From d9e62f58371d1bcafc94edfa1eed836d90627bfe Mon Sep 17 00:00:00 2001
From: danilchap <danilchap@webrtc.org>
Date: Thu, 14 Jan 2016 14:55:19 -0800
Subject: [PATCH] Fixed sending Rtp packets with non zero csrcs and certain
 extensions. Added test that fails because of given issue.

BUG=webrtc:5413
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1586523003

Cr-Commit-Position: refs/heads/master@{#11258}
---
 webrtc/modules/rtp_rtcp/source/rtp_sender.cc  | 17 +++----
 .../rtp_rtcp/source/rtp_sender_unittest.cc    | 44 ++++++++++++++++++-
 2 files changed, 47 insertions(+), 14 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index f4933afdd9..19d53c5459 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1456,8 +1456,9 @@ bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
 
   HeaderExtension header_extension(type);
 
-  size_t block_pos =
-      kRtpHeaderLength + rtp_header.numCSRCs + extension_block_pos;
+  size_t extension_pos =
+      kRtpHeaderLength + rtp_header.numCSRCs * sizeof(uint32_t);
+  size_t block_pos = extension_pos + extension_block_pos;
   if (rtp_packet_length < block_pos + header_extension.length ||
       rtp_header.headerLength < block_pos + header_extension.length) {
     LOG(LS_WARNING) << "Failed to find extension position for " << type
@@ -1466,8 +1467,8 @@ bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
   }
 
   // Verify that header contains extension.
-  if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
-        (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
+  if (!(rtp_packet[extension_pos] == 0xBE &&
+        rtp_packet[extension_pos + 1] == 0xDE)) {
     LOG(LS_WARNING) << "Failed to find extension position for " << type
                     << "as hdr extension not found.";
     return false;
@@ -1494,14 +1495,6 @@ RTPSender::ExtensionStatus RTPSender::VerifyExtension(
                                    rtp_packet_length, rtp_header, &block_pos))
     return ExtensionStatus::kError;
 
-  // Verify that header contains extension.
-  if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
-        (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
-    LOG(LS_WARNING)
-        << "Failed to update absolute send time, hdr extension not found.";
-    return ExtensionStatus::kError;
-  }
-
   // Verify first byte in block.
   const uint8_t first_block_byte = (id << 4) + (extension_length_bytes - 2);
   if (rtp_packet[block_pos] != first_block_byte)
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index c0703244af..4e652d7b95 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -146,16 +146,22 @@ class RtpSenderTest : public ::testing::Test {
   uint8_t packet_[kMaxPacketLength];
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
-    VerifyRTPHeaderCommon(rtp_header, kMarkerBit);
+    VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
   }
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) {
+    VerifyRTPHeaderCommon(rtp_header, marker_bit, 0);
+  }
+
+  void VerifyRTPHeaderCommon(const RTPHeader& rtp_header,
+                             bool marker_bit,
+                             uint8_t number_of_csrcs) {
     EXPECT_EQ(marker_bit, rtp_header.markerBit);
     EXPECT_EQ(payload_, rtp_header.payloadType);
     EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
     EXPECT_EQ(kTimestamp, rtp_header.timestamp);
     EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
-    EXPECT_EQ(0, rtp_header.numCSRCs);
+    EXPECT_EQ(number_of_csrcs, rtp_header.numCSRCs);
     EXPECT_EQ(0U, rtp_header.paddingLength);
   }
 
@@ -550,6 +556,40 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
 }
 
+TEST_F(RtpSenderTestWithoutPacer,
+       BuildRTPPacketWithCSRCAndAudioLevelExtension) {
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
+  std::vector<uint32_t> csrcs;
+  csrcs.push_back(0x23456789);
+  rtp_sender_->SetCsrcs(csrcs);
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+      packet_, kPayload, kMarkerBit, kTimestamp, 0));
+
+  // Verify
+  webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
+  webrtc::RTPHeader rtp_header;
+
+  // Updating audio level is done in RTPSenderAudio, so simulate it here.
+  rtp_parser.Parse(&rtp_header);
+  EXPECT_TRUE(rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true,
+                                            kAudioLevel));
+
+  RtpHeaderExtensionMap map;
+  map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
+
+  ASSERT_TRUE(valid_rtp_header);
+  ASSERT_FALSE(rtp_parser.RTCP());
+  VerifyRTPHeaderCommon(rtp_header, kMarkerBit, csrcs.size());
+  EXPECT_EQ(length, rtp_header.headerLength);
+  EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
+  EXPECT_EQ(1u, rtp_header.numCSRCs);
+  EXPECT_EQ(csrcs[0], rtp_header.arrOfCSRCs[0]);
+}
+
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));