From mboxrd@z Thu Jan 1 00:00:00 1970 From: Andrew Eikum Subject: Re: Trying to understand alsa Date: Thu, 12 Jan 2012 16:12:18 -0600 Message-ID: <20120112221218.GE24575@foghorn.codeweavers.com> References: <1326404651.22083.28.camel@jonspc> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail.codeweavers.com (mail.codeweavers.com [216.251.189.131]) by alsa0.perex.cz (Postfix) with ESMTP id A770424398 for ; Thu, 12 Jan 2012 23:12:24 +0100 (CET) Content-Disposition: inline In-Reply-To: <1326404651.22083.28.camel@jonspc> 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: Jonathan Andrews Cc: alsa-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > I have an application that works using 512 sample packets of 22050Hz 16 > bit mono audio. The 'receiver' takes many audio streams from a network > via UDP, at the moment it pipes them into pulse. > > Can alsa buffer audio. At the moment every time I and set an audio > buffer size I get a negative response from > snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the > units alsa uses ... > You don't want to over-specify your requirements. You require a buffer size of "at least" 3 * 512 frames. So use set_buffer_size_min(). Otherwise ALSA will try to set exactly that buffer size, which can fail. Check the function signatures for units. Notice that set_buffer_size*() all take snd_pcm_uframes_t, that is, the number of frames you want to store. In ALSA terms, a "frame" is a set of a single sample for every channel. Since you have mono audio, a frame and a sample are actually the same unit (for 16 bit stereo audio, 1 frame = 2 samples = 32 bits). > What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x > 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K > Byte, 512 samples per buffer) in for playback. The packets are arriving > at roughly the correct rate, I just need a buffer to iron out any > jitter in network transmit, do I have to do this myself ? > > Can somebody help by telling me which numbers I push into which places > to make it work ? > > At the moment I get i keep getting a broken pipe, if I underrun how can > I make it just wait for me ? > If a packet arrives very late (and one will, eventually), you will underrun. That's unavoidable. You can check for SND_PCM_STATE_XRUN from snd_pcm_state(). It's undocumented, but you need to call snd_pcm_avail_update() first to get an accurate reading from snd_pcm_state(). When an underrun occurs, recover with snd_pcm_recover() and then start writing data again. If a packet arrives early, you'll need to check that the ALSA buffer isn't full (see snd_pcm_avail_update()), and store it within your application to write later if it is full. There's some mostly-accurate information here: http://0pointer.de/blog/projects/guide-to-sound-apis.html As with all ALSA documentation, it is confusing and often incorrect, but it's probably the most helpful document I've found. Good luck! Andrew