Multi frequency DTMF support - receiver side

webrtc/modules/audio_coding/neteq/dtmf_buffer.cc

 /*
  *  Copyright (c) 2012 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 81 matching lines @@
 // is not known when the first packet describing it is sent. To deal with that,
 // the RFC 4733 specifies that multiple packets are sent for one and the same
 // event as it is being created (typically, as the user is pressing the key).
 // These packets will all share the same start timestamp and event number,
 // while the duration will be the cumulative duration from the start. When
 // inserting a new event, the InsertEvent method tries to find a matching event
 // already in the buffer. If so, the new event is simply merged with the
 // existing one.
 int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
   if (event.event_no < 0 || event.event_no > 15 ||
-      event.volume < 0 || event.volume > 36 ||
+      event.volume < 0 || event.volume > 63 ||
       event.duration <= 0 || event.duration > 65535) {
     LOG(LS_WARNING) << "InsertEvent invalid parameters";
     return kInvalidEventParameters;
   }
   DtmfList::iterator it = buffer_.begin();
   while (it != buffer_.end()) {
     if (MergeEvents(it, event)) {
       // A matching event was found and the new event was merged.
       return kOK;
     }
@@ skipping 126 matching lines @@
 // that belong to the same events, and therefore sharing the same start
 // timestamp, have already been merged before the sort method is called.
 bool DtmfBuffer::CompareEvents(const DtmfEvent& a, const DtmfEvent& b) {
   if (a.timestamp == b.timestamp) {
     return a.event_no < b.event_no;
   }
   // Take wrap-around into account.
   return (static_cast<uint32_t>(b.timestamp - a.timestamp) < 0xFFFFFFFF / 2);
 }
 }  // namespace webrtc