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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">Thank you for your kind comments. I have blown up my connection with ISP every time I tried writing frames using ast_write for the equivalent of echo. I am
copying the relevant code fromconference.c for elucidation. Kindly forgive me for bringing up the App Konference code here.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">// loop over member list and retrieve incoming frames<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> for (listentry = conf->listheaders->speakerlistheader.next; listentry != &conf->listheaders->speakerlistheader; listentry = listentry->next)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> ast_conf_member *member = (ast_conf_member *)((char*)listentry - offsetof(ast_conf_member, speakerlistentry));<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> member_process_spoken_frames(conf,member,&spoken_frames,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> &listener_count, &speaker_count);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> // mix incoming frames and get batch of outgoing frames<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> conf_frame *send_frames = spoken_frames ? mix_frames(conf, spoken_frames, speaker_count, listener_count) : NULL;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> // loop over member list and send outgoing frames<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> for (listentry = conf->listheaders->memberlistheader.next; listentry != &conf->listheaders->memberlistheader; listentry = listentry->next)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> ast_conf_member *member = (ast_conf_member *)((char*)listentry - offsetof(ast_conf_member, memberlistentry));<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> /* ********************************************************this blows up the ISP connection<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> conf_frame * sendFrame=NULL;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> for(sendFrame=send_frames; sendFrame != NULL;sendFrame = send_frames->next) {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> ast_write(member->chan, sendFrame->fr);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> */<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> member_process_outgoing_frames(conf, member);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"> }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">member_process_outgoing_frames queues the frames to be sent out to the listeners. What I don't understand is why the ISP drops the connection.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">Thanks<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> asterisk-dev-bounces@lists.digium.com [mailto:asterisk-dev-bounces@lists.digium.com]
<b>On Behalf Of </b>Matthew Jordan<br>
<b>Sent:</b> Wednesday, December 24, 2014 11:30 AM<br>
<b>To:</b> Asterisk Developers Mailing List<br>
<b>Subject:</b> Re: [asterisk-dev] Volume Control<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Wed, Dec 24, 2014 at 1:14 PM, Murthy Gandikota <<a href="mailto:mgandikota@nts.net" target="_blank">mgandikota@nts.net</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">To correct myself, it has nothing to do with hardware. I am using ast_openstream acquire the stream
object. I then use ast_write to send a frame on the stream to all the conference members. Meanwhile another thread is waiting for events on the channel (ast_waitfor) and if a spoken frame is sensed, then it is being queued for the conference members. Here
is the catch: if I do ast_write(stream, NULL) and speak up on the phone, then I can hear an echo of myself. So it must be possible to do ast_write(stream, frame) and hear an echo, if the talk volume is sufficiently higher than the listen volume?</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"> </span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I have no idea what App Konference is doing with the media mixing that it is performing. ast_write and ast_waitfor have nothing to do with the act of mixing media however - those are merely calls to write media
to a channel and to wait for a file descriptor on a channel to be triggered. That actual act of mixing the media would have to be provided by something else.<br>
<br>
In bridge_softmix - which is a bridging module for the Bridging Framework that mixes media between n channels - we explicitly remove the source channel from the media that is mixed for it - so you don't hear yourself. But that's an implementation detail of
that mixer - other mixers could choose to echo back the audio from the speakers.<o:p></o:p></p>
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<p class="MsoNormal">Volume, again, is handled by something else completely different from mixing media. In the case of func_volume, an audiohook is used to manipulate the media prior to it being written down into the channel technology. The audiohook itself
is invoked via ast_write.<o:p></o:p></p>
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<p class="MsoNormal"><br>
-- <o:p></o:p></p>
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<p class="MsoNormal">Matthew Jordan<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Digium, Inc. | Engineering Manager<o:p></o:p></p>
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<p class="MsoNormal">445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<o:p></o:p></p>
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<p class="MsoNormal">Check us out at: <a href="http://digium.com" target="_blank">
http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><o:p></o:p></p>
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