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 1911 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.

[Taylor Brandstetter, 2017/05/17 01:30:04]

Can this TODO be fixed in this CL?

 }
 
+// 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>();
+  ASSERT_GT(media_stats.size(), 0);
+  EXPECT_TRUE(media_stats[0]->audio_level.is_defined());

[Taylor Brandstetter, 2017/05/17 01:30:04]

This relies on media_stats[0] being the audio track and not the video track,
which I don't think is guaranteed anywhere. Would be better to explicitly find
the one with a "kind"
(https://cs.chromium.org/chromium/src/third_party/webrtc/api/stats/rtcstats_ob...)
of "audio".

+}
+
 // 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
   // direction.
   caller()->AddAudioVideoMediaStream();
@@ skipping 958 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)