Enable send-side BWE by default for video in call tests.

webrtc/video/video_send_stream_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.
  */
 #include <algorithm>  // max
@@ skipping 222 matching lines @@
       observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
-      send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(RtpExtension(
-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
     }
 
     test::FakeEncoder encoder_;
   } test;
 
   RunBaseTest(&test);
@@ skipping 197 matching lines @@
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms =
               VideoSendStreamTest::kNackRtpHistoryMs;
     }
     send_config->encoder_settings.encoder = encoder_;
     send_config->encoder_settings.payload_name = payload_name_;
     send_config->rtp.ulpfec.red_payload_type =
         VideoSendStreamTest::kRedPayloadType;
     send_config->rtp.ulpfec.ulpfec_payload_type =
         VideoSendStreamTest::kUlpfecPayloadType;
-    if (header_extensions_enabled_) {
+    EXPECT_FALSE(send_config->rtp.extensions.empty());
+    if (!header_extensions_enabled_) {
+      send_config->rtp.extensions.clear();
+    } else {
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
     }
     (*receive_configs)[0].rtp.ulpfec.red_payload_type =
         send_config->rtp.ulpfec.red_payload_type;
     (*receive_configs)[0].rtp.ulpfec.ulpfec_payload_type =
         send_config->rtp.ulpfec.ulpfec_payload_type;
   }
 
   void PerformTest() override {
     EXPECT_EQ(expect_ulpfec_, Wait())
         << "Timed out waiting for ULPFEC and/or media packets.";
@@ skipping 128 matching lines @@
           (*receive_configs)[0].rtp.nack.rtp_history_ms =
               VideoSendStreamTest::kNackRtpHistoryMs;
     }
     send_config->encoder_settings.encoder = encoder_;
     send_config->encoder_settings.payload_name = payload_name_;
     if (header_extensions_enabled_) {
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
+    } else {
+      send_config->rtp.extensions.clear();
     }
   }
 
   void PerformTest() override {
     EXPECT_TRUE(Wait())
         << "Timed out waiting for FlexFEC and/or media packets.";
   }
 
   VideoEncoder* const encoder_;
   std::string payload_name_;
@@ skipping 635 matching lines @@
       config.link_capacity_kbps = kCapacityKbps;
       config.queue_delay_ms = kNetworkDelayMs;
       return new test::PacketTransport(sender_call, this,
                                        test::PacketTransport::kSender, config);
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
-      send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
       // Turn on RTX.
       send_config->rtp.rtx.payload_type = kFakeVideoSendPayloadType;
       send_config->rtp.rtx.ssrcs.push_back(kVideoSendSsrcs[0]);
-
-      (*receive_configs)[0].rtp.extensions.clear();
-      (*receive_configs)[0].rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       test::kTransportSequenceNumberExtensionId));
-      (*receive_configs)[0].rtp.transport_cc = true;
     }
 
     void PerformTest() override {
       // TODO(isheriff): Some platforms do not ramp up as expected to full
       // capacity due to packet scheduling delays. Fix that before getting
       // rid of this.
       SleepMs(5000);
       {
         rtc::CritScope lock(&crit_);
         // Expect padding to be a small percentage of total bytes sent.
@@ skipping 1953 matching lines @@
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc