RTCPeerConnection.addTrack and removeTrack added (behind flag)

chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc

Index: chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc
diff --git a/chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc b/chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc
index bd591a994707151759292c68c2c31ff09259b054..9b5045c04c8b77469cb1832c4b6a18277a5ea383 100644
--- a/chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc
+++ b/chrome/browser/media/webrtc/webrtc_rtp_browsertest.cc
@@ -2,6 +2,9 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include <string>
+#include <vector>
+
 #include "base/command_line.h"
 #include "chrome/browser/media/webrtc/webrtc_browsertest_base.h"
 #include "content/public/common/content_switches.h"
@@ -63,3 +66,106 @@ IN_PROC_BROWSER_TEST_F(WebRtcRtpBrowserTest, GetReceivers) {
   VerifyRtpReceivers(left_tab_, 2);
   VerifyRtpReceivers(right_tab_, 6);
 }
+
+IN_PROC_BROWSER_TEST_F(WebRtcRtpBrowserTest, AddAndRemoveTracksWithoutStream) {
+  StartServerAndOpenTabs();
+
+  SetupPeerconnectionWithoutLocalStream(left_tab_);
+  SetupPeerconnectionWithoutLocalStream(right_tab_);
+
+  std::vector<std::string> ids =
+      CreateAndAddAudioAndVideoTrack(left_tab_, StreamArgumentType::NO_STREAM);
+  std::string audio_stream_id = ids[0];
+  std::string audio_track_id = ids[1];
+  std::string video_stream_id = ids[2];
+  std::string video_track_id = ids[3];
+  EXPECT_NE("null", audio_stream_id);
+  EXPECT_NE("null", audio_track_id);
+  EXPECT_NE("null", video_stream_id);
+  EXPECT_NE("null", video_track_id);
+  EXPECT_NE(audio_stream_id, video_stream_id);
+
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 2);
+  VerifyRtpReceivers(right_tab_, 2);
+
+  RemoveTrack(left_tab_, audio_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 1);
+  VerifyRtpReceivers(right_tab_, 1);
+
+  RemoveTrack(left_tab_, video_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 0);
+  VerifyRtpReceivers(right_tab_, 0);
+}
+
+IN_PROC_BROWSER_TEST_F(WebRtcRtpBrowserTest,
+                       AddAndRemoveTracksWithSharedStream) {
+  StartServerAndOpenTabs();
+
+  SetupPeerconnectionWithoutLocalStream(left_tab_);
+  SetupPeerconnectionWithoutLocalStream(right_tab_);
+
+  std::vector<std::string> ids = CreateAndAddAudioAndVideoTrack(
+      left_tab_, StreamArgumentType::SHARED_STREAM);
+  std::string audio_stream_id = ids[0];
+  std::string audio_track_id = ids[1];
+  std::string video_stream_id = ids[2];
+  std::string video_track_id = ids[3];
+  EXPECT_NE("null", audio_stream_id);
+  EXPECT_EQ(audio_stream_id, video_stream_id);
+  EXPECT_NE("null", audio_track_id);
+  EXPECT_NE("null", video_track_id);
+
+  NegotiateCall(left_tab_, right_tab_);
+  // TODO(hbos): Here and in other "AddAndRemoveTracks" tests: when ontrack and
+  // ended events are supported, verify that these are fired on the remote side
+  // when tracks are added and removed. https://crbug.com/webrtc/7933
+  VerifyRtpSenders(left_tab_, 2);
+  VerifyRtpReceivers(right_tab_, 2);
+
+  RemoveTrack(left_tab_, audio_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 1);
+  VerifyRtpReceivers(right_tab_, 1);
+
+  RemoveTrack(left_tab_, video_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 0);
+  VerifyRtpReceivers(right_tab_, 0);
+}
+
+IN_PROC_BROWSER_TEST_F(WebRtcRtpBrowserTest,
+                       AddAndRemoveTracksWithIndividualStreams) {
+  StartServerAndOpenTabs();
+
+  SetupPeerconnectionWithoutLocalStream(left_tab_);
+  SetupPeerconnectionWithoutLocalStream(right_tab_);
+
+  std::vector<std::string> ids = CreateAndAddAudioAndVideoTrack(
+      left_tab_, StreamArgumentType::INDIVIDUAL_STREAMS);
+  std::string audio_stream_id = ids[0];
+  std::string audio_track_id = ids[1];
+  std::string video_stream_id = ids[2];
+  std::string video_track_id = ids[3];
+  EXPECT_NE("null", audio_stream_id);
+  EXPECT_NE("null", audio_track_id);
+  EXPECT_NE("null", video_stream_id);
+  EXPECT_NE("null", video_track_id);
+  EXPECT_NE(audio_stream_id, video_stream_id);
+
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 2);
+  VerifyRtpReceivers(right_tab_, 2);
+
+  RemoveTrack(left_tab_, audio_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 1);
+  VerifyRtpReceivers(right_tab_, 1);
+
+  RemoveTrack(left_tab_, video_track_id);
+  NegotiateCall(left_tab_, right_tab_);
+  VerifyRtpSenders(left_tab_, 0);
+  VerifyRtpReceivers(right_tab_, 0);
+}