Fixing a segfault that can occur when changing the track of an RtpSender.

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 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.
  */
 
 #include <string>
 #include <utility>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/rtpreceiver.h"
 #include "webrtc/api/rtpsender.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/test/fakevideotracksource.h"
 #include "webrtc/api/videotracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/mediachannel.h"
 
 using ::testing::_;
 using ::testing::Exactly;
+using ::testing::InvokeWithoutArgs;
 using ::testing::Return;
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "video_1";
 static const char kAudioTrackId[] = "audio_1";
 static const uint32_t kVideoSsrc = 98;
 static const uint32_t kVideoSsrc2 = 100;
 static const uint32_t kAudioSsrc = 99;
 static const uint32_t kAudioSsrc2 = 101;
 
@@ skipping 368 matching lines @@
 }
 
 TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
   rtc::scoped_refptr<AudioTrackInterface> track =
       AudioTrack::Create(kAudioTrackId, nullptr);
   EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
   rtc::scoped_refptr<AudioRtpSender> sender =
       new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
   sender->SetSsrc(kAudioSsrc);
 
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+  // Expect that SetAudioSend will be called before the reference to the track
+  // is released.
+  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, nullptr))
+      .Times(1)
+      .WillOnce(InvokeWithoutArgs([&track] {
+        EXPECT_LT(2, track->AddRef());
+        track->Release();
+      }));
   EXPECT_TRUE(sender->SetTrack(nullptr));
 
   // Make sure it's SetTrack that called methods on the provider, and not the
   // destructor.
   EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
 }
 
 TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
-  AddVideoTrack();
-  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
+  rtc::scoped_refptr<VideoTrackSourceInterface> source(
+      FakeVideoTrackSource::Create());
+  rtc::scoped_refptr<VideoTrackInterface> track =
+      VideoTrack::Create(kVideoTrackId, source);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, track.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+      new VideoRtpSender(track, kStreamLabel1, &video_provider_);
   sender->SetSsrc(kVideoSsrc);
 
-  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr)).Times(1);
+  // Expect that SetSource will be called before the reference to the track
+  // is released.
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr))
+      .Times(1)
+      .WillOnce(InvokeWithoutArgs([&track] {
+        EXPECT_LT(2, track->AddRef());
+        track->Release();
+        return true;
+      }));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
   EXPECT_TRUE(sender->SetTrack(nullptr));
 
   // Make sure it's SetTrack that called methods on the provider, and not the
   // destructor.
   EXPECT_CALL(video_provider_, SetSource(_, _)).Times(0);
   EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
 }
 
 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
@@ skipping 52 matching lines @@
       .WillOnce(Return(RtpParameters()));
   EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   DestroyVideoRtpSender();
 }
 
 }  // namespace webrtc