Wire up send-side bandwidth estimation.

webrtc/video/video_quality_test.cc

Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index d440858155a9b4ebdc4b829454e1f5ec53392e54..cbd1e6065d6a54323817b776a72ae2dd23c903d5 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -32,6 +32,8 @@
 
 namespace webrtc {
 
+static const int kTransportSeqExtensionId =
+    VideoQualityTest::kAbsSendTimeExtensionId + 1;
 static const int kSendStatsPollingIntervalMs = 1000;
 static const int kPayloadTypeVP8 = 123;
 static const int kPayloadTypeVP9 = 124;
@@ -598,6 +600,15 @@ void VideoQualityTest::SetupFullStack(const Params& params,
   send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
   send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
 
+  send_config_.rtp.extensions.clear();
+  if (params.common.send_side_bwe) {
+    send_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumber, kTransportSeqExtensionId));
+  } else {
+    send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+  }
+
   // Automatically fill out streams[0] with params.
   VideoStream* stream = &encoder_config_.streams[0];
   stream->width = params.common.width;