Revert "Move stats ID generation from SSRC to local ID"

This reverts commit e357a4dd4e3b015f8281813f246de793589bd537.

Reason for revert: Looks like it's breaking some downstream projects.

Original change's description:
> Move stats ID generation from SSRC to local ID
> 
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
> 
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
> 
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}

TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I621c10236c02be01d82f4660168f0323b85e24af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8673
Reviewed-on: https://webrtc-review.googlesource.com/38681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21586}
diff --git a/api/rtpreceiverinterface.h b/api/rtpreceiverinterface.h
index ac2e090..311971d 100644
--- a/api/rtpreceiverinterface.h
+++ b/api/rtpreceiverinterface.h
@@ -127,13 +127,6 @@
   virtual std::vector<RtpSource> GetSources() const {
     return std::vector<RtpSource>();
   }
-  // TODO(hta): Remove default implementation or move function to
-  // an internal interface. content::FakeRtpReceiver in Chromium needs this.
-
-  // Returns an ID that changes if the attached track changes, but
-  // otherwise remains constant. Used to generate IDs for stats.
-  // The special value zero means that no track is attached.
-  virtual int AttachmentId() const { return 0; }
 
  protected:
   virtual ~RtpReceiverInterface() {}
@@ -153,7 +146,6 @@
   PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
   PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
   PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
-  PROXY_CONSTMETHOD0(int, AttachmentId);
   END_PROXY_MAP()
 
 }  // namespace webrtc
diff --git a/api/rtpsenderinterface.h b/api/rtpsenderinterface.h
index d253e89..a7fbbda 100644
--- a/api/rtpsenderinterface.h
+++ b/api/rtpsenderinterface.h
@@ -59,11 +59,6 @@
   // Returns null for a video sender.
   virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
 
-  // Returns an ID that changes every time SetTrack() is called, but
-  // otherwise remains constant. Used to generate IDs for stats.
-  // The special value zero means that no track is attached.
-  virtual int AttachmentId() const = 0;
-
  protected:
   virtual ~RtpSenderInterface() {}
 };
@@ -82,8 +77,7 @@
   PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
   PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
   PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
-  PROXY_CONSTMETHOD0(int, AttachmentId);
-  END_PROXY_MAP()
+END_PROXY_MAP()
 
 }  // namespace webrtc
 
diff --git a/api/test/mock_rtpreceiver.h b/api/test/mock_rtpreceiver.h
index 53c04b9..7097adc 100644
--- a/api/test/mock_rtpreceiver.h
+++ b/api/test/mock_rtpreceiver.h
@@ -29,7 +29,6 @@
   MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
   MOCK_METHOD1(SetObserver, void(RtpReceiverObserverInterface*));
   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
-  MOCK_CONST_METHOD0(AttachmentId, int());
 };
 
 }  // namespace webrtc
diff --git a/api/test/mock_rtpsender.h b/api/test/mock_rtpsender.h
index 35d048c..a89fa92 100644
--- a/api/test/mock_rtpsender.h
+++ b/api/test/mock_rtpsender.h
@@ -30,7 +30,6 @@
   MOCK_CONST_METHOD0(GetParameters, RtpParameters());
   MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
   MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
-  MOCK_CONST_METHOD0(AttachmentId, int());
 };
 
 }  // namespace webrtc