- Make RtpSenderAudio not inherit from DtmfQueue.

- Remove unused method DtmfQueue::ResetDTMF()

BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2365873002
Cr-Commit-Position: refs/heads/master@{#14376}

webrtc/modules/rtp_rtcp/source/dtmf_queue.cc

@@ -59,9 +59,4 @@ bool DTMFqueue::PendingDTMF() {
   rtc::CritScope lock(&dtmf_critsect_);
   return next_empty_index_ > 0;
 }
-
-void DTMFqueue::ResetDTMF() {
-  rtc::CritScope lock(&dtmf_critsect_);
-  next_empty_index_ = 0;
-}
 }  // namespace webrtc

webrtc/modules/rtp_rtcp/source/dtmf_queue.h

@@ -19,12 +19,11 @@ namespace webrtc {
 class DTMFqueue {
  public:
   DTMFqueue();
-  virtual ~DTMFqueue();
+  ~DTMFqueue();
 
   int32_t AddDTMF(uint8_t dtmf_key, uint16_t len, uint8_t level);
   int8_t NextDTMF(uint8_t* dtmf_key, uint16_t* len, uint8_t* level);
   bool PendingDTMF();
-  void ResetDTMF();
 
  private:
   rtc::CriticalSection dtmf_critsect_;

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -169,14 +169,14 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
   }
 
   // Check if we have pending DTMFs to send
-  if (!dtmf_event_is_on_ && PendingDTMF()) {
+  if (!dtmf_event_is_on_ && dtmf_queue_.PendingDTMF()) {
     int64_t delaySinceLastDTMF =
         clock_->TimeInMilliseconds() - dtmf_time_last_sent_;
 
     if (delaySinceLastDTMF > 100) {
       // New tone to play
       dtmf_timestamp_ = rtp_timestamp;
-      if (NextDTMF(&key, &dtmf_length_ms, &dtmf_level_) >= 0) {
+      if (dtmf_queue_.NextDTMF(&key, &dtmf_length_ms, &dtmf_level_) >= 0) {
         dtmf_event_first_packet_sent_ = false;
         dtmf_key_ = key;
         dtmf_length_samples_ = (kDtmfFrequencyHz / 1000) * dtmf_length_ms;
@@ -310,7 +310,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
       return -1;
     }
   }
-  return AddDTMF(key, time_ms, level);
+  return dtmf_queue_.AddDTMF(key, time_ms, level);
 }
 
 bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
@@ -22,10 +23,10 @@
 
 namespace webrtc {
 
-class RTPSenderAudio : public DTMFqueue {
+class RTPSenderAudio {
  public:
   RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
-  virtual ~RTPSenderAudio();
+  ~RTPSenderAudio();
 
   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                int8_t payload_type,
@@ -83,6 +84,7 @@ class RTPSenderAudio : public DTMFqueue {
   uint8_t dtmf_level_;
   int64_t dtmf_time_last_sent_;
   uint32_t dtmf_timestamp_last_sent_;
+  DTMFqueue dtmf_queue_;
 
   // VAD detection, used for marker bit.
   bool inband_vad_active_ GUARDED_BY(send_audio_critsect_);
@@ -96,6 +98,8 @@ class RTPSenderAudio : public DTMFqueue {
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
   OneTimeEvent first_packet_sent_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
 };
 
 }  // namespace webrtc