From mboxrd@z Thu Jan 1 00:00:00 1970 From: Clemens Ladisch Subject: Re: Getting the actual HW sample rate Date: Thu, 08 Sep 2011 10:10:53 +0200 Message-ID: <4E68788D.8040306@ladisch.de> References: Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from out4.smtp.messagingengine.com (out4.smtp.messagingengine.com [66.111.4.28]) by alsa0.perex.cz (Postfix) with ESMTP id 8635B24809 for ; Thu, 8 Sep 2011 10:10:27 +0200 (CEST) In-Reply-To: List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: alsa-devel-bounces@alsa-project.org Errors-To: alsa-devel-bounces@alsa-project.org To: Raymond Toy Cc: alsa-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org Raymond Toy wrote: > In my application, we want ... to avoid yet another sample rate > conversion when playing out the audio. Add the SND_PCM_NO_AUTO_RESAMPLE flag when calling snd_pcm_open, or use snd_pcm_hw_params_set_rate_resample. Regards, Clemens