Implementing auto pausing of video streams.

webrtc/call/rampup_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 322 matching lines @@
 
 RampUpDownUpTester::~RampUpDownUpTester() {}
 
 bool RampUpDownUpTester::PollStats() {
   if (send_stream_) {
     webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
     int transmit_bitrate_bps = 0;
     for (auto it : stats.substreams) {
       transmit_bitrate_bps += it.second.total_bitrate_bps;
     }
-
     EvolveTestState(transmit_bitrate_bps, stats.suspended);
   }
 
   return !event_.Wait(kPollIntervalMs);
 }
 
 Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
   Call::Config config;
   config.bitrate_config.min_bitrate_bps = 10000;
   return config;
@@ skipping 103 matching lines @@
   RampUpTester test(3, 0, 0, RtpExtension::kTimestampOffsetUri, true, true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTimestampOffsetUri, false, false);
   RunBaseTest(&test);
 }
 
-// Disabled on Mac due to flakiness, see
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5407
-#ifndef WEBRTC_MAC
-
 static const uint32_t kStartBitrateBps = 60000;
 
 // Disabled: https://bugs.chromium.org/p/webrtc/issues/detail?id=5576
 TEST_F(RampUpTest, DISABLED_UpDownUpOneStream) {
   RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
                           false, false);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, UpDownUpThreeStreams) {
@@ skipping 37 matching lines @@
 
 // TODO(holmer): Enable when audio bitrates are included in the bitrate
 //               allocation.
 TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
   RampUpDownUpTester test(3, 1, kStartBitrateBps,
                           RtpExtension::kTransportSequenceNumberUri, true,
                           false);
   RunBaseTest(&test);
 }
 
-#endif
-
 TEST_F(RampUpTest, AbsSendTimeSingleStream) {
   RampUpTester test(1, 0, 0, RtpExtension::kAbsSendTimeUri, false, false);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, AbsSendTimeSimulcast) {
   RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTimeUri, false, false);
   RunBaseTest(&test);
 }
 
@@ skipping 42 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumberUri, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc