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 thi
Zach Stein
2017/08/15 21:44:59
The following line is actually redundant with this
|
+ 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> an
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 exten
Zach Stein
2017/08/15 21:44:59
Done.
|
} // namespace webrtc |