RTCCodecStats added

webrtc/api/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 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.
  */
 
 #include "webrtc/api/rtcstatscollector.h"
 
 #include <memory>
 #include <ostream>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/mediastreamtrack.h"
-#include "webrtc/api/jsepsessiondescription.h"
+#include "webrtc/api/rtpparameters.h"
 #include "webrtc/api/stats/rtcstats_objects.h"
 #include "webrtc/api/stats/rtcstatsreport.h"
 #include "webrtc/api/test/mock_datachannel.h"
 #include "webrtc/api/test/mock_peerconnection.h"
 #include "webrtc/api/test/mock_webrtcsession.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
@@ skipping 14 matching lines @@
 using testing::ReturnRef;
 using testing::SetArgPointee;
 
 namespace webrtc {
 
 // These are used by gtest code, such as if |EXPECT_EQ| fails.
 void PrintTo(const RTCCertificateStats& stats, ::std::ostream* os) {
   *os << stats.ToString();
 }
 
+void PrintTo(const RTCCodecStats& stats, ::std::ostream* os) {
+  *os << stats.ToString();
+}
+
 void PrintTo(const RTCDataChannelStats& stats, ::std::ostream* os) {
   *os << stats.ToString();
 }
 
 void PrintTo(const RTCIceCandidatePairStats& stats, ::std::ostream* os) {
   *os << stats.ToString();
 }
 
 void PrintTo(const RTCLocalIceCandidateStats& stats, ::std::ostream* os) {
   *os << stats.ToString();
@@ skipping 734 matching lines @@
         if (transport_name == "transport")
           return remote_certinfo->certificate->ssl_certificate().GetReference();
         return static_cast<rtc::SSLCertificate*>(nullptr);
       }));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
   ExpectReportContainsCertificateInfo(report, *local_certinfo.get());
   ExpectReportContainsCertificateInfo(report, *remote_certinfo.get());
 }
 
+TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
+  MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
+  cricket::VoiceChannel voice_channel(
+      test_->worker_thread(), test_->network_thread(), test_->media_engine(),
+      voice_media_channel, nullptr, "VoiceContentName", false);
+
+  MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
+  cricket::VideoChannel video_channel(
+      test_->worker_thread(), test_->network_thread(), video_media_channel,
+      nullptr, "VideoContentName", false);

[Taylor Brandstetter, 2016/11/17 22:01:51]

To test the scenario I described where a payload type may be mapped to two
different things (for different RTP sessions), you could create another mock
video channel that uses a different transport.

+
+  // Audio
+  cricket::VoiceMediaInfo voice_media_info;
+
+  RtpCodecParameters inbound_audio_codec;
+  inbound_audio_codec.payload_type = 1;
+  inbound_audio_codec.mime_type = "opus";
+  inbound_audio_codec.clock_rate = 1337;
+  voice_media_info.receive_codecs.insert(
+      std::make_pair(inbound_audio_codec.payload_type, inbound_audio_codec));
+
+  RtpCodecParameters outbound_audio_codec;
+  outbound_audio_codec.payload_type = 2;
+  outbound_audio_codec.mime_type = "isac";
+  outbound_audio_codec.clock_rate = 1338;
+  voice_media_info.send_codecs.insert(
+      std::make_pair(outbound_audio_codec.payload_type, outbound_audio_codec));
+
+  EXPECT_CALL(*voice_media_channel, GetStats(_))
+      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
+
+  // Video
+  cricket::VideoMediaInfo video_media_info;
+
+  RtpCodecParameters inbound_video_codec;
+  inbound_video_codec.payload_type = 3;
+  inbound_video_codec.mime_type = "H264";
+  inbound_video_codec.clock_rate = 1339;
+  video_media_info.receive_codecs.insert(
+      std::make_pair(inbound_video_codec.payload_type, inbound_video_codec));
+
+  RtpCodecParameters outbound_video_codec;
+  outbound_video_codec.payload_type = 4;
+  outbound_video_codec.mime_type = "VP8";
+  outbound_video_codec.clock_rate = 1340;
+  video_media_info.send_codecs.insert(
+      std::make_pair(outbound_video_codec.payload_type, outbound_video_codec));
+
+  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";
+
+  EXPECT_CALL(test_->session(), GetTransportStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
+  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();
+
+  RTCCodecStats expected_inbound_audio_codec(
+      "RTCCodec_InboundAudio_1", report->timestamp_us());
+  expected_inbound_audio_codec.payload_type = 1;
+  expected_inbound_audio_codec.codec = "audio/opus";
+  expected_inbound_audio_codec.clock_rate = 1337;
+
+  RTCCodecStats expected_outbound_audio_codec(
+      "RTCCodec_OutboundAudio_2", report->timestamp_us());
+  expected_outbound_audio_codec.payload_type = 2;
+  expected_outbound_audio_codec.codec = "audio/isac";
+  expected_outbound_audio_codec.clock_rate = 1338;
+
+  RTCCodecStats expected_inbound_video_codec(
+      "RTCCodec_InboundVideo_3", report->timestamp_us());
+  expected_inbound_video_codec.payload_type = 3;
+  expected_inbound_video_codec.codec = "video/H264";
+  expected_inbound_video_codec.clock_rate = 1339;
+
+  RTCCodecStats expected_outbound_video_codec(
+      "RTCCodec_OutboundVideo_4", report->timestamp_us());
+  expected_outbound_video_codec.payload_type = 4;
+  expected_outbound_video_codec.codec = "video/VP8";
+  expected_outbound_video_codec.clock_rate = 1340;
+
+  ASSERT(report->Get(expected_inbound_audio_codec.id()));
+  EXPECT_EQ(expected_inbound_audio_codec,
+            report->Get(expected_inbound_audio_codec.id())->cast_to<
+                  RTCCodecStats>());
+
+  ASSERT(report->Get(expected_outbound_audio_codec.id()));
+  EXPECT_EQ(expected_outbound_audio_codec,
+            report->Get(expected_outbound_audio_codec.id())->cast_to<
+                  RTCCodecStats>());
+
+  ASSERT(report->Get(expected_inbound_video_codec.id()));
+  EXPECT_EQ(expected_inbound_video_codec,
+            report->Get(expected_inbound_video_codec.id())->cast_to<
+                  RTCCodecStats>());
+
+  ASSERT(report->Get(expected_outbound_video_codec.id()));
+  EXPECT_EQ(expected_outbound_video_codec,
+            report->Get(expected_outbound_video_codec.id())->cast_to<
+                  RTCCodecStats>());
+}
+
 TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
   std::unique_ptr<CertificateInfo> audio_local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
           std::vector<std::string>({ "(local) audio" }));
   audio_local_certinfo = CreateFakeCertificateAndInfoFromDers(
       audio_local_certinfo->ders);
   std::unique_ptr<CertificateInfo> audio_remote_certinfo =
       CreateFakeCertificateAndInfoFromDers(
           std::vector<std::string>({ "(remote) audio" }));
   audio_remote_certinfo = CreateFakeCertificateAndInfoFromDers(
@@ skipping 500 matching lines @@
       test_->worker_thread(), test_->network_thread(), test_->media_engine(),
       voice_media_channel, nullptr, "VoiceContentName", false);
 
   cricket::VoiceMediaInfo voice_media_info;
   voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
   voice_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   voice_media_info.receivers[0].local_stats[0].ssrc = 1;
   voice_media_info.receivers[0].packets_rcvd = 2;
   voice_media_info.receivers[0].bytes_rcvd = 3;
+  voice_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42);
   voice_media_info.receivers[0].jitter_ms = 4500;
   voice_media_info.receivers[0].fraction_lost = 5.5f;
   EXPECT_CALL(*voice_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
 
   SessionStats session_stats;
   session_stats.proxy_to_transport["VoiceContentName"] = "TransportName";
   session_stats.transport_stats["TransportName"].transport_name =
       "TransportName";
 
@@ skipping 10 matching lines @@
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCInboundRTPStreamStats expected_audio(
       "RTCInboundRTPAudioStream_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_InboundAudio_42";
   expected_audio.packets_received = 2;
   expected_audio.bytes_received = 3;
   expected_audio.jitter = 4.5;
   expected_audio.fraction_lost = 5.5;
 
   ASSERT(report->Get(expected_audio.id()));
   const RTCInboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   EXPECT_TRUE(report->Get(*expected_audio.transport_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), video_media_channel,
       nullptr, "VideoContentName", false);
 
   cricket::VideoMediaInfo video_media_info;
   video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
   video_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   video_media_info.receivers[0].local_stats[0].ssrc = 1;
   video_media_info.receivers[0].packets_rcvd = 2;
   video_media_info.receivers[0].bytes_rcvd = 3;
   video_media_info.receivers[0].fraction_lost = 4.5f;
+  video_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42);
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
 
   SessionStats session_stats;
   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(), GetTransportStats(_))
       .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
-  RTCInboundRTPStreamStats expected_audio(
+  RTCInboundRTPStreamStats expected_video(
       "RTCInboundRTPVideoStream_1", report->timestamp_us());
-  expected_audio.ssrc = "1";
-  expected_audio.is_remote = false;
-  expected_audio.media_type = "video";
-  expected_audio.transport_id = "RTCTransport_TransportName_" +
+  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_audio.packets_received = 2;
-  expected_audio.bytes_received = 3;
-  expected_audio.fraction_lost = 4.5;
+  expected_video.codec_id = "RTCCodec_InboundVideo_42";

[hta-webrtc, 2016/11/23 07:37:35]

Nit: Since IDs are defined by the spec to be opaque (any unique string is
conformant), the test should check that expected_video.codec_id exists in the
stats record and points to a codec stats object. Otherwise, the test will break
when you change the format of the video codec ID.

If this is hard to do, leave a TODO.

[hbos, 2016/11/23 09:40:59]

Done here and three other places. (in/out audio/video)

+  expected_video.packets_received = 2;
+  expected_video.bytes_received = 3;
+  expected_video.fraction_lost = 4.5;
 
-  ASSERT(report->Get(expected_audio.id()));
+  ASSERT(report->Get(expected_video.id()));
   const RTCInboundRTPStreamStats& audio = report->Get(
-      expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
-  EXPECT_EQ(audio, expected_audio);
+      expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
+  EXPECT_EQ(audio, expected_video);
 
-  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
+  EXPECT_TRUE(report->Get(*expected_video.transport_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), test_->media_engine(),
       voice_media_channel, nullptr, "VoiceContentName", false);
 
   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 = 4500;
+  voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
   EXPECT_CALL(*voice_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
 
   SessionStats session_stats;
   session_stats.proxy_to_transport["VoiceContentName"] = "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(), GetTransportStats(_))
       .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
   EXPECT_CALL(test_->session(), voice_channel())
       .WillRepeatedly(Return(&voice_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";

[hta-webrtc, 2016/11/23 07:37:35]

As above.

[hbos, 2016/11/23 09:40:59]

Done.

   expected_audio.packets_sent = 2;
   expected_audio.bytes_sent = 3;
   expected_audio.round_trip_time = 4.5;
 
   ASSERT(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   EXPECT_TRUE(report->Get(*expected_audio.transport_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), video_media_channel,
       nullptr, "VideoContentName", false);
 
   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 = 7500;
+  video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
 
   SessionStats session_stats;
   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(), GetTransportStats(_))
       .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   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.round_trip_time = 7.5;
 
   ASSERT(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
@@ skipping 114 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc