Multi frequency DTMF support - sender side

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index aaee73956ca59639fa855ec359fb209edbf63e05..9808901ff9326ad7c333339301448739fa7675b9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -57,7 +57,6 @@ class RTPSenderAudio {
  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
@@ -70,31 +69,30 @@ class RTPSenderAudio {
 
   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);