From mboxrd@z Thu Jan 1 00:00:00 1970 From: Jonathan Andrews Subject: Re: Trying to understand alsa Date: Thu, 12 Jan 2012 23:45:40 +0000 Message-ID: <1326411940.23310.47.camel@jonspc> Reply-To: jon@jonshouse.co.uk Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail.jonshouse.co.uk (unknown [91.84.151.33]) by alsa0.perex.cz (Postfix) with ESMTP id 2E27024402 for ; Fri, 13 Jan 2012 00:42:03 +0100 (CET) Received: from [10.10.10.6] (jonspc [10.10.10.6]) by mail.jonshouse.co.uk (8.14.1/8.13.8) with ESMTP id q0CNjeSE019613 for ; Thu, 12 Jan 2012 23:45:46 GMT 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-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org On Thu, 2012-01-12 at 16:12 -0600, Andrew Eikum wrote: > 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(). I cant find any reference to "set_buffer_size_min" in the ALSA API documentation I have or the link you provided ? snd_pcm_hw_params_set_buffer_time_near() is the closest and that seems to take an argument in useconds. ? http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > 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. Ok, thanks. > 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. Many thanks for the help, Jon