From mboxrd@z Thu Jan 1 00:00:00 1970 From: Raymond Yau Subject: Re: [RFC] disabling ALSA period interrupts Date: Tue, 4 May 2010 11:18:51 +0800 Message-ID: References: <1272645879.3220.177.camel@odin> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail-pv0-f179.google.com (mail-pv0-f179.google.com [74.125.83.179]) by alsa0.perex.cz (Postfix) with ESMTP id 2A3BC24472 for ; Tue, 4 May 2010 05:18:52 +0200 (CEST) Received: by pvg7 with SMTP id 7so502444pvg.38 for ; Mon, 03 May 2010 20:18:51 -0700 (PDT) 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: ALSA Development Mailing List List-Id: alsa-devel@alsa-project.org 2010/5/1 pl bossart > Hi Liam, > > > How do you handle any clock drift here between the HDA hardware > > interface clock and the PA timer ? > > Good question. This is already handled by PulseAudio. The timer isn't > programmed with a fixed value but is adapted precisely to track > differences between system time and audio time. seem underrun occur during this adjustment but PA just print "cutting sleeping time" and did not provide any information about the sleeping time for debugging PulseAudio relies on > snd_pcm_avail() to query the number of samples played, and correlates > timer values with samples. This mean that cs46xx and those drivers which cannot provide accurate number of sample played will not work with PA ? -Pierre >