Refactor RtpSender and SSRCDatabase a bit.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index a672a06398a5772b9ef64914914a5946e8410536..b65e811133dd4603bf3187d93b3fd9daa7989756 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -16,6 +16,7 @@
 #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"
@@ -31,7 +32,6 @@
 namespace webrtc {
 
 class BitrateAggregator;
-class CriticalSectionWrapper;
 class RTPSenderAudio;
 class RTPSenderVideo;
 class RtcEventLog;
@@ -98,7 +98,8 @@ class RTPSender : public RTPSenderInterface {
             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();
@@ -196,7 +197,7 @@ class RTPSender : public RTPSenderInterface {
                                   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,
@@ -386,22 +387,24 @@ class RTPSender : public RTPSenderInterface {
                       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_);
@@ -440,7 +443,7 @@ class RTPSender : public RTPSenderInterface {
   // 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_);