Reland of Add functionality which limits the number of bytes on the network.

webrtc/video/end_to_end_tests.cc

Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 106b5137123100c11b18d6ae3bf066075349b562..b2558f559a3c56a7755c59d073910d91e7aefedb 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1831,9 +1831,9 @@
 
 class TransportFeedbackTester : public test::EndToEndTest {
  public:
-  explicit TransportFeedbackTester(bool feedback_enabled,
-                                   size_t num_video_streams,
-                                   size_t num_audio_streams)
+  TransportFeedbackTester(bool feedback_enabled,
+                          size_t num_video_streams,
+                          size_t num_audio_streams)
       : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
         feedback_enabled_(feedback_enabled),
         num_video_streams_(num_video_streams),
@@ -1925,6 +1925,80 @@
 
 TEST_F(EndToEndTest, AudioVideoReceivesTransportFeedback) {
   TransportFeedbackTester test(true, 1, 1);
+  RunBaseTest(&test);
+}
+
+TEST_F(EndToEndTest, StopsSendingMediaWithoutFeedback) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-CwndExperiment/Enabled/");
+
+  class TransportFeedbackTester : public test::EndToEndTest {
+   public:
+    TransportFeedbackTester(size_t num_video_streams, size_t num_audio_streams)
+        : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
+          num_video_streams_(num_video_streams),
+          num_audio_streams_(num_audio_streams),
+          media_sent_(0),
+          padding_sent_(0) {
+      // Only one stream of each supported for now.
+      EXPECT_LE(num_video_streams, 1u);
+      EXPECT_LE(num_audio_streams, 1u);
+    }
+
+   protected:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      const bool only_padding =
+          header.headerLength + header.paddingLength == length;
+      rtc::CritScope lock(&crit_);
+      if (only_padding) {
+        ++padding_sent_;
+      } else {
+        ++media_sent_;
+        EXPECT_LT(media_sent_, 40) << "Media sent without feedback.";
+      }
+
+      return SEND_PACKET;
+    }
+
+    Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
+      rtc::CritScope lock(&crit_);
+      if (media_sent_ > 20 && HasTransportFeedback(data, length)) {
+        return DROP_PACKET;
+      }
+      return SEND_PACKET;
+    }
+
+    bool HasTransportFeedback(const uint8_t* data, size_t length) const {
+      test::RtcpPacketParser parser;
+      EXPECT_TRUE(parser.Parse(data, length));
+      return parser.transport_feedback()->num_packets() > 0;
+    }
+
+    Call::Config GetSenderCallConfig() override {
+      Call::Config config = EndToEndTest::GetSenderCallConfig();
+      config.bitrate_config.max_bitrate_bps = 300000;
+      return config;
+    }
+
+    void PerformTest() override {
+      const int64_t kDisabledFeedbackTimeoutMs = 10000;
+      observation_complete_.Wait(kDisabledFeedbackTimeoutMs);
+      rtc::CritScope lock(&crit_);
+      EXPECT_GT(padding_sent_, 0);
+    }
+
+    size_t GetNumVideoStreams() const override { return num_video_streams_; }
+    size_t GetNumAudioStreams() const override { return num_audio_streams_; }
+
+   private:
+    const size_t num_video_streams_;
+    const size_t num_audio_streams_;
+    rtc::CriticalSection crit_;
+    int media_sent_ GUARDED_BY(crit_);
+    int padding_sent_ GUARDED_BY(crit_);
+  } test(1, 0);
   RunBaseTest(&test);
 }
 
@@ -2408,8 +2482,8 @@
     if (success)
       return;
   }
-  RTC_DCHECK(success) << "Failed to perform mid call probing (" << kMaxAttempts
-                      << " attempts).";
+  EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
+                       << " attempts).";
 }
 
 TEST_F(EndToEndTest, VerifyNackStats) {
@@ -4194,12 +4268,17 @@
           receiver_call_(nullptr),
           sender_state_(kNetworkUp),
           sender_rtp_(0),
+          sender_padding_(0),
           sender_rtcp_(0),
           receiver_rtcp_(0),
           down_frames_(0) {}
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&test_crit_);
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      if (length == header.headerLength + header.paddingLength)
+        ++sender_padding_;
       ++sender_rtp_;
       packet_event_.Set();
       return SEND_PACKET;
@@ -4324,7 +4403,8 @@
         int64_t time_now_ms = clock_->TimeInMilliseconds();
         rtc::CritScope lock(&test_crit_);
         if (sender_down) {
-          ASSERT_LE(sender_rtp_ - initial_sender_rtp, kNumAcceptedDowntimeRtp)
+          ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
+                    kNumAcceptedDowntimeRtp)
               << "RTP sent during sender-side downtime.";
           ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
                     kNumAcceptedDowntimeRtcp)
@@ -4359,6 +4439,7 @@
     Call* receiver_call_;
     NetworkState sender_state_ GUARDED_BY(test_crit_);
     int sender_rtp_ GUARDED_BY(test_crit_);
+    int sender_padding_ GUARDED_BY(test_crit_);
     int sender_rtcp_ GUARDED_BY(test_crit_);
     int receiver_rtcp_ GUARDED_BY(test_crit_);
     int down_frames_ GUARDED_BY(test_crit_);