Fix uploading of available send bitrate statistics.

webrtc/pc/statscollector_unittest.cc

 /*
  *  Copyright 2014 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 882 matching lines @@
   EXPECT_CALL(*media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(stats_read),
                       Return(true)));
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
   std::string result = ExtractSsrcStatsValue(reports,
       StatsReport::kStatsValueNameBytesSent);
   EXPECT_EQ(kBytesSentString, result);
 }
 
-// Test that BWE information is reported via stats.
-TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
+// Test that audio BWE information is reported via stats.
+TEST_F(StatsCollectorTest, AudioBandwidthEstimationInfoIsReported) {
   StatsCollectorForTest stats(&pc_);
 
   EXPECT_CALL(session_, GetLocalCertificate(_, _))
+  .WillRepeatedly(Return(false));
+  EXPECT_CALL(session_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
+  .WillRepeatedly(Return(nullptr));
+
+  const char kAudioChannelName[] = "audio";
+
+  InitSessionStats(kAudioChannelName);
+  EXPECT_CALL(session_, GetStats(_)).WillRepeatedly(Invoke(
+                                                           [this](const ChannelNamePairs&) {
+                                                             return std::unique_ptr<SessionStats>(
+                                                                                                  new SessionStats(session_stats_));
+                                                           }));

[stefan-webrtc, 2017/06/19 13:07:05]

Run 'git cl format' to get nice formatting of this code.

[alexnarest1, 2017/06/19 13:55:22]

Done.

+
+  MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();

[stefan-webrtc, 2017/06/19 13:07:05]

I believe this is leaking. Either use "std::unique_ptr<MockVoiceMediaChannel>
media_channel" or "MockVoiceMediaChannel media_channel".

[alexnarest1, 2017/06/19 13:55:22]

It will not leak because BaseChannel will release it
https://cs.corp.google.com/piper///depot/google3/third_party/webrtc/files/sta...

+  cricket::VoiceChannel voice_channel(
+                                      worker_thread_, network_thread_, nullptr, nullptr, media_channel,
+                                      kAudioChannelName, kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
+
+  StatsReports reports;  // returned values.
+  cricket::VoiceSenderInfo voice_sender_info;
+  cricket::VoiceMediaInfo stats_read;
+  // Set up an SSRC just to test that we get both kinds of stats back: SSRC and
+  // BWE.
+  const int64_t kBytesSent = 12345678901234LL;
+  const std::string kBytesSentString("12345678901234");
+
+  AddOutgoingAudioTrackStats();
+  stats.AddStream(stream_);
+
+  // Construct a stats value to read.
+  voice_sender_info.add_ssrc(1234);
+  voice_sender_info.bytes_sent = kBytesSent;
+  stats_read.senders.push_back(voice_sender_info);
+
+  webrtc::Call::Stats call_stats;
+  const int kSendBandwidth = 1234567;
+  const int kRecvBandwidth = 12345678;
+  const int kPacerDelay = 123;
+  call_stats.send_bandwidth_bps = kSendBandwidth;
+  call_stats.recv_bandwidth_bps = kRecvBandwidth;
+  call_stats.pacer_delay_ms = kPacerDelay;
+  EXPECT_CALL(session_, GetCallStats()).WillRepeatedly(Return(call_stats));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(Return(&voice_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*media_channel, GetStats(_))
+  .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
+
+  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
+  stats.GetStats(NULL, &reports);
+  std::string result = ExtractSsrcStatsValue(reports,
+                                             StatsReport::kStatsValueNameBytesSent);
+  EXPECT_EQ(kBytesSentString, result);
+  result = ExtractBweStatsValue(
+                                reports, StatsReport::kStatsValueNameAvailableSendBandwidth);
+  EXPECT_EQ(rtc::ToString(kSendBandwidth), result);
+  result = ExtractBweStatsValue(
+                                reports, StatsReport::kStatsValueNameAvailableReceiveBandwidth);
+  EXPECT_EQ(rtc::ToString(kRecvBandwidth), result);
+  result =
+  ExtractBweStatsValue(reports, StatsReport::kStatsValueNameBucketDelay);
+  EXPECT_EQ(rtc::ToString(kPacerDelay), result);
+}
+
+// Test that video BWE information is reported via stats.
+TEST_F(StatsCollectorTest, VideoBandwidthEstimationInfoIsReported) {
+  StatsCollectorForTest stats(&pc_);
+
+  EXPECT_CALL(session_, GetLocalCertificate(_, _))
       .WillRepeatedly(Return(false));
   EXPECT_CALL(session_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
       .WillRepeatedly(Return(nullptr));
 
   const char kVideoChannelName[] = "video";
 
   InitSessionStats(kVideoChannelName);
   EXPECT_CALL(session_, GetStats(_)).WillRepeatedly(Invoke(
       [this](const ChannelNamePairs&) {
         return std::unique_ptr<SessionStats>(
@@ skipping 1116 matching lines @@
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
   EXPECT_EQ(rtc::ToString(video_receiver_info.frames_decoded),
             ExtractSsrcStatsValue(reports,
                                   StatsReport::kStatsValueNameFramesDecoded));
   EXPECT_EQ(rtc::ToString(*video_receiver_info.qp_sum),
             ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameQpSum));
 }
 
 }  // namespace webrtc