Add media related stats (audio level etc.) to unsignaled streams.

webrtc/pc/peerconnection_integrationtest.cc

 /*
  *  Copyright 2012 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 98 matching lines @@
 // attribute from received SDP, simulating a legacy endpoint.
 void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
   for (ContentInfo& content : desc->contents()) {
     MediaContentDescription* media_desc =
         static_cast<MediaContentDescription*>(content.description);
     media_desc->mutable_streams().clear();
   }
   desc->set_msid_supported(false);
 }
 
+int FindFirstMediaStatsIndexByKind(
+    const std::string& kind,
+    const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
+        media_stats_vec) {
+  for (size_t i = 0; i < media_stats_vec.size(); i++) {
+    if (media_stats_vec[i]->kind.ValueToString() == kind) {
+      return i;
+    }
+  }
+  return -1;
+}
+
 class SignalingMessageReceiver {
  public:
   virtual void ReceiveSdpMessage(const std::string& type,
                                  const std::string& msg) = 0;
   virtual void ReceiveIceMessage(const std::string& sdp_mid,
                                  int sdp_mline_index,
                                  const std::string& msg) = 0;
 
  protected:
   SignalingMessageReceiver() {}
@@ skipping 1790 matching lines @@
   // We received a frame, so we should have nonzero "bytes received" stats for
   // the unsignaled stream, if stats are working for it.
   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
       callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
   auto inbound_stream_stats =
       report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
   ASSERT_EQ(1U, inbound_stream_stats.size());
   ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
   ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
-  // TODO(deadbeef): Test that track_id is defined. This is not currently
-  // working since SSRCs are used to match RtpReceivers (and their tracks) with
-  // received stream stats in TrackMediaInfoMap.
+  ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
+}
+
+// Test that we can successfully get the media related stats (audio level
+// etc.) for the unsignaled stream.
+TEST_F(PeerConnectionIntegrationTest,
+       GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioVideoMediaStream();
+  // Remove SSRCs and MSIDs from the received offer SDP.
+  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for one audio frame to be received by the callee.
+  ExpectNewFramesReceivedWithWait(0, 0, 1, 1, kMaxWaitForFramesMs);
+
+  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
+      callee()->NewGetStats();
+  ASSERT_NE(nullptr, report);
+
+  auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
+  auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
+  ASSERT_GE(audio_index, 0);
+  EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
 }
 
 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
   PeerConnectionFactory::Options dtls_10_options;
   dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                       dtls_10_options));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
@@ skipping 960 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
                                   kMaxWaitForFramesMs);
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)