Multi frequency DTMF support - sender side

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

 /*
  *  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 39 matching lines @@
   // header-extension-for-audio-level-indication.
   // Valid range is [0,100]. Actual value is negative.
   int32_t SetAudioLevel(uint8_t level_dbov);
 
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
  protected:
   bool SendTelephoneEventPacket(
       bool ended,
-      int8_t dtmf_payload_type,
       uint32_t dtmf_timestamp,
       uint16_t duration,
       bool marker_bit);  // set on first packet in talk burst
 
   bool MarkerBit(FrameType frame_type, int8_t payload_type);
 
  private:
   Clock* const clock_;
   RTPSender* const rtp_sender_;
 
   rtc::CriticalSection send_audio_critsect_;
 
-  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_);
+  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_) = 160;
 
   // DTMF.
-  bool dtmf_event_is_on_;
-  bool dtmf_event_first_packet_sent_;
-  int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_);
-  uint32_t dtmf_timestamp_;
-  uint8_t dtmf_key_;
-  uint32_t dtmf_length_samples_;
-  uint8_t dtmf_level_;
-  int64_t dtmf_time_last_sent_;
-  uint32_t dtmf_timestamp_last_sent_;
-  DTMFqueue dtmf_queue_;
+  bool dtmf_event_is_on_ = false;
+  bool dtmf_event_first_packet_sent_ = false;
+  int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
+  uint32_t dtmf_timestamp_ = 0;
+  uint32_t dtmf_length_samples_ = 0;
+  int64_t dtmf_time_last_sent_ = 0;
+  uint32_t dtmf_timestamp_last_sent_ = 0;
+  DtmfQueue::Event dtmf_current_event_;
+  DtmfQueue dtmf_queue_;
 
   // VAD detection, used for marker bit.
-  bool inband_vad_active_ GUARDED_BY(send_audio_critsect_);
-  int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_);
-  int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_);
-  int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_);
-  int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_);
-  int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_);
+  bool inband_vad_active_ GUARDED_BY(send_audio_critsect_) = false;
+  int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
+  int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
+  int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
+  int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
+  int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
 
   // Audio level indication.
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
-  uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
+  uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_) = 0;

[stefan-webrtc, 2016/11/08 15:05:39]

We've tended to prefer initializer lists as splitting initialization between
initializer lists and header files can be confusing (can't easily look at one
place to know if something is initialized).

[the sun, 2016/11/11 12:07:24]

Ah yes, that's why I set clock_ and rtp_sender_ to nullptr even though they're
also set in the ctor. I'm reverting to that. Initializing at declaration is
preferable because then you only need to look in one place to see both what
members the class has and whether they're initialized. It also avoids needing to
duplicate init lists in case of multiple ctors (not the case here though).

   OneTimeEvent first_packet_sent_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_