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rntmfgkgk / csipsimple

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In-Call dialpad should have haptic and tone feedback #139

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?
1. Place a call
2. Wait till called party answers (Talking)
3. Open the dialpad
4. Press dialpad keys

What is the expected output? What do you see instead?

Expected: Tone and/or haptic feedback like the dialer's keys now have
Got: neither

Please use labels and text to provide additional information.

The existing prefs for dialing feedback could control tone and haptic feedback 
on the in-call dialpad. No need for new prefs.

Original issue reported on code.google.com by dc3de...@gmail.com on 11 Aug 2010 at 2:49

GoogleCodeExporter commented 9 years ago

Original comment by dc3de...@gmail.com on 11 Aug 2010 at 6:39

GoogleCodeExporter commented 9 years ago 
See r195 - Refactored dialing feedback (tone and haptic) into new class 
utils/DialingFeedback.java. This is called from both Dialer and InCallActivity. 
I think I was correct in not calling setVolumeControlStream on the 
STREAM_VOICE_CALL in the InCallActivity.

Original comment by dc3de...@gmail.com on 11 Aug 2010 at 6:47

GoogleCodeExporter commented 9 years ago 
Sounds good.

Yes your right, while in call, stream volume control should be managed by 
inCall activity.

Besides one important thing you may not noticed is that the inCall dialer send 
DTMF sounds using the pjsip library. So while in call, not needed to give audio 
feedback (and it can become really unstable to add audio on the top of the 
pjsip streams)

So what I advise that if boolean inCall is true in DialingFeedback to set 
dialPressTone to false : we must make sure that DTMF is only sent by pjsip 
stack.
(Testing that vibrator doesn't produce sound that is mixed with DTMF sent by 
pjsip could be great too... else let me know, I can remove the conference port 
from the micro while dialing a DTMF char).

Original comment by r3gis...@gmail.com on 11 Aug 2010 at 7:15

GoogleCodeExporter commented 9 years ago 
Well, without the TOneGenerator sounds, I hear nothing when I press the in-call 
dialpad. I did see that it is sending DTMF via pjsip stack. However, that DTMF 
is inaudible by the local user. I use the ToneGenerator now to play DTMF 
through the earpiece/speaker/BT. I don't think it is going out to the 
remote/callee. I listened at the other end and heard the same thing (clean 
DTMF) whether I have tone feedback enabled or not. So I conclude that the tone 
feedback is not actually going over the connection to the other end, it is only 
being played at the local end. Again, without that, I hear nothing when 
pressing the in-call dialpad buttons.

The vibrator does not produce any sound that can be heard by the remote end, so 
this is OK also.

Original comment by dc3de...@gmail.com on 11 Aug 2010 at 8:00

GoogleCodeExporter commented 9 years ago 
Oh, yes sorry you're absolutely right. All the more so as output audio is not 
mixed with micro input (excepted on the Sony X10... :/ )

Well should be ok. There is just a little fear about sound mixing on device on 
which there is already issues, but actually not due to this improvement.

Original comment by r3gis...@gmail.com on 11 Aug 2010 at 8:50

GoogleCodeExporter commented 9 years ago 
See r196 - Tone/haptic pref changes not seen until program restart. Fixed this.

Original comment by dc3de...@gmail.com on 11 Aug 2010 at 11:02

GoogleCodeExporter commented 9 years ago 
Just a note: there are three ways to indicate numeric buttons presses in SIP: 
RFC2833 (RTP Payload for DTMF Digits...), RFC2976 (SIP INFO method) and 
"inline". Only the last one involves real DTMF sound signal being encoded with 
the current codec and sent over the RTP audio stream. I think that is rarely 
used these days. The other two send the button press events in as either 
special (not audio) RTP payload or as a SIP message.

Original comment by egcros...@gmail.com on 14 Aug 2010 at 9:46

GoogleCodeExporter commented 9 years ago 
Sending dtmf using the audio stream is the more compatible way. 
Many sip providers that does pstn gateway doesnt provide support for RFC2833 or 
RFC2976.
But, pjsip support both this features (see : 
http://trac.pjsip.org/repos/wiki/FAQ#dtmf). Well it's not yet implemented since 
inline sending is in most case   enough to dial dtmf codes. So we can imagine 
in the future to support both method described in rfc2833 and 2976 but for now 
that's not really necessary.

Original comment by r3gis...@gmail.com on 14 Aug 2010 at 12:41