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..7a3bd449e776a1ad8817e924a8ee3da37d5a8c79 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
@@ -255,4 +255,54 @@ 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.
+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(

[danilchap, 2017/08/15 08:18:29]

ASSERT_THAT
otherwise next line might crash if this expectation doesn't hold because sources
is empty

[Zach Stein, 2017/08/15 21:44:59]

The following line is actually redundant with this EXPECT_THAT now (I wrote the
following before I added the RtpSource constructor and operator== and forgot to
delete it). I'll just rely on the EXPECT_THAT(..., UnorderedElementsAre(...)).

+                           time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
+  ASSERT_EQ(rtc::Optional<uint8_t>(10), sources.begin()->audio_level());

[danilchap, 2017/08/15 08:18:29]

there is a compare operator between optional<T> and T, i.e. you can write just
EXPECT_EQ(10, sources.front().audio_level());
(here ASSERT might be EXPECT if you want to continue test should it fail)

[Zach Stein, 2017/08/15 21:44:59]

Acknowledged.

+
+  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)));
+}
+

[danilchap, 2017/08/15 08:18:29]

may be add a test that absent of audio level extensions resets audio level

[Zach Stein, 2017/08/15 21:44:59]

Done.

 }  // namespace webrtc