyeonwoo Oh. I?m developing video conference software using speex library. Can you give me some sample codes for acoustic echo canceller and noise reduction? I tried noise reduction API, but failed. Normal encode/decode API is working well. Thank you very much for your good job. I?m very pleased with speex. Best regards, Hyeonwoo Oh