Fix uploading of available send bitrate statistics.

webrtc/pc/statscollector_unittest.cc

Index: webrtc/pc/statscollector_unittest.cc
diff --git a/webrtc/pc/statscollector_unittest.cc b/webrtc/pc/statscollector_unittest.cc
index e9a2c5c9d56deddf7dc48b563ce391b0b688a6a8..165457c9c82493b1e181c4f36958e5c724f969a8 100644
--- a/webrtc/pc/statscollector_unittest.cc
+++ b/webrtc/pc/statscollector_unittest.cc
@@ -900,8 +900,75 @@ TEST_F(StatsCollectorTest, BytesCounterHandles64Bits) {
   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_));
+      }));
+
+  MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
+  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(_, _))