Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.

webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
index b0531c3c51a820ec7fd361b7653b06f1df00b586..14be2dd0a214c8c37038613b060a9a67a85339e2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
@@ -255,4 +255,82 @@ TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
             csrc_sources.begin()->timestamp_ms());
 }
 
+// The audio level from the RTPHeader extension should be stored in the
+// RTPSource with the matching SSRC.

[Taylor Brandstetter, 2017/08/24 20:40:57]

nit: RtpSource

[Zach Stein, 2017/08/24 21:14:53]

Done.

+TEST_F(RtpReceiverTest, GetSourcesContainsAudioLevelExtension) {
+  RTPHeader header;
+  int64_t time1_ms = fake_clock_.TimeInMilliseconds();
+  header.payloadType = kPcmuPayloadType;
+  header.ssrc = kSsrc1;
+  header.timestamp = rtp_timestamp(time1_ms);
+  header.extension.hasAudioLevel = true;
+  header.extension.audioLevel = 10;
+  PayloadUnion payload_specific = {AudioPayload()};
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+      header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+  auto sources = rtp_receiver_->GetSources();
+  EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
+                           time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
+

[Taylor Brandstetter, 2017/08/24 20:40:57]

May help to have some comments for the different sections of this test. For
example: "Receive a packet from a different SSRC with a different audio level,
and ensure that both are remembered"

[Zach Stein, 2017/08/24 21:14:53]

Done.

+  fake_clock_.AdvanceTimeMilliseconds(1);
+  int64_t time2_ms = fake_clock_.TimeInMilliseconds();
+  header.ssrc = kSsrc2;
+  header.timestamp = rtp_timestamp(time2_ms);
+  header.extension.hasAudioLevel = true;
+  header.extension.audioLevel = 20;
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+      header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+  sources = rtp_receiver_->GetSources();
+  EXPECT_THAT(sources,
+              UnorderedElementsAre(
+                  RtpSource(time1_ms, kSsrc1, RtpSourceType::SSRC, 10),
+                  RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  int64_t time3_ms = fake_clock_.TimeInMilliseconds();
+  header.ssrc = kSsrc1;
+  header.timestamp = rtp_timestamp(time3_ms);
+  header.extension.hasAudioLevel = true;
+  header.extension.audioLevel = 30;
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+      header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+  sources = rtp_receiver_->GetSources();
+  EXPECT_THAT(sources,
+              UnorderedElementsAre(
+                  RtpSource(time3_ms, kSsrc1, RtpSourceType::SSRC, 30),
+                  RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
+}
+
+TEST_F(RtpReceiverTest,
+       MissingAudioLevelHeaderExtensionClearsRtpSourceAudioLevel) {
+  RTPHeader header;
+  int64_t time1_ms = fake_clock_.TimeInMilliseconds();
+  header.payloadType = kPcmuPayloadType;
+  header.ssrc = kSsrc1;
+  header.timestamp = rtp_timestamp(time1_ms);
+  header.extension.hasAudioLevel = true;
+  header.extension.audioLevel = 10;
+  PayloadUnion payload_specific = {AudioPayload()};
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+      header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+  auto sources = rtp_receiver_->GetSources();
+  EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
+                           time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  int64_t time2_ms = fake_clock_.TimeInMilliseconds();
+  header.timestamp = rtp_timestamp(time2_ms);
+  header.extension.hasAudioLevel = false;
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+      header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+  sources = rtp_receiver_->GetSources();
+  EXPECT_THAT(sources, UnorderedElementsAre(
+                           RtpSource(time2_ms, kSsrc1, RtpSourceType::SSRC)));
+}

[Taylor Brandstetter, 2017/08/24 20:40:57]

Good tests!

+
 }  // namespace webrtc