From 5c58f63d3fbce3f894a583a438c164b00c0b15dc Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Thu, 23 May 2013 13:36:55 +0000
Subject: [PATCH] Fix regression where retransmission bitrate is no longer
 estimated.

BUG=1813
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1530004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 webrtc/modules/rtp_rtcp/source/rtp_sender.cc    |  4 ++--
 .../rtp_rtcp/source/rtp_sender_unittest.cc      |  2 +-
 .../auto_test/source/vie_autotest_rtp_rtcp.cc   | 17 ++++++++++-------
 3 files changed, 13 insertions(+), 10 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index ed2412e18..a0e27a8bd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -519,12 +519,12 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
                                    length)) {
       // We can't send the packet right now.
       // We will be called when it is time.
-      return 0;
+      return length;
     }
   }
 
   if (SendPacketToNetwork(buffer_to_send_ptr, length)) {
-    return 0;
+    return length;
   }
   return -1;
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 9ee59e229..17416ef7c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -413,7 +413,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
-  EXPECT_EQ(0, rtp_sender_->ReSendPacket(kSeqNum));
+  EXPECT_EQ(rtp_length, rtp_sender_->ReSendPacket(kSeqNum));
   EXPECT_EQ(0, transport_.packets_sent_);
 
   rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms);
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index 03d5b5eb9..f9fe5dcf1 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -115,6 +115,10 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     ViETest::Log("Set start sequence number: %u", startSequenceNumber);
     EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
         tbChannel.videoChannel, startSequenceNumber));
+    const unsigned int kVideoSsrc = 123456;
+    // Set an SSRC to avoid issues with collisions.
+    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, kVideoSsrc,
+                                            webrtc::kViEStreamTypeNormal, 0));
 
     myTransport.EnableSequenceNumberCheck();
 
@@ -263,7 +267,7 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
 
     if (FLAGS_include_timing_dependent_tests) {
       EXPECT_GT(sentTotalBitrate, 0u);
-      EXPECT_GE(sentFecBitrate, 10u);
+      EXPECT_GT(sentFecBitrate, 0u);
       EXPECT_EQ(sentNackBitrate, 0u);
     }
 
@@ -279,12 +283,11 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
         tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
         sentFecBitrate, sentNackBitrate));
 
-    // TODO(holmer): Write a non-flaky verification of this API.
-    // numberOfErrors += ViETest::TestError(sentTotalBitrate > 0 &&
-    //                                      sentFecBitrate == 0 &&
-    //                                      sentNackBitrate > 0,
-    //                                      "ERROR: %s at line %d",
-    //                                      __FUNCTION__, __LINE__);
+    if (FLAGS_include_timing_dependent_tests) {
+      EXPECT_GT(sentTotalBitrate, 0u);
+      EXPECT_EQ(sentFecBitrate, 0u);
+      EXPECT_GT(sentNackBitrate, 0u);
+    }
 
     EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
     EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));