<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:x="urn:schemas-microsoft-com:office:excel" xmlns:p="urn:schemas-microsoft-com:office:powerpoint" xmlns:a="urn:schemas-microsoft-com:office:access" xmlns:dt="uuid:C2F41010-65B3-11d1-A29F-00AA00C14882" xmlns:s="uuid:BDC6E3F0-6DA3-11d1-A2A3-00AA00C14882" xmlns:rs="urn:schemas-microsoft-com:rowset" xmlns:z="#RowsetSchema" xmlns:b="urn:schemas-microsoft-com:office:publisher" xmlns:ss="urn:schemas-microsoft-com:office:spreadsheet" xmlns:c="urn:schemas-microsoft-com:office:component:spreadsheet" xmlns:odc="urn:schemas-microsoft-com:office:odc" xmlns:oa="urn:schemas-microsoft-com:office:activation" xmlns:html="http://www.w3.org/TR/REC-html40" xmlns:q="http://schemas.xmlsoap.org/soap/envelope/" xmlns:rtc="http://microsoft.com/officenet/conferencing" xmlns:D="DAV:" xmlns:Repl="http://schemas.microsoft.com/repl/" xmlns:mt="http://schemas.microsoft.com/sharepoint/soap/meetings/" xmlns:x2="http://schemas.microsoft.com/office/excel/2003/xml" xmlns:ppda="http://www.passport.com/NameSpace.xsd" xmlns:ois="http://schemas.microsoft.com/sharepoint/soap/ois/" xmlns:dir="http://schemas.microsoft.com/sharepoint/soap/directory/" xmlns:ds="http://www.w3.org/2000/09/xmldsig#" xmlns:dsp="http://schemas.microsoft.com/sharepoint/dsp" xmlns:udc="http://schemas.microsoft.com/data/udc" xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:sub="http://schemas.microsoft.com/sharepoint/soap/2002/1/alerts/" xmlns:ec="http://www.w3.org/2001/04/xmlenc#" xmlns:sp="http://schemas.microsoft.com/sharepoint/" xmlns:sps="http://schemas.microsoft.com/sharepoint/soap/" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns:udcs="http://schemas.microsoft.com/data/udc/soap" xmlns:udcxf="http://schemas.microsoft.com/data/udc/xmlfile" xmlns:udcp2p="http://schemas.microsoft.com/data/udc/parttopart" xmlns:wf="http://schemas.microsoft.com/sharepoint/soap/workflow/" xmlns:dsss="http://schemas.microsoft.com/office/2006/digsig-setup" xmlns:dssi="http://schemas.microsoft.com/office/2006/digsig" xmlns:mdssi="http://schemas.openxmlformats.org/package/2006/digital-signature" xmlns:mver="http://schemas.openxmlformats.org/markup-compatibility/2006" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns:mrels="http://schemas.openxmlformats.org/package/2006/relationships" xmlns:spwp="http://microsoft.com/sharepoint/webpartpages" xmlns:ex12t="http://schemas.microsoft.com/exchange/services/2006/types" xmlns:ex12m="http://schemas.microsoft.com/exchange/services/2006/messages" xmlns:pptsl="http://schemas.microsoft.com/sharepoint/soap/SlideLibrary/" xmlns:spsl="http://microsoft.com/webservices/SharePointPortalServer/PublishedLinksService" xmlns:Z="urn:schemas-microsoft-com:" xmlns:st="" 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=ZH-CN link=blue vlink=purple style='text-justify-trim:punctuation'><div class=WordSection1>I found when moving a running live stream (of Gtalk) from an ALSA sink to a new created Bluetooth HSP sink, buffer underruns and rewind flood can happen and so I can no longer hear the voice from the other side. I’ve tried the following methods but still cannot avoid buffer underrun 100%. Could anyone share any idea about other methods? <o:p></o:p><o:p> </o:p>Here are the methods I tried:<o:p></o:p><![if !supportLists]>1.       <![endif]>Using the application original requested latency when moving sinks, with patch of Arun <a href="http://lists.freedesktop.org/archives/pulseaudio-discuss/2011-June/010291.html">http://lists.freedesktop.org/archives/pulseaudio-discuss/2011-June/010291.html</a><o:p></o:p>But, I found the original latency (200ms) is still not enough to overcome excessive underrun. After I increase the audio latency to 500ms, there are no excessive underruns for most time. But not 100% safe. <o:p></o:p><o:p> </o:p><![if !supportLists]>2.       <![endif]>Force the implementor to rewind by itself when underrun happens. (original audio latenty 200ms is used on moving the sink)<o:p></o:p>I found for the BT HSP sink cannot really rewind, because It does not implement “request_rewind” function on the sink and its “sink->thread_info.max_rewind” is 0.  <o:p></o:p>So I modified handle_seek() function in the “protocol-native.c” to let the implement rewind by itself:<o:p></o:p><o:p> </o:p>if (indexw < indexr) {<o:p></o:p>            /* OK, the sink already asked for this data, so<o:p></o:p>             * let's have it usk us again */<o:p></o:p><o:p> </o:p>            pa_log_debug("Requesting rewind due to rewrite.!");<o:p></o:p>            pa_sink_input_request_rewind(s->sink_input, (size_t) (indexr - indexw), TRUE, FALSE, FALSE);<o:p></o:p><o:p> </o:p>            <o:p></o:p>            + /* amanda: to fight BT underrun */<o:p></o:p>            + if(s->sink_input->sink)<o:p></o:p>            +{<o:p></o:p>            +    pa_sink * sink = s->sink_input->sink;<o:p></o:p>            +    if(!sink->request_rewind && !sink->thread_info.max_rewind)<o:p></o:p>            +    {<o:p></o:p>            +        /* the sink cannot handle rewind, implementor rewind by itself */                                       <o:p></o:p>            +       sink_input_process_rewind_cb(s->sink_input, (size_t) (indexr - indexw));<o:p></o:p>            +      }<o:p></o:p>            +}<o:p></o:p>            <o:p></o:p>        }<o:p></o:p>Unfortunately, excessive rewind cannot be avoided and the voice is frequently interrupted.<o:p></o:p><o:p> </o:p>What other method can I try? How to ask the application to feed more data? <o:p></o:p>Does the application write data during sink moving so that there are some data buffered for the destination sink? <o:p></o:p><o:p> </o:p>Great thanks!<o:p></o:p>Amanda<o:p></o:p><o:p> </o:p></div></body></html>