Refactor RtpSender and SSRCDatabase a bit.

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 <utility>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.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/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 {
 
 class BitrateAggregator;
-class CriticalSectionWrapper;
 class RTPSenderAudio;
 class RTPSenderVideo;
 class RtcEventLog;
 
 class RTPSenderInterface {
  public:
   RTPSenderInterface() {}
   virtual ~RTPSenderInterface() {}
 
   enum CVOMode {
@@ skipping 46 matching lines @@
   RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
             RtpPacketSender* paced_sender,
             TransportSequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log);
+            RtcEventLog* event_log,
+            SSRCDatabase* ssrc_database);
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
@@ skipping 77 matching lines @@
     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());
+      EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
 
   bool UpdateAudioLevel(uint8_t* rtp_packet,
                         size_t rtp_packet_length,
                         const RTPHeader& rtp_header,
                         bool is_voiced,
                         uint8_t dBov) const;
 
   bool UpdateVideoRotation(uint8_t* rtp_packet,
                            size_t rtp_packet_length,
                            const RTPHeader& rtp_header,
@@ skipping 169 matching lines @@
                                          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_;
+  Clock* const clock_;
+  const int64_t clock_delta_ms_;
   Random random_ GUARDED_BY(send_critsect_);
 
-  rtc::scoped_ptr<BitrateAggregator> bitrates_;
+  // TODO(tommi): BitrateAggregator doesn't need to be allocated this way.
+  // it could be a regular member variable.

[stefan-webrtc, 2016/01/23 17:38:40]

I guess the same is true for RTPSenderAudio/Video below too?

[tommi, 2016/01/24 11:51:49]

Moved the BitrateAggregator class declaration.

For RTPSenderAudio/Video we need to keep it as is do to the |bool audio|
variable passed to the ctor - or can it be done some other way?

[stefan-webrtc, 2016/01/25 10:09:32]

Ah, ok. Let's leave it as is for now then.

+  const rtc::scoped_ptr<BitrateAggregator> bitrates_;
   Bitrate total_bitrate_sent_;
 
   const bool audio_configured_;
-  rtc::scoped_ptr<RTPSenderAudio> audio_;
-  rtc::scoped_ptr<RTPSenderVideo> video_;
+  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::scoped_ptr<CriticalSectionWrapper> send_critsect_;
+  rtc::CriticalSection 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_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
@@ skipping 18 matching lines @@
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
   RtcEventLog* const event_log_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
   uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
-  SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_);
+  SSRCDatabase* const ssrc_db_;
   uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
   bool sequence_number_forced_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
   bool ssrc_forced_ GUARDED_BY(send_critsect_);
   uint32_t ssrc_ GUARDED_BY(send_critsect_);
   uint32_t timestamp_ GUARDED_BY(send_critsect_);
   int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
   int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
   bool media_has_been_sent_ GUARDED_BY(send_critsect_);
@@ skipping 13 matching lines @@
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> 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_