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.
// loop over member list and retrieve incoming frames
for (listentry = conf->listheaders->speakerlistheader.next; listentry
!= &conf->listheaders->speakerlistheader; listentry = listentry->next)
{
ast_conf_member *member = (ast_conf_member *)((char*)listentry
- offsetof(ast_conf_member, speakerlistentry));
member_process_spoken_frames(conf,member,&spoken_frames,
&listener_count, &speaker_count);
}
// mix incoming frames and get batch of outgoing frames
conf_frame *send_frames = spoken_frames ? mix_frames(conf,
spoken_frames, speaker_count, listener_count) : NULL;
// loop over member list and send outgoing frames
for (listentry = conf->listheaders->memberlistheader.next; listentry !=
&conf->listheaders->memberlistheader; listentry = listentry->next)
{
ast_conf_member *member = (ast_conf_member *)((char*)listentry
- offsetof(ast_conf_member, memberlistentry));
/* ********************************************************this blows
up the ISP connection
conf_frame * sendFrame=NULL;
for(sendFrame=send_frames; sendFrame != NULL;sendFrame =
send_frames->next) {
ast_write(member->chan, sendFrame->fr);
}
*/
member_process_outgoing_frames(conf, member);
}
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.
Thanks
From: [email protected]
[mailto:[email protected]] On Behalf Of Matthew Jordan
Sent: Wednesday, December 24, 2014 11:30 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Volume Control
On Wed, Dec 24, 2014 at 1:14 PM, Murthy Gandikota
<[email protected]<mailto:[email protected]>> wrote:
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?
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.
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.
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.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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