Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ )
Reason for revert:
Breaks downstream build.
Original issue's description:
> RTCInboundRTPStreamStats.qpSum collected.
>
> This was previously only collected for local tracks
> (RTCOutboundRTPStreamStats.qpSum).
>
> Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
>
> This CL also improves some testing in rtcstatscollector_unittest.cc.
> Default and non-default values are tested in the same unittests,
> removing the test that was specific to default-values, which was
> otherwise code duplication.
>
> BUG=webrtc:7065
>
> Review-Url: https://codereview.webrtc.org/2675943002
> Cr-Commit-Position: refs/heads/master@{#16477}
> Committed: https://chromium.googlesource.com/external/webrtc/+/cd195bea5e38a61e73b08f82445560d9136cd9da
TBR=sakal@webrtc.org,hta@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7065
Review-Url: https://codereview.webrtc.org/2687483002 .
Cr-Original-Commit-Position: refs/heads/master@{#16479}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: ed02c6d68f31baed3cfb7656062895a4d1cb7e65
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index 624d23e..715da56 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1792,9 +1792,10 @@
expected_audio.fraction_lost = 5.5;
ASSERT_TRUE(report->Get(expected_audio.id()));
- EXPECT_EQ(
- report->Get(expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(),
- expected_audio);
+ const RTCInboundRTPStreamStats& audio = report->Get(
+ expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
+ EXPECT_EQ(audio, expected_audio);
+
EXPECT_TRUE(report->Get(*expected_audio.track_id));
EXPECT_TRUE(report->Get(*expected_audio.transport_id));
EXPECT_TRUE(report->Get(*expected_audio.codec_id));
@@ -1825,7 +1826,6 @@
video_media_info.receivers[0].plis_sent = 6;
video_media_info.receivers[0].nacks_sent = 7;
video_media_info.receivers[0].frames_decoded = 8;
- video_media_info.receivers[0].qp_sum = rtc::Optional<uint64_t>();
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
@@ -1876,29 +1876,15 @@
expected_video.packets_lost = 42;
expected_video.fraction_lost = 4.5;
expected_video.frames_decoded = 8;
- // |expected_video.qp_sum| should be undefined.
ASSERT_TRUE(report->Get(expected_video.id()));
- EXPECT_EQ(
- report->Get(expected_video.id())->cast_to<RTCInboundRTPStreamStats>(),
- expected_video);
+ const RTCInboundRTPStreamStats& video = report->Get(
+ expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
+ EXPECT_EQ(video, expected_video);
- // Set previously undefined values and "GetStats" again.
- video_media_info.receivers[0].qp_sum = rtc::Optional<uint64_t>(9);
- expected_video.qp_sum = 9;
-
- EXPECT_CALL(*video_media_channel, GetStats(_))
- .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
- collector_->ClearCachedStatsReport();
- report = GetStatsReport();
-
- ASSERT_TRUE(report->Get(expected_video.id()));
- EXPECT_EQ(
- report->Get(expected_video.id())->cast_to<RTCInboundRTPStreamStats>(),
- expected_video);
EXPECT_TRUE(report->Get(*expected_video.track_id));
EXPECT_TRUE(report->Get(*expected_video.transport_id));
- EXPECT_TRUE(report->Get(*expected_video.codec_id));
+ EXPECT_TRUE(report->Get(*video.codec_id));
}
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
@@ -1918,7 +1904,7 @@
voice_media_info.senders[0].local_stats[0].ssrc = 1;
voice_media_info.senders[0].packets_sent = 2;
voice_media_info.senders[0].bytes_sent = 3;
- voice_media_info.senders[0].rtt_ms = -1;
+ voice_media_info.senders[0].rtt_ms = 4500;
voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
RtpCodecParameters codec_parameters;
@@ -1964,26 +1950,13 @@
expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
expected_audio.packets_sent = 2;
expected_audio.bytes_sent = 3;
- // |expected_audio.round_trip_time| should be undefined.
-
- ASSERT_TRUE(report->Get(expected_audio.id()));
- EXPECT_EQ(
- report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
- expected_audio);
-
- // Set previously undefined values and "GetStats" again.
- voice_media_info.senders[0].rtt_ms = 4500;
expected_audio.round_trip_time = 4.5;
- EXPECT_CALL(*voice_media_channel, GetStats(_))
- .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
- collector_->ClearCachedStatsReport();
- report = GetStatsReport();
-
ASSERT_TRUE(report->Get(expected_audio.id()));
- EXPECT_EQ(
- report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
- expected_audio);
+ const RTCOutboundRTPStreamStats& audio = report->Get(
+ expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
+ EXPECT_EQ(audio, expected_audio);
+
EXPECT_TRUE(report->Get(*expected_audio.track_id));
EXPECT_TRUE(report->Get(*expected_audio.transport_id));
EXPECT_TRUE(report->Get(*expected_audio.codec_id));
@@ -2009,10 +1982,10 @@
video_media_info.senders[0].nacks_rcvd = 4;
video_media_info.senders[0].packets_sent = 5;
video_media_info.senders[0].bytes_sent = 6;
- video_media_info.senders[0].rtt_ms = -1;
+ video_media_info.senders[0].rtt_ms = 7500;
video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
video_media_info.senders[0].frames_encoded = 8;
- video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>();
+ video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>(16);
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
@@ -2060,35 +2033,122 @@
expected_video.nack_count = 4;
expected_video.packets_sent = 5;
expected_video.bytes_sent = 6;
- expected_video.frames_encoded = 8;
- // |expected_video.round_trip_time| and |expected_video.qp_sum| should be
- // undefined.
-
- ASSERT_TRUE(report->Get(expected_video.id()));
- EXPECT_EQ(
- report->Get(expected_video.id())->cast_to<RTCOutboundRTPStreamStats>(),
- expected_video);
-
- // Set previously undefined values and "GetStats" again.
- video_media_info.senders[0].rtt_ms = 7500;
- video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>(9);
expected_video.round_trip_time = 7.5;
- expected_video.qp_sum = 9;
-
- EXPECT_CALL(*video_media_channel, GetStats(_))
- .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
- collector_->ClearCachedStatsReport();
- report = GetStatsReport();
+ expected_video.frames_encoded = 8;
+ expected_video.qp_sum = 16;
ASSERT_TRUE(report->Get(expected_video.id()));
- EXPECT_EQ(
- report->Get(expected_video.id())->cast_to<RTCOutboundRTPStreamStats>(),
- expected_video);
+ const RTCOutboundRTPStreamStats& video = report->Get(
+ expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
+ EXPECT_EQ(video, expected_video);
+
EXPECT_TRUE(report->Get(*expected_video.track_id));
EXPECT_TRUE(report->Get(*expected_video.transport_id));
EXPECT_TRUE(report->Get(*expected_video.codec_id));
}
+TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
+ MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
+ cricket::VoiceChannel voice_channel(
+ test_->worker_thread(), test_->network_thread(),
+ test_->signaling_thread(), test_->media_engine(), voice_media_channel,
+ "VoiceContentName", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
+ MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
+ cricket::VideoChannel video_channel(
+ test_->worker_thread(), test_->network_thread(),
+ test_->signaling_thread(), video_media_channel, "VideoContentName",
+ kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
+
+ cricket::VoiceMediaInfo voice_media_info;
+ voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
+ voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
+ voice_media_info.senders[0].local_stats[0].ssrc = 1;
+ voice_media_info.senders[0].packets_sent = 2;
+ voice_media_info.senders[0].bytes_sent = 3;
+ voice_media_info.senders[0].rtt_ms = -1;
+ voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+
+ cricket::VideoMediaInfo video_media_info;
+ video_media_info.senders.push_back(cricket::VideoSenderInfo());
+ video_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
+ video_media_info.senders[0].local_stats[0].ssrc = 1;
+ video_media_info.senders[0].firs_rcvd = 2;
+ video_media_info.senders[0].plis_rcvd = 3;
+ video_media_info.senders[0].nacks_rcvd = 4;
+ video_media_info.senders[0].packets_sent = 5;
+ video_media_info.senders[0].bytes_sent = 6;
+ video_media_info.senders[0].rtt_ms = -1;
+ video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+ video_media_info.senders[0].frames_encoded = 7;
+
+ EXPECT_CALL(*voice_media_channel, GetStats(_))
+ .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
+ EXPECT_CALL(*video_media_channel, GetStats(_))
+ .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
+
+ SessionStats session_stats;
+ session_stats.proxy_to_transport["VoiceContentName"] = "TransportName";
+ session_stats.proxy_to_transport["VideoContentName"] = "TransportName";
+ session_stats.transport_stats["TransportName"].transport_name =
+ "TransportName";
+
+ // Make sure the associated |RTCTransportStats| is created.
+ cricket::TransportChannelStats channel_stats;
+ channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
+ session_stats.transport_stats["TransportName"].channel_stats.push_back(
+ channel_stats);
+
+ EXPECT_CALL(test_->session(), GetStats(_)).WillRepeatedly(Invoke(
+ [&session_stats](const ChannelNamePairs&) {
+ return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
+ }));
+ EXPECT_CALL(test_->session(), voice_channel())
+ .WillRepeatedly(Return(&voice_channel));
+ EXPECT_CALL(test_->session(), video_channel())
+ .WillRepeatedly(Return(&video_channel));
+
+ rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+ RTCOutboundRTPStreamStats expected_audio(
+ "RTCOutboundRTPAudioStream_1", report->timestamp_us());
+ expected_audio.ssrc = 1;
+ expected_audio.is_remote = false;
+ expected_audio.media_type = "audio";
+ expected_audio.transport_id = "RTCTransport_TransportName_" +
+ rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
+ expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
+ expected_audio.packets_sent = 2;
+ expected_audio.bytes_sent = 3;
+ // |expected_audio.round_trip_time| should be undefined.
+
+ ASSERT_TRUE(report->Get(expected_audio.id()));
+ const RTCOutboundRTPStreamStats& audio = report->Get(
+ expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
+ EXPECT_EQ(audio, expected_audio);
+
+ RTCOutboundRTPStreamStats expected_video(
+ "RTCOutboundRTPVideoStream_1", report->timestamp_us());
+ expected_video.ssrc = 1;
+ expected_video.is_remote = false;
+ expected_video.media_type = "video";
+ expected_video.transport_id = "RTCTransport_TransportName_" +
+ rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
+ expected_video.codec_id = "RTCCodec_OutboundVideo_42";
+ expected_video.fir_count = 2;
+ expected_video.pli_count = 3;
+ expected_video.nack_count = 4;
+ expected_video.packets_sent = 5;
+ expected_video.bytes_sent = 6;
+ expected_video.frames_encoded = 7;
+ // |expected_video.round_trip_time| should be undefined.
+ // |expected_video.qp_sum| should be undefined.
+
+ ASSERT_TRUE(report->Get(expected_video.id()));
+ const RTCOutboundRTPStreamStats& video = report->Get(
+ expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
+ EXPECT_EQ(video, expected_video);
+}
+
TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
std::unique_ptr<cricket::Candidate> rtp_local_candidate = CreateFakeCandidate(
"42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42);
