Add a flags field to video timing extension.

webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index 1075d9ea352bbdd487eb808f328f56865e2d4a1a..21cf2445e9a1cb5c17cda24a183f24e73d819e60 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -32,6 +32,7 @@ constexpr uint8_t kTransmissionOffsetExtensionId = 1;
 constexpr uint8_t kAudioLevelExtensionId = 9;
 constexpr uint8_t kRtpStreamIdExtensionId = 0xa;
 constexpr uint8_t kRtpMidExtensionId = 0xb;
+constexpr uint8_t kVideoTimingExtensionId = 0xc;
 constexpr int32_t kTimeOffset = 0x56ce;
 constexpr bool kVoiceActive = true;
 constexpr uint8_t kAudioLevel = 0x5a;
@@ -498,4 +499,91 @@ TEST(RtpPacketTest, RawExtensionFunctionsAcceptZeroIdAndReturnFalse) {
   EXPECT_THAT(packet.AllocateRawExtension(kInvalidId, 3), IsEmpty());
 }
 
+TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) {
+  // Create a packet with video frame timing extension populated.
+  RtpPacketToSend::ExtensionManager send_extensions;
+  send_extensions.Register(kRtpExtensionVideoTiming, kVideoTimingExtensionId);
+  RtpPacketToSend send_packet(&send_extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  VideoSendTiming timing;
+  timing.encode_start_delta_ms = 1;
+  timing.packetization_finish_delta_ms = 5;
+  timing.flags =
+      TimingFrameFlags::kTriggeredByTimer + TimingFrameFlags::kTriggeredBySize;
+
+  send_packet.SetExtension<VideoTimingExtension>(timing);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<VideoTimingExtension>(kVideoTimingExtensionId);
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  VideoSendTiming receivied_timing;
+  EXPECT_TRUE(
+      receive_packet.GetExtension<VideoTimingExtension>(&receivied_timing));
+
+  // Only check first and last timestamp (covered by other tests) plus flags.
+  EXPECT_EQ(receivied_timing.encode_start_delta_ms,
+            timing.encode_start_delta_ms);
+  EXPECT_EQ(receivied_timing.packetization_finish_delta_ms,
+            timing.packetization_finish_delta_ms);
+  EXPECT_EQ(receivied_timing.flags, timing.flags);
+}
+
+TEST(RtpPacketTest, ParseLegacyTimingFrameExtension) {
+  // Create a packet with video frame timing extension populated.
+  RtpPacketToSend::ExtensionManager send_extensions;
+  send_extensions.Register(kRtpExtensionVideoTiming, kVideoTimingExtensionId);
+  RtpPacketToSend send_packet(&send_extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  VideoSendTiming timing;
+  timing.encode_start_delta_ms = 1;
+  timing.packetization_finish_delta_ms = 5;
+  timing.flags =
+      TimingFrameFlags::kTriggeredByTimer + TimingFrameFlags::kTriggeredBySize;
+
+  send_packet.SetExtension<VideoTimingExtension>(timing);
+
+  // Modify the sent packet so that the old VideoTimingExtension format is used.

[danilchap, 2017/08/11 13:51:16]

May be it is simpler and more robust to assemble such packet manually (specially
if you find it is better to put flags first)

[sprang_webrtc, 2017/08/11 15:06:34]

Done.

+  rtc::ArrayView<const uint8_t> raw_extension =
+      send_packet.GetRawExtension(kVideoTimingExtensionId);
+  rtc::CopyOnWriteBuffer buffer = send_packet.Buffer();
+  ptrdiff_t start_offset = raw_extension.cbegin() - buffer.cdata();
+
+  // Turn last byte of header extension into padding.
+  const uint8_t kExtensionLength = VideoTimingExtension::kValueSizeBytes;
+  buffer[start_offset + kExtensionLength - 1] = 0;
+  // Validate one-byte header and reduce length by one.
+  const uint8_t onebyte_header = buffer[start_offset - 1];
+  EXPECT_EQ(onebyte_header,
+            (kVideoTimingExtensionId << 4) | (kExtensionLength - 1));
+  buffer[send_packet.headers_size() - kExtensionLength] =
+      (kVideoTimingExtensionId << 4) | (kExtensionLength - 2);
+
+  // Parse the modified packet.
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<VideoTimingExtension>(kVideoTimingExtensionId);
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(buffer));
+  VideoSendTiming receivied_timing;
+  EXPECT_TRUE(
+      receive_packet.GetExtension<VideoTimingExtension>(&receivied_timing));
+
+  // Check first and last timestamp are still OK. Flags should now be 0.
+  EXPECT_EQ(receivied_timing.encode_start_delta_ms,
+            timing.encode_start_delta_ms);
+  EXPECT_EQ(receivied_timing.packetization_finish_delta_ms,
+            timing.packetization_finish_delta_ms);
+  EXPECT_EQ(receivied_timing.flags, 0);
+}
+
 }  // namespace webrtc