RTCPeerConnection.addTrack and removeTrack added (behind flag)

chrome/browser/media/webrtc/webrtc_browsertest_base.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CHROME_BROWSER_MEDIA_WEBRTC_WEBRTC_BROWSERTEST_BASE_H_
 #define CHROME_BROWSER_MEDIA_WEBRTC_WEBRTC_BROWSERTEST_BASE_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 29 matching lines @@
   static const char kAudioVideoCallConstraints720p[];
 
   static const char kOkGotStream[];
   static const char kFailedWithPermissionDeniedError[];
   static const char kFailedWithPermissionDismissedError[];
 
   static const char kUseDefaultCertKeygen[];
   static const char kUseDefaultAudioCodec[];
   static const char kUseDefaultVideoCodec[];
 
+  static const char kUndefined[];
+
+  enum class StreamArgumentType {
+    NO_STREAM,
+    SHARED_STREAM,
+    INDIVIDUAL_STREAMS
+  };
+
  protected:
   WebRtcTestBase();
   ~WebRtcTestBase() override;
 
   // These all require that the loaded page fulfills the public interface in
   // chrome/test/data/webrtc/getusermedia.js.
   // If an error is reported back from the getUserMedia call, these functions
   // will return false.
   // The ...AndAccept()/...AndDeny()/...AndDismiss() functions expect that a
   // prompt will be shown (i.e. the current origin in the tab_contents doesn't
@@ skipping 131 matching lines @@
   // Add 'usedtx=1' to the offer SDP.
   void EnableOpusDtx(content::WebContents* tab) const;
 
   void CreateAndAddStreams(content::WebContents* tab, size_t count) const;
   void VerifyRtpSenders(content::WebContents* tab,
                         base::Optional<size_t> expected_num_tracks =
                             base::Optional<size_t>()) const;
   void VerifyRtpReceivers(content::WebContents* tab,
                           base::Optional<size_t> expected_num_tracks =
                               base::Optional<size_t>()) const;
+  std::vector<std::string> CreateAndAddAudioAndVideoTrack(
+      content::WebContents* tab,
+      StreamArgumentType stream_argument_type) const;
+  void RemoveTrack(content::WebContents* tab,
+                   const std::string& track_id) const;
+  bool HasLocalStreamWithTrack(content::WebContents* tab,
+                               const std::string& stream_id,
+                               const std::string& track_id) const;
+  bool HasRemoteStreamWithTrack(content::WebContents* tab,
+                                const std::string& stream_id,
+                                const std::string& track_id) const;
+  bool HasSenderWithTrack(content::WebContents* tab,
+                          std::string track_id) const;
+  bool HasReceiverWithTrack(content::WebContents* tab,
+                            std::string track_id) const;
+  size_t GetNegotiationNeededCount(content::WebContents* tab) const;
 
  private:
   void CloseInfoBarInTab(content::WebContents* tab_contents,
                          infobars::InfoBar* infobar) const;
 
   std::string CreateLocalOffer(content::WebContents* from_tab) const;
   std::string CreateAnswer(std::string local_offer,
                            content::WebContents* to_tab) const;
   void ReceiveAnswer(const std::string& answer,
                      content::WebContents* from_tab) const;
   void GatherAndSendIceCandidates(content::WebContents* from_tab,
                                   content::WebContents* to_tab) const;
+  bool HasStreamWithTrack(content::WebContents* tab,
+                          const char* function_name,
+                          std::string stream_id,
+                          std::string track_id) const;
 
   infobars::InfoBar* GetUserMediaAndWaitForInfoBar(
       content::WebContents* tab_contents,
       const std::string& constraints) const;
 
   bool detect_errors_in_javascript_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcTestBase);
 };
 
 #endif  // CHROME_BROWSER_MEDIA_WEBRTC_WEBRTC_BROWSERTEST_BASE_H_