Change minimum DTMF event duration to be 40 milliseconds

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 84 matching lines @@
 const int kIsacMaxBitrateBps = 56000;
 
 // Default audio dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details.
 // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
 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;

[the sun, 2017/02/14 21:57:35]

Refer to https://w3c.github.io/webrtc-pc/#rtcdtmfsender in the CL description
(+a comment here).

Also, you should update the limits in webrtc/pc/dtmfsender.cc and add the
missing unit tests.

 const int kMaxTelephoneEventDuration = 60000;   // Actual limit is 2^16
 
 const int kMinPayloadType = 0;
 const int kMaxPayloadType = 127;
 
 class ProxySink : public webrtc::AudioSinkInterface {
  public:
   ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }
 
   void OnData(const Data& audio) override { sink_->OnData(audio); }
@@ skipping 2582 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE