Make RtpSenderAudio not inherit from DtmfQueue

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 30 matching lines @@
       dtmf_time_last_sent_(0),
       dtmf_timestamp_last_sent_(0),
       inband_vad_active_(false),
       cngnb_payload_type_(-1),
       cngwb_payload_type_(-1),
       cngswb_payload_type_(-1),
       cngfb_payload_type_(-1),
       last_payload_type_(-1),
       audio_level_dbov_(0) {}
 
-RTPSenderAudio::~RTPSenderAudio() {}
-
 int RTPSenderAudio::AudioFrequency() const {
   return kDtmfFrequencyHz;
 }
 
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
   rtc::CritScope cs(&send_audio_critsect_);
   packet_size_samples_ = packet_size_samples;
   return 0;
@@ skipping 99 matching lines @@
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
     rtc::CritScope cs(&send_audio_critsect_);
     audio_level_dbov = audio_level_dbov_;
     dtmf_payload_type = dtmf_payload_type_;
     packet_size_samples = packet_size_samples_;
   }
 
   // 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;
         dtmf_event_is_on_ = true;
       }
     }
   }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it
@@ skipping 113 matching lines @@
 int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,
                                            uint8_t level) {
   {
     rtc::CritScope lock(&send_audio_critsect_);
     if (dtmf_payload_type_ < 0) {
       // TelephoneEvent payloadtype not configured
       return -1;
     }
   }
-  return AddDTMF(key, time_ms, level);
+  return dtmf_queue_.AddDTMF(key, time_ms, level);
 }
 
 bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                                               int8_t dtmf_payload_type,
                                               uint32_t dtmf_timestamp,
                                               uint16_t duration,
                                               bool marker_bit) {
   uint8_t send_count = 1;
   bool result = true;
 
@@ skipping 41 matching lines @@
         TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
         "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber());
     result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
                                         RtpPacketSender::kHighPriority);
     send_count--;
   } while (send_count > 0 && result);
 
   return result;
 }
 }  // namespace webrtc