Reland of Adding the ability to create an RtpSender without a track.

(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )

Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

Review URL: https://codereview.webrtc.org/1468113002

Cr-Commit-Position: refs/heads/master@{#10790}

talk/app/webrtc/peerconnectioninterface.h

@@ -334,6 +334,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
       AudioTrackInterface* track) = 0;
 
   // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
+  // |kind| must be "audio" or "video".
+  virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+      const std::string& kind) {
+    return rtc::scoped_refptr<RtpSenderInterface>();
+  }
+
   virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
       const {
     return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();