<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"> <head> <meta http-equiv=Content-Type content="text/html; charset=us-ascii"> <meta name=Generator content="Microsoft Word 12 (filtered medium)"> <style>  </style>  </head> <body lang=EN-US link=blue vlink=purple> <div class=Section1> Hi, I’m trying to write an audio sink for OpenAL (if such a thing already exists, please point me to it, because I haven’t found it! J ). I’m subclassing GstAudioSink and am a bit confused regarding some of the values passed in the prepare() function (GstRingBufferSpec struct) and would appreciate a bit of direction. From the docs:<o:p></o:p> <o:p> </o:p>   guint64  latency_time;        /* the required/actual latency time, this is the<o:p></o:p>                                 * actual the size of one segment and the<o:p></o:p>                                 * minimum possible latency we can achieve. */<o:p></o:p>   guint64  buffer_time;         /* the required/actual time of the buffer, this is<o:p></o:p>                                 * the total size of the buffer and maximum<o:p></o:p>                                 * latency we can compensate for. */<o:p></o:p>   gint     segsize;             /* size of one buffer segment in bytes, this value<o:p></o:p>                                 * should be chosen to match latency_time as<o:p></o:p>                                 * well as possible. */<o:p></o:p>   gint     segtotal;            /* total number of segments, this value is the<o:p></o:p>                                 * number of segments of @segsize and should be<o:p></o:p>                                 * chosen so that it matches buffer_time as<o:p></o:p>                                 * close as possible. */<o:p></o:p> <o:p> </o:p> All 4 values are under [in/out] so I don’t know if it’s inputting recommended values with the option of tweaking them? Regarding latency_time/buffer_time, I’m leaving them as defaults (they can be modified thru the properties) and by default I get values of 10,000/0 respectively with my test. Segsize/segtotal I take as “buffer size” and “number of buffers”, with OpenAL I have to pre-allocate buffers and while running I write to the buffer and queue them (and unqueue processed buffers). And so in my test I get 882 segsize and 20 segtotal so I pre-allocate 20 OpenAL buffers and fill/queue them during the write() call but what appears to be happening is that all the buffers gets full before even one buffer gets processed by OpenAL so I end up overrunning my buffers. Should I be stalling the write() call? Also, do those values make sense? When I normally stream a sound in OpenAL (not synced to video) I would allocate fairly large buffers, but not so many (2-3 buffers large enough to keep about 2-3 seconds buffered).  But here from what I understand it’s suggesting I allocate buffers of only a few samples each (i.e. 882 / 4 = 220 samples).<o:p></o:p> <o:p> </o:p> At this point I’m not too worried about latency/delays; I’m just trying to get something playing sound so that I get a better feel as to how this interface works.<o:p></o:p> <o:p> </o:p> Any input appreciated!<o:p></o:p> <o:p> </o:p> Thanks<o:p></o:p> Denis<o:p></o:p> </div> <pre>CONFIDENTIALITY NOTICE: The contents of this email are confidential and for the exclusive use of the intended recipient. If you receive this email in error, please delete it from your system immediately and notify us either by email, telephone or fax. You should not copy, forward, or otherwise disclose the content of the email. </pre></body> </html>