Add support for transport wide sequence numbers

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 53 matching lines @@
 };
 
 class RTPSender : public RTPSenderInterface {
  public:
   RTPSender(int32_t id,
             bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
             PacedSender* paced_sender,
+            PacketRouter* packet_router,
+            SendTimeObserver* send_time_observer,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer);
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
   uint32_t VideoBitrateSent() const;
@@ skipping 60 matching lines @@
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionTotalLength() const;
 
   uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const;
 
   uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
   uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
   uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const;
-  uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer) const;
+  uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer,
+                                                uint16_t sequence_number) const;
+
+  // Verifies that the specified extension is registered, and that it is
+  // present in rtp packet. If extension is not registered kNotRegistered is
+  // returned. If extension cannot be found in the rtp header, or if it is
+  // malformed, kError is returned. Otherwise *extension_offset is set to the
+  // offset of the extension from the beginning of the rtp packet and kOk is
+  // returned.
+  enum class ExtensionStatus {
+    kNotRegistered,
+    kOk,
+    kError,
+  };
+  ExtensionStatus VerifyExtension(RTPExtensionType extension_type,
+                                  uint8_t* rtp_packet,
+                                  size_t rtp_packet_length,
+                                  const RTPHeader& rtp_header,
+                                  size_t extension_length_bytes,
+                                  size_t* extension_offset) const
+      EXCLUSIVE_LOCKS_REQUIRED(send_critsect_.get());
 
   bool UpdateAudioLevel(uint8_t* rtp_packet,
                         size_t rtp_packet_length,
                         const RTPHeader& rtp_header,
                         bool is_voiced,
                         uint8_t dBov) const;
 
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
@@ skipping 153 matching lines @@
                                    size_t* position) const;
 
   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.
+  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_;
 
   int32_t id_;
 
   const bool audio_configured_;
   rtc::scoped_ptr<RTPSenderAudio> audio_;
   rtc::scoped_ptr<RTPSenderVideo> video_;
 
-  PacedSender *paced_sender_;
+  PacedSender* const paced_sender_;
+  PacketRouter* const packet_router_;
+  SendTimeObserver* const send_time_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_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
@@ skipping 51 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_