Avoid circular dependency rtp_rtcp <-> paced_sender

webrtc/modules/rtp_rtcp/source/rtp_sender.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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #include <assert.h>
 #include <math.h>
 
 #include <map>
 
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 
 #define MAX_INIT_RTP_SEQ_NUMBER 32767  // 2^15 -1.
 
@@ skipping 32 matching lines @@
   // Returns the next sequence number to use for a packet and allocates
   // 'packets_to_send' number of sequence numbers. It's important all allocated
   // sequence numbers are used in sequence to avoid perceived packet loss.
   virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
   virtual uint16_t PacketOverHead() const = 0;
   virtual uint16_t ActualSendBitrateKbit() const = 0;
 
-  virtual int32_t SendToNetwork(
-      uint8_t *data_buffer, size_t payload_length, size_t rtp_header_length,
-      int64_t capture_time_ms, StorageType storage,
-      PacedSender::Priority priority) = 0;
+  virtual int32_t SendToNetwork(uint8_t* data_buffer,
+                                size_t payload_length,
+                                size_t rtp_header_length,
+                                int64_t capture_time_ms,
+                                StorageType storage,
+                                RtpPacketSender::Priority priority) = 0;
 
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    VideoRotation rotation) const = 0;
   virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0;
   virtual CVOMode ActivateCVORtpHeaderExtension() = 0;
 };
 
 class RTPSender : public RTPSenderInterface {
  public:
   RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
-            PacedSender* paced_sender,
-            PacketRouter* packet_router,
+            RtpPacketSender* paced_sender,
+            SequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer);
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
@@ skipping 145 matching lines @@
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
   int32_t SendToNetwork(uint8_t* data_buffer,
                         size_t payload_length,
                         size_t rtp_header_length,
                         int64_t capture_time_ms,
                         StorageType storage,
-                        PacedSender::Priority priority) override;
+                        RtpPacketSender::Priority priority) override;
 
   // Audio.
 
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   // Set audio packet size, used to determine when it's time to send a DTMF
   // packet in silence (CNG).
   int32_t SetAudioPacketSize(uint16_t packet_size_samples);
 
@@ skipping 92 matching lines @@
 
   void UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
                                     size_t rtp_packet_length,
                                     const RTPHeader& rtp_header,
                                     int64_t time_diff_ms) const;
   void UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                               size_t rtp_packet_length,
                               const RTPHeader& rtp_header,
                               int64_t now_ms) const;
   // Update the transport sequence number of the packet using a new sequence
-  // number allocated by PacketRouter. Returns the assigned sequence number,
-  // or 0 if extension could not be updated.
+  // number allocated by SequenceNumberAllocator. Returns the assigned sequence
+  // number, or 0 if extension could not be updated.
   uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet,
                                          size_t rtp_packet_length,
                                          const RTPHeader& rtp_header) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
                       size_t packet_length,
                       const RTPHeader& header,
                       bool is_rtx,
                       bool is_retransmit);
   bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
 
   Clock* clock_;
   int64_t clock_delta_ms_;
 
   rtc::scoped_ptr<BitrateAggregator> bitrates_;
   Bitrate total_bitrate_sent_;
 
   const bool audio_configured_;
   rtc::scoped_ptr<RTPSenderAudio> audio_;
   rtc::scoped_ptr<RTPSenderVideo> video_;
 
-  PacedSender* const paced_sender_;
-  PacketRouter* const packet_router_;
+  RtpPacketSender* const paced_sender_;
+  SequenceNumberAllocator* const sequence_number_allocator_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_;
 
   Transport *transport_;
   bool sending_media_ GUARDED_BY(send_critsect_);
 
   size_t max_payload_length_;
   uint16_t packet_over_head_;
 
@@ skipping 52 matching lines @@
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_