Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_receiver_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 46 matching lines @@
   bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
 
   int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
 
   int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const override;
 
+  // We do not allow codecs to have multiple payload types for audio, so we
+  // need to override the default behavior (which is to do nothing).
+  void PossiblyRemoveExistingPayloadType(
+      RtpUtility::PayloadTypeMap* payload_type_map,
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      size_t payload_name_length,
+      uint32_t frequency,
+      uint8_t channels,
+      uint32_t rate) const;
+
   // We need to look out for special payload types here and sometimes reset
   // statistics. In addition we sometimes need to tweak the frequency.
   void CheckPayloadChanged(int8_t payload_type,
                            PayloadUnion* specific_payload,
                            bool* should_discard_changes) override;
 
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
  private:
   int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
                                   const uint8_t* payload_data,
                                   size_t payload_length,
                                   const AudioPayload& audio_specific,
                                   bool is_red);
 
+  uint32_t last_received_frequency_;
+
   bool telephone_event_forward_to_decoder_;
   int8_t telephone_event_payload_type_;
   std::set<uint8_t> telephone_event_reported_;
 
   int8_t cng_nb_payload_type_;
   int8_t cng_wb_payload_type_;
   int8_t cng_swb_payload_type_;
   int8_t cng_fb_payload_type_;
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
 
   ThreadUnsafeOneTimeEvent first_packet_received_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_