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

webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
@@ skipping 237 matching lines @@
             ssrc_sources.begin()->timestamp_ms());
 
   auto csrc_sources = rtp_receiver_impl->csrc_sources_for_testing();
   ASSERT_EQ(1u, csrc_sources.size());
   EXPECT_EQ(kCsrc1, csrc_sources.begin()->source_id());
   EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
   EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
             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