Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc

 /*
  *  Copyright (c) 2016 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 "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace {
 
+using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
 using ::testing::make_tuple;
 
 const int8_t kPayloadType = 100;
 const uint32_t kSsrc = 0x12345678;
 const uint16_t kSeqNum = 88;
 const uint32_t kTimestamp = 0x65431278;
 
 }  // namespace
 
@@ skipping 111 matching lines @@
   RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
   RTPHeader header;
 
   EXPECT_TRUE(parser.Parse(&header, &extensions));
 
   // Parse should ignore extension.
   EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
   EXPECT_EQ(sizeof(kPacket), header.headerLength);
 }
 
-TEST(RtpHeaderParser, ParseAll6Extensions) {
+TEST(RtpHeaderParser, ParseAll8Extensions) {
   const uint8_t kAudioLevel = 0x5a;
   // clang-format off
   const uint8_t kPacket[] = {
       0x90, kPayloadType, 0x00, kSeqNum,
       0x65, 0x43, 0x12, 0x78,  // kTimestamp.
       0x12, 0x34, 0x56, 0x78,  // kSsrc.
-      0xbe, 0xde, 0x00, 0x05,  // Extension of size 5x32bit word.
+      0xbe, 0xde, 0x00, 0x08,  // Extension of size 8x32bit words.
       0x40, 0x80|kAudioLevel,  // AudioLevel.
       0x22, 0x01, 0x56, 0xce,  // TransmissionOffset.
       0x62, 0x12, 0x34, 0x56,  // AbsoluteSendTime.
       0x81, 0xce, 0xab,        // TransportSequenceNumber.
       0xa0, 0x03,              // VideoRotation.
       0xb2, 0x12, 0x48, 0x76,  // PlayoutDelayLimits.
-      0x00,                    // Padding to 32bit boundary.
+      0xc2, 'r', 't', 'x',     // RtpStreamId
+      0xd5, 's', 't', 'r', 'e', 'a', 'm',  // RepairedRtpStreamId
+      0x00, 0x00,              // Padding to 32bit boundary.
   };
   // clang-format on
   ASSERT_EQ(sizeof(kPacket) % 4, 0u);
 
   RtpHeaderExtensionMap extensions;
   extensions.Register<TransmissionOffset>(2);
   extensions.Register<AudioLevel>(4);
   extensions.Register<AbsoluteSendTime>(6);
   extensions.Register<TransportSequenceNumber>(8);
   extensions.Register<VideoOrientation>(0xa);
   extensions.Register<PlayoutDelayLimits>(0xb);
+  extensions.Register<RtpStreamId>(0xc);
+  extensions.Register<RepairedRtpStreamId>(0xd);
   RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
   RTPHeader header;
 
   EXPECT_TRUE(parser.Parse(&header, &extensions));
 
   EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
   EXPECT_EQ(0x156ce, header.extension.transmissionTimeOffset);
 
   EXPECT_TRUE(header.extension.hasAudioLevel);
   EXPECT_TRUE(header.extension.voiceActivity);
   EXPECT_EQ(kAudioLevel, header.extension.audioLevel);
 
   EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
   EXPECT_EQ(0x123456U, header.extension.absoluteSendTime);
 
   EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
   EXPECT_EQ(0xceab, header.extension.transportSequenceNumber);
 
   EXPECT_TRUE(header.extension.hasVideoRotation);
   EXPECT_EQ(kVideoRotation_270, header.extension.videoRotation);
 
   EXPECT_EQ(0x124 * PlayoutDelayLimits::kGranularityMs,
             header.extension.playout_delay.min_ms);
   EXPECT_EQ(0x876 * PlayoutDelayLimits::kGranularityMs,
             header.extension.playout_delay.max_ms);
+  EXPECT_EQ(header.extension.stream_id_size, 3U);
+  // Join buffer + size using ArrayView because ::testing::make_tuple calls
+  // strlen(const char* rsid), but rsid is not 0-terminated causing msan error.
+  EXPECT_THAT(rtc::MakeArrayView(header.extension.stream_id,
+                                 header.extension.stream_id_size),
+              ElementsAre('r', 't', 'x'));
+  EXPECT_THAT(rtc::MakeArrayView(header.extension.repaired_stream_id,
+                                 header.extension.repaired_stream_id_size),
+              ElementsAre('s', 't', 'r', 'e', 'a', 'm'));
 }
 
 TEST(RtpHeaderParser, ParseWithCsrcsExtensionAndPadding) {
   const uint8_t kPacketPaddingSize = 8;
   const uint32_t kCsrcs[] = {0x34567890, 0x32435465};
   const size_t kPayloadSize = 7;
   // clang-format off
   const uint8_t kPacket[] = {
     0xb2, kPayloadType, 0x00, kSeqNum,
     0x65, 0x43, 0x12, 0x78,  // kTimestamp.
@@ skipping 19 matching lines @@
   EXPECT_THAT(make_tuple(header.arrOfCSRCs, header.numCSRCs),
               ElementsAreArray(kCsrcs));
   EXPECT_EQ(kPacketPaddingSize, header.paddingLength);
   EXPECT_THAT(sizeof(kPacket) - kPayloadSize - kPacketPaddingSize,
               header.headerLength);
   EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
   EXPECT_EQ(0x56ce, header.extension.transmissionTimeOffset);
 }
 
 }  // namespace webrtc