From mboxrd@z Thu Jan 1 00:00:00 1970 From: Joe Hsu Subject: What's the right way to do??? resample?? Date: Thu, 31 Mar 2005 10:49:04 +0800 Message-ID: <7fe2059905033018492546d763@mail.gmail.com> Reply-To: Joe Hsu Mime-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Return-path: Sender: alsa-devel-admin@lists.sourceforge.net Errors-To: alsa-devel-admin@lists.sourceforge.net List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , List-Archive: To: alsa-devel@lists.sourceforge.net List-Id: alsa-devel@alsa-project.org I am a beginner on Linux audio programming. When recording from one audio codec and playing back from another audio codec in real time, there must be some asynchronous problems.(Buffer overrun for recording or buffer underrun for playback). I believe this problem must have some answers. Hence should I resample audio in real time dynamically so that asynchronous conditions would be eliminated? If so, what resample library would be suitable?? Libsamplerate or Libresample??? Thanks if anyone can give me some hints. (My recording device is 8000HZ and my playback device also works at 8000HZ) -- The sun is shinny but the ice is slippery. ------------------------------------------------------- This SF.net email is sponsored by Demarc: A global provider of Threat Management Solutions. Download our HomeAdmin security software for free today! http://www.demarc.com/Info/Sentarus/hamr30