From 9a394f0649d39737a791d17e73ad2603e3d162fb Mon Sep 17 00:00:00 2001
From: hbos <hbos@webrtc.org>
Date: Wed, 14 Dec 2016 07:58:22 -0800
Subject: [PATCH] Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement]
 default value.

Due to the Chromium implementation[1] of GetAudioProcesssingStats,
echoReturnLoss and echoReturnLossEnhancement could default to -100 when
no value was available. This should be improved by using rtc::Optional
or AudioProcessorInterface::GetStats being able to return false, but
this requires a bunch of refactoring.

In the meantime we "blacklist" the value -100 which is a nonsense value
anyway. In that case echoReturnLoss[Enhancement] is correctly left
undefined.

[1] https://cs.chromium.org/chromium/src/content/renderer/media/media_stream_audio_processor_options.cc?sq=package:chromium&dr=C&rcl=1481530670&l=461

BUG=chromium:669877

Review-Url: https://codereview.webrtc.org/2573443002
Cr-Commit-Position: refs/heads/master@{#15611}
---
 webrtc/api/rtcstatscollector.cc          | 12 ++++---
 webrtc/api/rtcstatscollector_unittest.cc | 40 ++++++++++++++++++++++++
 2 files changed, 48 insertions(+), 4 deletions(-)

diff --git a/webrtc/api/rtcstatscollector.cc b/webrtc/api/rtcstatscollector.cc
index 9998d9aafe..5289c4820b 100644
--- a/webrtc/api/rtcstatscollector.cc
+++ b/webrtc/api/rtcstatscollector.cc
@@ -317,10 +317,14 @@ void ProduceMediaStreamAndTrackStats(
       if (audio_track->GetAudioProcessor()) {
         AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
         audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats);
-        audio_track_stats->echo_return_loss = static_cast<double>(
-            audio_processor_stats.echo_return_loss);
-        audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
-            audio_processor_stats.echo_return_loss_enhancement);
+        if (audio_processor_stats.echo_return_loss != -100) {
+          audio_track_stats->echo_return_loss = static_cast<double>(
+              audio_processor_stats.echo_return_loss);
+        }
+        if (audio_processor_stats.echo_return_loss_enhancement != -100) {
+          audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
+              audio_processor_stats.echo_return_loss_enhancement);
+        }
       }
       report->AddStats(std::move(audio_track_stats));
     }
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index 278d945378..6d0b56fc0f 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -1211,6 +1211,46 @@ TEST_F(RTCStatsCollectorTest,
                 RTCMediaStreamTrackStats>());
 }
 
+TEST_F(RTCStatsCollectorTest,
+       CollectRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio_Defaults) {
+  rtc::scoped_refptr<StreamCollection> local_streams =
+      StreamCollection::Create();
+  EXPECT_CALL(test_->pc(), local_streams())
+      .WillRepeatedly(Return(local_streams));
+
+  rtc::scoped_refptr<MediaStream> local_stream =
+      MediaStream::Create("LocalStreamLabel");
+  local_streams->AddStream(local_stream);
+
+  // Local audio track
+  AudioProcessorInterface::AudioProcessorStats local_audio_processor_stats;
+  local_audio_processor_stats.echo_return_loss = -100;
+  local_audio_processor_stats.echo_return_loss_enhancement = -100;
+  rtc::scoped_refptr<FakeAudioTrackForStats> local_audio_track =
+      FakeAudioTrackForStats::Create(
+          "LocalAudioTrackID",
+          MediaStreamTrackInterface::TrackState::kEnded,
+          32767,
+          new FakeAudioProcessorForStats(local_audio_processor_stats));
+  local_stream->AddTrack(local_audio_track);
+
+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+  RTCMediaStreamTrackStats expected_local_audio_track(
+      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us());
+  expected_local_audio_track.track_identifier = local_audio_track->id();
+  expected_local_audio_track.remote_source = false;
+  expected_local_audio_track.ended = true;
+  expected_local_audio_track.detached = false;
+  expected_local_audio_track.audio_level = 1.0;
+  // Should be undefined: |expected_local_audio_track.echo_return_loss| and
+  // |expected_local_audio_track.echo_return_loss_enhancement|.
+  EXPECT_TRUE(report->Get(expected_local_audio_track.id()));
+  EXPECT_EQ(expected_local_audio_track,
+            report->Get(expected_local_audio_track.id())->cast_to<
+                RTCMediaStreamTrackStats>());
+}
+
 TEST_F(RTCStatsCollectorTest,
        CollectRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video) {
   rtc::scoped_refptr<StreamCollection> local_streams =