Refactor NACK bitrate allocation

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 <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
+#include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_annotations.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/playout_delay_oracle.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"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
@@ skipping 20 matching lines @@
   // This returns the expected header length taking into consideration
   // the optional RTP header extensions that may not be currently active.
   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 uint16_t ActualSendBitrateKbit() = 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,
@@ skipping 14 matching lines @@
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
             RtcEventLog* event_log,
             SendPacketObserver* send_packet_observer);
 
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
-  uint16_t ActualSendBitrateKbit() const override;
+  uint16_t ActualSendBitrateKbit() override;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
-  uint32_t NackOverheadRate() const;
+  uint32_t NackOverheadRate();
 
   void SetTargetBitrate(uint32_t bitrate);
   uint32_t GetTargetBitrate();
 
   // Includes size of RTP and FEC headers.
   size_t MaxDataPayloadLength() const override;
 
   int32_t RegisterPayload(const char* payload_name,
                           const int8_t payload_type,
                           const uint32_t frequency,
@@ skipping 103 matching lines @@
   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;
 
   int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
 
-  bool ProcessNACKBitRate(uint32_t now);
-
   // Feedback to decide when to stop sending playout delay.
   void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
 
   // RTX.
   void SetRtxStatus(int mode);
   int RtxStatus() const;
 
   uint32_t RtxSsrc() const;
   void SetRtxSsrc(uint32_t ssrc);
 
@@ skipping 66 matching lines @@
   size_t SendPadData(size_t bytes,
                      bool timestamp_provided,
                      uint32_t timestamp,
                      int64_t capture_time_ms,
                      int probe_cluster_id);
 
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
-  uint32_t BitrateSent() const;
+  uint32_t BitrateSent();
 
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
   bool ActivateCVORtpHeaderExtension() override;
 
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
   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 send_over_rtx,
                             bool is_retransmit,
                             int probe_cluster_id);
 
   // 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, int probe_cluster_id);
@@ skipping 44 matching lines @@
 
   bool AllocateTransportSequenceNumber(int* packet_id) 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;
 
-  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);
-
-   private:
-    // We assume that these observers are called on the same thread, which is
-    // true for RtpSender as they are called on the Process thread.
-    class BitrateObserver : public Bitrate::Observer {
-     public:
-      explicit BitrateObserver(const BitrateAggregator& aggregator);
-
-      // Implements Bitrate::Observer.
-      void BitrateUpdated(const BitrateStatistics& stats) override;
-      const BitrateStatistics& statistics() const;
-
-     private:
-      BitrateStatistics statistics_;
-      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_;
+  rtc::CriticalSection stats_critsect_;

[danilchap, 2016/06/23 12:46:12]

may be use already existent statistics_crit_  instead (and avoid locking two
critsect in UpdateRtpStats)

[sprang_webrtc, 2016/06/28 09:12:33]

Done.

+  BitrateStatisticsObserver* const bitrate_callback_;
+  RateStatistics total_bitrate_sent_ GUARDED_BY(stats_critsect_);
+  RateStatistics nack_bitrate_sent_ GUARDED_BY(stats_critsect_);
 
   const bool audio_configured_;
   const std::unique_ptr<RTPSenderAudio> audio_;
   const std::unique_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_;
   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_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_;
   int32_t transmission_time_offset_;
   uint32_t absolute_send_time_;
   VideoRotation rotation_;
   bool video_rotation_active_;
   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_;
-
   // Tracks the current request for playout delay limits from application
   // and decides whether the current RTP frame should include the playout
   // delay extension on header.
   PlayoutDelayOracle playout_delay_oracle_;
   bool playout_delay_active_ GUARDED_BY(send_critsect_);
 
   RTPPacketHistory packet_history_;
 
   // Statistics
   rtc::CriticalSection statistics_crit_;
@@ skipping 34 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_