From mboxrd@z Thu Jan 1 00:00:00 1970 From: Clemens Ladisch Subject: Re: using ALSA one call Date: Wed, 11 Feb 2009 09:55:00 +0100 Message-ID: <49929264.1060503@ladisch.de> References: <1234269014.49917356f0157@mail.inbox.lv> <49919765.4000706@ladisch.de> <1234306055.499204078ce73@mail.inbox.lv> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from out1.smtp.messagingengine.com (out1.smtp.messagingengine.com [66.111.4.25]) by alsa0.perex.cz (Postfix) with ESMTP id 40BAC10388C for ; Wed, 11 Feb 2009 09:55:08 +0100 (CET) In-Reply-To: <1234306055.499204078ce73@mail.inbox.lv> 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: Alfs Kurmis Cc: alsa-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org Alfs Kurmis wrote: > As given here > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g6aa164ed37308d66bcc079f5cd265a09 > latency required overall latency in us (0 = optimum latency for players) > i have try to set optimum latency = 0; That documentation is wrong; 0 is not allowed. Just use 0.5 s, or some larger value. > But why i can not play my wave on notebook sound device "hw:0,0" ?? Because the "hw" device goes straight to the hardware and disallows any conversion of sample rate/format. Use "default", or something like "default:0" to select a specific card. > snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, > channelz, samplerate , 0, 500000) ... and setting soft_resample to 0 disallows resampling in any case. HTH Clemens