[Draft] RtpPacket sketched.

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 <list>
 #include <map>
+#include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/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_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions_manager.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.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"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
 class RTPSenderAudio;
 class RTPSenderVideo;
@@ skipping 26 matching lines @@
   virtual size_t RTPHeaderLength() const = 0;
   // 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 ActualSendBitrateKbit() const = 0;
 
-  virtual int32_t SendToNetwork(uint8_t* data_buffer,
+  virtual int32_t SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
                                 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;
@@ skipping 74 matching lines @@
   // RTP header extension
   int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
   int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
   void SetVideoRotation(VideoRotation rotation);
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
+  template <typename Extension>
+  bool RegisterRtpHeaderExtension(uint8_t id) {
+    return rtp_manager_.Register<Extension>(id);
+  }
+  bool RegisterRtpHeaderExtension(const std::string& name,
+                                  MediaType media,
+                                  uint8_t id) {
+    return rtp_manager_.RegisterByName(name, media, id);
+  }
+
   size_t RtpHeaderExtensionTotalLength() const;
+  std::unique_ptr<RtpPacketToSend> CreatePacket() const;
+  void ReserveExtensions(RtpPacketToSend* packet) 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,
                                                 uint16_t sequence_number) const;
 
@@ skipping 30 matching lines @@
   bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
                         bool retransmission);
   size_t TimeToSendPadding(size_t bytes);
 
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
   void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
 
-  void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
-
-  bool StorePackets() const;
+  void SetStorePacketsSize(uint16_t number_to_store);
 
   int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
 
   bool ProcessNACKBitRate(uint32_t now);
 
   // RTX.
   void SetRtxStatus(int mode);
   int RtxStatus() const;
 
   uint32_t RtxSsrc() const;
   void SetRtxSsrc(uint32_t ssrc);
 
   void SetRtxPayloadType(int payload_type, int associated_payload_type);
 
   // Functions wrapping RTPSenderInterface.
   int32_t BuildRTPheader(uint8_t* data_buffer,
                          int8_t payload_type,
                          bool marker_bit,
                          uint32_t capture_timestamp,
                          int64_t capture_time_ms,
                          const bool timestamp_provided = true,
                          const bool inc_sequence_number = true) override;
+  void BuildRtpHeader(RtpPacketToSend* packet,
+                      uint32_t capture_timestamp,
+                      bool reserve_extensions);
 
   size_t RTPHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
-  int32_t SendToNetwork(uint8_t* data_buffer,
+  int32_t SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
                         size_t payload_length,
-                        size_t rtp_header_length,
                         int64_t capture_time_ms,
                         StorageType storage,
                         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
@@ skipping 55 matching lines @@
   size_t CreateRtpHeader(uint8_t* header,
                          int8_t payload_type,
                          uint32_t ssrc,
                          bool marker_bit,
                          uint32_t timestamp,
                          uint16_t sequence_number,
                          const std::vector<uint32_t>& csrcs) const;
 
   void UpdateNACKBitRate(uint32_t bytes, int64_t now);
 
-  bool PrepareAndSendPacket(uint8_t* buffer,
-                            size_t length,
-                            int64_t capture_time_ms,
+  bool PrepareAndSendPacket(RtpPacketToSend* packet,
                             bool send_over_rtx,
                             bool is_retransmit);
 
   // Return the number of bytes sent.  Note that both of these functions may
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes);
 
   void BuildPaddingPacket(uint8_t* packet,
                           size_t header_length,
                           size_t padding_length);
 
-  void BuildRtxPacket(uint8_t* buffer, size_t* length,
-                      uint8_t* buffer_rtx);
+  std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
+      const RtpPacketToSend& packet);
 
-  bool SendPacketToNetwork(const uint8_t* packet,
-                           size_t size,
-                           const PacketOptions& options);
+  bool SendPacketToNetwork(RtpPacketToSend* packet);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
   bool FindHeaderExtensionPosition(RTPExtensionType type,
                                    const uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    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 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;
+  uint16_t UpdateTransportSequenceNumber(RtpPacketToSend* rtp_packet) const;
 
-  void UpdateRtpStats(const uint8_t* buffer,
-                      size_t packet_length,
-                      const RTPHeader& header,
+  void UpdateRtpStats(const RtpPacketToSend& packet,
                       bool is_rtx,
                       bool is_retransmit);
-  bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
+  bool IsFecPacket(const RtpPacketToSend& packet) const;
 
   class BitrateAggregator {
    public:
     explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback);
 
     void OnStatsUpdated() const;
 
     Bitrate::Observer* total_bitrate_observer();
     Bitrate::Observer* retransmit_bitrate_observer();
     void set_ssrc(uint32_t ssrc);
@@ skipping 14 matching lines @@
       const BitrateAggregator& aggregator_;
     };
 
     BitrateStatisticsObserver* const callback_;
     BitrateObserver total_bitrate_observer_;
     BitrateObserver retransmit_bitrate_observer_;
     uint32_t ssrc_;
   };
 
   Clock* const clock_;
-  const int64_t clock_delta_ms_;
   Random random_ GUARDED_BY(send_critsect_);
 
   BitrateAggregator bitrates_;
   Bitrate total_bitrate_sent_;
 
   const bool audio_configured_;
   const rtc::scoped_ptr<RTPSenderAudio> audio_;
   const rtc::scoped_ptr<RTPSenderVideo> video_;
 
   RtpPacketSender* const paced_sender_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::CriticalSection send_critsect_;
 
-  Transport *transport_;
+  Transport* transport_;
   bool sending_media_ GUARDED_BY(send_critsect_);
 
   size_t max_payload_length_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
+  // TODO(danilchap): rtp_header_extension_map_ and rtp_manager_ do same thing
+  // Remove older rtp_header_extension_map_ when all RtpPacket code switch to
+  // use rtp_manager_
   RtpHeaderExtensionMap rtp_header_extension_map_;
+  RtpHeaderExtensionsManager rtp_manager_;
+  rtc::ThreadChecker configuration_thread_;
+  rtc::ThreadChecker encoder_thread_;
+  rtc::ThreadChecker network_thread_;
   int32_t transmission_time_offset_;
   uint32_t absolute_send_time_;
   VideoRotation rotation_;
   CVOMode cvo_mode_;
   uint16_t transport_sequence_number_;
 
   // NACK
   uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
   size_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
   Bitrate nack_bitrate_;
@@ skipping 38 matching lines @@
   // that the target bitrate is still valid.
   rtc::CriticalSection target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_