Change minimum DTMF event duration to be 40 milliseconds

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index f96c8e741e04ad94075ad69add97fab351230e01..52386d70a89eeebb88434081219fb4e790abf1a3 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -102,7 +102,7 @@ const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
 // Constants from voice_engine_defines.h.
 const int kMinTelephoneEventCode = 0;           // RFC4733 (Section 2.3.1)
 const int kMaxTelephoneEventCode = 255;
-const int kMinTelephoneEventDuration = 100;
+const int kMinTelephoneEventDuration = 40;
 const int kMaxTelephoneEventDuration = 60000;   // Actual limit is 2^16

[Taylor Brandstetter, 2017/02/16 18:18:22]

As long as we're changing the min to match the spec, should we change the max?
This is 60 seconds instead of 6.

[the sun, 2017/02/16 19:54:58]

I don't think it makes sense to check ranges at all layers of the call chain, so
these should just be removed. This is not an API boundary.

DtmfSender::DoInsertDtmf() already ensures we're sticking to the WebRTC 1.0
spec, and AFAICT this is the only way we get to
WebRtcVoiceMediaChannel::InsertDtmf().

Channel::SendTelephoneEventOutband() DCHECKs the input is within the bounds
specified in RFC4733 (that's the limit of 65535 which somebody has rounded down
to 60s above). Those checks should be in the module producing the DTMF payloads
(RtpSenderAudio), however that's a story for a different CL.

So I suggest we remove these four constants and their use in
WebRtcVoiceMediaChannel::InsertDtmf().

[Taylor Brandstetter, 2017/02/16 22:46:40]

Sounds good to me.

 
 const int kMinPayloadType = 0;