similar to: Integration / Playback

I am using 'portaudio' and 'speex' to implement a speech audio capture/playback. Preprocess works fine but not the echo canceller (associated with preprocessor). I have read 'testecho.c' but it didn't help me. What I am doing bad? This is my implementation: One thread to capture and playback, both are different functions called in different times directly from

I am making a lightweight opus audio player. This is my portaudio callback function. But i hear a stammering song when i try to play a .opus song file. Can anybody say what is wrong or provide me link to any code which can play opus file written in c/c++(light). I tried to see some open source(vlc opus play) but was unable to follow. int MyCallback(const void* input, void* output, unsigned

Hi All, I try to compile the last version of iaxclient with the new build system, i also compiled speex 1.2b2 . When the makefile try to compile the sample code testcall I have the following mistales: gcc -I/d/Lib/iaxclient-2.0.1/portaudio-1.9/src/common -I/d/Lib/iaxclient-2.0.1/speex-1.2/include -o testcall.exe testcall.o ../../lib/.libs/libiaxclient.a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I was wondering if anyone had any advice on an audio library that is better suited to be paired with Celt in terms of latency. I'm working on an application that I would like to have running on both windows/Linux systems. I started by using the OpenAL library but I have run into an issue when feeding OpenAL small mono sample sizes that

Hello, I have dozens of video files that were encoded using the speex format of audio and can not seem to find any video playback devices that support speex lol. Literally every playback device I've encountered (mac, 10.10.2) plays the videos w no sound. Do you know of any playback devices for the mac format that support speex? -------------- next part -------------- An HTML attachment was

Speaking of portability and people more familiar with SDL: The reason we have two example players for theora is that monty wrote the original example around the OSS audio interface used on several free unix-like operating systems, which immediately excludes win32 and MacOS, even though it used SDL to display the frames. SDL also has audio support, and is quite widely ported. splayer replaces

Hi, Jean-Marc, and thanks for the quick reply. Let me just say I'm a huge fan of speex, and the work you've done. I actually barely understand what I'm reading so far in the source code and documentation, just enough to understand just how cool the algorithms are. LPC10 and MELP allow me to speed up speech with a simple hack on the decoder frame size. Playing fewer samples per

If you have a couple minutes, it probably wouldn't be hard to see where the bug is (I just don't have time this week). It would have to do with the granulepos computation in speexenc. Jean-Marc Le jeudi 07 juillet 2005 ? 08:41 +0300, David Resnick a ?crit : > Thank you very much for your immediate attention! > > I've exported my wav files to raw files before encoding and

Hi everyone, I'd like to present an open-source project I made for use in my company's projects: Theora Playback Library website: http://libtheoraplayer.sourceforge.net It's a multi-threaded library that decodes theora videos, decoding frames in advance (precaching), decodes any syncs audio and provides a very fast YUV->RGB conversion. One of the most important features is that

Thank you very much for your immediate attention! I've exported my wav files to raw files before encoding and the click is now gone. That sure beats discarding the last packet, which was the workaround "solution" I was going to implement... Best regards, David Resnick -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Thursday, July 07,

Hi Bill, Any attempt to alter speed by simple insert or dropping produces poor results. Even if you can get it to sound smooth, the resulting pitch shift is horrible. You really need to use a transform that alters speed smoothly, while maintaining the original pitch of the voice. If you look in my spandsp library you will find a module which does exactly this, using an algorithm called

I was able to easily hack in an option to play back at different speeds. For example, using "speexdec --speed 2.0 file.enc file.wav" plays back encoded file.enc at 2X speed. What I did was divide st->frameSize and st->subFrameSize by the speedup, and added a SPEEX_SET_SPEED decoder control for the nb_celp decoder. This produced speech that was 2X faster than the original.

Hi! I?ve just discovered Speex, and sorry if this isn?t the right place to ask this, but I found no other mailing list (some projects have a "users" and a "developers" list separately, I guess this isn?t the case with Speex. If it is, please let me know). First, how I got here: 1. I?m looking for some low-bitrate codec to reduce the size of some very long (think 4-5+ hrs of

Hi, Steve. I tried your the time_scale_tests program, and it works well! Especially for low speed changes, it's the best I've heard so far. For high speed increases, there is what sounds like static added to the sound output. I've attached two sound samples of high speed speech, which is a 4X speed up of a popular TTS voice in the blind community (voxin/Eloquence). I've sped

Actually, I just tried again with the wav file directly and it just seems to be a bug with the way the speexenc utility handles wav files. The problem doesn't occur with the equivalent raw file (which I used for testing). It's probably really stupid anyway. Jean-Marc Le mercredi 06 juillet 2005 ? 14:00 +0300, David Resnick a ?crit : > I am using speexenc/speexdec to convert WAV files

Hi all, I'm having a playback issue I can't eliminate. I'm encoding 16khz mono WAV files (voiceover recordings) to speex using a quality setting of 4. In any standalone speex player e.g. winamp or using speex frontend etc. the files play fine, but in a jspeex environment there's a subtle high pitched beeping type distortion, mostly on the transients. The sound is similar to

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

I'm getting garbled playback with decoded fragments and I'm hoping someone here can point me in the right direction to correcting the problem. Essentially I'm capturing audio from the microphone. I stream it over the net, but for testing purposes with this API I'm just grabbing the whole chunk and encoding / decoding it right away and then updating the sound buffer for

Great! I am trying now. Thanks Alex ----- Original Message ----- From: Fran?ois Guerraz To: ALEX NG Cc: Jim Crichton ; greg@mytantrum.com ; speex-dev@xiph.org Sent: Friday, May 25, 2007 11:27 PM Subject: Re: [Speex-dev] Re: compatibility issues. Hi, For the openSpeak project we use PortAudio V19 & speex. So if you want some code examples, you can look an pick in our

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).