Remove OnLocalSsrcChanged and rename EncoderStateFeedback.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

 /*
  *  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.
  */
 
@@ skipping 28 matching lines @@
                                const int32_t audioVideoOffset) {}
   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
   virtual void OnReceivedSLI(uint32_t ssrc,
                              uint8_t pictureId) {
     EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
   }
   virtual void OnReceivedRPSI(uint32_t ssrc,
                               uint64_t pictureId) {
     EXPECT_EQ(kTestPictureId, pictureId);
   }
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}
 
  private:
   RtpRtcp* _rtpRtcpModule;
 };
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
   explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
@@ skipping 208 matching lines @@
   EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
 
   EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
   EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
   EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
   EXPECT_EQ(0u, report_blocks[0].fractionLost);
 }
 
 }  // namespace
 }  // namespace webrtc