Refactor RtpSender and SSRCDatabase a bit.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <list>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 namespace webrtc {
 
 class ModuleRtpRtcpImpl : public RtpRtcp {
  public:
   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
 
   // Returns the number of milliseconds until the module want a worker thread to
   // call Process.
@@ skipping 295 matching lines @@
   void OnRequestSendReport();
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
   uint32_t BitrateReceivedNow() const;
 
   // Get remote SequenceNumber.
   uint16_t RemoteSequenceNumber() const;
 
+  SSRCDatabase ssrc_database_;

[mflodman, 2016/01/25 07:26:19]

Drive-by and Stefan might already have commented on this, but:
- The intention with the database is to ensure no ssrc is reused across all
sending and receiving streams.
- The new video API already checks this when creating send and receive streams,
so there is no real use case for the database in the video engine.
- VoiceEngine might still depend on this being checked across streams though, so
this change will then break currently use VoiceEngine functionality.

+
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* clock_;
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
@@ skipping 31 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_