Wire up BitrateAllocation to be sent as RTCP TargetBitrate

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 2417 matching lines @@
   VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
 }
 
 TEST_P(EndToEndTest, VerifyHistogramStatsWithScreenshare) {
   const bool kEnabledRtx = false;
   const bool kEnabledRed = false;
   const bool kScreenshare = true;
   VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
 }
 
 void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {

[danilchap, 2016/11/29 10:39:32]

rename or better fork test since now it is used to track target_bitrate too

[sprang_webrtc, 2016/11/29 12:24:01]

I briefly explored that but it turned out messier than I would like. Prefer not
to spent too much time on this. Renamed the test though and moved out
RtcpXrObserver way down to where it is actually used.

[danilchap, 2016/11/29 13:13:14]

ok

-  static const int kNumRtcpReportPacketsToObserve = 5;
+  static const int kNumRtcpReportPacketsToObserve = 10;
   class RtcpXrObserver : public test::EndToEndTest {
    public:
     explicit RtcpXrObserver(bool enable_rrtr)
         : EndToEndTest(kDefaultTimeoutMs),
           enable_rrtr_(enable_rrtr),
           sent_rtcp_sr_(0),
           sent_rtcp_rr_(0),
           sent_rtcp_rrtr_(0),
+          sent_rtcp_target_bitrate_(false),
           sent_rtcp_dlrr_(0) {}
 
    private:
     // Receive stream should send RR packets (and RRTR packets if enabled).
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));
 
       sent_rtcp_rr_ += parser.receiver_report()->num_packets();
       EXPECT_EQ(0, parser.sender_report()->num_packets());
       EXPECT_GE(1, parser.xr()->num_packets());
       if (parser.xr()->num_packets() > 0) {
         if (parser.xr()->rrtr())
           ++sent_rtcp_rrtr_;
         EXPECT_FALSE(parser.xr()->dlrr());
       }
 
       return SEND_PACKET;
     }
     // Send stream should send SR packets (and DLRR packets if enabled).
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));
 
       sent_rtcp_sr_ += parser.sender_report()->num_packets();
-      EXPECT_GE(1, parser.xr()->num_packets());
+      EXPECT_LE(parser.xr()->num_packets(), 2);  // dlrr and/or target bitrate.
       if (parser.xr()->num_packets() > 0) {
         EXPECT_FALSE(parser.xr()->rrtr());
         if (parser.xr()->dlrr())
           ++sent_rtcp_dlrr_;
+        if (parser.xr()->target_bitrate())
+          sent_rtcp_target_bitrate_ = true;
       }
 
       if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
-          sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
+          sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
+          sent_rtcp_target_bitrate_) {
         if (enable_rrtr_) {
-          EXPECT_LT(0, sent_rtcp_rrtr_);
-          EXPECT_LT(0, sent_rtcp_dlrr_);
+          EXPECT_GT(sent_rtcp_rrtr_, 0);

[danilchap, 2016/11/29 10:39:32]

Yep, no need to write expected value first this days. Can now order parameters
of EXPECT_* macros in a nicer way :)

+          EXPECT_GT(sent_rtcp_dlrr_, 0);
         } else {
-          EXPECT_EQ(0, sent_rtcp_rrtr_);
-          EXPECT_EQ(0, sent_rtcp_dlrr_);
+          EXPECT_EQ(sent_rtcp_rrtr_, 0);
+          EXPECT_EQ(sent_rtcp_dlrr_, 0);
         }
         observation_complete_.Set();
       }
       return SEND_PACKET;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
           enable_rrtr_;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for RTCP SR/RR packets to be sent.";
     }
 
     rtc::CriticalSection crit_;
     bool enable_rrtr_;
     int sent_rtcp_sr_;
     int sent_rtcp_rr_ GUARDED_BY(&crit_);
     int sent_rtcp_rrtr_ GUARDED_BY(&crit_);
+    bool sent_rtcp_target_bitrate_ GUARDED_BY(&crit_);
     int sent_rtcp_dlrr_;
   } test(enable_rrtr);
 
   RunBaseTest(&test);
 }
 
 void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
                                      bool send_single_ssrc_first) {
   class SendsSetSsrcs : public test::EndToEndTest {
    public:
@@ skipping 1516 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc